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<pre>Network Working Group                                       J. Rosenberg
Request for Comments: 3261                                   dynamicsoft
Obsoletes: <a href="./rfc2543">2543</a>                                           H. Schulzrinne
Category: Standards Track                                    Columbia U.
                                                            G. Camarillo
                                                                Ericsson
                                                             A. Johnston
                                                                WorldCom
                                                             J. Peterson
                                                                 Neustar
                                                               R. Sparks
                                                             dynamicsoft
                                                              M. Handley
                                                                    ICIR
                                                             E. Schooler
                                                                    AT&amp;T
                                                               June 2002

                    <span class="h1">SIP: Session Initiation Protocol</span>

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

Abstract

   This document describes Session Initiation Protocol (SIP), an
   application-layer control (signaling) protocol for creating,
   modifying, and terminating sessions with one or more participants.
   These sessions include Internet telephone calls, multimedia
   distribution, and multimedia conferences.

   SIP invitations used to create sessions carry session descriptions
   that allow participants to agree on a set of compatible media types.
   SIP makes use of elements called proxy servers to help route requests
   to the user's current location, authenticate and authorize users for
   services, implement provider call-routing policies, and provide
   features to users.  SIP also provides a registration function that
   allows users to upload their current locations for use by proxy
   servers.  SIP runs on top of several different transport protocols.



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 1]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-2" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Table of Contents

   <a href="#section-1">1</a>          Introduction ........................................    <a href="#page-8">8</a>
   <a href="#section-2">2</a>          Overview of SIP Functionality .......................    <a href="#page-9">9</a>
   <a href="#section-3">3</a>          Terminology .........................................   <a href="#page-10">10</a>
   <a href="#section-4">4</a>          Overview of Operation ...............................   <a href="#page-10">10</a>
   <a href="#section-5">5</a>          Structure of the Protocol ...........................   <a href="#page-18">18</a>
   <a href="#section-6">6</a>          Definitions .........................................   <a href="#page-20">20</a>
   <a href="#section-7">7</a>          SIP Messages ........................................   <a href="#page-26">26</a>
   <a href="#section-7.1">7.1</a>        Requests ............................................   <a href="#page-27">27</a>
   <a href="#section-7.2">7.2</a>        Responses ...........................................   <a href="#page-28">28</a>
   <a href="#section-7.3">7.3</a>        Header Fields .......................................   <a href="#page-29">29</a>
   <a href="#section-7.3.1">7.3.1</a>      Header Field Format .................................   <a href="#page-30">30</a>
   <a href="#section-7.3.2">7.3.2</a>      Header Field Classification .........................   <a href="#page-32">32</a>
   <a href="#section-7.3.3">7.3.3</a>      Compact Form ........................................   <a href="#page-32">32</a>
   <a href="#section-7.4">7.4</a>        Bodies ..............................................   <a href="#page-33">33</a>
   <a href="#section-7.4.1">7.4.1</a>      Message Body Type ...................................   <a href="#page-33">33</a>
   <a href="#section-7.4.2">7.4.2</a>      Message Body Length .................................   <a href="#page-33">33</a>
   <a href="#section-7.5">7.5</a>        Framing SIP Messages ................................   <a href="#page-34">34</a>
   <a href="#section-8">8</a>          General User Agent Behavior .........................   <a href="#page-34">34</a>
   <a href="#section-8.1">8.1</a>        UAC Behavior ........................................   <a href="#page-35">35</a>
   <a href="#section-8.1.1">8.1.1</a>      Generating the Request ..............................   <a href="#page-35">35</a>
   <a href="#section-8.1.1.1">8.1.1.1</a>    Request-URI .........................................   <a href="#page-35">35</a>
   <a href="#section-8.1.1.2">8.1.1.2</a>    To ..................................................   <a href="#page-36">36</a>
   <a href="#section-8.1.1.3">8.1.1.3</a>    From ................................................   <a href="#page-37">37</a>
   <a href="#section-8.1.1.4">8.1.1.4</a>    Call-ID .............................................   <a href="#page-37">37</a>
   <a href="#section-8.1.1.5">8.1.1.5</a>    CSeq ................................................   <a href="#page-38">38</a>
   <a href="#section-8.1.1.6">8.1.1.6</a>    Max-Forwards ........................................   <a href="#page-38">38</a>
   <a href="#section-8.1.1.7">8.1.1.7</a>    Via .................................................   <a href="#page-39">39</a>
   <a href="#section-8.1.1.8">8.1.1.8</a>    Contact .............................................   <a href="#page-40">40</a>
   <a href="#section-8.1.1.9">8.1.1.9</a>    Supported and Require ...............................   <a href="#page-40">40</a>
   <a href="#section-8.1.1.10">8.1.1.10</a>   Additional Message Components .......................   <a href="#page-41">41</a>
   <a href="#section-8.1.2">8.1.2</a>      Sending the Request .................................   <a href="#page-41">41</a>
   <a href="#section-8.1.3">8.1.3</a>      Processing Responses ................................   <a href="#page-42">42</a>
   <a href="#section-8.1.3.1">8.1.3.1</a>    Transaction Layer Errors ............................   <a href="#page-42">42</a>
   <a href="#section-8.1.3.2">8.1.3.2</a>    Unrecognized Responses ..............................   <a href="#page-42">42</a>
   <a href="#section-8.1.3.3">8.1.3.3</a>    Vias ................................................   <a href="#page-43">43</a>
   <a href="#section-8.1.3.4">8.1.3.4</a>    Processing 3xx Responses ............................   <a href="#page-43">43</a>
   <a href="#section-8.1.3.5">8.1.3.5</a>    Processing 4xx Responses ............................   <a href="#page-45">45</a>
   <a href="#section-8.2">8.2</a>        UAS Behavior ........................................   <a href="#page-46">46</a>
   <a href="#section-8.2.1">8.2.1</a>      Method Inspection ...................................   <a href="#page-46">46</a>
   <a href="#section-8.2.2">8.2.2</a>      Header Inspection ...................................   <a href="#page-46">46</a>
   <a href="#section-8.2.2.1">8.2.2.1</a>    To and Request-URI ..................................   <a href="#page-46">46</a>
   <a href="#section-8.2.2.2">8.2.2.2</a>    Merged Requests .....................................   <a href="#page-47">47</a>
   <a href="#section-8.2.2.3">8.2.2.3</a>    Require .............................................   <a href="#page-47">47</a>
   <a href="#section-8.2.3">8.2.3</a>      Content Processing ..................................   <a href="#page-48">48</a>
   <a href="#section-8.2.4">8.2.4</a>      Applying Extensions .................................   <a href="#page-49">49</a>
   <a href="#section-8.2.5">8.2.5</a>      Processing the Request ..............................   <a href="#page-49">49</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 2]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-3" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-8.2.6">8.2.6</a>      Generating the Response .............................   <a href="#page-49">49</a>
   <a href="#section-8.2.6.1">8.2.6.1</a>    Sending a Provisional Response ......................   <a href="#page-49">49</a>
   <a href="#section-8.2.6.2">8.2.6.2</a>    Headers and Tags ....................................   <a href="#page-50">50</a>
   <a href="#section-8.2.7">8.2.7</a>      Stateless UAS Behavior ..............................   <a href="#page-50">50</a>
   <a href="#section-8.3">8.3</a>        Redirect Servers ....................................   <a href="#page-51">51</a>
   <a href="#section-9">9</a>          Canceling a Request .................................   <a href="#page-53">53</a>
   <a href="#section-9.1">9.1</a>        Client Behavior .....................................   <a href="#page-53">53</a>
   <a href="#section-9.2">9.2</a>        Server Behavior .....................................   <a href="#page-55">55</a>
   <a href="#section-10">10</a>         Registrations .......................................   <a href="#page-56">56</a>
   <a href="#section-10.1">10.1</a>       Overview ............................................   <a href="#page-56">56</a>
   <a href="#section-10.2">10.2</a>       Constructing the REGISTER Request ...................   <a href="#page-57">57</a>
   <a href="#section-10.2.1">10.2.1</a>     Adding Bindings .....................................   <a href="#page-59">59</a>
   10.2.1.1   Setting the Expiration Interval of Contact Addresses    60
   <a href="#section-10.2.1.2">10.2.1.2</a>   Preferences among Contact Addresses .................   <a href="#page-61">61</a>
   <a href="#section-10.2.2">10.2.2</a>     Removing Bindings ...................................   <a href="#page-61">61</a>
   <a href="#section-10.2.3">10.2.3</a>     Fetching Bindings ...................................   <a href="#page-61">61</a>
   <a href="#section-10.2.4">10.2.4</a>     Refreshing Bindings .................................   <a href="#page-61">61</a>
   <a href="#section-10.2.5">10.2.5</a>     Setting the Internal Clock ..........................   <a href="#page-62">62</a>
   <a href="#section-10.2.6">10.2.6</a>     Discovering a Registrar .............................   <a href="#page-62">62</a>
   <a href="#section-10.2.7">10.2.7</a>     Transmitting a Request ..............................   <a href="#page-62">62</a>
   <a href="#section-10.2.8">10.2.8</a>     Error Responses .....................................   <a href="#page-63">63</a>
   <a href="#section-10.3">10.3</a>       Processing REGISTER Requests ........................   <a href="#page-63">63</a>
   <a href="#section-11">11</a>         Querying for Capabilities ...........................   <a href="#page-66">66</a>
   <a href="#section-11.1">11.1</a>       Construction of OPTIONS Request .....................   <a href="#page-67">67</a>
   <a href="#section-11.2">11.2</a>       Processing of OPTIONS Request .......................   <a href="#page-68">68</a>
   <a href="#section-12">12</a>         Dialogs .............................................   <a href="#page-69">69</a>
   <a href="#section-12.1">12.1</a>       Creation of a Dialog ................................   <a href="#page-70">70</a>
   <a href="#section-12.1.1">12.1.1</a>     UAS behavior ........................................   <a href="#page-70">70</a>
   <a href="#section-12.1.2">12.1.2</a>     UAC Behavior ........................................   <a href="#page-71">71</a>
   <a href="#section-12.2">12.2</a>       Requests within a Dialog ............................   <a href="#page-72">72</a>
   <a href="#section-12.2.1">12.2.1</a>     UAC Behavior ........................................   <a href="#page-73">73</a>
   <a href="#section-12.2.1.1">12.2.1.1</a>   Generating the Request ..............................   <a href="#page-73">73</a>
   <a href="#section-12.2.1.2">12.2.1.2</a>   Processing the Responses ............................   <a href="#page-75">75</a>
   <a href="#section-12.2.2">12.2.2</a>     UAS Behavior ........................................   <a href="#page-76">76</a>
   <a href="#section-12.3">12.3</a>       Termination of a Dialog .............................   <a href="#page-77">77</a>
   <a href="#section-13">13</a>         Initiating a Session ................................   <a href="#page-77">77</a>
   <a href="#section-13.1">13.1</a>       Overview ............................................   <a href="#page-77">77</a>
   <a href="#section-13.2">13.2</a>       UAC Processing ......................................   <a href="#page-78">78</a>
   <a href="#section-13.2.1">13.2.1</a>     Creating the Initial INVITE .........................   <a href="#page-78">78</a>
   <a href="#section-13.2.2">13.2.2</a>     Processing INVITE Responses .........................   <a href="#page-81">81</a>
   <a href="#section-13.2.2.1">13.2.2.1</a>   1xx Responses .......................................   <a href="#page-81">81</a>
   <a href="#section-13.2.2.2">13.2.2.2</a>   3xx Responses .......................................   <a href="#page-81">81</a>
   <a href="#section-13.2.2.3">13.2.2.3</a>   4xx, 5xx and 6xx Responses ..........................   <a href="#page-81">81</a>
   <a href="#section-13.2.2.4">13.2.2.4</a>   2xx Responses .......................................   <a href="#page-82">82</a>
   <a href="#section-13.3">13.3</a>       UAS Processing ......................................   <a href="#page-83">83</a>
   <a href="#section-13.3.1">13.3.1</a>     Processing of the INVITE ............................   <a href="#page-83">83</a>
   <a href="#section-13.3.1.1">13.3.1.1</a>   Progress ............................................   <a href="#page-84">84</a>
   <a href="#section-13.3.1.2">13.3.1.2</a>   The INVITE is Redirected ............................   <a href="#page-84">84</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 3]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-4" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-13.3.1.3">13.3.1.3</a>   The INVITE is Rejected ..............................   <a href="#page-85">85</a>
   <a href="#section-13.3.1.4">13.3.1.4</a>   The INVITE is Accepted ..............................   <a href="#page-85">85</a>
   <a href="#section-14">14</a>         Modifying an Existing Session .......................   <a href="#page-86">86</a>
   <a href="#section-14.1">14.1</a>       UAC Behavior ........................................   <a href="#page-86">86</a>
   <a href="#section-14.2">14.2</a>       UAS Behavior ........................................   <a href="#page-88">88</a>
   <a href="#section-15">15</a>         Terminating a Session ...............................   <a href="#page-89">89</a>
   <a href="#section-15.1">15.1</a>       Terminating a Session with a BYE Request ............   <a href="#page-90">90</a>
   <a href="#section-15.1.1">15.1.1</a>     UAC Behavior ........................................   <a href="#page-90">90</a>
   <a href="#section-15.1.2">15.1.2</a>     UAS Behavior ........................................   <a href="#page-91">91</a>
   <a href="#section-16">16</a>         Proxy Behavior ......................................   <a href="#page-91">91</a>
   <a href="#section-16.1">16.1</a>       Overview ............................................   <a href="#page-91">91</a>
   <a href="#section-16.2">16.2</a>       Stateful Proxy ......................................   <a href="#page-92">92</a>
   <a href="#section-16.3">16.3</a>       Request Validation ..................................   <a href="#page-94">94</a>
   <a href="#section-16.4">16.4</a>       Route Information Preprocessing .....................   <a href="#page-96">96</a>
   <a href="#section-16.5">16.5</a>       Determining Request Targets .........................   <a href="#page-97">97</a>
   <a href="#section-16.6">16.6</a>       Request Forwarding ..................................   <a href="#page-99">99</a>
   <a href="#section-16.7">16.7</a>       Response Processing .................................  <a href="#page-107">107</a>
   <a href="#section-16.8">16.8</a>       Processing Timer C ..................................  <a href="#page-114">114</a>
   <a href="#section-16.9">16.9</a>       Handling Transport Errors ...........................  <a href="#page-115">115</a>
   <a href="#section-16.10">16.10</a>      CANCEL Processing ...................................  <a href="#page-115">115</a>
   <a href="#section-16.11">16.11</a>      Stateless Proxy .....................................  <a href="#page-116">116</a>
   <a href="#section-16.12">16.12</a>      Summary of Proxy Route Processing ...................  <a href="#page-118">118</a>
   <a href="#section-16.12.1">16.12.1</a>    Examples ............................................  <a href="#page-118">118</a>
   <a href="#section-16.12.1.1">16.12.1.1</a>  Basic SIP Trapezoid .................................  <a href="#page-118">118</a>
   <a href="#section-16.12.1.2">16.12.1.2</a>  Traversing a Strict-Routing Proxy ...................  <a href="#page-120">120</a>
   <a href="#section-16.12.1.3">16.12.1.3</a>  Rewriting Record-Route Header Field Values ..........  <a href="#page-121">121</a>
   <a href="#section-17">17</a>         Transactions ........................................  <a href="#page-122">122</a>
   <a href="#section-17.1">17.1</a>       Client Transaction ..................................  <a href="#page-124">124</a>
   <a href="#section-17.1.1">17.1.1</a>     INVITE Client Transaction ...........................  <a href="#page-125">125</a>
   <a href="#section-17.1.1.1">17.1.1.1</a>   Overview of INVITE Transaction ......................  <a href="#page-125">125</a>
   <a href="#section-17.1.1.2">17.1.1.2</a>   Formal Description ..................................  <a href="#page-125">125</a>
   <a href="#section-17.1.1.3">17.1.1.3</a>   Construction of the ACK Request .....................  <a href="#page-129">129</a>
   <a href="#section-17.1.2">17.1.2</a>     Non-INVITE Client Transaction .......................  <a href="#page-130">130</a>
   <a href="#section-17.1.2.1">17.1.2.1</a>   Overview of the non-INVITE Transaction ..............  <a href="#page-130">130</a>
   <a href="#section-17.1.2.2">17.1.2.2</a>   Formal Description ..................................  <a href="#page-131">131</a>
   <a href="#section-17.1.3">17.1.3</a>     Matching Responses to Client Transactions ...........  <a href="#page-132">132</a>
   <a href="#section-17.1.4">17.1.4</a>     Handling Transport Errors ...........................  <a href="#page-133">133</a>
   <a href="#section-17.2">17.2</a>       Server Transaction ..................................  <a href="#page-134">134</a>
   <a href="#section-17.2.1">17.2.1</a>     INVITE Server Transaction ...........................  <a href="#page-134">134</a>
   <a href="#section-17.2.2">17.2.2</a>     Non-INVITE Server Transaction .......................  <a href="#page-137">137</a>
   <a href="#section-17.2.3">17.2.3</a>     Matching Requests to Server Transactions ............  <a href="#page-138">138</a>
   <a href="#section-17.2.4">17.2.4</a>     Handling Transport Errors ...........................  <a href="#page-141">141</a>
   <a href="#section-18">18</a>         Transport ...........................................  <a href="#page-141">141</a>
   <a href="#section-18.1">18.1</a>       Clients .............................................  <a href="#page-142">142</a>
   <a href="#section-18.1.1">18.1.1</a>     Sending Requests ....................................  <a href="#page-142">142</a>
   <a href="#section-18.1.2">18.1.2</a>     Receiving Responses .................................  <a href="#page-144">144</a>
   <a href="#section-18.2">18.2</a>       Servers .............................................  <a href="#page-145">145</a>
   <a href="#section-18.2.1">18.2.1</a>     Receiving Requests ..................................  <a href="#page-145">145</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 4]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-5" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-18.2.2">18.2.2</a>     Sending Responses ...................................  <a href="#page-146">146</a>
   <a href="#section-18.3">18.3</a>       Framing .............................................  <a href="#page-147">147</a>
   <a href="#section-18.4">18.4</a>       Error Handling ......................................  <a href="#page-147">147</a>
   <a href="#section-19">19</a>         Common Message Components ...........................  <a href="#page-147">147</a>
   <a href="#section-19.1">19.1</a>       SIP and SIPS Uniform Resource Indicators ............  <a href="#page-148">148</a>
   <a href="#section-19.1.1">19.1.1</a>     SIP and SIPS URI Components .........................  <a href="#page-148">148</a>
   <a href="#section-19.1.2">19.1.2</a>     Character Escaping Requirements .....................  <a href="#page-152">152</a>
   <a href="#section-19.1.3">19.1.3</a>     Example SIP and SIPS URIs ...........................  <a href="#page-153">153</a>
   <a href="#section-19.1.4">19.1.4</a>     URI Comparison ......................................  <a href="#page-153">153</a>
   <a href="#section-19.1.5">19.1.5</a>     Forming Requests from a URI .........................  <a href="#page-156">156</a>
   <a href="#section-19.1.6">19.1.6</a>     Relating SIP URIs and tel URLs ......................  <a href="#page-157">157</a>
   <a href="#section-19.2">19.2</a>       Option Tags .........................................  <a href="#page-158">158</a>
   <a href="#section-19.3">19.3</a>       Tags ................................................  <a href="#page-159">159</a>
   <a href="#section-20">20</a>         Header Fields .......................................  <a href="#page-159">159</a>
   <a href="#section-20.1">20.1</a>       Accept ..............................................  <a href="#page-161">161</a>
   <a href="#section-20.2">20.2</a>       Accept-Encoding .....................................  <a href="#page-163">163</a>
   <a href="#section-20.3">20.3</a>       Accept-Language .....................................  <a href="#page-164">164</a>
   <a href="#section-20.4">20.4</a>       Alert-Info ..........................................  <a href="#page-164">164</a>
   <a href="#section-20.5">20.5</a>       Allow ...............................................  <a href="#page-165">165</a>
   <a href="#section-20.6">20.6</a>       Authentication-Info .................................  <a href="#page-165">165</a>
   <a href="#section-20.7">20.7</a>       Authorization .......................................  <a href="#page-165">165</a>
   <a href="#section-20.8">20.8</a>       Call-ID .............................................  <a href="#page-166">166</a>
   <a href="#section-20.9">20.9</a>       Call-Info ...........................................  <a href="#page-166">166</a>
   <a href="#section-20.10">20.10</a>      Contact .............................................  <a href="#page-167">167</a>
   <a href="#section-20.11">20.11</a>      Content-Disposition .................................  <a href="#page-168">168</a>
   <a href="#section-20.12">20.12</a>      Content-Encoding ....................................  <a href="#page-169">169</a>
   <a href="#section-20.13">20.13</a>      Content-Language ....................................  <a href="#page-169">169</a>
   <a href="#section-20.14">20.14</a>      Content-Length ......................................  <a href="#page-169">169</a>
   <a href="#section-20.15">20.15</a>      Content-Type ........................................  <a href="#page-170">170</a>
   <a href="#section-20.16">20.16</a>      CSeq ................................................  <a href="#page-170">170</a>
   <a href="#section-20.17">20.17</a>      Date ................................................  <a href="#page-170">170</a>
   <a href="#section-20.18">20.18</a>      Error-Info ..........................................  <a href="#page-171">171</a>
   <a href="#section-20.19">20.19</a>      Expires .............................................  <a href="#page-171">171</a>
   <a href="#section-20.20">20.20</a>      From ................................................  <a href="#page-172">172</a>
   <a href="#section-20.21">20.21</a>      In-Reply-To .........................................  <a href="#page-172">172</a>
   <a href="#section-20.22">20.22</a>      Max-Forwards ........................................  <a href="#page-173">173</a>
   <a href="#section-20.23">20.23</a>      Min-Expires .........................................  <a href="#page-173">173</a>
   <a href="#section-20.24">20.24</a>      MIME-Version ........................................  <a href="#page-173">173</a>
   <a href="#section-20.25">20.25</a>      Organization ........................................  <a href="#page-174">174</a>
   <a href="#section-20.26">20.26</a>      Priority ............................................  <a href="#page-174">174</a>
   <a href="#section-20.27">20.27</a>      Proxy-Authenticate ..................................  <a href="#page-174">174</a>
   <a href="#section-20.28">20.28</a>      Proxy-Authorization .................................  <a href="#page-175">175</a>
   <a href="#section-20.29">20.29</a>      Proxy-Require .......................................  <a href="#page-175">175</a>
   <a href="#section-20.30">20.30</a>      Record-Route ........................................  <a href="#page-175">175</a>
   <a href="#section-20.31">20.31</a>      Reply-To ............................................  <a href="#page-176">176</a>
   <a href="#section-20.32">20.32</a>      Require .............................................  <a href="#page-176">176</a>
   <a href="#section-20.33">20.33</a>      Retry-After .........................................  <a href="#page-176">176</a>
   <a href="#section-20.34">20.34</a>      Route ...............................................  <a href="#page-177">177</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 5]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-6" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-20.35">20.35</a>      Server ..............................................  <a href="#page-177">177</a>
   <a href="#section-20.36">20.36</a>      Subject .............................................  <a href="#page-177">177</a>
   <a href="#section-20.37">20.37</a>      Supported ...........................................  <a href="#page-178">178</a>
   <a href="#section-20.38">20.38</a>      Timestamp ...........................................  <a href="#page-178">178</a>
   <a href="#section-20.39">20.39</a>      To ..................................................  <a href="#page-178">178</a>
   <a href="#section-20.40">20.40</a>      Unsupported .........................................  <a href="#page-179">179</a>
   <a href="#section-20.41">20.41</a>      User-Agent ..........................................  <a href="#page-179">179</a>
   <a href="#section-20.42">20.42</a>      Via .................................................  <a href="#page-179">179</a>
   <a href="#section-20.43">20.43</a>      Warning .............................................  <a href="#page-180">180</a>
   <a href="#section-20.44">20.44</a>      WWW-Authenticate ....................................  <a href="#page-182">182</a>
   <a href="#section-21">21</a>         Response Codes ......................................  <a href="#page-182">182</a>
   <a href="#section-21.1">21.1</a>       Provisional 1xx .....................................  <a href="#page-182">182</a>
   <a href="#section-21.1.1">21.1.1</a>     100 Trying ..........................................  <a href="#page-183">183</a>
   <a href="#section-21.1.2">21.1.2</a>     180 Ringing .........................................  <a href="#page-183">183</a>
   <a href="#section-21.1.3">21.1.3</a>     181 Call Is Being Forwarded .........................  <a href="#page-183">183</a>
   <a href="#section-21.1.4">21.1.4</a>     182 Queued ..........................................  <a href="#page-183">183</a>
   <a href="#section-21.1.5">21.1.5</a>     183 Session Progress ................................  <a href="#page-183">183</a>
   <a href="#section-21.2">21.2</a>       Successful 2xx ......................................  <a href="#page-183">183</a>
   <a href="#section-21.2.1">21.2.1</a>     200 OK ..............................................  <a href="#page-183">183</a>
   <a href="#section-21.3">21.3</a>       Redirection 3xx .....................................  <a href="#page-184">184</a>
   <a href="#section-21.3.1">21.3.1</a>     300 Multiple Choices ................................  <a href="#page-184">184</a>
   <a href="#section-21.3.2">21.3.2</a>     301 Moved Permanently ...............................  <a href="#page-184">184</a>
   <a href="#section-21.3.3">21.3.3</a>     302 Moved Temporarily ...............................  <a href="#page-184">184</a>
   <a href="#section-21.3.4">21.3.4</a>     305 Use Proxy .......................................  <a href="#page-185">185</a>
   <a href="#section-21.3.5">21.3.5</a>     380 Alternative Service .............................  <a href="#page-185">185</a>
   <a href="#section-21.4">21.4</a>       Request Failure 4xx .................................  <a href="#page-185">185</a>
   <a href="#section-21.4.1">21.4.1</a>     400 Bad Request .....................................  <a href="#page-185">185</a>
   <a href="#section-21.4.2">21.4.2</a>     401 Unauthorized ....................................  <a href="#page-185">185</a>
   <a href="#section-21.4.3">21.4.3</a>     402 Payment Required ................................  <a href="#page-186">186</a>
   <a href="#section-21.4.4">21.4.4</a>     403 Forbidden .......................................  <a href="#page-186">186</a>
   <a href="#section-21.4.5">21.4.5</a>     404 Not Found .......................................  <a href="#page-186">186</a>
   <a href="#section-21.4.6">21.4.6</a>     405 Method Not Allowed ..............................  <a href="#page-186">186</a>
   <a href="#section-21.4.7">21.4.7</a>     406 Not Acceptable ..................................  <a href="#page-186">186</a>
   <a href="#section-21.4.8">21.4.8</a>     407 Proxy Authentication Required ...................  <a href="#page-186">186</a>
   <a href="#section-21.4.9">21.4.9</a>     408 Request Timeout .................................  <a href="#page-186">186</a>
   <a href="#section-21.4.10">21.4.10</a>    410 Gone ............................................  <a href="#page-187">187</a>
   <a href="#section-21.4.11">21.4.11</a>    413 Request Entity Too Large ........................  <a href="#page-187">187</a>
   <a href="#section-21.4.12">21.4.12</a>    414 Request-URI Too Long ............................  <a href="#page-187">187</a>
   <a href="#section-21.4.13">21.4.13</a>    415 Unsupported Media Type ..........................  <a href="#page-187">187</a>
   <a href="#section-21.4.14">21.4.14</a>    416 Unsupported URI Scheme ..........................  <a href="#page-187">187</a>
   <a href="#section-21.4.15">21.4.15</a>    420 Bad Extension ...................................  <a href="#page-187">187</a>
   <a href="#section-21.4.16">21.4.16</a>    421 Extension Required ..............................  <a href="#page-188">188</a>
   <a href="#section-21.4.17">21.4.17</a>    423 Interval Too Brief ..............................  <a href="#page-188">188</a>
   <a href="#section-21.4.18">21.4.18</a>    480 Temporarily Unavailable .........................  <a href="#page-188">188</a>
   <a href="#section-21.4.19">21.4.19</a>    481 Call/Transaction Does Not Exist .................  <a href="#page-188">188</a>
   <a href="#section-21.4.20">21.4.20</a>    482 Loop Detected ...................................  <a href="#page-188">188</a>
   <a href="#section-21.4.21">21.4.21</a>    483 Too Many Hops ...................................  <a href="#page-189">189</a>
   <a href="#section-21.4.22">21.4.22</a>    484 Address Incomplete ..............................  <a href="#page-189">189</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 6]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-7" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-21.4.23">21.4.23</a>    485 Ambiguous .......................................  <a href="#page-189">189</a>
   <a href="#section-21.4.24">21.4.24</a>    486 Busy Here .......................................  <a href="#page-189">189</a>
   <a href="#section-21.4.25">21.4.25</a>    487 Request Terminated ..............................  <a href="#page-190">190</a>
   <a href="#section-21.4.26">21.4.26</a>    488 Not Acceptable Here .............................  <a href="#page-190">190</a>
   <a href="#section-21.4.27">21.4.27</a>    491 Request Pending .................................  <a href="#page-190">190</a>
   <a href="#section-21.4.28">21.4.28</a>    493 Undecipherable ..................................  <a href="#page-190">190</a>
   <a href="#section-21.5">21.5</a>       Server Failure 5xx ..................................  <a href="#page-190">190</a>
   <a href="#section-21.5.1">21.5.1</a>     500 Server Internal Error ...........................  <a href="#page-190">190</a>
   <a href="#section-21.5.2">21.5.2</a>     501 Not Implemented .................................  <a href="#page-191">191</a>
   <a href="#section-21.5.3">21.5.3</a>     502 Bad Gateway .....................................  <a href="#page-191">191</a>
   <a href="#section-21.5.4">21.5.4</a>     503 Service Unavailable .............................  <a href="#page-191">191</a>
   <a href="#section-21.5.5">21.5.5</a>     504 Server Time-out .................................  <a href="#page-191">191</a>
   <a href="#section-21.5.6">21.5.6</a>     505 Version Not Supported ...........................  <a href="#page-192">192</a>
   <a href="#section-21.5.7">21.5.7</a>     513 Message Too Large ...............................  <a href="#page-192">192</a>
   <a href="#section-21.6">21.6</a>       Global Failures 6xx .................................  <a href="#page-192">192</a>
   <a href="#section-21.6.1">21.6.1</a>     600 Busy Everywhere .................................  <a href="#page-192">192</a>
   <a href="#section-21.6.2">21.6.2</a>     603 Decline .........................................  <a href="#page-192">192</a>
   <a href="#section-21.6.3">21.6.3</a>     604 Does Not Exist Anywhere .........................  <a href="#page-192">192</a>
   <a href="#section-21.6.4">21.6.4</a>     606 Not Acceptable ..................................  <a href="#page-192">192</a>
   <a href="#section-22">22</a>         Usage of HTTP Authentication ........................  <a href="#page-193">193</a>
   <a href="#section-22.1">22.1</a>       Framework ...........................................  <a href="#page-193">193</a>
   <a href="#section-22.2">22.2</a>       User-to-User Authentication .........................  <a href="#page-195">195</a>
   <a href="#section-22.3">22.3</a>       Proxy-to-User Authentication ........................  <a href="#page-197">197</a>
   <a href="#section-22.4">22.4</a>       The Digest Authentication Scheme ....................  <a href="#page-199">199</a>
   <a href="#section-23">23</a>         S/MIME ..............................................  <a href="#page-201">201</a>
   <a href="#section-23.1">23.1</a>       S/MIME Certificates .................................  <a href="#page-201">201</a>
   <a href="#section-23.2">23.2</a>       S/MIME Key Exchange .................................  <a href="#page-202">202</a>
   <a href="#section-23.3">23.3</a>       Securing MIME bodies ................................  <a href="#page-205">205</a>
   23.4       SIP Header Privacy and Integrity using S/MIME:
              Tunneling SIP .......................................  <a href="#page-207">207</a>
   23.4.1     Integrity and Confidentiality Properties of SIP
              Headers .............................................  <a href="#page-207">207</a>
   <a href="#section-23.4.1.1">23.4.1.1</a>   Integrity ...........................................  <a href="#page-207">207</a>
   <a href="#section-23.4.1.2">23.4.1.2</a>   Confidentiality .....................................  <a href="#page-208">208</a>
   <a href="#section-23.4.2">23.4.2</a>     Tunneling Integrity and Authentication ..............  <a href="#page-209">209</a>
   <a href="#section-23.4.3">23.4.3</a>     Tunneling Encryption ................................  <a href="#page-211">211</a>
   <a href="#section-24">24</a>         Examples ............................................  <a href="#page-213">213</a>
   <a href="#section-24.1">24.1</a>       Registration ........................................  <a href="#page-213">213</a>
   <a href="#section-24.2">24.2</a>       Session Setup .......................................  <a href="#page-214">214</a>
   <a href="#section-25">25</a>         Augmented BNF for the SIP Protocol ..................  <a href="#page-219">219</a>
   <a href="#section-25.1">25.1</a>       Basic Rules .........................................  <a href="#page-219">219</a>
   26         Security Considerations: Threat Model and Security
              Usage Recommendations ...............................  <a href="#page-232">232</a>
   <a href="#section-26.1">26.1</a>       Attacks and Threat Models ...........................  <a href="#page-233">233</a>
   <a href="#section-26.1.1">26.1.1</a>     Registration Hijacking ..............................  <a href="#page-233">233</a>
   <a href="#section-26.1.2">26.1.2</a>     Impersonating a Server ..............................  <a href="#page-234">234</a>
   <a href="#section-26.1.3">26.1.3</a>     Tampering with Message Bodies .......................  <a href="#page-235">235</a>
   <a href="#section-26.1.4">26.1.4</a>     Tearing Down Sessions ...............................  <a href="#page-235">235</a>



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 7]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-8" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   <a href="#section-26.1.5">26.1.5</a>     Denial of Service and Amplification .................  <a href="#page-236">236</a>
   <a href="#section-26.2">26.2</a>       Security Mechanisms .................................  <a href="#page-237">237</a>
   <a href="#section-26.2.1">26.2.1</a>     Transport and Network Layer Security ................  <a href="#page-238">238</a>
   <a href="#section-26.2.2">26.2.2</a>     SIPS URI Scheme .....................................  <a href="#page-239">239</a>
   <a href="#section-26.2.3">26.2.3</a>     HTTP Authentication .................................  <a href="#page-240">240</a>
   <a href="#section-26.2.4">26.2.4</a>     S/MIME ..............................................  <a href="#page-240">240</a>
   <a href="#section-26.3">26.3</a>       Implementing Security Mechanisms ....................  <a href="#page-241">241</a>
   <a href="#section-26.3.1">26.3.1</a>     Requirements for Implementers of SIP ................  <a href="#page-241">241</a>
   <a href="#section-26.3.2">26.3.2</a>     Security Solutions ..................................  <a href="#page-242">242</a>
   <a href="#section-26.3.2.1">26.3.2.1</a>   Registration ........................................  <a href="#page-242">242</a>
   <a href="#section-26.3.2.2">26.3.2.2</a>   Interdomain Requests ................................  <a href="#page-243">243</a>
   <a href="#section-26.3.2.3">26.3.2.3</a>   Peer-to-Peer Requests ...............................  <a href="#page-245">245</a>
   <a href="#section-26.3.2.4">26.3.2.4</a>   DoS Protection ......................................  <a href="#page-246">246</a>
   <a href="#section-26.4">26.4</a>       Limitations .........................................  <a href="#page-247">247</a>
   <a href="#section-26.4.1">26.4.1</a>     HTTP Digest .........................................  <a href="#page-247">247</a>
   <a href="#section-26.4.2">26.4.2</a>     S/MIME ..............................................  <a href="#page-248">248</a>
   <a href="#section-26.4.3">26.4.3</a>     TLS .................................................  <a href="#page-249">249</a>
   <a href="#section-26.4.4">26.4.4</a>     SIPS URIs ...........................................  <a href="#page-249">249</a>
   <a href="#section-26.5">26.5</a>       Privacy .............................................  <a href="#page-251">251</a>
   <a href="#section-27">27</a>         IANA Considerations .................................  <a href="#page-252">252</a>
   <a href="#section-27.1">27.1</a>       Option Tags .........................................  <a href="#page-252">252</a>
   <a href="#section-27.2">27.2</a>       Warn-Codes ..........................................  <a href="#page-252">252</a>
   <a href="#section-27.3">27.3</a>       Header Field Names ..................................  <a href="#page-253">253</a>
   <a href="#section-27.4">27.4</a>       Method and Response Codes ...........................  <a href="#page-253">253</a>
   27.5       The "message/sip" MIME type.  .......................  <a href="#page-254">254</a>
   <a href="#section-27.6">27.6</a>       New Content-Disposition Parameter Registrations .....  <a href="#page-255">255</a>
   <a href="#section-28">28</a>         Changes From <a href="./rfc2543">RFC 2543</a> ...............................  <a href="#page-255">255</a>
   <a href="#section-28.1">28.1</a>       Major Functional Changes ............................  <a href="#page-255">255</a>
   <a href="#section-28.2">28.2</a>       Minor Functional Changes ............................  <a href="#page-260">260</a>
   <a href="#section-29">29</a>         Normative References ................................  <a href="#page-261">261</a>
   <a href="#section-30">30</a>         Informative References ..............................  <a href="#page-262">262</a>
   <a href="#appendix-A">A</a>          Table of Timer Values ...............................  <a href="#page-265">265</a>
   Acknowledgments ................................................  <a href="#page-266">266</a>
   Authors' Addresses .............................................  <a href="#page-267">267</a>
   Full Copyright Statement .......................................  <a href="#page-269">269</a>

<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a> Introduction</span>

   There are many applications of the Internet that require the creation
   and management of a session, where a session is considered an
   exchange of data between an association of participants.  The
   implementation of these applications is complicated by the practices
   of participants: users may move between endpoints, they may be
   addressable by multiple names, and they may communicate in several
   different media - sometimes simultaneously.  Numerous protocols have
   been authored that carry various forms of real-time multimedia
   session data such as voice, video, or text messages.  The Session
   Initiation Protocol (SIP) works in concert with these protocols by



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 8]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-9" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   enabling Internet endpoints (called user agents) to discover one
   another and to agree on a characterization of a session they would
   like to share.  For locating prospective session participants, and
   for other functions, SIP enables the creation of an infrastructure of
   network hosts (called proxy servers) to which user agents can send
   registrations, invitations to sessions, and other requests.  SIP is
   an agile, general-purpose tool for creating, modifying, and
   terminating sessions that works independently of underlying transport
   protocols and without dependency on the type of session that is being
   established.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a> Overview of SIP Functionality</span>

   SIP is an application-layer control protocol that can establish,
   modify, and terminate multimedia sessions (conferences) such as
   Internet telephony calls.  SIP can also invite participants to
   already existing sessions, such as multicast conferences.  Media can
   be added to (and removed from) an existing session.  SIP
   transparently supports name mapping and redirection services, which
   supports personal mobility [<a href="#ref-27" title="&quot;Emerging mobile and personal communication systems,&quot;">27</a>] - users can maintain a single
   externally visible identifier regardless of their network location.

   SIP supports five facets of establishing and terminating multimedia
   communications:

      User location: determination of the end system to be used for
           communication;

      User availability: determination of the willingness of the called
           party to engage in communications;

      User capabilities: determination of the media and media parameters
           to be used;

      Session setup: "ringing", establishment of session parameters at
           both called and calling party;

      Session management: including transfer and termination of
           sessions, modifying session parameters, and invoking
           services.

   SIP is not a vertically integrated communications system.  SIP is
   rather a component that can be used with other IETF protocols to
   build a complete multimedia architecture.  Typically, these
   architectures will include protocols such as the Real-time Transport
   Protocol (RTP) (<a href="./rfc1889">RFC 1889</a> [<a href="#ref-28" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">28</a>]) for transporting real-time data and
   providing QoS feedback, the Real-Time streaming protocol (RTSP) (<a href="./rfc2326">RFC</a>
   <a href="./rfc2326">2326</a> [<a href="#ref-29" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">29</a>]) for controlling delivery of streaming media, the Media



<span class="grey">Rosenberg, et. al.          Standards Track                     [Page 9]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-10" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Gateway Control Protocol (MEGACO) (<a href="./rfc3015">RFC 3015</a> [<a href="#ref-30" title="&quot;Megaco Protocol Version 1.0&quot;">30</a>]) for controlling
   gateways to the Public Switched Telephone Network (PSTN), and the
   Session Description Protocol (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-1" title="&quot;SDP: Session Description Protocol&quot;">1</a>]) for describing
   multimedia sessions.  Therefore, SIP should be used in conjunction
   with other protocols in order to provide complete services to the
   users.  However, the basic functionality and operation of SIP does
   not depend on any of these protocols.

   SIP does not provide services.  Rather, SIP provides primitives that
   can be used to implement different services.  For example, SIP can
   locate a user and deliver an opaque object to his current location.
   If this primitive is used to deliver a session description written in
   SDP, for instance, the endpoints can agree on the parameters of a
   session.  If the same primitive is used to deliver a photo of the
   caller as well as the session description, a "caller ID" service can
   be easily implemented.  As this example shows, a single primitive is
   typically used to provide several different services.

   SIP does not offer conference control services such as floor control
   or voting and does not prescribe how a conference is to be managed.
   SIP can be used to initiate a session that uses some other conference
   control protocol.  Since SIP messages and the sessions they establish
   can pass through entirely different networks, SIP cannot, and does
   not, provide any kind of network resource reservation capabilities.

   The nature of the services provided make security particularly
   important.  To that end, SIP provides a suite of security services,
   which include denial-of-service prevention, authentication (both user
   to user and proxy to user), integrity protection, and encryption and
   privacy services.

   SIP works with both IPv4 and IPv6.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a> Terminology</span>

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
   RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
   described in <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a> [<a href="#ref-2" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">2</a>] and indicate requirement levels for
   compliant SIP implementations.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a> Overview of Operation</span>

   This section introduces the basic operations of SIP using simple
   examples.  This section is tutorial in nature and does not contain
   any normative statements.





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   The first example shows the basic functions of SIP: location of an
   end point, signal of a desire to communicate, negotiation of session
   parameters to establish the session, and teardown of the session once
   established.

   Figure 1 shows a typical example of a SIP message exchange between
   two users, Alice and Bob.  (Each message is labeled with the letter
   "F" and a number for reference by the text.)  In this example, Alice
   uses a SIP application on her PC (referred to as a softphone) to call
   Bob on his SIP phone over the Internet.  Also shown are two SIP proxy
   servers that act on behalf of Alice and Bob to facilitate the session
   establishment.  This typical arrangement is often referred to as the
   "SIP trapezoid" as shown by the geometric shape of the dotted lines
   in Figure 1.

   Alice "calls" Bob using his SIP identity, a type of Uniform Resource
   Identifier (URI) called a SIP URI. SIP URIs are defined in <a href="#section-19.1">Section</a>
   <a href="#section-19.1">19.1</a>.  It has a similar form to an email address, typically
   containing a username and a host name.  In this case, it is
   sip:bob@biloxi.com, where biloxi.com is the domain of Bob's SIP
   service provider.  Alice has a SIP URI of sip:alice@atlanta.com.
   Alice might have typed in Bob's URI or perhaps clicked on a hyperlink
   or an entry in an address book.  SIP also provides a secure URI,
   called a SIPS URI.  An example would be sips:bob@biloxi.com.  A call
   made to a SIPS URI guarantees that secure, encrypted transport
   (namely TLS) is used to carry all SIP messages from the caller to the
   domain of the callee.  From there, the request is sent securely to
   the callee, but with security mechanisms that depend on the policy of
   the domain of the callee.

   SIP is based on an HTTP-like request/response transaction model.
   Each transaction consists of a request that invokes a particular
   method, or function, on the server and at least one response.  In
   this example, the transaction begins with Alice's softphone sending
   an INVITE request addressed to Bob's SIP URI.  INVITE is an example
   of a SIP method that specifies the action that the requestor (Alice)
   wants the server (Bob) to take.  The INVITE request contains a number
   of header fields.  Header fields are named attributes that provide
   additional information about a message.  The ones present in an
   INVITE include a unique identifier for the call, the destination
   address, Alice's address, and information about the type of session
   that Alice wishes to establish with Bob.  The INVITE (message F1 in
   Figure 1) might look like this:








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                     atlanta.com  . . . biloxi.com
                 .      proxy              proxy     .
               .                                       .
       Alice's  . . . . . . . . . . . . . . . . . . . .  Bob's
      softphone                                        SIP Phone
         |                |                |                |
         |    INVITE F1   |                |                |
         |---------------&gt;|    INVITE F2   |                |
         |  100 Trying F3 |---------------&gt;|    INVITE F4   |
         |&lt;---------------|  100 Trying F5 |---------------&gt;|
         |                |&lt;-------------- | 180 Ringing F6 |
         |                | 180 Ringing F7 |&lt;---------------|
         | 180 Ringing F8 |&lt;---------------|     200 OK F9  |
         |&lt;---------------|    200 OK F10  |&lt;---------------|
         |    200 OK F11  |&lt;---------------|                |
         |&lt;---------------|                |                |
         |                       ACK F12                    |
         |-------------------------------------------------&gt;|
         |                   Media Session                  |
         |&lt;================================================&gt;|
         |                       BYE F13                    |
         |&lt;-------------------------------------------------|
         |                     200 OK F14                   |
         |-------------------------------------------------&gt;|
         |                                                  |

         Figure 1: SIP session setup example with SIP trapezoid

      INVITE sip:bob@biloxi.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
      Max-Forwards: 70
      To: Bob &lt;sip:bob@biloxi.com&gt;
      From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710@pc33.atlanta.com
      CSeq: 314159 INVITE
      Contact: &lt;sip:alice@pc33.atlanta.com&gt;
      Content-Type: application/sdp
      Content-Length: 142

      (Alice's SDP not shown)

   The first line of the text-encoded message contains the method name
   (INVITE).  The lines that follow are a list of header fields.  This
   example contains a minimum required set.  The header fields are
   briefly described below:






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   Via contains the address (pc33.atlanta.com) at which Alice is
   expecting to receive responses to this request.  It also contains a
   branch parameter that identifies this transaction.

   To contains a display name (Bob) and a SIP or SIPS URI
   (sip:bob@biloxi.com) towards which the request was originally
   directed.  Display names are described in <a href="./rfc2822">RFC 2822</a> [<a href="#ref-3" title="&quot;Internet Message Format&quot;">3</a>].

   From also contains a display name (Alice) and a SIP or SIPS URI
   (sip:alice@atlanta.com) that indicate the originator of the request.
   This header field also has a tag parameter containing a random string
   (1928301774) that was added to the URI by the softphone.  It is used
   for identification purposes.

   Call-ID contains a globally unique identifier for this call,
   generated by the combination of a random string and the softphone's
   host name or IP address.  The combination of the To tag, From tag,
   and Call-ID completely defines a peer-to-peer SIP relationship
   between Alice and Bob and is referred to as a dialog.

   CSeq or Command Sequence contains an integer and a method name.  The
   CSeq number is incremented for each new request within a dialog and
   is a traditional sequence number.

   Contact contains a SIP or SIPS URI that represents a direct route to
   contact Alice, usually composed of a username at a fully qualified
   domain name (FQDN).  While an FQDN is preferred, many end systems do
   not have registered domain names, so IP addresses are permitted.
   While the Via header field tells other elements where to send the
   response, the Contact header field tells other elements where to send
   future requests.

   Max-Forwards serves to limit the number of hops a request can make on
   the way to its destination.  It consists of an integer that is
   decremented by one at each hop.

   Content-Type contains a description of the message body (not shown).

   Content-Length contains an octet (byte) count of the message body.

   The complete set of SIP header fields is defined in <a href="#section-20">Section 20</a>.

   The details of the session, such as the type of media, codec, or
   sampling rate, are not described using SIP.  Rather, the body of a
   SIP message contains a description of the session, encoded in some
   other protocol format.  One such format is the Session Description
   Protocol (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-1" title="&quot;SDP: Session Description Protocol&quot;">1</a>]).  This SDP message (not shown in the




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   example) is carried by the SIP message in a way that is analogous to
   a document attachment being carried by an email message, or a web
   page being carried in an HTTP message.

   Since the softphone does not know the location of Bob or the SIP
   server in the biloxi.com domain, the softphone sends the INVITE to
   the SIP server that serves Alice's domain, atlanta.com.  The address
   of the atlanta.com SIP server could have been configured in Alice's
   softphone, or it could have been discovered by DHCP, for example.

   The atlanta.com SIP server is a type of SIP server known as a proxy
   server.  A proxy server receives SIP requests and forwards them on
   behalf of the requestor.  In this example, the proxy server receives
   the INVITE request and sends a 100 (Trying) response back to Alice's
   softphone.  The 100 (Trying) response indicates that the INVITE has
   been received and that the proxy is working on her behalf to route
   the INVITE to the destination.  Responses in SIP use a three-digit
   code followed by a descriptive phrase.  This response contains the
   same To, From, Call-ID, CSeq and branch parameter in the Via as the
   INVITE, which allows Alice's softphone to correlate this response to
   the sent INVITE.  The atlanta.com proxy server locates the proxy
   server at biloxi.com, possibly by performing a particular type of DNS
   (Domain Name Service) lookup to find the SIP server that serves the
   biloxi.com domain.  This is described in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].  As a result, it
   obtains the IP address of the biloxi.com proxy server and forwards,
   or proxies, the INVITE request there.  Before forwarding the request,
   the atlanta.com proxy server adds an additional Via header field
   value that contains its own address (the INVITE already contains
   Alice's address in the first Via).  The biloxi.com proxy server
   receives the INVITE and responds with a 100 (Trying) response back to
   the atlanta.com proxy server to indicate that it has received the
   INVITE and is processing the request.  The proxy server consults a
   database, generically called a location service, that contains the
   current IP address of Bob.  (We shall see in the next section how
   this database can be populated.)  The biloxi.com proxy server adds
   another Via header field value with its own address to the INVITE and
   proxies it to Bob's SIP phone.

   Bob's SIP phone receives the INVITE and alerts Bob to the incoming
   call from Alice so that Bob can decide whether to answer the call,
   that is, Bob's phone rings.  Bob's SIP phone indicates this in a 180
   (Ringing) response, which is routed back through the two proxies in
   the reverse direction.  Each proxy uses the Via header field to
   determine where to send the response and removes its own address from
   the top.  As a result, although DNS and location service lookups were
   required to route the initial INVITE, the 180 (Ringing) response can
   be returned to the caller without lookups or without state being




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   maintained in the proxies.  This also has the desirable property that
   each proxy that sees the INVITE will also see all responses to the
   INVITE.

   When Alice's softphone receives the 180 (Ringing) response, it passes
   this information to Alice, perhaps using an audio ringback tone or by
   displaying a message on Alice's screen.

   In this example, Bob decides to answer the call.  When he picks up
   the handset, his SIP phone sends a 200 (OK) response to indicate that
   the call has been answered.  The 200 (OK) contains a message body
   with the SDP media description of the type of session that Bob is
   willing to establish with Alice.  As a result, there is a two-phase
   exchange of SDP messages: Alice sent one to Bob, and Bob sent one
   back to Alice.  This two-phase exchange provides basic negotiation
   capabilities and is based on a simple offer/answer model of SDP
   exchange.  If Bob did not wish to answer the call or was busy on
   another call, an error response would have been sent instead of the
   200 (OK), which would have resulted in no media session being
   established.  The complete list of SIP response codes is in <a href="#section-21">Section</a>
   <a href="#section-21">21</a>.  The 200 (OK) (message F9 in Figure 1) might look like this as
   Bob sends it out:

      SIP/2.0 200 OK
      Via: SIP/2.0/UDP server10.biloxi.com
         ;branch=z9hG4bKnashds8;received=192.0.2.3
      Via: SIP/2.0/UDP bigbox3.site3.atlanta.com
         ;branch=z9hG4bK77ef4c2312983.1;received=192.0.2.2
      Via: SIP/2.0/UDP pc33.atlanta.com
         ;branch=z9hG4bK776asdhds ;received=192.0.2.1
      To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
      From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710@pc33.atlanta.com
      CSeq: 314159 INVITE
      Contact: &lt;sip:bob@192.0.2.4&gt;
      Content-Type: application/sdp
      Content-Length: 131

      (Bob's SDP not shown)

   The first line of the response contains the response code (200) and
   the reason phrase (OK).  The remaining lines contain header fields.
   The Via, To, From, Call-ID, and CSeq header fields are copied from
   the INVITE request.  (There are three Via header field values - one
   added by Alice's SIP phone, one added by the atlanta.com proxy, and
   one added by the biloxi.com proxy.)  Bob's SIP phone has added a tag
   parameter to the To header field.  This tag will be incorporated by
   both endpoints into the dialog and will be included in all future



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   requests and responses in this call.  The Contact header field
   contains a URI at which Bob can be directly reached at his SIP phone.
   The Content-Type and Content-Length refer to the message body (not
   shown) that contains Bob's SDP media information.

   In addition to DNS and location service lookups shown in this
   example, proxy servers can make flexible "routing decisions" to
   decide where to send a request.  For example, if Bob's SIP phone
   returned a 486 (Busy Here) response, the biloxi.com proxy server
   could proxy the INVITE to Bob's voicemail server.  A proxy server can
   also send an INVITE to a number of locations at the same time.  This
   type of parallel search is known as forking.

   In this case, the 200 (OK) is routed back through the two proxies and
   is received by Alice's softphone, which then stops the ringback tone
   and indicates that the call has been answered.  Finally, Alice's
   softphone sends an acknowledgement message, ACK, to Bob's SIP phone
   to confirm the reception of the final response (200 (OK)).  In this
   example, the ACK is sent directly from Alice's softphone to Bob's SIP
   phone, bypassing the two proxies.  This occurs because the endpoints
   have learned each other's address from the Contact header fields
   through the INVITE/200 (OK) exchange, which was not known when the
   initial INVITE was sent.  The lookups performed by the two proxies
   are no longer needed, so the proxies drop out of the call flow.  This
   completes the INVITE/200/ACK three-way handshake used to establish
   SIP sessions.  Full details on session setup are in <a href="#section-13">Section 13</a>.

   Alice and Bob's media session has now begun, and they send media
   packets using the format to which they agreed in the exchange of SDP.
   In general, the end-to-end media packets take a different path from
   the SIP signaling messages.

   During the session, either Alice or Bob may decide to change the
   characteristics of the media session.  This is accomplished by
   sending a re-INVITE containing a new media description.  This re-
   INVITE references the existing dialog so that the other party knows
   that it is to modify an existing session instead of establishing a
   new session.  The other party sends a 200 (OK) to accept the change.
   The requestor responds to the 200 (OK) with an ACK.  If the other
   party does not accept the change, he sends an error response such as
   488 (Not Acceptable Here), which also receives an ACK.  However, the
   failure of the re-INVITE does not cause the existing call to fail -
   the session continues using the previously negotiated
   characteristics.  Full details on session modification are in <a href="#section-14">Section</a>
   <a href="#section-14">14</a>.






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   At the end of the call, Bob disconnects (hangs up) first and
   generates a BYE message.  This BYE is routed directly to Alice's
   softphone, again bypassing the proxies.  Alice confirms receipt of
   the BYE with a 200 (OK) response, which terminates the session and
   the BYE transaction.  No ACK is sent - an ACK is only sent in
   response to a response to an INVITE request.  The reasons for this
   special handling for INVITE will be discussed later, but relate to
   the reliability mechanisms in SIP, the length of time it can take for
   a ringing phone to be answered, and forking.  For this reason,
   request handling in SIP is often classified as either INVITE or non-
   INVITE, referring to all other methods besides INVITE.  Full details
   on session termination are in <a href="#section-15">Section 15</a>.

   <a href="#section-24.2">Section 24.2</a> describes the messages shown in Figure 1 in full.

   In some cases, it may be useful for proxies in the SIP signaling path
   to see all the messaging between the endpoints for the duration of
   the session.  For example, if the biloxi.com proxy server wished to
   remain in the SIP messaging path beyond the initial INVITE, it would
   add to the INVITE a required routing header field known as Record-
   Route that contained a URI resolving to the hostname or IP address of
   the proxy.  This information would be received by both Bob's SIP
   phone and (due to the Record-Route header field being passed back in
   the 200 (OK)) Alice's softphone and stored for the duration of the
   dialog.  The biloxi.com proxy server would then receive and proxy the
   ACK, BYE, and 200 (OK) to the BYE.  Each proxy can independently
   decide to receive subsequent messages, and those messages will pass
   through all proxies that elect to receive it.  This capability is
   frequently used for proxies that are providing mid-call features.

   Registration is another common operation in SIP.  Registration is one
   way that the biloxi.com server can learn the current location of Bob.
   Upon initialization, and at periodic intervals, Bob's SIP phone sends
   REGISTER messages to a server in the biloxi.com domain known as a SIP
   registrar.  The REGISTER messages associate Bob's SIP or SIPS URI
   (sip:bob@biloxi.com) with the machine into which he is currently
   logged (conveyed as a SIP or SIPS URI in the Contact header field).
   The registrar writes this association, also called a binding, to a
   database, called the location service, where it can be used by the
   proxy in the biloxi.com domain.  Often, a registrar server for a
   domain is co-located with the proxy for that domain.  It is an
   important concept that the distinction between types of SIP servers
   is logical, not physical.

   Bob is not limited to registering from a single device.  For example,
   both his SIP phone at home and the one in the office could send
   registrations.  This information is stored together in the location




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   service and allows a proxy to perform various types of searches to
   locate Bob.  Similarly, more than one user can be registered on a
   single device at the same time.

   The location service is just an abstract concept.  It generally
   contains information that allows a proxy to input a URI and receive a
   set of zero or more URIs that tell the proxy where to send the
   request.  Registrations are one way to create this information, but
   not the only way.  Arbitrary mapping functions can be configured at
   the discretion of the administrator.

   Finally, it is important to note that in SIP, registration is used
   for routing incoming SIP requests and has no role in authorizing
   outgoing requests.  Authorization and authentication are handled in
   SIP either on a request-by-request basis with a challenge/response
   mechanism, or by using a lower layer scheme as discussed in <a href="#section-26">Section</a>
   <a href="#section-26">26</a>.

   The complete set of SIP message details for this registration example
   is in <a href="#section-24.1">Section 24.1</a>.

   Additional operations in SIP, such as querying for the capabilities
   of a SIP server or client using OPTIONS, or canceling a pending
   request using CANCEL, will be introduced in later sections.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a> Structure of the Protocol</span>

   SIP is structured as a layered protocol, which means that its
   behavior is described in terms of a set of fairly independent
   processing stages with only a loose coupling between each stage.  The
   protocol behavior is described as layers for the purpose of
   presentation, allowing the description of functions common across
   elements in a single section.  It does not dictate an implementation
   in any way.  When we say that an element "contains" a layer, we mean
   it is compliant to the set of rules defined by that layer.

   Not every element specified by the protocol contains every layer.
   Furthermore, the elements specified by SIP are logical elements, not
   physical ones.  A physical realization can choose to act as different
   logical elements, perhaps even on a transaction-by-transaction basis.

   The lowest layer of SIP is its syntax and encoding.  Its encoding is
   specified using an augmented Backus-Naur Form grammar (BNF).  The
   complete BNF is specified in <a href="#section-25">Section 25</a>; an overview of a SIP
   message's structure can be found in <a href="#section-7">Section 7</a>.






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   The second layer is the transport layer.  It defines how a client
   sends requests and receives responses and how a server receives
   requests and sends responses over the network.  All SIP elements
   contain a transport layer.  The transport layer is described in
   <a href="#section-18">Section 18</a>.

   The third layer is the transaction layer.  Transactions are a
   fundamental component of SIP.  A transaction is a request sent by a
   client transaction (using the transport layer) to a server
   transaction, along with all responses to that request sent from the
   server transaction back to the client.  The transaction layer handles
   application-layer retransmissions, matching of responses to requests,
   and application-layer timeouts.  Any task that a user agent client
   (UAC) accomplishes takes place using a series of transactions.
   Discussion of transactions can be found in <a href="#section-17">Section 17</a>.  User agents
   contain a transaction layer, as do stateful proxies.  Stateless
   proxies do not contain a transaction layer.  The transaction layer
   has a client component (referred to as a client transaction) and a
   server component (referred to as a server transaction), each of which
   are represented by a finite state machine that is constructed to
   process a particular request.

   The layer above the transaction layer is called the transaction user
   (TU).  Each of the SIP entities, except the stateless proxy, is a
   transaction user.  When a TU wishes to send a request, it creates a
   client transaction instance and passes it the request along with the
   destination IP address, port, and transport to which to send the
   request.  A TU that creates a client transaction can also cancel it.
   When a client cancels a transaction, it requests that the server stop
   further processing, revert to the state that existed before the
   transaction was initiated, and generate a specific error response to
   that transaction.  This is done with a CANCEL request, which
   constitutes its own transaction, but references the transaction to be
   cancelled (<a href="#section-9">Section 9</a>).

   The SIP elements, that is, user agent clients and servers, stateless
   and stateful proxies and registrars, contain a core that
   distinguishes them from each other.  Cores, except for the stateless
   proxy, are transaction users.  While the behavior of the UAC and UAS
   cores depends on the method, there are some common rules for all
   methods (<a href="#section-8">Section 8</a>).  For a UAC, these rules govern the construction
   of a request; for a UAS, they govern the processing of a request and
   generating a response.  Since registrations play an important role in
   SIP, a UAS that handles a REGISTER is given the special name
   registrar.  <a href="#section-10">Section 10</a> describes UAC and UAS core behavior for the
   REGISTER method.  <a href="#section-11">Section 11</a> describes UAC and UAS core behavior for
   the OPTIONS method, used for determining the capabilities of a UA.




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   Certain other requests are sent within a dialog.  A dialog is a
   peer-to-peer SIP relationship between two user agents that persists
   for some time.  The dialog facilitates sequencing of messages and
   proper routing of requests between the user agents.  The INVITE
   method is the only way defined in this specification to establish a
   dialog.  When a UAC sends a request that is within the context of a
   dialog, it follows the common UAC rules as discussed in <a href="#section-8">Section 8</a> but
   also the rules for mid-dialog requests.  <a href="#section-12">Section 12</a> discusses dialogs
   and presents the procedures for their construction and maintenance,
   in addition to construction of requests within a dialog.

   The most important method in SIP is the INVITE method, which is used
   to establish a session between participants.  A session is a
   collection of participants, and streams of media between them, for
   the purposes of communication.  <a href="#section-13">Section 13</a> discusses how sessions are
   initiated, resulting in one or more SIP dialogs.  <a href="#section-14">Section 14</a>
   discusses how characteristics of that session are modified through
   the use of an INVITE request within a dialog.  Finally, <a href="#section-15">section 15</a>
   discusses how a session is terminated.

   The procedures of Sections <a href="#section-8">8</a>, <a href="#section-10">10</a>, <a href="#section-11">11</a>, <a href="#section-12">12</a>, <a href="#section-13">13</a>, <a href="#section-14">14</a>, and <a href="#section-15">15</a> deal
   entirely with the UA core (<a href="#section-9">Section 9</a> describes cancellation, which
   applies to both UA core and proxy core).  <a href="#section-16">Section 16</a> discusses the
   proxy element, which facilitates routing of messages between user
   agents.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a> Definitions</span>

   The following terms have special significance for SIP.

      Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user.

      Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
         logical entity that receives a request and processes it as a
         user agent server (UAS).  In order to determine how the request
         should be answered, it acts as a user agent client (UAC) and
         generates requests.  Unlike a proxy server, it maintains dialog
         state and must participate in all requests sent on the dialogs
         it has established.  Since it is a concatenation of a UAC and
         UAS, no explicit definitions are needed for its behavior.






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      Call: A call is an informal term that refers to some communication
         between peers, generally set up for the purposes of a
         multimedia conversation.

      Call Leg: Another name for a dialog [<a href="#ref-31" title="&quot;SIP: Session Initiation Protocol&quot;">31</a>]; no longer used in this
         specification.

      Call Stateful: A proxy is call stateful if it retains state for a
         dialog from the initiating INVITE to the terminating BYE
         request.  A call stateful proxy is always transaction stateful,
         but the converse is not necessarily true.

      Client: A client is any network element that sends SIP requests
         and receives SIP responses.  Clients may or may not interact
         directly with a human user.  User agent clients and proxies are
         clients.

      Conference: A multimedia session (see below) that contains
         multiple participants.

      Core: Core designates the functions specific to a particular type
         of SIP entity, i.e., specific to either a stateful or stateless
         proxy, a user agent or registrar.  All cores, except those for
         the stateless proxy, are transaction users.

      Dialog: A dialog is a peer-to-peer SIP relationship between two
         UAs that persists for some time.  A dialog is established by
         SIP messages, such as a 2xx response to an INVITE request.  A
         dialog is identified by a call identifier, local tag, and a
         remote tag.  A dialog was formerly known as a call leg in <a href="./rfc2543">RFC</a>
         <a href="./rfc2543">2543</a>.

      Downstream: A direction of message forwarding within a transaction
         that refers to the direction that requests flow from the user
         agent client to user agent server.

      Final Response: A response that terminates a SIP transaction, as
         opposed to a provisional response that does not.  All 2xx, 3xx,
         4xx, 5xx and 6xx responses are final.

      Header: A header is a component of a SIP message that conveys
         information about the message.  It is structured as a sequence
         of header fields.

      Header Field: A header field is a component of the SIP message
         header.  A header field can appear as one or more header field
         rows. Header field rows consist of a header field name and zero
         or more header field values. Multiple header field values on a



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         given header field row are separated by commas. Some header
         fields can only have a single header field value, and as a
         result, always appear as a single header field row.

      Header Field Value: A header field value is a single value; a
         header field consists of zero or more header field values.

      Home Domain: The domain providing service to a SIP user.
         Typically, this is the domain present in the URI in the
         address-of-record of a registration.

      Informational Response: Same as a provisional response.

      Initiator, Calling Party, Caller: The party initiating a session
         (and dialog) with an INVITE request.  A caller retains this
         role from the time it sends the initial INVITE that established
         a dialog until the termination of that dialog.

      Invitation: An INVITE request.

      Invitee, Invited User, Called Party, Callee: The party that
         receives an INVITE request for the purpose of establishing a
         new session.  A callee retains this role from the time it
         receives the INVITE until the termination of the dialog
         established by that INVITE.

      Location Service: A location service is used by a SIP redirect or
         proxy server to obtain information about a callee's possible
         location(s).  It contains a list of bindings of address-of-
         record keys to zero or more contact addresses.  The bindings
         can be created and removed in many ways; this specification
         defines a REGISTER method that updates the bindings.

      Loop: A request that arrives at a proxy, is forwarded, and later
         arrives back at the same proxy.  When it arrives the second
         time, its Request-URI is identical to the first time, and other
         header fields that affect proxy operation are unchanged, so
         that the proxy would make the same processing decision on the
         request it made the first time.  Looped requests are errors,
         and the procedures for detecting them and handling them are
         described by the protocol.

      Loose Routing: A proxy is said to be loose routing if it follows
         the procedures defined in this specification for processing of
         the Route header field.  These procedures separate the
         destination of the request (present in the Request-URI) from





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         the set of proxies that need to be visited along the way
         (present in the Route header field).  A proxy compliant to
         these mechanisms is also known as a loose router.

      Message: Data sent between SIP elements as part of the protocol.
         SIP messages are either requests or responses.

      Method: The method is the primary function that a request is meant
         to invoke on a server.  The method is carried in the request
         message itself.  Example methods are INVITE and BYE.

      Outbound Proxy: A proxy that receives requests from a client, even
         though it may not be the server resolved by the Request-URI.
         Typically, a UA is manually configured with an outbound proxy,
         or can learn about one through auto-configuration protocols.

      Parallel Search: In a parallel search, a proxy issues several
         requests to possible user locations upon receiving an incoming
         request.  Rather than issuing one request and then waiting for
         the final response before issuing the next request as in a
         sequential search, a parallel search issues requests without
         waiting for the result of previous requests.

      Provisional Response: A response used by the server to indicate
         progress, but that does not terminate a SIP transaction.  1xx
         responses are provisional, other responses are considered
         final.

      Proxy, Proxy Server: An intermediary entity that acts as both a
         server and a client for the purpose of making requests on
         behalf of other clients.  A proxy server primarily plays the
         role of routing, which means its job is to ensure that a
         request is sent to another entity "closer" to the targeted
         user.  Proxies are also useful for enforcing policy (for
         example, making sure a user is allowed to make a call).  A
         proxy interprets, and, if necessary, rewrites specific parts of
         a request message before forwarding it.

      Recursion: A client recurses on a 3xx response when it generates a
         new request to one or more of the URIs in the Contact header
         field in the response.

      Redirect Server: A redirect server is a user agent server that
         generates 3xx responses to requests it receives, directing the
         client to contact an alternate set of URIs.






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      Registrar: A registrar is a server that accepts REGISTER requests
         and places the information it receives in those requests into
         the location service for the domain it handles.

      Regular Transaction: A regular transaction is any transaction with
         a method other than INVITE, ACK, or CANCEL.

      Request: A SIP message sent from a client to a server, for the
         purpose of invoking a particular operation.

      Response: A SIP message sent from a server to a client, for
         indicating the status of a request sent from the client to the
         server.

      Ringback: Ringback is the signaling tone produced by the calling
         party's application indicating that a called party is being
         alerted (ringing).

      Route Set: A route set is a collection of ordered SIP or SIPS URI
         which represent a list of proxies that must be traversed when
         sending a particular request.  A route set can be learned,
         through headers like Record-Route, or it can be configured.

      Server: A server is a network element that receives requests in
         order to service them and sends back responses to those
         requests.  Examples of servers are proxies, user agent servers,
         redirect servers, and registrars.

      Sequential Search: In a sequential search, a proxy server attempts
         each contact address in sequence, proceeding to the next one
         only after the previous has generated a final response.  A 2xx
         or 6xx class final response always terminates a sequential
         search.

      Session: From the SDP specification: "A multimedia session is a
         set of multimedia senders and receivers and the data streams
         flowing from senders to receivers.  A multimedia conference is
         an example of a multimedia session." (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-1" title="&quot;SDP: Session Description Protocol&quot;">1</a>]) (A session
         as defined for SDP can comprise one or more RTP sessions.)  As
         defined, a callee can be invited several times, by different
         calls, to the same session.  If SDP is used, a session is
         defined by the concatenation of the SDP user name, session id,
         network type, address type, and address elements in the origin
         field.

      SIP Transaction: A SIP transaction occurs between a client and a
         server and comprises all messages from the first request sent
         from the client to the server up to a final (non-1xx) response



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         sent from the server to the client.  If the request is INVITE
         and the final response is a non-2xx, the transaction also
         includes an ACK to the response.  The ACK for a 2xx response to
         an INVITE request is a separate transaction.

      Spiral: A spiral is a SIP request that is routed to a proxy,
         forwarded onwards, and arrives once again at that proxy, but
         this time differs in a way that will result in a different
         processing decision than the original request.  Typically, this
         means that the request's Request-URI differs from its previous
         arrival.  A spiral is not an error condition, unlike a loop.  A
         typical cause for this is call forwarding.  A user calls
         joe@example.com.  The example.com proxy forwards it to Joe's
         PC, which in turn, forwards it to bob@example.com.  This
         request is proxied back to the example.com proxy.  However,
         this is not a loop.  Since the request is targeted at a
         different user, it is considered a spiral, and is a valid
         condition.

      Stateful Proxy: A logical entity that maintains the client and
         server transaction state machines defined by this specification
         during the processing of a request, also known as a transaction
         stateful proxy.  The behavior of a stateful proxy is further
         defined in <a href="#section-16">Section 16</a>.  A (transaction) stateful proxy is not
         the same as a call stateful proxy.

      Stateless Proxy: A logical entity that does not maintain the
         client or server transaction state machines defined in this
         specification when it processes requests.  A stateless proxy
         forwards every request it receives downstream and every
         response it receives upstream.

      Strict Routing: A proxy is said to be strict routing if it follows
         the Route processing rules of <a href="./rfc2543">RFC 2543</a> and many prior work in
         progress versions of this RFC.  That rule caused proxies to
         destroy the contents of the Request-URI when a Route header
         field was present.  Strict routing behavior is not used in this
         specification, in favor of a loose routing behavior.  Proxies
         that perform strict routing are also known as strict routers.

      Target Refresh Request: A target refresh request sent within a
         dialog is defined as a request that can modify the remote
         target of the dialog.

      Transaction User (TU): The layer of protocol processing that
         resides above the transaction layer.  Transaction users include
         the UAC core, UAS core, and proxy core.




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      Upstream: A direction of message forwarding within a transaction
         that refers to the direction that responses flow from the user
         agent server back to the user agent client.

      URL-encoded: A character string encoded according to <a href="./rfc2396#section-2.4">RFC 2396,
         Section&nbsp;2.4</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>].

      User Agent Client (UAC): A user agent client is a logical entity
         that creates a new request, and then uses the client
         transaction state machinery to send it.  The role of UAC lasts
         only for the duration of that transaction.  In other words, if
         a piece of software initiates a request, it acts as a UAC for
         the duration of that transaction.  If it receives a request
         later, it assumes the role of a user agent server for the
         processing of that transaction.

      UAC Core: The set of processing functions required of a UAC that
         reside above the transaction and transport layers.

      User Agent Server (UAS): A user agent server is a logical entity
         that generates a response to a SIP request.  The response
         accepts, rejects, or redirects the request.  This role lasts
         only for the duration of that transaction.  In other words, if
         a piece of software responds to a request, it acts as a UAS for
         the duration of that transaction.  If it generates a request
         later, it assumes the role of a user agent client for the
         processing of that transaction.

      UAS Core: The set of processing functions required at a UAS that
         resides above the transaction and transport layers.

      User Agent (UA): A logical entity that can act as both a user
         agent client and user agent server.

   The role of UAC and UAS, as well as proxy and redirect servers, are
   defined on a transaction-by-transaction basis.  For example, the user
   agent initiating a call acts as a UAC when sending the initial INVITE
   request and as a UAS when receiving a BYE request from the callee.
   Similarly, the same software can act as a proxy server for one
   request and as a redirect server for the next request.

   Proxy, location, and registrar servers defined above are logical
   entities; implementations MAY combine them into a single application.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a> SIP Messages</span>

   SIP is a text-based protocol and uses the UTF-8 charset (<a href="./rfc2279">RFC 2279</a>
   [<a href="#ref-7" title="&quot;UTF-8, a transformation format of ISO 10646&quot;">7</a>]).



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   A SIP message is either a request from a client to a server, or a
   response from a server to a client.

   Both Request (<a href="#section-7.1">section 7.1</a>) and Response (<a href="#section-7.2">section 7.2</a>) messages use
   the basic format of <a href="./rfc2822">RFC 2822</a> [<a href="#ref-3" title="&quot;Internet Message Format&quot;">3</a>], even though the syntax differs in
   character set and syntax specifics.  (SIP allows header fields that
   would not be valid <a href="./rfc2822">RFC 2822</a> header fields, for example.)  Both types
   of messages consist of a start-line, one or more header fields, an
   empty line indicating the end of the header fields, and an optional
   message-body.

         generic-message  =  start-line
                             *message-header
                             CRLF
                             [ message-body ]
         start-line       =  Request-Line / Status-Line

   The start-line, each message-header line, and the empty line MUST be
   terminated by a carriage-return line-feed sequence (CRLF).  Note that
   the empty line MUST be present even if the message-body is not.

   Except for the above difference in character sets, much of SIP's
   message and header field syntax is identical to HTTP/1.1.  Rather
   than repeating the syntax and semantics here, we use [HX.Y] to refer
   to Section X.Y of the current HTTP/1.1 specification (<a href="./rfc2616">RFC 2616</a> [<a href="#ref-8" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">8</a>]).

   However, SIP is not an extension of HTTP.

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a> Requests</span>

   SIP requests are distinguished by having a Request-Line for a start-
   line.  A Request-Line contains a method name, a Request-URI, and the
   protocol version separated by a single space (SP) character.

   The Request-Line ends with CRLF.  No CR or LF are allowed except in
   the end-of-line CRLF sequence.  No linear whitespace (LWS) is allowed
   in any of the elements.

         Request-Line  =  Method SP Request-URI SP SIP-Version CRLF

      Method: This specification defines six methods: REGISTER for
           registering contact information, INVITE, ACK, and CANCEL for
           setting up sessions, BYE for terminating sessions, and
           OPTIONS for querying servers about their capabilities.  SIP
           extensions, documented in standards track RFCs, may define
           additional methods.





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      Request-URI: The Request-URI is a SIP or SIPS URI as described in
           <a href="#section-19.1">Section 19.1</a> or a general URI (<a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>]).  It indicates
           the user or service to which this request is being addressed.
           The Request-URI MUST NOT contain unescaped spaces or control
           characters and MUST NOT be enclosed in "&lt;&gt;".

           SIP elements MAY support Request-URIs with schemes other than
           "sip" and "sips", for example the "tel" URI scheme of <a href="./rfc2806">RFC</a>
           <a href="./rfc2806">2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>].  SIP elements MAY translate non-SIP URIs using any
           mechanism at their disposal, resulting in SIP URI, SIPS URI,
           or some other scheme.

      SIP-Version: Both request and response messages include the
           version of SIP in use, and follow [H3.1] (with HTTP replaced
           by SIP, and HTTP/1.1 replaced by SIP/2.0) regarding version
           ordering, compliance requirements, and upgrading of version
           numbers.  To be compliant with this specification,
           applications sending SIP messages MUST include a SIP-Version
           of "SIP/2.0".  The SIP-Version string is case-insensitive,
           but implementations MUST send upper-case.

           Unlike HTTP/1.1, SIP treats the version number as a literal
           string.  In practice, this should make no difference.

<span class="h3"><a class="selflink" id="section-7.2" href="#section-7.2">7.2</a> Responses</span>

   SIP responses are distinguished from requests by having a Status-Line
   as their start-line.  A Status-Line consists of the protocol version
   followed by a numeric Status-Code and its associated textual phrase,
   with each element separated by a single SP character.

   No CR or LF is allowed except in the final CRLF sequence.

      Status-Line  =  SIP-Version SP Status-Code SP Reason-Phrase CRLF

   The Status-Code is a 3-digit integer result code that indicates the
   outcome of an attempt to understand and satisfy a request.  The
   Reason-Phrase is intended to give a short textual description of the
   Status-Code.  The Status-Code is intended for use by automata,
   whereas the Reason-Phrase is intended for the human user.  A client
   is not required to examine or display the Reason-Phrase.

   While this specification suggests specific wording for the reason
   phrase, implementations MAY choose other text, for example, in the
   language indicated in the Accept-Language header field of the
   request.





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   The first digit of the Status-Code defines the class of response.
   The last two digits do not have any categorization role.  For this
   reason, any response with a status code between 100 and 199 is
   referred to as a "1xx response", any response with a status code
   between 200 and 299 as a "2xx response", and so on.  SIP/2.0 allows
   six values for the first digit:

      1xx: Provisional -- request received, continuing to process the
           request;

      2xx: Success -- the action was successfully received, understood,
           and accepted;

      3xx: Redirection -- further action needs to be taken in order to
           complete the request;

      4xx: Client Error -- the request contains bad syntax or cannot be
           fulfilled at this server;

      5xx: Server Error -- the server failed to fulfill an apparently
           valid request;

      6xx: Global Failure -- the request cannot be fulfilled at any
           server.

   <a href="#section-21">Section 21</a> defines these classes and describes the individual codes.

<span class="h3"><a class="selflink" id="section-7.3" href="#section-7.3">7.3</a> Header Fields</span>

   SIP header fields are similar to HTTP header fields in both syntax
   and semantics.  In particular, SIP header fields follow the [<a href="#ref-H4.2">H4.2</a>]
   definitions of syntax for the message-header and the rules for
   extending header fields over multiple lines.  However, the latter is
   specified in HTTP with implicit whitespace and folding.  This
   specification conforms to <a href="./rfc2234">RFC 2234</a> [<a href="#ref-10" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">10</a>] and uses only explicit
   whitespace and folding as an integral part of the grammar.

   [<a id="ref-H4.2">H4.2</a>] also specifies that multiple header fields of the same field
   name whose value is a comma-separated list can be combined into one
   header field.  That applies to SIP as well, but the specific rule is
   different because of the different grammars.  Specifically, any SIP
   header whose grammar is of the form

      header  =  "header-name" HCOLON header-value *(COMMA header-value)

   allows for combining header fields of the same name into a comma-
   separated list.  The Contact header field allows a comma-separated
   list unless the header field value is "*".



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<span class="h4"><a class="selflink" id="section-7.3.1" href="#section-7.3.1">7.3.1</a> Header Field Format</span>

   Header fields follow the same generic header format as that given in
   <a href="./rfc2822#section-2.2">Section&nbsp;2.2 of RFC 2822</a> [<a href="#ref-3" title="&quot;Internet Message Format&quot;">3</a>].  Each header field consists of a field
   name followed by a colon (":") and the field value.

      field-name: field-value

   The formal grammar for a message-header specified in <a href="#section-25">Section 25</a>
   allows for an arbitrary amount of whitespace on either side of the
   colon; however, implementations should avoid spaces between the field
   name and the colon and use a single space (SP) between the colon and
   the field-value.

      Subject:            lunch
      Subject      :      lunch
      Subject            :lunch
      Subject: lunch

   Thus, the above are all valid and equivalent, but the last is the
   preferred form.

   Header fields can be extended over multiple lines by preceding each
   extra line with at least one SP or horizontal tab (HT).  The line
   break and the whitespace at the beginning of the next line are
   treated as a single SP character.  Thus, the following are
   equivalent:

      Subject: I know you're there, pick up the phone and talk to me!
      Subject: I know you're there,
               pick up the phone
               and talk to me!

   The relative order of header fields with different field names is not
   significant.  However, it is RECOMMENDED that header fields which are
   needed for proxy processing (Via, Route, Record-Route, Proxy-Require,
   Max-Forwards, and Proxy-Authorization, for example) appear towards
   the top of the message to facilitate rapid parsing.  The relative
   order of header field rows with the same field name is important.
   Multiple header field rows with the same field-name MAY be present in
   a message if and only if the entire field-value for that header field
   is defined as a comma-separated list (that is, if follows the grammar
   defined in <a href="#section-7.3">Section 7.3</a>).  It MUST be possible to combine the multiple
   header field rows into one "field-name: field-value" pair, without
   changing the semantics of the message, by appending each subsequent
   field-value to the first, each separated by a comma.  The exceptions
   to this rule are the WWW-Authenticate, Authorization, Proxy-
   Authenticate, and Proxy-Authorization header fields.  Multiple header



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   field rows with these names MAY be present in a message, but since
   their grammar does not follow the general form listed in <a href="#section-7.3">Section 7.3</a>,
   they MUST NOT be combined into a single header field row.

   Implementations MUST be able to process multiple header field rows
   with the same name in any combination of the single-value-per-line or
   comma-separated value forms.

   The following groups of header field rows are valid and equivalent:

      Route: &lt;sip:alice@atlanta.com&gt;
      Subject: Lunch
      Route: &lt;sip:bob@biloxi.com&gt;
      Route: &lt;sip:carol@chicago.com&gt;

      Route: &lt;sip:alice@atlanta.com&gt;, &lt;sip:bob@biloxi.com&gt;
      Route: &lt;sip:carol@chicago.com&gt;
      Subject: Lunch

      Subject: Lunch
      Route: &lt;sip:alice@atlanta.com&gt;, &lt;sip:bob@biloxi.com&gt;,
             &lt;sip:carol@chicago.com&gt;

   Each of the following blocks is valid but not equivalent to the
   others:

      Route: &lt;sip:alice@atlanta.com&gt;
      Route: &lt;sip:bob@biloxi.com&gt;
      Route: &lt;sip:carol@chicago.com&gt;

      Route: &lt;sip:bob@biloxi.com&gt;
      Route: &lt;sip:alice@atlanta.com&gt;
      Route: &lt;sip:carol@chicago.com&gt;

      Route: &lt;sip:alice@atlanta.com&gt;,&lt;sip:carol@chicago.com&gt;,
             &lt;sip:bob@biloxi.com&gt;

   The format of a header field-value is defined per header-name.  It
   will always be either an opaque sequence of TEXT-UTF8 octets, or a
   combination of whitespace, tokens, separators, and quoted strings.
   Many existing header fields will adhere to the general form of a
   value followed by a semi-colon separated sequence of parameter-name,
   parameter-value pairs:

         field-name: field-value *(;parameter-name=parameter-value)






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   Even though an arbitrary number of parameter pairs may be attached to
   a header field value, any given parameter-name MUST NOT appear more
   than once.

   When comparing header fields, field names are always case-
   insensitive.  Unless otherwise stated in the definition of a
   particular header field, field values, parameter names, and parameter
   values are case-insensitive.  Tokens are always case-insensitive.
   Unless specified otherwise, values expressed as quoted strings are
   case-sensitive.  For example,

      Contact: &lt;sip:alice@atlanta.com&gt;;expires=3600

   is equivalent to

      CONTACT: &lt;sip:alice@atlanta.com&gt;;ExPiReS=3600

   and

      Content-Disposition: session;handling=optional

   is equivalent to

      content-disposition: Session;HANDLING=OPTIONAL

   The following two header fields are not equivalent:

      Warning: 370 devnull "Choose a bigger pipe"
      Warning: 370 devnull "CHOOSE A BIGGER PIPE"

<span class="h4"><a class="selflink" id="section-7.3.2" href="#section-7.3.2">7.3.2</a> Header Field Classification</span>

   Some header fields only make sense in requests or responses.  These
   are called request header fields and response header fields,
   respectively.  If a header field appears in a message not matching
   its category (such as a request header field in a response), it MUST
   be ignored.  <a href="#section-20">Section 20</a> defines the classification of each header
   field.

<span class="h4"><a class="selflink" id="section-7.3.3" href="#section-7.3.3">7.3.3</a> Compact Form</span>

   SIP provides a mechanism to represent common header field names in an
   abbreviated form.  This may be useful when messages would otherwise
   become too large to be carried on the transport available to it
   (exceeding the maximum transmission unit (MTU) when using UDP, for
   example).  These compact forms are defined in <a href="#section-20">Section 20</a>.  A compact
   form MAY be substituted for the longer form of a header field name at
   any time without changing the semantics of the message.  A header



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   field name MAY appear in both long and short forms within the same
   message.  Implementations MUST accept both the long and short forms
   of each header name.

<span class="h3"><a class="selflink" id="section-7.4" href="#section-7.4">7.4</a> Bodies</span>

   Requests, including new requests defined in extensions to this
   specification, MAY contain message bodies unless otherwise noted.
   The interpretation of the body depends on the request method.

   For response messages, the request method and the response status
   code determine the type and interpretation of any message body.  All
   responses MAY include a body.

<span class="h4"><a class="selflink" id="section-7.4.1" href="#section-7.4.1">7.4.1</a> Message Body Type</span>

   The Internet media type of the message body MUST be given by the
   Content-Type header field.  If the body has undergone any encoding
   such as compression, then this MUST be indicated by the Content-
   Encoding header field; otherwise, Content-Encoding MUST be omitted.
   If applicable, the character set of the message body is indicated as
   part of the Content-Type header-field value.

   The "multipart" MIME type defined in <a href="./rfc2046">RFC 2046</a> [<a href="#ref-11" title="&quot;Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types&quot;">11</a>] MAY be used within
   the body of the message.  Implementations that send requests
   containing multipart message bodies MUST send a session description
   as a non-multipart message body if the remote implementation requests
   this through an Accept header field that does not contain multipart.

   SIP messages MAY contain binary bodies or body parts. When no
   explicit charset parameter is provided by the sender, media subtypes
   of the "text" type are defined to have a default charset value of
   "UTF-8".

<span class="h4"><a class="selflink" id="section-7.4.2" href="#section-7.4.2">7.4.2</a> Message Body Length</span>

   The body length in bytes is provided by the Content-Length header
   field.  <a href="#section-20.14">Section 20.14</a> describes the necessary contents of this header
   field in detail.

   The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
   (Note: The chunked encoding modifies the body of a message in order
   to transfer it as a series of chunks, each with its own size
   indicator.)







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<span class="h3"><a class="selflink" id="section-7.5" href="#section-7.5">7.5</a> Framing SIP Messages</span>

   Unlike HTTP, SIP implementations can use UDP or other unreliable
   datagram protocols.  Each such datagram carries one request or
   response.  See <a href="#section-18">Section 18</a> on constraints on usage of unreliable
   transports.

   Implementations processing SIP messages over stream-oriented
   transports MUST ignore any CRLF appearing before the start-line
   [H4.1].

      The Content-Length header field value is used to locate the end of
      each SIP message in a stream.  It will always be present when SIP
      messages are sent over stream-oriented transports.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a> General User Agent Behavior</span>

   A user agent represents an end system.  It contains a user agent
   client (UAC), which generates requests, and a user agent server
   (UAS), which responds to them.  A UAC is capable of generating a
   request based on some external stimulus (the user clicking a button,
   or a signal on a PSTN line) and processing a response.  A UAS is
   capable of receiving a request and generating a response based on
   user input, external stimulus, the result of a program execution, or
   some other mechanism.

   When a UAC sends a request, the request passes through some number of
   proxy servers, which forward the request towards the UAS. When the
   UAS generates a response, the response is forwarded towards the UAC.

   UAC and UAS procedures depend strongly on two factors.  First, based
   on whether the request or response is inside or outside of a dialog,
   and second, based on the method of a request.  Dialogs are discussed
   thoroughly in <a href="#section-12">Section 12</a>; they represent a peer-to-peer relationship
   between user agents and are established by specific SIP methods, such
   as INVITE.

   In this section, we discuss the method-independent rules for UAC and
   UAS behavior when processing requests that are outside of a dialog.
   This includes, of course, the requests which themselves establish a
   dialog.

   Security procedures for requests and responses outside of a dialog
   are described in <a href="#section-26">Section 26</a>.  Specifically, mechanisms exist for the
   UAS and UAC to mutually authenticate.  A limited set of privacy
   features are also supported through encryption of bodies using
   S/MIME.




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<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a> UAC Behavior</span>

   This section covers UAC behavior outside of a dialog.

<span class="h4"><a class="selflink" id="section-8.1.1" href="#section-8.1.1">8.1.1</a> Generating the Request</span>

   A valid SIP request formulated by a UAC MUST, at a minimum, contain
   the following header fields: To, From, CSeq, Call-ID, Max-Forwards,
   and Via; all of these header fields are mandatory in all SIP
   requests.  These six header fields are the fundamental building
   blocks of a SIP message, as they jointly provide for most of the
   critical message routing services including the addressing of
   messages, the routing of responses, limiting message propagation,
   ordering of messages, and the unique identification of transactions.
   These header fields are in addition to the mandatory request line,
   which contains the method, Request-URI, and SIP version.

   Examples of requests sent outside of a dialog include an INVITE to
   establish a session (<a href="#section-13">Section 13</a>) and an OPTIONS to query for
   capabilities (<a href="#section-11">Section 11</a>).

<span class="h5"><a class="selflink" id="section-8.1.1.1" href="#section-8.1.1.1">8.1.1.1</a> Request-URI</span>

   The initial Request-URI of the message SHOULD be set to the value of
   the URI in the To field.  One notable exception is the REGISTER
   method; behavior for setting the Request-URI of REGISTER is given in
   <a href="#section-10">Section 10</a>.  It may also be undesirable for privacy reasons or
   convenience to set these fields to the same value (especially if the
   originating UA expects that the Request-URI will be changed during
   transit).

   In some special circumstances, the presence of a pre-existing route
   set can affect the Request-URI of the message.  A pre-existing route
   set is an ordered set of URIs that identify a chain of servers, to
   which a UAC will send outgoing requests that are outside of a dialog.
   Commonly, they are configured on the UA by a user or service provider
   manually, or through some other non-SIP mechanism.  When a provider
   wishes to configure a UA with an outbound proxy, it is RECOMMENDED
   that this be done by providing it with a pre-existing route set with
   a single URI, that of the outbound proxy.

   When a pre-existing route set is present, the procedures for
   populating the Request-URI and Route header field detailed in <a href="#section-12.2.1.1">Section</a>
   <a href="#section-12.2.1.1">12.2.1.1</a> MUST be followed (even though there is no dialog), using the
   desired Request-URI as the remote target URI.






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<span class="h5"><a class="selflink" id="section-8.1.1.2" href="#section-8.1.1.2">8.1.1.2</a> To</span>

   The To header field first and foremost specifies the desired
   "logical" recipient of the request, or the address-of-record of the
   user or resource that is the target of this request.  This may or may
   not be the ultimate recipient of the request.  The To header field
   MAY contain a SIP or SIPS URI, but it may also make use of other URI
   schemes (the tel URL (<a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>]), for example) when appropriate.
   All SIP implementations MUST support the SIP URI scheme.  Any
   implementation that supports TLS MUST support the SIPS URI scheme.
   The To header field allows for a display name.

   A UAC may learn how to populate the To header field for a particular
   request in a number of ways.  Usually the user will suggest the To
   header field through a human interface, perhaps inputting the URI
   manually or selecting it from some sort of address book.  Frequently,
   the user will not enter a complete URI, but rather a string of digits
   or letters (for example, "bob").  It is at the discretion of the UA
   to choose how to interpret this input.  Using the string to form the
   user part of a SIP URI implies that the UA wishes the name to be
   resolved in the domain to the right-hand side (RHS) of the at-sign in
   the SIP URI (for instance, sip:bob@example.com).  Using the string to
   form the user part of a SIPS URI implies that the UA wishes to
   communicate securely, and that the name is to be resolved in the
   domain to the RHS of the at-sign.  The RHS will frequently be the
   home domain of the requestor, which allows for the home domain to
   process the outgoing request.  This is useful for features like
   "speed dial" that require interpretation of the user part in the home
   domain.  The tel URL may be used when the UA does not wish to specify
   the domain that should interpret a telephone number that has been
   input by the user.  Rather, each domain through which the request
   passes would be given that opportunity.  As an example, a user in an
   airport might log in and send requests through an outbound proxy in
   the airport.  If they enter "411" (this is the phone number for local
   directory assistance in the United States), that needs to be
   interpreted and processed by the outbound proxy in the airport, not
   the user's home domain.  In this case, tel:411 would be the right
   choice.

   A request outside of a dialog MUST NOT contain a To tag; the tag in
   the To field of a request identifies the peer of the dialog.  Since
   no dialog is established, no tag is present.

   For further information on the To header field, see <a href="#section-20.39">Section 20.39</a>.
   The following is an example of a valid To header field:

      To: Carol &lt;sip:carol@chicago.com&gt;




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<span class="h5"><a class="selflink" id="section-8.1.1.3" href="#section-8.1.1.3">8.1.1.3</a> From</span>

   The From header field indicates the logical identity of the initiator
   of the request, possibly the user's address-of-record.  Like the To
   header field, it contains a URI and optionally a display name.  It is
   used by SIP elements to determine which processing rules to apply to
   a request (for example, automatic call rejection).  As such, it is
   very important that the From URI not contain IP addresses or the FQDN
   of the host on which the UA is running, since these are not logical
   names.

   The From header field allows for a display name.  A UAC SHOULD use
   the display name "Anonymous", along with a syntactically correct, but
   otherwise meaningless URI (like sip:thisis@anonymous.invalid), if the
   identity of the client is to remain hidden.

   Usually, the value that populates the From header field in requests
   generated by a particular UA is pre-provisioned by the user or by the
   administrators of the user's local domain.  If a particular UA is
   used by multiple users, it might have switchable profiles that
   include a URI corresponding to the identity of the profiled user.
   Recipients of requests can authenticate the originator of a request
   in order to ascertain that they are who their From header field
   claims they are (see <a href="#section-22">Section 22</a> for more on authentication).

   The From field MUST contain a new "tag" parameter, chosen by the UAC.
   See <a href="#section-19.3">Section 19.3</a> for details on choosing a tag.

   For further information on the From header field, see <a href="#section-20.20">Section 20.20</a>.
   Examples:

      From: "Bob" &lt;sips:bob@biloxi.com&gt; ;tag=a48s
      From: sip:+12125551212@phone2net.com;tag=887s
      From: Anonymous &lt;sip:c8oqz84zk7z@privacy.org&gt;;tag=hyh8

<span class="h5"><a class="selflink" id="section-8.1.1.4" href="#section-8.1.1.4">8.1.1.4</a> Call-ID</span>

   The Call-ID header field acts as a unique identifier to group
   together a series of messages.  It MUST be the same for all requests
   and responses sent by either UA in a dialog.  It SHOULD be the same
   in each registration from a UA.

   In a new request created by a UAC outside of any dialog, the Call-ID
   header field MUST be selected by the UAC as a globally unique
   identifier over space and time unless overridden by method-specific
   behavior.  All SIP UAs must have a means to guarantee that the Call-
   ID header fields they produce will not be inadvertently generated by
   any other UA.  Note that when requests are retried after certain



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   failure responses that solicit an amendment to a request (for
   example, a challenge for authentication), these retried requests are
   not considered new requests, and therefore do not need new Call-ID
   header fields; see <a href="#section-8.1.3.5">Section 8.1.3.5</a>.

   Use of cryptographically random identifiers (<a href="./rfc1750">RFC 1750</a> [<a href="#ref-12" title="&quot;Randomness Recommendations for Security&quot;">12</a>]) in the
   generation of Call-IDs is RECOMMENDED.  Implementations MAY use the
   form "localid@host".  Call-IDs are case-sensitive and are simply
   compared byte-by-byte.

      Using cryptographically random identifiers provides some
      protection against session hijacking and reduces the likelihood of
      unintentional Call-ID collisions.

   No provisioning or human interface is required for the selection of
   the Call-ID header field value for a request.

   For further information on the Call-ID header field, see <a href="#section-20.8">Section</a>
   <a href="#section-20.8">20.8</a>.

   Example:

      Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com

<span class="h5"><a class="selflink" id="section-8.1.1.5" href="#section-8.1.1.5">8.1.1.5</a> CSeq</span>

   The CSeq header field serves as a way to identify and order
   transactions.  It consists of a sequence number and a method.  The
   method MUST match that of the request.  For non-REGISTER requests
   outside of a dialog, the sequence number value is arbitrary.  The
   sequence number value MUST be expressible as a 32-bit unsigned
   integer and MUST be less than 2**31.  As long as it follows the above
   guidelines, a client may use any mechanism it would like to select
   CSeq header field values.

   <a href="#section-12.2.1.1">Section 12.2.1.1</a> discusses construction of the CSeq for requests
   within a dialog.

   Example:

      CSeq: 4711 INVITE










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<span class="h5"><a class="selflink" id="section-8.1.1.6" href="#section-8.1.1.6">8.1.1.6</a> Max-Forwards</span>

   The Max-Forwards header field serves to limit the number of hops a
   request can transit on the way to its destination.  It consists of an
   integer that is decremented by one at each hop.  If the Max-Forwards
   value reaches 0 before the request reaches its destination, it will
   be rejected with a 483(Too Many Hops) error response.

   A UAC MUST insert a Max-Forwards header field into each request it
   originates with a value that SHOULD be 70.  This number was chosen to
   be sufficiently large to guarantee that a request would not be
   dropped in any SIP network when there were no loops, but not so large
   as to consume proxy resources when a loop does occur.  Lower values
   should be used with caution and only in networks where topologies are
   known by the UA.

<span class="h5"><a class="selflink" id="section-8.1.1.7" href="#section-8.1.1.7">8.1.1.7</a> Via</span>

   The Via header field indicates the transport used for the transaction
   and identifies the location where the response is to be sent.  A Via
   header field value is added only after the transport that will be
   used to reach the next hop has been selected (which may involve the
   usage of the procedures in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>]).

   When the UAC creates a request, it MUST insert a Via into that
   request.  The protocol name and protocol version in the header field
   MUST be SIP and 2.0, respectively.  The Via header field value MUST
   contain a branch parameter.  This parameter is used to identify the
   transaction created by that request.  This parameter is used by both
   the client and the server.

   The branch parameter value MUST be unique across space and time for
   all requests sent by the UA.  The exceptions to this rule are CANCEL
   and ACK for non-2xx responses.  As discussed below, a CANCEL request
   will have the same value of the branch parameter as the request it
   cancels.  As discussed in <a href="#section-17.1.1.3">Section 17.1.1.3</a>, an ACK for a non-2xx
   response will also have the same branch ID as the INVITE whose
   response it acknowledges.

      The uniqueness property of the branch ID parameter, to facilitate
      its use as a transaction ID, was not part of <a href="./rfc2543">RFC 2543</a>.

   The branch ID inserted by an element compliant with this
   specification MUST always begin with the characters "z9hG4bK".  These
   7 characters are used as a magic cookie (7 is deemed sufficient to
   ensure that an older <a href="./rfc2543">RFC 2543</a> implementation would not pick such a
   value), so that servers receiving the request can determine that the
   branch ID was constructed in the fashion described by this



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   specification (that is, globally unique).  Beyond this requirement,
   the precise format of the branch token is implementation-defined.

   The Via header maddr, ttl, and sent-by components will be set when
   the request is processed by the transport layer (<a href="#section-18">Section 18</a>).

   Via processing for proxies is described in <a href="#section-16.6">Section 16.6</a> Item 8 and
   <a href="#section-16.7">Section 16.7</a> Item 3.

<span class="h5"><a class="selflink" id="section-8.1.1.8" href="#section-8.1.1.8">8.1.1.8</a> Contact</span>

   The Contact header field provides a SIP or SIPS URI that can be used
   to contact that specific instance of the UA for subsequent requests.
   The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog.  For the methods defined in this specification, that includes
   only the INVITE request.  For these requests, the scope of the
   Contact is global.  That is, the Contact header field value contains
   the URI at which the UA would like to receive requests, and this URI
   MUST be valid even if used in subsequent requests outside of any
   dialogs.

   If the Request-URI or top Route header field value contains a SIPS
   URI, the Contact header field MUST contain a SIPS URI as well.

   For further information on the Contact header field, see <a href="#section-20.10">Section</a>
   <a href="#section-20.10">20.10</a>.

<span class="h5"><a class="selflink" id="section-8.1.1.9" href="#section-8.1.1.9">8.1.1.9</a> Supported and Require</span>

   If the UAC supports extensions to SIP that can be applied by the
   server to the response, the UAC SHOULD include a Supported header
   field in the request listing the option tags (<a href="#section-19.2">Section 19.2</a>) for those
   extensions.

   The option tags listed MUST only refer to extensions defined in
   standards-track RFCs.  This is to prevent servers from insisting that
   clients implement non-standard, vendor-defined features in order to
   receive service.  Extensions defined by experimental and
   informational RFCs are explicitly excluded from usage with the
   Supported header field in a request, since they too are often used to
   document vendor-defined extensions.

   If the UAC wishes to insist that a UAS understand an extension that
   the UAC will apply to the request in order to process the request, it
   MUST insert a Require header field into the request listing the
   option tag for that extension.  If the UAC wishes to apply an
   extension to the request and insist that any proxies that are



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   traversed understand that extension, it MUST insert a Proxy-Require
   header field into the request listing the option tag for that
   extension.

   As with the Supported header field, the option tags in the Require
   and Proxy-Require header fields MUST only refer to extensions defined
   in standards-track RFCs.

<span class="h5"><a class="selflink" id="section-8.1.1.10" href="#section-8.1.1.10">8.1.1.10</a> Additional Message Components</span>

   After a new request has been created, and the header fields described
   above have been properly constructed, any additional optional header
   fields are added, as are any header fields specific to the method.

   SIP requests MAY contain a MIME-encoded message-body.  Regardless of
   the type of body that a request contains, certain header fields must
   be formulated to characterize the contents of the body.  For further
   information on these header fields, see Sections <a href="#section-20.11">20.11</a> through <a href="#section-20.15">20.15</a>.

<span class="h4"><a class="selflink" id="section-8.1.2" href="#section-8.1.2">8.1.2</a> Sending the Request</span>

   The destination for the request is then computed.  Unless there is
   local policy specifying otherwise, the destination MUST be determined
   by applying the DNS procedures described in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] as follows.  If the
   first element in the route set indicated a strict router (resulting
   in forming the request as described in <a href="#section-12.2.1.1">Section 12.2.1.1</a>), the
   procedures MUST be applied to the Request-URI of the request.
   Otherwise, the procedures are applied to the first Route header field
   value in the request (if one exists), or to the request's Request-URI
   if there is no Route header field present.  These procedures yield an
   ordered set of address, port, and transports to attempt.  Independent
   of which URI is used as input to the procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>], if the
   Request-URI specifies a SIPS resource, the UAC MUST follow the
   procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] as if the input URI were a SIPS URI.

   Local policy MAY specify an alternate set of destinations to attempt.
   If the Request-URI contains a SIPS URI, any alternate destinations
   MUST be contacted with TLS.  Beyond that, there are no restrictions
   on the alternate destinations if the request contains no Route header
   field.  This provides a simple alternative to a pre-existing route
   set as a way to specify an outbound proxy.  However, that approach
   for configuring an outbound proxy is NOT RECOMMENDED; a pre-existing
   route set with a single URI SHOULD be used instead.  If the request
   contains a Route header field, the request SHOULD be sent to the
   locations derived from its topmost value, but MAY be sent to any
   server that the UA is certain will honor the Route and Request-URI
   policies specified in this document (as opposed to those in <a href="./rfc2543">RFC</a>
   <a href="./rfc2543">2543</a>).  In particular, a UAC configured with an outbound proxy SHOULD



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   attempt to send the request to the location indicated in the first
   Route header field value instead of adopting the policy of sending
   all messages to the outbound proxy.

      This ensures that outbound proxies that do not add Record-Route
      header field values will drop out of the path of subsequent
      requests.  It allows endpoints that cannot resolve the first Route
      URI to delegate that task to an outbound proxy.

   The UAC SHOULD follow the procedures defined in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] for stateful
   elements, trying each address until a server is contacted.  Each try
   constitutes a new transaction, and therefore each carries a different
   topmost Via header field value with a new branch parameter.
   Furthermore, the transport value in the Via header field is set to
   whatever transport was determined for the target server.

<span class="h4"><a class="selflink" id="section-8.1.3" href="#section-8.1.3">8.1.3</a> Processing Responses</span>

   Responses are first processed by the transport layer and then passed
   up to the transaction layer.  The transaction layer performs its
   processing and then passes the response up to the TU.  The majority
   of response processing in the TU is method specific.  However, there
   are some general behaviors independent of the method.

<span class="h5"><a class="selflink" id="section-8.1.3.1" href="#section-8.1.3.1">8.1.3.1</a> Transaction Layer Errors</span>

   In some cases, the response returned by the transaction layer will
   not be a SIP message, but rather a transaction layer error.  When a
   timeout error is received from the transaction layer, it MUST be
   treated as if a 408 (Request Timeout) status code has been received.
   If a fatal transport error is reported by the transport layer
   (generally, due to fatal ICMP errors in UDP or connection failures in
   TCP), the condition MUST be treated as a 503 (Service Unavailable)
   status code.

<span class="h5"><a class="selflink" id="section-8.1.3.2" href="#section-8.1.3.2">8.1.3.2</a> Unrecognized Responses</span>

   A UAC MUST treat any final response it does not recognize as being
   equivalent to the x00 response code of that class, and MUST be able
   to process the x00 response code for all classes.  For example, if a
   UAC receives an unrecognized response code of 431, it can safely
   assume that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code.  A
   UAC MUST treat any provisional response different than 100 that it
   does not recognize as 183 (Session Progress).  A UAC MUST be able to
   process 100 and 183 responses.





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<span class="h5"><a class="selflink" id="section-8.1.3.3" href="#section-8.1.3.3">8.1.3.3</a> Vias</span>

   If more than one Via header field value is present in a response, the
   UAC SHOULD discard the message.

      The presence of additional Via header field values that precede
      the originator of the request suggests that the message was
      misrouted or possibly corrupted.

<span class="h5"><a class="selflink" id="section-8.1.3.4" href="#section-8.1.3.4">8.1.3.4</a> Processing 3xx Responses</span>

   Upon receipt of a redirection response (for example, a 301 response
   status code), clients SHOULD use the URI(s) in the Contact header
   field to formulate one or more new requests based on the redirected
   request.  This process is similar to that of a proxy recursing on a
   3xx class response as detailed in Sections <a href="#section-16.5">16.5</a> and <a href="#section-16.6">16.6</a>.  A client
   starts with an initial target set containing exactly one URI, the
   Request-URI of the original request.  If a client wishes to formulate
   new requests based on a 3xx class response to that request, it places
   the URIs to try into the target set.  Subject to the restrictions in
   this specification, a client can choose which Contact URIs it places
   into the target set.  As with proxy recursion, a client processing
   3xx class responses MUST NOT add any given URI to the target set more
   than once.  If the original request had a SIPS URI in the Request-
   URI, the client MAY choose to recurse to a non-SIPS URI, but SHOULD
   inform the user of the redirection to an insecure URI.

      Any new request may receive 3xx responses themselves containing
      the original URI as a contact.  Two locations can be configured to
      redirect to each other.  Placing any given URI in the target set
      only once prevents infinite redirection loops.

   As the target set grows, the client MAY generate new requests to the
   URIs in any order.  A common mechanism is to order the set by the "q"
   parameter value from the Contact header field value.  Requests to the
   URIs MAY be generated serially or in parallel.  One approach is to
   process groups of decreasing q-values serially and process the URIs
   in each q-value group in parallel.  Another is to perform only serial
   processing in decreasing q-value order, arbitrarily choosing between
   contacts of equal q-value.

   If contacting an address in the list results in a failure, as defined
   in the next paragraph, the element moves to the next address in the
   list, until the list is exhausted.  If the list is exhausted, then
   the request has failed.






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   Failures SHOULD be detected through failure response codes (codes
   greater than 399); for network errors the client transaction will
   report any transport layer failures to the transaction user.  Note
   that some response codes (detailed in 8.1.3.5) indicate that the
   request can be retried; requests that are reattempted should not be
   considered failures.

   When a failure for a particular contact address is received, the
   client SHOULD try the next contact address.  This will involve
   creating a new client transaction to deliver a new request.

   In order to create a request based on a contact address in a 3xx
   response, a UAC MUST copy the entire URI from the target set into the
   Request-URI, except for the "method-param" and "header" URI
   parameters (see <a href="#section-19.1.1">Section 19.1.1</a> for a definition of these parameters).
   It uses the "header" parameters to create header field values for the
   new request, overwriting header field values associated with the
   redirected request in accordance with the guidelines in <a href="#section-19.1.5">Section</a>
   <a href="#section-19.1.5">19.1.5</a>.

   Note that in some instances, header fields that have been
   communicated in the contact address may instead append to existing
   request header fields in the original redirected request.  As a
   general rule, if the header field can accept a comma-separated list
   of values, then the new header field value MAY be appended to any
   existing values in the original redirected request.  If the header
   field does not accept multiple values, the value in the original
   redirected request MAY be overwritten by the header field value
   communicated in the contact address.  For example, if a contact
   address is returned with the following value:

      sip:user@host?Subject=foo&amp;Call-Info=&lt;<a href="http://www.foo.com">http://www.foo.com</a>&gt;

   Then any Subject header field in the original redirected request is
   overwritten, but the HTTP URL is merely appended to any existing
   Call-Info header field values.

   It is RECOMMENDED that the UAC reuse the same To, From, and Call-ID
   used in the original redirected request, but the UAC MAY also choose
   to update the Call-ID header field value for new requests, for
   example.

   Finally, once the new request has been constructed, it is sent using
   a new client transaction, and therefore MUST have a new branch ID in
   the top Via field as discussed in <a href="#section-8.1.1.7">Section 8.1.1.7</a>.






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   In all other respects, requests sent upon receipt of a redirect
   response SHOULD re-use the header fields and bodies of the original
   request.

   In some instances, Contact header field values may be cached at UAC
   temporarily or permanently depending on the status code received and
   the presence of an expiration interval; see Sections <a href="#section-21.3.2">21.3.2</a> and
   21.3.3.

<span class="h5"><a class="selflink" id="section-8.1.3.5" href="#section-8.1.3.5">8.1.3.5</a> Processing 4xx Responses</span>

   Certain 4xx response codes require specific UA processing,
   independent of the method.

   If a 401 (Unauthorized) or 407 (Proxy Authentication Required)
   response is received, the UAC SHOULD follow the authorization
   procedures of <a href="#section-22.2">Section 22.2</a> and <a href="#section-22.3">Section 22.3</a> to retry the request with
   credentials.

   If a 413 (Request Entity Too Large) response is received (<a href="#section-21.4.11">Section</a>
   <a href="#section-21.4.11">21.4.11</a>), the request contained a body that was longer than the UAS
   was willing to accept.  If possible, the UAC SHOULD retry the
   request, either omitting the body or using one of a smaller length.

   If a 415 (Unsupported Media Type) response is received (<a href="#section-21.4.13">Section</a>
   <a href="#section-21.4.13">21.4.13</a>), the request contained media types not supported by the UAS.
   The UAC SHOULD retry sending the request, this time only using
   content with types listed in the Accept header field in the response,
   with encodings listed in the Accept-Encoding header field in the
   response, and with languages listed in the Accept-Language in the
   response.

   If a 416 (Unsupported URI Scheme) response is received (<a href="#section-21.4.14">Section</a>
   <a href="#section-21.4.14">21.4.14</a>), the Request-URI used a URI scheme not supported by the
   server.  The client SHOULD retry the request, this time, using a SIP
   URI.

   If a 420 (Bad Extension) response is received (<a href="#section-21.4.15">Section 21.4.15</a>), the
   request contained a Require or Proxy-Require header field listing an
   option-tag for a feature not supported by a proxy or UAS.  The UAC
   SHOULD retry the request, this time omitting any extensions listed in
   the Unsupported header field in the response.

   In all of the above cases, the request is retried by creating a new
   request with the appropriate modifications.  This new request
   constitutes a new transaction and SHOULD have the same value of the
   Call-ID, To, and From of the previous request, but the CSeq should
   contain a new sequence number that is one higher than the previous.



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   With other 4xx responses, including those yet to be defined, a retry
   may or may not be possible depending on the method and the use case.

<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a> UAS Behavior</span>

   When a request outside of a dialog is processed by a UAS, there is a
   set of processing rules that are followed, independent of the method.
   <a href="#section-12">Section 12</a> gives guidance on how a UAS can tell whether a request is
   inside or outside of a dialog.

   Note that request processing is atomic.  If a request is accepted,
   all state changes associated with it MUST be performed.  If it is
   rejected, all state changes MUST NOT be performed.

   UASs SHOULD process the requests in the order of the steps that
   follow in this section (that is, starting with authentication, then
   inspecting the method, the header fields, and so on throughout the
   remainder of this section).

<span class="h4"><a class="selflink" id="section-8.2.1" href="#section-8.2.1">8.2.1</a> Method Inspection</span>

   Once a request is authenticated (or authentication is skipped), the
   UAS MUST inspect the method of the request.  If the UAS recognizes
   but does not support the method of a request, it MUST generate a 405
   (Method Not Allowed) response.  Procedures for generating responses
   are described in <a href="#section-8.2.6">Section 8.2.6</a>.  The UAS MUST also add an Allow
   header field to the 405 (Method Not Allowed) response.  The Allow
   header field MUST list the set of methods supported by the UAS
   generating the message.  The Allow header field is presented in
   <a href="#section-20.5">Section 20.5</a>.

   If the method is one supported by the server, processing continues.

<span class="h4"><a class="selflink" id="section-8.2.2" href="#section-8.2.2">8.2.2</a> Header Inspection</span>

   If a UAS does not understand a header field in a request (that is,
   the header field is not defined in this specification or in any
   supported extension), the server MUST ignore that header field and
   continue processing the message.  A UAS SHOULD ignore any malformed
   header fields that are not necessary for processing requests.

<span class="h5"><a class="selflink" id="section-8.2.2.1" href="#section-8.2.2.1">8.2.2.1</a> To and Request-URI</span>

   The To header field identifies the original recipient of the request
   designated by the user identified in the From field.  The original
   recipient may or may not be the UAS processing the request, due to
   call forwarding or other proxy operations.  A UAS MAY apply any
   policy it wishes to determine whether to accept requests when the To



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   header field is not the identity of the UAS.  However, it is
   RECOMMENDED that a UAS accept requests even if they do not recognize
   the URI scheme (for example, a tel: URI) in the To header field, or
   if the To header field does not address a known or current user of
   this UAS.  If, on the other hand, the UAS decides to reject the
   request, it SHOULD generate a response with a 403 (Forbidden) status
   code and pass it to the server transaction for transmission.

   However, the Request-URI identifies the UAS that is to process the
   request.  If the Request-URI uses a scheme not supported by the UAS,
   it SHOULD reject the request with a 416 (Unsupported URI Scheme)
   response.  If the Request-URI does not identify an address that the
   UAS is willing to accept requests for, it SHOULD reject the request
   with a 404 (Not Found) response.  Typically, a UA that uses the
   REGISTER method to bind its address-of-record to a specific contact
   address will see requests whose Request-URI equals that contact
   address.  Other potential sources of received Request-URIs include
   the Contact header fields of requests and responses sent by the UA
   that establish or refresh dialogs.

<span class="h5"><a class="selflink" id="section-8.2.2.2" href="#section-8.2.2.2">8.2.2.2</a> Merged Requests</span>

   If the request has no tag in the To header field, the UAS core MUST
   check the request against ongoing transactions.  If the From tag,
   Call-ID, and CSeq exactly match those associated with an ongoing
   transaction, but the request does not match that transaction (based
   on the matching rules in <a href="#section-17.2.3">Section 17.2.3</a>), the UAS core SHOULD
   generate a 482 (Loop Detected) response and pass it to the server
   transaction.

      The same request has arrived at the UAS more than once, following
      different paths, most likely due to forking.  The UAS processes
      the first such request received and responds with a 482 (Loop
      Detected) to the rest of them.

<span class="h5"><a class="selflink" id="section-8.2.2.3" href="#section-8.2.2.3">8.2.2.3</a> Require</span>

   Assuming the UAS decides that it is the proper element to process the
   request, it examines the Require header field, if present.

   The Require header field is used by a UAC to tell a UAS about SIP
   extensions that the UAC expects the UAS to support in order to
   process the request properly.  Its format is described in <a href="#section-20.32">Section</a>
   <a href="#section-20.32">20.32</a>.  If a UAS does not understand an option-tag listed in a
   Require header field, it MUST respond by generating a response with
   status code 420 (Bad Extension).  The UAS MUST add an Unsupported
   header field, and list in it those options it does not understand
   amongst those in the Require header field of the request.



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   Note that Require and Proxy-Require MUST NOT be used in a SIP CANCEL
   request, or in an ACK request sent for a non-2xx response.  These
   header fields MUST be ignored if they are present in these requests.

   An ACK request for a 2xx response MUST contain only those Require and
   Proxy-Require values that were present in the initial request.

   Example:

      UAC-&gt;UAS:   INVITE sip:watson@bell-telephone.com SIP/2.0
                  Require: 100rel

      UAS-&gt;UAC:   SIP/2.0 420 Bad Extension
                  Unsupported: 100rel

      This behavior ensures that the client-server interaction will
      proceed without delay when all options are understood by both
      sides, and only slow down if options are not understood (as in the
      example above).  For a well-matched client-server pair, the
      interaction proceeds quickly, saving a round-trip often required
      by negotiation mechanisms.  In addition, it also removes ambiguity
      when the client requires features that the server does not
      understand.  Some features, such as call handling fields, are only
      of interest to end systems.

<span class="h4"><a class="selflink" id="section-8.2.3" href="#section-8.2.3">8.2.3</a> Content Processing</span>

   Assuming the UAS understands any extensions required by the client,
   the UAS examines the body of the message, and the header fields that
   describe it.  If there are any bodies whose type (indicated by the
   Content-Type), language (indicated by the Content-Language) or
   encoding (indicated by the Content-Encoding) are not understood, and
   that body part is not optional (as indicated by the Content-
   Disposition header field), the UAS MUST reject the request with a 415
   (Unsupported Media Type) response.  The response MUST contain an
   Accept header field listing the types of all bodies it understands,
   in the event the request contained bodies of types not supported by
   the UAS.  If the request contained content encodings not understood
   by the UAS, the response MUST contain an Accept-Encoding header field
   listing the encodings understood by the UAS.  If the request
   contained content with languages not understood by the UAS, the
   response MUST contain an Accept-Language header field indicating the
   languages understood by the UAS.  Beyond these checks, body handling
   depends on the method and type.  For further information on the
   processing of content-specific header fields, see <a href="#section-7.4">Section 7.4</a> as well
   as <a href="#section-20.11">Section 20.11</a> through 20.15.





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<span class="h4"><a class="selflink" id="section-8.2.4" href="#section-8.2.4">8.2.4</a> Applying Extensions</span>

   A UAS that wishes to apply some extension when generating the
   response MUST NOT do so unless support for that extension is
   indicated in the Supported header field in the request.  If the
   desired extension is not supported, the server SHOULD rely only on
   baseline SIP and any other extensions supported by the client.  In
   rare circumstances, where the server cannot process the request
   without the extension, the server MAY send a 421 (Extension Required)
   response.  This response indicates that the proper response cannot be
   generated without support of a specific extension.  The needed
   extension(s) MUST be included in a Require header field in the
   response.  This behavior is NOT RECOMMENDED, as it will generally
   break interoperability.

   Any extensions applied to a non-421 response MUST be listed in a
   Require header field included in the response.  Of course, the server
   MUST NOT apply extensions not listed in the Supported header field in
   the request.  As a result of this, the Require header field in a
   response will only ever contain option tags defined in standards-
   track RFCs.

<span class="h4"><a class="selflink" id="section-8.2.5" href="#section-8.2.5">8.2.5</a> Processing the Request</span>

   Assuming all of the checks in the previous subsections are passed,
   the UAS processing becomes method-specific.  <a href="#section-10">Section 10</a> covers the
   REGISTER request, <a href="#section-11">Section 11</a> covers the OPTIONS request, <a href="#section-13">Section 13</a>
   covers the INVITE request, and <a href="#section-15">Section 15</a> covers the BYE request.

<span class="h4"><a class="selflink" id="section-8.2.6" href="#section-8.2.6">8.2.6</a> Generating the Response</span>

   When a UAS wishes to construct a response to a request, it follows
   the general procedures detailed in the following subsections.
   Additional behaviors specific to the response code in question, which
   are not detailed in this section, may also be required.

   Once all procedures associated with the creation of a response have
   been completed, the UAS hands the response back to the server
   transaction from which it received the request.

<span class="h5"><a class="selflink" id="section-8.2.6.1" href="#section-8.2.6.1">8.2.6.1</a> Sending a Provisional Response</span>

   One largely non-method-specific guideline for the generation of
   responses is that UASs SHOULD NOT issue a provisional response for a
   non-INVITE request.  Rather, UASs SHOULD generate a final response to
   a non-INVITE request as soon as possible.





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   When a 100 (Trying) response is generated, any Timestamp header field
   present in the request MUST be copied into this 100 (Trying)
   response.  If there is a delay in generating the response, the UAS
   SHOULD add a delay value into the Timestamp value in the response.
   This value MUST contain the difference between the time of sending of
   the response and receipt of the request, measured in seconds.

<span class="h5"><a class="selflink" id="section-8.2.6.2" href="#section-8.2.6.2">8.2.6.2</a> Headers and Tags</span>

   The From field of the response MUST equal the From header field of
   the request.  The Call-ID header field of the response MUST equal the
   Call-ID header field of the request.  The CSeq header field of the
   response MUST equal the CSeq field of the request.  The Via header
   field values in the response MUST equal the Via header field values
   in the request and MUST maintain the same ordering.

   If a request contained a To tag in the request, the To header field
   in the response MUST equal that of the request.  However, if the To
   header field in the request did not contain a tag, the URI in the To
   header field in the response MUST equal the URI in the To header
   field; additionally, the UAS MUST add a tag to the To header field in
   the response (with the exception of the 100 (Trying) response, in
   which a tag MAY be present).  This serves to identify the UAS that is
   responding, possibly resulting in a component of a dialog ID.  The
   same tag MUST be used for all responses to that request, both final
   and provisional (again excepting the 100 (Trying)).  Procedures for
   the generation of tags are defined in <a href="#section-19.3">Section 19.3</a>.

<span class="h4"><a class="selflink" id="section-8.2.7" href="#section-8.2.7">8.2.7</a> Stateless UAS Behavior</span>

   A stateless UAS is a UAS that does not maintain transaction state.
   It replies to requests normally, but discards any state that would
   ordinarily be retained by a UAS after a response has been sent.  If a
   stateless UAS receives a retransmission of a request, it regenerates
   the response and resends it, just as if it were replying to the first
   instance of the request. A UAS cannot be stateless unless the request
   processing for that method would always result in the same response
   if the requests are identical. This rules out stateless registrars,
   for example.  Stateless UASs do not use a transaction layer; they
   receive requests directly from the transport layer and send responses
   directly to the transport layer.

   The stateless UAS role is needed primarily to handle unauthenticated
   requests for which a challenge response is issued.  If
   unauthenticated requests were handled statefully, then malicious
   floods of unauthenticated requests could create massive amounts of





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   transaction state that might slow or completely halt call processing
   in a UAS, effectively creating a denial of service condition; for
   more information see <a href="#section-26.1.5">Section 26.1.5</a>.

   The most important behaviors of a stateless UAS are the following:

      o  A stateless UAS MUST NOT send provisional (1xx) responses.

      o  A stateless UAS MUST NOT retransmit responses.

      o  A stateless UAS MUST ignore ACK requests.

      o  A stateless UAS MUST ignore CANCEL requests.

      o  To header tags MUST be generated for responses in a stateless
         manner - in a manner that will generate the same tag for the
         same request consistently.  For information on tag construction
         see <a href="#section-19.3">Section 19.3</a>.

   In all other respects, a stateless UAS behaves in the same manner as
   a stateful UAS.  A UAS can operate in either a stateful or stateless
   mode for each new request.

<span class="h3"><a class="selflink" id="section-8.3" href="#section-8.3">8.3</a> Redirect Servers</span>

   In some architectures it may be desirable to reduce the processing
   load on proxy servers that are responsible for routing requests, and
   improve signaling path robustness, by relying on redirection.

   Redirection allows servers to push routing information for a request
   back in a response to the client, thereby taking themselves out of
   the loop of further messaging for this transaction while still aiding
   in locating the target of the request.  When the originator of the
   request receives the redirection, it will send a new request based on
   the URI(s) it has received.  By propagating URIs from the core of the
   network to its edges, redirection allows for considerable network
   scalability.

   A redirect server is logically constituted of a server transaction
   layer and a transaction user that has access to a location service of
   some kind (see <a href="#section-10">Section 10</a> for more on registrars and location
   services).  This location service is effectively a database
   containing mappings between a single URI and a set of one or more
   alternative locations at which the target of that URI can be found.

   A redirect server does not issue any SIP requests of its own.  After
   receiving a request other than CANCEL, the server either refuses the
   request or gathers the list of alternative locations from the



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   location service and returns a final response of class 3xx.  For
   well-formed CANCEL requests, it SHOULD return a 2xx response.  This
   response ends the SIP transaction.  The redirect server maintains
   transaction state for an entire SIP transaction.  It is the
   responsibility of clients to detect forwarding loops between redirect
   servers.

   When a redirect server returns a 3xx response to a request, it
   populates the list of (one or more) alternative locations into the
   Contact header field.  An "expires" parameter to the Contact header
   field values may also be supplied to indicate the lifetime of the
   Contact data.

   The Contact header field contains URIs giving the new locations or
   user names to try, or may simply specify additional transport
   parameters.  A 301 (Moved Permanently) or 302 (Moved Temporarily)
   response may also give the same location and username that was
   targeted by the initial request but specify additional transport
   parameters such as a different server or multicast address to try, or
   a change of SIP transport from UDP to TCP or vice versa.

   However, redirect servers MUST NOT redirect a request to a URI equal
   to the one in the Request-URI; instead, provided that the URI does
   not point to itself, the server MAY proxy the request to the
   destination URI, or MAY reject it with a 404.

      If a client is using an outbound proxy, and that proxy actually
      redirects requests, a potential arises for infinite redirection
      loops.

   Note that a Contact header field value MAY also refer to a different
   resource than the one originally called.  For example, a SIP call
   connected to PSTN gateway may need to deliver a special informational
   announcement such as "The number you have dialed has been changed."

   A Contact response header field can contain any suitable URI
   indicating where the called party can be reached, not limited to SIP
   URIs.  For example, it could contain URIs for phones, fax, or irc (if
   they were defined) or a mailto:  (<a href="./rfc2368">RFC 2368</a> [<a href="#ref-32" title="&quot;The mailto URL scheme&quot;">32</a>]) URL.  <a href="#section-26.4.4">Section 26.4.4</a>
   discusses implications and limitations of redirecting a SIPS URI to a
   non-SIPS URI.

   The "expires" parameter of a Contact header field value indicates how
   long the URI is valid.  The value of the parameter is a number
   indicating seconds.  If this parameter is not provided, the value of
   the Expires header field determines how long the URI is valid.
   Malformed values SHOULD be treated as equivalent to 3600.




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      This provides a modest level of backwards compatibility with <a href="./rfc2543">RFC</a>
      <a href="./rfc2543">2543</a>, which allowed absolute times in this header field.  If an
      absolute time is received, it will be treated as malformed, and
      then default to 3600.

   Redirect servers MUST ignore features that are not understood
   (including unrecognized header fields, any unknown option tags in
   Require, or even method names) and proceed with the redirection of
   the request in question.

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a> Canceling a Request</span>

   The previous section has discussed general UA behavior for generating
   requests and processing responses for requests of all methods.  In
   this section, we discuss a general purpose method, called CANCEL.

   The CANCEL request, as the name implies, is used to cancel a previous
   request sent by a client.  Specifically, it asks the UAS to cease
   processing the request and to generate an error response to that
   request.  CANCEL has no effect on a request to which a UAS has
   already given a final response.  Because of this, it is most useful
   to CANCEL requests to which it can take a server long time to
   respond.  For this reason, CANCEL is best for INVITE requests, which
   can take a long time to generate a response.  In that usage, a UAS
   that receives a CANCEL request for an INVITE, but has not yet sent a
   final response, would "stop ringing", and then respond to the INVITE
   with a specific error response (a 487).

   CANCEL requests can be constructed and sent by both proxies and user
   agent clients.  <a href="#section-15">Section 15</a> discusses under what conditions a UAC
   would CANCEL an INVITE request, and <a href="#section-16.10">Section 16.10</a> discusses proxy
   usage of CANCEL.

   A stateful proxy responds to a CANCEL, rather than simply forwarding
   a response it would receive from a downstream element.  For that
   reason, CANCEL is referred to as a "hop-by-hop" request, since it is
   responded to at each stateful proxy hop.

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a> Client Behavior</span>

   A CANCEL request SHOULD NOT be sent to cancel a request other than
   INVITE.

      Since requests other than INVITE are responded to immediately,
      sending a CANCEL for a non-INVITE request would always create a
      race condition.





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   The following procedures are used to construct a CANCEL request.  The
   Request-URI, Call-ID, To, the numeric part of CSeq, and From header
   fields in the CANCEL request MUST be identical to those in the
   request being cancelled, including tags.  A CANCEL constructed by a
   client MUST have only a single Via header field value matching the
   top Via value in the request being cancelled.  Using the same values
   for these header fields allows the CANCEL to be matched with the
   request it cancels (<a href="#section-9.2">Section 9.2</a> indicates how such matching occurs).
   However, the method part of the CSeq header field MUST have a value
   of CANCEL.  This allows it to be identified and processed as a
   transaction in its own right (See <a href="#section-17">Section 17</a>).

   If the request being cancelled contains a Route header field, the
   CANCEL request MUST include that Route header field's values.

      This is needed so that stateless proxies are able to route CANCEL
      requests properly.

   The CANCEL request MUST NOT contain any Require or Proxy-Require
   header fields.

   Once the CANCEL is constructed, the client SHOULD check whether it
   has received any response (provisional or final) for the request
   being cancelled (herein referred to as the "original request").

   If no provisional response has been received, the CANCEL request MUST
   NOT be sent; rather, the client MUST wait for the arrival of a
   provisional response before sending the request.  If the original
   request has generated a final response, the CANCEL SHOULD NOT be
   sent, as it is an effective no-op, since CANCEL has no effect on
   requests that have already generated a final response.  When the
   client decides to send the CANCEL, it creates a client transaction
   for the CANCEL and passes it the CANCEL request along with the
   destination address, port, and transport.  The destination address,
   port, and transport for the CANCEL MUST be identical to those used to
   send the original request.

      If it was allowed to send the CANCEL before receiving a response
      for the previous request, the server could receive the CANCEL
      before the original request.

   Note that both the transaction corresponding to the original request
   and the CANCEL transaction will complete independently.  However, a
   UAC canceling a request cannot rely on receiving a 487 (Request
   Terminated) response for the original request, as an <a href="./rfc2543">RFC 2543</a>-
   compliant UAS will not generate such a response.  If there is no
   final response for the original request in 64*T1 seconds (T1 is




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   defined in <a href="#section-17.1.1.1">Section 17.1.1.1</a>), the client SHOULD then consider the
   original transaction cancelled and SHOULD destroy the client
   transaction handling the original request.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a> Server Behavior</span>

   The CANCEL method requests that the TU at the server side cancel a
   pending transaction.  The TU determines the transaction to be
   cancelled by taking the CANCEL request, and then assuming that the
   request method is anything but CANCEL or ACK and applying the
   transaction matching procedures of <a href="#section-17.2.3">Section 17.2.3</a>.  The matching
   transaction is the one to be cancelled.

   The processing of a CANCEL request at a server depends on the type of
   server.  A stateless proxy will forward it, a stateful proxy might
   respond to it and generate some CANCEL requests of its own, and a UAS
   will respond to it.  See <a href="#section-16.10">Section 16.10</a> for proxy treatment of CANCEL.

   A UAS first processes the CANCEL request according to the general UAS
   processing described in <a href="#section-8.2">Section 8.2</a>.  However, since CANCEL requests
   are hop-by-hop and cannot be resubmitted, they cannot be challenged
   by the server in order to get proper credentials in an Authorization
   header field.  Note also that CANCEL requests do not contain a
   Require header field.

   If the UAS did not find a matching transaction for the CANCEL
   according to the procedure above, it SHOULD respond to the CANCEL
   with a 481 (Call Leg/Transaction Does Not Exist).  If the transaction
   for the original request still exists, the behavior of the UAS on
   receiving a CANCEL request depends on whether it has already sent a
   final response for the original request.  If it has, the CANCEL
   request has no effect on the processing of the original request, no
   effect on any session state, and no effect on the responses generated
   for the original request.  If the UAS has not issued a final response
   for the original request, its behavior depends on the method of the
   original request.  If the original request was an INVITE, the UAS
   SHOULD immediately respond to the INVITE with a 487 (Request
   Terminated).  A CANCEL request has no impact on the processing of
   transactions with any other method defined in this specification.

   Regardless of the method of the original request, as long as the
   CANCEL matched an existing transaction, the UAS answers the CANCEL
   request itself with a 200 (OK) response.  This response is
   constructed following the procedures described in <a href="#section-8.2.6">Section 8.2.6</a>
   noting that the To tag of the response to the CANCEL and the To tag
   in the response to the original request SHOULD be the same.  The
   response to CANCEL is passed to the server transaction for
   transmission.



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<span class="h2"><a class="selflink" id="section-10" href="#section-10">10</a> Registrations</span>

<span class="h3"><a class="selflink" id="section-10.1" href="#section-10.1">10.1</a> Overview</span>

   SIP offers a discovery capability.  If a user wants to initiate a
   session with another user, SIP must discover the current host(s) at
   which the destination user is reachable.  This discovery process is
   frequently accomplished by SIP network elements such as proxy servers
   and redirect servers which are responsible for receiving a request,
   determining where to send it based on knowledge of the location of
   the user, and then sending it there.  To do this, SIP network
   elements consult an abstract service known as a location service,
   which provides address bindings for a particular domain.  These
   address bindings map an incoming SIP or SIPS URI, sip:bob@biloxi.com,
   for example, to one or more URIs that are somehow "closer" to the
   desired user, sip:bob@engineering.biloxi.com, for example.
   Ultimately, a proxy will consult a location service that maps a
   received URI to the user agent(s) at which the desired recipient is
   currently residing.

   Registration creates bindings in a location service for a particular
   domain that associates an address-of-record URI with one or more
   contact addresses.  Thus, when a proxy for that domain receives a
   request whose Request-URI matches the address-of-record, the proxy
   will forward the request to the contact addresses registered to that
   address-of-record.  Generally, it only makes sense to register an
   address-of-record at a domain's location service when requests for
   that address-of-record would be routed to that domain.  In most
   cases, this means that the domain of the registration will need to
   match the domain in the URI of the address-of-record.

   There are many ways by which the contents of the location service can
   be established.  One way is administratively.  In the above example,
   Bob is known to be a member of the engineering department through
   access to a corporate database.  However, SIP provides a mechanism
   for a UA to create a binding explicitly.  This mechanism is known as
   registration.

   Registration entails sending a REGISTER request to a special type of
   UAS known as a registrar.  A registrar acts as the front end to the
   location service for a domain, reading and writing mappings based on
   the contents of REGISTER requests.  This location service is then
   typically consulted by a proxy server that is responsible for routing
   requests for that domain.

   An illustration of the overall registration process is given in
   Figure 2.  Note that the registrar and proxy server are logical roles
   that can be played by a single device in a network; for purposes of



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   clarity the two are separated in this illustration.  Also note that
   UAs may send requests through a proxy server in order to reach a
   registrar if the two are separate elements.

   SIP does not mandate a particular mechanism for implementing the
   location service.  The only requirement is that a registrar for some
   domain MUST be able to read and write data to the location service,
   and a proxy or a redirect server for that domain MUST be capable of
   reading that same data.  A registrar MAY be co-located with a
   particular SIP proxy server for the same domain.

<span class="h3"><a class="selflink" id="section-10.2" href="#section-10.2">10.2</a> Constructing the REGISTER Request</span>

   REGISTER requests add, remove, and query bindings.  A REGISTER
   request can add a new binding between an address-of-record and one or
   more contact addresses.  Registration on behalf of a particular
   address-of-record can be performed by a suitably authorized third
   party.  A client can also remove previous bindings or query to
   determine which bindings are currently in place for an address-of-
   record.

   Except as noted, the construction of the REGISTER request and the
   behavior of clients sending a REGISTER request is identical to the
   general UAC behavior described in <a href="#section-8.1">Section 8.1</a> and <a href="#section-17.1">Section 17.1</a>.

   A REGISTER request does not establish a dialog.  A UAC MAY include a
   Route header field in a REGISTER request based on a pre-existing
   route set as described in <a href="#section-8.1">Section 8.1</a>.  The Record-Route header field
   has no meaning in REGISTER requests or responses, and MUST be ignored
   if present.  In particular, the UAC MUST NOT create a new route set
   based on the presence or absence of a Record-Route header field in
   any response to a REGISTER request.

   The following header fields, except Contact, MUST be included in a
   REGISTER request.  A Contact header field MAY be included:

      Request-URI: The Request-URI names the domain of the location
           service for which the registration is meant (for example,
           "sip:chicago.com").  The "userinfo" and "@" components of the
           SIP URI MUST NOT be present.

      To: The To header field contains the address of record whose
           registration is to be created, queried, or modified.  The To
           header field and the Request-URI field typically differ, as
           the former contains a user name.  This address-of-record MUST
           be a SIP URI or SIPS URI.





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      From: The From header field contains the address-of-record of the
           person responsible for the registration.  The value is the
           same as the To header field unless the request is a third-
           party registration.

      Call-ID: All registrations from a UAC SHOULD use the same Call-ID
           header field value for registrations sent to a particular
           registrar.

           If the same client were to use different Call-ID values, a
           registrar could not detect whether a delayed REGISTER request
           might have arrived out of order.

      CSeq: The CSeq value guarantees proper ordering of REGISTER
           requests.  A UA MUST increment the CSeq value by one for each
           REGISTER request with the same Call-ID.

      Contact: REGISTER requests MAY contain a Contact header field with
           zero or more values containing address bindings.

   UAs MUST NOT send a new registration (that is, containing new Contact
   header field values, as opposed to a retransmission) until they have
   received a final response from the registrar for the previous one or
   the previous REGISTER request has timed out.



























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                                                 bob
                                               +----+
                                               | UA |
                                               |    |
                                               +----+
                                                  |
                                                  |3)INVITE
                                                  |   carol@chicago.com
         chicago.com        +--------+            V
         +---------+ 2)Store|Location|4)Query +-----+
         |Registrar|=======&gt;| Service|&lt;=======|Proxy|sip.chicago.com
         +---------+        +--------+=======&gt;+-----+
               A                      5)Resp      |
               |                                  |
               |                                  |
     1)REGISTER|                                  |
               |                                  |
            +----+                                |
            | UA |&lt;-------------------------------+
   cube2214a|    |                            6)INVITE
            +----+                    carol@cube2214a.chicago.com
             carol

                      Figure 2: REGISTER example

      The following Contact header parameters have a special meaning in
           REGISTER requests:

      action: The "action" parameter from <a href="./rfc2543">RFC 2543</a> has been deprecated.
           UACs SHOULD NOT use the "action" parameter.

      expires: The "expires" parameter indicates how long the UA would
           like the binding to be valid.  The value is a number
           indicating seconds.  If this parameter is not provided, the
           value of the Expires header field is used instead.
           Implementations MAY treat values larger than 2**32-1
           (4294967295 seconds or 136 years) as equivalent to 2**32-1.
           Malformed values SHOULD be treated as equivalent to 3600.

<span class="h4"><a class="selflink" id="section-10.2.1" href="#section-10.2.1">10.2.1</a> Adding Bindings</span>

   The REGISTER request sent to a registrar includes the contact
   address(es) to which SIP requests for the address-of-record should be
   forwarded.  The address-of-record is included in the To header field
   of the REGISTER request.






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   The Contact header field values of the request typically consist of
   SIP or SIPS URIs that identify particular SIP endpoints (for example,
   "sip:carol@cube2214a.chicago.com"), but they MAY use any URI scheme.
   A SIP UA can choose to register telephone numbers (with the tel URL,
   <a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>]) or email addresses (with a mailto URL, <a href="./rfc2368">RFC 2368</a> [<a href="#ref-32" title="&quot;The mailto URL scheme&quot;">32</a>])
   as Contacts for an address-of-record, for example.

   For example, Carol, with address-of-record "sip:carol@chicago.com",
   would register with the SIP registrar of the domain chicago.com.  Her
   registrations would then be used by a proxy server in the chicago.com
   domain to route requests for Carol's address-of-record to her SIP
   endpoint.

   Once a client has established bindings at a registrar, it MAY send
   subsequent registrations containing new bindings or modifications to
   existing bindings as necessary.  The 2xx response to the REGISTER
   request will contain, in a Contact header field, a complete list of
   bindings that have been registered for this address-of-record at this
   registrar.

   If the address-of-record in the To header field of a REGISTER request
   is a SIPS URI, then any Contact header field values in the request
   SHOULD also be SIPS URIs.  Clients should only register non-SIPS URIs
   under a SIPS address-of-record when the security of the resource
   represented by the contact address is guaranteed by other means.
   This may be applicable to URIs that invoke protocols other than SIP,
   or SIP devices secured by protocols other than TLS.

   Registrations do not need to update all bindings.  Typically, a UA
   only updates its own contact addresses.

<span class="h5"><a class="selflink" id="section-10.2.1.1" href="#section-10.2.1.1">10.2.1.1</a> Setting the Expiration Interval of Contact Addresses</span>

   When a client sends a REGISTER request, it MAY suggest an expiration
   interval that indicates how long the client would like the
   registration to be valid.  (As described in <a href="#section-10.3">Section 10.3</a>, the
   registrar selects the actual time interval based on its local
   policy.)

   There are two ways in which a client can suggest an expiration
   interval for a binding: through an Expires header field or an
   "expires" Contact header parameter.  The latter allows expiration
   intervals to be suggested on a per-binding basis when more than one
   binding is given in a single REGISTER request, whereas the former
   suggests an expiration interval for all Contact header field values
   that do not contain the "expires" parameter.





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   If neither mechanism for expressing a suggested expiration time is
   present in a REGISTER, the client is indicating its desire for the
   server to choose.

<span class="h5"><a class="selflink" id="section-10.2.1.2" href="#section-10.2.1.2">10.2.1.2</a> Preferences among Contact Addresses</span>

   If more than one Contact is sent in a REGISTER request, the
   registering UA intends to associate all of the URIs in these Contact
   header field values with the address-of-record present in the To
   field.  This list can be prioritized with the "q" parameter in the
   Contact header field.  The "q" parameter indicates a relative
   preference for the particular Contact header field value compared to
   other bindings for this address-of-record.  <a href="#section-16.6">Section 16.6</a> describes
   how a proxy server uses this preference indication.

<span class="h4"><a class="selflink" id="section-10.2.2" href="#section-10.2.2">10.2.2</a> Removing Bindings</span>

   Registrations are soft state and expire unless refreshed, but can
   also be explicitly removed.  A client can attempt to influence the
   expiration interval selected by the registrar as described in <a href="#section-10.2.1">Section</a>
   <a href="#section-10.2.1">10.2.1</a>.  A UA requests the immediate removal of a binding by
   specifying an expiration interval of "0" for that contact address in
   a REGISTER request.  UAs SHOULD support this mechanism so that
   bindings can be removed before their expiration interval has passed.

   The REGISTER-specific Contact header field value of "*" applies to
   all registrations, but it MUST NOT be used unless the Expires header
   field is present with a value of "0".

      Use of the "*" Contact header field value allows a registering UA
      to remove all bindings associated with an address-of-record
      without knowing their precise values.

<span class="h4"><a class="selflink" id="section-10.2.3" href="#section-10.2.3">10.2.3</a> Fetching Bindings</span>

   A success response to any REGISTER request contains the complete list
   of existing bindings, regardless of whether the request contained a
   Contact header field.  If no Contact header field is present in a
   REGISTER request, the list of bindings is left unchanged.

<span class="h4"><a class="selflink" id="section-10.2.4" href="#section-10.2.4">10.2.4</a> Refreshing Bindings</span>

   Each UA is responsible for refreshing the bindings that it has
   previously established.  A UA SHOULD NOT refresh bindings set up by
   other UAs.






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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The 200 (OK) response from the registrar contains a list of Contact
   fields enumerating all current bindings.  The UA compares each
   contact address to see if it created the contact address, using
   comparison rules in <a href="#section-19.1.4">Section 19.1.4</a>.  If so, it updates the expiration
   time interval according to the expires parameter or, if absent, the
   Expires field value.  The UA then issues a REGISTER request for each
   of its bindings before the expiration interval has elapsed.  It MAY
   combine several updates into one REGISTER request.

   A UA SHOULD use the same Call-ID for all registrations during a
   single boot cycle.  Registration refreshes SHOULD be sent to the same
   network address as the original registration, unless redirected.

<span class="h4"><a class="selflink" id="section-10.2.5" href="#section-10.2.5">10.2.5</a> Setting the Internal Clock</span>

   If the response for a REGISTER request contains a Date header field,
   the client MAY use this header field to learn the current time in
   order to set any internal clocks.

<span class="h4"><a class="selflink" id="section-10.2.6" href="#section-10.2.6">10.2.6</a> Discovering a Registrar</span>

   UAs can use three ways to determine the address to which to send
   registrations:  by configuration, using the address-of-record, and
   multicast.  A UA can be configured, in ways beyond the scope of this
   specification, with a registrar address.  If there is no configured
   registrar address, the UA SHOULD use the host part of the address-
   of-record as the Request-URI and address the request there, using the
   normal SIP server location mechanisms [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].  For example, the UA for
   the user "sip:carol@chicago.com" addresses the REGISTER request to
   "sip:chicago.com".

   Finally, a UA can be configured to use multicast.  Multicast
   registrations are addressed to the well-known "all SIP servers"
   multicast address "sip.mcast.net" (224.0.1.75 for IPv4).  No well-
   known IPv6 multicast address has been allocated; such an allocation
   will be documented separately when needed.  SIP UAs MAY listen to
   that address and use it to become aware of the location of other
   local users (see [<a href="#ref-33" title="&quot;A multicast user directory service for synchronous rendezvous,&quot;">33</a>]); however, they do not respond to the request.

      Multicast registration may be inappropriate in some environments,
      for example, if multiple businesses share the same local area
      network.

<span class="h4"><a class="selflink" id="section-10.2.7" href="#section-10.2.7">10.2.7</a> Transmitting a Request</span>

   Once the REGISTER method has been constructed, and the destination of
   the message identified, UACs follow the procedures described in
   <a href="#section-8.1.2">Section 8.1.2</a> to hand off the REGISTER to the transaction layer.



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   If the transaction layer returns a timeout error because the REGISTER
   yielded no response, the UAC SHOULD NOT immediately re-attempt a
   registration to the same registrar.

      An immediate re-attempt is likely to also timeout.  Waiting some
      reasonable time interval for the conditions causing the timeout to
      be corrected reduces unnecessary load on the network.  No specific
      interval is mandated.

<span class="h4"><a class="selflink" id="section-10.2.8" href="#section-10.2.8">10.2.8</a> Error Responses</span>

   If a UA receives a 423 (Interval Too Brief) response, it MAY retry
   the registration after making the expiration interval of all contact
   addresses in the REGISTER request equal to or greater than the
   expiration interval within the Min-Expires header field of the 423
   (Interval Too Brief) response.

<span class="h3"><a class="selflink" id="section-10.3" href="#section-10.3">10.3</a> Processing REGISTER Requests</span>

   A registrar is a UAS that responds to REGISTER requests and maintains
   a list of bindings that are accessible to proxy servers and redirect
   servers within its administrative domain.  A registrar handles
   requests according to <a href="#section-8.2">Section 8.2</a> and <a href="#section-17.2">Section 17.2</a>, but it accepts
   only REGISTER requests.  A registrar MUST not generate 6xx responses.

   A registrar MAY redirect REGISTER requests as appropriate.  One
   common usage would be for a registrar listening on a multicast
   interface to redirect multicast REGISTER requests to its own unicast
   interface with a 302 (Moved Temporarily) response.

   Registrars MUST ignore the Record-Route header field if it is
   included in a REGISTER request.  Registrars MUST NOT include a
   Record-Route header field in any response to a REGISTER request.

      A registrar might receive a request that traversed a proxy which
      treats REGISTER as an unknown request and which added a Record-
      Route header field value.

   A registrar has to know (for example, through configuration) the set
   of domain(s) for which it maintains bindings.  REGISTER requests MUST
   be processed by a registrar in the order that they are received.
   REGISTER requests MUST also be processed atomically, meaning that a
   particular REGISTER request is either processed completely or not at
   all.  Each REGISTER message MUST be processed independently of any
   other registration or binding changes.






<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 63]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-64" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   When receiving a REGISTER request, a registrar follows these steps:

      1. The registrar inspects the Request-URI to determine whether it
         has access to bindings for the domain identified in the
         Request-URI.  If not, and if the server also acts as a proxy
         server, the server SHOULD forward the request to the addressed
         domain, following the general behavior for proxying messages
         described in <a href="#section-16">Section 16</a>.

      2. To guarantee that the registrar supports any necessary
         extensions, the registrar MUST process the Require header field
         values as described for UASs in <a href="#section-8.2.2">Section 8.2.2</a>.

      3. A registrar SHOULD authenticate the UAC.  Mechanisms for the
         authentication of SIP user agents are described in <a href="#section-22">Section 22</a>.
         Registration behavior in no way overrides the generic
         authentication framework for SIP.  If no authentication
         mechanism is available, the registrar MAY take the From address
         as the asserted identity of the originator of the request.

      4. The registrar SHOULD determine if the authenticated user is
         authorized to modify registrations for this address-of-record.
         For example, a registrar might consult an authorization
         database that maps user names to a list of addresses-of-record
         for which that user has authorization to modify bindings.  If
         the authenticated user is not authorized to modify bindings,
         the registrar MUST return a 403 (Forbidden) and skip the
         remaining steps.

         In architectures that support third-party registration, one
         entity may be responsible for updating the registrations
         associated with multiple addresses-of-record.

      5. The registrar extracts the address-of-record from the To header
         field of the request.  If the address-of-record is not valid
         for the domain in the Request-URI, the registrar MUST send a
         404 (Not Found) response and skip the remaining steps.  The URI
         MUST then be converted to a canonical form.  To do that, all
         URI parameters MUST be removed (including the user-param), and
         any escaped characters MUST be converted to their unescaped
         form.  The result serves as an index into the list of bindings.










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      6. The registrar checks whether the request contains the Contact
         header field.  If not, it skips to the last step.  If the
         Contact header field is present, the registrar checks if there
         is one Contact field value that contains the special value "*"
         and an Expires field.  If the request has additional Contact
         fields or an expiration time other than zero, the request is
         invalid, and the server MUST return a 400 (Invalid Request) and
         skip the remaining steps.  If not, the registrar checks whether
         the Call-ID agrees with the value stored for each binding.  If
         not, it MUST remove the binding.  If it does agree, it MUST
         remove the binding only if the CSeq in the request is higher
         than the value stored for that binding.  Otherwise, the update
         MUST be aborted and the request fails.

      7. The registrar now processes each contact address in the Contact
         header field in turn.  For each address, it determines the
         expiration interval as follows:

         -  If the field value has an "expires" parameter, that value
            MUST be taken as the requested expiration.

         -  If there is no such parameter, but the request has an
            Expires header field, that value MUST be taken as the
            requested expiration.

         -  If there is neither, a locally-configured default value MUST
            be taken as the requested expiration.

         The registrar MAY choose an expiration less than the requested
         expiration interval.  If and only if the requested expiration
         interval is greater than zero AND smaller than one hour AND
         less than a registrar-configured minimum, the registrar MAY
         reject the registration with a response of 423 (Interval Too
         Brief).  This response MUST contain a Min-Expires header field
         that states the minimum expiration interval the registrar is
         willing to honor.  It then skips the remaining steps.

         Allowing the registrar to set the registration interval
         protects it against excessively frequent registration refreshes
         while limiting the state that it needs to maintain and
         decreasing the likelihood of registrations going stale.  The
         expiration interval of a registration is frequently used in the
         creation of services.  An example is a follow-me service, where
         the user may only be available at a terminal for a brief
         period.  Therefore, registrars should accept brief
         registrations; a request should only be rejected if the
         interval is so short that the refreshes would degrade registrar
         performance.



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-66" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


         For each address, the registrar then searches the list of
         current bindings using the URI comparison rules.  If the
         binding does not exist, it is tentatively added.  If the
         binding does exist, the registrar checks the Call-ID value.  If
         the Call-ID value in the existing binding differs from the
         Call-ID value in the request, the binding MUST be removed if
         the expiration time is zero and updated otherwise.  If they are
         the same, the registrar compares the CSeq value.  If the value
         is higher than that of the existing binding, it MUST update or
         remove the binding as above.  If not, the update MUST be
         aborted and the request fails.

         This algorithm ensures that out-of-order requests from the same
         UA are ignored.

         Each binding record records the Call-ID and CSeq values from
         the request.

         The binding updates MUST be committed (that is, made visible to
         the proxy or redirect server) if and only if all binding
         updates and additions succeed.  If any one of them fails (for
         example, because the back-end database commit failed), the
         request MUST fail with a 500 (Server Error) response and all
         tentative binding updates MUST be removed.

      8. The registrar returns a 200 (OK) response.  The response MUST
         contain Contact header field values enumerating all current
         bindings.  Each Contact value MUST feature an "expires"
         parameter indicating its expiration interval chosen by the
         registrar.  The response SHOULD include a Date header field.

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a> Querying for Capabilities</span>

   The SIP method OPTIONS allows a UA to query another UA or a proxy
   server as to its capabilities.  This allows a client to discover
   information about the supported methods, content types, extensions,
   codecs, etc. without "ringing" the other party.  For example, before
   a client inserts a Require header field into an INVITE listing an
   option that it is not certain the destination UAS supports, the
   client can query the destination UAS with an OPTIONS to see if this
   option is returned in a Supported header field.  All UAs MUST support
   the OPTIONS method.

   The target of the OPTIONS request is identified by the Request-URI,
   which could identify another UA or a SIP server.  If the OPTIONS is
   addressed to a proxy server, the Request-URI is set without a user
   part, similar to the way a Request-URI is set for a REGISTER request.




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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Alternatively, a server receiving an OPTIONS request with a Max-
   Forwards header field value of 0 MAY respond to the request
   regardless of the Request-URI.

      This behavior is common with HTTP/1.1.  This behavior can be used
      as a "traceroute" functionality to check the capabilities of
      individual hop servers by sending a series of OPTIONS requests
      with incremented Max-Forwards values.

   As is the case for general UA behavior, the transaction layer can
   return a timeout error if the OPTIONS yields no response.  This may
   indicate that the target is unreachable and hence unavailable.

   An OPTIONS request MAY be sent as part of an established dialog to
   query the peer on capabilities that may be utilized later in the
   dialog.

<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a> Construction of OPTIONS Request</span>

   An OPTIONS request is constructed using the standard rules for a SIP
   request as discussed in <a href="#section-8.1.1">Section 8.1.1</a>.

   A Contact header field MAY be present in an OPTIONS.

   An Accept header field SHOULD be included to indicate the type of
   message body the UAC wishes to receive in the response.  Typically,
   this is set to a format that is used to describe the media
   capabilities of a UA, such as SDP (application/sdp).

   The response to an OPTIONS request is assumed to be scoped to the
   Request-URI in the original request.  However, only when an OPTIONS
   is sent as part of an established dialog is it guaranteed that future
   requests will be received by the server that generated the OPTIONS
   response.

   Example OPTIONS request:

      OPTIONS sip:carol@chicago.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
      Max-Forwards: 70
      To: &lt;sip:carol@chicago.com&gt;
      From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 63104 OPTIONS
      Contact: &lt;sip:alice@pc33.atlanta.com&gt;
      Accept: application/sdp
      Content-Length: 0




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-68" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a> Processing of OPTIONS Request</span>

   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in <a href="#section-8.2.6">Section 8.2.6</a>.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.

   An OPTIONS request received within a dialog generates a 200 (OK)
   response that is identical to one constructed outside a dialog and
   does not have any impact on the dialog.

   This use of OPTIONS has limitations due to the differences in proxy
   handling of OPTIONS and INVITE requests.  While a forked INVITE can
   result in multiple 200 (OK) responses being returned, a forked
   OPTIONS will only result in a single 200 (OK) response, since it is
   treated by proxies using the non-INVITE handling.  See <a href="#section-16.7">Section 16.7</a>
   for the normative details.

   If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.

   Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
   fields SHOULD be present in a 200 (OK) response to an OPTIONS
   request.  If the response is generated by a proxy, the Allow header
   field SHOULD be omitted as it is ambiguous since a proxy is method
   agnostic.  Contact header fields MAY be present in a 200 (OK)
   response and have the same semantics as in a 3xx response.  That is,
   they may list a set of alternative names and methods of reaching the
   user.  A Warning header field MAY be present.

   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).  If the types
   include one that can describe media capabilities, the UAS SHOULD
   include a body in the response for that purpose.  Details on the
   construction of such a body in the case of application/sdp are
   described in [<a href="#ref-13" title="&quot;An Offer/Answer Model with SDP&quot;">13</a>].









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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-69" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Example OPTIONS response generated by a UAS (corresponding to the
   request in <a href="#section-11.1">Section 11.1</a>):

      SIP/2.0 200 OK
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKhjhs8ass877
       ;received=192.0.2.4
      To: &lt;sip:carol@chicago.com&gt;;tag=93810874
      From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 63104 OPTIONS
      Contact: &lt;sip:carol@chicago.com&gt;
      Contact: &lt;mailto:carol@chicago.com&gt;
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
      Accept: application/sdp
      Accept-Encoding: gzip
      Accept-Language: en
      Supported: foo
      Content-Type: application/sdp
      Content-Length: 274

      (SDP not shown)

<span class="h2"><a class="selflink" id="section-12" href="#section-12">12</a> Dialogs</span>

   A key concept for a user agent is that of a dialog.  A dialog
   represents a peer-to-peer SIP relationship between two user agents
   that persists for some time.  The dialog facilitates sequencing of
   messages between the user agents and proper routing of requests
   between both of them.  The dialog represents a context in which to
   interpret SIP messages.  <a href="#section-8">Section 8</a> discussed method independent UA
   processing for requests and responses outside of a dialog.  This
   section discusses how those requests and responses are used to
   construct a dialog, and then how subsequent requests and responses
   are sent within a dialog.

   A dialog is identified at each UA with a dialog ID, which consists of
   a Call-ID value, a local tag and a remote tag.  The dialog ID at each
   UA involved in the dialog is not the same.  Specifically, the local
   tag at one UA is identical to the remote tag at the peer UA.  The
   tags are opaque tokens that facilitate the generation of unique
   dialog IDs.

   A dialog ID is also associated with all responses and with any
   request that contains a tag in the To field.  The rules for computing
   the dialog ID of a message depend on whether the SIP element is a UAC
   or UAS.  For a UAC, the Call-ID value of the dialog ID is set to the
   Call-ID of the message, the remote tag is set to the tag in the To
   field of the message, and the local tag is set to the tag in the From



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   field of the message (these rules apply to both requests and
   responses).  As one would expect for a UAS, the Call-ID value of the
   dialog ID is set to the Call-ID of the message, the remote tag is set
   to the tag in the From field of the message, and the local tag is set
   to the tag in the To field of the message.

   A dialog contains certain pieces of state needed for further message
   transmissions within the dialog.  This state consists of the dialog
   ID, a local sequence number (used to order requests from the UA to
   its peer), a remote sequence number (used to order requests from its
   peer to the UA), a local URI, a remote URI, remote target, a boolean
   flag called "secure", and a route set, which is an ordered list of
   URIs.  The route set is the list of servers that need to be traversed
   to send a request to the peer.  A dialog can also be in the "early"
   state, which occurs when it is created with a provisional response,
   and then transition to the "confirmed" state when a 2xx final
   response arrives.  For other responses, or if no response arrives at
   all on that dialog, the early dialog terminates.

<span class="h3"><a class="selflink" id="section-12.1" href="#section-12.1">12.1</a> Creation of a Dialog</span>

   Dialogs are created through the generation of non-failure responses
   to requests with specific methods.  Within this specification, only
   2xx and 101-199 responses with a To tag, where the request was
   INVITE, will establish a dialog.  A dialog established by a non-final
   response to a request is in the "early" state and it is called an
   early dialog.  Extensions MAY define other means for creating
   dialogs.  <a href="#section-13">Section 13</a> gives more details that are specific to the
   INVITE method.  Here, we describe the process for creation of dialog
   state that is not dependent on the method.

   UAs MUST assign values to the dialog ID components as described
   below.

<span class="h4"><a class="selflink" id="section-12.1.1" href="#section-12.1.1">12.1.1</a> UAS behavior</span>

   When a UAS responds to a request with a response that establishes a
   dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Route
   header field values from the request into the response (including the
   URIs, URI parameters, and any Record-Route header field parameters,
   whether they are known or unknown to the UAS) and MUST maintain the
   order of those values.  The UAS MUST add a Contact header field to
   the response.  The Contact header field contains an address where the
   UAS would like to be contacted for subsequent requests in the dialog
   (which includes the ACK for a 2xx response in the case of an INVITE).
   Generally, the host portion of this URI is the IP address or FQDN of
   the host.  The URI provided in the Contact header field MUST be a SIP
   or SIPS URI.  If the request that initiated the dialog contained a



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   SIPS URI in the Request-URI or in the top Record-Route header field
   value, if there was any, or the Contact header field if there was no
   Record-Route header field, the Contact header field in the response
   MUST be a SIPS URI.  The URI SHOULD have global scope (that is, the
   same URI can be used in messages outside this dialog).  The same way,
   the scope of the URI in the Contact header field of the INVITE is not
   limited to this dialog either.  It can therefore be used in messages
   to the UAC even outside this dialog.

   The UAS then constructs the state of the dialog.  This state MUST be
   maintained for the duration of the dialog.

   If the request arrived over TLS, and the Request-URI contained a SIPS
   URI, the "secure" flag is set to TRUE.

   The route set MUST be set to the list of URIs in the Record-Route
   header field from the request, taken in order and preserving all URI
   parameters.  If no Record-Route header field is present in the
   request, the route set MUST be set to the empty set.  This route set,
   even if empty, overrides any pre-existing route set for future
   requests in this dialog.  The remote target MUST be set to the URI
   from the Contact header field of the request.

   The remote sequence number MUST be set to the value of the sequence
   number in the CSeq header field of the request.  The local sequence
   number MUST be empty.  The call identifier component of the dialog ID
   MUST be set to the value of the Call-ID in the request.  The local
   tag component of the dialog ID MUST be set to the tag in the To field
   in the response to the request (which always includes a tag), and the
   remote tag component of the dialog ID MUST be set to the tag from the
   From field in the request.  A UAS MUST be prepared to receive a
   request without a tag in the From field, in which case the tag is
   considered to have a value of null.

      This is to maintain backwards compatibility with <a href="./rfc2543">RFC 2543</a>, which
      did not mandate From tags.

   The remote URI MUST be set to the URI in the From field, and the
   local URI MUST be set to the URI in the To field.

<span class="h4"><a class="selflink" id="section-12.1.2" href="#section-12.1.2">12.1.2</a> UAC Behavior</span>

   When a UAC sends a request that can establish a dialog (such as an
   INVITE) it MUST provide a SIP or SIPS URI with global scope (i.e.,
   the same SIP URI can be used in messages outside this dialog) in the
   Contact header field of the request.  If the request has a Request-
   URI or a topmost Route header field value with a SIPS URI, the
   Contact header field MUST contain a SIPS URI.



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   When a UAC receives a response that establishes a dialog, it
   constructs the state of the dialog.  This state MUST be maintained
   for the duration of the dialog.

   If the request was sent over TLS, and the Request-URI contained a
   SIPS URI, the "secure" flag is set to TRUE.

   The route set MUST be set to the list of URIs in the Record-Route
   header field from the response, taken in reverse order and preserving
   all URI parameters.  If no Record-Route header field is present in
   the response, the route set MUST be set to the empty set.  This route
   set, even if empty, overrides any pre-existing route set for future
   requests in this dialog.  The remote target MUST be set to the URI
   from the Contact header field of the response.

   The local sequence number MUST be set to the value of the sequence
   number in the CSeq header field of the request.  The remote sequence
   number MUST be empty (it is established when the remote UA sends a
   request within the dialog).  The call identifier component of the
   dialog ID MUST be set to the value of the Call-ID in the request.
   The local tag component of the dialog ID MUST be set to the tag in
   the From field in the request, and the remote tag component of the
   dialog ID MUST be set to the tag in the To field of the response.  A
   UAC MUST be prepared to receive a response without a tag in the To
   field, in which case the tag is considered to have a value of null.

      This is to maintain backwards compatibility with <a href="./rfc2543">RFC 2543</a>, which
      did not mandate To tags.

   The remote URI MUST be set to the URI in the To field, and the local
   URI MUST be set to the URI in the From field.

<span class="h3"><a class="selflink" id="section-12.2" href="#section-12.2">12.2</a> Requests within a Dialog</span>

   Once a dialog has been established between two UAs, either of them
   MAY initiate new transactions as needed within the dialog.  The UA
   sending the request will take the UAC role for the transaction.  The
   UA receiving the request will take the UAS role.  Note that these may
   be different roles than the UAs held during the transaction that
   established the dialog.

   Requests within a dialog MAY contain Record-Route and Contact header
   fields.  However, these requests do not cause the dialog's route set
   to be modified, although they may modify the remote target URI.
   Specifically, requests that are not target refresh requests do not
   modify the dialog's remote target URI, and requests that are target
   refresh requests do.  For dialogs that have been established with an




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   INVITE, the only target refresh request defined is re-INVITE (see
   <a href="#section-14">Section 14</a>).  Other extensions may define different target refresh
   requests for dialogs established in other ways.

      Note that an ACK is NOT a target refresh request.

   Target refresh requests only update the dialog's remote target URI,
   and not the route set formed from the Record-Route.  Updating the
   latter would introduce severe backwards compatibility problems with
   <a href="./rfc2543">RFC 2543</a>-compliant systems.

<span class="h4"><a class="selflink" id="section-12.2.1" href="#section-12.2.1">12.2.1</a> UAC Behavior</span>

<span class="h5"><a class="selflink" id="section-12.2.1.1" href="#section-12.2.1.1">12.2.1.1</a> Generating the Request</span>

   A request within a dialog is constructed by using many of the
   components of the state stored as part of the dialog.

   The URI in the To field of the request MUST be set to the remote URI
   from the dialog state.  The tag in the To header field of the request
   MUST be set to the remote tag of the dialog ID.  The From URI of the
   request MUST be set to the local URI from the dialog state.  The tag
   in the From header field of the request MUST be set to the local tag
   of the dialog ID.  If the value of the remote or local tags is null,
   the tag parameter MUST be omitted from the To or From header fields,
   respectively.

      Usage of the URI from the To and From fields in the original
      request within subsequent requests is done for backwards
      compatibility with <a href="./rfc2543">RFC 2543</a>, which used the URI for dialog
      identification.  In this specification, only the tags are used for
      dialog identification.  It is expected that mandatory reflection
      of the original To and From URI in mid-dialog requests will be
      deprecated in a subsequent revision of this specification.

   The Call-ID of the request MUST be set to the Call-ID of the dialog.
   Requests within a dialog MUST contain strictly monotonically
   increasing and contiguous CSeq sequence numbers (increasing-by-one)
   in each direction (excepting ACK and CANCEL of course, whose numbers
   equal the requests being acknowledged or cancelled).  Therefore, if
   the local sequence number is not empty, the value of the local
   sequence number MUST be incremented by one, and this value MUST be
   placed into the CSeq header field.  If the local sequence number is
   empty, an initial value MUST be chosen using the guidelines of
   <a href="#section-8.1.1.5">Section 8.1.1.5</a>.  The method field in the CSeq header field value
   MUST match the method of the request.





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      With a length of 32 bits, a client could generate, within a single
      call, one request a second for about 136 years before needing to
      wrap around.  The initial value of the sequence number is chosen
      so that subsequent requests within the same call will not wrap
      around.  A non-zero initial value allows clients to use a time-
      based initial sequence number.  A client could, for example,
      choose the 31 most significant bits of a 32-bit second clock as an
      initial sequence number.

   The UAC uses the remote target and route set to build the Request-URI
   and Route header field of the request.

   If the route set is empty, the UAC MUST place the remote target URI
   into the Request-URI.  The UAC MUST NOT add a Route header field to
   the request.

   If the route set is not empty, and the first URI in the route set
   contains the lr parameter (see <a href="#section-19.1.1">Section 19.1.1</a>), the UAC MUST place
   the remote target URI into the Request-URI and MUST include a Route
   header field containing the route set values in order, including all
   parameters.

   If the route set is not empty, and its first URI does not contain the
   lr parameter, the UAC MUST place the first URI from the route set
   into the Request-URI, stripping any parameters that are not allowed
   in a Request-URI.  The UAC MUST add a Route header field containing
   the remainder of the route set values in order, including all
   parameters.  The UAC MUST then place the remote target URI into the
   Route header field as the last value.

   For example, if the remote target is sip:user@remoteua and the route
   set contains:

      &lt;sip:proxy1&gt;,&lt;sip:proxy2&gt;,&lt;sip:proxy3;lr&gt;,&lt;sip:proxy4&gt;

   The request will be formed with the following Request-URI and Route
   header field:

   METHOD sip:proxy1
   Route: &lt;sip:proxy2&gt;,&lt;sip:proxy3;lr&gt;,&lt;sip:proxy4&gt;,&lt;sip:user@remoteua&gt;

      If the first URI of the route set does not contain the lr
      parameter, the proxy indicated does not understand the routing
      mechanisms described in this document and will act as specified in
      <a href="./rfc2543">RFC 2543</a>, replacing the Request-URI with the first Route header
      field value it receives while forwarding the message.  Placing the
      Request-URI at the end of the Route header field preserves the




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      information in that Request-URI across the strict router (it will
      be returned to the Request-URI when the request reaches a loose-
      router).

   A UAC SHOULD include a Contact header field in any target refresh
   requests within a dialog, and unless there is a need to change it,
   the URI SHOULD be the same as used in previous requests within the
   dialog.  If the "secure" flag is true, that URI MUST be a SIPS URI.
   As discussed in <a href="#section-12.2.2">Section 12.2.2</a>, a Contact header field in a target
   refresh request updates the remote target URI.  This allows a UA to
   provide a new contact address, should its address change during the
   duration of the dialog.

   However, requests that are not target refresh requests do not affect
   the remote target URI for the dialog.

   The rest of the request is formed as described in <a href="#section-8.1.1">Section 8.1.1</a>.

   Once the request has been constructed, the address of the server is
   computed and the request is sent, using the same procedures for
   requests outside of a dialog (<a href="#section-8.1.2">Section 8.1.2</a>).

      The procedures in <a href="#section-8.1.2">Section 8.1.2</a> will normally result in the
      request being sent to the address indicated by the topmost Route
      header field value or the Request-URI if no Route header field is
      present.  Subject to certain restrictions, they allow the request
      to be sent to an alternate address (such as a default outbound
      proxy not represented in the route set).

<span class="h5"><a class="selflink" id="section-12.2.1.2" href="#section-12.2.1.2">12.2.1.2</a> Processing the Responses</span>

   The UAC will receive responses to the request from the transaction
   layer.  If the client transaction returns a timeout, this is treated
   as a 408 (Request Timeout) response.

   The behavior of a UAC that receives a 3xx response for a request sent
   within a dialog is the same as if the request had been sent outside a
   dialog.  This behavior is described in <a href="#section-8.1.3.4">Section 8.1.3.4</a>.

      Note, however, that when the UAC tries alternative locations, it
      still uses the route set for the dialog to build the Route header
      of the request.

   When a UAC receives a 2xx response to a target refresh request, it
   MUST replace the dialog's remote target URI with the URI from the
   Contact header field in that response, if present.





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   If the response for a request within a dialog is a 481
   (Call/Transaction Does Not Exist) or a 408 (Request Timeout), the UAC
   SHOULD terminate the dialog.  A UAC SHOULD also terminate a dialog if
   no response at all is received for the request (the client
   transaction would inform the TU about the timeout.)

      For INVITE initiated dialogs, terminating the dialog consists of
      sending a BYE.

<span class="h4"><a class="selflink" id="section-12.2.2" href="#section-12.2.2">12.2.2</a> UAS Behavior</span>

   Requests sent within a dialog, as any other requests, are atomic.  If
   a particular request is accepted by the UAS, all the state changes
   associated with it are performed.  If the request is rejected, none
   of the state changes are performed.

      Note that some requests, such as INVITEs, affect several pieces of
      state.

   The UAS will receive the request from the transaction layer.  If the
   request has a tag in the To header field, the UAS core computes the
   dialog identifier corresponding to the request and compares it with
   existing dialogs.  If there is a match, this is a mid-dialog request.
   In that case, the UAS first applies the same processing rules for
   requests outside of a dialog, discussed in <a href="#section-8.2">Section 8.2</a>.

   If the request has a tag in the To header field, but the dialog
   identifier does not match any existing dialogs, the UAS may have
   crashed and restarted, or it may have received a request for a
   different (possibly failed) UAS (the UASs can construct the To tags
   so that a UAS can identify that the tag was for a UAS for which it is
   providing recovery).  Another possibility is that the incoming
   request has been simply misrouted.  Based on the To tag, the UAS MAY
   either accept or reject the request.  Accepting the request for
   acceptable To tags provides robustness, so that dialogs can persist
   even through crashes.  UAs wishing to support this capability must
   take into consideration some issues such as choosing monotonically
   increasing CSeq sequence numbers even across reboots, reconstructing
   the route set, and accepting out-of-range RTP timestamps and sequence
   numbers.

   If the UAS wishes to reject the request because it does not wish to
   recreate the dialog, it MUST respond to the request with a 481
   (Call/Transaction Does Not Exist) status code and pass that to the
   server transaction.






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   Requests that do not change in any way the state of a dialog may be
   received within a dialog (for example, an OPTIONS request).  They are
   processed as if they had been received outside the dialog.

   If the remote sequence number is empty, it MUST be set to the value
   of the sequence number in the CSeq header field value in the request.
   If the remote sequence number was not empty, but the sequence number
   of the request is lower than the remote sequence number, the request
   is out of order and MUST be rejected with a 500 (Server Internal
   Error) response.  If the remote sequence number was not empty, and
   the sequence number of the request is greater than the remote
   sequence number, the request is in order.  It is possible for the
   CSeq sequence number to be higher than the remote sequence number by
   more than one.  This is not an error condition, and a UAS SHOULD be
   prepared to receive and process requests with CSeq values more than
   one higher than the previous received request.  The UAS MUST then set
   the remote sequence number to the value of the sequence number in the
   CSeq header field value in the request.

      If a proxy challenges a request generated by the UAC, the UAC has
      to resubmit the request with credentials.  The resubmitted request
      will have a new CSeq number.  The UAS will never see the first
      request, and thus, it will notice a gap in the CSeq number space.
      Such a gap does not represent any error condition.

   When a UAS receives a target refresh request, it MUST replace the
   dialog's remote target URI with the URI from the Contact header field
   in that request, if present.

<span class="h3"><a class="selflink" id="section-12.3" href="#section-12.3">12.3</a> Termination of a Dialog</span>

   Independent of the method, if a request outside of a dialog generates
   a non-2xx final response, any early dialogs created through
   provisional responses to that request are terminated.  The mechanism
   for terminating confirmed dialogs is method specific.  In this
   specification, the BYE method terminates a session and the dialog
   associated with it.  See <a href="#section-15">Section 15</a> for details.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a> Initiating a Session</span>

<span class="h3"><a class="selflink" id="section-13.1" href="#section-13.1">13.1</a> Overview</span>

   When a user agent client desires to initiate a session (for example,
   audio, video, or a game), it formulates an INVITE request.  The
   INVITE request asks a server to establish a session.  This request
   may be forwarded by proxies, eventually arriving at one or more UAS
   that can potentially accept the invitation.  These UASs will
   frequently need to query the user about whether to accept the



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   invitation.  After some time, those UASs can accept the invitation
   (meaning the session is to be established) by sending a 2xx response.
   If the invitation is not accepted, a 3xx, 4xx, 5xx or 6xx response is
   sent, depending on the reason for the rejection.  Before sending a
   final response, the UAS can also send provisional responses (1xx) to
   advise the UAC of progress in contacting the called user.

   After possibly receiving one or more provisional responses, the UAC
   will get one or more 2xx responses or one non-2xx final response.
   Because of the protracted amount of time it can take to receive final
   responses to INVITE, the reliability mechanisms for INVITE
   transactions differ from those of other requests (like OPTIONS).
   Once it receives a final response, the UAC needs to send an ACK for
   every final response it receives.  The procedure for sending this ACK
   depends on the type of response.  For final responses between 300 and
   699, the ACK processing is done in the transaction layer and follows
   one set of rules (See <a href="#section-17">Section 17</a>).  For 2xx responses, the ACK is
   generated by the UAC core.

   A 2xx response to an INVITE establishes a session, and it also
   creates a dialog between the UA that issued the INVITE and the UA
   that generated the 2xx response.  Therefore, when multiple 2xx
   responses are received from different remote UAs (because the INVITE
   forked), each 2xx establishes a different dialog.  All these dialogs
   are part of the same call.

   This section provides details on the establishment of a session using
   INVITE.  A UA that supports INVITE MUST also support ACK, CANCEL and
   BYE.

<span class="h3"><a class="selflink" id="section-13.2" href="#section-13.2">13.2</a> UAC Processing</span>

<span class="h4"><a class="selflink" id="section-13.2.1" href="#section-13.2.1">13.2.1</a> Creating the Initial INVITE</span>

   Since the initial INVITE represents a request outside of a dialog,
   its construction follows the procedures of <a href="#section-8.1.1">Section 8.1.1</a>.  Additional
   processing is required for the specific case of INVITE.

   An Allow header field (<a href="#section-20.5">Section 20.5</a>) SHOULD be present in the INVITE.
   It indicates what methods can be invoked within a dialog, on the UA
   sending the INVITE, for the duration of the dialog.  For example, a
   UA capable of receiving INFO requests within a dialog [<a href="#ref-34" title="&quot;The SIP INFO Method&quot;">34</a>] SHOULD
   include an Allow header field listing the INFO method.

   A Supported header field (<a href="#section-20.37">Section 20.37</a>) SHOULD be present in the
   INVITE.  It enumerates all the extensions understood by the UAC.





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   An Accept (<a href="#section-20.1">Section 20.1</a>) header field MAY be present in the INVITE.
   It indicates which Content-Types are acceptable to the UA, in both
   the response received by it, and in any subsequent requests sent to
   it within dialogs established by the INVITE.  The Accept header field
   is especially useful for indicating support of various session
   description formats.

   The UAC MAY add an Expires header field (<a href="#section-20.19">Section 20.19</a>) to limit the
   validity of the invitation.  If the time indicated in the Expires
   header field is reached and no final answer for the INVITE has been
   received, the UAC core SHOULD generate a CANCEL request for the
   INVITE, as per <a href="#section-9">Section 9</a>.

   A UAC MAY also find it useful to add, among others, Subject (<a href="#section-20.36">Section</a>
   <a href="#section-20.36">20.36</a>), Organization (<a href="#section-20.25">Section 20.25</a>) and User-Agent (<a href="#section-20.41">Section 20.41</a>)
   header fields.  They all contain information related to the INVITE.

   The UAC MAY choose to add a message body to the INVITE.  <a href="#section-8.1.1.10">Section</a>
   <a href="#section-8.1.1.10">8.1.1.10</a> deals with how to construct the header fields -- Content-
   Type among others -- needed to describe the message body.

   There are special rules for message bodies that contain a session
   description - their corresponding Content-Disposition is "session".
   SIP uses an offer/answer model where one UA sends a session
   description, called the offer, which contains a proposed description
   of the session.  The offer indicates the desired communications means
   (audio, video, games), parameters of those means (such as codec
   types) and addresses for receiving media from the answerer.  The
   other UA responds with another session description, called the
   answer, which indicates which communications means are accepted, the
   parameters that apply to those means, and addresses for receiving
   media from the offerer. An offer/answer exchange is within the
   context of a dialog, so that if a SIP INVITE results in multiple
   dialogs, each is a separate offer/answer exchange.  The offer/answer
   model defines restrictions on when offers and answers can be made
   (for example, you cannot make a new offer while one is in progress).
   This results in restrictions on where the offers and answers can
   appear in SIP messages.  In this specification, offers and answers
   can only appear in INVITE requests and responses, and ACK.  The usage
   of offers and answers is further restricted.  For the initial INVITE
   transaction, the rules are:

      o  The initial offer MUST be in either an INVITE or, if not there,
         in the first reliable non-failure message from the UAS back to
         the UAC.  In this specification, that is the final 2xx
         response.





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      o  If the initial offer is in an INVITE, the answer MUST be in a
         reliable non-failure message from UAS back to UAC which is
         correlated to that INVITE.  For this specification, that is
         only the final 2xx response to that INVITE.  That same exact
         answer MAY also be placed in any provisional responses sent
         prior to the answer.  The UAC MUST treat the first session
         description it receives as the answer, and MUST ignore any
         session descriptions in subsequent responses to the initial
         INVITE.

      o  If the initial offer is in the first reliable non-failure
         message from the UAS back to UAC, the answer MUST be in the
         acknowledgement for that message (in this specification, ACK
         for a 2xx response).

      o  After having sent or received an answer to the first offer, the
         UAC MAY generate subsequent offers in requests based on rules
         specified for that method, but only if it has received answers
         to any previous offers, and has not sent any offers to which it
         hasn't gotten an answer.

      o  Once the UAS has sent or received an answer to the initial
         offer, it MUST NOT generate subsequent offers in any responses
         to the initial INVITE.  This means that a UAS based on this
         specification alone can never generate subsequent offers until
         completion of the initial transaction.

   Concretely, the above rules specify two exchanges for UAs compliant
   to this specification alone - the offer is in the INVITE, and the
   answer in the 2xx (and possibly in a 1xx as well, with the same
   value), or the offer is in the 2xx, and the answer is in the ACK.
   All user agents that support INVITE MUST support these two exchanges.

   The Session Description Protocol (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-1" title="&quot;SDP: Session Description Protocol&quot;">1</a>]) MUST be
   supported by all user agents as a means to describe sessions, and its
   usage for constructing offers and answers MUST follow the procedures
   defined in [<a href="#ref-13" title="&quot;An Offer/Answer Model with SDP&quot;">13</a>].

   The restrictions of the offer-answer model just described only apply
   to bodies whose Content-Disposition header field value is "session".
   Therefore, it is possible that both the INVITE and the ACK contain a
   body message (for example, the INVITE carries a photo (Content-
   Disposition: render) and the ACK a session description (Content-
   Disposition: session)).

   If the Content-Disposition header field is missing, bodies of
   Content-Type application/sdp imply the disposition "session", while
   other content types imply "render".



<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 80]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-81" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Once the INVITE has been created, the UAC follows the procedures
   defined for sending requests outside of a dialog (<a href="#section-8">Section 8</a>).  This
   results in the construction of a client transaction that will
   ultimately send the request and deliver responses to the UAC.

<span class="h4"><a class="selflink" id="section-13.2.2" href="#section-13.2.2">13.2.2</a> Processing INVITE Responses</span>

   Once the INVITE has been passed to the INVITE client transaction, the
   UAC waits for responses for the INVITE.  If the INVITE client
   transaction returns a timeout rather than a response the TU acts as
   if a 408 (Request Timeout) response had been received, as described
   in <a href="#section-8.1.3">Section 8.1.3</a>.

<span class="h5"><a class="selflink" id="section-13.2.2.1" href="#section-13.2.2.1">13.2.2.1</a> 1xx Responses</span>

   Zero, one or multiple provisional responses may arrive before one or
   more final responses are received.  Provisional responses for an
   INVITE request can create "early dialogs".  If a provisional response
   has a tag in the To field, and if the dialog ID of the response does
   not match an existing dialog, one is constructed using the procedures
   defined in <a href="#section-12.1.2">Section 12.1.2</a>.

   The early dialog will only be needed if the UAC needs to send a
   request to its peer within the dialog before the initial INVITE
   transaction completes.  Header fields present in a provisional
   response are applicable as long as the dialog is in the early state
   (for example, an Allow header field in a provisional response
   contains the methods that can be used in the dialog while this is in
   the early state).

<span class="h5"><a class="selflink" id="section-13.2.2.2" href="#section-13.2.2.2">13.2.2.2</a> 3xx Responses</span>

   A 3xx response may contain one or more Contact header field values
   providing new addresses where the callee might be reachable.
   Depending on the status code of the 3xx response (see <a href="#section-21.3">Section 21.3</a>),
   the UAC MAY choose to try those new addresses.

<span class="h5"><a class="selflink" id="section-13.2.2.3" href="#section-13.2.2.3">13.2.2.3</a> 4xx, 5xx and 6xx Responses</span>

   A single non-2xx final response may be received for the INVITE.  4xx,
   5xx and 6xx responses may contain a Contact header field value
   indicating the location where additional information about the error
   can be found.  Subsequent final responses (which would only arrive
   under error conditions) MUST be ignored.

   All early dialogs are considered terminated upon reception of the
   non-2xx final response.




<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 81]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-82" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   After having received the non-2xx final response the UAC core
   considers the INVITE transaction completed.  The INVITE client
   transaction handles the generation of ACKs for the response (see
   <a href="#section-17">Section 17</a>).

<span class="h5"><a class="selflink" id="section-13.2.2.4" href="#section-13.2.2.4">13.2.2.4</a> 2xx Responses</span>

   Multiple 2xx responses may arrive at the UAC for a single INVITE
   request due to a forking proxy.  Each response is distinguished by
   the tag parameter in the To header field, and each represents a
   distinct dialog, with a distinct dialog identifier.

   If the dialog identifier in the 2xx response matches the dialog
   identifier of an existing dialog, the dialog MUST be transitioned to
   the "confirmed" state, and the route set for the dialog MUST be
   recomputed based on the 2xx response using the procedures of <a href="#section-12.2.1.2">Section</a>
   <a href="#section-12.2.1.2">12.2.1.2</a>.  Otherwise, a new dialog in the "confirmed" state MUST be
   constructed using the procedures of <a href="#section-12.1.2">Section 12.1.2</a>.

      Note that the only piece of state that is recomputed is the route
      set.  Other pieces of state such as the highest sequence numbers
      (remote and local) sent within the dialog are not recomputed.  The
      route set only is recomputed for backwards compatibility.  <a href="./rfc2543">RFC</a>
      <a href="./rfc2543">2543</a> did not mandate mirroring of the Record-Route header field in
      a 1xx, only 2xx.  However, we cannot update the entire state of
      the dialog, since mid-dialog requests may have been sent within
      the early dialog, modifying the sequence numbers, for example.

   The UAC core MUST generate an ACK request for each 2xx received from
   the transaction layer.  The header fields of the ACK are constructed
   in the same way as for any request sent within a dialog (see <a href="#section-12">Section</a>
   <a href="#section-12">12</a>) with the exception of the CSeq and the header fields related to
   authentication.  The sequence number of the CSeq header field MUST be
   the same as the INVITE being acknowledged, but the CSeq method MUST
   be ACK.  The ACK MUST contain the same credentials as the INVITE.  If
   the 2xx contains an offer (based on the rules above), the ACK MUST
   carry an answer in its body.  If the offer in the 2xx response is not
   acceptable, the UAC core MUST generate a valid answer in the ACK and
   then send a BYE immediately.

   Once the ACK has been constructed, the procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] are used to
   determine the destination address, port and transport.  However, the
   request is passed to the transport layer directly for transmission,
   rather than a client transaction.  This is because the UAC core
   handles retransmissions of the ACK, not the transaction layer.  The
   ACK MUST be passed to the client transport every time a
   retransmission of the 2xx final response that triggered the ACK
   arrives.



<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 82]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-83" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The UAC core considers the INVITE transaction completed 64*T1 seconds
   after the reception of the first 2xx response.  At this point all the
   early dialogs that have not transitioned to established dialogs are
   terminated.  Once the INVITE transaction is considered completed by
   the UAC core, no more new 2xx responses are expected to arrive.

   If, after acknowledging any 2xx response to an INVITE, the UAC does
   not want to continue with that dialog, then the UAC MUST terminate
   the dialog by sending a BYE request as described in <a href="#section-15">Section 15</a>.

<span class="h3"><a class="selflink" id="section-13.3" href="#section-13.3">13.3</a> UAS Processing</span>

<span class="h4"><a class="selflink" id="section-13.3.1" href="#section-13.3.1">13.3.1</a> Processing of the INVITE</span>

   The UAS core will receive INVITE requests from the transaction layer.
   It first performs the request processing procedures of <a href="#section-8.2">Section 8.2</a>,
   which are applied for both requests inside and outside of a dialog.

   Assuming these processing states are completed without generating a
   response, the UAS core performs the additional processing steps:

      1. If the request is an INVITE that contains an Expires header
         field, the UAS core sets a timer for the number of seconds
         indicated in the header field value.  When the timer fires, the
         invitation is considered to be expired.  If the invitation
         expires before the UAS has generated a final response, a 487
         (Request Terminated) response SHOULD be generated.

      2. If the request is a mid-dialog request, the method-independent
         processing described in <a href="#section-12.2.2">Section 12.2.2</a> is first applied.  It
         might also modify the session; <a href="#section-14">Section 14</a> provides details.

      3. If the request has a tag in the To header field but the dialog
         identifier does not match any of the existing dialogs, the UAS
         may have crashed and restarted, or may have received a request
         for a different (possibly failed) UAS.  <a href="#section-12.2.2">Section 12.2.2</a> provides
         guidelines to achieve a robust behavior under such a situation.

   Processing from here forward assumes that the INVITE is outside of a
   dialog, and is thus for the purposes of establishing a new session.

   The INVITE may contain a session description, in which case the UAS
   is being presented with an offer for that session.  It is possible
   that the user is already a participant in that session, even though
   the INVITE is outside of a dialog.  This can happen when a user is
   invited to the same multicast conference by multiple other
   participants.  If desired, the UAS MAY use identifiers within the
   session description to detect this duplication.  For example, SDP



<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 83]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-84" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   contains a session id and version number in the origin (o) field.  If
   the user is already a member of the session, and the session
   parameters contained in the session description have not changed, the
   UAS MAY silently accept the INVITE (that is, send a 2xx response
   without prompting the user).

   If the INVITE does not contain a session description, the UAS is
   being asked to participate in a session, and the UAC has asked that
   the UAS provide the offer of the session.  It MUST provide the offer
   in its first non-failure reliable message back to the UAC.  In this
   specification, that is a 2xx response to the INVITE.

   The UAS can indicate progress, accept, redirect, or reject the
   invitation.  In all of these cases, it formulates a response using
   the procedures described in <a href="#section-8.2.6">Section 8.2.6</a>.

<span class="h5"><a class="selflink" id="section-13.3.1.1" href="#section-13.3.1.1">13.3.1.1</a> Progress</span>

   If the UAS is not able to answer the invitation immediately, it can
   choose to indicate some kind of progress to the UAC (for example, an
   indication that a phone is ringing).  This is accomplished with a
   provisional response between 101 and 199.  These provisional
   responses establish early dialogs and therefore follow the procedures
   of <a href="#section-12.1.1">Section 12.1.1</a> in addition to those of <a href="#section-8.2.6">Section 8.2.6</a>.  A UAS MAY
   send as many provisional responses as it likes.  Each of these MUST
   indicate the same dialog ID.  However, these will not be delivered
   reliably.

   If the UAS desires an extended period of time to answer the INVITE,
   it will need to ask for an "extension" in order to prevent proxies
   from canceling the transaction.  A proxy has the option of canceling
   a transaction when there is a gap of 3 minutes between responses in a
   transaction.  To prevent cancellation, the UAS MUST send a non-100
   provisional response at every minute, to handle the possibility of
   lost provisional responses.

      An INVITE transaction can go on for extended durations when the
      user is placed on hold, or when interworking with PSTN systems
      which allow communications to take place without answering the
      call.  The latter is common in Interactive Voice Response (IVR)
      systems.

<span class="h5"><a class="selflink" id="section-13.3.1.2" href="#section-13.3.1.2">13.3.1.2</a> The INVITE is Redirected</span>

   If the UAS decides to redirect the call, a 3xx response is sent.  A
   300 (Multiple Choices), 301 (Moved Permanently) or 302 (Moved
   Temporarily) response SHOULD contain a Contact header field




<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 84]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-85" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   containing one or more URIs of new addresses to be tried.  The
   response is passed to the INVITE server transaction, which will deal
   with its retransmissions.

<span class="h5"><a class="selflink" id="section-13.3.1.3" href="#section-13.3.1.3">13.3.1.3</a> The INVITE is Rejected</span>

   A common scenario occurs when the callee is currently not willing or
   able to take additional calls at this end system.  A 486 (Busy Here)
   SHOULD be returned in such a scenario.  If the UAS knows that no
   other end system will be able to accept this call, a 600 (Busy
   Everywhere) response SHOULD be sent instead.  However, it is unlikely
   that a UAS will be able to know this in general, and thus this
   response will not usually be used.  The response is passed to the
   INVITE server transaction, which will deal with its retransmissions.

   A UAS rejecting an offer contained in an INVITE SHOULD return a 488
   (Not Acceptable Here) response.  Such a response SHOULD include a
   Warning header field value explaining why the offer was rejected.

<span class="h5"><a class="selflink" id="section-13.3.1.4" href="#section-13.3.1.4">13.3.1.4</a> The INVITE is Accepted</span>

   The UAS core generates a 2xx response.  This response establishes a
   dialog, and therefore follows the procedures of <a href="#section-12.1.1">Section 12.1.1</a> in
   addition to those of <a href="#section-8.2.6">Section 8.2.6</a>.

   A 2xx response to an INVITE SHOULD contain the Allow header field and
   the Supported header field, and MAY contain the Accept header field.
   Including these header fields allows the UAC to determine the
   features and extensions supported by the UAS for the duration of the
   call, without probing.

   If the INVITE request contained an offer, and the UAS had not yet
   sent an answer, the 2xx MUST contain an answer.  If the INVITE did
   not contain an offer, the 2xx MUST contain an offer if the UAS had
   not yet sent an offer.

   Once the response has been constructed, it is passed to the INVITE
   server transaction.  Note, however, that the INVITE server
   transaction will be destroyed as soon as it receives this final
   response and passes it to the transport.  Therefore, it is necessary
   to periodically pass the response directly to the transport until the
   ACK arrives.  The 2xx response is passed to the transport with an
   interval that starts at T1 seconds and doubles for each
   retransmission until it reaches T2 seconds (T1 and T2 are defined in
   <a href="#section-17">Section 17</a>).  Response retransmissions cease when an ACK request for
   the response is received.  This is independent of whatever transport
   protocols are used to send the response.




<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 85]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-86" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      Since 2xx is retransmitted end-to-end, there may be hops between
      UAS and UAC that are UDP.  To ensure reliable delivery across
      these hops, the response is retransmitted periodically even if the
      transport at the UAS is reliable.

   If the server retransmits the 2xx response for 64*T1 seconds without
   receiving an ACK, the dialog is confirmed, but the session SHOULD be
   terminated.  This is accomplished with a BYE, as described in <a href="#section-15">Section</a>
   <a href="#section-15">15</a>.

<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a> Modifying an Existing Session</span>

   A successful INVITE request (see <a href="#section-13">Section 13</a>) establishes both a
   dialog between two user agents and a session using the offer-answer
   model.  <a href="#section-12">Section 12</a> explains how to modify an existing dialog using a
   target refresh request (for example, changing the remote target URI
   of the dialog).  This section describes how to modify the actual
   session.  This modification can involve changing addresses or ports,
   adding a media stream, deleting a media stream, and so on.  This is
   accomplished by sending a new INVITE request within the same dialog
   that established the session.  An INVITE request sent within an
   existing dialog is known as a re-INVITE.

      Note that a single re-INVITE can modify the dialog and the
      parameters of the session at the same time.

   Either the caller or callee can modify an existing session.

   The behavior of a UA on detection of media failure is a matter of
   local policy.  However, automated generation of re-INVITE or BYE is
   NOT RECOMMENDED to avoid flooding the network with traffic when there
   is congestion.  In any case, if these messages are sent
   automatically, they SHOULD be sent after some randomized interval.

      Note that the paragraph above refers to automatically generated
      BYEs and re-INVITEs.  If the user hangs up upon media failure, the
      UA would send a BYE request as usual.

<span class="h3"><a class="selflink" id="section-14.1" href="#section-14.1">14.1</a> UAC Behavior</span>

   The same offer-answer model that applies to session descriptions in
   INVITEs (<a href="#section-13.2.1">Section 13.2.1</a>) applies to re-INVITEs.  As a result, a UAC
   that wants to add a media stream, for example, will create a new
   offer that contains this media stream, and send that in an INVITE
   request to its peer.  It is important to note that the full
   description of the session, not just the change, is sent.  This
   supports stateless session processing in various elements, and
   supports failover and recovery capabilities.  Of course, a UAC MAY



<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 86]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-87" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   send a re-INVITE with no session description, in which case the first
   reliable non-failure response to the re-INVITE will contain the offer
   (in this specification, that is a 2xx response).

   If the session description format has the capability for version
   numbers, the offerer SHOULD indicate that the version of the session
   description has changed.

   The To, From, Call-ID, CSeq, and Request-URI of a re-INVITE are set
   following the same rules as for regular requests within an existing
   dialog, described in <a href="#section-12">Section 12</a>.

   A UAC MAY choose not to add an Alert-Info header field or a body with
   Content-Disposition "alert" to re-INVITEs because UASs do not
   typically alert the user upon reception of a re-INVITE.

   Unlike an INVITE, which can fork, a re-INVITE will never fork, and
   therefore, only ever generate a single final response.  The reason a
   re-INVITE will never fork is that the Request-URI identifies the
   target as the UA instance it established the dialog with, rather than
   identifying an address-of-record for the user.

   Note that a UAC MUST NOT initiate a new INVITE transaction within a
   dialog while another INVITE transaction is in progress in either
   direction.

      1. If there is an ongoing INVITE client transaction, the TU MUST
         wait until the transaction reaches the completed or terminated
         state before initiating the new INVITE.

      2. If there is an ongoing INVITE server transaction, the TU MUST
         wait until the transaction reaches the confirmed or terminated
         state before initiating the new INVITE.

   However, a UA MAY initiate a regular transaction while an INVITE
   transaction is in progress.  A UA MAY also initiate an INVITE
   transaction while a regular transaction is in progress.

   If a UA receives a non-2xx final response to a re-INVITE, the session
   parameters MUST remain unchanged, as if no re-INVITE had been issued.
   Note that, as stated in <a href="#section-12.2.1.2">Section 12.2.1.2</a>, if the non-2xx final
   response is a 481 (Call/Transaction Does Not Exist), or a 408
   (Request Timeout), or no response at all is received for the re-
   INVITE (that is, a timeout is returned by the INVITE client
   transaction), the UAC will terminate the dialog.






<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 87]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-88" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   If a UAC receives a 491 response to a re-INVITE, it SHOULD start a
   timer with a value T chosen as follows:

      1. If the UAC is the owner of the Call-ID of the dialog ID
         (meaning it generated the value), T has a randomly chosen value
         between 2.1 and 4 seconds in units of 10 ms.

      2. If the UAC is not the owner of the Call-ID of the dialog ID, T
         has a randomly chosen value of between 0 and 2 seconds in units
         of 10 ms.

   When the timer fires, the UAC SHOULD attempt the re-INVITE once more,
   if it still desires for that session modification to take place.  For
   example, if the call was already hung up with a BYE, the re-INVITE
   would not take place.

   The rules for transmitting a re-INVITE and for generating an ACK for
   a 2xx response to re-INVITE are the same as for the initial INVITE
   (<a href="#section-13.2.1">Section 13.2.1</a>).

<span class="h3"><a class="selflink" id="section-14.2" href="#section-14.2">14.2</a> UAS Behavior</span>

   <a href="#section-13.3.1">Section 13.3.1</a> describes the procedure for distinguishing incoming
   re-INVITEs from incoming initial INVITEs and handling a re-INVITE for
   an existing dialog.

   A UAS that receives a second INVITE before it sends the final
   response to a first INVITE with a lower CSeq sequence number on the
   same dialog MUST return a 500 (Server Internal Error) response to the
   second INVITE and MUST include a Retry-After header field with a
   randomly chosen value of between 0 and 10 seconds.

   A UAS that receives an INVITE on a dialog while an INVITE it had sent
   on that dialog is in progress MUST return a 491 (Request Pending)
   response to the received INVITE.

   If a UA receives a re-INVITE for an existing dialog, it MUST check
   any version identifiers in the session description or, if there are
   no version identifiers, the content of the session description to see
   if it has changed.  If the session description has changed, the UAS
   MUST adjust the session parameters accordingly, possibly after asking
   the user for confirmation.

      Versioning of the session description can be used to accommodate
      the capabilities of new arrivals to a conference, add or delete
      media, or change from a unicast to a multicast conference.





<span class="grey">Rosenberg, et. al.          Standards Track                    [Page 88]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-89" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   If the new session description is not acceptable, the UAS can reject
   it by returning a 488 (Not Acceptable Here) response for the re-
   INVITE.  This response SHOULD include a Warning header field.

   If a UAS generates a 2xx response and never receives an ACK, it
   SHOULD generate a BYE to terminate the dialog.

   A UAS MAY choose not to generate 180 (Ringing) responses for a re-
   INVITE because UACs do not typically render this information to the
   user.  For the same reason, UASs MAY choose not to use an Alert-Info
   header field or a body with Content-Disposition "alert" in responses
   to a re-INVITE.

   A UAS providing an offer in a 2xx (because the INVITE did not contain
   an offer) SHOULD construct the offer as if the UAS were making a
   brand new call, subject to the constraints of sending an offer that
   updates an existing session, as described in [<a href="#ref-13" title="&quot;An Offer/Answer Model with SDP&quot;">13</a>] in the case of SDP.
   Specifically, this means that it SHOULD include as many media formats
   and media types that the UA is willing to support.  The UAS MUST
   ensure that the session description overlaps with its previous
   session description in media formats, transports, or other parameters
   that require support from the peer.  This is to avoid the need for
   the peer to reject the session description.  If, however, it is
   unacceptable to the UAC, the UAC SHOULD generate an answer with a
   valid session description, and then send a BYE to terminate the
   session.

<span class="h2"><a class="selflink" id="section-15" href="#section-15">15</a> Terminating a Session</span>

   This section describes the procedures for terminating a session
   established by SIP.  The state of the session and the state of the
   dialog are very closely related.  When a session is initiated with an
   INVITE, each 1xx or 2xx response from a distinct UAS creates a
   dialog, and if that response completes the offer/answer exchange, it
   also creates a session.  As a result, each session is "associated"
   with a single dialog - the one which resulted in its creation.  If an
   initial INVITE generates a non-2xx final response, that terminates
   all sessions (if any) and all dialogs (if any) that were created
   through responses to the request.  By virtue of completing the
   transaction, a non-2xx final response also prevents further sessions
   from being created as a result of the INVITE.  The BYE request is
   used to terminate a specific session or attempted session.  In this
   case, the specific session is the one with the peer UA on the other
   side of the dialog.  When a BYE is received on a dialog, any session
   associated with that dialog SHOULD terminate.  A UA MUST NOT send a
   BYE outside of a dialog.  The caller's UA MAY send a BYE for either
   confirmed or early dialogs, and the callee's UA MAY send a BYE on
   confirmed dialogs, but MUST NOT send a BYE on early dialogs.



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   However, the callee's UA MUST NOT send a BYE on a confirmed dialog
   until it has received an ACK for its 2xx response or until the server
   transaction times out.  If no SIP extensions have defined other
   application layer states associated with the dialog, the BYE also
   terminates the dialog.

   The impact of a non-2xx final response to INVITE on dialogs and
   sessions makes the use of CANCEL attractive.  The CANCEL attempts to
   force a non-2xx response to the INVITE (in particular, a 487).
   Therefore, if a UAC wishes to give up on its call attempt entirely,
   it can send a CANCEL.  If the INVITE results in 2xx final response(s)
   to the INVITE, this means that a UAS accepted the invitation while
   the CANCEL was in progress.  The UAC MAY continue with the sessions
   established by any 2xx responses, or MAY terminate them with BYE.

      The notion of "hanging up" is not well defined within SIP.  It is
      specific to a particular, albeit common, user interface.
      Typically, when the user hangs up, it indicates a desire to
      terminate the attempt to establish a session, and to terminate any
      sessions already created.  For the caller's UA, this would imply a
      CANCEL request if the initial INVITE has not generated a final
      response, and a BYE to all confirmed dialogs after a final
      response.  For the callee's UA, it would typically imply a BYE;
      presumably, when the user picked up the phone, a 2xx was
      generated, and so hanging up would result in a BYE after the ACK
      is received.  This does not mean a user cannot hang up before
      receipt of the ACK, it just means that the software in his phone
      needs to maintain state for a short while in order to clean up
      properly.  If the particular UI allows for the user to reject a
      call before its answered, a 403 (Forbidden) is a good way to
      express that.  As per the rules above, a BYE can't be sent.

<span class="h3"><a class="selflink" id="section-15.1" href="#section-15.1">15.1</a> Terminating a Session with a BYE Request</span>

<span class="h4"><a class="selflink" id="section-15.1.1" href="#section-15.1.1">15.1.1</a> UAC Behavior</span>

   A BYE request is constructed as would any other request within a
   dialog, as described in <a href="#section-12">Section 12</a>.

   Once the BYE is constructed, the UAC core creates a new non-INVITE
   client transaction, and passes it the BYE request.  The UAC MUST
   consider the session terminated (and therefore stop sending or
   listening for media) as soon as the BYE request is passed to the
   client transaction.  If the response for the BYE is a 481
   (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no






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   response at all is received for the BYE (that is, a timeout is
   returned by the client transaction), the UAC MUST consider the
   session and the dialog terminated.

<span class="h4"><a class="selflink" id="section-15.1.2" href="#section-15.1.2">15.1.2</a> UAS Behavior</span>

   A UAS first processes the BYE request according to the general UAS
   processing described in <a href="#section-8.2">Section 8.2</a>.  A UAS core receiving a BYE
   request checks if it matches an existing dialog.  If the BYE does not
   match an existing dialog, the UAS core SHOULD generate a 481
   (Call/Transaction Does Not Exist) response and pass that to the
   server transaction.

      This rule means that a BYE sent without tags by a UAC will be
      rejected.  This is a change from <a href="./rfc2543">RFC 2543</a>, which allowed BYE
      without tags.

   A UAS core receiving a BYE request for an existing dialog MUST follow
   the procedures of <a href="#section-12.2.2">Section 12.2.2</a> to process the request.  Once done,
   the UAS SHOULD terminate the session (and therefore stop sending and
   listening for media).  The only case where it can elect not to are
   multicast sessions, where participation is possible even if the other
   participant in the dialog has terminated its involvement in the
   session.  Whether or not it ends its participation on the session,
   the UAS core MUST generate a 2xx response to the BYE, and MUST pass
   that to the server transaction for transmission.

   The UAS MUST still respond to any pending requests received for that
   dialog.  It is RECOMMENDED that a 487 (Request Terminated) response
   be generated to those pending requests.

<span class="h2"><a class="selflink" id="section-16" href="#section-16">16</a> Proxy Behavior</span>

<span class="h3"><a class="selflink" id="section-16.1" href="#section-16.1">16.1</a> Overview</span>

   SIP proxies are elements that route SIP requests to user agent
   servers and SIP responses to user agent clients.  A request may
   traverse several proxies on its way to a UAS.  Each will make routing
   decisions, modifying the request before forwarding it to the next
   element.  Responses will route through the same set of proxies
   traversed by the request in the reverse order.

   Being a proxy is a logical role for a SIP element.  When a request
   arrives, an element that can play the role of a proxy first decides
   if it needs to respond to the request on its own.  For instance, the
   request may be malformed or the element may need credentials from the
   client before acting as a proxy.  The element MAY respond with any




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   appropriate error code.  When responding directly to a request, the
   element is playing the role of a UAS and MUST behave as described in
   <a href="#section-8.2">Section 8.2</a>.

   A proxy can operate in either a stateful or stateless mode for each
   new request.  When stateless, a proxy acts as a simple forwarding
   element.  It forwards each request downstream to a single element
   determined by making a targeting and routing decision based on the
   request.  It simply forwards every response it receives upstream.  A
   stateless proxy discards information about a message once the message
   has been forwarded.  A stateful proxy remembers information
   (specifically, transaction state) about each incoming request and any
   requests it sends as a result of processing the incoming request.  It
   uses this information to affect the processing of future messages
   associated with that request.  A stateful proxy MAY choose to "fork"
   a request, routing it to multiple destinations.  Any request that is
   forwarded to more than one location MUST be handled statefully.

   In some circumstances, a proxy MAY forward requests using stateful
   transports (such as TCP) without being transaction-stateful.  For
   instance, a proxy MAY forward a request from one TCP connection to
   another transaction statelessly as long as it places enough
   information in the message to be able to forward the response down
   the same connection the request arrived on.  Requests forwarded
   between different types of transports where the proxy's TU must take
   an active role in ensuring reliable delivery on one of the transports
   MUST be forwarded transaction statefully.

   A stateful proxy MAY transition to stateless operation at any time
   during the processing of a request, so long as it did not do anything
   that would otherwise prevent it from being stateless initially
   (forking, for example, or generation of a 100 response).  When
   performing such a transition, all state is simply discarded.  The
   proxy SHOULD NOT initiate a CANCEL request.

   Much of the processing involved when acting statelessly or statefully
   for a request is identical.  The next several subsections are written
   from the point of view of a stateful proxy.  The last section calls
   out those places where a stateless proxy behaves differently.

<span class="h3"><a class="selflink" id="section-16.2" href="#section-16.2">16.2</a> Stateful Proxy</span>

   When stateful, a proxy is purely a SIP transaction processing engine.
   Its behavior is modeled here in terms of the server and client
   transactions defined in <a href="#section-17">Section 17</a>.  A stateful proxy has a server
   transaction associated with one or more client transactions by a
   higher layer proxy processing component (see figure 3), known as a
   proxy core.  An incoming request is processed by a server



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   transaction.  Requests from the server transaction are passed to a
   proxy core.  The proxy core determines where to route the request,
   choosing one or more next-hop locations.  An outgoing request for
   each next-hop location is processed by its own associated client
   transaction.  The proxy core collects the responses from the client
   transactions and uses them to send responses to the server
   transaction.

   A stateful proxy creates a new server transaction for each new
   request received.  Any retransmissions of the request will then be
   handled by that server transaction per <a href="#section-17">Section 17</a>.  The proxy core
   MUST behave as a UAS with respect to sending an immediate provisional
   on that server transaction (such as 100 Trying) as described in
   <a href="#section-8.2.6">Section 8.2.6</a>.  Thus, a stateful proxy SHOULD NOT generate 100
   (Trying) responses to non-INVITE requests.

   This is a model of proxy behavior, not of software.  An
   implementation is free to take any approach that replicates the
   external behavior this model defines.

   For all new requests, including any with unknown methods, an element
   intending to proxy the request MUST:

      1. Validate the request (<a href="#section-16.3">Section 16.3</a>)

      2. Preprocess routing information (<a href="#section-16.4">Section 16.4</a>)

      3. Determine target(s) for the request (<a href="#section-16.5">Section 16.5</a>)

            +--------------------+
            |                    | +---+
            |                    | | C |
            |                    | | T |
            |                    | +---+
      +---+ |       Proxy        | +---+   CT = Client Transaction
      | S | |  "Higher" Layer    | | C |
      | T | |                    | | T |   ST = Server Transaction
      +---+ |                    | +---+
            |                    | +---+
            |                    | | C |
            |                    | | T |
            |                    | +---+
            +--------------------+

               Figure 3: Stateful Proxy Model






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      4. Forward the request to each target (<a href="#section-16.6">Section 16.6</a>)

      5. Process all responses (<a href="#section-16.7">Section 16.7</a>)

<span class="h3"><a class="selflink" id="section-16.3" href="#section-16.3">16.3</a> Request Validation</span>

   Before an element can proxy a request, it MUST verify the message's
   validity.  A valid message must pass the following checks:

      1. Reasonable Syntax

      2. URI scheme

      3. Max-Forwards

      4. (Optional) Loop Detection

      5. Proxy-Require

      6. Proxy-Authorization

   If any of these checks fail, the element MUST behave as a user agent
   server (see <a href="#section-8.2">Section 8.2</a>) and respond with an error code.

   Notice that a proxy is not required to detect merged requests and
   MUST NOT treat merged requests as an error condition.  The endpoints
   receiving the requests will resolve the merge as described in <a href="#section-8.2.2.2">Section</a>
   <a href="#section-8.2.2.2">8.2.2.2</a>.

   1. Reasonable syntax check

      The request MUST be well-formed enough to be handled with a server
      transaction.  Any components involved in the remainder of these
      Request Validation steps or the Request Forwarding section MUST be
      well-formed.  Any other components, well-formed or not, SHOULD be
      ignored and remain unchanged when the message is forwarded.  For
      instance, an element would not reject a request because of a
      malformed Date header field.  Likewise, a proxy would not remove a
      malformed Date header field before forwarding a request.

      This protocol is designed to be extended.  Future extensions may
      define new methods and header fields at any time.  An element MUST
      NOT refuse to proxy a request because it contains a method or
      header field it does not know about.







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   2. URI scheme check

      If the Request-URI has a URI whose scheme is not understood by the
      proxy, the proxy SHOULD reject the request with a 416 (Unsupported
      URI Scheme) response.

   3. Max-Forwards check

      The Max-Forwards header field (<a href="#section-20.22">Section 20.22</a>) is used to limit the
      number of elements a SIP request can traverse.

      If the request does not contain a Max-Forwards header field, this
      check is passed.

      If the request contains a Max-Forwards header field with a field
      value greater than zero, the check is passed.

      If the request contains a Max-Forwards header field with a field
      value of zero (0), the element MUST NOT forward the request.  If
      the request was for OPTIONS, the element MAY act as the final
      recipient and respond per <a href="#section-11">Section 11</a>.  Otherwise, the element MUST
      return a 483 (Too many hops) response.

   4. Optional Loop Detection check

      An element MAY check for forwarding loops before forwarding a
      request.  If the request contains a Via header field with a sent-
      by value that equals a value placed into previous requests by the
      proxy, the request has been forwarded by this element before.  The
      request has either looped or is legitimately spiraling through the
      element.  To determine if the request has looped, the element MAY
      perform the branch parameter calculation described in Step 8 of
      <a href="#section-16.6">Section 16.6</a> on this message and compare it to the parameter
      received in that Via header field.  If the parameters match, the
      request has looped.  If they differ, the request is spiraling, and
      processing continues.  If a loop is detected, the element MAY
      return a 482 (Loop Detected) response.

   5. Proxy-Require check

      Future extensions to this protocol may introduce features that
      require special handling by proxies.  Endpoints will include a
      Proxy-Require header field in requests that use these features,
      telling the proxy not to process the request unless the feature is
      understood.






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      If the request contains a Proxy-Require header field (<a href="#section-20.29">Section</a>
      <a href="#section-20.29">20.29</a>) with one or more option-tags this element does not
      understand, the element MUST return a 420 (Bad Extension)
      response.  The response MUST include an Unsupported (<a href="#section-20.40">Section</a>
      <a href="#section-20.40">20.40</a>) header field listing those option-tags the element did not
      understand.

   6. Proxy-Authorization check

      If an element requires credentials before forwarding a request,
      the request MUST be inspected as described in <a href="#section-22.3">Section 22.3</a>.  That
      section also defines what the element must do if the inspection
      fails.

<span class="h3"><a class="selflink" id="section-16.4" href="#section-16.4">16.4</a> Route Information Preprocessing</span>

   The proxy MUST inspect the Request-URI of the request.  If the
   Request-URI of the request contains a value this proxy previously
   placed into a Record-Route header field (see <a href="#section-16.6">Section 16.6</a> item 4),
   the proxy MUST replace the Request-URI in the request with the last
   value from the Route header field, and remove that value from the
   Route header field.  The proxy MUST then proceed as if it received
   this modified request.

      This will only happen when the element sending the request to the
      proxy (which may have been an endpoint) is a strict router.  This
      rewrite on receive is necessary to enable backwards compatibility
      with those elements.  It also allows elements following this
      specification to preserve the Request-URI through strict-routing
      proxies (see <a href="#section-12.2.1.1">Section 12.2.1.1</a>).

      This requirement does not obligate a proxy to keep state in order
      to detect URIs it previously placed in Record-Route header fields.
      Instead, a proxy need only place enough information in those URIs
      to recognize them as values it provided when they later appear.

   If the Request-URI contains a maddr parameter, the proxy MUST check
   to see if its value is in the set of addresses or domains the proxy
   is configured to be responsible for.  If the Request-URI has a maddr
   parameter with a value the proxy is responsible for, and the request
   was received using the port and transport indicated (explicitly or by
   default) in the Request-URI, the proxy MUST strip the maddr and any
   non-default port or transport parameter and continue processing as if
   those values had not been present in the request.







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      A request may arrive with a maddr matching the proxy, but on a
      port or transport different from that indicated in the URI.  Such
      a request needs to be forwarded to the proxy using the indicated
      port and transport.

   If the first value in the Route header field indicates this proxy,
   the proxy MUST remove that value from the request.

<span class="h3"><a class="selflink" id="section-16.5" href="#section-16.5">16.5</a> Determining Request Targets</span>

   Next, the proxy calculates the target(s) of the request.  The set of
   targets will either be predetermined by the contents of the request
   or will be obtained from an abstract location service.  Each target
   in the set is represented as a URI.

   If the Request-URI of the request contains an maddr parameter, the
   Request-URI MUST be placed into the target set as the only target
   URI, and the proxy MUST proceed to <a href="#section-16.6">Section 16.6</a>.

   If the domain of the Request-URI indicates a domain this element is
   not responsible for, the Request-URI MUST be placed into the target
   set as the only target, and the element MUST proceed to the task of
   Request Forwarding (<a href="#section-16.6">Section 16.6</a>).

      There are many circumstances in which a proxy might receive a
      request for a domain it is not responsible for.  A firewall proxy
      handling outgoing calls (the way HTTP proxies handle outgoing
      requests) is an example of where this is likely to occur.

   If the target set for the request has not been predetermined as
   described above, this implies that the element is responsible for the
   domain in the Request-URI, and the element MAY use whatever mechanism
   it desires to determine where to send the request.  Any of these
   mechanisms can be modeled as accessing an abstract Location Service.
   This may consist of obtaining information from a location service
   created by a SIP Registrar, reading a database, consulting a presence
   server, utilizing other protocols, or simply performing an
   algorithmic substitution on the Request-URI.  When accessing the
   location service constructed by a registrar, the Request-URI MUST
   first be canonicalized as described in <a href="#section-10.3">Section 10.3</a> before being used
   as an index.  The output of these mechanisms is used to construct the
   target set.

   If the Request-URI does not provide sufficient information for the
   proxy to determine the target set, it SHOULD return a 485 (Ambiguous)
   response.  This response SHOULD contain a Contact header field
   containing URIs of new addresses to be tried.  For example, an INVITE




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   to sip:John.Smith@company.com may be ambiguous at a proxy whose
   location service has multiple John Smiths listed.  See <a href="#section-21.4.23">Section</a>
   <a href="#section-21.4.23">21.4.23</a> for details.

   Any information in or about the request or the current environment of
   the element MAY be used in the construction of the target set.  For
   instance, different sets may be constructed depending on contents or
   the presence of header fields and bodies, the time of day of the
   request's arrival, the interface on which the request arrived,
   failure of previous requests, or even the element's current level of
   utilization.

   As potential targets are located through these services, their URIs
   are added to the target set.  Targets can only be placed in the
   target set once.  If a target URI is already present in the set
   (based on the definition of equality for the URI type), it MUST NOT
   be added again.

   A proxy MUST NOT add additional targets to the target set if the
   Request-URI of the original request does not indicate a resource this
   proxy is responsible for.

      A proxy can only change the Request-URI of a request during
      forwarding if it is responsible for that URI.  If the proxy is not
      responsible for that URI, it will not recurse on 3xx or 416
      responses as described below.

   If the Request-URI of the original request indicates a resource this
   proxy is responsible for, the proxy MAY continue to add targets to
   the set after beginning Request Forwarding.  It MAY use any
   information obtained during that processing to determine new targets.
   For instance, a proxy may choose to incorporate contacts obtained in
   a redirect response (3xx) into the target set.  If a proxy uses a
   dynamic source of information while building the target set (for
   instance, if it consults a SIP Registrar), it SHOULD monitor that
   source for the duration of processing the request.  New locations
   SHOULD be added to the target set as they become available.  As
   above, any given URI MUST NOT be added to the set more than once.

      Allowing a URI to be added to the set only once reduces
      unnecessary network traffic, and in the case of incorporating
      contacts from redirect requests prevents infinite recursion.

   For example, a trivial location service is a "no-op", where the
   target URI is equal to the incoming request URI.  The request is sent
   to a specific next hop proxy for further processing.  During request





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   forwarding of <a href="#section-16.6">Section 16.6</a>, Item 6, the identity of that next hop,
   expressed as a SIP or SIPS URI, is inserted as the top-most Route
   header field value into the request.

   If the Request-URI indicates a resource at this proxy that does not
   exist, the proxy MUST return a 404 (Not Found) response.

   If the target set remains empty after applying all of the above, the
   proxy MUST return an error response, which SHOULD be the 480
   (Temporarily Unavailable) response.

<span class="h3"><a class="selflink" id="section-16.6" href="#section-16.6">16.6</a> Request Forwarding</span>

   As soon as the target set is non-empty, a proxy MAY begin forwarding
   the request.  A stateful proxy MAY process the set in any order.  It
   MAY process multiple targets serially, allowing each client
   transaction to complete before starting the next.  It MAY start
   client transactions with every target in parallel.  It also MAY
   arbitrarily divide the set into groups, processing the groups
   serially and processing the targets in each group in parallel.

   A common ordering mechanism is to use the qvalue parameter of targets
   obtained from Contact header fields (see <a href="#section-20.10">Section 20.10</a>).  Targets are
   processed from highest qvalue to lowest.  Targets with equal qvalues
   may be processed in parallel.

   A stateful proxy must have a mechanism to maintain the target set as
   responses are received and associate the responses to each forwarded
   request with the original request.  For the purposes of this model,
   this mechanism is a "response context" created by the proxy layer
   before forwarding the first request.

   For each target, the proxy forwards the request following these
   steps:

      1.  Make a copy of the received request

      2.  Update the Request-URI

      3.  Update the Max-Forwards header field

      4.  Optionally add a Record-route header field value

      5.  Optionally add additional header fields

      6.  Postprocess routing information

      7.  Determine the next-hop address, port, and transport



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      8.  Add a Via header field value

      9.  Add a Content-Length header field if necessary

      10. Forward the new request

      11. Set timer C

   Each of these steps is detailed below:

      1. Copy request

         The proxy starts with a copy of the received request.  The copy
         MUST initially contain all of the header fields from the
         received request.  Fields not detailed in the processing
         described below MUST NOT be removed.  The copy SHOULD maintain
         the ordering of the header fields as in the received request.
         The proxy MUST NOT reorder field values with a common field
         name (See <a href="#section-7.3.1">Section 7.3.1</a>).  The proxy MUST NOT add to, modify,
         or remove the message body.

         An actual implementation need not perform a copy; the primary
         requirement is that the processing for each next hop begin with
         the same request.

      2. Request-URI

         The Request-URI in the copy's start line MUST be replaced with
         the URI for this target.  If the URI contains any parameters
         not allowed in a Request-URI, they MUST be removed.

         This is the essence of a proxy's role.  This is the mechanism
         through which a proxy routes a request toward its destination.

         In some circumstances, the received Request-URI is placed into
         the target set without being modified.  For that target, the
         replacement above is effectively a no-op.

      3. Max-Forwards

         If the copy contains a Max-Forwards header field, the proxy
         MUST decrement its value by one (1).

         If the copy does not contain a Max-Forwards header field, the
         proxy MUST add one with a field value, which SHOULD be 70.

         Some existing UAs will not provide a Max-Forwards header field
         in a request.



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      4. Record-Route

         If this proxy wishes to remain on the path of future requests
         in a dialog created by this request (assuming the request
         creates a dialog), it MUST insert a Record-Route header field
         value into the copy before any existing Record-Route header
         field values, even if a Route header field is already present.

         Requests establishing a dialog may contain a preloaded Route
         header field.

         If this request is already part of a dialog, the proxy SHOULD
         insert a Record-Route header field value if it wishes to remain
         on the path of future requests in the dialog.  In normal
         endpoint operation as described in <a href="#section-12">Section 12</a>, these Record-
         Route header field values will not have any effect on the route
         sets used by the endpoints.

         The proxy will remain on the path if it chooses to not insert a
         Record-Route header field value into requests that are already
         part of a dialog.  However, it would be removed from the path
         when an endpoint that has failed reconstitutes the dialog.

         A proxy MAY insert a Record-Route header field value into any
         request.  If the request does not initiate a dialog, the
         endpoints will ignore the value.  See <a href="#section-12">Section 12</a> for details on
         how endpoints use the Record-Route header field values to
         construct Route header fields.

         Each proxy in the path of a request chooses whether to add a
         Record-Route header field value independently - the presence of
         a Record-Route header field in a request does not obligate this
         proxy to add a value.

         The URI placed in the Record-Route header field value MUST be a
         SIP or SIPS URI.  This URI MUST contain an lr parameter (see
         <a href="#section-19.1.1">Section 19.1.1</a>).  This URI MAY be different for each
         destination the request is forwarded to.  The URI SHOULD NOT
         contain the transport parameter unless the proxy has knowledge
         (such as in a private network) that the next downstream element
         that will be in the path of subsequent requests supports that
         transport.

         The URI this proxy provides will be used by some other element
         to make a routing decision.  This proxy, in general, has no way
         of knowing the capabilities of that element, so it must
         restrict itself to the mandatory elements of a SIP
         implementation: SIP URIs and either the TCP or UDP transports.



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         The URI placed in the Record-Route header field MUST resolve to
         the element inserting it (or a suitable stand-in) when the
         server location procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] are applied to it, so that
         subsequent requests reach the same SIP element.  If the
         Request-URI contains a SIPS URI, or the topmost Route header
         field value (after the post processing of bullet 6) contains a
         SIPS URI, the URI placed into the Record-Route header field
         MUST be a SIPS URI.  Furthermore, if the request was not
         received over TLS, the proxy MUST insert a Record-Route header
         field.  In a similar fashion, a proxy that receives a request
         over TLS, but generates a request without a SIPS URI in the
         Request-URI or topmost Route header field value (after the post
         processing of bullet 6), MUST insert a Record-Route header
         field that is not a SIPS URI.

         A proxy at a security perimeter must remain on the perimeter
         throughout the dialog.

         If the URI placed in the Record-Route header field needs to be
         rewritten when it passes back through in a response, the URI
         MUST be distinct enough to locate at that time.  (The request
         may spiral through this proxy, resulting in more than one
         Record-Route header field value being added).  Item 8 of
         <a href="#section-16.7">Section 16.7</a> recommends a mechanism to make the URI
         sufficiently distinct.

         The proxy MAY include parameters in the Record-Route header
         field value.  These will be echoed in some responses to the
         request such as the 200 (OK) responses to INVITE.  Such
         parameters may be useful for keeping state in the message
         rather than the proxy.

         If a proxy needs to be in the path of any type of dialog (such
         as one straddling a firewall), it SHOULD add a Record-Route
         header field value to every request with a method it does not
         understand since that method may have dialog semantics.

         The URI a proxy places into a Record-Route header field is only
         valid for the lifetime of any dialog created by the transaction
         in which it occurs.  A dialog-stateful proxy, for example, MAY
         refuse to accept future requests with that value in the
         Request-URI after the dialog has terminated.  Non-dialog-
         stateful proxies, of course, have no concept of when the dialog
         has terminated, but they MAY encode enough information in the
         value to compare it against the dialog identifier of future
         requests and MAY reject requests not matching that information.
         Endpoints MUST NOT use a URI obtained from a Record-Route
         header field outside the dialog in which it was provided.  See



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         <a href="#section-12">Section 12</a> for more information on an endpoint's use of
         Record-Route header fields.

         Record-routing may be required by certain services where the
         proxy needs to observe all messages in a dialog.  However, it
         slows down processing and impairs scalability and thus proxies
         should only record-route if required for a particular service.

         The Record-Route process is designed to work for any SIP
         request that initiates a dialog.  INVITE is the only such
         request in this specification, but extensions to the protocol
         MAY define others.

      5. Add Additional Header Fields

         The proxy MAY add any other appropriate header fields to the
         copy at this point.

      6. Postprocess routing information

         A proxy MAY have a local policy that mandates that a request
         visit a specific set of proxies before being delivered to the
         destination.  A proxy MUST ensure that all such proxies are
         loose routers.  Generally, this can only be known with
         certainty if the proxies are within the same administrative
         domain.  This set of proxies is represented by a set of URIs
         (each of which contains the lr parameter).  This set MUST be
         pushed into the Route header field of the copy ahead of any
         existing values, if present.  If the Route header field is
         absent, it MUST be added, containing that list of URIs.

         If the proxy has a local policy that mandates that the request
         visit one specific proxy, an alternative to pushing a Route
         value into the Route header field is to bypass the forwarding
         logic of item 10 below, and instead just send the request to
         the address, port, and transport for that specific proxy.  If
         the request has a Route header field, this alternative MUST NOT
         be used unless it is known that next hop proxy is a loose
         router.  Otherwise, this approach MAY be used, but the Route
         insertion mechanism above is preferred for its robustness,
         flexibility, generality and consistency of operation.
         Furthermore, if the Request-URI contains a SIPS URI, TLS MUST
         be used to communicate with that proxy.

         If the copy contains a Route header field, the proxy MUST
         inspect the URI in its first value.  If that URI does not
         contain an lr parameter, the proxy MUST modify the copy as
         follows:



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         -  The proxy MUST place the Request-URI into the Route header
            field as the last value.

         -  The proxy MUST then place the first Route header field value
            into the Request-URI and remove that value from the Route
            header field.

         Appending the Request-URI to the Route header field is part of
         a mechanism used to pass the information in that Request-URI
         through strict-routing elements.  "Popping" the first Route
         header field value into the Request-URI formats the message the
         way a strict-routing element expects to receive it (with its
         own URI in the Request-URI and the next location to visit in
         the first Route header field value).

      7. Determine Next-Hop Address, Port, and Transport

         The proxy MAY have a local policy to send the request to a
         specific IP address, port, and transport, independent of the
         values of the Route and Request-URI.  Such a policy MUST NOT be
         used if the proxy is not certain that the IP address, port, and
         transport correspond to a server that is a loose router.
         However, this mechanism for sending the request through a
         specific next hop is NOT RECOMMENDED; instead a Route header
         field should be used for that purpose as described above.

         In the absence of such an overriding mechanism, the proxy
         applies the procedures listed in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] as follows to determine
         where to send the request.  If the proxy has reformatted the
         request to send to a strict-routing element as described in
         step 6 above, the proxy MUST apply those procedures to the
         Request-URI of the request.  Otherwise, the proxy MUST apply
         the procedures to the first value in the Route header field, if
         present, else the Request-URI.  The procedures will produce an
         ordered set of (address, port, transport) tuples.
         Independently of which URI is being used as input to the
         procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>], if the Request-URI specifies a SIPS
         resource, the proxy MUST follow the procedures of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] as if the
         input URI were a SIPS URI.

         As described in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>], the proxy MUST attempt to deliver the
         message to the first tuple in that set, and proceed through the
         set in order until the delivery attempt succeeds.

         For each tuple attempted, the proxy MUST format the message as
         appropriate for the tuple and send the request using a new
         client transaction as detailed in steps 8 through 10.




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         Since each attempt uses a new client transaction, it represents
         a new branch.  Thus, the branch parameter provided with the Via
         header field inserted in step 8 MUST be different for each
         attempt.

         If the client transaction reports failure to send the request
         or a timeout from its state machine, the proxy continues to the
         next address in that ordered set.  If the ordered set is
         exhausted, the request cannot be forwarded to this element in
         the target set.  The proxy does not need to place anything in
         the response context, but otherwise acts as if this element of
         the target set returned a 408 (Request Timeout) final response.

      8. Add a Via header field value

         The proxy MUST insert a Via header field value into the copy
         before the existing Via header field values.  The construction
         of this value follows the same guidelines of <a href="#section-8.1.1.7">Section 8.1.1.7</a>.
         This implies that the proxy will compute its own branch
         parameter, which will be globally unique for that branch, and
         contain the requisite magic cookie. Note that this implies that
         the branch parameter will be different for different instances
         of a spiraled or looped request through a proxy.

         Proxies choosing to detect loops have an additional constraint
         in the value they use for construction of the branch parameter.
         A proxy choosing to detect loops SHOULD create a branch
         parameter separable into two parts by the implementation.  The
         first part MUST satisfy the constraints of <a href="#section-8.1.1.7">Section 8.1.1.7</a> as
         described above.  The second is used to perform loop detection
         and distinguish loops from spirals.

         Loop detection is performed by verifying that, when a request
         returns to a proxy, those fields having an impact on the
         processing of the request have not changed.  The value placed
         in this part of the branch parameter SHOULD reflect all of
         those fields (including any Route, Proxy-Require and Proxy-
         Authorization header fields).  This is to ensure that if the
         request is routed back to the proxy and one of those fields
         changes, it is treated as a spiral and not a loop (see <a href="#section-16.3">Section</a>
         <a href="#section-16.3">16.3</a>).  A common way to create this value is to compute a
         cryptographic hash of the To tag, From tag, Call-ID header
         field, the Request-URI of the request received (before
         translation), the topmost Via header, and the sequence number
         from the CSeq header field, in addition to any Proxy-Require
         and Proxy-Authorization header fields that may be present.  The





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         algorithm used to compute the hash is implementation-dependent,
         but MD5 (<a href="./rfc1321">RFC 1321</a> [<a href="#ref-35" title="&quot;The MD5 Message-Digest Algorithm&quot;">35</a>]), expressed in hexadecimal, is a
         reasonable choice.  (Base64 is not permissible for a token.)

         If a proxy wishes to detect loops, the "branch" parameter it
         supplies MUST depend on all information affecting processing of
         a request, including the incoming Request-URI and any header
         fields affecting the request's admission or routing.  This is
         necessary to distinguish looped requests from requests whose
         routing parameters have changed before returning to this
         server.

         The request method MUST NOT be included in the calculation of
         the branch parameter.  In particular, CANCEL and ACK requests
         (for non-2xx responses) MUST have the same branch value as the
         corresponding request they cancel or acknowledge.  The branch
         parameter is used in correlating those requests at the server
         handling them (see Sections <a href="#section-17.2.3">17.2.3</a> and <a href="#section-9.2">9.2</a>).

      9. Add a Content-Length header field if necessary

         If the request will be sent to the next hop using a stream-
         based transport and the copy contains no Content-Length header
         field, the proxy MUST insert one with the correct value for the
         body of the request (see <a href="#section-20.14">Section 20.14</a>).

      10. Forward Request

         A stateful proxy MUST create a new client transaction for this
         request as described in <a href="#section-17.1">Section 17.1</a> and instructs the
         transaction to send the request using the address, port and
         transport determined in step 7.

      11. Set timer C

         In order to handle the case where an INVITE request never
         generates a final response, the TU uses a timer which is called
         timer C.  Timer C MUST be set for each client transaction when
         an INVITE request is proxied.  The timer MUST be larger than 3
         minutes.  <a href="#section-16.7">Section 16.7</a> bullet 2 discusses how this timer is
         updated with provisional responses, and <a href="#section-16.8">Section 16.8</a> discusses
         processing when it fires.









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<span class="h3"><a class="selflink" id="section-16.7" href="#section-16.7">16.7</a> Response Processing</span>

   When a response is received by an element, it first tries to locate a
   client transaction (<a href="#section-17.1.3">Section 17.1.3</a>) matching the response.  If none
   is found, the element MUST process the response (even if it is an
   informational response) as a stateless proxy (described below).  If a
   match is found, the response is handed to the client transaction.

      Forwarding responses for which a client transaction (or more
      generally any knowledge of having sent an associated request) is
      not found improves robustness.  In particular, it ensures that
      "late" 2xx responses to INVITE requests are forwarded properly.

   As client transactions pass responses to the proxy layer, the
   following processing MUST take place:

      1.  Find the appropriate response context

      2.  Update timer C for provisional responses

      3.  Remove the topmost Via

      4.  Add the response to the response context

      5.  Check to see if this response should be forwarded immediately

      6.  When necessary, choose the best final response from the
          response context

   If no final response has been forwarded after every client
   transaction associated with the response context has been terminated,
   the proxy must choose and forward the "best" response from those it
   has seen so far.

   The following processing MUST be performed on each response that is
   forwarded.  It is likely that more than one response to each request
   will be forwarded: at least each provisional and one final response.

      7.  Aggregate authorization header field values if necessary

      8.  Optionally rewrite Record-Route header field values

      9.  Forward the response

      10. Generate any necessary CANCEL requests






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   Each of the above steps are detailed below:

      1.  Find Context

         The proxy locates the "response context" it created before
         forwarding the original request using the key described in
         <a href="#section-16.6">Section 16.6</a>.  The remaining processing steps take place in
         this context.

      2.  Update timer C for provisional responses

         For an INVITE transaction, if the response is a provisional
         response with status codes 101 to 199 inclusive (i.e., anything
         but 100), the proxy MUST reset timer C for that client
         transaction.  The timer MAY be reset to a different value, but
         this value MUST be greater than 3 minutes.

      3.  Via

         The proxy removes the topmost Via header field value from the
         response.

         If no Via header field values remain in the response, the
         response was meant for this element and MUST NOT be forwarded.
         The remainder of the processing described in this section is
         not performed on this message, the UAC processing rules
         described in <a href="#section-8.1.3">Section 8.1.3</a> are followed instead (transport
         layer processing has already occurred).

         This will happen, for instance, when the element generates
         CANCEL requests as described in <a href="#section-10">Section 10</a>.

      4.  Add response to context

         Final responses received are stored in the response context
         until a final response is generated on the server transaction
         associated with this context.  The response may be a candidate
         for the best final response to be returned on that server
         transaction.  Information from this response may be needed in
         forming the best response, even if this response is not chosen.

         If the proxy chooses to recurse on any contacts in a 3xx
         response by adding them to the target set, it MUST remove them
         from the response before adding the response to the response
         context.  However, a proxy SHOULD NOT recurse to a non-SIPS URI
         if the Request-URI of the original request was a SIPS URI.  If





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         the proxy recurses on all of the contacts in a 3xx response,
         the proxy SHOULD NOT add the resulting contactless response to
         the response context.

         Removing the contact before adding the response to the response
         context prevents the next element upstream from retrying a
         location this proxy has already attempted.

         3xx responses may contain a mixture of SIP, SIPS, and non-SIP
         URIs.  A proxy may choose to recurse on the SIP and SIPS URIs
         and place the remainder into the response context to be
         returned, potentially in the final response.

         If a proxy receives a 416 (Unsupported URI Scheme) response to
         a request whose Request-URI scheme was not SIP, but the scheme
         in the original received request was SIP or SIPS (that is, the
         proxy changed the scheme from SIP or SIPS to something else
         when it proxied a request), the proxy SHOULD add a new URI to
         the target set.  This URI SHOULD be a SIP URI version of the
         non-SIP URI that was just tried.  In the case of the tel URL,
         this is accomplished by placing the telephone-subscriber part
         of the tel URL into the user part of the SIP URI, and setting
         the hostpart to the domain where the prior request was sent.
         See <a href="#section-19.1.6">Section 19.1.6</a> for more detail on forming SIP URIs from tel
         URLs.

         As with a 3xx response, if a proxy "recurses" on the 416 by
         trying a SIP or SIPS URI instead, the 416 response SHOULD NOT
         be added to the response context.

      5.  Check response for forwarding

         Until a final response has been sent on the server transaction,
         the following responses MUST be forwarded immediately:

         -  Any provisional response other than 100 (Trying)

         -  Any 2xx response

         If a 6xx response is received, it is not immediately forwarded,
         but the stateful proxy SHOULD cancel all client pending
         transactions as described in <a href="#section-10">Section 10</a>, and it MUST NOT create
         any new branches in this context.

         This is a change from <a href="./rfc2543">RFC 2543</a>, which mandated that the proxy
         was to forward the 6xx response immediately.  For an INVITE
         transaction, this approach had the problem that a 2xx response
         could arrive on another branch, in which case the proxy would



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         have to forward the 2xx.  The result was that the UAC could
         receive a 6xx response followed by a 2xx response, which should
         never be allowed to happen.  Under the new rules, upon
         receiving a 6xx, a proxy will issue a CANCEL request, which
         will generally result in 487 responses from all outstanding
         client transactions, and then at that point the 6xx is
         forwarded upstream.

         After a final response has been sent on the server transaction,
         the following responses MUST be forwarded immediately:

         -  Any 2xx response to an INVITE request

         A stateful proxy MUST NOT immediately forward any other
         responses.  In particular, a stateful proxy MUST NOT forward
         any 100 (Trying) response.  Those responses that are candidates
         for forwarding later as the "best" response have been gathered
         as described in step "Add Response to Context".

         Any response chosen for immediate forwarding MUST be processed
         as described in steps "Aggregate Authorization Header Field
         Values" through "Record-Route".

         This step, combined with the next, ensures that a stateful
         proxy will forward exactly one final response to a non-INVITE
         request, and either exactly one non-2xx response or one or more
         2xx responses to an INVITE request.

      6.  Choosing the best response

         A stateful proxy MUST send a final response to a response
         context's server transaction if no final responses have been
         immediately forwarded by the above rules and all client
         transactions in this response context have been terminated.

         The stateful proxy MUST choose the "best" final response among
         those received and stored in the response context.

         If there are no final responses in the context, the proxy MUST
         send a 408 (Request Timeout) response to the server
         transaction.

         Otherwise, the proxy MUST forward a response from the responses
         stored in the response context.  It MUST choose from the 6xx
         class responses if any exist in the context.  If no 6xx class
         responses are present, the proxy SHOULD choose from the lowest
         response class stored in the response context.  The proxy MAY
         select any response within that chosen class.  The proxy SHOULD



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         give preference to responses that provide information affecting
         resubmission of this request, such as 401, 407, 415, 420, and
         484 if the 4xx class is chosen.

         A proxy which receives a 503 (Service Unavailable) response
         SHOULD NOT forward it upstream unless it can determine that any
         subsequent requests it might proxy will also generate a 503.
         In other words, forwarding a 503 means that the proxy knows it
         cannot service any requests, not just the one for the Request-
         URI in the request which generated the 503.  If the only
         response that was received is a 503, the proxy SHOULD generate
         a 500 response and forward that upstream.

         The forwarded response MUST be processed as described in steps
         "Aggregate Authorization Header Field Values" through "Record-
         Route".

         For example, if a proxy forwarded a request to 4 locations, and
         received 503, 407, 501, and 404 responses, it may choose to
         forward the 407 (Proxy Authentication Required) response.

         1xx and 2xx responses may be involved in the establishment of
         dialogs.  When a request does not contain a To tag, the To tag
         in the response is used by the UAC to distinguish multiple
         responses to a dialog creating request.  A proxy MUST NOT
         insert a tag into the To header field of a 1xx or 2xx response
         if the request did not contain one.  A proxy MUST NOT modify
         the tag in the To header field of a 1xx or 2xx response.

         Since a proxy may not insert a tag into the To header field of
         a 1xx response to a request that did not contain one, it cannot
         issue non-100 provisional responses on its own.  However, it
         can branch the request to a UAS sharing the same element as the
         proxy.  This UAS can return its own provisional responses,
         entering into an early dialog with the initiator of the
         request.  The UAS does not have to be a discreet process from
         the proxy.  It could be a virtual UAS implemented in the same
         code space as the proxy.

         3-6xx responses are delivered hop-by-hop.  When issuing a 3-6xx
         response, the element is effectively acting as a UAS, issuing
         its own response, usually based on the responses received from
         downstream elements.  An element SHOULD preserve the To tag
         when simply forwarding a 3-6xx response to a request that did
         not contain a To tag.

         A proxy MUST NOT modify the To tag in any forwarded response to
         a request that contains a To tag.



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         While it makes no difference to the upstream elements if the
         proxy replaced the To tag in a forwarded 3-6xx response,
         preserving the original tag may assist with debugging.

         When the proxy is aggregating information from several
         responses, choosing a To tag from among them is arbitrary, and
         generating a new To tag may make debugging easier.  This
         happens, for instance, when combining 401 (Unauthorized) and
         407 (Proxy Authentication Required) challenges, or combining
         Contact values from unencrypted and unauthenticated 3xx
         responses.

      7.  Aggregate Authorization Header Field Values

         If the selected response is a 401 (Unauthorized) or 407 (Proxy
         Authentication Required), the proxy MUST collect any WWW-
         Authenticate and Proxy-Authenticate header field values from
         all other 401 (Unauthorized) and 407 (Proxy Authentication
         Required) responses received so far in this response context
         and add them to this response without modification before
         forwarding.  The resulting 401 (Unauthorized) or 407 (Proxy
         Authentication Required) response could have several WWW-
         Authenticate AND Proxy-Authenticate header field values.

         This is necessary because any or all of the destinations the
         request was forwarded to may have requested credentials.  The
         client needs to receive all of those challenges and supply
         credentials for each of them when it retries the request.
         Motivation for this behavior is provided in <a href="#section-26">Section 26</a>.

      8.  Record-Route

         If the selected response contains a Record-Route header field
         value originally provided by this proxy, the proxy MAY choose
         to rewrite the value before forwarding the response.  This
         allows the proxy to provide different URIs for itself to the
         next upstream and downstream elements.  A proxy may choose to
         use this mechanism for any reason.  For instance, it is useful
         for multi-homed hosts.

         If the proxy received the request over TLS, and sent it out
         over a non-TLS connection, the proxy MUST rewrite the URI in
         the Record-Route header field to be a SIPS URI.  If the proxy
         received the request over a non-TLS connection, and sent it out
         over TLS, the proxy MUST rewrite the URI in the Record-Route
         header field to be a SIP URI.





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


         The new URI provided by the proxy MUST satisfy the same
         constraints on URIs placed in Record-Route header fields in
         requests (see Step 4 of <a href="#section-16.6">Section 16.6</a>) with the following
         modifications:

         The URI SHOULD NOT contain the transport parameter unless the
         proxy has knowledge that the next upstream (as opposed to
         downstream) element that will be in the path of subsequent
         requests supports that transport.

         When a proxy does decide to modify the Record-Route header
         field in the response, one of the operations it performs is
         locating the Record-Route value that it had inserted.  If the
         request spiraled, and the proxy inserted a Record-Route value
         in each iteration of the spiral, locating the correct value in
         the response (which must be the proper iteration in the reverse
         direction) is tricky.  The rules above recommend that a proxy
         wishing to rewrite Record-Route header field values insert
         sufficiently distinct URIs into the Record-Route header field
         so that the right one may be selected for rewriting.  A
         RECOMMENDED mechanism to achieve this is for the proxy to
         append a unique identifier for the proxy instance to the user
         portion of the URI.

         When the response arrives, the proxy modifies the first
         Record-Route whose identifier matches the proxy instance.  The
         modification results in a URI without this piece of data
         appended to the user portion of the URI.  Upon the next
         iteration, the same algorithm (find the topmost Record-Route
         header field value with the parameter) will correctly extract
         the next Record-Route header field value inserted by that
         proxy.

         Not every response to a request to which a proxy adds a
         Record-Route header field value will contain a Record-Route
         header field.  If the response does contain a Record-Route
         header field, it will contain the value the proxy added.

      9.  Forward response

         After performing the processing described in steps "Aggregate
         Authorization Header Field Values" through "Record-Route", the
         proxy MAY perform any feature specific manipulations on the
         selected response.  The proxy MUST NOT add to, modify, or
         remove the message body.  Unless otherwise specified, the proxy
         MUST NOT remove any header field values other than the Via
         header field value discussed in <a href="#section-16.7">Section 16.7</a> Item 3.  In
         particular, the proxy MUST NOT remove any "received" parameter



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         it may have added to the next Via header field value while
         processing the request associated with this response.  The
         proxy MUST pass the response to the server transaction
         associated with the response context.  This will result in the
         response being sent to the location now indicated in the
         topmost Via header field value.  If the server transaction is
         no longer available to handle the transmission, the element
         MUST forward the response statelessly by sending it to the
         server transport.  The server transaction might indicate
         failure to send the response or signal a timeout in its state
         machine.  These errors would be logged for diagnostic purposes
         as appropriate, but the protocol requires no remedial action
         from the proxy.

         The proxy MUST maintain the response context until all of its
         associated transactions have been terminated, even after
         forwarding a final response.

      10. Generate CANCELs

         If the forwarded response was a final response, the proxy MUST
         generate a CANCEL request for all pending client transactions
         associated with this response context.  A proxy SHOULD also
         generate a CANCEL request for all pending client transactions
         associated with this response context when it receives a 6xx
         response.  A pending client transaction is one that has
         received a provisional response, but no final response (it is
         in the proceeding state) and has not had an associated CANCEL
         generated for it.  Generating CANCEL requests is described in
         <a href="#section-9.1">Section 9.1</a>.

         The requirement to CANCEL pending client transactions upon
         forwarding a final response does not guarantee that an endpoint
         will not receive multiple 200 (OK) responses to an INVITE.  200
         (OK) responses on more than one branch may be generated before
         the CANCEL requests can be sent and processed.  Further, it is
         reasonable to expect that a future extension may override this
         requirement to issue CANCEL requests.

<span class="h3"><a class="selflink" id="section-16.8" href="#section-16.8">16.8</a> Processing Timer C</span>

   If timer C should fire, the proxy MUST either reset the timer with
   any value it chooses, or terminate the client transaction.  If the
   client transaction has received a provisional response, the proxy
   MUST generate a CANCEL request matching that transaction.  If the
   client transaction has not received a provisional response, the proxy
   MUST behave as if the transaction received a 408 (Request Timeout)
   response.



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Allowing the proxy to reset the timer allows the proxy to dynamically
   extend the transaction's lifetime based on current conditions (such
   as utilization) when the timer fires.

<span class="h3"><a class="selflink" id="section-16.9" href="#section-16.9">16.9</a> Handling Transport Errors</span>

   If the transport layer notifies a proxy of an error when it tries to
   forward a request (see <a href="#section-18.4">Section 18.4</a>), the proxy MUST behave as if the
   forwarded request received a 503 (Service Unavailable) response.

   If the proxy is notified of an error when forwarding a response, it
   drops the response.  The proxy SHOULD NOT cancel any outstanding
   client transactions associated with this response context due to this
   notification.

      If a proxy cancels its outstanding client transactions, a single
      malicious or misbehaving client can cause all transactions to fail
      through its Via header field.

<span class="h3"><a class="selflink" id="section-16.10" href="#section-16.10">16.10</a> CANCEL Processing</span>

   A stateful proxy MAY generate a CANCEL to any other request it has
   generated at any time (subject to receiving a provisional response to
   that request as described in <a href="#section-9.1">section 9.1</a>).  A proxy MUST cancel any
   pending client transactions associated with a response context when
   it receives a matching CANCEL request.

   A stateful proxy MAY generate CANCEL requests for pending INVITE
   client transactions based on the period specified in the INVITE's
   Expires header field elapsing.  However, this is generally
   unnecessary since the endpoints involved will take care of signaling
   the end of the transaction.

   While a CANCEL request is handled in a stateful proxy by its own
   server transaction, a new response context is not created for it.
   Instead, the proxy layer searches its existing response contexts for
   the server transaction handling the request associated with this
   CANCEL.  If a matching response context is found, the element MUST
   immediately return a 200 (OK) response to the CANCEL request.  In
   this case, the element is acting as a user agent server as defined in
   <a href="#section-8.2">Section 8.2</a>.  Furthermore, the element MUST generate CANCEL requests
   for all pending client transactions in the context as described in
   <a href="#section-16.7">Section 16.7</a> step 10.

   If a response context is not found, the element does not have any
   knowledge of the request to apply the CANCEL to.  It MUST statelessly
   forward the CANCEL request (it may have statelessly forwarded the
   associated request previously).



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-116" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h3"><a class="selflink" id="section-16.11" href="#section-16.11">16.11</a> Stateless Proxy</span>

   When acting statelessly, a proxy is a simple message forwarder.  Much
   of the processing performed when acting statelessly is the same as
   when behaving statefully.  The differences are detailed here.

   A stateless proxy does not have any notion of a transaction, or of
   the response context used to describe stateful proxy behavior.
   Instead, the stateless proxy takes messages, both requests and
   responses, directly from the transport layer (See <a href="#section-18">section 18</a>).  As a
   result, stateless proxies do not retransmit messages on their own.
   They do, however, forward all retransmissions they receive (they do
   not have the ability to distinguish a retransmission from the
   original message).  Furthermore, when handling a request statelessly,
   an element MUST NOT generate its own 100 (Trying) or any other
   provisional response.

   A stateless proxy MUST validate a request as described in <a href="#section-16.3">Section</a>
   <a href="#section-16.3">16.3</a>

   A stateless proxy MUST follow the request processing steps described
   in Sections <a href="#section-16.4">16.4</a> through <a href="#section-16.5">16.5</a> with the following exception:

      o  A stateless proxy MUST choose one and only one target from the
         target set.  This choice MUST only rely on fields in the
         message and time-invariant properties of the server.  In
         particular, a retransmitted request MUST be forwarded to the
         same destination each time it is processed.  Furthermore,
         CANCEL and non-Routed ACK requests MUST generate the same
         choice as their associated INVITE.

   A stateless proxy MUST follow the request processing steps described
   in <a href="#section-16.6">Section 16.6</a> with the following exceptions:

      o  The requirement for unique branch IDs across space and time
         applies to stateless proxies as well.  However, a stateless
         proxy cannot simply use a random number generator to compute
         the first component of the branch ID, as described in <a href="#section-16.6">Section</a>
         <a href="#section-16.6">16.6</a> bullet 8.  This is because retransmissions of a request
         need to have the same value, and a stateless proxy cannot tell
         a retransmission from the original request.  Therefore, the
         component of the branch parameter that makes it unique MUST be
         the same each time a retransmitted request is forwarded.  Thus
         for a stateless proxy, the branch parameter MUST be computed as
         a combinatoric function of message parameters which are
         invariant on retransmission.





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


         The stateless proxy MAY use any technique it likes to guarantee
         uniqueness of its branch IDs across transactions.  However, the
         following procedure is RECOMMENDED.  The proxy examines the
         branch ID in the topmost Via header field of the received
         request.  If it begins with the magic cookie, the first
         component of the branch ID of the outgoing request is computed
         as a hash of the received branch ID.  Otherwise, the first
         component of the branch ID is computed as a hash of the topmost
         Via, the tag in the To header field, the tag in the From header
         field, the Call-ID header field, the CSeq number (but not
         method), and the Request-URI from the received request.  One of
         these fields will always vary across two different
         transactions.

      o  All other message transformations specified in <a href="#section-16.6">Section 16.6</a>
         MUST result in the same transformation of a retransmitted
         request.  In particular, if the proxy inserts a Record-Route
         value or pushes URIs into the Route header field, it MUST place
         the same values in retransmissions of the request.  As for the
         Via branch parameter, this implies that the transformations
         MUST be based on time-invariant configuration or
         retransmission-invariant properties of the request.

      o  A stateless proxy determines where to forward the request as
         described for stateful proxies in <a href="#section-16.6">Section 16.6</a> Item 10.  The
         request is sent directly to the transport layer instead of
         through a client transaction.

         Since a stateless proxy must forward retransmitted requests to
         the same destination and add identical branch parameters to
         each of them, it can only use information from the message
         itself and time-invariant configuration data for those
         calculations.  If the configuration state is not time-invariant
         (for example, if a routing table is updated) any requests that
         could be affected by the change may not be forwarded
         statelessly during an interval equal to the transaction timeout
         window before or after the change.  The method of processing
         the affected requests in that interval is an implementation
         decision.  A common solution is to forward them transaction
         statefully.

   Stateless proxies MUST NOT perform special processing for CANCEL
   requests.  They are processed by the above rules as any other
   requests.  In particular, a stateless proxy applies the same Route
   header field processing to CANCEL requests that it applies to any
   other request.





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Response processing as described in <a href="#section-16.7">Section 16.7</a> does not apply to a
   proxy behaving statelessly.  When a response arrives at a stateless
   proxy, the proxy MUST inspect the sent-by value in the first
   (topmost) Via header field value.  If that address matches the proxy,
   (it equals a value this proxy has inserted into previous requests)
   the proxy MUST remove that header field value from the response and
   forward the result to the location indicated in the next Via header
   field value.  The proxy MUST NOT add to, modify, or remove the
   message body.  Unless specified otherwise, the proxy MUST NOT remove
   any other header field values.  If the address does not match the
   proxy, the message MUST be silently discarded.

<span class="h3"><a class="selflink" id="section-16.12" href="#section-16.12">16.12</a> Summary of Proxy Route Processing</span>

   In the absence of local policy to the contrary, the processing a
   proxy performs on a request containing a Route header field can be
   summarized in the following steps.

      1.  The proxy will inspect the Request-URI.  If it indicates a
          resource owned by this proxy, the proxy will replace it with
          the results of running a location service.  Otherwise, the
          proxy will not change the Request-URI.

      2.  The proxy will inspect the URI in the topmost Route header
          field value.  If it indicates this proxy, the proxy removes it
          from the Route header field (this route node has been
          reached).

      3.  The proxy will forward the request to the resource indicated
          by the URI in the topmost Route header field value or in the
          Request-URI if no Route header field is present.  The proxy
          determines the address, port and transport to use when
          forwarding the request by applying the procedures in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] to
          that URI.

   If no strict-routing elements are encountered on the path of the
   request, the Request-URI will always indicate the target of the
   request.

<span class="h4"><a class="selflink" id="section-16.12.1" href="#section-16.12.1">16.12.1</a> Examples</span>

<span class="h5"><a class="selflink" id="section-16.12.1.1" href="#section-16.12.1.1">16.12.1.1</a> Basic SIP Trapezoid</span>

   This scenario is the basic SIP trapezoid, U1 -&gt; P1 -&gt; P2 -&gt; U2, with
   both proxies record-routing.  Here is the flow.






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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   U1 sends:

      INVITE sip:callee@domain.com SIP/2.0
      Contact: sip:caller@u1.example.com

   to P1.  P1 is an outbound proxy.  P1 is not responsible for
   domain.com, so it looks it up in DNS and sends it there.  It also
   adds a Record-Route header field value:

      INVITE sip:callee@domain.com SIP/2.0
      Contact: sip:caller@u1.example.com
      Record-Route: &lt;sip:p1.example.com;lr&gt;

   P2 gets this.  It is responsible for domain.com so it runs a location
   service and rewrites the Request-URI.  It also adds a Record-Route
   header field value.  There is no Route header field, so it resolves
   the new Request-URI to determine where to send the request:

      INVITE sip:callee@u2.domain.com SIP/2.0
      Contact: sip:caller@u1.example.com
      Record-Route: &lt;sip:p2.domain.com;lr&gt;
      Record-Route: &lt;sip:p1.example.com;lr&gt;

   The callee at u2.domain.com gets this and responds with a 200 OK:

      SIP/2.0 200 OK
      Contact: sip:callee@u2.domain.com
      Record-Route: &lt;sip:p2.domain.com;lr&gt;
      Record-Route: &lt;sip:p1.example.com;lr&gt;

   The callee at u2 also sets its dialog state's remote target URI to
   sip:caller@u1.example.com and its route set to:

      (&lt;sip:p2.domain.com;lr&gt;,&lt;sip:p1.example.com;lr&gt;)

   This is forwarded by P2 to P1 to U1 as normal.  Now, U1 sets its
   dialog state's remote target URI to sip:callee@u2.domain.com and its
   route set to:

      (&lt;sip:p1.example.com;lr&gt;,&lt;sip:p2.domain.com;lr&gt;)

   Since all the route set elements contain the lr parameter, U1
   constructs the following BYE request:

      BYE sip:callee@u2.domain.com SIP/2.0
      Route: &lt;sip:p1.example.com;lr&gt;,&lt;sip:p2.domain.com;lr&gt;





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   As any other element (including proxies) would do, it resolves the
   URI in the topmost Route header field value using DNS to determine
   where to send the request.  This goes to P1.  P1 notices that it is
   not responsible for the resource indicated in the Request-URI so it
   doesn't change it.  It does see that it is the first value in the
   Route header field, so it removes that value, and forwards the
   request to P2:

      BYE sip:callee@u2.domain.com SIP/2.0
      Route: &lt;sip:p2.domain.com;lr&gt;

   P2 also notices it is not responsible for the resource indicated by
   the Request-URI (it is responsible for domain.com, not
   u2.domain.com), so it doesn't change it.  It does see itself in the
   first Route header field value, so it removes it and forwards the
   following to u2.domain.com based on a DNS lookup against the
   Request-URI:

      BYE sip:callee@u2.domain.com SIP/2.0

<span class="h5"><a class="selflink" id="section-16.12.1.2" href="#section-16.12.1.2">16.12.1.2</a> Traversing a Strict-Routing Proxy</span>

   In this scenario, a dialog is established across four proxies, each
   of which adds Record-Route header field values.  The third proxy
   implements the strict-routing procedures specified in <a href="./rfc2543">RFC 2543</a> and
   many works in progress.

      U1-&gt;P1-&gt;P2-&gt;P3-&gt;P4-&gt;U2

   The INVITE arriving at U2 contains:

      INVITE sip:callee@u2.domain.com SIP/2.0
      Contact: sip:caller@u1.example.com
      Record-Route: &lt;sip:p4.domain.com;lr&gt;
      Record-Route: &lt;sip:p3.middle.com&gt;
      Record-Route: &lt;sip:p2.example.com;lr&gt;
      Record-Route: &lt;sip:p1.example.com;lr&gt;

   Which U2 responds to with a 200 OK.  Later, U2 sends the following
   BYE request to P4 based on the first Route header field value.

      BYE sip:caller@u1.example.com SIP/2.0
      Route: &lt;sip:p4.domain.com;lr&gt;
      Route: &lt;sip:p3.middle.com&gt;
      Route: &lt;sip:p2.example.com;lr&gt;
      Route: &lt;sip:p1.example.com;lr&gt;





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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   P4 is not responsible for the resource indicated in the Request-URI
   so it will leave it alone.  It notices that it is the element in the
   first Route header field value so it removes it.  It then prepares to
   send the request based on the now first Route header field value of
   sip:p3.middle.com, but it notices that this URI does not contain the
   lr parameter, so before sending, it reformats the request to be:

      BYE sip:p3.middle.com SIP/2.0
      Route: &lt;sip:p2.example.com;lr&gt;
      Route: &lt;sip:p1.example.com;lr&gt;
      Route: &lt;sip:caller@u1.example.com&gt;

   P3 is a strict router, so it forwards the following to P2:

      BYE sip:p2.example.com;lr SIP/2.0
      Route: &lt;sip:p1.example.com;lr&gt;
      Route: &lt;sip:caller@u1.example.com&gt;

   P2 sees the request-URI is a value it placed into a Record-Route
   header field, so before further processing, it rewrites the request
   to be:

      BYE sip:caller@u1.example.com SIP/2.0
      Route: &lt;sip:p1.example.com;lr&gt;

   P2 is not responsible for u1.example.com, so it sends the request to
   P1 based on the resolution of the Route header field value.

   P1 notices itself in the topmost Route header field value, so it
   removes it, resulting in:

      BYE sip:caller@u1.example.com SIP/2.0

   Since P1 is not responsible for u1.example.com and there is no Route
   header field, P1 will forward the request to u1.example.com based on
   the Request-URI.

<span class="h5"><a class="selflink" id="section-16.12.1.3" href="#section-16.12.1.3">16.12.1.3</a> Rewriting Record-Route Header Field Values</span>

   In this scenario, U1 and U2 are in different private namespaces and
   they enter a dialog through a proxy P1, which acts as a gateway
   between the namespaces.

      U1-&gt;P1-&gt;U2







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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   U1 sends:

      INVITE sip:callee@gateway.leftprivatespace.com SIP/2.0
      Contact: &lt;sip:caller@u1.leftprivatespace.com&gt;

   P1 uses its location service and sends the following to U2:

      INVITE sip:callee@rightprivatespace.com SIP/2.0
      Contact: &lt;sip:caller@u1.leftprivatespace.com&gt;
      Record-Route: &lt;sip:gateway.rightprivatespace.com;lr&gt;

   U2 sends this 200 (OK) back to P1:

      SIP/2.0 200 OK
      Contact: &lt;sip:callee@u2.rightprivatespace.com&gt;
      Record-Route: &lt;sip:gateway.rightprivatespace.com;lr&gt;

   P1 rewrites its Record-Route header parameter to provide a value that
   U1 will find useful, and sends the following to U1:

      SIP/2.0 200 OK
      Contact: &lt;sip:callee@u2.rightprivatespace.com&gt;
      Record-Route: &lt;sip:gateway.leftprivatespace.com;lr&gt;

   Later, U1 sends the following BYE request to P1:

      BYE sip:callee@u2.rightprivatespace.com SIP/2.0
      Route: &lt;sip:gateway.leftprivatespace.com;lr&gt;

   which P1 forwards to U2 as:

      BYE sip:callee@u2.rightprivatespace.com SIP/2.0

<span class="h2"><a class="selflink" id="section-17" href="#section-17">17</a> Transactions</span>

   SIP is a transactional protocol: interactions between components take
   place in a series of independent message exchanges.  Specifically, a
   SIP transaction consists of a single request and any responses to
   that request, which include zero or more provisional responses and
   one or more final responses.  In the case of a transaction where the
   request was an INVITE (known as an INVITE transaction), the
   transaction also includes the ACK only if the final response was not
   a 2xx response.  If the response was a 2xx, the ACK is not considered
   part of the transaction.

      The reason for this separation is rooted in the importance of
      delivering all 200 (OK) responses to an INVITE to the UAC.  To
      deliver them all to the UAC, the UAS alone takes responsibility



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      for retransmitting them (see <a href="#section-13.3.1.4">Section 13.3.1.4</a>), and the UAC alone
      takes responsibility for acknowledging them with ACK (see <a href="#section-13.2.2.4">Section</a>
      <a href="#section-13.2.2.4">13.2.2.4</a>).  Since this ACK is retransmitted only by the UAC, it is
      effectively considered its own transaction.

   Transactions have a client side and a server side.  The client side
   is known as a client transaction and the server side as a server
   transaction.  The client transaction sends the request, and the
   server transaction sends the response.  The client and server
   transactions are logical functions that are embedded in any number of
   elements.  Specifically, they exist within user agents and stateful
   proxy servers.  Consider the example in <a href="#section-4">Section 4</a>.  In this example,
   the UAC executes the client transaction, and its outbound proxy
   executes the server transaction.  The outbound proxy also executes a
   client transaction, which sends the request to a server transaction
   in the inbound proxy.  That proxy also executes a client transaction,
   which in turn sends the request to a server transaction in the UAS.
   This is shown in Figure 4.

   +---------+        +---------+        +---------+        +---------+
   |      +-+|Request |+-+   +-+|Request |+-+   +-+|Request |+-+      |
   |      |C||-------&gt;||S|   |C||-------&gt;||S|   |C||-------&gt;||S|      |
   |      |l||        ||e|   |l||        ||e|   |l||        ||e|      |
   |      |i||        ||r|   |i||        ||r|   |i||        ||r|      |
   |      |e||        ||v|   |e||        ||v|   |e||        ||v|      |
   |      |n||        ||e|   |n||        ||e|   |n||        ||e|      |
   |      |t||        ||r|   |t||        ||r|   |t||        ||r|      |
   |      | ||        || |   | ||        || |   | ||        || |      |
   |      |T||        ||T|   |T||        ||T|   |T||        ||T|      |
   |      |r||        ||r|   |r||        ||r|   |r||        ||r|      |
   |      |a||        ||a|   |a||        ||a|   |a||        ||a|      |
   |      |n||        ||n|   |n||        ||n|   |n||        ||n|      |
   |      |s||Response||s|   |s||Response||s|   |s||Response||s|      |
   |      +-+|&lt;-------|+-+   +-+|&lt;-------|+-+   +-+|&lt;-------|+-+      |
   +---------+        +---------+        +---------+        +---------+
      UAC               Outbound           Inbound              UAS
                        Proxy               Proxy

                  Figure 4: Transaction relationships

   A stateless proxy does not contain a client or server transaction.
   The transaction exists between the UA or stateful proxy on one side,
   and the UA or stateful proxy on the other side.  As far as SIP
   transactions are concerned, stateless proxies are effectively
   transparent.  The purpose of the client transaction is to receive a
   request from the element in which the client is embedded (call this
   element the "Transaction User" or TU; it can be a UA or a stateful
   proxy), and reliably deliver the request to a server transaction.



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   The client transaction is also responsible for receiving responses
   and delivering them to the TU, filtering out any response
   retransmissions or disallowed responses (such as a response to ACK).
   Additionally, in the case of an INVITE request, the client
   transaction is responsible for generating the ACK request for any
   final response accepting a 2xx response.

   Similarly, the purpose of the server transaction is to receive
   requests from the transport layer and deliver them to the TU.  The
   server transaction filters any request retransmissions from the
   network.  The server transaction accepts responses from the TU and
   delivers them to the transport layer for transmission over the
   network.  In the case of an INVITE transaction, it absorbs the ACK
   request for any final response excepting a 2xx response.

   The 2xx response and its ACK receive special treatment.  This
   response is retransmitted only by a UAS, and its ACK generated only
   by the UAC.  This end-to-end treatment is needed so that a caller
   knows the entire set of users that have accepted the call.  Because
   of this special handling, retransmissions of the 2xx response are
   handled by the UA core, not the transaction layer.  Similarly,
   generation of the ACK for the 2xx is handled by the UA core.  Each
   proxy along the path merely forwards each 2xx response to INVITE and
   its corresponding ACK.

<span class="h3"><a class="selflink" id="section-17.1" href="#section-17.1">17.1</a> Client Transaction</span>

   The client transaction provides its functionality through the
   maintenance of a state machine.

   The TU communicates with the client transaction through a simple
   interface.  When the TU wishes to initiate a new transaction, it
   creates a client transaction and passes it the SIP request to send
   and an IP address, port, and transport to which to send it.  The
   client transaction begins execution of its state machine.  Valid
   responses are passed up to the TU from the client transaction.

   There are two types of client transaction state machines, depending
   on the method of the request passed by the TU.  One handles client
   transactions for INVITE requests.  This type of machine is referred
   to as an INVITE client transaction.  Another type handles client
   transactions for all requests except INVITE and ACK.  This is
   referred to as a non-INVITE client transaction.  There is no client
   transaction for ACK.  If the TU wishes to send an ACK, it passes one
   directly to the transport layer for transmission.






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   The INVITE transaction is different from those of other methods
   because of its extended duration.  Normally, human input is required
   in order to respond to an INVITE.  The long delays expected for
   sending a response argue for a three-way handshake.  On the other
   hand, requests of other methods are expected to complete rapidly.
   Because of the non-INVITE transaction's reliance on a two-way
   handshake, TUs SHOULD respond immediately to non-INVITE requests.

<span class="h4"><a class="selflink" id="section-17.1.1" href="#section-17.1.1">17.1.1</a> INVITE Client Transaction</span>

<span class="h5"><a class="selflink" id="section-17.1.1.1" href="#section-17.1.1.1">17.1.1.1</a> Overview of INVITE Transaction</span>

   The INVITE transaction consists of a three-way handshake.  The client
   transaction sends an INVITE, the server transaction sends responses,
   and the client transaction sends an ACK.  For unreliable transports
   (such as UDP), the client transaction retransmits requests at an
   interval that starts at T1 seconds and doubles after every
   retransmission.  T1 is an estimate of the round-trip time (RTT), and
   it defaults to 500 ms.  Nearly all of the transaction timers
   described here scale with T1, and changing T1 adjusts their values.
   The request is not retransmitted over reliable transports.  After
   receiving a 1xx response, any retransmissions cease altogether, and
   the client waits for further responses.  The server transaction can
   send additional 1xx responses, which are not transmitted reliably by
   the server transaction.  Eventually, the server transaction decides
   to send a final response.  For unreliable transports, that response
   is retransmitted periodically, and for reliable transports, it is
   sent once.  For each final response that is received at the client
   transaction, the client transaction sends an ACK, the purpose of
   which is to quench retransmissions of the response.

<span class="h5"><a class="selflink" id="section-17.1.1.2" href="#section-17.1.1.2">17.1.1.2</a> Formal Description</span>

   The state machine for the INVITE client transaction is shown in
   Figure 5.  The initial state, "calling", MUST be entered when the TU
   initiates a new client transaction with an INVITE request.  The
   client transaction MUST pass the request to the transport layer for
   transmission (see <a href="#section-18">Section 18</a>).  If an unreliable transport is being
   used, the client transaction MUST start timer A with a value of T1.
   If a reliable transport is being used, the client transaction SHOULD
   NOT start timer A (Timer A controls request retransmissions).  For
   any transport, the client transaction MUST start timer B with a value
   of 64*T1 seconds (Timer B controls transaction timeouts).

   When timer A fires, the client transaction MUST retransmit the
   request by passing it to the transport layer, and MUST reset the
   timer with a value of 2*T1.  The formal definition of retransmit




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   within the context of the transaction layer is to take the message
   previously sent to the transport layer and pass it to the transport
   layer once more.

   When timer A fires 2*T1 seconds later, the request MUST be
   retransmitted again (assuming the client transaction is still in this
   state).  This process MUST continue so that the request is
   retransmitted with intervals that double after each transmission.
   These retransmissions SHOULD only be done while the client
   transaction is in the "calling" state.

   The default value for T1 is 500 ms.  T1 is an estimate of the RTT
   between the client and server transactions.  Elements MAY (though it
   is NOT RECOMMENDED) use smaller values of T1 within closed, private
   networks that do not permit general Internet connection.  T1 MAY be
   chosen larger, and this is RECOMMENDED if it is known in advance
   (such as on high latency access links) that the RTT is larger.
   Whatever the value of T1, the exponential backoffs on retransmissions
   described in this section MUST be used.

   If the client transaction is still in the "Calling" state when timer
   B fires, the client transaction SHOULD inform the TU that a timeout
   has occurred.  The client transaction MUST NOT generate an ACK.  The
   value of 64*T1 is equal to the amount of time required to send seven
   requests in the case of an unreliable transport.

   If the client transaction receives a provisional response while in
   the "Calling" state, it transitions to the "Proceeding" state. In the
   "Proceeding" state, the client transaction SHOULD NOT retransmit the
   request any longer. Furthermore, the provisional response MUST be
   passed to the TU.  Any further provisional responses MUST be passed
   up to the TU while in the "Proceeding" state.

   When in either the "Calling" or "Proceeding" states, reception of a
   response with status code from 300-699 MUST cause the client
   transaction to transition to "Completed".  The client transaction
   MUST pass the received response up to the TU, and the client
   transaction MUST generate an ACK request, even if the transport is
   reliable (guidelines for constructing the ACK from the response are
   given in <a href="#section-17.1.1.3">Section 17.1.1.3</a>) and then pass the ACK to the transport
   layer for transmission.  The ACK MUST be sent to the same address,
   port, and transport to which the original request was sent.  The
   client transaction SHOULD start timer D when it enters the
   "Completed" state, with a value of at least 32 seconds for unreliable
   transports, and a value of zero seconds for reliable transports.
   Timer D reflects the amount of time that the server transaction can
   remain in the "Completed" state when unreliable transports are used.
   This is equal to Timer H in the INVITE server transaction, whose



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   default is 64*T1.  However, the client transaction does not know the
   value of T1 in use by the server transaction, so an absolute minimum
   of 32s is used instead of basing Timer D on T1.

   Any retransmissions of the final response that are received while in
   the "Completed" state MUST cause the ACK to be re-passed to the
   transport layer for retransmission, but the newly received response
   MUST NOT be passed up to the TU.  A retransmission of the response is
   defined as any response which would match the same client transaction
   based on the rules of <a href="#section-17.1.3">Section 17.1.3</a>.









































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                               |INVITE from TU
             Timer A fires     |INVITE sent
             Reset A,          V                      Timer B fires
             INVITE sent +-----------+                or Transport Err.
               +---------|           |---------------+inform TU
               |         |  Calling  |               |
               +--------&gt;|           |--------------&gt;|
                         +-----------+ 2xx           |
                            |  |       2xx to TU     |
                            |  |1xx                  |
    300-699 +---------------+  |1xx to TU            |
   ACK sent |                  |                     |
resp. to TU |  1xx             V                     |
            |  1xx to TU  -----------+               |
            |  +---------|           |               |
            |  |         |Proceeding |--------------&gt;|
            |  +--------&gt;|           | 2xx           |
            |            +-----------+ 2xx to TU     |
            |       300-699    |                     |
            |       ACK sent,  |                     |
            |       resp. to TU|                     |
            |                  |                     |      NOTE:
            |  300-699         V                     |
            |  ACK sent  +-----------+Transport Err. |  transitions
            |  +---------|           |Inform TU      |  labeled with
            |  |         | Completed |--------------&gt;|  the event
            |  +--------&gt;|           |               |  over the action
            |            +-----------+               |  to take
            |              ^   |                     |
            |              |   | Timer D fires       |
            +--------------+   | -                   |
                               |                     |
                               V                     |
                         +-----------+               |
                         |           |               |
                         | Terminated|&lt;--------------+
                         |           |
                         +-----------+

                 Figure 5: INVITE client transaction

   If timer D fires while the client transaction is in the "Completed"
   state, the client transaction MUST move to the terminated state.

   When in either the "Calling" or "Proceeding" states, reception of a
   2xx response MUST cause the client transaction to enter the
   "Terminated" state, and the response MUST be passed up to the TU.
   The handling of this response depends on whether the TU is a proxy



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   core or a UAC core.  A UAC core will handle generation of the ACK for
   this response, while a proxy core will always forward the 200 (OK)
   upstream.  The differing treatment of 200 (OK) between proxy and UAC
   is the reason that handling of it does not take place in the
   transaction layer.

   The client transaction MUST be destroyed the instant it enters the
   "Terminated" state.  This is actually necessary to guarantee correct
   operation.  The reason is that 2xx responses to an INVITE are treated
   differently; each one is forwarded by proxies, and the ACK handling
   in a UAC is different.  Thus, each 2xx needs to be passed to a proxy
   core (so that it can be forwarded) and to a UAC core (so it can be
   acknowledged).  No transaction layer processing takes place.
   Whenever a response is received by the transport, if the transport
   layer finds no matching client transaction (using the rules of
   <a href="#section-17.1.3">Section 17.1.3</a>), the response is passed directly to the core.  Since
   the matching client transaction is destroyed by the first 2xx,
   subsequent 2xx will find no match and therefore be passed to the
   core.

<span class="h5"><a class="selflink" id="section-17.1.1.3" href="#section-17.1.1.3">17.1.1.3</a> Construction of the ACK Request</span>

   This section specifies the construction of ACK requests sent within
   the client transaction.  A UAC core that generates an ACK for 2xx
   MUST instead follow the rules described in <a href="#section-13">Section 13</a>.

   The ACK request constructed by the client transaction MUST contain
   values for the Call-ID, From, and Request-URI that are equal to the
   values of those header fields in the request passed to the transport
   by the client transaction (call this the "original request").  The To
   header field in the ACK MUST equal the To header field in the
   response being acknowledged, and therefore will usually differ from
   the To header field in the original request by the addition of the
   tag parameter.  The ACK MUST contain a single Via header field, and
   this MUST be equal to the top Via header field of the original
   request.  The CSeq header field in the ACK MUST contain the same
   value for the sequence number as was present in the original request,
   but the method parameter MUST be equal to "ACK".













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   If the INVITE request whose response is being acknowledged had Route
   header fields, those header fields MUST appear in the ACK.  This is
   to ensure that the ACK can be routed properly through any downstream
   stateless proxies.

   Although any request MAY contain a body, a body in an ACK is special
   since the request cannot be rejected if the body is not understood.
   Therefore, placement of bodies in ACK for non-2xx is NOT RECOMMENDED,
   but if done, the body types are restricted to any that appeared in
   the INVITE, assuming that the response to the INVITE was not 415.  If
   it was, the body in the ACK MAY be any type listed in the Accept
   header field in the 415.

   For example, consider the following request:

   INVITE sip:bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff
   To: Bob &lt;sip:bob@biloxi.com&gt;
   From: Alice &lt;sip:alice@atlanta.com&gt;;tag=88sja8x
   Max-Forwards: 70
   Call-ID: 987asjd97y7atg
   CSeq: 986759 INVITE

   The ACK request for a non-2xx final response to this request would
   look like this:

   ACK sip:bob@biloxi.com SIP/2.0
   Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKkjshdyff
   To: Bob &lt;sip:bob@biloxi.com&gt;;tag=99sa0xk
   From: Alice &lt;sip:alice@atlanta.com&gt;;tag=88sja8x
   Max-Forwards: 70
   Call-ID: 987asjd97y7atg
   CSeq: 986759 ACK

<span class="h4"><a class="selflink" id="section-17.1.2" href="#section-17.1.2">17.1.2</a> Non-INVITE Client Transaction</span>

<span class="h5"><a class="selflink" id="section-17.1.2.1" href="#section-17.1.2.1">17.1.2.1</a> Overview of the non-INVITE Transaction</span>

   Non-INVITE transactions do not make use of ACK.  They are simple
   request-response interactions.  For unreliable transports, requests
   are retransmitted at an interval which starts at T1 and doubles until
   it hits T2.  If a provisional response is received, retransmissions
   continue for unreliable transports, but at an interval of T2.  The
   server transaction retransmits the last response it sent, which can
   be a provisional or final response, only when a retransmission of the
   request is received.  This is why request retransmissions need to
   continue even after a provisional response; they are to ensure
   reliable delivery of the final response.



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   Unlike an INVITE transaction, a non-INVITE transaction has no special
   handling for the 2xx response.  The result is that only a single 2xx
   response to a non-INVITE is ever delivered to a UAC.

<span class="h5"><a class="selflink" id="section-17.1.2.2" href="#section-17.1.2.2">17.1.2.2</a> Formal Description</span>

   The state machine for the non-INVITE client transaction is shown in
   Figure 6.  It is very similar to the state machine for INVITE.

   The "Trying" state is entered when the TU initiates a new client
   transaction with a request.  When entering this state, the client
   transaction SHOULD set timer F to fire in 64*T1 seconds.  The request
   MUST be passed to the transport layer for transmission.  If an
   unreliable transport is in use, the client transaction MUST set timer
   E to fire in T1 seconds.  If timer E fires while still in this state,
   the timer is reset, but this time with a value of MIN(2*T1, T2).
   When the timer fires again, it is reset to a MIN(4*T1, T2).  This
   process continues so that retransmissions occur with an exponentially
   increasing interval that caps at T2.  The default value of T2 is 4s,
   and it represents the amount of time a non-INVITE server transaction
   will take to respond to a request, if it does not respond
   immediately.  For the default values of T1 and T2, this results in
   intervals of 500 ms, 1 s, 2 s, 4 s, 4 s, 4 s, etc.

   If Timer F fires while the client transaction is still in the
   "Trying" state, the client transaction SHOULD inform the TU about the
   timeout, and then it SHOULD enter the "Terminated" state.  If a
   provisional response is received while in the "Trying" state, the
   response MUST be passed to the TU, and then the client transaction
   SHOULD move to the "Proceeding" state.  If a final response (status
   codes 200-699) is received while in the "Trying" state, the response
   MUST be passed to the TU, and the client transaction MUST transition
   to the "Completed" state.

   If Timer E fires while in the "Proceeding" state, the request MUST be
   passed to the transport layer for retransmission, and Timer E MUST be
   reset with a value of T2 seconds.  If timer F fires while in the
   "Proceeding" state, the TU MUST be informed of a timeout, and the
   client transaction MUST transition to the terminated state.  If a
   final response (status codes 200-699) is received while in the
   "Proceeding" state, the response MUST be passed to the TU, and the
   client transaction MUST transition to the "Completed" state.

   Once the client transaction enters the "Completed" state, it MUST set
   Timer K to fire in T4 seconds for unreliable transports, and zero
   seconds for reliable transports.  The "Completed" state exists to
   buffer any additional response retransmissions that may be received
   (which is why the client transaction remains there only for



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   unreliable transports).  T4 represents the amount of time the network
   will take to clear messages between client and server transactions.
   The default value of T4 is 5s.  A response is a retransmission when
   it matches the same transaction, using the rules specified in <a href="#section-17.1.3">Section</a>
   <a href="#section-17.1.3">17.1.3</a>.  If Timer K fires while in this state, the client transaction
   MUST transition to the "Terminated" state.

   Once the transaction is in the terminated state, it MUST be destroyed
   immediately.

<span class="h4"><a class="selflink" id="section-17.1.3" href="#section-17.1.3">17.1.3</a> Matching Responses to Client Transactions</span>

   When the transport layer in the client receives a response, it has to
   determine which client transaction will handle the response, so that
   the processing of Sections <a href="#section-17.1.1">17.1.1</a> and <a href="#section-17.1.2">17.1.2</a> can take place.  The
   branch parameter in the top Via header field is used for this
   purpose.  A response matches a client transaction under two
   conditions:

      1.  If the response has the same value of the branch parameter in
          the top Via header field as the branch parameter in the top
          Via header field of the request that created the transaction.

      2.  If the method parameter in the CSeq header field matches the
          method of the request that created the transaction.  The
          method is needed since a CANCEL request constitutes a
          different transaction, but shares the same value of the branch
          parameter.

   If a request is sent via multicast, it is possible that it will
   generate multiple responses from different servers.  These responses
   will all have the same branch parameter in the topmost Via, but vary
   in the To tag.  The first response received, based on the rules
   above, will be used, and others will be viewed as retransmissions.
   That is not an error; multicast SIP provides only a rudimentary
   "single-hop-discovery-like" service that is limited to processing a
   single response.  See <a href="#section-18.1.1">Section 18.1.1</a> for details.














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<span class="h4"><a class="selflink" id="section-17.1.4" href="#section-17.1.4">17.1.4</a> Handling Transport Errors</span>

                                   |Request from TU
                                   |send request
               Timer E             V
               send request  +-----------+
                   +---------|           |-------------------+
                   |         |  Trying   |  Timer F          |
                   +--------&gt;|           |  or Transport Err.|
                             +-----------+  inform TU        |
                200-699         |  |                         |
                resp. to TU     |  |1xx                      |
                +---------------+  |resp. to TU              |
                |                  |                         |
                |   Timer E        V       Timer F           |
                |   send req +-----------+ or Transport Err. |
                |  +---------|           | inform TU         |
                |  |         |Proceeding |------------------&gt;|
                |  +--------&gt;|           |-----+             |
                |            +-----------+     |1xx          |
                |              |      ^        |resp to TU   |
                | 200-699      |      +--------+             |
                | resp. to TU  |                             |
                |              |                             |
                |              V                             |
                |            +-----------+                   |
                |            |           |                   |
                |            | Completed |                   |
                |            |           |                   |
                |            +-----------+                   |
                |              ^   |                         |
                |              |   | Timer K                 |
                +--------------+   | -                       |
                                   |                         |
                                   V                         |
             NOTE:           +-----------+                   |
                             |           |                   |
         transitions         | Terminated|&lt;------------------+
         labeled with        |           |
         the event           +-----------+
         over the action
         to take

                 Figure 6: non-INVITE client transaction

   When the client transaction sends a request to the transport layer to
   be sent, the following procedures are followed if the transport layer
   indicates a failure.



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-134" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The client transaction SHOULD inform the TU that a transport failure
   has occurred, and the client transaction SHOULD transition directly
   to the "Terminated" state.  The TU will handle the failover
   mechanisms described in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

<span class="h3"><a class="selflink" id="section-17.2" href="#section-17.2">17.2</a> Server Transaction</span>

   The server transaction is responsible for the delivery of requests to
   the TU and the reliable transmission of responses.  It accomplishes
   this through a state machine.  Server transactions are created by the
   core when a request is received, and transaction handling is desired
   for that request (this is not always the case).

   As with the client transactions, the state machine depends on whether
   the received request is an INVITE request.

<span class="h4"><a class="selflink" id="section-17.2.1" href="#section-17.2.1">17.2.1</a> INVITE Server Transaction</span>

   The state diagram for the INVITE server transaction is shown in
   Figure 7.

   When a server transaction is constructed for a request, it enters the
   "Proceeding" state.  The server transaction MUST generate a 100
   (Trying) response unless it knows that the TU will generate a
   provisional or final response within 200 ms, in which case it MAY
   generate a 100 (Trying) response.  This provisional response is
   needed to quench request retransmissions rapidly in order to avoid
   network congestion.  The 100 (Trying) response is constructed
   according to the procedures in <a href="#section-8.2.6">Section 8.2.6</a>, except that the
   insertion of tags in the To header field of the response (when none
   was present in the request) is downgraded from MAY to SHOULD NOT.
   The request MUST be passed to the TU.

   The TU passes any number of provisional responses to the server
   transaction.  So long as the server transaction is in the
   "Proceeding" state, each of these MUST be passed to the transport
   layer for transmission.  They are not sent reliably by the
   transaction layer (they are not retransmitted by it) and do not cause
   a change in the state of the server transaction.  If a request
   retransmission is received while in the "Proceeding" state, the most
   recent provisional response that was received from the TU MUST be
   passed to the transport layer for retransmission.  A request is a
   retransmission if it matches the same server transaction based on the
   rules of <a href="#section-17.2.3">Section 17.2.3</a>.

   If, while in the "Proceeding" state, the TU passes a 2xx response to
   the server transaction, the server transaction MUST pass this
   response to the transport layer for transmission.  It is not



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-135" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   retransmitted by the server transaction; retransmissions of 2xx
   responses are handled by the TU.  The server transaction MUST then
   transition to the "Terminated" state.

   While in the "Proceeding" state, if the TU passes a response with
   status code from 300 to 699 to the server transaction, the response
   MUST be passed to the transport layer for transmission, and the state
   machine MUST enter the "Completed" state.  For unreliable transports,
   timer G is set to fire in T1 seconds, and is not set to fire for
   reliable transports.

      This is a change from <a href="./rfc2543">RFC 2543</a>, where responses were always
      retransmitted, even over reliable transports.

   When the "Completed" state is entered, timer H MUST be set to fire in
   64*T1 seconds for all transports.  Timer H determines when the server
   transaction abandons retransmitting the response.  Its value is
   chosen to equal Timer B, the amount of time a client transaction will
   continue to retry sending a request.  If timer G fires, the response
   is passed to the transport layer once more for retransmission, and
   timer G is set to fire in MIN(2*T1, T2) seconds.  From then on, when
   timer G fires, the response is passed to the transport again for
   transmission, and timer G is reset with a value that doubles, unless
   that value exceeds T2, in which case it is reset with the value of
   T2.  This is identical to the retransmit behavior for requests in the
   "Trying" state of the non-INVITE client transaction.  Furthermore,
   while in the "Completed" state, if a request retransmission is
   received, the server SHOULD pass the response to the transport for
   retransmission.

   If an ACK is received while the server transaction is in the
   "Completed" state, the server transaction MUST transition to the
   "Confirmed" state.  As Timer G is ignored in this state, any
   retransmissions of the response will cease.

   If timer H fires while in the "Completed" state, it implies that the
   ACK was never received.  In this case, the server transaction MUST
   transition to the "Terminated" state, and MUST indicate to the TU
   that a transaction failure has occurred.












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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-136" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


                               |INVITE
                               |pass INV to TU
            INVITE             V send 100 if TU won't in 200ms
            send response+-----------+
                +--------|           |--------+101-199 from TU
                |        | Proceeding|        |send response
                +-------&gt;|           |&lt;-------+
                         |           |          Transport Err.
                         |           |          Inform TU
                         |           |---------------&gt;+
                         +-----------+                |
            300-699 from TU |     |2xx from TU        |
            send response   |     |send response      |
                            |     +------------------&gt;+
                            |                         |
            INVITE          V          Timer G fires  |
            send response+-----------+ send response  |
                +--------|           |--------+       |
                |        | Completed |        |       |
                +-------&gt;|           |&lt;-------+       |
                         +-----------+                |
                            |     |                   |
                        ACK |     |                   |
                        -   |     +------------------&gt;+
                            |        Timer H fires    |
                            V        or Transport Err.|
                         +-----------+  Inform TU     |
                         |           |                |
                         | Confirmed |                |
                         |           |                |
                         +-----------+                |
                               |                      |
                               |Timer I fires         |
                               |-                     |
                               |                      |
                               V                      |
                         +-----------+                |
                         |           |                |
                         | Terminated|&lt;---------------+
                         |           |
                         +-----------+

              Figure 7: INVITE server transaction








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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The purpose of the "Confirmed" state is to absorb any additional ACK
   messages that arrive, triggered from retransmissions of the final
   response.  When this state is entered, timer I is set to fire in T4
   seconds for unreliable transports, and zero seconds for reliable
   transports.  Once timer I fires, the server MUST transition to the
   "Terminated" state.

   Once the transaction is in the "Terminated" state, it MUST be
   destroyed immediately.  As with client transactions, this is needed
   to ensure reliability of the 2xx responses to INVITE.

<span class="h4"><a class="selflink" id="section-17.2.2" href="#section-17.2.2">17.2.2</a> Non-INVITE Server Transaction</span>

   The state machine for the non-INVITE server transaction is shown in
   Figure 8.

   The state machine is initialized in the "Trying" state and is passed
   a request other than INVITE or ACK when initialized.  This request is
   passed up to the TU.  Once in the "Trying" state, any further request
   retransmissions are discarded.  A request is a retransmission if it
   matches the same server transaction, using the rules specified in
   <a href="#section-17.2.3">Section 17.2.3</a>.

   While in the "Trying" state, if the TU passes a provisional response
   to the server transaction, the server transaction MUST enter the
   "Proceeding" state.  The response MUST be passed to the transport
   layer for transmission.  Any further provisional responses that are
   received from the TU while in the "Proceeding" state MUST be passed
   to the transport layer for transmission.  If a retransmission of the
   request is received while in the "Proceeding" state, the most
   recently sent provisional response MUST be passed to the transport
   layer for retransmission.  If the TU passes a final response (status
   codes 200-699) to the server while in the "Proceeding" state, the
   transaction MUST enter the "Completed" state, and the response MUST
   be passed to the transport layer for transmission.

   When the server transaction enters the "Completed" state, it MUST set
   Timer J to fire in 64*T1 seconds for unreliable transports, and zero
   seconds for reliable transports.  While in the "Completed" state, the
   server transaction MUST pass the final response to the transport
   layer for retransmission whenever a retransmission of the request is
   received.  Any other final responses passed by the TU to the server
   transaction MUST be discarded while in the "Completed" state.  The
   server transaction remains in this state until Timer J fires, at
   which point it MUST transition to the "Terminated" state.

   The server transaction MUST be destroyed the instant it enters the
   "Terminated" state.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 137]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-138" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h4"><a class="selflink" id="section-17.2.3" href="#section-17.2.3">17.2.3</a> Matching Requests to Server Transactions</span>

   When a request is received from the network by the server, it has to
   be matched to an existing transaction.  This is accomplished in the
   following manner.

   The branch parameter in the topmost Via header field of the request
   is examined.  If it is present and begins with the magic cookie
   "z9hG4bK", the request was generated by a client transaction
   compliant to this specification.  Therefore, the branch parameter
   will be unique across all transactions sent by that client.  The
   request matches a transaction if:

      1. the branch parameter in the request is equal to the one in the
         top Via header field of the request that created the
         transaction, and

      2. the sent-by value in the top Via of the request is equal to the
         one in the request that created the transaction, and

      3. the method of the request matches the one that created the
         transaction, except for ACK, where the method of the request
         that created the transaction is INVITE.

   This matching rule applies to both INVITE and non-INVITE transactions
   alike.

      The sent-by value is used as part of the matching process because
      there could be accidental or malicious duplication of branch
      parameters from different clients.

   If the branch parameter in the top Via header field is not present,
   or does not contain the magic cookie, the following procedures are
   used.  These exist to handle backwards compatibility with <a href="./rfc2543">RFC 2543</a>
   compliant implementations.

   The INVITE request matches a transaction if the Request-URI, To tag,
   From tag, Call-ID, CSeq, and top Via header field match those of the
   INVITE request which created the transaction.  In this case, the
   INVITE is a retransmission of the original one that created the
   transaction.  The ACK request matches a transaction if the Request-
   URI, From tag, Call-ID, CSeq number (not the method), and top Via
   header field match those of the INVITE request which created the
   transaction, and the To tag of the ACK matches the To tag of the
   response sent by the server transaction.  Matching is done based on
   the matching rules defined for each of those header fields.
   Inclusion of the tag in the To header field in the ACK matching
   process helps disambiguate ACK for 2xx from ACK for other responses



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-139" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   at a proxy, which may have forwarded both responses (This can occur
   in unusual conditions.  Specifically, when a proxy forked a request,
   and then crashes, the responses may be delivered to another proxy,
   which might end up forwarding multiple responses upstream).  An ACK
   request that matches an INVITE transaction matched by a previous ACK
   is considered a retransmission of that previous ACK.













































<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 139]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-140" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


                                  |Request received
                                  |pass to TU
                                  V
                            +-----------+
                            |           |
                            | Trying    |-------------+
                            |           |             |
                            +-----------+             |200-699 from TU
                                  |                   |send response
                                  |1xx from TU        |
                                  |send response      |
                                  |                   |
               Request            V      1xx from TU  |
               send response+-----------+send response|
                   +--------|           |--------+    |
                   |        | Proceeding|        |    |
                   +-------&gt;|           |&lt;-------+    |
            +&lt;--------------|           |             |
            |Trnsprt Err    +-----------+             |
            |Inform TU            |                   |
            |                     |                   |
            |                     |200-699 from TU    |
            |                     |send response      |
            |  Request            V                   |
            |  send response+-----------+             |
            |      +--------|           |             |
            |      |        | Completed |&lt;------------+
            |      +-------&gt;|           |
            +&lt;--------------|           |
            |Trnsprt Err    +-----------+
            |Inform TU            |
            |                     |Timer J fires
            |                     |-
            |                     |
            |                     V
            |               +-----------+
            |               |           |
            +--------------&gt;| Terminated|
                            |           |
                            +-----------+

                Figure 8: non-INVITE server transaction

   For all other request methods, a request is matched to a transaction
   if the Request-URI, To tag, From tag, Call-ID, CSeq (including the
   method), and top Via header field match those of the request that
   created the transaction.  Matching is done based on the matching




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-141" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   rules defined for each of those header fields.  When a non-INVITE
   request matches an existing transaction, it is a retransmission of
   the request that created that transaction.

   Because the matching rules include the Request-URI, the server cannot
   match a response to a transaction.  When the TU passes a response to
   the server transaction, it must pass it to the specific server
   transaction for which the response is targeted.

<span class="h4"><a class="selflink" id="section-17.2.4" href="#section-17.2.4">17.2.4</a> Handling Transport Errors</span>

   When the server transaction sends a response to the transport layer
   to be sent, the following procedures are followed if the transport
   layer indicates a failure.

   First, the procedures in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] are followed, which attempt to deliver
   the response to a backup.  If those should all fail, based on the
   definition of failure in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>], the server transaction SHOULD inform
   the TU that a failure has occurred, and SHOULD transition to the
   terminated state.

<span class="h2"><a class="selflink" id="section-18" href="#section-18">18</a> Transport</span>

   The transport layer is responsible for the actual transmission of
   requests and responses over network transports.  This includes
   determination of the connection to use for a request or response in
   the case of connection-oriented transports.

   The transport layer is responsible for managing persistent
   connections for transport protocols like TCP and SCTP, or TLS over
   those, including ones opened to the transport layer.  This includes
   connections opened by the client or server transports, so that
   connections are shared between client and server transport functions.
   These connections are indexed by the tuple formed from the address,
   port, and transport protocol at the far end of the connection.  When
   a connection is opened by the transport layer, this index is set to
   the destination IP, port and transport.  When the connection is
   accepted by the transport layer, this index is set to the source IP
   address, port number, and transport.  Note that, because the source
   port is often ephemeral, but it cannot be known whether it is
   ephemeral or selected through procedures in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>], connections accepted
   by the transport layer will frequently not be reused.  The result is
   that two proxies in a "peering" relationship using a connection-
   oriented transport frequently will have two connections in use, one
   for transactions initiated in each direction.






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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   It is RECOMMENDED that connections be kept open for some
   implementation-defined duration after the last message was sent or
   received over that connection.  This duration SHOULD at least equal
   the longest amount of time the element would need in order to bring a
   transaction from instantiation to the terminated state.  This is to
   make it likely that transactions are completed over the same
   connection on which they are initiated (for example, request,
   response, and in the case of INVITE, ACK for non-2xx responses).
   This usually means at least 64*T1 (see <a href="#section-17.1.1.1">Section 17.1.1.1</a> for a
   definition of T1).  However, it could be larger in an element that
   has a TU using a large value for timer C (bullet 11 of <a href="#section-16.6">Section 16.6</a>),
   for example.

   All SIP elements MUST implement UDP and TCP.  SIP elements MAY
   implement other protocols.

      Making TCP mandatory for the UA is a substantial change from <a href="./rfc2543">RFC</a>
      <a href="./rfc2543">2543</a>.  It has arisen out of the need to handle larger messages,
      which MUST use TCP, as discussed below.  Thus, even if an element
      never sends large messages, it may receive one and needs to be
      able to handle them.

<span class="h3"><a class="selflink" id="section-18.1" href="#section-18.1">18.1</a> Clients</span>

<span class="h4"><a class="selflink" id="section-18.1.1" href="#section-18.1.1">18.1.1</a> Sending Requests</span>

   The client side of the transport layer is responsible for sending the
   request and receiving responses.  The user of the transport layer
   passes the client transport the request, an IP address, port,
   transport, and possibly TTL for multicast destinations.

   If a request is within 200 bytes of the path MTU, or if it is larger
   than 1300 bytes and the path MTU is unknown, the request MUST be sent
   using an <a href="./rfc2914">RFC 2914</a> [<a href="#ref-43" title="&quot;Congestion Control Principles&quot;">43</a>] congestion controlled transport protocol, such
   as TCP. If this causes a change in the transport protocol from the
   one indicated in the top Via, the value in the top Via MUST be
   changed.  This prevents fragmentation of messages over UDP and
   provides congestion control for larger messages.  However,
   implementations MUST be able to handle messages up to the maximum
   datagram packet size.  For UDP, this size is 65,535 bytes, including
   IP and UDP headers.

      The 200 byte "buffer" between the message size and the MTU
      accommodates the fact that the response in SIP can be larger than
      the request.  This happens due to the addition of Record-Route
      header field values to the responses to INVITE, for example.  With
      the extra buffer, the response can be about 170 bytes larger than
      the request, and still not be fragmented on IPv4 (about 30 bytes



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-143" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      is consumed by IP/UDP, assuming no IPSec).  1300 is chosen when
      path MTU is not known, based on the assumption of a 1500 byte
      Ethernet MTU.

   If an element sends a request over TCP because of these message size
   constraints, and that request would have otherwise been sent over
   UDP, if the attempt to establish the connection generates either an
   ICMP Protocol Not Supported, or results in a TCP reset, the element
   SHOULD retry the request, using UDP.  This is only to provide
   backwards compatibility with <a href="./rfc2543">RFC 2543</a> compliant implementations that
   do not support TCP.  It is anticipated that this behavior will be
   deprecated in a future revision of this specification.

   A client that sends a request to a multicast address MUST add the
   "maddr" parameter to its Via header field value containing the
   destination multicast address, and for IPv4, SHOULD add the "ttl"
   parameter with a value of 1.  Usage of IPv6 multicast is not defined
   in this specification, and will be a subject of future
   standardization when the need arises.

   These rules result in a purposeful limitation of multicast in SIP.
   Its primary function is to provide a "single-hop-discovery-like"
   service, delivering a request to a group of homogeneous servers,
   where it is only required to process the response from any one of
   them.  This functionality is most useful for registrations.  In fact,
   based on the transaction processing rules in <a href="#section-17.1.3">Section 17.1.3</a>, the
   client transaction will accept the first response, and view any
   others as retransmissions because they all contain the same Via
   branch identifier.

   Before a request is sent, the client transport MUST insert a value of
   the "sent-by" field into the Via header field.  This field contains
   an IP address or host name, and port.  The usage of an FQDN is
   RECOMMENDED.  This field is used for sending responses under certain
   conditions, described below.  If the port is absent, the default
   value depends on the transport.  It is 5060 for UDP, TCP and SCTP,
   5061 for TLS.

   For reliable transports, the response is normally sent on the
   connection on which the request was received.  Therefore, the client
   transport MUST be prepared to receive the response on the same
   connection used to send the request.  Under error conditions, the
   server may attempt to open a new connection to send the response.  To
   handle this case, the transport layer MUST also be prepared to
   receive an incoming connection on the source IP address from which
   the request was sent and port number in the "sent-by" field.  It also





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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-144" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   MUST be prepared to receive incoming connections on any address and
   port that would be selected by a server based on the procedures
   described in Section 5 of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

   For unreliable unicast transports, the client transport MUST be
   prepared to receive responses on the source IP address from which the
   request is sent (as responses are sent back to the source address)
   and the port number in the "sent-by" field.  Furthermore, as with
   reliable transports, in certain cases the response will be sent
   elsewhere.  The client MUST be prepared to receive responses on any
   address and port that would be selected by a server based on the
   procedures described in Section 5 of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

   For multicast, the client transport MUST be prepared to receive
   responses on the same multicast group and port to which the request
   is sent (that is, it needs to be a member of the multicast group it
   sent the request to.)

   If a request is destined to an IP address, port, and transport to
   which an existing connection is open, it is RECOMMENDED that this
   connection be used to send the request, but another connection MAY be
   opened and used.

   If a request is sent using multicast, it is sent to the group
   address, port, and TTL provided by the transport user.  If a request
   is sent using unicast unreliable transports, it is sent to the IP
   address and port provided by the transport user.

<span class="h4"><a class="selflink" id="section-18.1.2" href="#section-18.1.2">18.1.2</a> Receiving Responses</span>

   When a response is received, the client transport examines the top
   Via header field value.  If the value of the "sent-by" parameter in
   that header field value does not correspond to a value that the
   client transport is configured to insert into requests, the response
   MUST be silently discarded.

   If there are any client transactions in existence, the client
   transport uses the matching procedures of <a href="#section-17.1.3">Section 17.1.3</a> to attempt
   to match the response to an existing transaction.  If there is a
   match, the response MUST be passed to that transaction.  Otherwise,
   the response MUST be passed to the core (whether it be stateless
   proxy, stateful proxy, or UA) for further processing.  Handling of
   these "stray" responses is dependent on the core (a proxy will
   forward them, while a UA will discard, for example).







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<span class="h3"><a class="selflink" id="section-18.2" href="#section-18.2">18.2</a> Servers</span>

<span class="h4"><a class="selflink" id="section-18.2.1" href="#section-18.2.1">18.2.1</a> Receiving Requests</span>

   A server SHOULD be prepared to receive requests on any IP address,
   port and transport combination that can be the result of a DNS lookup
   on a SIP or SIPS URI [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] that is handed out for the purposes of
   communicating with that server.  In this context, "handing out"
   includes placing a URI in a Contact header field in a REGISTER
   request or a redirect response, or in a Record-Route header field in
   a request or response.  A URI can also be "handed out" by placing it
   on a web page or business card.  It is also RECOMMENDED that a server
   listen for requests on the default SIP ports (5060 for TCP and UDP,
   5061 for TLS over TCP) on all public interfaces.  The typical
   exception would be private networks, or when multiple server
   instances are running on the same host.  For any port and interface
   that a server listens on for UDP, it MUST listen on that same port
   and interface for TCP.  This is because a message may need to be sent
   using TCP, rather than UDP, if it is too large.  As a result, the
   converse is not true.  A server need not listen for UDP on a
   particular address and port just because it is listening on that same
   address and port for TCP.  There may, of course, be other reasons why
   a server needs to listen for UDP on a particular address and port.

   When the server transport receives a request over any transport, it
   MUST examine the value of the "sent-by" parameter in the top Via
   header field value.  If the host portion of the "sent-by" parameter
   contains a domain name, or if it contains an IP address that differs
   from the packet source address, the server MUST add a "received"
   parameter to that Via header field value.  This parameter MUST
   contain the source address from which the packet was received.  This
   is to assist the server transport layer in sending the response,
   since it must be sent to the source IP address from which the request
   came.

   Consider a request received by the server transport which looks like,
   in part:

      INVITE sip:bob@Biloxi.com SIP/2.0
      Via: SIP/2.0/UDP bobspc.biloxi.com:5060

   The request is received with a source IP address of 192.0.2.4.
   Before passing the request up, the transport adds a "received"
   parameter, so that the request would look like, in part:

      INVITE sip:bob@Biloxi.com SIP/2.0
      Via: SIP/2.0/UDP bobspc.biloxi.com:5060;received=192.0.2.4




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   Next, the server transport attempts to match the request to a server
   transaction.  It does so using the matching rules described in
   <a href="#section-17.2.3">Section 17.2.3</a>.  If a matching server transaction is found, the
   request is passed to that transaction for processing.  If no match is
   found, the request is passed to the core, which may decide to
   construct a new server transaction for that request.  Note that when
   a UAS core sends a 2xx response to INVITE, the server transaction is
   destroyed.  This means that when the ACK arrives, there will be no
   matching server transaction, and based on this rule, the ACK is
   passed to the UAS core, where it is processed.

<span class="h4"><a class="selflink" id="section-18.2.2" href="#section-18.2.2">18.2.2</a> Sending Responses</span>

   The server transport uses the value of the top Via header field in
   order to determine where to send a response.  It MUST follow the
   following process:

      o  If the "sent-protocol" is a reliable transport protocol such as
         TCP or SCTP, or TLS over those, the response MUST be sent using
         the existing connection to the source of the original request
         that created the transaction, if that connection is still open.
         This requires the server transport to maintain an association
         between server transactions and transport connections.  If that
         connection is no longer open, the server SHOULD open a
         connection to the IP address in the "received" parameter, if
         present, using the port in the "sent-by" value, or the default
         port for that transport, if no port is specified.  If that
         connection attempt fails, the server SHOULD use the procedures
         in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] for servers in order to determine the IP address and
         port to open the connection and send the response to.

      o  Otherwise, if the Via header field value contains a "maddr"
         parameter, the response MUST be forwarded to the address listed
         there, using the port indicated in "sent-by", or port 5060 if
         none is present.  If the address is a multicast address, the
         response SHOULD be sent using the TTL indicated in the "ttl"
         parameter, or with a TTL of 1 if that parameter is not present.

      o  Otherwise (for unreliable unicast transports), if the top Via
         has a "received" parameter, the response MUST be sent to the
         address in the "received" parameter, using the port indicated
         in the "sent-by" value, or using port 5060 if none is specified
         explicitly.  If this fails, for example, elicits an ICMP "port
         unreachable" response, the procedures of Section 5 of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>]
         SHOULD be used to determine where to send the response.






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      o  Otherwise, if it is not receiver-tagged, the response MUST be
         sent to the address indicated by the "sent-by" value, using the
         procedures in Section 5 of [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

<span class="h3"><a class="selflink" id="section-18.3" href="#section-18.3">18.3</a> Framing</span>

   In the case of message-oriented transports (such as UDP), if the
   message has a Content-Length header field, the message body is
   assumed to contain that many bytes.  If there are additional bytes in
   the transport packet beyond the end of the body, they MUST be
   discarded.  If the transport packet ends before the end of the
   message body, this is considered an error.  If the message is a
   response, it MUST be discarded.  If the message is a request, the
   element SHOULD generate a 400 (Bad Request) response.  If the message
   has no Content-Length header field, the message body is assumed to
   end at the end of the transport packet.

   In the case of stream-oriented transports such as TCP, the Content-
   Length header field indicates the size of the body.  The Content-
   Length header field MUST be used with stream oriented transports.

<span class="h3"><a class="selflink" id="section-18.4" href="#section-18.4">18.4</a> Error Handling</span>

   Error handling is independent of whether the message was a request or
   response.

   If the transport user asks for a message to be sent over an
   unreliable transport, and the result is an ICMP error, the behavior
   depends on the type of ICMP error.  Host, network, port or protocol
   unreachable errors, or parameter problem errors SHOULD cause the
   transport layer to inform the transport user of a failure in sending.
   Source quench and TTL exceeded ICMP errors SHOULD be ignored.

   If the transport user asks for a request to be sent over a reliable
   transport, and the result is a connection failure, the transport
   layer SHOULD inform the transport user of a failure in sending.

<span class="h2"><a class="selflink" id="section-19" href="#section-19">19</a> Common Message Components</span>

   There are certain components of SIP messages that appear in various
   places within SIP messages (and sometimes, outside of them) that
   merit separate discussion.









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<span class="h3"><a class="selflink" id="section-19.1" href="#section-19.1">19.1</a> SIP and SIPS Uniform Resource Indicators</span>

   A SIP or SIPS URI identifies a communications resource.  Like all
   URIs, SIP and SIPS URIs may be placed in web pages, email messages,
   or printed literature.  They contain sufficient information to
   initiate and maintain a communication session with the resource.

   Examples of communications resources include the following:

      o  a user of an online service

      o  an appearance on a multi-line phone

      o  a mailbox on a messaging system

      o  a PSTN number at a gateway service

      o  a group (such as "sales" or "helpdesk") in an organization

   A SIPS URI specifies that the resource be contacted securely.  This
   means, in particular, that TLS is to be used between the UAC and the
   domain that owns the URI.  From there, secure communications are used
   to reach the user, where the specific security mechanism depends on
   the policy of the domain.  Any resource described by a SIP URI can be
   "upgraded" to a SIPS URI by just changing the scheme, if it is
   desired to communicate with that resource securely.

<span class="h4"><a class="selflink" id="section-19.1.1" href="#section-19.1.1">19.1.1</a> SIP and SIPS URI Components</span>

   The "sip:" and "sips:" schemes follow the guidelines in <a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>].
   They use a form similar to the mailto URL, allowing the specification
   of SIP request-header fields and the SIP message-body.  This makes it
   possible to specify the subject, media type, or urgency of sessions
   initiated by using a URI on a web page or in an email message.  The
   formal syntax for a SIP or SIPS URI is presented in <a href="#section-25">Section 25</a>.  Its
   general form, in the case of a SIP URI, is:

      sip:user:password@host:port;uri-parameters?headers

   The format for a SIPS URI is the same, except that the scheme is
   "sips" instead of sip.  These tokens, and some of the tokens in their
   expansions, have the following meanings:

      user: The identifier of a particular resource at the host being
         addressed.  The term "host" in this context frequently refers
         to a domain.  The "userinfo" of a URI consists of this user
         field, the password field, and the @ sign following them.  The
         userinfo part of a URI is optional and MAY be absent when the



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         destination host does not have a notion of users or when the
         host itself is the resource being identified.  If the @ sign is
         present in a SIP or SIPS URI, the user field MUST NOT be empty.

         If the host being addressed can process telephone numbers, for
         instance, an Internet telephony gateway, a telephone-
         subscriber field defined in <a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>] MAY be used to
         populate the user field.  There are special escaping rules for
         encoding telephone-subscriber fields in SIP and SIPS URIs
         described in <a href="#section-19.1.2">Section 19.1.2</a>.

      password: A password associated with the user.  While the SIP and
         SIPS URI syntax allows this field to be present, its use is NOT
         RECOMMENDED, because the passing of authentication information
         in clear text (such as URIs) has proven to be a security risk
         in almost every case where it has been used.  For instance,
         transporting a PIN number in this field exposes the PIN.

         Note that the password field is just an extension of the user
         portion.  Implementations not wishing to give special
         significance to the password portion of the field MAY simply
         treat "user:password" as a single string.

      host: The host providing the SIP resource.  The host part contains
         either a fully-qualified domain name or numeric IPv4 or IPv6
         address.  Using the fully-qualified domain name form is
         RECOMMENDED whenever possible.

      port: The port number where the request is to be sent.

      URI parameters: Parameters affecting a request constructed from
         the URI.

         URI parameters are added after the hostport component and are
         separated by semi-colons.

         URI parameters take the form:

            parameter-name "=" parameter-value

         Even though an arbitrary number of URI parameters may be
         included in a URI, any given parameter-name MUST NOT appear
         more than once.

         This extensible mechanism includes the transport, maddr, ttl,
         user, method and lr parameters.





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         The transport parameter determines the transport mechanism to
         be used for sending SIP messages, as specified in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].  SIP can
         use any network transport protocol.  Parameter names are
         defined for UDP (<a href="./rfc768">RFC 768</a> [<a href="#ref-14" title="&quot;User Datagram Protocol&quot;">14</a>]), TCP (<a href="./rfc761">RFC 761</a> [<a href="#ref-15" title="&quot;DoD Standard Transmission Control Protocol&quot;">15</a>]), and SCTP
         (<a href="./rfc2960">RFC 2960</a> [<a href="#ref-16" title="&quot;Stream Control Transmission Protocol&quot;">16</a>]).  For a SIPS URI, the transport parameter MUST
         indicate a reliable transport.

         The maddr parameter indicates the server address to be
         contacted for this user, overriding any address derived from
         the host field.  When an maddr parameter is present, the port
         and transport components of the URI apply to the address
         indicated in the maddr parameter value.  [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] describes the
         proper interpretation of the transport, maddr, and hostport in
         order to obtain the destination address, port, and transport
         for sending a request.

         The maddr field has been used as a simple form of loose source
         routing.  It allows a URI to specify a proxy that must be
         traversed en-route to the destination.  Continuing to use the
         maddr parameter this way is strongly discouraged (the
         mechanisms that enable it are deprecated).  Implementations
         should instead use the Route mechanism described in this
         document, establishing a pre-existing route set if necessary
         (see <a href="#section-8.1.1.1">Section 8.1.1.1</a>).  This provides a full URI to describe
         the node to be traversed.

         The ttl parameter determines the time-to-live value of the UDP
         multicast packet and MUST only be used if maddr is a multicast
         address and the transport protocol is UDP.  For example, to
         specify a call to alice@atlanta.com using multicast to
         239.255.255.1 with a ttl of 15, the following URI would be
         used:

            sip:alice@atlanta.com;maddr=239.255.255.1;ttl=15

         The set of valid telephone-subscriber strings is a subset of
         valid user strings.  The user URI parameter exists to
         distinguish telephone numbers from user names that happen to
         look like telephone numbers.  If the user string contains a
         telephone number formatted as a telephone-subscriber, the user
         parameter value "phone" SHOULD be present.  Even without this
         parameter, recipients of SIP and SIPS URIs MAY interpret the
         pre-@ part as a telephone number if local restrictions on the
         name space for user name allow it.

         The method of the SIP request constructed from the URI can be
         specified with the method parameter.




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         The lr parameter, when present, indicates that the element
         responsible for this resource implements the routing mechanisms
         specified in this document.  This parameter will be used in the
         URIs proxies place into Record-Route header field values, and
         may appear in the URIs in a pre-existing route set.

         This parameter is used to achieve backwards compatibility with
         systems implementing the strict-routing mechanisms of <a href="./rfc2543">RFC 2543</a>
         and the rfc2543bis drafts up to bis-05.  An element preparing
         to send a request based on a URI not containing this parameter
         can assume the receiving element implements strict-routing and
         reformat the message to preserve the information in the
         Request-URI.

         Since the uri-parameter mechanism is extensible, SIP elements
         MUST silently ignore any uri-parameters that they do not
         understand.

      Headers: Header fields to be included in a request constructed
         from the URI.

         Headers fields in the SIP request can be specified with the "?"
         mechanism within a URI.  The header names and values are
         encoded in ampersand separated hname = hvalue pairs.  The
         special hname "body" indicates that the associated hvalue is
         the message-body of the SIP request.

   Table 1 summarizes the use of SIP and SIPS URI components based on
   the context in which the URI appears.  The external column describes
   URIs appearing anywhere outside of a SIP message, for instance on a
   web page or business card.  Entries marked "m" are mandatory, those
   marked "o" are optional, and those marked "-" are not allowed.
   Elements processing URIs SHOULD ignore any disallowed components if
   they are present.  The second column indicates the default value of
   an optional element if it is not present.  "--" indicates that the
   element is either not optional, or has no default value.

   URIs in Contact header fields have different restrictions depending
   on the context in which the header field appears.  One set applies to
   messages that establish and maintain dialogs (INVITE and its 200 (OK)
   response).  The other applies to registration and redirection
   messages (REGISTER, its 200 (OK) response, and 3xx class responses to
   any method).








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<span class="h4"><a class="selflink" id="section-19.1.2" href="#section-19.1.2">19.1.2</a> Character Escaping Requirements</span>

                                                       dialog
                                          reg./redir. Contact/
              default  Req.-URI  To  From  Contact   R-R/Route  external
user          --          o      o    o       o          o         o
password      --          o      o    o       o          o         o
host          --          m      m    m       m          m         m
port          (1)         o      -    -       o          o         o
user-param    ip          o      o    o       o          o         o
method        INVITE      -      -    -       -          -         o
maddr-param   --          o      -    -       o          o         o
ttl-param     1           o      -    -       o          -         o
transp.-param (2)         o      -    -       o          o         o
lr-param      --          o      -    -       -          o         o
other-param   --          o      o    o       o          o         o
headers       --          -      -    -       o          -         o

   (1): The default port value is transport and scheme dependent.  The
   default  is  5060  for  sip: using UDP, TCP, or SCTP.  The default is
   5061 for sip: using TLS over TCP and sips: over TCP.

   (2): The default transport is scheme dependent.  For sip:, it is UDP.
   For sips:, it is TCP.

   Table 1: Use and default values of URI components for SIP header
   field values, Request-URI and references

   SIP follows the requirements and guidelines of <a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>] when
   defining the set of characters that must be escaped in a SIP URI, and
   uses its ""%" HEX HEX" mechanism for escaping.  From <a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>]:

      The set of characters actually reserved within any given URI
      component is defined by that component.  In general, a character
      is reserved if the semantics of the URI changes if the character
      is replaced with its escaped US-ASCII encoding [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>].  Excluded US-
      ASCII characters (<a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>]), such as space and control
      characters and characters used as URI delimiters, also MUST be
      escaped.  URIs MUST NOT contain unescaped space and control
      characters.

   For each component, the set of valid BNF expansions defines exactly
   which characters may appear unescaped.  All other characters MUST be
   escaped.

   For example, "@" is not in the set of characters in the user
   component, so the user "j@s0n" must have at least the @ sign encoded,
   as in "j%40s0n".



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   Expanding the hname and hvalue tokens in <a href="#section-25">Section 25</a> show that all URI
   reserved characters in header field names and values MUST be escaped.

   The telephone-subscriber subset of the user component has special
   escaping considerations.  The set of characters not reserved in the
   <a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>] description of telephone-subscriber contains a number of
   characters in various syntax elements that need to be escaped when
   used in SIP URIs.  Any characters occurring in a telephone-subscriber
   that do not appear in an expansion of the BNF for the user rule MUST
   be escaped.

   Note that character escaping is not allowed in the host component of
   a SIP or SIPS URI (the % character is not valid in its expansion).
   This is likely to change in the future as requirements for
   Internationalized Domain Names are finalized.  Current
   implementations MUST NOT attempt to improve robustness by treating
   received escaped characters in the host component as literally
   equivalent to their unescaped counterpart.  The behavior required to
   meet the requirements of IDN may be significantly different.

<span class="h4"><a class="selflink" id="section-19.1.3" href="#section-19.1.3">19.1.3</a> Example SIP and SIPS URIs</span>

   sip:alice@atlanta.com
   sip:alice:secretword@atlanta.com;transport=tcp
   sips:alice@atlanta.com?subject=project%20x&amp;priority=urgent
   sip:+1-212-555-1212:1234@gateway.com;user=phone
   sips:1212@gateway.com
   sip:alice@192.0.2.4
   sip:atlanta.com;method=REGISTER?to=alice%40atlanta.com
   sip:alice;day=tuesday@atlanta.com

   The last sample URI above has a user field value of
   "alice;day=tuesday".  The escaping rules defined above allow a
   semicolon to appear unescaped in this field.  For the purposes of
   this protocol, the field is opaque.  The structure of that value is
   only useful to the SIP element responsible for the resource.

<span class="h4"><a class="selflink" id="section-19.1.4" href="#section-19.1.4">19.1.4</a> URI Comparison</span>

   Some operations in this specification require determining whether two
   SIP or SIPS URIs are equivalent.  In this specification, registrars
   need to compare bindings in Contact URIs in REGISTER requests (see
   <a href="#section-10.3">Section 10.3</a>.).  SIP and SIPS URIs are compared for equality
   according to the following rules:

      o  A SIP and SIPS URI are never equivalent.





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      o  Comparison of the userinfo of SIP and SIPS URIs is case-
         sensitive.  This includes userinfo containing passwords or
         formatted as telephone-subscribers.  Comparison of all other
         components of the URI is case-insensitive unless explicitly
         defined otherwise.

      o  The ordering of parameters and header fields is not significant
         in comparing SIP and SIPS URIs.

      o  Characters other than those in the "reserved" set (see <a href="./rfc2396">RFC 2396</a>
         [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>]) are equivalent to their ""%" HEX HEX" encoding.

      o  An IP address that is the result of a DNS lookup of a host name
         does not match that host name.

      o  For two URIs to be equal, the user, password, host, and port
         components must match.

         A URI omitting the user component will not match a URI that
         includes one.  A URI omitting the password component will not
         match a URI that includes one.

         A URI omitting any component with a default value will not
         match a URI explicitly containing that component with its
         default value.  For instance, a URI omitting the optional port
         component will not match a URI explicitly declaring port 5060.
         The same is true for the transport-parameter, ttl-parameter,
         user-parameter, and method components.

            Defining sip:user@host to not be equivalent to
            sip:user@host:5060 is a change from <a href="./rfc2543">RFC 2543</a>.  When deriving
            addresses from URIs, equivalent addresses are expected from
            equivalent URIs.  The URI sip:user@host:5060 will always
            resolve to port 5060.  The URI sip:user@host may resolve to
            other ports through the DNS SRV mechanisms detailed in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

      o  URI uri-parameter components are compared as follows:

         -  Any uri-parameter appearing in both URIs must match.

         -  A user, ttl, or method uri-parameter appearing in only one
            URI never matches, even if it contains the default value.

         -  A URI that includes an maddr parameter will not match a URI
            that contains no maddr parameter.

         -  All other uri-parameters appearing in only one URI are
            ignored when comparing the URIs.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 154]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-155" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      o  URI header components are never ignored.  Any present header
         component MUST be present in both URIs and match for the URIs
         to match.  The matching rules are defined for each header field
         in <a href="#section-20">Section 20</a>.

   The URIs within each of the following sets are equivalent:

   sip:%61lice@atlanta.com;transport=TCP
   sip:alice@AtLanTa.CoM;Transport=tcp

   sip:carol@chicago.com
   sip:carol@chicago.com;newparam=5
   sip:carol@chicago.com;security=on

   sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob%40biloxi.com
   sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob%40biloxi.com

   sip:alice@atlanta.com?subject=project%20x&amp;priority=urgent
   sip:alice@atlanta.com?priority=urgent&amp;subject=project%20x

   The URIs within each of the following sets are not equivalent:

   SIP:ALICE@AtLanTa.CoM;Transport=udp             (different usernames)
   sip:alice@AtLanTa.CoM;Transport=UDP

   sip:bob@biloxi.com                   (can resolve to different ports)
   sip:bob@biloxi.com:5060

   sip:bob@biloxi.com              (can resolve to different transports)
   sip:bob@biloxi.com;transport=udp

   sip:bob@biloxi.com     (can resolve to different port and transports)
   sip:bob@biloxi.com:6000;transport=tcp

   sip:carol@chicago.com                    (different header component)
   sip:carol@chicago.com?Subject=next%20meeting

   sip:bob@phone21.boxesbybob.com   (even though that's what
   sip:bob@192.0.2.4                 phone21.boxesbybob.com resolves to)

   Note that equality is not transitive:

      o  sip:carol@chicago.com and sip:carol@chicago.com;security=on are
         equivalent

      o  sip:carol@chicago.com and sip:carol@chicago.com;security=off
         are equivalent




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 155]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-156" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      o  sip:carol@chicago.com;security=on and
         sip:carol@chicago.com;security=off are not equivalent

<span class="h4"><a class="selflink" id="section-19.1.5" href="#section-19.1.5">19.1.5</a> Forming Requests from a URI</span>

   An implementation needs to take care when forming requests directly
   from a URI.  URIs from business cards, web pages, and even from
   sources inside the protocol such as registered contacts may contain
   inappropriate header fields or body parts.

   An implementation MUST include any provided transport, maddr, ttl, or
   user parameter in the Request-URI of the formed request.  If the URI
   contains a method parameter, its value MUST be used as the method of
   the request.  The method parameter MUST NOT be placed in the
   Request-URI.  Unknown URI parameters MUST be placed in the message's
   Request-URI.

   An implementation SHOULD treat the presence of any headers or body
   parts in the URI as a desire to include them in the message, and
   choose to honor the request on a per-component basis.

   An implementation SHOULD NOT honor these obviously dangerous header
   fields: From, Call-ID, CSeq, Via, and Record-Route.

   An implementation SHOULD NOT honor any requested Route header field
   values in order to not be used as an unwitting agent in malicious
   attacks.

   An implementation SHOULD NOT honor requests to include header fields
   that may cause it to falsely advertise its location or capabilities.
   These include: Accept, Accept-Encoding, Accept-Language, Allow,
   Contact (in its dialog usage), Organization, Supported, and User-
   Agent.

   An implementation SHOULD verify the accuracy of any requested
   descriptive header fields, including: Content-Disposition, Content-
   Encoding, Content-Language, Content-Length, Content-Type, Date,
   Mime-Version, and Timestamp.

   If the request formed from constructing a message from a given URI is
   not a valid SIP request, the URI is invalid.  An implementation MUST
   NOT proceed with transmitting the request.  It should instead pursue
   the course of action due an invalid URI in the context it occurs.

      The constructed request can be invalid in many ways.  These
      include, but are not limited to, syntax error in header fields,
      invalid combinations of URI parameters, or an incorrect
      description of the message body.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 156]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-157" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Sending a request formed from a given URI may require capabilities
   unavailable to the implementation.  The URI might indicate use of an
   unimplemented transport or extension, for example.  An implementation
   SHOULD refuse to send these requests rather than modifying them to
   match their capabilities.  An implementation MUST NOT send a request
   requiring an extension that it does not support.

      For example, such a request can be formed through the presence of
      a Require header parameter or a method URI parameter with an
      unknown or explicitly unsupported value.

<span class="h4"><a class="selflink" id="section-19.1.6" href="#section-19.1.6">19.1.6</a> Relating SIP URIs and tel URLs</span>

   When a tel URL (<a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>]) is converted to a SIP or SIPS URI, the
   entire telephone-subscriber portion of the tel URL, including any
   parameters, is placed into the userinfo part of the SIP or SIPS URI.

   Thus, tel:+358-555-1234567;postd=pp22 becomes

      sip:+358-555-1234567;postd=pp22@foo.com;user=phone

   or
      sips:+358-555-1234567;postd=pp22@foo.com;user=phone

   not
      sip:+358-555-1234567@foo.com;postd=pp22;user=phone

   or

      sips:+358-555-1234567@foo.com;postd=pp22;user=phone

   In general, equivalent "tel" URLs converted to SIP or SIPS URIs in
   this fashion may not produce equivalent SIP or SIPS URIs.  The
   userinfo of SIP and SIPS URIs are compared as a case-sensitive
   string.  Variance in case-insensitive portions of tel URLs and
   reordering of tel URL parameters does not affect tel URL equivalence,
   but does affect the equivalence of SIP URIs formed from them.

   For example,

      tel:+358-555-1234567;postd=pp22
      tel:+358-555-1234567;POSTD=PP22

   are equivalent, while

      sip:+358-555-1234567;postd=pp22@foo.com;user=phone
      sip:+358-555-1234567;POSTD=PP22@foo.com;user=phone




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 157]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-158" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   are not.

   Likewise,

      tel:+358-555-1234567;postd=pp22;isub=1411
      tel:+358-555-1234567;isub=1411;postd=pp22

   are equivalent, while

      sip:+358-555-1234567;postd=pp22;isub=1411@foo.com;user=phone
      sip:+358-555-1234567;isub=1411;postd=pp22@foo.com;user=phone

   are not.

   To mitigate this problem, elements constructing telephone-subscriber
   fields to place in the userinfo part of a SIP or SIPS URI SHOULD fold
   any case-insensitive portion of telephone-subscriber to lower case,
   and order the telephone-subscriber parameters lexically by parameter
   name, excepting isdn-subaddress and post-dial, which occur first and
   in that order.  (All components of a tel URL except for future-
   extension parameters are defined to be compared case-insensitive.)

   Following this suggestion, both

      tel:+358-555-1234567;postd=pp22
      tel:+358-555-1234567;POSTD=PP22

      become

        sip:+358-555-1234567;postd=pp22@foo.com;user=phone

   and both

        tel:+358-555-1234567;tsp=a.b;phone-context=5
        tel:+358-555-1234567;phone-context=5;tsp=a.b

      become

        sip:+358-555-1234567;phone-context=5;tsp=a.b@foo.com;user=phone

<span class="h3"><a class="selflink" id="section-19.2" href="#section-19.2">19.2</a> Option Tags</span>

   Option tags are unique identifiers used to designate new options
   (extensions) in SIP.  These tags are used in Require (<a href="#section-20.32">Section 20.32</a>),
   Proxy-Require (<a href="#section-20.29">Section 20.29</a>), Supported (<a href="#section-20.37">Section 20.37</a>) and
   Unsupported (<a href="#section-20.40">Section 20.40</a>) header fields.  Note that these options
   appear as parameters in those header fields in an option-tag = token
   form (see <a href="#section-25">Section 25</a> for the definition of token).



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 158]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-159" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Option tags are defined in standards track RFCs.  This is a change
   from past practice, and is instituted to ensure continuing multi-
   vendor interoperability (see discussion in <a href="#section-20.32">Section 20.32</a> and <a href="#section-20.37">Section</a>
   <a href="#section-20.37">20.37</a>).  An IANA registry of option tags is used to ensure easy
   reference.

<span class="h3"><a class="selflink" id="section-19.3" href="#section-19.3">19.3</a> Tags</span>

   The "tag" parameter is used in the To and From header fields of SIP
   messages.  It serves as a general mechanism to identify a dialog,
   which is the combination of the Call-ID along with two tags, one from
   each participant in the dialog.  When a UA sends a request outside of
   a dialog, it contains a From tag only, providing "half" of the dialog
   ID.  The dialog is completed from the response(s), each of which
   contributes the second half in the To header field.  The forking of
   SIP requests means that multiple dialogs can be established from a
   single request.  This also explains the need for the two-sided dialog
   identifier; without a contribution from the recipients, the
   originator could not disambiguate the multiple dialogs established
   from a single request.

   When a tag is generated by a UA for insertion into a request or
   response, it MUST be globally unique and cryptographically random
   with at least 32 bits of randomness.  A property of this selection
   requirement is that a UA will place a different tag into the From
   header of an INVITE than it would place into the To header of the
   response to the same INVITE.  This is needed in order for a UA to
   invite itself to a session, a common case for "hairpinning" of calls
   in PSTN gateways.  Similarly, two INVITEs for different calls will
   have different From tags, and two responses for different calls will
   have different To tags.

   Besides the requirement for global uniqueness, the algorithm for
   generating a tag is implementation-specific.  Tags are helpful in
   fault tolerant systems, where a dialog is to be recovered on an
   alternate server after a failure.  A UAS can select the tag in such a
   way that a backup can recognize a request as part of a dialog on the
   failed server, and therefore determine that it should attempt to
   recover the dialog and any other state associated with it.

<span class="h2"><a class="selflink" id="section-20" href="#section-20">20</a> Header Fields</span>

   The general syntax for header fields is covered in <a href="#section-7.3">Section 7.3</a>.  This
   section lists the full set of header fields along with notes on
   syntax, meaning, and usage.  Throughout this section, we use [HX.Y]
   to refer to Section X.Y of the current HTTP/1.1 specification <a href="./rfc2616">RFC</a>
   <a href="./rfc2616">2616</a> [<a href="#ref-8" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">8</a>].  Examples of each header field are given.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 159]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-160" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Information about header fields in relation to methods and proxy
   processing is summarized in Tables 2 and 3.

   The "where" column describes the request and response types in which
   the header field can be used.  Values in this column are:

      R: header field may only appear in requests;

      r: header field may only appear in responses;

      2xx, 4xx, etc.: A numerical value or range indicates response
           codes with which the header field can be used;

      c: header field is copied from the request to the response.

      An empty entry in the "where" column indicates that the header
           field may be present in all requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field:

      a: A proxy can add or concatenate the header field if not present.

      m: A proxy can modify an existing header field value.

      d: A proxy can delete a header field value.

      r: A proxy must be able to read the header field, and thus this
           header field cannot be encrypted.

   The next six columns relate to the presence of a header field in a
   method:

      c: Conditional; requirements on the header field depend on the
           context of the message.

      m: The header field is mandatory.

      m*: The header field SHOULD be sent, but clients/servers need to
           be prepared to receive messages without that header field.

      o: The header field is optional.

      t: The header field SHOULD be sent, but clients/servers need to be
           prepared to receive messages without that header field.

           If a stream-based protocol (such as TCP) is used as a
           transport, then the header field MUST be sent.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 160]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-161" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      *: The header field is required if the message body is not empty.
           See Sections <a href="#section-20.14">20.14</a>, <a href="#section-20.15">20.15</a> and <a href="#section-7.4">7.4</a> for details.

      -: The header field is not applicable.

   "Optional" means that an element MAY include the header field in a
   request or response, and a UA MAY ignore the header field if present
   in the request or response (The exception to this rule is the Require
   header field discussed in 20.32).  A "mandatory" header field MUST be
   present in a request, and MUST be understood by the UAS receiving the
   request.  A mandatory response header field MUST be present in the
   response, and the header field MUST be understood by the UAC
   processing the response.  "Not applicable" means that the header
   field MUST NOT be present in a request.  If one is placed in a
   request by mistake, it MUST be ignored by the UAS receiving the
   request.  Similarly, a header field labeled "not applicable" for a
   response means that the UAS MUST NOT place the header field in the
   response, and the UAC MUST ignore the header field in the response.

   A UA SHOULD ignore extension header parameters that are not
   understood.

   A compact form of some common header field names is also defined for
   use when overall message size is an issue.

   The Contact, From, and To header fields contain a URI.  If the URI
   contains a comma, question mark or semicolon, the URI MUST be
   enclosed in angle brackets (&lt; and &gt;).  Any URI parameters are
   contained within these brackets.  If the URI is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URI parameters.

<span class="h3"><a class="selflink" id="section-20.1" href="#section-20.1">20.1</a> Accept</span>

   The Accept header field follows the syntax defined in [H14.1].  The
   semantics are also identical, with the exception that if no Accept
   header field is present, the server SHOULD assume a default value of
   application/sdp.

   An empty Accept header field means that no formats are acceptable.











<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 161]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-162" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Example:

      Header field          where   proxy ACK BYE CAN INV OPT REG
      ___________________________________________________________
      Accept                  R            -   o   -   o   m*  o
      Accept                 2xx           -   -   -   o   m*  o
      Accept                 415           -   c   -   c   c   c
      Accept-Encoding         R            -   o   -   o   o   o
      Accept-Encoding        2xx           -   -   -   o   m*  o
      Accept-Encoding        415           -   c   -   c   c   c
      Accept-Language         R            -   o   -   o   o   o
      Accept-Language        2xx           -   -   -   o   m*  o
      Accept-Language        415           -   c   -   c   c   c
      Alert-Info              R      ar    -   -   -   o   -   -
      Alert-Info             180     ar    -   -   -   o   -   -
      Allow                   R            -   o   -   o   o   o
      Allow                  2xx           -   o   -   m*  m*  o
      Allow                   r            -   o   -   o   o   o
      Allow                  405           -   m   -   m   m   m
      Authentication-Info    2xx           -   o   -   o   o   o
      Authorization           R            o   o   o   o   o   o
      Call-ID                 c       r    m   m   m   m   m   m
      Call-Info                      ar    -   -   -   o   o   o
      Contact                 R            o   -   -   m   o   o
      Contact                1xx           -   -   -   o   -   -
      Contact                2xx           -   -   -   m   o   o
      Contact                3xx      d    -   o   -   o   o   o
      Contact                485           -   o   -   o   o   o
      Content-Disposition                  o   o   -   o   o   o
      Content-Encoding                     o   o   -   o   o   o
      Content-Language                     o   o   -   o   o   o
      Content-Length                 ar    t   t   t   t   t   t
      Content-Type                         *   *   -   *   *   *
      CSeq                    c       r    m   m   m   m   m   m
      Date                            a    o   o   o   o   o   o
      Error-Info           300-699    a    -   o   o   o   o   o
      Expires                              -   -   -   o   -   o
      From                    c       r    m   m   m   m   m   m
      In-Reply-To             R            -   -   -   o   -   -
      Max-Forwards            R      amr   m   m   m   m   m   m
      Min-Expires            423           -   -   -   -   -   m
      MIME-Version                         o   o   -   o   o   o
      Organization                   ar    -   -   -   o   o   o

             Table 2: Summary of header fields, A--O






<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 162]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-163" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Header field              where       proxy ACK BYE CAN INV OPT REG
   ___________________________________________________________________
   Priority                    R          ar    -   -   -   o   -   -
   Proxy-Authenticate         407         ar    -   m   -   m   m   m
   Proxy-Authenticate         401         ar    -   o   o   o   o   o
   Proxy-Authorization         R          dr    o   o   -   o   o   o
   Proxy-Require               R          ar    -   o   -   o   o   o
   Record-Route                R          ar    o   o   o   o   o   -
   Record-Route             2xx,18x       mr    -   o   o   o   o   -
   Reply-To                                     -   -   -   o   -   -
   Require                                ar    -   c   -   c   c   c
   Retry-After          404,413,480,486         -   o   o   o   o   o
                            500,503             -   o   o   o   o   o
                            600,603             -   o   o   o   o   o
   Route                       R          adr   c   c   c   c   c   c
   Server                      r                -   o   o   o   o   o
   Subject                     R                -   -   -   o   -   -
   Supported                   R                -   o   o   m*  o   o
   Supported                  2xx               -   o   o   m*  m*  o
   Timestamp                                    o   o   o   o   o   o
   To                        c(1)          r    m   m   m   m   m   m
   Unsupported                420               -   m   -   m   m   m
   User-Agent                                   o   o   o   o   o   o
   Via                         R          amr   m   m   m   m   m   m
   Via                        rc          dr    m   m   m   m   m   m
   Warning                     r                -   o   o   o   o   o
   WWW-Authenticate           401         ar    -   m   -   m   m   m
   WWW-Authenticate           407         ar    -   o   -   o   o   o

   Table 3: Summary of header fields, P--Z; (1): copied with possible
   addition of tag

      Accept: application/sdp;level=1, application/x-private, text/html

<span class="h3"><a class="selflink" id="section-20.2" href="#section-20.2">20.2</a> Accept-Encoding</span>

   The Accept-Encoding header field is similar to Accept, but restricts
   the content-codings [H3.5] that are acceptable in the response.  See
   [H14.3].  The semantics in SIP are identical to those defined in
   [H14.3].

   An empty Accept-Encoding header field is permissible.  It is
   equivalent to Accept-Encoding: identity, that is, only the identity
   encoding, meaning no encoding, is permissible.

   If no Accept-Encoding header field is present, the server SHOULD
   assume a default value of identity.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 163]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-164" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   This differs slightly from the HTTP definition, which indicates that
   when not present, any encoding can be used, but the identity encoding
   is preferred.

   Example:

      Accept-Encoding: gzip

<span class="h3"><a class="selflink" id="section-20.3" href="#section-20.3">20.3</a> Accept-Language</span>

   The Accept-Language header field is used in requests to indicate the
   preferred languages for reason phrases, session descriptions, or
   status responses carried as message bodies in the response.  If no
   Accept-Language header field is present, the server SHOULD assume all
   languages are acceptable to the client.

   The Accept-Language header field follows the syntax defined in
   [H14.4].  The rules for ordering the languages based on the "q"
   parameter apply to SIP as well.

   Example:

      Accept-Language: da, en-gb;q=0.8, en;q=0.7

<span class="h3"><a class="selflink" id="section-20.4" href="#section-20.4">20.4</a> Alert-Info</span>

   When present in an INVITE request, the Alert-Info header field
   specifies an alternative ring tone to the UAS.  When present in a 180
   (Ringing) response, the Alert-Info header field specifies an
   alternative ringback tone to the UAC.  A typical usage is for a proxy
   to insert this header field to provide a distinctive ring feature.

   The Alert-Info header field can introduce security risks.  These
   risks and the ways to handle them are discussed in <a href="#section-20.9">Section 20.9</a>,
   which discusses the Call-Info header field since the risks are
   identical.

   In addition, a user SHOULD be able to disable this feature
   selectively.

      This helps prevent disruptions that could result from the use of
      this header field by untrusted elements.

   Example:

      Alert-Info: &lt;http://www.example.com/sounds/moo.wav&gt;





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 164]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-165" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h3"><a class="selflink" id="section-20.5" href="#section-20.5">20.5</a> Allow</span>

   The Allow header field lists the set of methods supported by the UA
   generating the message.

   All methods, including ACK and CANCEL, understood by the UA MUST be
   included in the list of methods in the Allow header field, when
   present.  The absence of an Allow header field MUST NOT be
   interpreted to mean that the UA sending the message supports no
   methods.   Rather, it implies that the UA is not providing any
   information on what methods it supports.

   Supplying an Allow header field in responses to methods other than
   OPTIONS reduces the number of messages needed.

   Example:

      Allow: INVITE, ACK, OPTIONS, CANCEL, BYE

<span class="h3"><a class="selflink" id="section-20.6" href="#section-20.6">20.6</a> Authentication-Info</span>

   The Authentication-Info header field provides for mutual
   authentication with HTTP Digest.  A UAS MAY include this header field
   in a 2xx response to a request that was successfully authenticated
   using digest based on the Authorization header field.

   Syntax and semantics follow those specified in <a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>].

   Example:

      Authentication-Info: nextnonce="47364c23432d2e131a5fb210812c"

<span class="h3"><a class="selflink" id="section-20.7" href="#section-20.7">20.7</a> Authorization</span>

   The Authorization header field contains authentication credentials of
   a UA.  <a href="#section-22.2">Section 22.2</a> overviews the use of the Authorization header
   field, and <a href="#section-22.4">Section 22.4</a> describes the syntax and semantics when used
   with HTTP authentication.

   This header field, along with Proxy-Authorization, breaks the general
   rules about multiple header field values.  Although not a comma-
   separated list, this header field name may be present multiple times,
   and MUST NOT be combined into a single header line using the usual
   rules described in <a href="#section-7.3">Section 7.3</a>.







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   In the example below, there are no quotes around the Digest
   parameter:

      Authorization: Digest username="Alice", realm="atlanta.com",
       nonce="84a4cc6f3082121f32b42a2187831a9e",
       response="7587245234b3434cc3412213e5f113a5432"

<span class="h3"><a class="selflink" id="section-20.8" href="#section-20.8">20.8</a> Call-ID</span>

   The Call-ID header field uniquely identifies a particular invitation
   or all registrations of a particular client.  A single multimedia
   conference can give rise to several calls with different Call-IDs,
   for example, if a user invites a single individual several times to
   the same (long-running) conference.  Call-IDs are case-sensitive and
   are simply compared byte-by-byte.

   The compact form of the Call-ID header field is i.

   Examples:

      Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@biloxi.com
      i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@192.0.2.4

<span class="h3"><a class="selflink" id="section-20.9" href="#section-20.9">20.9</a> Call-Info</span>

   The Call-Info header field provides additional information about the
   caller or callee, depending on whether it is found in a request or
   response.  The purpose of the URI is described by the "purpose"
   parameter.  The "icon" parameter designates an image suitable as an
   iconic representation of the caller or callee.  The "info" parameter
   describes the caller or callee in general, for example, through a web
   page.  The "card" parameter provides a business card, for example, in
   vCard [<a href="#ref-36" title="&quot;vCard MIME Directory Profile&quot;">36</a>] or LDIF [<a href="#ref-37" title="&quot;The LDAP Data Interchange Format (LDIF) - Technical Specification&quot;">37</a>] formats.  Additional tokens can be registered
   using IANA and the procedures in <a href="#section-27">Section 27</a>.

   Use of the Call-Info header field can pose a security risk.  If a
   callee fetches the URIs provided by a malicious caller, the callee
   may be at risk for displaying inappropriate or offensive content,
   dangerous or illegal content, and so on.  Therefore, it is
   RECOMMENDED that a UA only render the information in the Call-Info
   header field if it can verify the authenticity of the element that
   originated the header field and trusts that element.  This need not
   be the peer UA; a proxy can insert this header field into requests.

   Example:

   Call-Info: &lt;http://wwww.example.com/alice/photo.jpg&gt; ;purpose=icon,
     &lt;http://www.example.com/alice/&gt; ;purpose=info



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<span class="h3"><a class="selflink" id="section-20.10" href="#section-20.10">20.10</a> Contact</span>

   A Contact header field value provides a URI whose meaning depends on
   the type of request or response it is in.

   A Contact header field value can contain a display name, a URI with
   URI parameters, and header parameters.

   This document defines the Contact parameters "q" and "expires".
   These parameters are only used when the Contact is present in a
   REGISTER request or response, or in a 3xx response.  Additional
   parameters may be defined in other specifications.

   When the header field value contains a display name, the URI
   including all URI parameters is enclosed in "&lt;" and "&gt;".  If no "&lt;"
   and "&gt;" are present, all parameters after the URI are header
   parameters, not URI parameters.  The display name can be tokens, or a
   quoted string, if a larger character set is desired.

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, semicolon, or question
   mark.  There may or may not be LWS between the display-name and the
   "&lt;".

   These rules for parsing a display name, URI and URI parameters, and
   header parameters also apply for the header fields To and From.

      The Contact header field has a role similar to the Location header
      field in HTTP.  However, the HTTP header field only allows one
      address, unquoted.  Since URIs can contain commas and semicolons
      as reserved characters, they can be mistaken for header or
      parameter delimiters, respectively.

   The compact form of the Contact header field is m (for "moved").

   Examples:

      Contact: "Mr. Watson" &lt;sip:watson@worcester.bell-telephone.com&gt;
         ;q=0.7; expires=3600,
         "Mr. Watson" &lt;mailto:watson@bell-telephone.com&gt; ;q=0.1
      m: &lt;sips:bob@192.0.2.4&gt;;expires=60










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<span class="h3"><a class="selflink" id="section-20.11" href="#section-20.11">20.11</a> Content-Disposition</span>

   The Content-Disposition header field describes how the message body
   or, for multipart messages, a message body part is to be interpreted
   by the UAC or UAS.  This SIP header field extends the MIME Content-
   Type (<a href="./rfc2183">RFC 2183</a> [<a href="#ref-18" title="&quot;Communicating Presentation Information in Internet Messages: The Content-Disposition Header Field&quot;">18</a>]).

   Several new "disposition-types" of the Content-Disposition header are
   defined by SIP.  The value "session" indicates that the body part
   describes a session, for either calls or early (pre-call) media.  The
   value "render" indicates that the body part should be displayed or
   otherwise rendered to the user.  Note that the value "render" is used
   rather than "inline" to avoid the connotation that the MIME body is
   displayed as a part of the rendering of the entire message (since the
   MIME bodies of SIP messages oftentimes are not displayed to users).
   For backward-compatibility, if the Content-Disposition header field
   is missing, the server SHOULD assume bodies of Content-Type
   application/sdp are the disposition "session", while other content
   types are "render".

   The disposition type "icon" indicates that the body part contains an
   image suitable as an iconic representation of the caller or callee
   that could be rendered informationally by a user agent when a message
   has been received, or persistently while a dialog takes place.  The
   value "alert" indicates that the body part contains information, such
   as an audio clip, that should be rendered by the user agent in an
   attempt to alert the user to the receipt of a request, generally a
   request that initiates a dialog; this alerting body could for example
   be rendered as a ring tone for a phone call after a 180 Ringing
   provisional response has been sent.

   Any MIME body with a "disposition-type" that renders content to the
   user should only be processed when a message has been properly
   authenticated.

   The handling parameter, handling-param, describes how the UAS should
   react if it receives a message body whose content type or disposition
   type it does not understand.  The parameter has defined values of
   "optional" and "required".  If the handling parameter is missing, the
   value "required" SHOULD be assumed.  The handling parameter is
   described in <a href="./rfc3204">RFC 3204</a> [<a href="#ref-19" title="&quot;MIME media types for ISUP and QSIG Objects&quot;">19</a>].

   If this header field is missing, the MIME type determines the default
   content disposition.  If there is none, "render" is assumed.

   Example:

      Content-Disposition: session



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<span class="h3"><a class="selflink" id="section-20.12" href="#section-20.12">20.12</a> Content-Encoding</span>

   The Content-Encoding header field is used as a modifier to the
   "media-type".  When present, its value indicates what additional
   content codings have been applied to the entity-body, and thus what
   decoding mechanisms MUST be applied in order to obtain the media-type
   referenced by the Content-Type header field.  Content-Encoding is
   primarily used to allow a body to be compressed without losing the
   identity of its underlying media type.

   If multiple encodings have been applied to an entity-body, the
   content codings MUST be listed in the order in which they were
   applied.

   All content-coding values are case-insensitive.  IANA acts as a
   registry for content-coding value tokens.  See [H3.5] for a
   definition of the syntax for content-coding.

   Clients MAY apply content encodings to the body in requests.  A
   server MAY apply content encodings to the bodies in responses.  The
   server MUST only use encodings listed in the Accept-Encoding header
   field in the request.

   The compact form of the Content-Encoding header field is e.
   Examples:

      Content-Encoding: gzip
      e: tar

<span class="h3"><a class="selflink" id="section-20.13" href="#section-20.13">20.13</a> Content-Language</span>

   See [H14.12]. Example:

      Content-Language: fr

<span class="h3"><a class="selflink" id="section-20.14" href="#section-20.14">20.14</a> Content-Length</span>

   The Content-Length header field indicates the size of the message-
   body, in decimal number of octets, sent to the recipient.
   Applications SHOULD use this field to indicate the size of the
   message-body to be transferred, regardless of the media type of the
   entity.  If a stream-based protocol (such as TCP) is used as
   transport, the header field MUST be used.

   The size of the message-body does not include the CRLF separating
   header fields and body.  Any Content-Length greater than or equal to
   zero is a valid value.  If no body is present in a message, then the
   Content-Length header field value MUST be set to zero.



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      The ability to omit Content-Length simplifies the creation of
      cgi-like scripts that dynamically generate responses.

   The compact form of the header field is l.

   Examples:

      Content-Length: 349
      l: 173

<span class="h3"><a class="selflink" id="section-20.15" href="#section-20.15">20.15</a> Content-Type</span>

   The Content-Type header field indicates the media type of the
   message-body sent to the recipient.  The "media-type" element is
   defined in [H3.7].  The Content-Type header field MUST be present if
   the body is not empty.  If the body is empty, and a Content-Type
   header field is present, it indicates that the body of the specific
   type has zero length (for example, an empty audio file).

   The compact form of the header field is c.

   Examples:

      Content-Type: application/sdp
      c: text/html; charset=ISO-8859-4

<span class="h3"><a class="selflink" id="section-20.16" href="#section-20.16">20.16</a> CSeq</span>

   A CSeq header field in a request contains a single decimal sequence
   number and the request method.  The sequence number MUST be
   expressible as a 32-bit unsigned integer.  The method part of CSeq is
   case-sensitive.  The CSeq header field serves to order transactions
   within a dialog, to provide a means to uniquely identify
   transactions, and to differentiate between new requests and request
   retransmissions.  Two CSeq header fields are considered equal if the
   sequence number and the request method are identical.  Example:

      CSeq: 4711 INVITE

<span class="h3"><a class="selflink" id="section-20.17" href="#section-20.17">20.17</a> Date</span>

   The Date header field contains the date and time.  Unlike HTTP/1.1,
   SIP only supports the most recent <a href="./rfc1123">RFC 1123</a> [<a href="#ref-20" title="&quot;Requirements for Internet Hosts - Application and Support&quot;">20</a>] format for dates.  As
   in [H3.3], SIP restricts the time zone in SIP-date to "GMT", while
   <a href="./rfc1123">RFC 1123</a> allows any time zone.  An <a href="./rfc1123">RFC 1123</a> date is case-sensitive.

   The Date header field reflects the time when the request or response
   is first sent.



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      The Date header field can be used by simple end systems without a
      battery-backed clock to acquire a notion of current time.
      However, in its GMT form, it requires clients to know their offset
      from GMT.

   Example:

      Date: Sat, 13 Nov 2010 23:29:00 GMT

<span class="h3"><a class="selflink" id="section-20.18" href="#section-20.18">20.18</a> Error-Info</span>

   The Error-Info header field provides a pointer to additional
   information about the error status response.

      SIP UACs have user interface capabilities ranging from pop-up
      windows and audio on PC softclients to audio-only on "black"
      phones or endpoints connected via gateways.  Rather than forcing a
      server generating an error to choose between sending an error
      status code with a detailed reason phrase and playing an audio
      recording, the Error-Info header field allows both to be sent.
      The UAC then has the choice of which error indicator to render to
      the caller.

   A UAC MAY treat a SIP or SIPS URI in an Error-Info header field as if
   it were a Contact in a redirect and generate a new INVITE, resulting
   in a recorded announcement session being established.  A non-SIP URI
   MAY be rendered to the user.

   Examples:

      SIP/2.0 404 The number you have dialed is not in service
      Error-Info: &lt;sip:not-in-service-recording@atlanta.com&gt;

<span class="h3"><a class="selflink" id="section-20.19" href="#section-20.19">20.19</a> Expires</span>

   The Expires header field gives the relative time after which the
   message (or content) expires.

   The precise meaning of this is method dependent.

   The expiration time in an INVITE does not affect the duration of the
   actual session that may result from the invitation.  Session
   description protocols may offer the ability to express time limits on
   the session duration, however.

   The value of this field is an integral number of seconds (in decimal)
   between 0 and (2**32)-1, measured from the receipt of the request.




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   Example:

      Expires: 5

<span class="h3"><a class="selflink" id="section-20.20" href="#section-20.20">20.20</a> From</span>

   The From header field indicates the initiator of the request.  This
   may be different from the initiator of the dialog.  Requests sent by
   the callee to the caller use the callee's address in the From header
   field.

   The optional "display-name" is meant to be rendered by a human user
   interface.  A system SHOULD use the display name "Anonymous" if the
   identity of the client is to remain hidden.  Even if the "display-
   name" is empty, the "name-addr" form MUST be used if the "addr-spec"
   contains a comma, question mark, or semicolon.  Syntax issues are
   discussed in <a href="#section-7.3.1">Section 7.3.1</a>.

   Two From header fields are equivalent if their URIs match, and their
   parameters match. Extension parameters in one header field, not
   present in the other are ignored for the purposes of comparison. This
   means that the display name and presence or absence of angle brackets
   do not affect matching.

   See <a href="#section-20.10">Section 20.10</a> for the rules for parsing a display name, URI and
   URI parameters, and header field parameters.

   The compact form of the From header field is f.

   Examples:

      From: "A. G. Bell" &lt;sip:agb@bell-telephone.com&gt; ;tag=a48s
      From: sip:+12125551212@server.phone2net.com;tag=887s
      f: Anonymous &lt;sip:c8oqz84zk7z@privacy.org&gt;;tag=hyh8

<span class="h3"><a class="selflink" id="section-20.21" href="#section-20.21">20.21</a> In-Reply-To</span>

   The In-Reply-To header field enumerates the Call-IDs that this call
   references or returns.  These Call-IDs may have been cached by the
   client then included in this header field in a return call.

      This allows automatic call distribution systems to route return
      calls to the originator of the first call.  This also allows
      callees to filter calls, so that only return calls for calls they
      originated will be accepted.  This field is not a substitute for
      request authentication.





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   Example:

      In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com

<span class="h3"><a class="selflink" id="section-20.22" href="#section-20.22">20.22</a> Max-Forwards</span>

   The Max-Forwards header field must be used with any SIP method to
   limit the number of proxies or gateways that can forward the request
   to the next downstream server.  This can also be useful when the
   client is attempting to trace a request chain that appears to be
   failing or looping in mid-chain.

   The Max-Forwards value is an integer in the range 0-255 indicating
   the remaining number of times this request message is allowed to be
   forwarded.  This count is decremented by each server that forwards
   the request.  The recommended initial value is 70.

   This header field should be inserted by elements that can not
   otherwise guarantee loop detection.  For example, a B2BUA should
   insert a Max-Forwards header field.

   Example:

      Max-Forwards: 6

<span class="h3"><a class="selflink" id="section-20.23" href="#section-20.23">20.23</a> Min-Expires</span>

   The Min-Expires header field conveys the minimum refresh interval
   supported for soft-state elements managed by that server.  This
   includes Contact header fields that are stored by a registrar.  The
   header field contains a decimal integer number of seconds from 0 to
   (2**32)-1.  The use of the header field in a 423 (Interval Too Brief)
   response is described in Sections <a href="#section-10.2.8">10.2.8</a>, <a href="#section-10.3">10.3</a>, and <a href="#section-21.4.17">21.4.17</a>.

   Example:

      Min-Expires: 60

<span class="h3"><a class="selflink" id="section-20.24" href="#section-20.24">20.24</a> MIME-Version</span>

   See [H19.4.1].

   Example:

      MIME-Version: 1.0






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<span class="h3"><a class="selflink" id="section-20.25" href="#section-20.25">20.25</a> Organization</span>

   The Organization header field conveys the name of the organization to
   which the SIP element issuing the request or response belongs.

      The field MAY be used by client software to filter calls.

   Example:

      Organization: Boxes by Bob

<span class="h3"><a class="selflink" id="section-20.26" href="#section-20.26">20.26</a> Priority</span>

   The Priority header field indicates the urgency of the request as
   perceived by the client.  The Priority header field describes the
   priority that the SIP request should have to the receiving human or
   its agent.  For example, it may be factored into decisions about call
   routing and acceptance.  For these decisions, a message containing no
   Priority header field SHOULD be treated as if it specified a Priority
   of "normal".  The Priority header field does not influence the use of
   communications resources such as packet forwarding priority in
   routers or access to circuits in PSTN gateways.  The header field can
   have the values "non-urgent", "normal", "urgent", and "emergency",
   but additional values can be defined elsewhere.  It is RECOMMENDED
   that the value of "emergency" only be used when life, limb, or
   property are in imminent danger.  Otherwise, there are no semantics
   defined for this header field.

      These are the values of <a href="./rfc2076">RFC 2076</a> [<a href="#ref-38" title="&quot;Common Internet Message Headers&quot;">38</a>], with the addition of
      "emergency".

   Examples:

      Subject: A tornado is heading our way!
      Priority: emergency

   or

      Subject: Weekend plans
      Priority: non-urgent

<span class="h3"><a class="selflink" id="section-20.27" href="#section-20.27">20.27</a> Proxy-Authenticate</span>

   A Proxy-Authenticate header field value contains an authentication
   challenge.

   The use of this header field is defined in [H14.33].  See <a href="#section-22.3">Section</a>
   <a href="#section-22.3">22.3</a> for further details on its usage.



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   Example:

      Proxy-Authenticate: Digest realm="atlanta.com",
       domain="sip:ss1.carrier.com", qop="auth",
       nonce="f84f1cec41e6cbe5aea9c8e88d359",
       opaque="", stale=FALSE, algorithm=MD5

<span class="h3"><a class="selflink" id="section-20.28" href="#section-20.28">20.28</a> Proxy-Authorization</span>

   The Proxy-Authorization header field allows the client to identify
   itself (or its user) to a proxy that requires authentication.  A
   Proxy-Authorization field value consists of credentials containing
   the authentication information of the user agent for the proxy and/or
   realm of the resource being requested.

   See <a href="#section-22.3">Section 22.3</a> for a definition of the usage of this header field.

   This header field, along with Authorization, breaks the general rules
   about multiple header field names.  Although not a comma-separated
   list, this header field name may be present multiple times, and MUST
   NOT be combined into a single header line using the usual rules
   described in <a href="#section-7.3.1">Section 7.3.1</a>.

   Example:

   Proxy-Authorization: Digest username="Alice", realm="atlanta.com",
      nonce="c60f3082ee1212b402a21831ae",
      response="245f23415f11432b3434341c022"

<span class="h3"><a class="selflink" id="section-20.29" href="#section-20.29">20.29</a> Proxy-Require</span>

   The Proxy-Require header field is used to indicate proxy-sensitive
   features that must be supported by the proxy.  See <a href="#section-20.32">Section 20.32</a> for
   more details on the mechanics of this message and a usage example.

   Example:

      Proxy-Require: foo

<span class="h3"><a class="selflink" id="section-20.30" href="#section-20.30">20.30</a> Record-Route</span>

   The Record-Route header field is inserted by proxies in a request to
   force future requests in the dialog to be routed through the proxy.

   Examples of its use with the Route header field are described in
   Sections <a href="#section-16.12.1">16.12.1</a>.





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   Example:

      Record-Route: &lt;sip:server10.biloxi.com;lr&gt;,
                    &lt;sip:bigbox3.site3.atlanta.com;lr&gt;

<span class="h3"><a class="selflink" id="section-20.31" href="#section-20.31">20.31</a> Reply-To</span>

   The Reply-To header field contains a logical return URI that may be
   different from the From header field.  For example, the URI MAY be
   used to return missed calls or unestablished sessions.  If the user
   wished to remain anonymous, the header field SHOULD either be omitted
   from the request or populated in such a way that does not reveal any
   private information.

   Even if the "display-name" is empty, the "name-addr" form MUST be
   used if the "addr-spec" contains a comma, question mark, or
   semicolon.  Syntax issues are discussed in <a href="#section-7.3.1">Section 7.3.1</a>.

   Example:

      Reply-To: Bob &lt;sip:bob@biloxi.com&gt;

<span class="h3"><a class="selflink" id="section-20.32" href="#section-20.32">20.32</a> Require</span>

   The Require header field is used by UACs to tell UASs about options
   that the UAC expects the UAS to support in order to process the
   request.  Although an optional header field, the Require MUST NOT be
   ignored if it is present.

   The Require header field contains a list of option tags, described in
   <a href="#section-19.2">Section 19.2</a>.  Each option tag defines a SIP extension that MUST be
   understood to process the request.  Frequently, this is used to
   indicate that a specific set of extension header fields need to be
   understood.  A UAC compliant to this specification MUST only include
   option tags corresponding to standards-track RFCs.

   Example:

      Require: 100rel

<span class="h3"><a class="selflink" id="section-20.33" href="#section-20.33">20.33</a> Retry-After</span>

   The Retry-After header field can be used with a 500 (Server Internal
   Error) or 503 (Service Unavailable) response to indicate how long the
   service is expected to be unavailable to the requesting client and
   with a 404 (Not Found), 413 (Request Entity Too Large), 480
   (Temporarily Unavailable), 486 (Busy Here), 600 (Busy), or 603




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   (Decline) response to indicate when the called party anticipates
   being available again.  The value of this field is a positive integer
   number of seconds (in decimal) after the time of the response.

   An optional comment can be used to indicate additional information
   about the time of callback.  An optional "duration" parameter
   indicates how long the called party will be reachable starting at the
   initial time of availability.  If no duration parameter is given, the
   service is assumed to be available indefinitely.

   Examples:

      Retry-After: 18000;duration=3600
      Retry-After: 120 (I'm in a meeting)

<span class="h3"><a class="selflink" id="section-20.34" href="#section-20.34">20.34</a> Route</span>

   The Route header field is used to force routing for a request through
   the listed set of proxies.  Examples of the use of the Route header
   field are in <a href="#section-16.12.1">Section 16.12.1</a>.

   Example:

      Route: &lt;sip:bigbox3.site3.atlanta.com;lr&gt;,
             &lt;sip:server10.biloxi.com;lr&gt;

<span class="h3"><a class="selflink" id="section-20.35" href="#section-20.35">20.35</a> Server</span>

   The Server header field contains information about the software used
   by the UAS to handle the request.

   Revealing the specific software version of the server might allow the
   server to become more vulnerable to attacks against software that is
   known to contain security holes.  Implementers SHOULD make the Server
   header field a configurable option.

   Example:

      Server: HomeServer v2

<span class="h3"><a class="selflink" id="section-20.36" href="#section-20.36">20.36</a> Subject</span>

   The Subject header field provides a summary or indicates the nature
   of the call, allowing call filtering without having to parse the
   session description.  The session description does not have to use
   the same subject indication as the invitation.

   The compact form of the Subject header field is s.



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   Example:

      Subject: Need more boxes
      s: Tech Support

<span class="h3"><a class="selflink" id="section-20.37" href="#section-20.37">20.37</a> Supported</span>

   The Supported header field enumerates all the extensions supported by
   the UAC or UAS.

   The Supported header field contains a list of option tags, described
   in <a href="#section-19.2">Section 19.2</a>, that are understood by the UAC or UAS.  A UA
   compliant to this specification MUST only include option tags
   corresponding to standards-track RFCs.  If empty, it means that no
   extensions are supported.

   The compact form of the Supported header field is k.

   Example:

      Supported: 100rel

<span class="h3"><a class="selflink" id="section-20.38" href="#section-20.38">20.38</a> Timestamp</span>

   The Timestamp header field describes when the UAC sent the request to
   the UAS.

   See <a href="#section-8.2.6">Section 8.2.6</a> for details on how to generate a response to a
   request that contains the header field.  Although there is no
   normative behavior defined here that makes use of the header, it
   allows for extensions or SIP applications to obtain RTT estimates.

   Example:

      Timestamp: 54

<span class="h3"><a class="selflink" id="section-20.39" href="#section-20.39">20.39</a> To</span>

   The To header field specifies the logical recipient of the request.

   The optional "display-name" is meant to be rendered by a human-user
   interface.  The "tag" parameter serves as a general mechanism for
   dialog identification.

   See <a href="#section-19.3">Section 19.3</a> for details of the "tag" parameter.






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   Comparison of To header fields for equality is identical to
   comparison of From header fields.  See <a href="#section-20.10">Section 20.10</a> for the rules
   for parsing a display name, URI and URI parameters, and header field
   parameters.

   The compact form of the To header field is t.

   The following are examples of valid To header fields:

      To: The Operator &lt;sip:operator@cs.columbia.edu&gt;;tag=287447
      t: sip:+12125551212@server.phone2net.com

<span class="h3"><a class="selflink" id="section-20.40" href="#section-20.40">20.40</a> Unsupported</span>

   The Unsupported header field lists the features not supported by the
   UAS.  See <a href="#section-20.32">Section 20.32</a> for motivation.

   Example:

      Unsupported: foo

<span class="h3"><a class="selflink" id="section-20.41" href="#section-20.41">20.41</a> User-Agent</span>

   The User-Agent header field contains information about the UAC
   originating the request.  The semantics of this header field are
   defined in [H14.43].

   Revealing the specific software version of the user agent might allow
   the user agent to become more vulnerable to attacks against software
   that is known to contain security holes.  Implementers SHOULD make
   the User-Agent header field a configurable option.

   Example:

      User-Agent: Softphone Beta1.5

<span class="h3"><a class="selflink" id="section-20.42" href="#section-20.42">20.42</a> Via</span>

   The Via header field indicates the path taken by the request so far
   and indicates the path that should be followed in routing responses.
   The branch ID parameter in the Via header field values serves as a
   transaction identifier, and is used by proxies to detect loops.

   A Via header field value contains the transport protocol used to send
   the message, the client's host name or network address, and possibly
   the port number at which it wishes to receive responses.  A Via
   header field value can also contain parameters such as "maddr",
   "ttl", "received", and "branch", whose meaning and use are described



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   in other sections.  For implementations compliant to this
   specification, the value of the branch parameter MUST start with the
   magic cookie "z9hG4bK", as discussed in <a href="#section-8.1.1.7">Section 8.1.1.7</a>.

   Transport protocols defined here are "UDP", "TCP", "TLS", and "SCTP".
   "TLS" means TLS over TCP.  When a request is sent to a SIPS URI, the
   protocol still indicates "SIP", and the transport protocol is TLS.

Via: SIP/2.0/UDP erlang.bell-telephone.com:5060;branch=z9hG4bK87asdks7
Via: SIP/2.0/UDP 192.0.2.1:5060 ;received=192.0.2.207
     ;branch=z9hG4bK77asjd

   The compact form of the Via header field is v.

   In this example, the message originated from a multi-homed host with
   two addresses, 192.0.2.1 and 192.0.2.207.  The sender guessed wrong
   as to which network interface would be used.  Erlang.bell-
   telephone.com noticed the mismatch and added a parameter to the
   previous hop's Via header field value, containing the address that
   the packet actually came from.

   The host or network address and port number are not required to
   follow the SIP URI syntax.  Specifically, LWS on either side of the
   ":" or "/" is allowed, as shown here:

      Via: SIP / 2.0 / UDP first.example.com: 4000;ttl=16
        ;maddr=224.2.0.1 ;branch=z9hG4bKa7c6a8dlze.1

   Even though this specification mandates that the branch parameter be
   present in all requests, the BNF for the header field indicates that
   it is optional.  This allows interoperation with <a href="./rfc2543">RFC 2543</a> elements,
   which did not have to insert the branch parameter.

   Two Via header fields are equal if their sent-protocol and sent-by
   fields are equal, both have the same set of parameters, and the
   values of all parameters are equal.

<span class="h3"><a class="selflink" id="section-20.43" href="#section-20.43">20.43</a> Warning</span>

   The Warning header field is used to carry additional information
   about the status of a response.  Warning header field values are sent
   with responses and contain a three-digit warning code, host name, and
   warning text.

   The "warn-text" should be in a natural language that is most likely
   to be intelligible to the human user receiving the response.  This
   decision can be based on any available knowledge, such as the
   location of the user, the Accept-Language field in a request, or the



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   Content-Language field in a response.  The default language is i-
   default [<a href="#ref-21" title="&quot;IETF Policy on Character Sets and Languages&quot;">21</a>].

   The currently-defined "warn-code"s are listed below, with a
   recommended warn-text in English and a description of their meaning.
   These warnings describe failures induced by the session description.
   The first digit of warning codes beginning with "3" indicates
   warnings specific to SIP.  Warnings 300 through 329 are reserved for
   indicating problems with keywords in the session description, 330
   through 339 are warnings related to basic network services requested
   in the session description, 370 through 379 are warnings related to
   quantitative QoS parameters requested in the session description, and
   390 through 399 are miscellaneous warnings that do not fall into one
   of the above categories.

      300 Incompatible network protocol: One or more network protocols
          contained in the session description are not available.

      301 Incompatible network address formats: One or more network
          address formats contained in the session description are not
          available.

      302 Incompatible transport protocol: One or more transport
          protocols described in the session description are not
          available.

      303 Incompatible bandwidth units: One or more bandwidth
          measurement units contained in the session description were
          not understood.

      304 Media type not available: One or more media types contained in
          the session description are not available.

      305 Incompatible media format: One or more media formats contained
          in the session description are not available.

      306 Attribute not understood: One or more of the media attributes
          in the session description are not supported.

      307 Session description parameter not understood: A parameter
          other than those listed above was not understood.

      330 Multicast not available: The site where the user is located
          does not support multicast.

      331 Unicast not available: The site where the user is located does
          not support unicast communication (usually due to the presence
          of a firewall).



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      370 Insufficient bandwidth: The bandwidth specified in the session
          description or defined by the media exceeds that known to be
          available.

      399 Miscellaneous warning: The warning text can include arbitrary
          information to be presented to a human user or logged.  A
          system receiving this warning MUST NOT take any automated
          action.

             1xx and 2xx have been taken by HTTP/1.1.

   Additional "warn-code"s can be defined through IANA, as defined in
   <a href="#section-27.2">Section 27.2</a>.

   Examples:

      Warning: 307 isi.edu "Session parameter 'foo' not understood"
      Warning: 301 isi.edu "Incompatible network address type 'E.164'"

<span class="h3"><a class="selflink" id="section-20.44" href="#section-20.44">20.44</a> WWW-Authenticate</span>

   A WWW-Authenticate header field value contains an authentication
   challenge.  See <a href="#section-22.2">Section 22.2</a> for further details on its usage.

   Example:

      WWW-Authenticate: Digest realm="atlanta.com",
        domain="sip:boxesbybob.com", qop="auth",
        nonce="f84f1cec41e6cbe5aea9c8e88d359",
        opaque="", stale=FALSE, algorithm=MD5

<span class="h2"><a class="selflink" id="section-21" href="#section-21">21</a> Response Codes</span>

   The response codes are consistent with, and extend, HTTP/1.1 response
   codes.  Not all HTTP/1.1 response codes are appropriate, and only
   those that are appropriate are given here.  Other HTTP/1.1 response
   codes SHOULD NOT be used.  Also, SIP defines a new class, 6xx.

<span class="h3"><a class="selflink" id="section-21.1" href="#section-21.1">21.1</a> Provisional 1xx</span>

   Provisional responses, also known as informational responses,
   indicate that the server contacted is performing some further action
   and does not yet have a definitive response.  A server sends a 1xx
   response if it expects to take more than 200 ms to obtain a final
   response.  Note that 1xx responses are not transmitted reliably.
   They never cause the client to send an ACK.  Provisional (1xx)
   responses MAY contain message bodies, including session descriptions.




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<span class="h4"><a class="selflink" id="section-21.1.1" href="#section-21.1.1">21.1.1</a> 100 Trying</span>

   This response indicates that the request has been received by the
   next-hop server and that some unspecified action is being taken on
   behalf of this call (for example, a database is being consulted).
   This response, like all other provisional responses, stops
   retransmissions of an INVITE by a UAC.  The 100 (Trying) response is
   different from other provisional responses, in that it is never
   forwarded upstream by a stateful proxy.

<span class="h4"><a class="selflink" id="section-21.1.2" href="#section-21.1.2">21.1.2</a> 180 Ringing</span>

   The UA receiving the INVITE is trying to alert the user.  This
   response MAY be used to initiate local ringback.

<span class="h4"><a class="selflink" id="section-21.1.3" href="#section-21.1.3">21.1.3</a> 181 Call Is Being Forwarded</span>

   A server MAY use this status code to indicate that the call is being
   forwarded to a different set of destinations.

<span class="h4"><a class="selflink" id="section-21.1.4" href="#section-21.1.4">21.1.4</a> 182 Queued</span>

   The called party is temporarily unavailable, but the server has
   decided to queue the call rather than reject it.  When the callee
   becomes available, it will return the appropriate final status
   response.  The reason phrase MAY give further details about the
   status of the call, for example, "5 calls queued; expected waiting
   time is 15 minutes".  The server MAY issue several 182 (Queued)
   responses to update the caller about the status of the queued call.

<span class="h4"><a class="selflink" id="section-21.1.5" href="#section-21.1.5">21.1.5</a> 183 Session Progress</span>

   The 183 (Session Progress) response is used to convey information
   about the progress of the call that is not otherwise classified.  The
   Reason-Phrase, header fields, or message body MAY be used to convey
   more details about the call progress.

<span class="h3"><a class="selflink" id="section-21.2" href="#section-21.2">21.2</a> Successful 2xx</span>

   The request was successful.

<span class="h4"><a class="selflink" id="section-21.2.1" href="#section-21.2.1">21.2.1</a> 200 OK</span>

   The request has succeeded.  The information returned with the
   response depends on the method used in the request.






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<span class="h3"><a class="selflink" id="section-21.3" href="#section-21.3">21.3</a> Redirection 3xx</span>

   3xx responses give information about the user's new location, or
   about alternative services that might be able to satisfy the call.

<span class="h4"><a class="selflink" id="section-21.3.1" href="#section-21.3.1">21.3.1</a> 300 Multiple Choices</span>

   The address in the request resolved to several choices, each with its
   own specific location, and the user (or UA) can select a preferred
   communication end point and redirect its request to that location.

   The response MAY include a message body containing a list of resource
   characteristics and location(s) from which the user or UA can choose
   the one most appropriate, if allowed by the Accept request header
   field.  However, no MIME types have been defined for this message
   body.

   The choices SHOULD also be listed as Contact fields (<a href="#section-20.10">Section 20.10</a>).
   Unlike HTTP, the SIP response MAY contain several Contact fields or a
   list of addresses in a Contact field.  UAs MAY use the Contact header
   field value for automatic redirection or MAY ask the user to confirm
   a choice.  However, this specification does not define any standard
   for such automatic selection.

      This status response is appropriate if the callee can be reached
      at several different locations and the server cannot or prefers
      not to proxy the request.

<span class="h4"><a class="selflink" id="section-21.3.2" href="#section-21.3.2">21.3.2</a> 301 Moved Permanently</span>

   The user can no longer be found at the address in the Request-URI,
   and the requesting client SHOULD retry at the new address given by
   the Contact header field (<a href="#section-20.10">Section 20.10</a>).  The requestor SHOULD
   update any local directories, address books, and user location caches
   with this new value and redirect future requests to the address(es)
   listed.

<span class="h4"><a class="selflink" id="section-21.3.3" href="#section-21.3.3">21.3.3</a> 302 Moved Temporarily</span>

   The requesting client SHOULD retry the request at the new address(es)
   given by the Contact header field (<a href="#section-20.10">Section 20.10</a>).  The Request-URI
   of the new request uses the value of the Contact header field in the
   response.








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   The duration of the validity of the Contact URI can be indicated
   through an Expires (<a href="#section-20.19">Section 20.19</a>) header field or an expires
   parameter in the Contact header field.  Both proxies and UAs MAY
   cache this URI for the duration of the expiration time.  If there is
   no explicit expiration time, the address is only valid once for
   recursing, and MUST NOT be cached for future transactions.

   If the URI cached from the Contact header field fails, the Request-
   URI from the redirected request MAY be tried again a single time.

      The temporary URI may have become out-of-date sooner than the
      expiration time, and a new temporary URI may be available.

<span class="h4"><a class="selflink" id="section-21.3.4" href="#section-21.3.4">21.3.4</a> 305 Use Proxy</span>

   The requested resource MUST be accessed through the proxy given by
   the Contact field.  The Contact field gives the URI of the proxy.
   The recipient is expected to repeat this single request via the
   proxy.  305 (Use Proxy) responses MUST only be generated by UASs.

<span class="h4"><a class="selflink" id="section-21.3.5" href="#section-21.3.5">21.3.5</a> 380 Alternative Service</span>

   The call was not successful, but alternative services are possible.

   The alternative services are described in the message body of the
   response.  Formats for such bodies are not defined here, and may be
   the subject of future standardization.

<span class="h3"><a class="selflink" id="section-21.4" href="#section-21.4">21.4</a> Request Failure 4xx</span>

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (for example, adding appropriate authorization).
   However, the same request to a different server might be successful.

<span class="h4"><a class="selflink" id="section-21.4.1" href="#section-21.4.1">21.4.1</a> 400 Bad Request</span>

   The request could not be understood due to malformed syntax.  The
   Reason-Phrase SHOULD identify the syntax problem in more detail, for
   example, "Missing Call-ID header field".

<span class="h4"><a class="selflink" id="section-21.4.2" href="#section-21.4.2">21.4.2</a> 401 Unauthorized</span>

   The request requires user authentication.  This response is issued by
   UASs and registrars, while 407 (Proxy Authentication Required) is
   used by proxy servers.





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<span class="h4"><a class="selflink" id="section-21.4.3" href="#section-21.4.3">21.4.3</a> 402 Payment Required</span>

   Reserved for future use.

<span class="h4"><a class="selflink" id="section-21.4.4" href="#section-21.4.4">21.4.4</a> 403 Forbidden</span>

   The server understood the request, but is refusing to fulfill it.
   Authorization will not help, and the request SHOULD NOT be repeated.

<span class="h4"><a class="selflink" id="section-21.4.5" href="#section-21.4.5">21.4.5</a> 404 Not Found</span>

   The server has definitive information that the user does not exist at
   the domain specified in the Request-URI.  This status is also
   returned if the domain in the Request-URI does not match any of the
   domains handled by the recipient of the request.

<span class="h4"><a class="selflink" id="section-21.4.6" href="#section-21.4.6">21.4.6</a> 405 Method Not Allowed</span>

   The method specified in the Request-Line is understood, but not
   allowed for the address identified by the Request-URI.

   The response MUST include an Allow header field containing a list of
   valid methods for the indicated address.

<span class="h4"><a class="selflink" id="section-21.4.7" href="#section-21.4.7">21.4.7</a> 406 Not Acceptable</span>

   The resource identified by the request is only capable of generating
   response entities that have content characteristics not acceptable
   according to the Accept header field sent in the request.

<span class="h4"><a class="selflink" id="section-21.4.8" href="#section-21.4.8">21.4.8</a> 407 Proxy Authentication Required</span>

   This code is similar to 401 (Unauthorized), but indicates that the
   client MUST first authenticate itself with the proxy.  SIP access
   authentication is explained in Sections <a href="#section-26">26</a> and <a href="#section-22.3">22.3</a>.

   This status code can be used for applications where access to the
   communication channel (for example, a telephony gateway) rather than
   the callee requires authentication.

<span class="h4"><a class="selflink" id="section-21.4.9" href="#section-21.4.9">21.4.9</a> 408 Request Timeout</span>

   The server could not produce a response within a suitable amount of
   time, for example, if it could not determine the location of the user
   in time.  The client MAY repeat the request without modifications at
   any later time.





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<span class="h4"><a class="selflink" id="section-21.4.10" href="#section-21.4.10">21.4.10</a> 410 Gone</span>

   The requested resource is no longer available at the server and no
   forwarding address is known.  This condition is expected to be
   considered permanent.  If the server does not know, or has no
   facility to determine, whether or not the condition is permanent, the
   status code 404 (Not Found) SHOULD be used instead.

<span class="h4"><a class="selflink" id="section-21.4.11" href="#section-21.4.11">21.4.11</a> 413 Request Entity Too Large</span>

   The server is refusing to process a request because the request
   entity-body is larger than the server is willing or able to process.
   The server MAY close the connection to prevent the client from
   continuing the request.

   If the condition is temporary, the server SHOULD include a Retry-
   After header field to indicate that it is temporary and after what
   time the client MAY try again.

<span class="h4"><a class="selflink" id="section-21.4.12" href="#section-21.4.12">21.4.12</a> 414 Request-URI Too Long</span>

   The server is refusing to service the request because the Request-URI
   is longer than the server is willing to interpret.

<span class="h4"><a class="selflink" id="section-21.4.13" href="#section-21.4.13">21.4.13</a> 415 Unsupported Media Type</span>

   The server is refusing to service the request because the message
   body of the request is in a format not supported by the server for
   the requested method.  The server MUST return a list of acceptable
   formats using the Accept, Accept-Encoding, or Accept-Language header
   field, depending on the specific problem with the content.  UAC
   processing of this response is described in <a href="#section-8.1.3.5">Section 8.1.3.5</a>.

<span class="h4"><a class="selflink" id="section-21.4.14" href="#section-21.4.14">21.4.14</a> 416 Unsupported URI Scheme</span>

   The server cannot process the request because the scheme of the URI
   in the Request-URI is unknown to the server.  Client processing of
   this response is described in <a href="#section-8.1.3.5">Section 8.1.3.5</a>.

<span class="h4"><a class="selflink" id="section-21.4.15" href="#section-21.4.15">21.4.15</a> 420 Bad Extension</span>

   The server did not understand the protocol extension specified in a
   Proxy-Require (<a href="#section-20.29">Section 20.29</a>) or Require (<a href="#section-20.32">Section 20.32</a>) header
   field.  The server MUST include a list of the unsupported extensions
   in an Unsupported header field in the response.  UAC processing of
   this response is described in <a href="#section-8.1.3.5">Section 8.1.3.5</a>.





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<span class="h4"><a class="selflink" id="section-21.4.16" href="#section-21.4.16">21.4.16</a> 421 Extension Required</span>

   The UAS needs a particular extension to process the request, but this
   extension is not listed in a Supported header field in the request.
   Responses with this status code MUST contain a Require header field
   listing the required extensions.

   A UAS SHOULD NOT use this response unless it truly cannot provide any
   useful service to the client.  Instead, if a desirable extension is
   not listed in the Supported header field, servers SHOULD process the
   request using baseline SIP capabilities and any extensions supported
   by the client.

<span class="h4"><a class="selflink" id="section-21.4.17" href="#section-21.4.17">21.4.17</a> 423 Interval Too Brief</span>

   The server is rejecting the request because the expiration time of
   the resource refreshed by the request is too short.  This response
   can be used by a registrar to reject a registration whose Contact
   header field expiration time was too small.  The use of this response
   and the related Min-Expires header field are described in Sections
   10.2.8, 10.3, and 20.23.

<span class="h4"><a class="selflink" id="section-21.4.18" href="#section-21.4.18">21.4.18</a> 480 Temporarily Unavailable</span>

   The callee's end system was contacted successfully but the callee is
   currently unavailable (for example, is not logged in, logged in but
   in a state that precludes communication with the callee, or has
   activated the "do not disturb" feature).  The response MAY indicate a
   better time to call in the Retry-After header field.  The user could
   also be available elsewhere (unbeknownst to this server).  The reason
   phrase SHOULD indicate a more precise cause as to why the callee is
   unavailable.  This value SHOULD be settable by the UA.  Status 486
   (Busy Here) MAY be used to more precisely indicate a particular
   reason for the call failure.

   This status is also returned by a redirect or proxy server that
   recognizes the user identified by the Request-URI, but does not
   currently have a valid forwarding location for that user.

<span class="h4"><a class="selflink" id="section-21.4.19" href="#section-21.4.19">21.4.19</a> 481 Call/Transaction Does Not Exist</span>

   This status indicates that the UAS received a request that does not
   match any existing dialog or transaction.

<span class="h4"><a class="selflink" id="section-21.4.20" href="#section-21.4.20">21.4.20</a> 482 Loop Detected</span>

   The server has detected a loop (<a href="#section-16.3">Section 16.3</a> Item 4).




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<span class="h4"><a class="selflink" id="section-21.4.21" href="#section-21.4.21">21.4.21</a> 483 Too Many Hops</span>

   The server received a request that contains a Max-Forwards (<a href="#section-20.22">Section</a>
   <a href="#section-20.22">20.22</a>) header field with the value zero.

<span class="h4"><a class="selflink" id="section-21.4.22" href="#section-21.4.22">21.4.22</a> 484 Address Incomplete</span>

   The server received a request with a Request-URI that was incomplete.
   Additional information SHOULD be provided in the reason phrase.

      This status code allows overlapped dialing.  With overlapped
      dialing, the client does not know the length of the dialing
      string.  It sends strings of increasing lengths, prompting the
      user for more input, until it no longer receives a 484 (Address
      Incomplete) status response.

<span class="h4"><a class="selflink" id="section-21.4.23" href="#section-21.4.23">21.4.23</a> 485 Ambiguous</span>

   The Request-URI was ambiguous.  The response MAY contain a listing of
   possible unambiguous addresses in Contact header fields.  Revealing
   alternatives can infringe on privacy of the user or the organization.
   It MUST be possible to configure a server to respond with status 404
   (Not Found) or to suppress the listing of possible choices for
   ambiguous Request-URIs.

   Example response to a request with the Request-URI
   sip:lee@example.com:

      SIP/2.0 485 Ambiguous
      Contact: Carol Lee &lt;sip:carol.lee@example.com&gt;
      Contact: Ping Lee &lt;sip:p.lee@example.com&gt;
      Contact: Lee M. Foote &lt;sips:lee.foote@example.com&gt;

      Some email and voice mail systems provide this functionality.  A
      status code separate from 3xx is used since the semantics are
      different: for 300, it is assumed that the same person or service
      will be reached by the choices provided.  While an automated
      choice or sequential search makes sense for a 3xx response, user
      intervention is required for a 485 (Ambiguous) response.

<span class="h4"><a class="selflink" id="section-21.4.24" href="#section-21.4.24">21.4.24</a> 486 Busy Here</span>

   The callee's end system was contacted successfully, but the callee is
   currently not willing or able to take additional calls at this end
   system.  The response MAY indicate a better time to call in the
   Retry-After header field.  The user could also be available





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   elsewhere, such as through a voice mail service.  Status 600 (Busy
   Everywhere) SHOULD be used if the client knows that no other end
   system will be able to accept this call.

<span class="h4"><a class="selflink" id="section-21.4.25" href="#section-21.4.25">21.4.25</a> 487 Request Terminated</span>

   The request was terminated by a BYE or CANCEL request.  This response
   is never returned for a CANCEL request itself.

<span class="h4"><a class="selflink" id="section-21.4.26" href="#section-21.4.26">21.4.26</a> 488 Not Acceptable Here</span>

   The response has the same meaning as 606 (Not Acceptable), but only
   applies to the specific resource addressed by the Request-URI and the
   request may succeed elsewhere.

   A message body containing a description of media capabilities MAY be
   present in the response, which is formatted according to the Accept
   header field in the INVITE (or application/sdp if not present), the
   same as a message body in a 200 (OK) response to an OPTIONS request.

<span class="h4"><a class="selflink" id="section-21.4.27" href="#section-21.4.27">21.4.27</a> 491 Request Pending</span>

   The request was received by a UAS that had a pending request within
   the same dialog.  <a href="#section-14.2">Section 14.2</a> describes how such "glare" situations
   are resolved.

<span class="h4"><a class="selflink" id="section-21.4.28" href="#section-21.4.28">21.4.28</a> 493 Undecipherable</span>

   The request was received by a UAS that contained an encrypted MIME
   body for which the recipient does not possess or will not provide an
   appropriate decryption key.  This response MAY have a single body
   containing an appropriate public key that should be used to encrypt
   MIME bodies sent to this UA.  Details of the usage of this response
   code can be found in <a href="#section-23.2">Section 23.2</a>.

<span class="h3"><a class="selflink" id="section-21.5" href="#section-21.5">21.5</a> Server Failure 5xx</span>

   5xx responses are failure responses given when a server itself has
   erred.

<span class="h4"><a class="selflink" id="section-21.5.1" href="#section-21.5.1">21.5.1</a> 500 Server Internal Error</span>

   The server encountered an unexpected condition that prevented it from
   fulfilling the request.  The client MAY display the specific error
   condition and MAY retry the request after several seconds.

   If the condition is temporary, the server MAY indicate when the
   client may retry the request using the Retry-After header field.



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<span class="h4"><a class="selflink" id="section-21.5.2" href="#section-21.5.2">21.5.2</a> 501 Not Implemented</span>

   The server does not support the functionality required to fulfill the
   request.  This is the appropriate response when a UAS does not
   recognize the request method and is not capable of supporting it for
   any user.  (Proxies forward all requests regardless of method.)

   Note that a 405 (Method Not Allowed) is sent when the server
   recognizes the request method, but that method is not allowed or
   supported.

<span class="h4"><a class="selflink" id="section-21.5.3" href="#section-21.5.3">21.5.3</a> 502 Bad Gateway</span>

   The server, while acting as a gateway or proxy, received an invalid
   response from the downstream server it accessed in attempting to
   fulfill the request.

<span class="h4"><a class="selflink" id="section-21.5.4" href="#section-21.5.4">21.5.4</a> 503 Service Unavailable</span>

   The server is temporarily unable to process the request due to a
   temporary overloading or maintenance of the server.  The server MAY
   indicate when the client should retry the request in a Retry-After
   header field.  If no Retry-After is given, the client MUST act as if
   it had received a 500 (Server Internal Error) response.

   A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
   attempt to forward the request to an alternate server.  It SHOULD NOT
   forward any other requests to that server for the duration specified
   in the Retry-After header field, if present.

   Servers MAY refuse the connection or drop the request instead of
   responding with 503 (Service Unavailable).

<span class="h4"><a class="selflink" id="section-21.5.5" href="#section-21.5.5">21.5.5</a> 504 Server Time-out</span>

   The server did not receive a timely response from an external server
   it accessed in attempting to process the request.  408 (Request
   Timeout) should be used instead if there was no response within the
   period specified in the Expires header field from the upstream
   server.

<span class="h4"><a class="selflink" id="section-21.5.6" href="#section-21.5.6">21.5.6</a> 505 Version Not Supported</span>

   The server does not support, or refuses to support, the SIP protocol
   version that was used in the request.  The server is indicating that
   it is unable or unwilling to complete the request using the same
   major version as the client, other than with this error message.




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<span class="h4"><a class="selflink" id="section-21.5.7" href="#section-21.5.7">21.5.7</a> 513 Message Too Large</span>

   The server was unable to process the request since the message length
   exceeded its capabilities.

<span class="h3"><a class="selflink" id="section-21.6" href="#section-21.6">21.6</a> Global Failures 6xx</span>

   6xx responses indicate that a server has definitive information about
   a particular user, not just the particular instance indicated in the
   Request-URI.

<span class="h4"><a class="selflink" id="section-21.6.1" href="#section-21.6.1">21.6.1</a> 600 Busy Everywhere</span>

   The callee's end system was contacted successfully but the callee is
   busy and does not wish to take the call at this time.  The response
   MAY indicate a better time to call in the Retry-After header field.
   If the callee does not wish to reveal the reason for declining the
   call, the callee uses status code 603 (Decline) instead.  This status
   response is returned only if the client knows that no other end point
   (such as a voice mail system) will answer the request.  Otherwise,
   486 (Busy Here) should be returned.

<span class="h4"><a class="selflink" id="section-21.6.2" href="#section-21.6.2">21.6.2</a> 603 Decline</span>

   The callee's machine was successfully contacted but the user
   explicitly does not wish to or cannot participate.  The response MAY
   indicate a better time to call in the Retry-After header field.  This
   status response is returned only if the client knows that no other
   end point will answer the request.

<span class="h4"><a class="selflink" id="section-21.6.3" href="#section-21.6.3">21.6.3</a> 604 Does Not Exist Anywhere</span>

   The server has authoritative information that the user indicated in
   the Request-URI does not exist anywhere.

<span class="h4"><a class="selflink" id="section-21.6.4" href="#section-21.6.4">21.6.4</a> 606 Not Acceptable</span>

   The user's agent was contacted successfully but some aspects of the
   session description such as the requested media, bandwidth, or
   addressing style were not acceptable.

   A 606 (Not Acceptable) response means that the user wishes to
   communicate, but cannot adequately support the session described.
   The 606 (Not Acceptable) response MAY contain a list of reasons in a
   Warning header field describing why the session described cannot be
   supported.  Warning reason codes are listed in <a href="#section-20.43">Section 20.43</a>.





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   A message body containing a description of media capabilities MAY be
   present in the response, which is formatted according to the Accept
   header field in the INVITE (or application/sdp if not present), the
   same as a message body in a 200 (OK) response to an OPTIONS request.

   It is hoped that negotiation will not frequently be needed, and when
   a new user is being invited to join an already existing conference,
   negotiation may not be possible.  It is up to the invitation
   initiator to decide whether or not to act on a 606 (Not Acceptable)
   response.

   This status response is returned only if the client knows that no
   other end point will answer the request.

<span class="h2"><a class="selflink" id="section-22" href="#section-22">22</a> Usage of HTTP Authentication</span>

   SIP provides a stateless, challenge-based mechanism for
   authentication that is based on authentication in HTTP.  Any time
   that a proxy server or UA receives a request (with the exceptions
   given in <a href="#section-22.1">Section 22.1</a>), it MAY challenge the initiator of the request
   to provide assurance of its identity.  Once the originator has been
   identified, the recipient of the request SHOULD ascertain whether or
   not this user is authorized to make the request in question.  No
   authorization systems are recommended or discussed in this document.

   The "Digest" authentication mechanism described in this section
   provides message authentication and replay protection only, without
   message integrity or confidentiality.  Protective measures above and
   beyond those provided by Digest need to be taken to prevent active
   attackers from modifying SIP requests and responses.

   Note that due to its weak security, the usage of "Basic"
   authentication has been deprecated.  Servers MUST NOT accept
   credentials using the "Basic" authorization scheme, and servers also
   MUST NOT challenge with "Basic".  This is a change from <a href="./rfc2543">RFC 2543</a>.

<span class="h3"><a class="selflink" id="section-22.1" href="#section-22.1">22.1</a> Framework</span>

   The framework for SIP authentication closely parallels that of HTTP
   (<a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>]).  In particular, the BNF for auth-scheme, auth-param,
   challenge, realm, realm-value, and credentials is identical (although
   the usage of "Basic" as a scheme is not permitted).  In SIP, a UAS
   uses the 401 (Unauthorized) response to challenge the identity of a
   UAC.  Additionally, registrars and redirect servers MAY make use of
   401 (Unauthorized) responses for authentication, but proxies MUST
   NOT, and instead MAY use the 407 (Proxy Authentication Required)





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   response.  The requirements for inclusion of the Proxy-Authenticate,
   Proxy-Authorization, WWW-Authenticate, and Authorization in the
   various messages are identical to those described in <a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>].

   Since SIP does not have the concept of a canonical root URL, the
   notion of protection spaces is interpreted differently in SIP.  The
   realm string alone defines the protection domain.  This is a change
   from <a href="./rfc2543">RFC 2543</a>, in which the Request-URI and the realm together
   defined the protection domain.

      This previous definition of protection domain caused some amount
      of confusion since the Request-URI sent by the UAC and the
      Request-URI received by the challenging server might be different,
      and indeed the final form of the Request-URI might not be known to
      the UAC.  Also, the previous definition depended on the presence
      of a SIP URI in the Request-URI and seemed to rule out alternative
      URI schemes (for example, the tel URL).

   Operators of user agents or proxy servers that will authenticate
   received requests MUST adhere to the following guidelines for
   creation of a realm string for their server:

      o  Realm strings MUST be globally unique.  It is RECOMMENDED that
         a realm string contain a hostname or domain name, following the
         recommendation in <a href="./rfc2617#section-3.2.1">Section&nbsp;3.2.1 of RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>].

      o  Realm strings SHOULD present a human-readable identifier that
         can be rendered to a user.

   For example:

      INVITE sip:bob@biloxi.com SIP/2.0
      Authorization: Digest realm="biloxi.com", &lt;...&gt;

   Generally, SIP authentication is meaningful for a specific realm, a
   protection domain.  Thus, for Digest authentication, each such
   protection domain has its own set of usernames and passwords.  If a
   server does not require authentication for a particular request, it
   MAY accept a default username, "anonymous", which has no password
   (password of "").  Similarly, UACs representing many users, such as
   PSTN gateways, MAY have their own device-specific username and
   password, rather than accounts for particular users, for their realm.

   While a server can legitimately challenge most SIP requests, there
   are two requests defined by this document that require special
   handling for authentication: ACK and CANCEL.





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   Under an authentication scheme that uses responses to carry values
   used to compute nonces (such as Digest), some problems come up for
   any requests that take no response, including ACK.  For this reason,
   any credentials in the INVITE that were accepted by a server MUST be
   accepted by that server for the ACK.  UACs creating an ACK message
   will duplicate all of the Authorization and Proxy-Authorization
   header field values that appeared in the INVITE to which the ACK
   corresponds.  Servers MUST NOT attempt to challenge an ACK.

   Although the CANCEL method does take a response (a 2xx), servers MUST
   NOT attempt to challenge CANCEL requests since these requests cannot
   be resubmitted.  Generally, a CANCEL request SHOULD be accepted by a
   server if it comes from the same hop that sent the request being
   canceled (provided that some sort of transport or network layer
   security association, as described in <a href="#section-26.2.1">Section 26.2.1</a>, is in place).

   When a UAC receives a challenge, it SHOULD render to the user the
   contents of the "realm" parameter in the challenge (which appears in
   either a WWW-Authenticate header field or Proxy-Authenticate header
   field) if the UAC device does not already know of a credential for
   the realm in question.  A service provider that pre-configures UAs
   with credentials for its realm should be aware that users will not
   have the opportunity to present their own credentials for this realm
   when challenged at a pre-configured device.

   Finally, note that even if a UAC can locate credentials that are
   associated with the proper realm, the potential exists that these
   credentials may no longer be valid or that the challenging server
   will not accept these credentials for whatever reason (especially
   when "anonymous" with no password is submitted).  In this instance a
   server may repeat its challenge, or it may respond with a 403
   Forbidden.  A UAC MUST NOT re-attempt requests with the credentials
   that have just been rejected (though the request may be retried if
   the nonce was stale).

<span class="h3"><a class="selflink" id="section-22.2" href="#section-22.2">22.2</a> User-to-User Authentication</span>

   When a UAS receives a request from a UAC, the UAS MAY authenticate
   the originator before the request is processed.  If no credentials
   (in the Authorization header field) are provided in the request, the
   UAS can challenge the originator to provide credentials by rejecting
   the request with a 401 (Unauthorized) status code.

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages.  The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the realm.




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   An example of the WWW-Authenticate header field in a 401 challenge
   is:

      WWW-Authenticate: Digest
              realm="biloxi.com",
              qop="auth,auth-int",
              nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
              opaque="5ccc069c403ebaf9f0171e9517f40e41"

   When the originating UAC receives the 401 (Unauthorized), it SHOULD,
   if it is able, re-originate the request with the proper credentials.
   The UAC may require input from the originating user before
   proceeding.  Once authentication credentials have been supplied
   (either directly by the user, or discovered in an internal keyring),
   UAs SHOULD cache the credentials for a given value of the To header
   field and "realm" and attempt to re-use these values on the next
   request for that destination.  UAs MAY cache credentials in any way
   they would like.

   If no credentials for a realm can be located, UACs MAY attempt to
   retry the request with a username of "anonymous" and no password (a
   password of "").

   Once credentials have been located, any UA that wishes to
   authenticate itself with a UAS or registrar -- usually, but not
   necessarily, after receiving a 401 (Unauthorized) response -- MAY do
   so by including an Authorization header field with the request.  The
   Authorization field value consists of credentials containing the
   authentication information of the UA for the realm of the resource
   being requested as well as parameters required in support of
   authentication and replay protection.

   An example of the Authorization header field is:

      Authorization: Digest username="bob",
              realm="biloxi.com",
              nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
              uri="sip:bob@biloxi.com",
              qop=auth,
              nc=00000001,
              cnonce="0a4f113b",
              response="6629fae49393a05397450978507c4ef1",
              opaque="5ccc069c403ebaf9f0171e9517f40e41"

   When a UAC resubmits a request with its credentials after receiving a
   401 (Unauthorized) or 407 (Proxy Authentication Required) response,
   it MUST increment the CSeq header field value as it would normally
   when sending an updated request.



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<span class="h3"><a class="selflink" id="section-22.3" href="#section-22.3">22.3</a> Proxy-to-User Authentication</span>

   Similarly, when a UAC sends a request to a proxy server, the proxy
   server MAY authenticate the originator before the request is
   processed.  If no credentials (in the Proxy-Authorization header
   field) are provided in the request, the proxy can challenge the
   originator to provide credentials by rejecting the request with a 407
   (Proxy Authentication Required) status code.  The proxy MUST populate
   the 407 (Proxy Authentication Required) message with a Proxy-
   Authenticate header field value applicable to the proxy for the
   requested resource.

   The use of Proxy-Authenticate and Proxy-Authorization parallel that
   described in [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>], with one difference.  Proxies MUST NOT add values
   to the Proxy-Authorization header field.  All 407 (Proxy
   Authentication Required) responses MUST be forwarded upstream toward
   the UAC following the procedures for any other response.  It is the
   UAC's responsibility to add the Proxy-Authorization header field
   value containing credentials for the realm of the proxy that has
   asked for authentication.

      If a proxy were to resubmit a request adding a Proxy-Authorization
      header field value, it would need to increment the CSeq in the new
      request.  However, this would cause the UAC that submitted the
      original request to discard a response from the UAS, as the CSeq
      value would be different.

   When the originating UAC receives the 407 (Proxy Authentication
   Required) it SHOULD, if it is able, re-originate the request with the
   proper credentials.  It should follow the same procedures for the
   display of the "realm" parameter that are given above for responding
   to 401.

   If no credentials for a realm can be located, UACs MAY attempt to
   retry the request with a username of "anonymous" and no password (a
   password of "").

   The UAC SHOULD also cache the credentials used in the re-originated
   request.

   The following rule is RECOMMENDED for proxy credential caching:

   If a UA receives a Proxy-Authenticate header field value in a 401/407
   response to a request with a particular Call-ID, it should
   incorporate credentials for that realm in all subsequent requests
   that contain the same Call-ID.  These credentials MUST NOT be cached
   across dialogs; however, if a UA is configured with the realm of its
   local outbound proxy, when one exists, then the UA MAY cache



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   credentials for that realm across dialogs.  Note that this does mean
   a future request in a dialog could contain credentials that are not
   needed by any proxy along the Route header path.

   Any UA that wishes to authenticate itself to a proxy server --
   usually, but not necessarily, after receiving a 407 (Proxy
   Authentication Required) response -- MAY do so by including a Proxy-
   Authorization header field value with the request.  The Proxy-
   Authorization request-header field allows the client to identify
   itself (or its user) to a proxy that requires authentication.  The
   Proxy-Authorization header field value consists of credentials
   containing the authentication information of the UA for the proxy
   and/or realm of the resource being requested.

   A Proxy-Authorization header field value applies only to the proxy
   whose realm is identified in the "realm" parameter (this proxy may
   previously have demanded authentication using the Proxy-Authenticate
   field).  When multiple proxies are used in a chain, a Proxy-
   Authorization header field value MUST NOT be consumed by any proxy
   whose realm does not match the "realm" parameter specified in that
   value.

   Note that if an authentication scheme that does not support realms is
   used in the Proxy-Authorization header field, a proxy server MUST
   attempt to parse all Proxy-Authorization header field values to
   determine whether one of them has what the proxy server considers to
   be valid credentials.  Because this is potentially very time-
   consuming in large networks, proxy servers SHOULD use an
   authentication scheme that supports realms in the Proxy-Authorization
   header field.

   If a request is forked (as described in <a href="#section-16.7">Section 16.7</a>), various proxy
   servers and/or UAs may wish to challenge the UAC.  In this case, the
   forking proxy server is responsible for aggregating these challenges
   into a single response.  Each WWW-Authenticate and Proxy-Authenticate
   value received in responses to the forked request MUST be placed into
   the single response that is sent by the forking proxy to the UA; the
   ordering of these header field values is not significant.

      When a proxy server issues a challenge in response to a request,
      it will not proxy the request until the UAC has retried the
      request with valid credentials.  A forking proxy may forward a
      request simultaneously to multiple proxy servers that require
      authentication, each of which in turn will not forward the request
      until the originating UAC has authenticated itself in their
      respective realm.  If the UAC does not provide credentials for





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      each challenge, the proxy servers that issued the challenges will
      not forward requests to the UA where the destination user might be
      located, and therefore, the virtues of forking are largely lost.

   When resubmitting its request in response to a 401 (Unauthorized) or
   407 (Proxy Authentication Required) that contains multiple
   challenges, a UAC MAY include an Authorization value for each WWW-
   Authenticate value and a Proxy-Authorization value for each Proxy-
   Authenticate value for which the UAC wishes to supply a credential.
   As noted above, multiple credentials in a request SHOULD be
   differentiated by the "realm" parameter.

   It is possible for multiple challenges associated with the same realm
   to appear in the same 401 (Unauthorized) or 407 (Proxy Authentication
   Required).  This can occur, for example, when multiple proxies within
   the same administrative domain, which use a common realm, are reached
   by a forking request.  When it retries a request, a UAC MAY therefore
   supply multiple credentials in Authorization or Proxy-Authorization
   header fields with the same "realm" parameter value.  The same
   credentials SHOULD be used for the same realm.

<span class="h3"><a class="selflink" id="section-22.4" href="#section-22.4">22.4</a> The Digest Authentication Scheme</span>

   This section describes the modifications and clarifications required
   to apply the HTTP Digest authentication scheme to SIP.  The SIP
   scheme usage is almost completely identical to that for HTTP [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>].

   Since <a href="./rfc2543">RFC 2543</a> is based on HTTP Digest as defined in <a href="./rfc2069">RFC 2069</a> [<a href="#ref-39" title="&quot;An Extension to HTTP: Digest Access Authentication&quot;">39</a>],
   SIP servers supporting <a href="./rfc2617">RFC 2617</a> MUST ensure they are backwards
   compatible with <a href="./rfc2069">RFC 2069</a>.  Procedures for this backwards
   compatibility are specified in <a href="./rfc2617">RFC 2617</a>.  Note, however, that SIP
   servers MUST NOT accept or request Basic authentication.

   The rules for Digest authentication follow those defined in [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>],
   with "HTTP/1.1" replaced by "SIP/2.0" in addition to the following
   differences:

      1.  The URI included in the challenge has the following BNF:

          URI  =  SIP-URI / SIPS-URI

      2.  The BNF in <a href="./rfc2617">RFC 2617</a> has an error in that the 'uri' parameter
          of the Authorization header field for HTTP Digest








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          authentication is not enclosed in quotation marks.  (The
          example in <a href="./rfc2617#section-3.5">Section&nbsp;3.5 of RFC 2617</a> is correct.)  For SIP, the
          'uri' MUST be enclosed in quotation marks.

      3.  The BNF for digest-uri-value is:

          digest-uri-value  =  Request-URI ; as defined in <a href="#section-25">Section 25</a>

      4.  The example procedure for choosing a nonce based on Etag does
          not work for SIP.

      5.  The text in <a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>] regarding cache operation does not
          apply to SIP.

      6.  <a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>] requires that a server check that the URI in the
          request line and the URI included in the Authorization header
          field point to the same resource.  In a SIP context, these two
          URIs may refer to different users, due to forwarding at some
          proxy.  Therefore, in SIP, a server MAY check that the
          Request-URI in the Authorization header field value
          corresponds to a user for whom the server is willing to accept
          forwarded or direct requests, but it is not necessarily a
          failure if the two fields are not equivalent.

      7.  As a clarification to the calculation of the A2 value for
          message integrity assurance in the Digest authentication
          scheme, implementers should assume, when the entity-body is
          empty (that is, when SIP messages have no body) that the hash
          of the entity-body resolves to the MD5 hash of an empty
          string, or:

             H(entity-body) = MD5("") =
          "d41d8cd98f00b204e9800998ecf8427e"

      8.  <a href="./rfc2617">RFC 2617</a> notes that a cnonce value MUST NOT be sent in an
          Authorization (and by extension Proxy-Authorization) header
          field if no qop directive has been sent.  Therefore, any
          algorithms that have a dependency on the cnonce (including
          "MD5-Sess") require that the qop directive be sent.  Use of
          the "qop" parameter is optional in <a href="./rfc2617">RFC 2617</a> for the purposes
          of backwards compatibility with <a href="./rfc2069">RFC 2069</a>; since <a href="./rfc2543">RFC 2543</a> was
          based on <a href="./rfc2069">RFC 2069</a>, the "qop" parameter must unfortunately
          remain optional for clients and servers to receive.  However,
          servers MUST always send a "qop" parameter in WWW-Authenticate
          and Proxy-Authenticate header field values.  If a client
          receives a "qop" parameter in a challenge header field, it
          MUST send the "qop" parameter in any resulting authorization
          header field.



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   <a href="./rfc2543">RFC 2543</a> did not allow usage of the Authentication-Info header field
   (it effectively used <a href="./rfc2069">RFC 2069</a>).  However, we now allow usage of this
   header field, since it provides integrity checks over the bodies and
   provides mutual authentication.  <a href="./rfc2617">RFC 2617</a> [<a href="#ref-17" title="&quot;HTTP authentication: Basic and Digest Access Authentication&quot;">17</a>] defines mechanisms for
   backwards compatibility using the qop attribute in the request.
   These mechanisms MUST be used by a server to determine if the client
   supports the new mechanisms in <a href="./rfc2617">RFC 2617</a> that were not specified in
   <a href="./rfc2069">RFC 2069</a>.

<span class="h2"><a class="selflink" id="section-23" href="#section-23">23</a> S/MIME</span>

   SIP messages carry MIME bodies and the MIME standard includes
   mechanisms for securing MIME contents to ensure both integrity and
   confidentiality (including the 'multipart/signed' and
   'application/pkcs7-mime' MIME types, see <a href="./rfc1847">RFC 1847</a> [<a href="#ref-22" title="&quot;Security Multiparts for MIME: Multipart/Signed and Multipart/Encrypted&quot;">22</a>], <a href="./rfc2630">RFC 2630</a> [<a href="#ref-23" title="&quot;Cryptographic Message Syntax&quot;">23</a>]
   and <a href="./rfc2633">RFC 2633</a> [<a href="#ref-24" title="&quot;S/MIME Version 3 Message Specification&quot;">24</a>]).  Implementers should note, however, that there
   may be rare network intermediaries (not typical proxy servers) that
   rely on viewing or modifying the bodies of SIP messages (especially
   SDP), and that secure MIME may prevent these sorts of intermediaries
   from functioning.

      This applies particularly to certain types of firewalls.

      The PGP mechanism for encrypting the header fields and bodies of
      SIP messages described in <a href="./rfc2543">RFC 2543</a> has been deprecated.

<span class="h3"><a class="selflink" id="section-23.1" href="#section-23.1">23.1</a> S/MIME Certificates</span>

   The certificates that are used to identify an end-user for the
   purposes of S/MIME differ from those used by servers in one important
   respect - rather than asserting that the identity of the holder
   corresponds to a particular hostname, these certificates assert that
   the holder is identified by an end-user address.  This address is
   composed of the concatenation of the "userinfo" "@" and "domainname"
   portions of a SIP or SIPS URI (in other words, an email address of
   the form "bob@biloxi.com"), most commonly corresponding to a user's
   address-of-record.

   These certificates are also associated with keys that are used to
   sign or encrypt bodies of SIP messages.  Bodies are signed with the
   private key of the sender (who may include their public key with the
   message as appropriate), but bodies are encrypted with the public key
   of the intended recipient.  Obviously, senders must have
   foreknowledge of the public key of recipients in order to encrypt
   message bodies.  Public keys can be stored within a UA on a virtual
   keyring.





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   Each user agent that supports S/MIME MUST contain a keyring
   specifically for end-users' certificates.  This keyring should map
   between addresses of record and corresponding certificates.  Over
   time, users SHOULD use the same certificate when they populate the
   originating URI of signaling (the From header field) with the same
   address-of-record.

   Any mechanisms depending on the existence of end-user certificates
   are seriously limited in that there is virtually no consolidated
   authority today that provides certificates for end-user applications.
   However, users SHOULD acquire certificates from known public
   certificate authorities.  As an alternative, users MAY create self-
   signed certificates.  The implications of self-signed certificates
   are explored further in <a href="#section-26.4.2">Section 26.4.2</a>.  Implementations may also use
   pre-configured certificates in deployments in which a previous trust
   relationship exists between all SIP entities.

   Above and beyond the problem of acquiring an end-user certificate,
   there are few well-known centralized directories that distribute
   end-user certificates.  However, the holder of a certificate SHOULD
   publish their certificate in any public directories as appropriate.
   Similarly, UACs SHOULD support a mechanism for importing (manually or
   automatically) certificates discovered in public directories
   corresponding to the target URIs of SIP requests.

<span class="h3"><a class="selflink" id="section-23.2" href="#section-23.2">23.2</a> S/MIME Key Exchange</span>

   SIP itself can also be used as a means to distribute public keys in
   the following manner.

   Whenever the CMS SignedData message is used in S/MIME for SIP, it
   MUST contain the certificate bearing the public key necessary to
   verify the signature.

   When a UAC sends a request containing an S/MIME body that initiates a
   dialog, or sends a non-INVITE request outside the context of a
   dialog, the UAC SHOULD structure the body as an S/MIME
   'multipart/signed' CMS SignedData body.  If the desired CMS service
   is EnvelopedData (and the public key of the target user is known),
   the UAC SHOULD send the EnvelopedData message encapsulated within a
   SignedData message.

   When a UAS receives a request containing an S/MIME CMS body that
   includes a certificate, the UAS SHOULD first validate the
   certificate, if possible, with any available root certificates for
   certificate authorities.  The UAS SHOULD also determine the subject
   of the certificate (for S/MIME, the SubjectAltName will contain the
   appropriate identity) and compare this value to the From header field



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   of the request.  If the certificate cannot be verified, because it is
   self-signed, or signed by no known authority, or if it is verifiable
   but its subject does not correspond to the From header field of
   request, the UAS MUST notify its user of the status of the
   certificate (including the subject of the certificate, its signer,
   and any key fingerprint information) and request explicit permission
   before proceeding.  If the certificate was successfully verified and
   the subject of the certificate corresponds to the From header field
   of the SIP request, or if the user (after notification) explicitly
   authorizes the use of the certificate, the UAS SHOULD add this
   certificate to a local keyring, indexed by the address-of-record of
   the holder of the certificate.

   When a UAS sends a response containing an S/MIME body that answers
   the first request in a dialog, or a response to a non-INVITE request
   outside the context of a dialog, the UAS SHOULD structure the body as
   an S/MIME 'multipart/signed' CMS SignedData body.  If the desired CMS
   service is EnvelopedData, the UAS SHOULD send the EnvelopedData
   message encapsulated within a SignedData message.

   When a UAC receives a response containing an S/MIME CMS body that
   includes a certificate, the UAC SHOULD first validate the
   certificate, if possible, with any appropriate root certificate.  The
   UAC SHOULD also determine the subject of the certificate and compare
   this value to the To field of the response; although the two may very
   well be different, and this is not necessarily indicative of a
   security breach.  If the certificate cannot be verified because it is
   self-signed, or signed by no known authority, the UAC MUST notify its
   user of the status of the certificate (including the subject of the
   certificate, its signator, and any key fingerprint information) and
   request explicit permission before proceeding.  If the certificate
   was successfully verified, and the subject of the certificate
   corresponds to the To header field in the response, or if the user
   (after notification) explicitly authorizes the use of the
   certificate, the UAC SHOULD add this certificate to a local keyring,
   indexed by the address-of-record of the holder of the certificate.
   If the UAC had not transmitted its own certificate to the UAS in any
   previous transaction, it SHOULD use a CMS SignedData body for its
   next request or response.

   On future occasions, when the UA receives requests or responses that
   contain a From header field corresponding to a value in its keyring,
   the UA SHOULD compare the certificate offered in these messages with
   the existing certificate in its keyring.  If there is a discrepancy,
   the UA MUST notify its user of a change of the certificate
   (preferably in terms that indicate that this is a potential security
   breach) and acquire the user's permission before continuing to




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   process the signaling.  If the user authorizes this certificate, it
   SHOULD be added to the keyring alongside any previous value(s) for
   this address-of-record.

   Note well however, that this key exchange mechanism does not
   guarantee the secure exchange of keys when self-signed certificates,
   or certificates signed by an obscure authority, are used - it is
   vulnerable to well-known attacks.  In the opinion of the authors,
   however, the security it provides is proverbially better than
   nothing; it is in fact comparable to the widely used SSH application.
   These limitations are explored in greater detail in <a href="#section-26.4.2">Section 26.4.2</a>.

   If a UA receives an S/MIME body that has been encrypted with a public
   key unknown to the recipient, it MUST reject the request with a 493
   (Undecipherable) response.  This response SHOULD contain a valid
   certificate for the respondent (corresponding, if possible, to any
   address of record given in the To header field of the rejected
   request) within a MIME body with a 'certs-only' "smime-type"
   parameter.

   A 493 (Undecipherable) sent without any certificate indicates that
   the respondent cannot or will not utilize S/MIME encrypted messages,
   though they may still support S/MIME signatures.

   Note that a user agent that receives a request containing an S/MIME
   body that is not optional (with a Content-Disposition header
   "handling" parameter of "required") MUST reject the request with a
   415 Unsupported Media Type response if the MIME type is not
   understood.  A user agent that receives such a response when S/MIME
   is sent SHOULD notify its user that the remote device does not
   support S/MIME, and it MAY subsequently resend the request without
   S/MIME, if appropriate; however, this 415 response may constitute a
   downgrade attack.

   If a user agent sends an S/MIME body in a request, but receives a
   response that contains a MIME body that is not secured, the UAC
   SHOULD notify its user that the session could not be secured.
   However, if a user agent that supports S/MIME receives a request with
   an unsecured body, it SHOULD NOT respond with a secured body, but if
   it expects S/MIME from the sender (for example, because the sender's
   From header field value corresponds to an identity on its keychain),
   the UAS SHOULD notify its user that the session could not be secured.

   A number of conditions that arise in the previous text call for the
   notification of the user when an anomalous certificate-management
   event occurs.  Users might well ask what they should do under these
   circumstances.  First and foremost, an unexpected change in a
   certificate, or an absence of security when security is expected, are



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   causes for caution but not necessarily indications that an attack is
   in progress.  Users might abort any connection attempt or refuse a
   connection request they have received; in telephony parlance, they
   could hang up and call back.  Users may wish to find an alternate
   means to contact the other party and confirm that their key has
   legitimately changed.  Note that users are sometimes compelled to
   change their certificates, for example when they suspect that the
   secrecy of their private key has been compromised.  When their
   private key is no longer private, users must legitimately generate a
   new key and re-establish trust with any users that held their old
   key.

   Finally, if during the course of a dialog a UA receives a certificate
   in a CMS SignedData message that does not correspond with the
   certificates previously exchanged during a dialog, the UA MUST notify
   its user of the change, preferably in terms that indicate that this
   is a potential security breach.

<span class="h3"><a class="selflink" id="section-23.3" href="#section-23.3">23.3</a> Securing MIME bodies</span>

   There are two types of secure MIME bodies that are of interest to
   SIP: use of these bodies should follow the S/MIME specification [<a href="#ref-24" title="&quot;S/MIME Version 3 Message Specification&quot;">24</a>]
   with a few variations.

      o  "multipart/signed" MUST be used only with CMS detached
         signatures.

            This allows backwards compatibility with non-S/MIME-
            compliant recipients.

      o  S/MIME bodies SHOULD have a Content-Disposition header field,
         and the value of the "handling" parameter SHOULD be "required."

      o  If a UAC has no certificate on its keyring associated with the
         address-of-record to which it wants to send a request, it
         cannot send an encrypted "application/pkcs7-mime" MIME message.
         UACs MAY send an initial request such as an OPTIONS message
         with a CMS detached signature in order to solicit the
         certificate of the remote side (the signature SHOULD be over a
         "message/sip" body of the type described in <a href="#section-23.4">Section 23.4</a>).

            Note that future standardization work on S/MIME may define
            non-certificate based keys.

      o  Senders of S/MIME bodies SHOULD use the "SMIMECapabilities"
         (see Section 2.5.2 of [<a href="#ref-24" title="&quot;S/MIME Version 3 Message Specification&quot;">24</a>]) attribute to express their
         capabilities and preferences for further communications.  Note
         especially that senders MAY use the "preferSignedData"



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         capability to encourage receivers to respond with CMS
         SignedData messages (for example, when sending an OPTIONS
         request as described above).

      o  S/MIME implementations MUST at a minimum support SHA1 as a
         digital signature algorithm, and 3DES as an encryption
         algorithm.  All other signature and encryption algorithms MAY
         be supported.  Implementations can negotiate support for these
         algorithms with the "SMIMECapabilities" attribute.

      o  Each S/MIME body in a SIP message SHOULD be signed with only
         one certificate.  If a UA receives a message with multiple
         signatures, the outermost signature should be treated as the
         single certificate for this body.  Parallel signatures SHOULD
         NOT be used.

         The following is an example of an encrypted S/MIME SDP body
         within a SIP message:

        INVITE sip:bob@biloxi.com SIP/2.0
        Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
        To: Bob &lt;sip:bob@biloxi.com&gt;
        From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
        Call-ID: a84b4c76e66710
        CSeq: 314159 INVITE
        Max-Forwards: 70
        Contact: &lt;sip:alice@pc33.atlanta.com&gt;
        Content-Type: application/pkcs7-mime; smime-type=enveloped-data;
             name=smime.p7m
        Content-Disposition: attachment; filename=smime.p7m
           handling=required

      *******************************************************
      * Content-Type: application/sdp                       *
      *                                                     *
      * v=0                                                 *
      * o=alice 53655765 2353687637 IN IP4 pc33.atlanta.com *
      * s=-                                                 *
      * t=0 0                                               *
      * c=IN IP4 pc33.atlanta.com                           *
      * m=audio 3456 RTP/AVP 0 1 3 99                       *
      * a=rtpmap:0 PCMU/8000                                *
      *******************************************************








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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-207" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h3"><a class="selflink" id="section-23.4" href="#section-23.4">23.4</a> SIP Header Privacy and Integrity using S/MIME: Tunneling SIP</span>

   As a means of providing some degree of end-to-end authentication,
   integrity or confidentiality for SIP header fields, S/MIME can
   encapsulate entire SIP messages within MIME bodies of type
   "message/sip" and then apply MIME security to these bodies in the
   same manner as typical SIP bodies.  These encapsulated SIP requests
   and responses do not constitute a separate dialog or transaction,
   they are a copy of the "outer" message that is used to verify
   integrity or to supply additional information.

   If a UAS receives a request that contains a tunneled "message/sip"
   S/MIME body, it SHOULD include a tunneled "message/sip" body in the
   response with the same smime-type.

   Any traditional MIME bodies (such as SDP) SHOULD be attached to the
   "inner" message so that they can also benefit from S/MIME security.
   Note that "message/sip" bodies can be sent as a part of a MIME
   "multipart/mixed" body if any unsecured MIME types should also be
   transmitted in a request.

<span class="h4"><a class="selflink" id="section-23.4.1" href="#section-23.4.1">23.4.1</a> Integrity and Confidentiality Properties of SIP Headers</span>

   When the S/MIME integrity or confidentiality mechanisms are used,
   there may be discrepancies between the values in the "inner" message
   and values in the "outer" message.  The rules for handling any such
   differences for all of the header fields described in this document
   are given in this section.

   Note that for the purposes of loose timestamping, all SIP messages
   that tunnel "message/sip" SHOULD contain a Date header in both the
   "inner" and "outer" headers.

<span class="h5"><a class="selflink" id="section-23.4.1.1" href="#section-23.4.1.1">23.4.1.1</a> Integrity</span>

   Whenever integrity checks are performed, the integrity of a header
   field should be determined by matching the value of the header field
   in the signed body with that in the "outer" messages using the
   comparison rules of SIP as described in 20.

   Header fields that can be legitimately modified by proxy servers are:
   Request-URI, Via, Record-Route, Route, Max-Forwards, and Proxy-
   Authorization.  If these header fields are not intact end-to-end,
   implementations SHOULD NOT consider this a breach of security.
   Changes to any other header fields defined in this document
   constitute an integrity violation; users MUST be notified of a
   discrepancy.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 207]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-208" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h5"><a class="selflink" id="section-23.4.1.2" href="#section-23.4.1.2">23.4.1.2</a> Confidentiality</span>

   When messages are encrypted, header fields may be included in the
   encrypted body that are not present in the "outer" message.

   Some header fields must always have a plaintext version because they
   are required header fields in requests and responses - these include:

   To, From, Call-ID, CSeq, Contact.  While it is probably not useful to
   provide an encrypted alternative for the Call-ID, CSeq, or Contact,
   providing an alternative to the information in the "outer" To or From
   is permitted.  Note that the values in an encrypted body are not used
   for the purposes of identifying transactions or dialogs - they are
   merely informational.  If the From header field in an encrypted body
   differs from the value in the "outer" message, the value within the
   encrypted body SHOULD be displayed to the user, but MUST NOT be used
   in the "outer" header fields of any future messages.

   Primarily, a user agent will want to encrypt header fields that have
   an end-to-end semantic, including: Subject, Reply-To, Organization,
   Accept, Accept-Encoding, Accept-Language, Alert-Info, Error-Info,
   Authentication-Info, Expires, In-Reply-To, Require, Supported,
   Unsupported, Retry-After, User-Agent, Server, and Warning.  If any of
   these header fields are present in an encrypted body, they should be
   used instead of any "outer" header fields, whether this entails
   displaying the header field values to users or setting internal
   states in the UA.  They SHOULD NOT however be used in the "outer"
   headers of any future messages.

   If present, the Date header field MUST always be the same in the
   "inner" and "outer" headers.

   Since MIME bodies are attached to the "inner" message,
   implementations will usually encrypt MIME-specific header fields,
   including: MIME-Version, Content-Type, Content-Length, Content-
   Language, Content-Encoding and Content-Disposition.  The "outer"
   message will have the proper MIME header fields for S/MIME bodies.
   These header fields (and any MIME bodies they preface) should be
   treated as normal MIME header fields and bodies received in a SIP
   message.

   It is not particularly useful to encrypt the following header fields:
   Min-Expires, Timestamp, Authorization, Priority, and WWW-
   Authenticate.  This category also includes those header fields that
   can be changed by proxy servers (described in the preceding section).
   UAs SHOULD never include these in an "inner" message if they are not





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 208]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-209" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   included in the "outer" message.  UAs that receive any of these
   header fields in an encrypted body SHOULD ignore the encrypted
   values.

   Note that extensions to SIP may define additional header fields; the
   authors of these extensions should describe the integrity and
   confidentiality properties of such header fields.  If a SIP UA
   encounters an unknown header field with an integrity violation, it
   MUST ignore the header field.

<span class="h4"><a class="selflink" id="section-23.4.2" href="#section-23.4.2">23.4.2</a> Tunneling Integrity and Authentication</span>

   Tunneling SIP messages within S/MIME bodies can provide integrity for
   SIP header fields if the header fields that the sender wishes to
   secure are replicated in a "message/sip" MIME body signed with a CMS
   detached signature.

   Provided that the "message/sip" body contains at least the
   fundamental dialog identifiers (To, From, Call-ID, CSeq), then a
   signed MIME body can provide limited authentication.  At the very
   least, if the certificate used to sign the body is unknown to the
   recipient and cannot be verified, the signature can be used to
   ascertain that a later request in a dialog was transmitted by the
   same certificate-holder that initiated the dialog.  If the recipient
   of the signed MIME body has some stronger incentive to trust the
   certificate (they were able to validate it, they acquired it from a
   trusted repository, or they have used it frequently) then the
   signature can be taken as a stronger assertion of the identity of the
   subject of the certificate.

   In order to eliminate possible confusions about the addition or
   subtraction of entire header fields, senders SHOULD replicate all
   header fields from the request within the signed body.  Any message
   bodies that require integrity protection MUST be attached to the
   "inner" message.

   If a Date header is present in a message with a signed body, the
   recipient SHOULD compare the header field value with its own internal
   clock, if applicable.  If a significant time discrepancy is detected
   (on the order of an hour or more), the user agent SHOULD alert the
   user to the anomaly, and note that it is a potential security breach.

   If an integrity violation in a message is detected by its recipient,
   the message MAY be rejected with a 403 (Forbidden) response if it is
   a request, or any existing dialog MAY be terminated.  UAs SHOULD
   notify users of this circumstance and request explicit guidance on
   how to proceed.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 209]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-210" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The following is an example of the use of a tunneled "message/sip"
   body:

      INVITE sip:bob@biloxi.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
      To: Bob &lt;sip:bob@biloxi.com&gt;
      From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 314159 INVITE
      Max-Forwards: 70
      Date: Thu, 21 Feb 2002 13:02:03 GMT
      Contact: &lt;sip:alice@pc33.atlanta.com&gt;
      Content-Type: multipart/signed;
        protocol="application/pkcs7-signature";
        micalg=sha1; boundary=boundary42
      Content-Length: 568

      --boundary42
      Content-Type: message/sip

      INVITE sip:bob@biloxi.com SIP/2.0
      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
      To: Bob &lt;bob@biloxi.com&gt;
      From: Alice &lt;alice@atlanta.com&gt;;tag=1928301774
      Call-ID: a84b4c76e66710
      CSeq: 314159 INVITE
      Max-Forwards: 70
      Date: Thu, 21 Feb 2002 13:02:03 GMT
      Contact: &lt;sip:alice@pc33.atlanta.com&gt;
      Content-Type: application/sdp
      Content-Length: 147

      v=0
      o=UserA 2890844526 2890844526 IN IP4 here.com
      s=Session SDP
      c=IN IP4 pc33.atlanta.com
      t=0 0
      m=audio 49172 RTP/AVP 0
      a=rtpmap:0 PCMU/8000

      --boundary42
      Content-Type: application/pkcs7-signature; name=smime.p7s
      Content-Transfer-Encoding: base64
      Content-Disposition: attachment; filename=smime.p7s;
         handling=required






<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 210]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-211" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6
      4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj
      n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4
      7GhIGfHfYT64VQbnj756

      --boundary42-

<span class="h4"><a class="selflink" id="section-23.4.3" href="#section-23.4.3">23.4.3</a> Tunneling Encryption</span>

   It may also be desirable to use this mechanism to encrypt a
   "message/sip" MIME body within a CMS EnvelopedData message S/MIME
   body, but in practice, most header fields are of at least some use to
   the network; the general use of encryption with S/MIME is to secure
   message bodies like SDP rather than message headers.  Some
   informational header fields, such as the Subject or Organization
   could perhaps warrant end-to-end security.  Headers defined by future
   SIP applications might also require obfuscation.

   Another possible application of encrypting header fields is selective
   anonymity.  A request could be constructed with a From header field
   that contains no personal information (for example,
   sip:anonymous@anonymizer.invalid).  However, a second From header
   field containing the genuine address-of-record of the originator
   could be encrypted within a "message/sip" MIME body where it will
   only be visible to the endpoints of a dialog.

      Note that if this mechanism is used for anonymity, the From header
      field will no longer be usable by the recipient of a message as an
      index to their certificate keychain for retrieving the proper
      S/MIME key to associated with the sender.  The message must first
      be decrypted, and the "inner" From header field MUST be used as an
      index.

   In order to provide end-to-end integrity, encrypted "message/sip"
   MIME bodies SHOULD be signed by the sender.  This creates a
   "multipart/signed" MIME body that contains an encrypted body and a
   signature, both of type "application/pkcs7-mime".














<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 211]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-212" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   In the following example, of an encrypted and signed message, the
   text boxed in asterisks ("*") is encrypted:

        INVITE sip:bob@biloxi.com SIP/2.0
        Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
        To: Bob &lt;sip:bob@biloxi.com&gt;
        From: Anonymous &lt;sip:anonymous@atlanta.com&gt;;tag=1928301774
        Call-ID: a84b4c76e66710
        CSeq: 314159 INVITE
        Max-Forwards: 70
        Date: Thu, 21 Feb 2002 13:02:03 GMT
        Contact: &lt;sip:pc33.atlanta.com&gt;
        Content-Type: multipart/signed;
          protocol="application/pkcs7-signature";
          micalg=sha1; boundary=boundary42
        Content-Length: 568

        --boundary42
        Content-Type: application/pkcs7-mime; smime-type=enveloped-data;
             name=smime.p7m
        Content-Transfer-Encoding: base64
        Content-Disposition: attachment; filename=smime.p7m
           handling=required
        Content-Length: 231

      ***********************************************************
      * Content-Type: message/sip                               *
      *                                                         *
      * INVITE sip:bob@biloxi.com SIP/2.0                       *
      * Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8 *
      * To: Bob &lt;bob@biloxi.com&gt;                                *
      * From: Alice &lt;alice@atlanta.com&gt;;tag=1928301774          *
      * Call-ID: a84b4c76e66710                                 *
      * CSeq: 314159 INVITE                                     *
      * Max-Forwards: 70                                        *
      * Date: Thu, 21 Feb 2002 13:02:03 GMT                     *
      * Contact: &lt;sip:alice@pc33.atlanta.com&gt;                   *
      *                                                         *
      * Content-Type: application/sdp                           *
      *                                                         *
      * v=0                                                     *
      * o=alice 53655765 2353687637 IN IP4 pc33.atlanta.com     *
      * s=Session SDP                                           *
      * t=0 0                                                   *
      * c=IN IP4 pc33.atlanta.com                               *
      * m=audio 3456 RTP/AVP 0 1 3 99                           *
      * a=rtpmap:0 PCMU/8000                                    *
      ***********************************************************



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 212]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-213" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


        --boundary42
        Content-Type: application/pkcs7-signature; name=smime.p7s
        Content-Transfer-Encoding: base64
        Content-Disposition: attachment; filename=smime.p7s;
           handling=required

        ghyHhHUujhJhjH77n8HHGTrfvbnj756tbB9HG4VQpfyF467GhIGfHfYT6
        4VQpfyF467GhIGfHfYT6jH77n8HHGghyHhHUujhJh756tbB9HGTrfvbnj
        n8HHGTrfvhJhjH776tbB9HG4VQbnj7567GhIGfHfYT6ghyHhHUujpfyF4
        7GhIGfHfYT64VQbnj756

        --boundary42-

<span class="h2"><a class="selflink" id="section-24" href="#section-24">24</a> Examples</span>

   In the following examples, we often omit the message body and the
   corresponding Content-Length and Content-Type header fields for
   brevity.

<span class="h3"><a class="selflink" id="section-24.1" href="#section-24.1">24.1</a> Registration</span>

   Bob registers on start-up.  The message flow is shown in Figure 9.
   Note that the authentication usually required for registration is not
   shown for simplicity.

                  biloxi.com         Bob's
                   registrar       softphone
                      |                |
                      |   REGISTER F1  |
                      |&lt;---------------|
                      |    200 OK F2   |
                      |---------------&gt;|

                  Figure 9: SIP Registration Example

   F1 REGISTER Bob -&gt; Registrar

       REGISTER sip:registrar.biloxi.com SIP/2.0
       Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
       Max-Forwards: 70
       To: Bob &lt;sip:bob@biloxi.com&gt;
       From: Bob &lt;sip:bob@biloxi.com&gt;;tag=456248
       Call-ID: 843817637684230@998sdasdh09
       CSeq: 1826 REGISTER
       Contact: &lt;sip:bob@192.0.2.4&gt;
       Expires: 7200
       Content-Length: 0




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 213]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-214" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The registration expires after two hours.  The registrar responds
   with a 200 OK:

   F2 200 OK Registrar -&gt; Bob

        SIP/2.0 200 OK
        Via: SIP/2.0/UDP bobspc.biloxi.com:5060;branch=z9hG4bKnashds7
         ;received=192.0.2.4
        To: Bob &lt;sip:bob@biloxi.com&gt;;tag=2493k59kd
        From: Bob &lt;sip:bob@biloxi.com&gt;;tag=456248
        Call-ID: 843817637684230@998sdasdh09
        CSeq: 1826 REGISTER
        Contact: &lt;sip:bob@192.0.2.4&gt;
        Expires: 7200
        Content-Length: 0

<span class="h3"><a class="selflink" id="section-24.2" href="#section-24.2">24.2</a> Session Setup</span>

   This example contains the full details of the example session setup
   in <a href="#section-4">Section 4</a>.  The message flow is shown in Figure 1.  Note that
   these flows show the minimum required set of header fields - some
   other header fields such as Allow and Supported would normally be
   present.

F1 INVITE Alice -&gt; atlanta.com proxy

INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
Max-Forwards: 70
To: Bob &lt;sip:bob@biloxi.com&gt;
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:alice@pc33.atlanta.com&gt;
Content-Type: application/sdp
Content-Length: 142

(Alice's SDP not shown)













<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 214]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-215" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


F2 100 Trying atlanta.com proxy -&gt; Alice

SIP/2.0 100 Trying
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Content-Length: 0

F3 INVITE atlanta.com proxy -&gt; biloxi.com proxy

INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
Max-Forwards: 69
To: Bob &lt;sip:bob@biloxi.com&gt;
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:alice@pc33.atlanta.com&gt;
Content-Type: application/sdp
Content-Length: 142

(Alice's SDP not shown)

F4 100 Trying biloxi.com proxy -&gt; atlanta.com proxy

SIP/2.0 100 Trying
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Content-Length: 0











<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 215]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-216" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


F5 INVITE biloxi.com proxy -&gt; Bob

INVITE sip:bob@192.0.2.4 SIP/2.0
Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
Max-Forwards: 68
To: Bob &lt;sip:bob@biloxi.com&gt;
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:alice@pc33.atlanta.com&gt;
Content-Type: application/sdp
Content-Length: 142

(Alice's SDP not shown)

F6 180 Ringing Bob -&gt; biloxi.com proxy

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1
 ;received=192.0.2.3
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
Contact: &lt;sip:bob@192.0.2.4&gt;
CSeq: 314159 INVITE
Content-Length: 0

F7 180 Ringing biloxi.com proxy -&gt; atlanta.com proxy

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
Contact: &lt;sip:bob@192.0.2.4&gt;
CSeq: 314159 INVITE
Content-Length: 0



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 216]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-217" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


F8 180 Ringing atlanta.com proxy -&gt; Alice

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
Contact: &lt;sip:bob@192.0.2.4&gt;
CSeq: 314159 INVITE
Content-Length: 0

F9 200 OK Bob -&gt; biloxi.com proxy

SIP/2.0 200 OK
Via: SIP/2.0/UDP server10.biloxi.com;branch=z9hG4bK4b43c2ff8.1
 ;received=192.0.2.3
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:bob@192.0.2.4&gt;
Content-Type: application/sdp
Content-Length: 131

(Bob's SDP not shown)

F10 200 OK biloxi.com proxy -&gt; atlanta.com proxy

SIP/2.0 200 OK
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com;branch=z9hG4bK77ef4c2312983.1
 ;received=192.0.2.2
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:bob@192.0.2.4&gt;
Content-Type: application/sdp
Content-Length: 131

(Bob's SDP not shown)




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 217]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-218" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


F11 200 OK atlanta.com proxy -&gt; Alice

SIP/2.0 200 OK
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds8
 ;received=192.0.2.1
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 INVITE
Contact: &lt;sip:bob@192.0.2.4&gt;
Content-Type: application/sdp
Content-Length: 131

(Bob's SDP not shown)

F12 ACK Alice -&gt; Bob

ACK sip:bob@192.0.2.4 SIP/2.0
Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bKnashds9
Max-Forwards: 70
To: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
From: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 314159 ACK
Content-Length: 0

   The media session between Alice and Bob is now established.

   Bob hangs up first.  Note that Bob's SIP phone maintains its own CSeq
   numbering space, which, in this example, begins with 231.  Since Bob
   is making the request, the To and From URIs and tags have been
   swapped.

F13 BYE Bob -&gt; Alice

BYE sip:alice@pc33.atlanta.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10
Max-Forwards: 70
From: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
To: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 231 BYE
Content-Length: 0








<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 218]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-219" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


F14 200 OK Alice -&gt; Bob

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.0.2.4;branch=z9hG4bKnashds10
From: Bob &lt;sip:bob@biloxi.com&gt;;tag=a6c85cf
To: Alice &lt;sip:alice@atlanta.com&gt;;tag=1928301774
Call-ID: a84b4c76e66710
CSeq: 231 BYE
Content-Length: 0

   The SIP Call Flows document [<a href="#ref-40" title="&quot;SIP Call Flow Examples&quot;">40</a>] contains further examples of SIP
   messages.

<span class="h2"><a class="selflink" id="section-25" href="#section-25">25</a>  Augmented BNF for the SIP Protocol</span>

   All of the mechanisms specified in this document are described in
   both prose and an augmented Backus-Naur Form (BNF) defined in <a href="./rfc2234">RFC</a>
   <a href="./rfc2234">2234</a> [<a href="#ref-10" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">10</a>].  <a href="./rfc2234#section-6.1">Section&nbsp;6.1 of RFC 2234</a> defines a set of core rules that
   are used by this specification, and not repeated here.  Implementers
   need to be familiar with the notation and content of <a href="./rfc2234">RFC 2234</a> in
   order to understand this specification.  Certain basic rules are in
   uppercase, such as SP, LWS, HTAB, CRLF, DIGIT, ALPHA, etc.  Angle
   brackets are used within definitions to clarify the use of rule
   names.

   The use of square brackets is redundant syntactically.  It is used as
   a semantic hint that the specific parameter is optional to use.

<span class="h3"><a class="selflink" id="section-25.1" href="#section-25.1">25.1</a> Basic Rules</span>

   The following rules are used throughout this specification to
   describe basic parsing constructs.  The US-ASCII coded character set
   is defined by ANSI X3.4-1986.

      alphanum  =  ALPHA / DIGIT
















<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 219]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-220" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Several rules are incorporated from <a href="./rfc2396">RFC 2396</a> [<a href="#ref-5" title="&quot;Uniform Resource Identifiers (URI): Generic Syntax&quot;">5</a>] but are updated to
   make them compliant with <a href="./rfc2234">RFC 2234</a> [<a href="#ref-10" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">10</a>].  These include:

      reserved    =  ";" / "/" / "?" / ":" / "@" / "&amp;" / "=" / "+"
                     / "$" / ","
      unreserved  =  alphanum / mark
      mark        =  "-" / "_" / "." / "!" / "~" / "*" / "'"
                     / "(" / ")"
      escaped     =  "%" HEXDIG HEXDIG

   SIP header field values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab.  All linear
   white space, including folding, has the same semantics as SP.  A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.
   This is intended to behave exactly as HTTP/1.1 as described in <a href="./rfc2616">RFC</a>
   <a href="./rfc2616">2616</a> [<a href="#ref-8" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">8</a>].  The SWS construct is used when linear white space is
   optional, generally between tokens and separators.

      LWS  =  [*WSP CRLF] 1*WSP ; linear whitespace
      SWS  =  [LWS] ; sep whitespace

   To separate the header name from the rest of value, a colon is used,
   which, by the above rule, allows whitespace before, but no line
   break, and whitespace after, including a linebreak.  The HCOLON
   defines this construct.

      HCOLON  =  *( SP / HTAB ) ":" SWS

   The TEXT-UTF8 rule is only used for descriptive field contents and
   values that are not intended to be interpreted by the message parser.
   Words of *TEXT-UTF8 contain characters from the UTF-8 charset (<a href="./rfc2279">RFC</a>
   <a href="./rfc2279">2279</a> [<a href="#ref-7" title="&quot;UTF-8, a transformation format of ISO 10646&quot;">7</a>]).  The TEXT-UTF8-TRIM rule is used for descriptive field
   contents that are n t quoted strings, where leading and trailing LWS
   is not meaningful.  In this regard, SIP differs from HTTP, which uses
   the ISO 8859-1 character set.

      TEXT-UTF8-TRIM  =  1*TEXT-UTF8char *(*LWS TEXT-UTF8char)
      TEXT-UTF8char   =  %x21-7E / UTF8-NONASCII
      UTF8-NONASCII   =  %xC0-DF 1UTF8-CONT
                      /  %xE0-EF 2UTF8-CONT
                      /  %xF0-F7 3UTF8-CONT
                      /  %xF8-Fb 4UTF8-CONT
                      /  %xFC-FD 5UTF8-CONT
      UTF8-CONT       =  %x80-BF






<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 220]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-221" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   A CRLF is allowed in the definition of TEXT-UTF8-TRIM only as part of
   a header field continuation.  It is expected that the folding LWS
   will be replaced with a single SP before interpretation of the TEXT-
   UTF8-TRIM value.

   Hexadecimal numeric characters are used in several protocol elements.
   Some elements (authentication) force hex alphas to be lower case.

      LHEX  =  DIGIT / %x61-66 ;lowercase a-f

   Many SIP header field values consist of words separated by LWS or
   special characters.  Unless otherwise stated, tokens are case-
   insensitive.  These special characters MUST be in a quoted string to
   be used within a parameter value.  The word construct is used in
   Call-ID to allow most separators to be used.

      token       =  1*(alphanum / "-" / "." / "!" / "%" / "*"
                     / "_" / "+" / "`" / "'" / "~" )
      separators  =  "(" / ")" / "&lt;" / "&gt;" / "@" /
                     "," / ";" / ":" / "\" / DQUOTE /
                     "/" / "[" / "]" / "?" / "=" /
                     "{" / "}" / SP / HTAB
      word        =  1*(alphanum / "-" / "." / "!" / "%" / "*" /
                     "_" / "+" / "`" / "'" / "~" /
                     "(" / ")" / "&lt;" / "&gt;" /
                     ":" / "\" / DQUOTE /
                     "/" / "[" / "]" / "?" /
                     "{" / "}" )

   When tokens are used or separators are used between elements,
   whitespace is often allowed before or after these characters:

      STAR    =  SWS "*" SWS ; asterisk
      SLASH   =  SWS "/" SWS ; slash
      EQUAL   =  SWS "=" SWS ; equal
      LPAREN  =  SWS "(" SWS ; left parenthesis
      RPAREN  =  SWS ")" SWS ; right parenthesis
      RAQUOT  =  "&gt;" SWS ; right angle quote
      LAQUOT  =  SWS "&lt;"; left angle quote
      COMMA   =  SWS "," SWS ; comma
      SEMI    =  SWS ";" SWS ; semicolon
      COLON   =  SWS ":" SWS ; colon
      LDQUOT  =  SWS DQUOTE; open double quotation mark
      RDQUOT  =  DQUOTE SWS ; close double quotation mark







<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 221]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-222" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Comments can be included in some SIP header fields by surrounding the
   comment text with parentheses.  Comments are only allowed in fields
   containing "comment" as part of their field value definition.  In all
   other fields, parentheses are considered part of the field value.

      comment  =  LPAREN *(ctext / quoted-pair / comment) RPAREN
      ctext    =  %x21-27 / %x2A-5B / %x5D-7E / UTF8-NONASCII
                  / LWS

   ctext includes all chars except left and right parens and backslash.
   A string of text is parsed as a single word if it is quoted using
   double-quote marks.  In quoted strings, quotation marks (") and
   backslashes (\) need to be escaped.

      quoted-string  =  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE
      qdtext         =  LWS / %x21 / %x23-5B / %x5D-7E
                        / UTF8-NONASCII

   The backslash character ("\") MAY be used as a single-character
   quoting mechanism only within quoted-string and comment constructs.
   Unlike HTTP/1.1, the characters CR and LF cannot be escaped by this
   mechanism to avoid conflict with line folding and header separation.

quoted-pair  =  "\" (%x00-09 / %x0B-0C
                / %x0E-7F)

SIP-URI          =  "sip:" [ userinfo ] hostport
                    uri-parameters [ headers ]
SIPS-URI         =  "sips:" [ userinfo ] hostport
                    uri-parameters [ headers ]
userinfo         =  ( user / telephone-subscriber ) [ ":" password ] "@"
user             =  1*( unreserved / escaped / user-unreserved )
user-unreserved  =  "&amp;" / "=" / "+" / "$" / "," / ";" / "?" / "/"
password         =  *( unreserved / escaped /
                    "&amp;" / "=" / "+" / "$" / "," )
hostport         =  host [ ":" port ]
host             =  hostname / IPv4address / IPv6reference
hostname         =  *( domainlabel "." ) toplabel [ "." ]
domainlabel      =  alphanum
                    / alphanum *( alphanum / "-" ) alphanum
toplabel         =  ALPHA / ALPHA *( alphanum / "-" ) alphanum










<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 222]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-223" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


IPv4address    =  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6reference  =  "[" IPv6address "]"
IPv6address    =  hexpart [ ":" IPv4address ]
hexpart        =  hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq ]
hexseq         =  hex4 *( ":" hex4)
hex4           =  1*4HEXDIG
port           =  1*DIGIT

   The BNF for telephone-subscriber can be found in <a href="./rfc2806">RFC 2806</a> [<a href="#ref-9" title="&quot;URLs for Telephone Calls&quot;">9</a>].  Note,
   however, that any characters allowed there that are not allowed in
   the user part of the SIP URI MUST be escaped.

uri-parameters    =  *( ";" uri-parameter)
uri-parameter     =  transport-param / user-param / method-param
                     / ttl-param / maddr-param / lr-param / other-param
transport-param   =  "transport="
                     ( "udp" / "tcp" / "sctp" / "tls"
                     / other-transport)
other-transport   =  token
user-param        =  "user=" ( "phone" / "ip" / other-user)
other-user        =  token
method-param      =  "method=" Method
ttl-param         =  "ttl=" ttl
maddr-param       =  "maddr=" host
lr-param          =  "lr"
other-param       =  pname [ "=" pvalue ]
pname             =  1*paramchar
pvalue            =  1*paramchar
paramchar         =  param-unreserved / unreserved / escaped
param-unreserved  =  "[" / "]" / "/" / ":" / "&amp;" / "+" / "$"

headers         =  "?" header *( "&amp;" header )
header          =  hname "=" hvalue
hname           =  1*( hnv-unreserved / unreserved / escaped )
hvalue          =  *( hnv-unreserved / unreserved / escaped )
hnv-unreserved  =  "[" / "]" / "/" / "?" / ":" / "+" / "$"

SIP-message    =  Request / Response
Request        =  Request-Line
                  *( message-header )
                  CRLF
                  [ message-body ]
Request-Line   =  Method SP Request-URI SP SIP-Version CRLF
Request-URI    =  SIP-URI / SIPS-URI / absoluteURI
absoluteURI    =  scheme ":" ( hier-part / opaque-part )
hier-part      =  ( net-path / abs-path ) [ "?" query ]
net-path       =  "//" authority [ abs-path ]
abs-path       =  "/" path-segments



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 223]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-224" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


opaque-part    =  uric-no-slash *uric
uric           =  reserved / unreserved / escaped
uric-no-slash  =  unreserved / escaped / ";" / "?" / ":" / "@"
                  / "&amp;" / "=" / "+" / "$" / ","
path-segments  =  segment *( "/" segment )
segment        =  *pchar *( ";" param )
param          =  *pchar
pchar          =  unreserved / escaped /
                  ":" / "@" / "&amp;" / "=" / "+" / "$" / ","
scheme         =  ALPHA *( ALPHA / DIGIT / "+" / "-" / "." )
authority      =  srvr / reg-name
srvr           =  [ [ userinfo "@" ] hostport ]
reg-name       =  1*( unreserved / escaped / "$" / ","
                  / ";" / ":" / "@" / "&amp;" / "=" / "+" )
query          =  *uric
SIP-Version    =  "SIP" "/" 1*DIGIT "." 1*DIGIT

message-header  =  (Accept
                /  Accept-Encoding
                /  Accept-Language
                /  Alert-Info
                /  Allow
                /  Authentication-Info
                /  Authorization
                /  Call-ID
                /  Call-Info
                /  Contact
                /  Content-Disposition
                /  Content-Encoding
                /  Content-Language
                /  Content-Length
                /  Content-Type
                /  CSeq
                /  Date
                /  Error-Info
                /  Expires
                /  From
                /  In-Reply-To
                /  Max-Forwards
                /  MIME-Version
                /  Min-Expires
                /  Organization
                /  Priority
                /  Proxy-Authenticate
                /  Proxy-Authorization
                /  Proxy-Require
                /  Record-Route
                /  Reply-To



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 224]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-225" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


                /  Require
                /  Retry-After
                /  Route
                /  Server
                /  Subject
                /  Supported
                /  Timestamp
                /  To
                /  Unsupported
                /  User-Agent
                /  Via
                /  Warning
                /  WWW-Authenticate
                /  extension-header) CRLF

INVITEm           =  %x49.4E.56.49.54.45 ; INVITE in caps
ACKm              =  %x41.43.4B ; ACK in caps
OPTIONSm          =  %x4F.50.54.49.4F.4E.53 ; OPTIONS in caps
BYEm              =  %x42.59.45 ; BYE in caps
CANCELm           =  %x43.41.4E.43.45.4C ; CANCEL in caps
REGISTERm         =  %x52.45.47.49.53.54.45.52 ; REGISTER in caps
Method            =  INVITEm / ACKm / OPTIONSm / BYEm
                     / CANCELm / REGISTERm
                     / extension-method
extension-method  =  token
Response          =  Status-Line
                     *( message-header )
                     CRLF
                     [ message-body ]

Status-Line     =  SIP-Version SP Status-Code SP Reason-Phrase CRLF
Status-Code     =  Informational
               /   Redirection
               /   Success
               /   Client-Error
               /   Server-Error
               /   Global-Failure
               /   extension-code
extension-code  =  3DIGIT
Reason-Phrase   =  *(reserved / unreserved / escaped
                   / UTF8-NONASCII / UTF8-CONT / SP / HTAB)

Informational  =  "100"  ;  Trying
              /   "180"  ;  Ringing
              /   "181"  ;  Call Is Being Forwarded
              /   "182"  ;  Queued
              /   "183"  ;  Session Progress




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 225]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-226" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Success  =  "200"  ;  OK

Redirection  =  "300"  ;  Multiple Choices
            /   "301"  ;  Moved Permanently
            /   "302"  ;  Moved Temporarily
            /   "305"  ;  Use Proxy
            /   "380"  ;  Alternative Service

Client-Error  =  "400"  ;  Bad Request
             /   "401"  ;  Unauthorized
             /   "402"  ;  Payment Required
             /   "403"  ;  Forbidden
             /   "404"  ;  Not Found
             /   "405"  ;  Method Not Allowed
             /   "406"  ;  Not Acceptable
             /   "407"  ;  Proxy Authentication Required
             /   "408"  ;  Request Timeout
             /   "410"  ;  Gone
             /   "413"  ;  Request Entity Too Large
             /   "414"  ;  Request-URI Too Large
             /   "415"  ;  Unsupported Media Type
             /   "416"  ;  Unsupported URI Scheme
             /   "420"  ;  Bad Extension
             /   "421"  ;  Extension Required
             /   "423"  ;  Interval Too Brief
             /   "480"  ;  Temporarily not available
             /   "481"  ;  Call Leg/Transaction Does Not Exist
             /   "482"  ;  Loop Detected
             /   "483"  ;  Too Many Hops
             /   "484"  ;  Address Incomplete
             /   "485"  ;  Ambiguous
             /   "486"  ;  Busy Here
             /   "487"  ;  Request Terminated
             /   "488"  ;  Not Acceptable Here
             /   "491"  ;  Request Pending
             /   "493"  ;  Undecipherable

Server-Error  =  "500"  ;  Internal Server Error
             /   "501"  ;  Not Implemented
             /   "502"  ;  Bad Gateway
             /   "503"  ;  Service Unavailable
             /   "504"  ;  Server Time-out
             /   "505"  ;  SIP Version not supported
             /   "513"  ;  Message Too Large







<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 226]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-227" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Global-Failure  =  "600"  ;  Busy Everywhere
               /   "603"  ;  Decline
               /   "604"  ;  Does not exist anywhere
               /   "606"  ;  Not Acceptable

Accept         =  "Accept" HCOLON
                   [ accept-range *(COMMA accept-range) ]
accept-range   =  media-range *(SEMI accept-param)
media-range    =  ( "*/*"
                  / ( m-type SLASH "*" )
                  / ( m-type SLASH m-subtype )
                  ) *( SEMI m-parameter )
accept-param   =  ("q" EQUAL qvalue) / generic-param
qvalue         =  ( "0" [ "." 0*3DIGIT ] )
                  / ( "1" [ "." 0*3("0") ] )
generic-param  =  token [ EQUAL gen-value ]
gen-value      =  token / host / quoted-string

Accept-Encoding  =  "Accept-Encoding" HCOLON
                     [ encoding *(COMMA encoding) ]
encoding         =  codings *(SEMI accept-param)
codings          =  content-coding / "*"
content-coding   =  token

Accept-Language  =  "Accept-Language" HCOLON
                     [ language *(COMMA language) ]
language         =  language-range *(SEMI accept-param)
language-range   =  ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" )

Alert-Info   =  "Alert-Info" HCOLON alert-param *(COMMA alert-param)
alert-param  =  LAQUOT absoluteURI RAQUOT *( SEMI generic-param )

Allow  =  "Allow" HCOLON [Method *(COMMA Method)]

Authorization     =  "Authorization" HCOLON credentials
credentials       =  ("Digest" LWS digest-response)
                     / other-response
digest-response   =  dig-resp *(COMMA dig-resp)
dig-resp          =  username / realm / nonce / digest-uri
                      / dresponse / algorithm / cnonce
                      / opaque / message-qop
                      / nonce-count / auth-param
username          =  "username" EQUAL username-value
username-value    =  quoted-string
digest-uri        =  "uri" EQUAL LDQUOT digest-uri-value RDQUOT
digest-uri-value  =  rquest-uri ; Equal to request-uri as specified
                     by HTTP/1.1
message-qop       =  "qop" EQUAL qop-value



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 227]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-228" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


cnonce            =  "cnonce" EQUAL cnonce-value
cnonce-value      =  nonce-value
nonce-count       =  "nc" EQUAL nc-value
nc-value          =  8LHEX
dresponse         =  "response" EQUAL request-digest
request-digest    =  LDQUOT 32LHEX RDQUOT
auth-param        =  auth-param-name EQUAL
                     ( token / quoted-string )
auth-param-name   =  token
other-response    =  auth-scheme LWS auth-param
                     *(COMMA auth-param)
auth-scheme       =  token

Authentication-Info  =  "Authentication-Info" HCOLON ainfo
                        *(COMMA ainfo)
ainfo                =  nextnonce / message-qop
                         / response-auth / cnonce
                         / nonce-count
nextnonce            =  "nextnonce" EQUAL nonce-value
response-auth        =  "rspauth" EQUAL response-digest
response-digest      =  LDQUOT *LHEX RDQUOT

Call-ID  =  ( "Call-ID" / "i" ) HCOLON callid
callid   =  word [ "@" word ]

Call-Info   =  "Call-Info" HCOLON info *(COMMA info)
info        =  LAQUOT absoluteURI RAQUOT *( SEMI info-param)
info-param  =  ( "purpose" EQUAL ( "icon" / "info"
               / "card" / token ) ) / generic-param

Contact        =  ("Contact" / "m" ) HCOLON
                  ( STAR / (contact-param *(COMMA contact-param)))
contact-param  =  (name-addr / addr-spec) *(SEMI contact-params)
name-addr      =  [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec      =  SIP-URI / SIPS-URI / absoluteURI
display-name   =  *(token LWS)/ quoted-string

contact-params     =  c-p-q / c-p-expires
                      / contact-extension
c-p-q              =  "q" EQUAL qvalue
c-p-expires        =  "expires" EQUAL delta-seconds
contact-extension  =  generic-param
delta-seconds      =  1*DIGIT

Content-Disposition   =  "Content-Disposition" HCOLON
                         disp-type *( SEMI disp-param )
disp-type             =  "render" / "session" / "icon" / "alert"
                         / disp-extension-token



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 228]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-229" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


disp-param            =  handling-param / generic-param
handling-param        =  "handling" EQUAL
                         ( "optional" / "required"
                         / other-handling )
other-handling        =  token
disp-extension-token  =  token

Content-Encoding  =  ( "Content-Encoding" / "e" ) HCOLON
                     content-coding *(COMMA content-coding)

Content-Language  =  "Content-Language" HCOLON
                     language-tag *(COMMA language-tag)
language-tag      =  primary-tag *( "-" subtag )
primary-tag       =  1*8ALPHA
subtag            =  1*8ALPHA

Content-Length  =  ( "Content-Length" / "l" ) HCOLON 1*DIGIT
Content-Type     =  ( "Content-Type" / "c" ) HCOLON media-type
media-type       =  m-type SLASH m-subtype *(SEMI m-parameter)
m-type           =  discrete-type / composite-type
discrete-type    =  "text" / "image" / "audio" / "video"
                    / "application" / extension-token
composite-type   =  "message" / "multipart" / extension-token
extension-token  =  ietf-token / x-token
ietf-token       =  token
x-token          =  "x-" token
m-subtype        =  extension-token / iana-token
iana-token       =  token
m-parameter      =  m-attribute EQUAL m-value
m-attribute      =  token
m-value          =  token / quoted-string

CSeq  =  "CSeq" HCOLON 1*DIGIT LWS Method

Date          =  "Date" HCOLON SIP-date
SIP-date      =  <a href="./rfc1123">rfc1123</a>-date
<a href="./rfc1123">rfc1123</a>-date  =  wkday "," SP date1 SP time SP "GMT"
date1         =  2DIGIT SP month SP 4DIGIT
                 ; day month year (e.g., 02 Jun 1982)
time          =  2DIGIT ":" 2DIGIT ":" 2DIGIT
                 ; 00:00:00 - 23:59:59
wkday         =  "Mon" / "Tue" / "Wed"
                 / "Thu" / "Fri" / "Sat" / "Sun"
month         =  "Jan" / "Feb" / "Mar" / "Apr"
                 / "May" / "Jun" / "Jul" / "Aug"
                 / "Sep" / "Oct" / "Nov" / "Dec"

Error-Info  =  "Error-Info" HCOLON error-uri *(COMMA error-uri)



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 229]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-230" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


error-uri   =  LAQUOT absoluteURI RAQUOT *( SEMI generic-param )

Expires     =  "Expires" HCOLON delta-seconds
From        =  ( "From" / "f" ) HCOLON from-spec
from-spec   =  ( name-addr / addr-spec )
               *( SEMI from-param )
from-param  =  tag-param / generic-param
tag-param   =  "tag" EQUAL token

In-Reply-To  =  "In-Reply-To" HCOLON callid *(COMMA callid)

Max-Forwards  =  "Max-Forwards" HCOLON 1*DIGIT

MIME-Version  =  "MIME-Version" HCOLON 1*DIGIT "." 1*DIGIT

Min-Expires  =  "Min-Expires" HCOLON delta-seconds

Organization  =  "Organization" HCOLON [TEXT-UTF8-TRIM]

Priority        =  "Priority" HCOLON priority-value
priority-value  =  "emergency" / "urgent" / "normal"
                   / "non-urgent" / other-priority
other-priority  =  token

Proxy-Authenticate  =  "Proxy-Authenticate" HCOLON challenge
challenge           =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                       / other-challenge
other-challenge     =  auth-scheme LWS auth-param
                       *(COMMA auth-param)
digest-cln          =  realm / domain / nonce
                        / opaque / stale / algorithm
                        / qop-options / auth-param
realm               =  "realm" EQUAL realm-value
realm-value         =  quoted-string
domain              =  "domain" EQUAL LDQUOT URI
                       *( 1*SP URI ) RDQUOT
URI                 =  absoluteURI / abs-path
nonce               =  "nonce" EQUAL nonce-value
nonce-value         =  quoted-string
opaque              =  "opaque" EQUAL quoted-string
stale               =  "stale" EQUAL ( "true" / "false" )
algorithm           =  "algorithm" EQUAL ( "MD5" / "MD5-sess"
                       / token )
qop-options         =  "qop" EQUAL LDQUOT qop-value
                       *("," qop-value) RDQUOT
qop-value           =  "auth" / "auth-int" / token

Proxy-Authorization  =  "Proxy-Authorization" HCOLON credentials



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 230]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-231" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Proxy-Require  =  "Proxy-Require" HCOLON option-tag
                  *(COMMA option-tag)
option-tag     =  token

Record-Route  =  "Record-Route" HCOLON rec-route *(COMMA rec-route)
rec-route     =  name-addr *( SEMI rr-param )
rr-param      =  generic-param

Reply-To      =  "Reply-To" HCOLON rplyto-spec
rplyto-spec   =  ( name-addr / addr-spec )
                 *( SEMI rplyto-param )
rplyto-param  =  generic-param
Require       =  "Require" HCOLON option-tag *(COMMA option-tag)

Retry-After  =  "Retry-After" HCOLON delta-seconds
                [ comment ] *( SEMI retry-param )

retry-param  =  ("duration" EQUAL delta-seconds)
                / generic-param

Route        =  "Route" HCOLON route-param *(COMMA route-param)
route-param  =  name-addr *( SEMI rr-param )

Server           =  "Server" HCOLON server-val *(LWS server-val)
server-val       =  product / comment
product          =  token [SLASH product-version]
product-version  =  token

Subject  =  ( "Subject" / "s" ) HCOLON [TEXT-UTF8-TRIM]

Supported  =  ( "Supported" / "k" ) HCOLON
              [option-tag *(COMMA option-tag)]

Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
               [ "." *(DIGIT) ] [ LWS delay ]
delay      =  *(DIGIT) [ "." *(DIGIT) ]

To        =  ( "To" / "t" ) HCOLON ( name-addr
             / addr-spec ) *( SEMI to-param )
to-param  =  tag-param / generic-param

Unsupported  =  "Unsupported" HCOLON option-tag *(COMMA option-tag)
User-Agent  =  "User-Agent" HCOLON server-val *(LWS server-val)








<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 231]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-232" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Via               =  ( "Via" / "v" ) HCOLON via-parm *(COMMA via-parm)
via-parm          =  sent-protocol LWS sent-by *( SEMI via-params )
via-params        =  via-ttl / via-maddr
                     / via-received / via-branch
                     / via-extension
via-ttl           =  "ttl" EQUAL ttl
via-maddr         =  "maddr" EQUAL host
via-received      =  "received" EQUAL (IPv4address / IPv6address)
via-branch        =  "branch" EQUAL token
via-extension     =  generic-param
sent-protocol     =  protocol-name SLASH protocol-version
                     SLASH transport
protocol-name     =  "SIP" / token
protocol-version  =  token
transport         =  "UDP" / "TCP" / "TLS" / "SCTP"
                     / other-transport
sent-by           =  host [ COLON port ]
ttl               =  1*3DIGIT ; 0 to 255

Warning        =  "Warning" HCOLON warning-value *(COMMA warning-value)
warning-value  =  warn-code SP warn-agent SP warn-text
warn-code      =  3DIGIT
warn-agent     =  hostport / pseudonym
                  ;  the name or pseudonym of the server adding
                  ;  the Warning header, for use in debugging
warn-text      =  quoted-string
pseudonym      =  token

WWW-Authenticate  =  "WWW-Authenticate" HCOLON challenge

extension-header  =  header-name HCOLON header-value
header-name       =  token
header-value      =  *(TEXT-UTF8char / UTF8-CONT / LWS)
message-body  =  *OCTET

<span class="h2"><a class="selflink" id="section-26" href="#section-26">26</a> Security Considerations: Threat Model and Security Usage</span>
<span class="h2">   Recommendations</span>

   SIP is not an easy protocol to secure.  Its use of intermediaries,
   its multi-faceted trust relationships, its expected usage between
   elements with no trust at all, and its user-to-user operation make
   security far from trivial.  Security solutions are needed that are
   deployable today, without extensive coordination, in a wide variety
   of environments and usages.  In order to meet these diverse needs,
   several distinct mechanisms applicable to different aspects and
   usages of SIP will be required.





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 232]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-233" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Note that the security of SIP signaling itself has no bearing on the
   security of protocols used in concert with SIP such as RTP, or with
   the security implications of any specific bodies SIP might carry
   (although MIME security plays a substantial role in securing SIP).
   Any media associated with a session can be encrypted end-to-end
   independently of any associated SIP signaling.  Media encryption is
   outside the scope of this document.

   The considerations that follow first examine a set of classic threat
   models that broadly identify the security needs of SIP.  The set of
   security services required to address these threats is then detailed,
   followed by an explanation of several security mechanisms that can be
   used to provide these services.  Next, the requirements for
   implementers of SIP are enumerated, along with exemplary deployments
   in which these security mechanisms could be used to improve the
   security of SIP.  Some notes on privacy conclude this section.

<span class="h3"><a class="selflink" id="section-26.1" href="#section-26.1">26.1</a> Attacks and Threat Models</span>

   This section details some threats that should be common to most
   deployments of SIP.  These threats have been chosen specifically to
   illustrate each of the security services that SIP requires.

   The following examples by no means provide an exhaustive list of the
   threats against SIP; rather, these are "classic" threats that
   demonstrate the need for particular security services that can
   potentially prevent whole categories of threats.

   These attacks assume an environment in which attackers can
   potentially read any packet on the network - it is anticipated that
   SIP will frequently be used on the public Internet.  Attackers on the
   network may be able to modify packets (perhaps at some compromised
   intermediary).  Attackers may wish to steal services, eavesdrop on
   communications, or disrupt sessions.

<span class="h4"><a class="selflink" id="section-26.1.1" href="#section-26.1.1">26.1.1</a> Registration Hijacking</span>

   The SIP registration mechanism allows a user agent to identify itself
   to a registrar as a device at which a user (designated by an address
   of record) is located.  A registrar assesses the identity asserted in
   the From header field of a REGISTER message to determine whether this
   request can modify the contact addresses associated with the
   address-of-record in the To header field.  While these two fields are
   frequently the same, there are many valid deployments in which a
   third-party may register contacts on a user's behalf.






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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-234" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The From header field of a SIP request, however, can be modified
   arbitrarily by the owner of a UA, and this opens the door to
   malicious registrations.  An attacker that successfully impersonates
   a party authorized to change contacts associated with an address-of-
   record could, for example, de-register all existing contacts for a
   URI and then register their own device as the appropriate contact
   address, thereby directing all requests for the affected user to the
   attacker's device.

   This threat belongs to a family of threats that rely on the absence
   of cryptographic assurance of a request's originator.  Any SIP UAS
   that represents a valuable service (a gateway that interworks SIP
   requests with traditional telephone calls, for example) might want to
   control access to its resources by authenticating requests that it
   receives.  Even end-user UAs, for example SIP phones, have an
   interest in ascertaining the identities of originators of requests.

   This threat demonstrates the need for security services that enable
   SIP entities to authenticate the originators of requests.

<span class="h4"><a class="selflink" id="section-26.1.2" href="#section-26.1.2">26.1.2</a> Impersonating a Server</span>

   The domain to which a request is destined is generally specified in
   the Request-URI.  UAs commonly contact a server in this domain
   directly in order to deliver a request.  However, there is always a
   possibility that an attacker could impersonate the remote server, and
   that the UA's request could be intercepted by some other party.

   For example, consider a case in which a redirect server at one
   domain, chicago.com, impersonates a redirect server at another
   domain, biloxi.com.  A user agent sends a request to biloxi.com, but
   the redirect server at chicago.com answers with a forged response
   that has appropriate SIP header fields for a response from
   biloxi.com.  The forged contact addresses in the redirection response
   could direct the originating UA to inappropriate or insecure
   resources, or simply prevent requests for biloxi.com from succeeding.

   This family of threats has a vast membership, many of which are
   critical.  As a converse to the registration hijacking threat,
   consider the case in which a registration sent to biloxi.com is
   intercepted by chicago.com, which replies to the intercepted
   registration with a forged 301 (Moved Permanently) response.  This
   response might seem to come from biloxi.com yet designate chicago.com
   as the appropriate registrar.  All future REGISTER requests from the
   originating UA would then go to chicago.com.

   Prevention of this threat requires a means by which UAs can
   authenticate the servers to whom they send requests.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 234]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-235" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h4"><a class="selflink" id="section-26.1.3" href="#section-26.1.3">26.1.3</a> Tampering with Message Bodies</span>

   As a matter of course, SIP UAs route requests through trusted proxy
   servers.  Regardless of how that trust is established (authentication
   of proxies is discussed elsewhere in this section), a UA may trust a
   proxy server to route a request, but not to inspect or possibly
   modify the bodies contained in that request.

   Consider a UA that is using SIP message bodies to communicate session
   encryption keys for a media session.  Although it trusts the proxy
   server of the domain it is contacting to deliver signaling properly,
   it may not want the administrators of that domain to be capable of
   decrypting any subsequent media session.  Worse yet, if the proxy
   server were actively malicious, it could modify the session key,
   either acting as a man-in-the-middle, or perhaps changing the
   security characteristics requested by the originating UA.

   This family of threats applies not only to session keys, but to most
   conceivable forms of content carried end-to-end in SIP.  These might
   include MIME bodies that should be rendered to the user, SDP, or
   encapsulated telephony signals, among others.  Attackers might
   attempt to modify SDP bodies, for example, in order to point RTP
   media streams to a wiretapping device in order to eavesdrop on
   subsequent voice communications.

   Also note that some header fields in SIP are meaningful end-to-end,
   for example, Subject.  UAs might be protective of these header fields
   as well as bodies (a malicious intermediary changing the Subject
   header field might make an important request appear to be spam, for
   example).  However, since many header fields are legitimately
   inspected or altered by proxy servers as a request is routed, not all
   header fields should be secured end-to-end.

   For these reasons, the UA might want to secure SIP message bodies,
   and in some limited cases header fields, end-to-end.  The security
   services required for bodies include confidentiality, integrity, and
   authentication.  These end-to-end services should be independent of
   the means used to secure interactions with intermediaries such as
   proxy servers.

<span class="h4"><a class="selflink" id="section-26.1.4" href="#section-26.1.4">26.1.4</a> Tearing Down Sessions</span>

   Once a dialog has been established by initial messaging, subsequent
   requests can be sent that modify the state of the dialog and/or
   session.  It is critical that principals in a session can be certain
   that such requests are not forged by attackers.





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 235]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-236" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Consider a case in which a third-party attacker captures some initial
   messages in a dialog shared by two parties in order to learn the
   parameters of the session (To tag, From tag, and so forth) and then
   inserts a BYE request into the session.  The attacker could opt to
   forge the request such that it seemed to come from either
   participant.  Once the BYE is received by its target, the session
   will be torn down prematurely.

   Similar mid-session threats include the transmission of forged re-
   INVITEs that alter the session (possibly to reduce session security
   or redirect media streams as part of a wiretapping attack).

   The most effective countermeasure to this threat is the
   authentication of the sender of the BYE.  In this instance, the
   recipient needs only know that the BYE came from the same party with
   whom the corresponding dialog was established (as opposed to
   ascertaining the absolute identity of the sender).  Also, if the
   attacker is unable to learn the parameters of the session due to
   confidentiality, it would not be possible to forge the BYE.  However,
   some intermediaries (like proxy servers) will need to inspect those
   parameters as the session is established.

<span class="h4"><a class="selflink" id="section-26.1.5" href="#section-26.1.5">26.1.5</a> Denial of Service and Amplification</span>

   Denial-of-service attacks focus on rendering a particular network
   element unavailable, usually by directing an excessive amount of
   network traffic at its interfaces.  A distributed denial-of-service
   attack allows one network user to cause multiple network hosts to
   flood a target host with a large amount of network traffic.

   In many architectures, SIP proxy servers face the public Internet in
   order to accept requests from worldwide IP endpoints.  SIP creates a
   number of potential opportunities for distributed denial-of-service
   attacks that must be recognized and addressed by the implementers and
   operators of SIP systems.

   Attackers can create bogus requests that contain a falsified source
   IP address and a corresponding Via header field that identify a
   targeted host as the originator of the request and then send this
   request to a large number of SIP network elements, thereby using
   hapless SIP UAs or proxies to generate denial-of-service traffic
   aimed at the target.

   Similarly, attackers might use falsified Route header field values in
   a request that identify the target host and then send such messages
   to forking proxies that will amplify messaging sent to the target.





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 236]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-237" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Record-Route could be used to similar effect when the attacker is
   certain that the SIP dialog initiated by the request will result in
   numerous transactions originating in the backwards direction.

   A number of denial-of-service attacks open up if REGISTER requests
   are not properly authenticated and authorized by registrars.
   Attackers could de-register some or all users in an administrative
   domain, thereby preventing these users from being invited to new
   sessions.  An attacker could also register a large number of contacts
   designating the same host for a given address-of-record in order to
   use the registrar and any associated proxy servers as amplifiers in a
   denial-of-service attack.  Attackers might also attempt to deplete
   available memory and disk resources of a registrar by registering
   huge numbers of bindings.

   The use of multicast to transmit SIP requests can greatly increase
   the potential for denial-of-service attacks.

   These problems demonstrate a general need to define architectures
   that minimize the risks of denial-of-service, and the need to be
   mindful in recommendations for security mechanisms of this class of
   attacks.

<span class="h3"><a class="selflink" id="section-26.2" href="#section-26.2">26.2</a> Security Mechanisms</span>

   From the threats described above, we gather that the fundamental
   security services required for the SIP protocol are: preserving the
   confidentiality and integrity of messaging, preventing replay attacks
   or message spoofing, providing for the authentication and privacy of
   the participants in a session, and preventing denial-of-service
   attacks.  Bodies within SIP messages separately require the security
   services of confidentiality, integrity, and authentication.

   Rather than defining new security mechanisms specific to SIP, SIP
   reuses wherever possible existing security models derived from the
   HTTP and SMTP space.

   Full encryption of messages provides the best means to preserve the
   confidentiality of signaling - it can also guarantee that messages
   are not modified by any malicious intermediaries.  However, SIP
   requests and responses cannot be naively encrypted end-to-end in
   their entirety because message fields such as the Request-URI, Route,
   and Via need to be visible to proxies in most network architectures
   so that SIP requests are routed correctly.  Note that proxy servers
   need to modify some features of messages as well (such as adding Via
   header field values) in order for SIP to function.  Proxy servers
   must therefore be trusted, to some degree, by SIP UAs.  To this
   purpose, low-layer security mechanisms for SIP are recommended, which



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 237]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-238" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   encrypt the entire SIP requests or responses on the wire on a hop-
   by-hop basis, and that allow endpoints to verify the identity of
   proxy servers to whom they send requests.

   SIP entities also have a need to identify one another in a secure
   fashion.  When a SIP endpoint asserts the identity of its user to a
   peer UA or to a proxy server, that identity should in some way be
   verifiable.  A cryptographic authentication mechanism is provided in
   SIP to address this requirement.

   An independent security mechanism for SIP message bodies supplies an
   alternative means of end-to-end mutual authentication, as well as
   providing a limit on the degree to which user agents must trust
   intermediaries.

<span class="h4"><a class="selflink" id="section-26.2.1" href="#section-26.2.1">26.2.1</a> Transport and Network Layer Security</span>

   Transport or network layer security encrypts signaling traffic,
   guaranteeing message confidentiality and integrity.

   Oftentimes, certificates are used in the establishment of lower-layer
   security, and these certificates can also be used to provide a means
   of authentication in many architectures.

   Two popular alternatives for providing security at the transport and
   network layer are, respectively, TLS [<a href="#ref-25" title="&quot;The TLS Protocol Version 1.0&quot;">25</a>] and IPSec [<a href="#ref-26" title="&quot;Security Architecture for the Internet Protocol&quot;">26</a>].

   IPSec is a set of network-layer protocol tools that collectively can
   be used as a secure replacement for traditional IP (Internet
   Protocol).  IPSec is most commonly used in architectures in which a
   set of hosts or administrative domains have an existing trust
   relationship with one another.  IPSec is usually implemented at the
   operating system level in a host, or on a security gateway that
   provides confidentiality and integrity for all traffic it receives
   from a particular interface (as in a VPN architecture).  IPSec can
   also be used on a hop-by-hop basis.

   In many architectures IPSec does not require integration with SIP
   applications; IPSec is perhaps best suited to deployments in which
   adding security directly to SIP hosts would be arduous.  UAs that
   have a pre-shared keying relationship with their first-hop proxy
   server are also good candidates to use IPSec.  Any deployment of
   IPSec for SIP would require an IPSec profile describing the protocol
   tools that would be required to secure SIP.  No such profile is given
   in this document.






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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   TLS provides transport-layer security over connection-oriented
   protocols (for the purposes of this document, TCP); "tls" (signifying
   TLS over TCP) can be specified as the desired transport protocol
   within a Via header field value or a SIP-URI.  TLS is most suited to
   architectures in which hop-by-hop security is required between hosts
   with no pre-existing trust association.  For example, Alice trusts
   her local proxy server, which after a certificate exchange decides to
   trust Bob's local proxy server, which Bob trusts, hence Bob and Alice
   can communicate securely.

   TLS must be tightly coupled with a SIP application.  Note that
   transport mechanisms are specified on a hop-by-hop basis in SIP, thus
   a UA that sends requests over TLS to a proxy server has no assurance
   that TLS will be used end-to-end.

   The TLS_RSA_WITH_AES_128_CBC_SHA ciphersuite [<a href="#ref-6" title="&quot;Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS)&quot;">6</a>] MUST be supported at
   a minimum by implementers when TLS is used in a SIP application.  For
   purposes of backwards compatibility, proxy servers, redirect servers,
   and registrars SHOULD support TLS_RSA_WITH_3DES_EDE_CBC_SHA.
   Implementers MAY also support any other ciphersuite.

<span class="h4"><a class="selflink" id="section-26.2.2" href="#section-26.2.2">26.2.2</a> SIPS URI Scheme</span>

   The SIPS URI scheme adheres to the syntax of the SIP URI (described
   in 19), although the scheme string is "sips" rather than "sip".  The
   semantics of SIPS are very different from the SIP URI, however.  SIPS
   allows resources to specify that they should be reached securely.

   A SIPS URI can be used as an address-of-record for a particular user
   - the URI by which the user is canonically known (on their business
   cards, in the From header field of their requests, in the To header
   field of REGISTER requests).  When used as the Request-URI of a
   request, the SIPS scheme signifies that each hop over which the
   request is forwarded, until the request reaches the SIP entity
   responsible for the domain portion of the Request-URI, must be
   secured with TLS; once it reaches the domain in question it is
   handled in accordance with local security and routing policy, quite
   possibly using TLS for any last hop to a UAS.  When used by the
   originator of a request (as would be the case if they employed a SIPS
   URI as the address-of-record of the target), SIPS dictates that the
   entire request path to the target domain be so secured.

   The SIPS scheme is applicable to many of the other ways in which SIP
   URIs are used in SIP today in addition to the Request-URI, including
   in addresses-of-record, contact addresses (the contents of Contact
   headers, including those of REGISTER methods), and Route headers.  In
   each instance, the SIPS URI scheme allows these existing fields to




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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   designate secure resources.  The manner in which a SIPS URI is
   dereferenced in any of these contexts has its own security properties
   which are detailed in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].

   The use of SIPS in particular entails that mutual TLS authentication
   SHOULD be employed, as SHOULD the ciphersuite
   TLS_RSA_WITH_AES_128_CBC_SHA.  Certificates received in the
   authentication process SHOULD be validated with root certificates
   held by the client; failure to validate a certificate SHOULD result
   in the failure of the request.

      Note that in the SIPS URI scheme, transport is independent of TLS,
      and thus "sips:alice@atlanta.com;transport=tcp" and
      "sips:alice@atlanta.com;transport=sctp" are both valid (although
      note that UDP is not a valid transport for SIPS).  The use of
      "transport=tls" has consequently been deprecated, partly because
      it was specific to a single hop of the request.  This is a change
      since <a href="./rfc2543">RFC 2543</a>.

   Users that distribute a SIPS URI as an address-of-record may elect to
   operate devices that refuse requests over insecure transports.

<span class="h4"><a class="selflink" id="section-26.2.3" href="#section-26.2.3">26.2.3</a> HTTP Authentication</span>

   SIP provides a challenge capability, based on HTTP authentication,
   that relies on the 401 and 407 response codes as well as header
   fields for carrying challenges and credentials.  Without significant
   modification, the reuse of the HTTP Digest authentication scheme in
   SIP allows for replay protection and one-way authentication.

   The usage of Digest authentication in SIP is detailed in <a href="#section-22">Section 22</a>.

<span class="h4"><a class="selflink" id="section-26.2.4" href="#section-26.2.4">26.2.4</a> S/MIME</span>

   As is discussed above, encrypting entire SIP messages end-to-end for
   the purpose of confidentiality is not appropriate because network
   intermediaries (like proxy servers) need to view certain header
   fields in order to route messages correctly, and if these
   intermediaries are excluded from security associations, then SIP
   messages will essentially be non-routable.

   However, S/MIME allows SIP UAs to encrypt MIME bodies within SIP,
   securing these bodies end-to-end without affecting message headers.
   S/MIME can provide end-to-end confidentiality and integrity for
   message bodies, as well as mutual authentication.  It is also
   possible to use S/MIME to provide a form of integrity and
   confidentiality for SIP header fields through SIP message tunneling.




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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The usage of S/MIME in SIP is detailed in <a href="#section-23">Section 23</a>.

<span class="h3"><a class="selflink" id="section-26.3" href="#section-26.3">26.3</a> Implementing Security Mechanisms</span>

<span class="h4"><a class="selflink" id="section-26.3.1" href="#section-26.3.1">26.3.1</a> Requirements for Implementers of SIP</span>

   Proxy servers, redirect servers, and registrars MUST implement TLS,
   and MUST support both mutual and one-way authentication.  It is
   strongly RECOMMENDED that UAs be capable initiating TLS; UAs MAY also
   be capable of acting as a TLS server.  Proxy servers, redirect
   servers, and registrars SHOULD possess a site certificate whose
   subject corresponds to their canonical hostname.  UAs MAY have
   certificates of their own for mutual authentication with TLS, but no
   provisions are set forth in this document for their use.  All SIP
   elements that support TLS MUST have a mechanism for validating
   certificates received during TLS negotiation; this entails possession
   of one or more root certificates issued by certificate authorities
   (preferably well-known distributors of site certificates comparable
   to those that issue root certificates for web browsers).

   All SIP elements that support TLS MUST also support the SIPS URI
   scheme.

   Proxy servers, redirect servers, registrars, and UAs MAY also
   implement IPSec or other lower-layer security protocols.

   When a UA attempts to contact a proxy server, redirect server, or
   registrar, the UAC SHOULD initiate a TLS connection over which it
   will send SIP messages.  In some architectures, UASs MAY receive
   requests over such TLS connections as well.

   Proxy servers, redirect servers, registrars, and UAs MUST implement
   Digest Authorization, encompassing all of the aspects required in 22.
   Proxy servers, redirect servers, and registrars SHOULD be configured
   with at least one Digest realm, and at least one "realm" string
   supported by a given server SHOULD correspond to the server's
   hostname or domainname.

   UAs MAY support the signing and encrypting of MIME bodies, and
   transference of credentials with S/MIME as described in <a href="#section-23">Section 23</a>.
   If a UA holds one or more root certificates of certificate
   authorities in order to validate certificates for TLS or IPSec, it
   SHOULD be capable of reusing these to verify S/MIME certificates, as
   appropriate.  A UA MAY hold root certificates specifically for
   validating S/MIME certificates.






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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      Note that is it anticipated that future security extensions may
      upgrade the normative strength associated with S/MIME as S/MIME
      implementations appear and the problem space becomes better
      understood.

<span class="h4"><a class="selflink" id="section-26.3.2" href="#section-26.3.2">26.3.2</a> Security Solutions</span>

   The operation of these security mechanisms in concert can follow the
   existing web and email security models to some degree.  At a high
   level, UAs authenticate themselves to servers (proxy servers,
   redirect servers, and registrars) with a Digest username and
   password; servers authenticate themselves to UAs one hop away, or to
   another server one hop away (and vice versa), with a site certificate
   delivered by TLS.

   On a peer-to-peer level, UAs trust the network to authenticate one
   another ordinarily; however, S/MIME can also be used to provide
   direct authentication when the network does not, or if the network
   itself is not trusted.

   The following is an illustrative example in which these security
   mechanisms are used by various UAs and servers to prevent the sorts
   of threats described in <a href="#section-26.1">Section 26.1</a>.  While implementers and network
   administrators MAY follow the normative guidelines given in the
   remainder of this section, these are provided only as example
   implementations.

<span class="h5"><a class="selflink" id="section-26.3.2.1" href="#section-26.3.2.1">26.3.2.1</a> Registration</span>

   When a UA comes online and registers with its local administrative
   domain, it SHOULD establish a TLS connection with its registrar
   (<a href="#section-10">Section 10</a> describes how the UA reaches its registrar).  The
   registrar SHOULD offer a certificate to the UA, and the site
   identified by the certificate MUST correspond with the domain in
   which the UA intends to register; for example, if the UA intends to
   register the address-of-record 'alice@atlanta.com', the site
   certificate must identify a host within the atlanta.com domain (such
   as sip.atlanta.com).  When it receives the TLS Certificate message,
   the UA SHOULD verify the certificate and inspect the site identified
   by the certificate.  If the certificate is invalid, revoked, or if it
   does not identify the appropriate party, the UA MUST NOT send the
   REGISTER message and otherwise proceed with the registration.

      When a valid certificate has been provided by the registrar, the
      UA knows that the registrar is not an attacker who might redirect
      the UA, steal passwords, or attempt any similar attacks.





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   The UA then creates a REGISTER request that SHOULD be addressed to a
   Request-URI corresponding to the site certificate received from the
   registrar.  When the UA sends the REGISTER request over the existing
   TLS connection, the registrar SHOULD challenge the request with a 401
   (Proxy Authentication Required) response.  The "realm" parameter
   within the Proxy-Authenticate header field of the response SHOULD
   correspond to the domain previously given by the site certificate.
   When the UAC receives the challenge, it SHOULD either prompt the user
   for credentials or take an appropriate credential from a keyring
   corresponding to the "realm" parameter in the challenge.  The
   username of this credential SHOULD correspond with the "userinfo"
   portion of the URI in the To header field of the REGISTER request.
   Once the Digest credentials have been inserted into an appropriate
   Proxy-Authorization header field, the REGISTER should be resubmitted
   to the registrar.

      Since the registrar requires the user agent to authenticate
      itself, it would be difficult for an attacker to forge REGISTER
      requests for the user's address-of-record.  Also note that since
      the REGISTER is sent over a confidential TLS connection, attackers
      will not be able to intercept the REGISTER to record credentials
      for any possible replay attack.

   Once the registration has been accepted by the registrar, the UA
   SHOULD leave this TLS connection open provided that the registrar
   also acts as the proxy server to which requests are sent for users in
   this administrative domain.  The existing TLS connection will be
   reused to deliver incoming requests to the UA that has just completed
   registration.

      Because the UA has already authenticated the server on the other
      side of the TLS connection, all requests that come over this
      connection are known to have passed through the proxy server -
      attackers cannot create spoofed requests that appear to have been
      sent through that proxy server.

<span class="h5"><a class="selflink" id="section-26.3.2.2" href="#section-26.3.2.2">26.3.2.2</a> Interdomain Requests</span>

   Now let's say that Alice's UA would like to initiate a session with a
   user in a remote administrative domain, namely "bob@biloxi.com".  We
   will also say that the local administrative domain (atlanta.com) has
   a local outbound proxy.

   The proxy server that handles inbound requests for an administrative
   domain MAY also act as a local outbound proxy; for simplicity's sake
   we'll assume this to be the case for atlanta.com (otherwise the user
   agent would initiate a new TLS connection to a separate server at
   this point).  Assuming that the client has completed the registration



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   process described in the preceding section, it SHOULD reuse the TLS
   connection to the local proxy server when it sends an INVITE request
   to another user.  The UA SHOULD reuse cached credentials in the
   INVITE to avoid prompting the user unnecessarily.

   When the local outbound proxy server has validated the credentials
   presented by the UA in the INVITE, it SHOULD inspect the Request-URI
   to determine how the message should be routed (see [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>]).  If the
   "domainname" portion of the Request-URI had corresponded to the local
   domain (atlanta.com) rather than biloxi.com, then the proxy server
   would have consulted its location service to determine how best to
   reach the requested user.

      Had "alice@atlanta.com" been attempting to contact, say,
      "alex@atlanta.com", the local proxy would have proxied to the
      request to the TLS connection Alex had established with the
      registrar when he registered.  Since Alex would receive this
      request over his authenticated channel, he would be assured that
      Alice's request had been authorized by the proxy server of the
      local administrative domain.

   However, in this instance the Request-URI designates a remote domain.
   The local outbound proxy server at atlanta.com SHOULD therefore
   establish a TLS connection with the remote proxy server at
   biloxi.com.  Since both of the participants in this TLS connection
   are servers that possess site certificates, mutual TLS authentication
   SHOULD occur.  Each side of the connection SHOULD verify and inspect
   the certificate of the other, noting the domain name that appears in
   the certificate for comparison with the header fields of SIP
   messages.  The atlanta.com proxy server, for example, SHOULD verify
   at this stage that the certificate received from the remote side
   corresponds with the biloxi.com domain.  Once it has done so, and TLS
   negotiation has completed, resulting in a secure channel between the
   two proxies, the atlanta.com proxy can forward the INVITE request to
   biloxi.com.

   The proxy server at biloxi.com SHOULD inspect the certificate of the
   proxy server at atlanta.com in turn and compare the domain asserted
   by the certificate with the "domainname" portion of the From header
   field in the INVITE request.  The biloxi proxy MAY have a strict
   security policy that requires it to reject requests that do not match
   the administrative domain from which they have been proxied.

      Such security policies could be instituted to prevent the SIP
      equivalent of SMTP 'open relays' that are frequently exploited to
      generate spam.





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   This policy, however, only guarantees that the request came from the
   domain it ascribes to itself; it does not allow biloxi.com to
   ascertain how atlanta.com authenticated Alice.  Only if biloxi.com
   has some other way of knowing atlanta.com's authentication policies
   could it possibly ascertain how Alice proved her identity.
   biloxi.com might then institute an even stricter policy that forbids
   requests that come from domains that are not known administratively
   to share a common authentication policy with biloxi.com.

   Once the INVITE has been approved by the biloxi proxy, the proxy
   server SHOULD identify the existing TLS channel, if any, associated
   with the user targeted by this request (in this case
   "bob@biloxi.com").  The INVITE should be proxied through this channel
   to Bob.  Since the request is received over a TLS connection that had
   previously been authenticated as the biloxi proxy, Bob knows that the
   From header field was not tampered with and that atlanta.com has
   validated Alice, although not necessarily whether or not to trust
   Alice's identity.

   Before they forward the request, both proxy servers SHOULD add a
   Record-Route header field to the request so that all future requests
   in this dialog will pass through the proxy servers.  The proxy
   servers can thereby continue to provide security services for the
   lifetime of this dialog.  If the proxy servers do not add themselves
   to the Record-Route, future messages will pass directly end-to-end
   between Alice and Bob without any security services (unless the two
   parties agree on some independent end-to-end security such as
   S/MIME).  In this respect the SIP trapezoid model can provide a nice
   structure where conventions of agreement between the site proxies can
   provide a reasonably secure channel between Alice and Bob.

      An attacker preying on this architecture would, for example, be
      unable to forge a BYE request and insert it into the signaling
      stream between Bob and Alice because the attacker has no way of
      ascertaining the parameters of the session and also because the
      integrity mechanism transitively protects the traffic between
      Alice and Bob.

<span class="h5"><a class="selflink" id="section-26.3.2.3" href="#section-26.3.2.3">26.3.2.3</a> Peer-to-Peer Requests</span>

   Alternatively, consider a UA asserting the identity
   "carol@chicago.com" that has no local outbound proxy.  When Carol
   wishes to send an INVITE to "bob@biloxi.com", her UA SHOULD initiate
   a TLS connection with the biloxi proxy directly (using the mechanism
   described in [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>] to determine how to best to reach the given
   Request-URI).  When her UA receives a certificate from the biloxi
   proxy, it SHOULD be verified normally before she passes her INVITE
   across the TLS connection.  However, Carol has no means of proving



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   her identity to the biloxi proxy, but she does have a CMS-detached
   signature over a "message/sip" body in the INVITE.  It is unlikely in
   this instance that Carol would have any credentials in the biloxi.com
   realm, since she has no formal association with biloxi.com.  The
   biloxi proxy MAY also have a strict policy that precludes it from
   even bothering to challenge requests that do not have biloxi.com in
   the "domainname" portion of the From header field - it treats these
   users as unauthenticated.

   The biloxi proxy has a policy for Bob that all non-authenticated
   requests should be redirected to the appropriate contact address
   registered against 'bob@biloxi.com', namely &lt;sip:bob@192.0.2.4&gt;.
   Carol receives the redirection response over the TLS connection she
   established with the biloxi proxy, so she trusts the veracity of the
   contact address.

   Carol SHOULD then establish a TCP connection with the designated
   address and send a new INVITE with a Request-URI containing the
   received contact address (recomputing the signature in the body as
   the request is readied).  Bob receives this INVITE on an insecure
   interface, but his UA inspects and, in this instance, recognizes the
   From header field of the request and subsequently matches a locally
   cached certificate with the one presented in the signature of the
   body of the INVITE.  He replies in similar fashion, authenticating
   himself to Carol, and a secure dialog begins.

      Sometimes firewalls or NATs in an administrative domain could
      preclude the establishment of a direct TCP connection to a UA.  In
      these cases, proxy servers could also potentially relay requests
      to UAs in a way that has no trust implications (for example,
      forgoing an existing TLS connection and forwarding the request
      over cleartext TCP) as local policy dictates.

<span class="h5"><a class="selflink" id="section-26.3.2.4" href="#section-26.3.2.4">26.3.2.4</a> DoS Protection</span>

   In order to minimize the risk of a denial-of-service attack against
   architectures using these security solutions, implementers should
   take note of the following guidelines.

   When the host on which a SIP proxy server is operating is routable
   from the public Internet, it SHOULD be deployed in an administrative
   domain with defensive operational policies (blocking source-routed
   traffic, preferably filtering ping traffic).  Both TLS and IPSec can
   also make use of bastion hosts at the edges of administrative domains
   that participate in the security associations to aggregate secure
   tunnels and sockets.  These bastion hosts can also take the brunt of
   denial-of-service attacks, ensuring that SIP hosts within the
   administrative domain are not encumbered with superfluous messaging.



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   No matter what security solutions are deployed, floods of messages
   directed at proxy servers can lock up proxy server resources and
   prevent desirable traffic from reaching its destination.  There is a
   computational expense associated with processing a SIP transaction at
   a proxy server, and that expense is greater for stateful proxy
   servers than it is for stateless proxy servers.  Therefore, stateful
   proxies are more susceptible to flooding than stateless proxy
   servers.

   UAs and proxy servers SHOULD challenge questionable requests with
   only a single 401 (Unauthorized) or 407 (Proxy Authentication
   Required), forgoing the normal response retransmission algorithm, and
   thus behaving statelessly towards unauthenticated requests.

      Retransmitting the 401 (Unauthorized) or 407 (Proxy Authentication
      Required) status response amplifies the problem of an attacker
      using a falsified header field value (such as Via) to direct
      traffic to a third party.

   In summary, the mutual authentication of proxy servers through
   mechanisms such as TLS significantly reduces the potential for rogue
   intermediaries to introduce falsified requests or responses that can
   deny service.  This commensurately makes it harder for attackers to
   make innocent SIP nodes into agents of amplification.

<span class="h3"><a class="selflink" id="section-26.4" href="#section-26.4">26.4</a> Limitations</span>

   Although these security mechanisms, when applied in a judicious
   manner, can thwart many threats, there are limitations in the scope
   of the mechanisms that must be understood by implementers and network
   operators.

<span class="h4"><a class="selflink" id="section-26.4.1" href="#section-26.4.1">26.4.1</a> HTTP Digest</span>

   One of the primary limitations of using HTTP Digest in SIP is that
   the integrity mechanisms in Digest do not work very well for SIP.
   Specifically, they offer protection of the Request-URI and the method
   of a message, but not for any of the header fields that UAs would
   most likely wish to secure.

   The existing replay protection mechanisms described in <a href="./rfc2617">RFC 2617</a> also
   have some limitations for SIP.  The next-nonce mechanism, for
   example, does not support pipelined requests.  The nonce-count
   mechanism should be used for replay protection.

   Another limitation of HTTP Digest is the scope of realms.  Digest is
   valuable when a user wants to authenticate themselves to a resource
   with which they have a pre-existing association, like a service



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   provider of which the user is a customer (which is quite a common
   scenario and thus Digest provides an extremely useful function).  By
   way of contrast, the scope of TLS is interdomain or multirealm, since
   certificates are often globally verifiable, so that the UA can
   authenticate the server with no pre-existing association.

<span class="h4"><a class="selflink" id="section-26.4.2" href="#section-26.4.2">26.4.2</a> S/MIME</span>

   The largest outstanding defect with the S/MIME mechanism is the lack
   of a prevalent public key infrastructure for end users.  If self-
   signed certificates (or certificates that cannot be verified by one
   of the participants in a dialog) are used, the SIP-based key exchange
   mechanism described in <a href="#section-23.2">Section 23.2</a> is susceptible to a man-in-the-
   middle attack with which an attacker can potentially inspect and
   modify S/MIME bodies.  The attacker needs to intercept the first
   exchange of keys between the two parties in a dialog, remove the
   existing CMS-detached signatures from the request and response, and
   insert a different CMS-detached signature containing a certificate
   supplied by the attacker (but which seems to be a certificate for the
   proper address-of-record).  Each party will think they have exchanged
   keys with the other, when in fact each has the public key of the
   attacker.

   It is important to note that the attacker can only leverage this
   vulnerability on the first exchange of keys between two parties - on
   subsequent occasions, the alteration of the key would be noticeable
   to the UAs.  It would also be difficult for the attacker to remain in
   the path of all future dialogs between the two parties over time (as
   potentially days, weeks, or years pass).

   SSH is susceptible to the same man-in-the-middle attack on the first
   exchange of keys; however, it is widely acknowledged that while SSH
   is not perfect, it does improve the security of connections.  The use
   of key fingerprints could provide some assistance to SIP, just as it
   does for SSH.  For example, if two parties use SIP to establish a
   voice communications session, each could read off the fingerprint of
   the key they received from the other, which could be compared against
   the original.  It would certainly be more difficult for the man-in-
   the-middle to emulate the voices of the participants than their
   signaling (a practice that was used with the Clipper chip-based
   secure telephone).

   The S/MIME mechanism allows UAs to send encrypted requests without
   preamble if they possess a certificate for the destination address-
   of-record on their keyring.  However, it is possible that any
   particular device registered for an address-of-record will not hold
   the certificate that has been previously employed by the device's
   current user, and that it will therefore be unable to process an



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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   encrypted request properly, which could lead to some avoidable error
   signaling.  This is especially likely when an encrypted request is
   forked.

   The keys associated with S/MIME are most useful when associated with
   a particular user (an address-of-record) rather than a device (a UA).
   When users move between devices, it may be difficult to transport
   private keys securely between UAs; how such keys might be acquired by
   a device is outside the scope of this document.

   Another, more prosaic difficulty with the S/MIME mechanism is that it
   can result in very large messages, especially when the SIP tunneling
   mechanism described in <a href="#section-23.4">Section 23.4</a> is used.  For that reason, it is
   RECOMMENDED that TCP should be used as a transport protocol when
   S/MIME tunneling is employed.

<span class="h4"><a class="selflink" id="section-26.4.3" href="#section-26.4.3">26.4.3</a> TLS</span>

   The most commonly voiced concern about TLS is that it cannot run over
   UDP; TLS requires a connection-oriented underlying transport
   protocol, which for the purposes of this document means TCP.

   It may also be arduous for a local outbound proxy server and/or
   registrar to maintain many simultaneous long-lived TLS connections
   with numerous UAs.  This introduces some valid scalability concerns,
   especially for intensive ciphersuites.  Maintaining redundancy of
   long-lived TLS connections, especially when a UA is solely
   responsible for their establishment, could also be cumbersome.

   TLS only allows SIP entities to authenticate servers to which they
   are adjacent; TLS offers strictly hop-by-hop security.  Neither TLS,
   nor any other mechanism specified in this document, allows clients to
   authenticate proxy servers to whom they cannot form a direct TCP
   connection.

<span class="h4"><a class="selflink" id="section-26.4.4" href="#section-26.4.4">26.4.4</a> SIPS URIs</span>

   Actually using TLS on every segment of a request path entails that
   the terminating UAS must be reachable over TLS (perhaps registering
   with a SIPS URI as a contact address).  This is the preferred use of
   SIPS.  Many valid architectures, however, use TLS to secure part of
   the request path, but rely on some other mechanism for the final hop
   to a UAS, for example.  Thus SIPS cannot guarantee that TLS usage
   will be truly end-to-end.  Note that since many UAs will not accept
   incoming TLS connections, even those UAs that do support TLS may be
   required to maintain persistent TLS connections as described in the
   TLS limitations section above in order to receive requests over TLS
   as a UAS.



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-250" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Location services are not required to provide a SIPS binding for a
   SIPS Request-URI.  Although location services are commonly populated
   by user registrations (as described in <a href="#section-10.2.1">Section 10.2.1</a>), various other
   protocols and interfaces could conceivably supply contact addresses
   for an AOR, and these tools are free to map SIPS URIs to SIP URIs as
   appropriate.  When queried for bindings, a location service returns
   its contact addresses without regard for whether it received a
   request with a SIPS Request-URI.  If a redirect server is accessing
   the location service, it is up to the entity that processes the
   Contact header field of a redirection to determine the propriety of
   the contact addresses.

   Ensuring that TLS will be used for all of the request segments up to
   the target domain is somewhat complex.  It is possible that
   cryptographically authenticated proxy servers along the way that are
   non-compliant or compromised may choose to disregard the forwarding
   rules associated with SIPS (and the general forwarding rules in
   <a href="#section-16.6">Section 16.6</a>).  Such malicious intermediaries could, for example,
   retarget a request from a SIPS URI to a SIP URI in an attempt to
   downgrade security.

   Alternatively, an intermediary might legitimately retarget a request
   from a SIP to a SIPS URI.  Recipients of a request whose Request-URI
   uses the SIPS URI scheme thus cannot assume on the basis of the
   Request-URI alone that SIPS was used for the entire request path
   (from the client onwards).

   To address these concerns, it is RECOMMENDED that recipients of a
   request whose Request-URI contains a SIP or SIPS URI inspect the To
   header field value to see if it contains a SIPS URI (though note that
   it does not constitute a breach of security if this URI has the same
   scheme but is not equivalent to the URI in the To header field).
   Although clients may choose to populate the Request-URI and To header
   field of a request differently, when SIPS is used this disparity
   could be interpreted as a possible security violation, and the
   request could consequently be rejected by its recipient.  Recipients
   MAY also inspect the Via header chain in order to double-check
   whether or not TLS was used for the entire request path until the
   local administrative domain was reached.  S/MIME may also be used by
   the originating UAC to help ensure that the original form of the To
   header field is carried end-to-end.

   If the UAS has reason to believe that the scheme of the Request-URI
   has been improperly modified in transit, the UA SHOULD notify its
   user of a potential security breach.






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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-251" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   As a further measure to prevent downgrade attacks, entities that
   accept only SIPS requests MAY also refuse connections on insecure
   ports.

   End users will undoubtedly discern the difference between SIPS and
   SIP URIs, and they may manually edit them in response to stimuli.
   This can either benefit or degrade security.  For example, if an
   attacker corrupts a DNS cache, inserting a fake record set that
   effectively removes all SIPS records for a proxy server, then any
   SIPS requests that traverse this proxy server may fail.  When a user,
   however, sees that repeated calls to a SIPS AOR are failing, they
   could on some devices manually convert the scheme from SIPS to SIP
   and retry.  Of course, there are some safeguards against this (if the
   destination UA is truly paranoid it could refuse all non-SIPS
   requests), but it is a limitation worth noting.  On the bright side,
   users might also divine that 'SIPS' would be valid even when they are
   presented only with a SIP URI.

<span class="h3"><a class="selflink" id="section-26.5" href="#section-26.5">26.5</a> Privacy</span>

   SIP messages frequently contain sensitive information about their
   senders - not just what they have to say, but with whom they
   communicate, when they communicate and for how long, and from where
   they participate in sessions.  Many applications and their users
   require that this sort of private information be hidden from any
   parties that do not need to know it.

   Note that there are also less direct ways in which private
   information can be divulged.  If a user or service chooses to be
   reachable at an address that is guessable from the person's name and
   organizational affiliation (which describes most addresses-of-
   record), the traditional method of ensuring privacy by having an
   unlisted "phone number" is compromised.  A user location service can
   infringe on the privacy of the recipient of a session invitation by
   divulging their specific whereabouts to the caller; an implementation
   consequently SHOULD be able to restrict, on a per-user basis, what
   kind of location and availability information is given out to certain
   classes of callers.  This is a whole class of problem that is
   expected to be studied further in ongoing SIP work.

   In some cases, users may want to conceal personal information in
   header fields that convey identity.  This can apply not only to the
   From and related headers representing the originator of the request,
   but also the To - it may not be appropriate to convey to the final
   destination a speed-dialing nickname, or an unexpanded identifier for
   a group of targets, either of which would be removed from the
   Request-URI as the request is routed, but not changed in the To




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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-252" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   header field if the two were initially identical.  Thus it MAY be
   desirable for privacy reasons to create a To header field that
   differs from the Request-URI.

<span class="h2"><a class="selflink" id="section-27" href="#section-27">27</a> IANA Considerations</span>

   All method names, header field names, status codes, and option tags
   used in SIP applications are registered with IANA through
   instructions in an IANA Considerations section in an RFC.

   The specification instructs the IANA to create four new sub-
   registries under <a href="http://www.iana">http://www.iana</a>.org/assignments/sip-parameters:
   Option Tags, Warning Codes (warn-codes), Methods and Response Codes,
   added to the sub-registry of Header Fields that is already present
   there.

<span class="h3"><a class="selflink" id="section-27.1" href="#section-27.1">27.1</a> Option Tags</span>

   This specification establishes the Option Tags sub-registry under
   <a href="http://www.iana.org/assignments/sip-parameters">http://www.iana.org/assignments/sip-parameters</a>.

   Option tags are used in header fields such as Require, Supported,
   Proxy-Require, and Unsupported in support of SIP compatibility
   mechanisms for extensions (<a href="#section-19.2">Section 19.2</a>).  The option tag itself is a
   string that is associated with a particular SIP option (that is, an
   extension).  It identifies the option to SIP endpoints.

   Option tags are registered by the IANA when they are published in
   standards track RFCs.  The IANA Considerations section of the RFC
   must include the following information, which appears in the IANA
   registry along with the RFC number of the publication.

      o  Name of the option tag.  The name MAY be of any length, but
         SHOULD be no more than twenty characters long.  The name MUST
         consist of alphanum (<a href="#section-25">Section 25</a>) characters only.

      o  Descriptive text that describes the extension.

<span class="h3"><a class="selflink" id="section-27.2" href="#section-27.2">27.2</a> Warn-Codes</span>

   This specification establishes the Warn-codes sub-registry under
   <a href="http://www.iana.org/assignments/sip-parameters">http://www.iana.org/assignments/sip-parameters</a> and initiates its
   population with the warn-codes listed in <a href="#section-20.43">Section 20.43</a>.  Additional
   warn-codes are registered by RFC publication.







<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 252]</span></pre>
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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   The descriptive text for the table of warn-codes is:

   Warning codes provide information supplemental to the status code in
   SIP response messages when the failure of the transaction results
   from a Session Description Protocol (SDP) (<a href="./rfc2327">RFC 2327</a> [<a href="#ref-1" title="&quot;SDP: Session Description Protocol&quot;">1</a>]) problem.

   The "warn-code" consists of three digits.  A first digit of "3"
   indicates warnings specific to SIP.  Until a future specification
   describes uses of warn-codes other than 3xx, only 3xx warn-codes may
   be registered.

   Warnings 300 through 329 are reserved for indicating problems with
   keywords in the session description, 330 through 339 are warnings
   related to basic network services requested in the session
   description, 370 through 379 are warnings related to quantitative QoS
   parameters requested in the session description, and 390 through 399
   are miscellaneous warnings that do not fall into one of the above
   categories.

<span class="h3"><a class="selflink" id="section-27.3" href="#section-27.3">27.3</a> Header Field Names</span>

   This obsoletes the IANA instructions about the header sub-registry
   under <a href="http://www.iana.org/assignments/sip-parameters">http://www.iana.org/assignments/sip-parameters</a>.

   The following information needs to be provided in an RFC publication
   in order to register a new header field name:

      o  The RFC number in which the header is registered;

      o  the name of the header field being registered;

      o  a compact form version for that header field, if one is
         defined;

   Some common and widely used header fields MAY be assigned one-letter
   compact forms (<a href="#section-7.3.3">Section 7.3.3</a>).  Compact forms can only be assigned
   after SIP working group review, followed by RFC publication.

<span class="h3"><a class="selflink" id="section-27.4" href="#section-27.4">27.4</a> Method and Response Codes</span>

   This specification establishes the Method and Response-Code sub-
   registries under <a href="http://www.iana.org/assignments/sip-parameters">http://www.iana.org/assignments/sip-parameters</a> and
   initiates their population as follows.  The initial Methods table is:








<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 253]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-254" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


         INVITE                   [<a href="./rfc3261">RFC3261</a>]
         ACK                      [<a href="./rfc3261">RFC3261</a>]
         BYE                      [<a href="./rfc3261">RFC3261</a>]
         CANCEL                   [<a href="./rfc3261">RFC3261</a>]
         REGISTER                 [<a href="./rfc3261">RFC3261</a>]
         OPTIONS                  [<a href="./rfc3261">RFC3261</a>]
         INFO                     [<a href="./rfc2976">RFC2976</a>]

   The response code table is initially populated from <a href="#section-21">Section 21</a>, the
   portions labeled Informational, Success, Redirection, Client-Error,
   Server-Error, and Global-Failure.  The table has the following
   format:

      Type (e.g., Informational)
            Number    Default Reason Phrase         [<a href="./rfc3261">RFC3261</a>]

   The following information needs to be provided in an RFC publication
   in order to register a new response code or method:

      o  The RFC number in which the method or response code is
         registered;

      o  the number of the response code or name of the method being
         registered;

      o  the default reason phrase for that response code, if
         applicable;

<span class="h3"><a class="selflink" id="section-27.5" href="#section-27.5">27.5</a> The "message/sip" MIME type.</span>

   This document registers the "message/sip" MIME media type in order to
   allow SIP messages to be tunneled as bodies within SIP, primarily for
   end-to-end security purposes.  This media type is defined by the
   following information:

      Media type name: message
      Media subtype name: sip
      Required parameters: none

      Optional parameters: version
         version: The SIP-Version number of the enclosed message (e.g.,
         "2.0").  If not present, the version defaults to "2.0".
      Encoding scheme: SIP messages consist of an 8-bit header
         optionally followed by a binary MIME data object.  As such, SIP
         messages must be treated as binary.  Under normal circumstances
         SIP messages are transported over binary-capable transports, no
         special encodings are needed.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 254]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-255" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


      Security considerations: see below
         Motivation and examples of this usage as a security mechanism
         in concert with S/MIME are given in 23.4.

<span class="h3"><a class="selflink" id="section-27.6" href="#section-27.6">27.6</a> New Content-Disposition Parameter Registrations</span>

   This document also registers four new Content-Disposition header
   "disposition-types": alert, icon, session and render.  The authors
   request that these values be recorded in the IANA registry for
   Content-Dispositions.

   Descriptions of these "disposition-types", including motivation and
   examples, are given in <a href="#section-20.11">Section 20.11</a>.

   Short descriptions suitable for the IANA registry are:

      alert     the body is a custom ring tone to alert the user
      icon      the body is displayed as an icon to the user
      render    the body should be displayed to the user
      session   the body describes a communications session, for
                example, as <a href="./rfc2327">RFC 2327</a> SDP body

<span class="h2"><a class="selflink" id="section-28" href="#section-28">28</a> Changes From <a href="./rfc2543">RFC 2543</a></span>

   This RFC revises <a href="./rfc2543">RFC 2543</a>.  It is mostly backwards compatible with
   <a href="./rfc2543">RFC 2543</a>.  The changes described here fix many errors discovered in
   <a href="./rfc2543">RFC 2543</a> and provide information on scenarios not detailed in <a href="./rfc2543">RFC</a>
   <a href="./rfc2543">2543</a>.  The protocol has been presented in a more cleanly layered
   model here.

   We break the differences into functional behavior that is a
   substantial change from <a href="./rfc2543">RFC 2543</a>, which has impact on
   interoperability or correct operation in some cases, and functional
   behavior that is different from <a href="./rfc2543">RFC 2543</a> but not a potential source
   of interoperability problems.  There have been countless
   clarifications as well, which are not documented here.

<span class="h3"><a class="selflink" id="section-28.1" href="#section-28.1">28.1</a> Major Functional Changes</span>

   o  When a UAC wishes to terminate a call before it has been answered,
      it sends CANCEL.  If the original INVITE still returns a 2xx, the
      UAC then sends BYE.  BYE can only be sent on an existing call leg
      (now called a dialog in this RFC), whereas it could be sent at any
      time in <a href="./rfc2543">RFC 2543</a>.

   o  The SIP BNF was converted to be <a href="./rfc2234">RFC 2234</a> compliant.





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 255]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-256" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   o  SIP URL BNF was made more general, allowing a greater set of
      characters in the user part.  Furthermore, comparison rules were
      simplified to be primarily case-insensitive, and detailed handling
      of comparison in the presence of parameters was described.  The
      most substantial change is that a URI with a parameter with the
      default value does not match a URI without that parameter.

   o  Removed Via hiding.  It had serious trust issues, since it relied
      on the next hop to perform the obfuscation process.  Instead, Via
      hiding can be done as a local implementation choice in stateful
      proxies, and thus is no longer documented.

   o  In <a href="./rfc2543">RFC 2543</a>, CANCEL and INVITE transactions were intermingled.
      They are separated now.  When a user sends an INVITE and then a
      CANCEL, the INVITE transaction still terminates normally.  A UAS
      needs to respond to the original INVITE request with a 487
      response.

   o  Similarly, CANCEL and BYE transactions were intermingled; <a href="./rfc2543">RFC 2543</a>
      allowed the UAS not to send a response to INVITE when a BYE was
      received.  That is disallowed here.  The original INVITE needs a
      response.

   o  In <a href="./rfc2543">RFC 2543</a>, UAs needed to support only UDP.  In this RFC, UAs
      need to support both UDP and TCP.

   o  In <a href="./rfc2543">RFC 2543</a>, a forking proxy only passed up one challenge from
      downstream elements in the event of multiple challenges.  In this
      RFC, proxies are supposed to collect all challenges and place them
      into the forwarded response.

   o  In Digest credentials, the URI needs to be quoted; this is unclear
      from <a href="./rfc2617">RFC 2617</a> and <a href="./rfc2069">RFC 2069</a> which are both inconsistent on it.

   o  SDP processing has been split off into a separate specification
      [<a href="#ref-13" title="&quot;An Offer/Answer Model with SDP&quot;">13</a>], and more fully specified as a formal offer/answer exchange
      process that is effectively tunneled through SIP.  SDP is allowed
      in INVITE/200 or 200/ACK for baseline SIP implementations; <a href="./rfc2543">RFC</a>
      <a href="./rfc2543">2543</a> alluded to the ability to use it in INVITE, 200, and ACK in a
      single transaction, but this was not well specified.  More complex
      SDP usages are allowed in extensions.










<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 256]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-257" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   o  Added full support for IPv6 in URIs and in the Via header field.
      Support for IPv6 in Via has required that its header field
      parameters allow the square bracket and colon characters.  These
      characters were previously not permitted.  In theory, this could
      cause interop problems with older implementations.  However, we
      have observed that most implementations accept any non-control
      ASCII character in these parameters.

   o  DNS SRV procedure is now documented in a separate specification
      [<a href="#ref-4" title="&quot;SIP: Locating SIP Servers&quot;">4</a>].  This procedure uses both SRV and NAPTR resource records and
      no longer combines data from across SRV records as described in
      <a href="./rfc2543">RFC 2543</a>.

   o  Loop detection has been made optional, supplanted by a mandatory
      usage of Max-Forwards.  The loop detection procedure in <a href="./rfc2543">RFC 2543</a>
      had a serious bug which would report "spirals" as an error
      condition when it was not.  The optional loop detection procedure
      is more fully and correctly specified here.

   o  Usage of tags is now mandatory (they were optional in <a href="./rfc2543">RFC 2543</a>),
      as they are now the fundamental building blocks of dialog
      identification.

   o  Added the Supported header field, allowing for clients to indicate
      what extensions are supported to a server, which can apply those
      extensions to the response, and indicate their usage with a
      Require in the response.

   o  Extension parameters were missing from the BNF for several header
      fields, and they have been added.

   o  Handling of Route and Record-Route construction was very
      underspecified in <a href="./rfc2543">RFC 2543</a>, and also not the right approach.  It
      has been substantially reworked in this specification (and made
      vastly simpler), and this is arguably the largest change.
      Backwards compatibility is still provided for deployments that do
      not use "pre-loaded routes", where the initial request has a set
      of Route header field values obtained in some way outside of
      Record-Route.  In those situations, the new mechanism is not
      interoperable.

   o  In <a href="./rfc2543">RFC 2543</a>, lines in a message could be terminated with CR, LF,
      or CRLF.  This specification only allows CRLF.








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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-258" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   o  Usage of Route in CANCEL and ACK was not well defined in <a href="./rfc2543">RFC 2543</a>.
      It is now well specified; if a request had a Route header field,
      its CANCEL or ACK for a non-2xx response to the request need to
      carry the same Route header field values.  ACKs for 2xx responses
      use the Route values learned from the Record-Route of the 2xx
      responses.

   o  <a href="./rfc2543">RFC 2543</a> allowed multiple requests in a single UDP packet.  This
      usage has been removed.

   o  Usage of absolute time in the Expires header field and parameter
      has been removed.  It caused interoperability problems in elements
      that were not time synchronized, a common occurrence.  Relative
      times are used instead.

   o  The branch parameter of the Via header field value is now
      mandatory for all elements to use.  It now plays the role of a
      unique transaction identifier.  This avoids the complex and bug-
      laden transaction identification rules from <a href="./rfc2543">RFC 2543</a>.  A magic
      cookie is used in the parameter value to determine if the previous
      hop has made the parameter globally unique, and comparison falls
      back to the old rules when it is not present.  Thus,
      interoperability is assured.

   o  In <a href="./rfc2543">RFC 2543</a>, closure of a TCP connection was made equivalent to a
      CANCEL.  This was nearly impossible to implement (and wrong) for
      TCP connections between proxies.  This has been eliminated, so
      that there is no coupling between TCP connection state and SIP
      processing.

   o  <a href="./rfc2543">RFC 2543</a> was silent on whether a UA could initiate a new
      transaction to a peer while another was in progress.  That is now
      specified here.  It is allowed for non-INVITE requests, disallowed
      for INVITE.

   o  PGP was removed.  It was not sufficiently specified, and not
      compatible with the more complete PGP MIME.  It was replaced with
      S/MIME.

   o  Added the "sips" URI scheme for end-to-end TLS.  This scheme is
      not backwards compatible with <a href="./rfc2543">RFC 2543</a>.  Existing elements that
      receive a request with a SIPS URI scheme in the Request-URI will
      likely reject the request.  This is actually a feature; it ensures
      that a call to a SIPS URI is only delivered if all path hops can
      be secured.






<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 258]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-259" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   o  Additional security features were added with TLS, and these are
      described in a much larger and complete security considerations
      section.

   o  In <a href="./rfc2543">RFC 2543</a>, a proxy was not required to forward provisional
      responses from 101 to 199 upstream.  This was changed to MUST.
      This is important, since many subsequent features depend on
      delivery of all provisional responses from 101 to 199.

   o  Little was said about the 503 response code in <a href="./rfc2543">RFC 2543</a>.  It has
      since found substantial use in indicating failure or overload
      conditions in proxies.  This requires somewhat special treatment.
      Specifically, receipt of a 503 should trigger an attempt to
      contact the next element in the result of a DNS SRV lookup.  Also,
      503 response is only forwarded upstream by a proxy under certain
      conditions.

   o  <a href="./rfc2543">RFC 2543</a> defined, but did no sufficiently specify, a mechanism for
      UA authentication of a server.  That has been removed.  Instead,
      the mutual authentication procedures of <a href="./rfc2617">RFC 2617</a> are allowed.

   o  A UA cannot send a BYE for a call until it has received an ACK for
      the initial INVITE.  This was allowed in <a href="./rfc2543">RFC 2543</a> but leads to a
      potential race condition.

   o  A UA or proxy cannot send CANCEL for a transaction until it gets a
      provisional response for the request.  This was allowed in <a href="./rfc2543">RFC</a>
      <a href="./rfc2543">2543</a> but leads to potential race conditions.

   o  The action parameter in registrations has been deprecated.  It was
      insufficient for any useful services, and caused conflicts when
      application processing was applied in proxies.

   o  <a href="./rfc2543">RFC 2543</a> had a number of special cases for multicast.  For
      example, certain responses were suppressed, timers were adjusted,
      and so on.  Multicast now plays a more limited role, and the
      protocol operation is unaffected by usage of multicast as opposed
      to unicast.  The limitations as a result of that are documented.

   o  Basic authentication has been removed entirely and its usage
      forbidden.










<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 259]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-260" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   o  Proxies no longer forward a 6xx immediately on receiving it.
      Instead, they CANCEL pending branches immediately.  This avoids a
      potential race condition that would result in a UAC getting a 6xx
      followed by a 2xx.  In all cases except this race condition, the
      result will be the same - the 6xx is forwarded upstream.

   o  <a href="./rfc2543">RFC 2543</a> did not address the problem of request merging.  This
      occurs when a request forks at a proxy and later rejoins at an
      element.  Handling of merging is done only at a UA, and procedures
      are defined for rejecting all but the first request.

<span class="h3"><a class="selflink" id="section-28.2" href="#section-28.2">28.2</a> Minor Functional Changes</span>

   o  Added the Alert-Info, Error-Info, and Call-Info header fields for
      optional content presentation to users.

   o  Added the Content-Language, Content-Disposition and MIME-Version
      header fields.

   o  Added a "glare handling" mechanism to deal with the case where
      both parties send each other a re-INVITE simultaneously.  It uses
      the new 491 (Request Pending) error code.

   o  Added the In-Reply-To and Reply-To header fields for supporting
      the return of missed calls or messages at a later time.

   o  Added TLS and SCTP as valid SIP transports.

   o  There were a variety of mechanisms described for handling failures
      at any time during a call; those are now generally unified.  BYE
      is sent to terminate.

   o  <a href="./rfc2543">RFC 2543</a> mandated retransmission of INVITE responses over TCP, but
      noted it was really only needed for 2xx.  That was an artifact of
      insufficient protocol layering.  With a more coherent transaction
      layer defined here, that is no longer needed.  Only 2xx responses
      to INVITEs are retransmitted over TCP.

   o  Client and server transaction machines are now driven based on
      timeouts rather than retransmit counts.  This allows the state
      machines to be properly specified for TCP and UDP.

   o  The Date header field is used in REGISTER responses to provide a
      simple means for auto-configuration of dates in user agents.

   o  Allowed a registrar to reject registrations with expirations that
      are too short in duration.  Defined the 423 response code and the
      Min-Expires for this purpose.



<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 260]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-261" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


<span class="h2"><a class="selflink" id="section-29" href="#section-29">29</a> Normative References</span>

   [<a id="ref-1">1</a>]  Handley, M. and V. Jacobson, "SDP: Session Description
        Protocol", <a href="./rfc2327">RFC 2327</a>, April 1998.

   [<a id="ref-2">2</a>]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-3">3</a>]  Resnick, P., "Internet Message Format", <a href="./rfc2822">RFC 2822</a>, April 2001.

   [<a id="ref-4">4</a>]  Rosenberg, J. and H. Schulzrinne, "SIP: Locating SIP Servers",
        <a href="./rfc3263">RFC 3263</a>, June 2002.

   [<a id="ref-5">5</a>]  Berners-Lee, T., Fielding, R. and L. Masinter, "Uniform Resource
        Identifiers (URI): Generic Syntax", <a href="./rfc2396">RFC 2396</a>, August 1998.

   [<a id="ref-6">6</a>]  Chown, P., "Advanced Encryption Standard (AES) Ciphersuites for
        Transport Layer Security (TLS)", <a href="./rfc3268">RFC 3268</a>, June 2002.

   [<a id="ref-7">7</a>]  Yergeau, F., "UTF-8, a transformation format of ISO 10646", <a href="./rfc2279">RFC</a>
        <a href="./rfc2279">2279</a>, January 1998.

   [<a id="ref-8">8</a>]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H., Masinter, L.,
        Leach, P. and T. Berners-Lee, "Hypertext Transfer Protocol --
        HTTP/1.1", <a href="./rfc2616">RFC 2616</a>, June 1999.

   [<a id="ref-9">9</a>]  Vaha-Sipila, A., "URLs for Telephone Calls", <a href="./rfc2806">RFC 2806</a>, April
        2000.

   [<a id="ref-10">10</a>] Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", <a href="./rfc2234">RFC 2234</a>, November 1997.

   [<a id="ref-11">11</a>] Freed, F. and N. Borenstein, "Multipurpose Internet Mail
        Extensions (MIME) Part Two: Media Types", <a href="./rfc2046">RFC 2046</a>, November
        1996.

   [<a id="ref-12">12</a>] Eastlake, D., Crocker, S. and J. Schiller, "Randomness
        Recommendations for Security", <a href="./rfc1750">RFC 1750</a>, December 1994.

   [<a id="ref-13">13</a>] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
        SDP", <a href="./rfc3264">RFC 3264</a>, June 2002.

   [<a id="ref-14">14</a>] Postel, J., "User Datagram Protocol", STD 6, <a href="./rfc768">RFC 768</a>, August
        1980.

   [<a id="ref-15">15</a>] Postel, J., "DoD Standard Transmission Control Protocol", <a href="./rfc761">RFC</a>
        <a href="./rfc761">761</a>, January 1980.




<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 261]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-262" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   [<a id="ref-16">16</a>] Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer,
        H., Taylor, T., Rytina, I., Kalla, M., Zhang, L. and V. Paxson,
        "Stream Control Transmission Protocol", <a href="./rfc2960">RFC 2960</a>, October 2000.

   [<a id="ref-17">17</a>] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
        Leach, P., Luotonen, A. and L. Stewart, "HTTP authentication:
        Basic and Digest Access Authentication", <a href="./rfc2617">RFC 2617</a>, June 1999.

   [<a id="ref-18">18</a>] Troost, R., Dorner, S. and K. Moore, "Communicating Presentation
        Information in Internet Messages: The Content-Disposition Header
        Field", <a href="./rfc2183">RFC 2183</a>, August 1997.

   [<a id="ref-19">19</a>] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F.,
        Watson, M. and M. Zonoun, "MIME media types for ISUP and QSIG
        Objects", <a href="./rfc3204">RFC 3204</a>, December 2001.

   [<a id="ref-20">20</a>] Braden, R., "Requirements for Internet Hosts - Application and
        Support", STD 3, <a href="./rfc1123">RFC 1123</a>, October 1989.

   [<a id="ref-21">21</a>] Alvestrand, H., "IETF Policy on Character Sets and Languages",
        <a href="https://www.rfc-editor.org/bcp/bcp18">BCP 18</a>, <a href="./rfc2277">RFC 2277</a>, January 1998.

   [<a id="ref-22">22</a>] Galvin, J., Murphy, S., Crocker, S. and N. Freed, "Security
        Multiparts for MIME: Multipart/Signed and Multipart/Encrypted",
        <a href="./rfc1847">RFC 1847</a>, October 1995.

   [<a id="ref-23">23</a>] Housley, R., "Cryptographic Message Syntax", <a href="./rfc2630">RFC 2630</a>, June
        1999.

   [<a id="ref-24">24</a>] Ramsdell B., "S/MIME Version 3 Message Specification", <a href="./rfc2633">RFC 2633</a>,
        June 1999.

   [<a id="ref-25">25</a>] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", <a href="./rfc2246">RFC</a>
        <a href="./rfc2246">2246</a>, January 1999.

   [<a id="ref-26">26</a>] Kent, S. and R. Atkinson, "Security Architecture for the
        Internet Protocol", <a href="./rfc2401">RFC 2401</a>, November 1998.

<span class="h2"><a class="selflink" id="section-30" href="#section-30">30</a> Informative References</span>

   [<a id="ref-27">27</a>] R. Pandya, "Emerging mobile and personal communication systems,"
        IEEE Communications Magazine, Vol. 33, pp. 44--52, June 1995.

   [<a id="ref-28">28</a>] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
        "RTP:  A Transport Protocol for Real-Time Applications", <a href="./rfc1889">RFC</a>
        <a href="./rfc1889">1889</a>, January 1996.





<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 262]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-263" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   [<a id="ref-29">29</a>] Schulzrinne, H., Rao, R. and R. Lanphier, "Real Time Streaming
        Protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.

   [<a id="ref-30">30</a>] Cuervo, F., Greene, N., Rayhan, A., Huitema, C., Rosen, B. and
        J. Segers, "Megaco Protocol Version 1.0", <a href="./rfc3015">RFC 3015</a>, November
        2000.

   [<a id="ref-31">31</a>] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
        "SIP: Session Initiation Protocol", <a href="./rfc2543">RFC 2543</a>, March 1999.

   [<a id="ref-32">32</a>] Hoffman, P., Masinter, L. and J. Zawinski, "The mailto URL
        scheme", <a href="./rfc2368">RFC 2368</a>, July 1998.

   [<a id="ref-33">33</a>] E. M. Schooler, "A multicast user directory service for
        synchronous rendezvous," Master's Thesis CS-TR-96-18, Department
        of Computer Science, California Institute of Technology,
        Pasadena, California, Aug. 1996.

   [<a id="ref-34">34</a>] Donovan, S., "The SIP INFO Method", <a href="./rfc2976">RFC 2976</a>, October 2000.

   [<a id="ref-35">35</a>] Rivest, R., "The MD5 Message-Digest Algorithm", <a href="./rfc1321">RFC 1321</a>, April
        1992.

   [<a id="ref-36">36</a>] Dawson, F. and T. Howes, "vCard MIME Directory Profile", <a href="./rfc2426">RFC</a>
        <a href="./rfc2426">2426</a>, September 1998.

   [<a id="ref-37">37</a>] Good, G., "The LDAP Data Interchange Format (LDIF) - Technical
        Specification", <a href="./rfc2849">RFC 2849</a>, June 2000.

   [<a id="ref-38">38</a>] Palme, J., "Common Internet Message Headers",  <a href="./rfc2076">RFC 2076</a>,
        February 1997.

   [<a id="ref-39">39</a>] Franks, J., Hallam-Baker, P., Hostetler, J., Leach, P.,
        Luotonen, A., Sink, E. and L. Stewart, "An Extension to HTTP:
        Digest Access Authentication", <a href="./rfc2069">RFC 2069</a>, January 1997.

   [<a id="ref-40">40</a>] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., Willis,
        D., Rosenberg, J., Summers, K. and H. Schulzrinne, "SIP Call
        Flow Examples", Work in Progress.

   [<a id="ref-41">41</a>] E. M. Schooler, "Case study: multimedia conference control in a
        packet-switched teleconferencing system," Journal of
        Internetworking:  Research and Experience, Vol. 4, pp. 99--120,
        June 1993.  ISI reprint series ISI/RS-93-359.







<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 263]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-264" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   [<a id="ref-42">42</a>] H. Schulzrinne, "Personal mobility for multimedia services in
        the Internet," in European Workshop on Interactive Distributed
        Multimedia Systems and Services (IDMS), (Berlin, Germany), Mar.
        1996.

   [<a id="ref-43">43</a>] Floyd, S., "Congestion Control Principles", <a href="./rfc2914">RFC 2914</a>, September
        2000.












































<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 264]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-265" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


A Table of Timer Values

   Table 4 summarizes the meaning and defaults of the various timers
   used by this specification.

Timer    Value            Section               Meaning
----------------------------------------------------------------------
T1       500ms default    <a href="#section-17.1.1.1">Section 17.1.1.1</a>     RTT Estimate
T2       4s               <a href="#section-17.1.2.2">Section 17.1.2.2</a>     The maximum retransmit
                                               interval for non-INVITE
                                               requests and INVITE
                                               responses
T4       5s               <a href="#section-17.1.2.2">Section 17.1.2.2</a>     Maximum duration a
                                               message will
                                               remain in the network
Timer A  initially T1     <a href="#section-17.1.1.2">Section 17.1.1.2</a>     INVITE request retransmit
                                               interval, for UDP only
Timer B  64*T1            <a href="#section-17.1.1.2">Section 17.1.1.2</a>     INVITE transaction
                                               timeout timer
Timer C  &gt; 3min           <a href="#section-16.6">Section 16.6</a>         proxy INVITE transaction
                           bullet 11            timeout
Timer D  &gt; 32s for UDP    <a href="#section-17.1.1.2">Section 17.1.1.2</a>     Wait time for response
         0s for TCP/SCTP                       retransmits
Timer E  initially T1     <a href="#section-17.1.2.2">Section 17.1.2.2</a>     non-INVITE request
                                               retransmit interval,
                                               UDP only
Timer F  64*T1            <a href="#section-17.1.2.2">Section 17.1.2.2</a>     non-INVITE transaction
                                               timeout timer
Timer G  initially T1     <a href="#section-17.2.1">Section 17.2.1</a>       INVITE response
                                               retransmit interval
Timer H  64*T1            <a href="#section-17.2.1">Section 17.2.1</a>       Wait time for
                                               ACK receipt
Timer I  T4 for UDP       <a href="#section-17.2.1">Section 17.2.1</a>       Wait time for
         0s for TCP/SCTP                       ACK retransmits
Timer J  64*T1 for UDP    <a href="#section-17.2.2">Section 17.2.2</a>       Wait time for
         0s for TCP/SCTP                       non-INVITE request
                                               retransmits
Timer K  T4 for UDP       <a href="#section-17.1.2.2">Section 17.1.2.2</a>     Wait time for
         0s for TCP/SCTP                       response retransmits

                   Table 4: Summary of timers










<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 265]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-266" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Acknowledgments

   We wish to thank the members of the IETF MMUSIC and SIP WGs for their
   comments and suggestions.  Detailed comments were provided by Ofir
   Arkin, Brian Bidulock, Jim Buller, Neil Deason, Dave Devanathan,
   Keith Drage, Bill Fenner, Cedric Fluckiger, Yaron Goland, John
   Hearty, Bernie Hoeneisen, Jo Hornsby, Phil Hoffer, Christian Huitema,
   Hisham Khartabil, Jean Jervis, Gadi Karmi, Peter Kjellerstedt, Anders
   Kristensen, Jonathan Lennox, Gethin Liddell, Allison Mankin, William
   Marshall, Rohan Mahy, Keith Moore, Vern Paxson, Bob Penfield, Moshe
   J. Sambol, Chip Sharp, Igor Slepchin, Eric Tremblay, and Rick
   Workman.

   Brian Rosen provided the compiled BNF.

   Jean Mahoney provided technical writing assistance.

   This work is based, inter alia, on [<a href="#ref-41" title="&quot;Case study: multimedia conference control in a packet-switched teleconferencing system,&quot;">41</a>,<a href="#ref-42" title="&quot;Personal mobility for multimedia services in the Internet,&quot;">42</a>].

































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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Authors' Addresses

   Authors addresses are listed alphabetically for the editors, the
   writers, and then the original authors of <a href="./rfc2543">RFC 2543</a>.  All listed
   authors actively contributed large amounts of text to this document.

   Jonathan Rosenberg
   dynamicsoft
   72 Eagle Rock Ave
   East Hanover, NJ 07936
   USA

   EMail:  jdrosen@dynamicsoft.com


   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA

   EMail:  schulzrinne@cs.columbia.edu


   Gonzalo Camarillo
   Ericsson
   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland

   EMail:  Gonzalo.Camarillo@ericsson.com


   Alan Johnston
   WorldCom
   100 South 4th Street
   St. Louis, MO 63102
   USA

   EMail:  alan.johnston@wcom.com










<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 267]</span></pre>
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<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


   Jon Peterson
   NeuStar, Inc
   1800 Sutter Street, Suite 570
   Concord, CA 94520
   USA

   EMail:  jon.peterson@neustar.com


   Robert Sparks
   dynamicsoft, Inc.
   5100 Tennyson Parkway
   Suite 1200
   Plano, Texas 75024
   USA

   EMail:  rsparks@dynamicsoft.com


   Mark Handley
   International Computer Science Institute
   1947 Center St, Suite 600
   Berkeley, CA 94704
   USA

   EMail:  mjh@icir.org


   Eve Schooler
   AT&amp;T Labs-Research
   75 Willow Road
   Menlo Park, CA 94025
   USA

   EMail: schooler@research.att.com
















<span class="grey">Rosenberg, et. al.          Standards Track                   [Page 268]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-269" ></span>
<span class="grey"><a href="./rfc3261">RFC 3261</a>            SIP: Session Initiation Protocol           June 2002</span>


Full Copyright Statement

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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</pre>