<pre>Network Working Group                                   C. Jennings, Ed.
Request for Comments: 5626                                 Cisco Systems
Updates: <a href="./rfc3261">3261</a>, <a href="./rfc3327">3327</a>                                         R. Mahy, Ed.
Category: Standards Track                                   Unaffiliated
                                                           F. Audet, Ed.
                                                              Skype Labs
                                                            October 2009


                 <span class="h1">Managing Client-Initiated Connections</span>
                <span class="h1">in the Session Initiation Protocol (SIP)</span>

Abstract

   The Session Initiation Protocol (SIP) allows proxy servers to
   initiate TCP connections or to send asynchronous UDP datagrams to
   User Agents in order to deliver requests.  However, in a large number
   of real deployments, many practical considerations, such as the
   existence of firewalls and Network Address Translators (NATs) or the
   use of TLS with server-provided certificates, prevent servers from
   connecting to User Agents in this way.  This specification defines
   behaviors for User Agents, registrars, and proxy servers that allow
   requests to be delivered on existing connections established by the
   User Agent.  It also defines keep-alive behaviors needed to keep NAT
   bindings open and specifies the usage of multiple connections from
   the User Agent to its registrar.

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) 2009 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in <a href="#section-4">Section 4</a>.e of




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   the Trust Legal Provisions and are provided without warranty as
   described in the BSD License.

   This document may contain material from IETF Documents or IETF
   Contributions published or made publicly available before November
   10, 2008.  The person(s) controlling the copyright in some of this
   material may not have granted the IETF Trust the right to allow
   modifications of such material outside the IETF Standards Process.
   Without obtaining an adequate license from the person(s) controlling
   the copyright in such materials, this document may not be modified
   outside the IETF Standards Process, and derivative works of it may
   not be created outside the IETF Standards Process, except to format
   it for publication as an RFC or to translate it into languages other
   than English.

Table of Contents

   <a href="#section-1">1</a>.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-4">4</a>
   <a href="#section-2">2</a>.  Conventions and Terminology  . . . . . . . . . . . . . . . . .  <a href="#page-5">5</a>
     <a href="#section-2.1">2.1</a>.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-5">5</a>
   <a href="#section-3">3</a>.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-6">6</a>
     <a href="#section-3.1">3.1</a>.  Summary of Mechanism . . . . . . . . . . . . . . . . . . .  <a href="#page-6">6</a>
     <a href="#section-3.2">3.2</a>.  Single Registrar and UA  . . . . . . . . . . . . . . . . .  <a href="#page-7">7</a>
     <a href="#section-3.3">3.3</a>.  Multiple Connections from a User Agent . . . . . . . . . .  <a href="#page-8">8</a>
     <a href="#section-3.4">3.4</a>.  Edge Proxies . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-10">10</a>
     <a href="#section-3.5">3.5</a>.  Keep-Alive Technique . . . . . . . . . . . . . . . . . . . <a href="#page-11">11</a>
       <a href="#section-3.5.1">3.5.1</a>.  CRLF Keep-Alive Technique  . . . . . . . . . . . . . . <a href="#page-12">12</a>
       <a href="#section-3.5.2">3.5.2</a>.  STUN Keep-Alive Technique  . . . . . . . . . . . . . . <a href="#page-12">12</a>
   <a href="#section-4">4</a>.  User Agent Procedures  . . . . . . . . . . . . . . . . . . . . <a href="#page-13">13</a>
     <a href="#section-4.1">4.1</a>.  Instance ID Creation . . . . . . . . . . . . . . . . . . . <a href="#page-13">13</a>
     <a href="#section-4.2">4.2</a>.  Registrations  . . . . . . . . . . . . . . . . . . . . . . <a href="#page-14">14</a>
       <a href="#section-4.2.1">4.2.1</a>.  Initial Registrations  . . . . . . . . . . . . . . . . <a href="#page-14">14</a>
       <a href="#section-4.2.2">4.2.2</a>.  Subsequent REGISTER Requests . . . . . . . . . . . . . <a href="#page-16">16</a>
       <a href="#section-4.2.3">4.2.3</a>.  Third-Party Registrations  . . . . . . . . . . . . . . <a href="#page-17">17</a>
     <a href="#section-4.3">4.3</a>.  Sending Non-REGISTER Requests  . . . . . . . . . . . . . . <a href="#page-17">17</a>
     <a href="#section-4.4">4.4</a>.  Keep-Alives and Detecting Flow Failure . . . . . . . . . . <a href="#page-18">18</a>
       <a href="#section-4.4.1">4.4.1</a>.  Keep-Alive with CRLF . . . . . . . . . . . . . . . . . <a href="#page-19">19</a>
       <a href="#section-4.4.2">4.4.2</a>.  Keep-Alive with STUN . . . . . . . . . . . . . . . . . <a href="#page-21">21</a>
     <a href="#section-4.5">4.5</a>.  Flow Recovery  . . . . . . . . . . . . . . . . . . . . . . <a href="#page-21">21</a>
   <a href="#section-5">5</a>.  Edge Proxy Procedures  . . . . . . . . . . . . . . . . . . . . <a href="#page-22">22</a>
     <a href="#section-5.1">5.1</a>.  Processing Register Requests . . . . . . . . . . . . . . . <a href="#page-22">22</a>
     <a href="#section-5.2">5.2</a>.  Generating Flow Tokens . . . . . . . . . . . . . . . . . . <a href="#page-23">23</a>
     <a href="#section-5.3">5.3</a>.  Forwarding Non-REGISTER Requests . . . . . . . . . . . . . <a href="#page-23">23</a>
       <a href="#section-5.3.1">5.3.1</a>.  Processing Incoming Requests . . . . . . . . . . . . . <a href="#page-24">24</a>
       <a href="#section-5.3.2">5.3.2</a>.  Processing Outgoing Requests . . . . . . . . . . . . . <a href="#page-24">24</a>
     <a href="#section-5.4">5.4</a>.  Edge Proxy Keep-Alive Handling . . . . . . . . . . . . . . <a href="#page-25">25</a>
   <a href="#section-6">6</a>.  Registrar Procedures . . . . . . . . . . . . . . . . . . . . . <a href="#page-25">25</a>
   <a href="#section-7">7</a>.  Authoritative Proxy Procedures: Forwarding Requests  . . . . . <a href="#page-27">27</a>



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   <a href="#section-8">8</a>.  STUN Keep-Alive Processing . . . . . . . . . . . . . . . . . . <a href="#page-28">28</a>
     <a href="#section-8.1">8.1</a>.  Use with SigComp . . . . . . . . . . . . . . . . . . . . . <a href="#page-29">29</a>
   <a href="#section-9">9</a>.  Example Message Flow . . . . . . . . . . . . . . . . . . . . . <a href="#page-30">30</a>
     <a href="#section-9.1">9.1</a>.  Subscription to Configuration Package  . . . . . . . . . . <a href="#page-30">30</a>
     <a href="#section-9.2">9.2</a>.  Registration . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-32">32</a>
     <a href="#section-9.3">9.3</a>.  Incoming Call and Proxy Crash  . . . . . . . . . . . . . . <a href="#page-34">34</a>
     <a href="#section-9.4">9.4</a>.  Re-Registration  . . . . . . . . . . . . . . . . . . . . . <a href="#page-37">37</a>
     <a href="#section-9.5">9.5</a>.  Outgoing Call  . . . . . . . . . . . . . . . . . . . . . . <a href="#page-38">38</a>
   <a href="#section-10">10</a>. Grammar  . . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
   <a href="#section-11">11</a>. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
     <a href="#section-11.1">11.1</a>. Flow-Timer Header Field  . . . . . . . . . . . . . . . . . <a href="#page-40">40</a>
     <a href="#section-11.2">11.2</a>. "reg-id" Contact Header Field Parameter  . . . . . . . . . <a href="#page-40">40</a>
     <a href="#section-11.3">11.3</a>. SIP/SIPS URI Parameters  . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
     <a href="#section-11.4">11.4</a>. SIP Option Tag . . . . . . . . . . . . . . . . . . . . . . <a href="#page-41">41</a>
     <a href="#section-11.5">11.5</a>. 430 (Flow Failed) Response Code  . . . . . . . . . . . . . <a href="#page-41">41</a>
     <a href="#section-11.6">11.6</a>. 439 (First Hop Lacks Outbound Support) Response Code . . . <a href="#page-42">42</a>
     <a href="#section-11.7">11.7</a>. Media Feature Tag  . . . . . . . . . . . . . . . . . . . . <a href="#page-42">42</a>
   <a href="#section-12">12</a>. Security Considerations  . . . . . . . . . . . . . . . . . . . <a href="#page-43">43</a>
   <a href="#section-13">13</a>. Operational Notes on Transports  . . . . . . . . . . . . . . . <a href="#page-44">44</a>
   <a href="#section-14">14</a>. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-44">44</a>
   <a href="#section-15">15</a>. Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
   <a href="#section-16">16</a>. References . . . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
     <a href="#section-16.1">16.1</a>. Normative References . . . . . . . . . . . . . . . . . . . <a href="#page-45">45</a>
     <a href="#section-16.2">16.2</a>. Informative References . . . . . . . . . . . . . . . . . . <a href="#page-47">47</a>
   <a href="#appendix-A">Appendix A</a>.  Default Flow Registration Backoff Times . . . . . . . <a href="#page-49">49</a>
   <a href="#appendix-B">Appendix B</a>.  ABNF  . . . . . . . . . . . . . . . . . . . . . . . . <a href="#page-49">49</a>

























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

   There are many environments for SIP [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] deployments in which
   the User Agent (UA) can form a connection to a registrar or proxy but
   in which connections in the reverse direction to the UA are not
   possible.  This can happen for several reasons, but the most likely
   is a NAT or a firewall in between the SIP UA and the proxy.  Many
   such devices will only allow outgoing connections.  This
   specification allows a SIP User Agent behind such a firewall or NAT
   to receive inbound traffic associated with registrations or dialogs
   that it initiates.

   Most IP phones and personal computers get their network
   configurations dynamically via a protocol such as the Dynamic Host
   Configuration Protocol (DHCP) [<a href="./rfc2131" title="&quot;Dynamic Host Configuration Protocol&quot;">RFC2131</a>].  These systems typically do
   not have a useful name in the Domain Name System (DNS) [<a href="./rfc1035" title="&quot;Domain names - implementation and specification&quot;">RFC1035</a>], and
   they almost never have a long-term, stable DNS name that is
   appropriate for use in the subjectAltName of a certificate, as
   required by [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  However, these systems can still act as a
   Transport Layer Security (TLS) [<a href="./rfc5246" title="&quot;The Transport Layer Security (TLS) Protocol Version 1.2&quot;">RFC5246</a>] client and form outbound
   connections to a proxy or registrar that authenticates with a server
   certificate.  The server can authenticate the UA using a shared
   secret in a digest challenge (as defined in <a href="./rfc3261#section-22">Section&nbsp;22 of RFC 3261</a>)
   over that TLS connection.  This specification allows a SIP User Agent
   who has to initiate the TLS connection to receive inbound traffic
   associated with registrations or dialogs that it initiates.

   The key idea of this specification is that when a UA sends a REGISTER
   request or a dialog-forming request, the proxy can later use this
   same network "flow" -- whether this is a bidirectional stream of UDP
   datagrams, a TCP connection, or an analogous concept in another
   transport protocol -- to forward any incoming requests that need to
   go to this UA in the context of the registration or dialog.

   For a UA to receive incoming requests, the UA has to connect to a
   server.  Since the server can't connect to the UA, the UA has to make
   sure that a flow is always active.  This requires the UA to detect
   when a flow fails.  Since such detection takes time and leaves a
   window of opportunity for missed incoming requests, this mechanism
   allows the UA to register over multiple flows at the same time.  This
   specification also defines two keep-alive schemes.  The keep-alive
   mechanism is used to keep NAT bindings fresh, and to allow the UA to
   detect when a flow has failed.








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<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Conventions and Terminology</span>

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [<a href="./rfc2119" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">RFC2119</a>].

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Definitions</span>

   Authoritative Proxy:  A proxy that handles non-REGISTER requests for
      a specific Address-of-Record (AOR), performs the logical Location
      Server lookup described in [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], and forwards those requests
      to specific Contact URIs.  (In [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>], the role that is
      authoritative for REGISTER requests for a specific AOR is a
      Registration Server.)

   Edge Proxy:  An edge proxy is any proxy that is located topologically
      between the registering User Agent and the Authoritative Proxy.
      The "first" edge proxy refers to the first edge proxy encountered
      when a UA sends a request.

   Flow:  A Flow is a transport-layer association between two hosts that
      is represented by the network address and port number of both ends
      and by the transport protocol.  For TCP, a flow is equivalent to a
      TCP connection.  For UDP a flow is a bidirectional stream of
      datagrams between a single pair of IP addresses and ports of both
      peers.  With TCP, a flow often has a one-to-one correspondence
      with a single file descriptor in the operating system.

   Flow Token:  An identifier that uniquely identifies a flow which can
      be included in a SIP URI (Uniform Resource Identifier [<a href="./rfc3986" title="&quot;Uniform Resource Identifier (URI): Generic Syntax&quot;">RFC3986</a>]).

   reg-id:  This refers to the value of a new header field parameter
      value for the Contact header field.  When a UA registers multiple
      times, each for a different flow, each concurrent registration
      gets a unique reg-id value.

   instance-id:  This specification uses the word instance-id to refer
      to the value of the "sip.instance" media feature tag which appears
      as a "+sip.instance" Contact header field parameter.  This is a
      Uniform Resource Name (URN) that uniquely identifies this specific
      UA instance.

   "ob" Parameter:  The "ob" parameter is a SIP URI parameter that has a
      different meaning depending on context.  In a Path header field
      value, it is used by the first edge proxy to indicate that a flow
      token was added to the URI.  In a Contact or Route header field
      value, it indicates that the UA would like other requests in the
      same dialog to be routed over the same flow.



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   outbound-proxy-set:  A set of SIP URIs (Uniform Resource Identifiers)
      that represents each of the outbound proxies (often edge proxies)
      with which the UA will attempt to maintain a direct flow.  The
      first URI in the set is often referred to as the primary outbound
      proxy and the second as the secondary outbound proxy.  There is no
      difference between any of the URIs in this set, nor does the
      primary/secondary terminology imply that one is preferred over the
      other.

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

   The mechanisms defined in this document are useful in several
   scenarios discussed below, including the simple co-located registrar
   and proxy, a User Agent desiring multiple connections to a resource
   (for redundancy, for example), and a system that uses edge proxies.

   This entire section is non-normative.

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Summary of Mechanism</span>

   Each UA has a unique instance-id that stays the same for this UA even
   if the UA reboots or is power cycled.  Each UA can register multiple
   times over different flows for the same SIP Address of Record (AOR)
   to achieve high reliability.  Each registration includes the
   instance-id for the UA and a reg-id label that is different for each
   flow.  The registrar can use the instance-id to recognize that two
   different registrations both correspond to the same UA.  The
   registrar can use the reg-id label to recognize whether a UA is
   creating a new flow or refreshing or replacing an old one, possibly
   after a reboot or a network failure.

   When a proxy goes to route a message to a UA for which it has a
   binding, it can use any one of the flows on which a successful
   registration has been completed.  A failure to deliver a request on a
   particular flow can be tried again on an alternate flow.  Proxies can
   determine which flows go to the same UA by comparing the instance-id.
   Proxies can tell that a flow replaces a previously abandoned flow by
   looking at the reg-id.

   When sending a dialog-forming request, a UA can also ask its first
   edge proxy to route subsequent requests in that dialog over the same
   flow.  This is necessary whether the UA has registered or not.

   UAs use a simple periodic message as a keep-alive mechanism to keep
   their flow to the proxy or registrar alive.  For connection-oriented
   transports such as TCP this is based on carriage-return and line-feed





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   sequences (CRLF), while for transports that are not connection
   oriented, this is accomplished by using a SIP-specific usage profile
   of STUN (Session Traversal Utilities for NAT) [<a href="./rfc5389" title="&quot;Session Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>].

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  Single Registrar and UA</span>

   In the topology shown below, a single server is acting as both a
   registrar and proxy.

      +-----------+
      | Registrar |
      | Proxy     |
      +-----+-----+
            |
            |
       +----+--+
       | User  |
       | Agent |
       +-------+

   User Agents that form only a single flow continue to register
   normally but include the instance-id as described in <a href="#section-4.1">Section 4.1</a>.
   The UA also includes a "reg-id" Contact header field parameter that
   is used to allow the registrar to detect and avoid keeping invalid
   contacts when a UA reboots or reconnects after its old connection has
   failed for some reason.

   For clarity, here is an example.  Bob's UA creates a new TCP flow to
   the registrar and sends the following REGISTER request.

   REGISTER sip:example.com SIP/2.0
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bK-bad0ce-11-1036
   Max-Forwards: 70
   From: Bob &lt;sip:bob@example.com&gt;;tag=d879h76
   To: Bob &lt;sip:bob@example.com&gt;
   Call-ID: 8921348ju72je840.204
   CSeq: 1 REGISTER
   Supported: path, outbound
   Contact: &lt;sip:line1@192.0.2.2;transport=tcp&gt;; reg-id=1;
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-000A95A0E128&gt;"
   Content-Length: 0

   The registrar challenges this registration to authenticate Bob.  When
   the registrar adds an entry for this contact under the AOR for Bob,
   the registrar also keeps track of the connection over which it
   received this registration.





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   The registrar saves the instance-id
   ("urn:uuid:00000000-0000-1000-8000-000A95A0E128") and reg-id ("1")
   along with the rest of the Contact header field.  If the instance-id
   and reg-id are the same as a previous registration for the same AOR,
   the registrar replaces the old Contact URI and flow information.
   This allows a UA that has rebooted to replace its previous
   registration for each flow with minimal impact on overall system
   load.

   When Alice sends a request to Bob, his authoritative proxy selects
   the target set.  The proxy forwards the request to elements in the
   target set based on the proxy's policy.  The proxy looks at the
   target set and uses the instance-id to understand if two targets both
   end up routing to the same UA.  When the proxy goes to forward a
   request to a given target, it looks and finds the flows over which it
   received the registration.  The proxy then forwards the request over
   an existing flow, instead of resolving the Contact URI using the
   procedures in [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>] and trying to form a new flow to that
   contact.

   As described in the next section, if the proxy has multiple flows
   that all go to this UA, the proxy can choose any one of the
   registration bindings for this AOR that has the same instance-id as
   the selected UA.

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Multiple Connections from a User Agent</span>

   There are various ways to deploy SIP to build a reliable and scalable
   system.  This section discusses one such design that is possible with
   the mechanisms in this specification.  Other designs are also
   possible.

   In the example system below, the logical outbound proxy/registrar for
   the domain is running on two hosts that share the appropriate state
   and can both provide registrar and outbound proxy functionality for
   the domain.  The UA will form connections to two of the physical
   hosts that can perform the authoritative proxy/registrar function for
   the domain.  Reliability is achieved by having the UA form two TCP
   connections to the domain.












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       +-------------------+
       | Domain            |
       | Logical Proxy/Reg |
       |                   |
       |+-----+     +-----+|
       ||Host1|     |Host2||
       |+-----+     +-----+|
       +---\------------/--+
            \          /
             \        /
              \      /
               \    /
              +------+
              | User |
              | Agent|
              +------+

   The UA is configured with multiple outbound proxy registration URIs.
   These URIs are configured into the UA through whatever the normal
   mechanism is to configure the proxy address and AOR in the UA.  If
   the AOR is alice@example.com, the outbound-proxy-set might look
   something like "sip:primary.example.com" and "sip:
   secondary.example.com".  Note that each URI in the outbound-proxy-set
   could resolve to several different physical hosts.  The
   administrative domain that created these URIs should ensure that the
   two URIs resolve to separate hosts.  These URIs are handled according
   to normal SIP processing rules, so mechanisms like DNS SRV [<a href="./rfc2782" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">RFC2782</a>]
   can be used to do load-balancing across a proxy farm.  The approach
   in this document does not prevent future extensions, such as the SIP
   UA configuration framework [<a href="#ref-CONFIG-FMWK" title="&quot;A Framework for Session Initiation Protocol User Agent Profile Delivery&quot;">CONFIG-FMWK</a>], from adding other ways for
   a User Agent to discover its outbound-proxy-set.

   The domain also needs to ensure that a request for the UA sent to
   Host1 or Host2 is then sent across the appropriate flow to the UA.
   The domain might choose to use the Path header approach (as described
   in the next section) to store this internal routing information on
   Host1 or Host2.

   When a single server fails, all the UAs that have a flow through it
   will detect a flow failure and try to reconnect.  This can cause
   large loads on the server.  When large numbers of hosts reconnect
   nearly simultaneously, this is referred to as the avalanche restart
   problem, and is further discussed in <a href="#section-4.5">Section 4.5</a>.  The multiple flows
   to many servers help reduce the load caused by the avalanche restart.
   If a UA has multiple flows, and one of the servers fails, the UA
   delays a recommended amount of time before trying to form a new





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   connection to replace the flow to the server that failed.  By
   spreading out the time used for all the UAs to reconnect to a server,
   the load on the server farm is reduced.

   Scalability is achieved by using DNS SRV [<a href="./rfc2782" title="&quot;A DNS RR for specifying the location of services (DNS SRV)&quot;">RFC2782</a>] to load-balance
   the primary connection across a set of machines that can service the
   primary connection, and also using DNS SRV to load-balance across a
   separate set of machines that can service the secondary connection.
   The deployment here requires that DNS is configured with one entry
   that resolves to all the primary hosts and another entry that
   resolves to all the secondary hosts.  While this introduces
   additional DNS configuration, the approach works and requires no
   additional SIP extensions to [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>].

   Another motivation for maintaining multiple flows between the UA and
   its registrar is related to multihomed UAs.  Such UAs can benefit
   from multiple connections from different interfaces to protect
   against the failure of an individual access link.

<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>.  Edge Proxies</span>

   Some SIP deployments use edge proxies such that the UA sends the
   REGISTER to an edge proxy that then forwards the REGISTER to the
   registrar.  There could be a NAT or firewall between the UA and the
   edge proxy.

                +---------+
                |Registrar|
                |Proxy    |
                +---------+
                 /      \
                /        \
               /          \
            +-----+     +-----+
            |Edge1|     |Edge2|
            +-----+     +-----+
               \           /
                \         /
        ----------------------------NAT/FW
                  \     /
                   \   /
                  +------+
                  |User  |
                  |Agent |
                  +------+






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   The edge proxy includes a Path header [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>] so that when the
   proxy/registrar later forwards a request to this UA, the request is
   routed through the edge proxy.

   These systems can use effectively the same mechanism as described in
   the previous sections but need to use the Path header.  When the edge
   proxy receives a registration, it needs to create an identifier value
   that is unique to this flow (and not a subsequent flow with the same
   addresses) and put this identifier in the Path header URI.  This
   identifier has two purposes.  First, it allows the edge proxy to map
   future requests back to the correct flow.  Second, because the
   identifier will only be returned if the user authenticates with the
   registrar successfully, it allows the edge proxy to indirectly check
   the user's authentication information via the registrar.  The
   identifier is placed in the user portion of a loose route in the Path
   header.  If the registration succeeds, the edge proxy needs to map
   future requests (that are routed to the identifier value from the
   Path header) to the associated flow.

   The term edge proxy is often used to refer to deployments where the
   edge proxy is in the same administrative domain as the registrar.
   However, in this specification we use the term to refer to any proxy
   between the UA and the registrar.  For example, the edge proxy may be
   inside an enterprise that requires its use, and the registrar could
   be from a service provider with no relationship to the enterprise.
   Regardless of whether they are in the same administrative domain,
   this specification requires that registrars and edge proxies support
   the Path header mechanism in [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>].

<span class="h3"><a class="selflink" id="section-3.5" href="#section-3.5">3.5</a>.  Keep-Alive Technique</span>

   This document describes two keep-alive mechanisms: a CRLF keep-alive
   and a STUN keep-alive.  Each of these mechanisms uses a client-to-
   server "ping" keep-alive and a corresponding server-to-client "pong"
   message.  This ping-pong sequence allows the client, and optionally
   the server, to tell if its flow is still active and useful for SIP
   traffic.  The server responds to pings by sending pongs.  If the
   client does not receive a pong in response to its ping (allowing for
   retransmission for STUN as described in <a href="#section-4.4.2">Section 4.4.2</a>), it declares
   the flow dead and opens a new flow in its place.

   This document also suggests timer values for these client keep-alive
   mechanisms.  These timer values were chosen to keep most NAT and
   firewall bindings open, to detect unresponsive servers within 2
   minutes, and to mitigate against the avalanche restart problem.
   However, the client may choose different timer values to suit its
   needs, for example to optimize battery life.  In some environments,




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   the server can also keep track of the time since a ping was received
   over a flow to guess the likelihood that the flow is still useful for
   delivering SIP messages.

   When the UA detects that a flow has failed or that the flow
   definition has changed, the UA needs to re-register and will use the
   back-off mechanism described in <a href="#section-4.5">Section 4.5</a> to provide congestion
   relief when a large number of agents simultaneously reboot.

   A keep-alive mechanism needs to keep NAT bindings refreshed; for
   connections, it also needs to detect failure of a connection; and for
   connectionless transports, it needs to detect flow failures including
   changes to the NAT public mapping.  For connection-oriented
   transports such as TCP [<a href="./rfc0793" title="&quot;Transmission Control Protocol&quot;">RFC0793</a>] and SCTP [<a href="./rfc4960" title="&quot;Stream Control Transmission Protocol&quot;">RFC4960</a>], this
   specification describes a keep-alive approach based on sending CRLFs.
   For connectionless transport, such as UDP [<a href="./rfc0768" title="&quot;User Datagram Protocol&quot;">RFC0768</a>], this
   specification describes using STUN [<a href="./rfc5389" title="&quot;Session Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>] over the same flow as
   the SIP traffic to perform the keep-alive.

   UAs and Proxies are also free to use native transport keep-alives;
   however, the application may not be able to set these timers on a
   per-connection basis, and the server certainly cannot make any
   assumption about what values are used.  Use of native transport
   keep-alives is outside the scope of this document.

<span class="h4"><a class="selflink" id="section-3.5.1" href="#section-3.5.1">3.5.1</a>.  CRLF Keep-Alive Technique</span>

   This approach can only be used with connection-oriented transports
   such as TCP or SCTP.  The client periodically sends a double-CRLF
   (the "ping") then waits to receive a single CRLF (the "pong").  If
   the client does not receive a "pong" within an appropriate amount of
   time, it considers the flow failed.

      Note: Sending a CRLF over a connection-oriented transport is
      backwards compatible (because of requirements in <a href="./rfc3261#section-7.5">Section&nbsp;7.5 of
      [RFC3261]</a>), but only implementations which support this
      specification will respond to a "ping" with a "pong".

<span class="h4"><a class="selflink" id="section-3.5.2" href="#section-3.5.2">3.5.2</a>.  STUN Keep-Alive Technique</span>

   This approach can only be used for connection-less transports, such
   as UDP.

   For connection-less transports, a flow definition could change
   because a NAT device in the network path reboots and the resulting
   public IP address or port mapping for the UA changes.  To detect
   this, STUN requests are sent over the same flow that is being used




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   for the SIP traffic.  The proxy or registrar acts as a limited
   Session Traversal Utilities for NAT (STUN) [<a href="./rfc5389" title="&quot;Session Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>] server on the
   SIP signaling port.

      Note: The STUN mechanism is very robust and allows the detection
      of a changed IP address and port.  Many other options were
      considered, but the SIP Working Group selected the STUN-based
      approach.  Approaches using SIP requests were abandoned because
      many believed that good performance and full backwards
      compatibility using this method were mutually exclusive.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  User Agent Procedures</span>

<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a>.  Instance ID Creation</span>

   Each UA MUST have an Instance Identifier Uniform Resource Name (URN)
   [<a href="./rfc2141" title="&quot;URN Syntax&quot;">RFC2141</a>] that uniquely identifies the device.  Usage of a URN
   provides a persistent and unique name for the UA instance.  It also
   provides an easy way to guarantee uniqueness within the AOR.  This
   URN MUST be persistent across power cycles of the device.  The
   instance ID MUST NOT change as the device moves from one network to
   another.

   A UA SHOULD create a Universally Unique Identifier (UUID) URN
   [<a href="./rfc4122" title="&quot;A Universally Unique IDentifier (UUID) URN Namespace&quot;">RFC4122</a>] as its instance-id.  The UUID URN allows for non-
   centralized computation of a URN based on time, unique names (such as
   a MAC address), or a random number generator.

      Note: A device like a "soft phone", when first installed, can
      generate a UUID [<a href="./rfc4122" title="&quot;A Universally Unique IDentifier (UUID) URN Namespace&quot;">RFC4122</a>] and then save this in persistent storage
      for all future use.  For a device such as a "hard phone", which
      will only ever have a single SIP UA present, the UUID can include
      the MAC address and be generated at any time because it is
      guaranteed that no other UUID is being generated at the same time
      on that physical device.  This means the value of the time
      component of the UUID can be arbitrarily selected to be any time
      less than the time when the device was manufactured.  A time of 0
      (as shown in the example in <a href="#section-3.2">Section 3.2</a>) is perfectly legal as
      long as the device knows no other UUIDs were generated at this
      time on this device.

   If a URN scheme other than UUID is used, the UA MUST only use URNs
   for which an RFC (from the IETF stream) defines how the specific URN
   needs to be constructed and used in the "+sip.instance" Contact
   header field parameter for outbound behavior.






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   To convey its instance-id in both requests and responses, the UA
   includes a "sip.instance" media feature tag as a UA characteristic
   [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>].  This media feature tag is encoded in the Contact header
   field as the "+sip.instance" Contact header field parameter.  One
   case where a UA could prefer to omit the "sip.instance" media feature
   tag is when it is making an anonymous request or some other privacy
   concern requires that the UA not reveal its identity.

      Note: [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>] defines equality rules for callee capabilities
      parameters, and according to that specification, the
      "sip.instance" media feature tag will be compared by case-
      sensitive string comparison.  This means that the URN will be
      encapsulated by angle brackets ("&lt;" and "&gt;") when it is placed
      within the quoted string value of the "+sip.instance" Contact
      header field parameter.  The case-sensitive matching rules apply
      only to the generic usages defined in the callee capabilities
      [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>] and the caller preferences [<a href="./rfc3841" title="&quot;Caller Preferences for the Session Initiation Protocol (SIP)&quot;">RFC3841</a>] specifications.
      When the instance ID is used in this specification, it is
      "extracted" from the value in the "sip.instance" media feature
      tag.  Thus, equality comparisons are performed using the rules for
      URN equality that are specific to the scheme in the URN.  If the
      element performing the comparisons does not understand the URN
      scheme, it performs the comparisons using the lexical equality
      rules defined in [<a href="./rfc2141" title="&quot;URN Syntax&quot;">RFC2141</a>].  Lexical equality could result in two
      URNs being considered unequal when they are actually equal.  In
      this specific usage of URNs, the only element that provides the
      URN is the SIP UA instance identified by that URN.  As a result,
      the UA instance has to provide lexically equivalent URNs in each
      registration it generates.  This is likely to be normal behavior
      in any case; clients are not likely to modify the value of the
      instance ID so that it remains functionally equivalent to (yet
      lexicographically different from) previous registrations.

<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a>.  Registrations</span>

<span class="h4"><a class="selflink" id="section-4.2.1" href="#section-4.2.1">4.2.1</a>.  Initial Registrations</span>

   At configuration time, UAs obtain one or more SIP URIs representing
   the default outbound-proxy-set.  This specification assumes the set
   is determined via any of a number of configuration mechanisms, and
   future specifications can define additional mechanisms such as using
   DNS to discover this set.  How the UA is configured is outside the
   scope of this specification.  However, a UA MUST support sets with at
   least two outbound proxy URIs and SHOULD support sets with up to four
   URIs.






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   For each outbound proxy URI in the set, the User Agent Client (UAC)
   SHOULD send a REGISTER request using this URI as the default outbound
   proxy.  (Alternatively, the UA could limit the number of flows formed
   to conserve battery power, for example).  If the set has more than
   one URI, the UAC MUST send a REGISTER request to at least two of the
   default outbound proxies from the set.  UAs that support this
   specification MUST include the outbound option tag in a Supported
   header field in a REGISTER request.  Each of these REGISTER requests
   will use a unique Call-ID.  Forming the route set for the request is
   outside the scope of this document, but typically results in sending
   the REGISTER such that the topmost Route header field contains a
   loose route to the outbound proxy URI.

   REGISTER requests, other than those described in <a href="#section-4.2.3">Section 4.2.3</a>, MUST
   include an instance-id media feature tag as specified in <a href="#section-4.1">Section 4.1</a>.

   A UAC conforming to this specification MUST include in the Contact
   header field, a "reg-id" parameter that is distinct from other
   "reg-id" parameters used in other registrations that use the same
   "+sip.instance" Contact header field parameter and AOR.  Each one of
   these registrations will form a new flow from the UA to the proxy.
   The sequence of reg-id values does not have to be sequential but MUST
   be exactly the same sequence of reg-id values each time the UA
   instance power cycles or reboots, so that the reg-id values will
   collide with the previously used reg-id values.  This is so the
   registrar can replace the older registrations.

      Note: The UAC can situationally decide whether to request outbound
      behavior by including or omitting the "reg-id" Contact header
      field parameter.  For example, imagine the outbound-proxy-set
      contains two proxies in different domains, EP1 and EP2.  If an
      outbound-style registration succeeded for a flow through EP1, the
      UA might decide to include 'outbound' in its Require header field
      when registering with EP2, in order to ensure consistency.
      Similarly, if the registration through EP1 did not support
      outbound, the UA might not register with EP2 at all.

   The UAC MUST support the Path header [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>] mechanism, and
   indicate its support by including the 'path' option-tag in a
   Supported header field value in its REGISTER requests.  Other than
   optionally examining the Path vector in the response, this is all
   that is required of the UAC to support Path.

   The UAC examines successful registration responses for the presence
   of an outbound option-tag in a Require header field value.  Presence
   of this option-tag indicates that the registrar is compliant with
   this specification, and that any edge proxies which needed to
   participate are also compliant.  If the registrar did not support



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   outbound, the UA has potentially registered an un-routable contact.
   It is the responsibility of the UA to remove any inappropriate
   Contacts.

   If outbound registration succeeded, as indicated by the presence of
   the outbound option-tag in the Require header field of a successful
   registration response, the UA begins sending keep-alives as described
   in <a href="#section-4.4">Section 4.4</a>.

      Note: The UA needs to honor 503 (Service Unavailable) responses to
      registrations as described in [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] and [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>].  In
      particular, implementors should note that when receiving a 503
      (Service Unavailable) response with a Retry-After header field,
      the UA is expected to wait the indicated amount of time and retry
      the registration.  A Retry-After header field value of 0 is valid
      and indicates the UA is expected to retry the REGISTER request
      immediately.  Implementations need to ensure that when retrying
      the REGISTER request, they revisit the DNS resolution results such
      that the UA can select an alternate host from the one chosen the
      previous time the URI was resolved.

   If the registering UA receives a 439 (First Hop Lacks Outbound
   Support) response to a REGISTER request, it MAY re-attempt
   registration without using the outbound mechanism (subject to local
   policy at the client).  If the client has one or more alternate
   outbound proxies available, it MAY re-attempt registration through
   such outbound proxies.  See <a href="#section-11.6">Section 11.6</a> for more information on the
   439 response code.

<span class="h4"><a class="selflink" id="section-4.2.2" href="#section-4.2.2">4.2.2</a>.  Subsequent REGISTER Requests</span>

   Registrations for refreshing a binding and for removing a binding use
   the same instance-id and reg-id values as the corresponding initial
   registration where the binding was added.  Registrations that merely
   refresh an existing binding are sent over the same flow as the
   original registration where the binding was added.

   If a re-registration is rejected with a recoverable error response,
   for example by a 503 (Service Unavailable) containing a Retry-After
   header, the UAC SHOULD NOT tear down the corresponding flow if the
   flow uses a connection-oriented transport such as TCP.  As long as
   "pongs" are received in response to "pings", the flow SHOULD be kept
   active until a non-recoverable error response is received.  This
   prevents unnecessary closing and opening of connections.







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<span class="h4"><a class="selflink" id="section-4.2.3" href="#section-4.2.3">4.2.3</a>.  Third-Party Registrations</span>

   In an initial registration or re-registration, a UA MUST NOT include
   a "reg-id" header field parameter in the Contact header field if the
   registering UA is not the same instance as the UA referred to by the
   target Contact header field.  (This practice is occasionally used to
   install forwarding policy into registrars.)

   A UAC also MUST NOT include an instance-id feature tag or "reg-id"
   Contact header field parameter in a request to un-register all
   Contacts (a single Contact header field value with the value of "*").

<span class="h3"><a class="selflink" id="section-4.3" href="#section-4.3">4.3</a>.  Sending Non-REGISTER Requests</span>

   When a UAC is about to send a request, it first performs normal
   processing to select the next hop URI.  The UA can use a variety of
   techniques to compute the route set and accordingly the next hop URI.
   Discussion of these techniques is outside the scope of this document.
   UAs that support this specification SHOULD include the outbound
   option tag in a Supported header field in a request that is not a
   REGISTER request.

   The UAC performs normal DNS resolution on the next hop URI (as
   described in [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>]) to find a protocol, IP address, and port.
   For protocols that don't use TLS, if the UAC has an existing flow to
   this IP address, and port with the correct protocol, then the UAC
   MUST use the existing connection.  For TLS protocols, there MUST also
   be a match between the host production in the next hop and one of the
   URIs contained in the subjectAltName in the peer certificate.  If the
   UAC cannot use one of the existing flows, then it SHOULD form a new
   flow by sending a datagram or opening a new connection to the next
   hop, as appropriate for the transport protocol.

   Typically, a UAC using the procedures of this document and sending a
   dialog-forming request will want all subsequent requests in the
   dialog to arrive over the same flow.  If the UAC is using a Globally
   Routable UA URI (GRUU) [<a href="./rfc5627" title="&quot;Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)&quot;">RFC5627</a>] that was instantiated using a
   Contact header field value that included an "ob" parameter, the UAC
   sends the request over the flow used for registration, and subsequent
   requests will arrive over that same flow.  If the UAC is not using
   such a GRUU, then the UAC adds an "ob" parameter to its Contact
   header field value.  This will cause all subsequent requests in the
   dialog to arrive over the flow instantiated by the dialog-forming
   request.  This case is typical when the request is sent prior to
   registration, such as in the initial subscription dialog for the
   configuration framework [<a href="#ref-CONFIG-FMWK" title="&quot;A Framework for Session Initiation Protocol User Agent Profile Delivery&quot;">CONFIG-FMWK</a>].





<span class="grey">Jennings, et al.            Standards Track                    [Page 17]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-18" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


      Note: If the UAC wants a UDP flow to work through NATs or
      firewalls, it still needs to put the 'rport' parameter [<a href="./rfc3581" title="&quot;An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing&quot;">RFC3581</a>]
      in its Via header field value, and send from the port it is
      prepared to receive on.  More general information about NAT
      traversal in SIP is described in [<a href="#ref-NAT-SCEN" title="&quot;Best Current Practices for NAT Traversal for Client-Server SIP&quot;">NAT-SCEN</a>].

<span class="h3"><a class="selflink" id="section-4.4" href="#section-4.4">4.4</a>.  Keep-Alives and Detecting Flow Failure</span>

   Keep-alives are used for refreshing NAT/firewall bindings and
   detecting flow failure.  Flows can fail for many reasons including
   the rebooting of NATs and the crashing of edge proxies.

   As described in <a href="#section-4.2">Section 4.2</a>, a UA that registers will begin sending
   keep-alives after an appropriate registration response.  A UA that
   does not register (for example, a PSTN gateway behind a firewall) can
   also send keep-alives under certain circumstances.

   Under specific circumstances, a UAC might be allowed to send STUN
   keep-alives even if the procedures in <a href="#section-4.2">Section 4.2</a> were not completed,
   provided that there is an explicit indication that the target first-
   hop SIP node supports STUN keep-alives.  For example, this applies to
   a non-registering UA or to a case where the UA registration
   succeeded, but the response did not include the outbound option-tag
   in the Require header field.

      Note: A UA can "always" send a double CRLF (a "ping") over
      connection-oriented transports as this is already allowed by
      <a href="./rfc3261#section-7.5">Section&nbsp;7.5 of [RFC3261]</a>.  However a UA that did not register
      using outbound registration cannot expect a CRLF in response (a
      "pong") unless the UA has an explicit indication that CRLF keep-
      alives are supported as described in this section.  Likewise, a UA
      that did not successfully register with outbound procedures needs
      explicit indication that the target first-hop SIP node supports
      STUN keep-alives before it can send any STUN messages.

   A configuration option indicating keep-alive support for a specific
   target is considered an explicit indication.  If these conditions are
   satisfied, the UA sends its keep-alives according to the same
   guidelines as those used when UAs register; these guidelines are
   described below.

   The UA needs to detect when a specific flow fails.  The UA actively
   tries to detect failure by periodically sending keep-alive messages
   using one of the techniques described in Sections <a href="#section-4.4.1">4.4.1</a> or <a href="#section-4.4.2">4.4.2</a>.  If
   a flow with a registration has failed, the UA follows the procedures
   in <a href="#section-4.2">Section 4.2</a> to form a new flow to replace the failed one.





<span class="grey">Jennings, et al.            Standards Track                    [Page 18]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-19" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   When a successful registration response contains the Flow-Timer
   header field, the value of this header field is the number of seconds
   the server is prepared to wait without seeing keep-alives before it
   could consider the corresponding flow dead.  Note that the server
   would wait for an amount of time larger than the Flow-Timer in order
   to have a grace period to account for transport delay.  The UA MUST
   send keep-alives at least as often as this number of seconds.  If the
   UA uses the server-recommended keep-alive frequency it SHOULD send
   its keep-alives so that the interval between each keep-alive is
   randomly distributed between 80% and 100% of the server-provided
   time.  For example, if the server suggests 120 seconds, the UA would
   send each keep-alive with a different frequency between 95 and 120
   seconds.

   If no Flow-Timer header field was present in a register response for
   this flow, the UA can send keep-alives at its discretion.  The
   sections below provide RECOMMENDED default values for these keep-
   alives.

   The client needs to perform normal [<a href="./rfc3263" title="&quot;Session Initiation Protocol (SIP): Locating SIP Servers&quot;">RFC3263</a>] SIP DNS resolution on
   the URI from the outbound-proxy-set to pick a transport.  Once a
   transport is selected, the UA selects the keep-alive approach that is
   recommended for that transport.

   <a href="#section-4.4.1">Section 4.4.1</a> describes a keep-alive mechanism for connection-
   oriented transports such as TCP or SCTP.  <a href="#section-4.4.2">Section 4.4.2</a> describes a
   keep-alive mechanism for connection-less transports such as UDP.
   Support for other transports such as DCCP [<a href="./rfc4340" title="&quot;Datagram Congestion Control Protocol (DCCP)&quot;">RFC4340</a>] is for further
   study.

<span class="h4"><a class="selflink" id="section-4.4.1" href="#section-4.4.1">4.4.1</a>.  Keep-Alive with CRLF</span>

   This approach MUST only be used with connection oriented transports
   such as TCP or SCTP; it MUST NOT be used with connection-less
   transports such as UDP.

   A User Agent that forms flows checks if the configured URI to which
   the UA is connecting resolves to a connection-oriented transport
   (e.g., TCP and TLS over TCP).

   For this mechanism, the client "ping" is a double-CRLF sequence, and
   the server "pong" is a single CRLF, as defined in the ABNF below:

   CRLF = CR LF
   double-CRLF = CR LF CR LF
   CR = %x0D
   LF = %x0A




<span class="grey">Jennings, et al.            Standards Track                    [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   The "ping" and "pong" need to be sent between SIP messages and cannot
   be sent in the middle of a SIP message.  If sending over TLS, the
   CRLFs are sent inside the TLS protected channel.  If sending over a
   SigComp [<a href="./rfc3320" title="&quot;Signaling Compression (SigComp)&quot;">RFC3320</a>] compressed data stream, the CRLF keep-alives are
   sent inside the compressed stream.  The double CRLF is considered a
   single SigComp message.  The specific mechanism for representing
   these characters is an implementation-specific matter to be handled
   by the SigComp compressor at the sending end.

   If a pong is not received within 10 seconds after sending a ping (or
   immediately after processing any incoming message being received when
   that 10 seconds expires), then the client MUST treat the flow as
   failed.  Clients MUST support this CRLF keep-alive.

      Note: This value of 10-second timeout was selected to be long
      enough that it allows plenty of time for a server to send a
      response even if the server is temporarily busy with an
      administrative activity.  At the same time, it was selected to be
      small enough that a UA registered to two redundant servers with
      unremarkable hardware uptime could still easily provide very high
      levels of overall reliability.  Although some Internet protocols
      are designed for round-trip times over 10 seconds, SIP for real-
      time communications is not really usable in these type of
      environments as users often abandon calls before waiting much more
      than a few seconds.

   When a Flow-Timer header field is not provided in the most recent
   success registration response, the proper selection of keep-alive
   frequency is primarily a trade-off between battery usage and
   availability.  The UA MUST select a random number between a fixed or
   configurable upper bound and a lower bound, where the lower bound is
   20% less then the upper bound.  The fixed upper bound or the default
   configurable upper bound SHOULD be 120 seconds (95 seconds for the
   lower bound) where battery power is not a concern and 840 seconds
   (672 seconds for the lower bound) where battery power is a concern.
   The random number will be different for each keep-alive "ping".

      Note on selection of time values: the 120-second upper bound was
      chosen based on the idea that for a good user experience, failures
      normally will be detected in this amount of time and a new
      connection will be set up.  The 14-minute upper bound for battery-
      powered devices was selected based on NATs with TCP timeouts as
      low as 15 minutes.  Operators that wish to change the relationship
      between load on servers and the expected time that a user might
      not receive inbound communications will probably adjust this time.
      The 95-second lower bound was chosen so that the jitter introduced
      will result in a relatively even load on the servers after 30
      minutes.



<span class="grey">Jennings, et al.            Standards Track                    [Page 20]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


<span class="h4"><a class="selflink" id="section-4.4.2" href="#section-4.4.2">4.4.2</a>.  Keep-Alive with STUN</span>

   This approach MUST only be used with connection-less transports, such
   as UDP; it MUST NOT be used for connection-oriented transports such
   as TCP and SCTP.

   A User Agent that forms flows checks if the configured URI to which
   the UA is connecting resolves to use the UDP transport.  The UA can
   periodically perform keep-alive checks by sending STUN [<a href="./rfc5389" title="&quot;Session Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>]
   Binding Requests over the flow as described in <a href="#section-8">Section 8</a>.  Clients
   MUST support STUN-based keep-alives.

   When a Flow-Timer header field is not included in a successful
   registration response, the time between each keep-alive request
   SHOULD be a random number between 24 and 29 seconds.

      Note on selection of time values: the upper bound of 29 seconds
      was selected, as many NATs have UDP timeouts as low as 30 seconds.
      The 24-second lower bound was selected so that after 10 minutes
      the jitter introduced by different timers will make the keep-alive
      requests unsynchronized to evenly spread the load on the servers.
      Note that the short NAT timeouts with UDP have a negative impact
      on battery life.

   If a STUN Binding Error Response is received, or if no Binding
   Response is received after 7 retransmissions (16 times the STUN "RTO"
   timer -- where RTO is an estimate of round-trip time), the UA
   considers the flow failed.  If the XOR-MAPPED-ADDRESS in the STUN
   Binding Response changes, the UA MUST treat this event as a failure
   on the flow.

<span class="h3"><a class="selflink" id="section-4.5" href="#section-4.5">4.5</a>.  Flow Recovery</span>

   When a flow used for registration (through a particular URI in the
   outbound-proxy-set) fails, the UA needs to form a new flow to replace
   the old flow and replace any registrations that were previously sent
   over this flow.  Each new registration MUST have the same reg-id
   value as the registration it replaces.  This is done in much the same
   way as forming a brand new flow as described in <a href="#section-4.2">Section 4.2</a>; however,
   if there is a failure in forming this flow, the UA needs to wait a
   certain amount of time before retrying to form a flow to this
   particular next hop.

   The amount of time to wait depends if the previous attempt at
   establishing a flow was successful.  For the purposes of this
   section, a flow is considered successful if outbound registration
   succeeded, and if keep-alives are in use on this flow, at least one
   subsequent keep-alive response was received.



<span class="grey">Jennings, et al.            Standards Track                    [Page 21]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-22" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   The number of seconds to wait is computed in the following way.  If
   all of the flows to every URI in the outbound proxy set have failed,
   the base-time is set to a lower value (with a default of 30 seconds);
   otherwise, in the case where at least one of the flows has not
   failed, the base-time is set to a higher value (with a default of 90
   seconds).  The upper-bound wait time (W) is computed by taking two
   raised to the power of the number of consecutive registration
   failures for that URI, and multiplying this by the base-time, up to a
   configurable maximum time (with a default of 1800 seconds).

   W = min (max-time, (base-time * (2 ^ consecutive-failures)))

   These times MAY be configurable in the UA.  The three times are:

   o  max-time with a default of 1800 seconds

   o  base-time (if all failed) with a default of 30 seconds

   o  base-time (if all have not failed) with a default of 90 seconds

   For example, if the base-time is 30 seconds, and there were three
   failures, then the upper-bound wait time is min(1800, 30*(2^3)) or
   240 seconds.  The actual amount of time the UA waits before retrying
   registration (the retry delay time) is computed by selecting a
   uniform random time between 50 and 100% of the upper-bound wait time.
   The UA MUST wait for at least the value of the retry delay time
   before trying another registration to form a new flow for that URI (a
   503 response to an earlier failed registration attempt with a Retry-
   After header field value may cause the UA to wait longer).

   To be explicitly clear on the boundary conditions: when the UA boots,
   it immediately tries to register.  If this fails and no registration
   on other flows succeed, the first retry happens somewhere between 30
   and 60 seconds after the failure of the first registration request.
   If the number of consecutive-failures is large enough that the
   maximum of 1800 seconds is reached, the UA will keep trying
   indefinitely with a random time of 15 to 30 minutes between each
   attempt.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Edge Proxy Procedures</span>

<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a>.  Processing Register Requests</span>

   When an edge proxy receives a registration request with a "reg-id"
   header field parameter in the Contact header field, it needs to
   determine if it (the edge proxy) will have to be visited for any
   subsequent requests sent to the User Agent identified in the Contact
   header field, or not.  If the edge proxy is the first hop, as



<span class="grey">Jennings, et al.            Standards Track                    [Page 22]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-23" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   indicated by the Via header field, it MUST insert its URI in a Path
   header field value as described in [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>].  If it is not the first
   hop, it might still decide to add itself to the Path header based on
   local policy.  In addition, if the edge proxy is the first SIP node
   after the UAC, the edge proxy either MUST store a "flow token"
   (containing information about the flow from the previous hop) in its
   Path URI or reject the request.  The flow token MUST be an identifier
   that is unique to this network flow.  The flow token MAY be placed in
   the userpart of the URI.  In addition, the first node MUST include an
   "ob" URI parameter in its Path header field value.  If the edge proxy
   is not the first SIP node after the UAC it MUST NOT place an "ob" URI
   parameter in a Path header field value.  The edge proxy can determine
   if it is the first hop by examining the Via header field.

<span class="h3"><a class="selflink" id="section-5.2" href="#section-5.2">5.2</a>.  Generating Flow Tokens</span>

   A trivial but impractical way to satisfy the flow token requirement
   in <a href="#section-5.1">Section 5.1</a> involves storing a mapping between an incrementing
   counter and the connection information; however, this would require
   the edge proxy to keep an infeasible amount of state.  It is unclear
   when this state could be removed, and the approach would have
   problems if the proxy crashed and lost the value of the counter.  A
   stateless example is provided below.  A proxy can use any algorithm
   it wants as long as the flow token is unique to a flow, the flow can
   be recovered from the token, and the token cannot be modified by
   attackers.

      Example Algorithm: When the proxy boots, it selects a 20-octet
      crypto random key called K that only the edge proxy knows.  A byte
      array, called S, is formed that contains the following information
      about the flow the request was received on: an enumeration
      indicating the protocol, the local IP address and port, the remote
      IP address and port.  The HMAC of S is computed using the key K
      and the HMAC-SHA1-80 algorithm, as defined in [<a href="./rfc2104" title="&quot;HMAC: Keyed-Hashing for Message Authentication&quot;">RFC2104</a>].  The
      concatenation of the HMAC and S are base64 encoded, as defined in
      [<a href="./rfc4648" title="&quot;The Base16, Base32, and Base64 Data Encodings&quot;">RFC4648</a>], and used as the flow identifier.  When using IPv4
      addresses, this will result in a 32-octet identifier.

<span class="h3"><a class="selflink" id="section-5.3" href="#section-5.3">5.3</a>.  Forwarding Non-REGISTER Requests</span>

   When an edge proxy receives a request, it applies normal routing
   procedures with the following additions.  If the edge proxy receives
   a request where the edge proxy is the host in the topmost Route
   header field value, and the Route header field value contains a flow
   token, the proxy follows the procedures of this section.  Otherwise
   the edge proxy skips the procedures in this section, removes itself
   from the Route header field, and continues processing the request.




<span class="grey">Jennings, et al.            Standards Track                    [Page 23]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-24" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   The proxy decodes the flow token and compares the flow in the flow
   token with the source of the request to determine if this is an
   "incoming" or "outgoing" request.

   If the flow in the flow token identified by the topmost Route header
   field value matches the source IP address and port of the request,
   the request is an "outgoing" request; otherwise, it is an "incoming"
   request.

<span class="h4"><a class="selflink" id="section-5.3.1" href="#section-5.3.1">5.3.1</a>.  Processing Incoming Requests</span>

   If the Route header value contains an "ob" URI parameter, the Route
   header was probably copied from the Path header in a registration.
   If the Route header value contains an "ob" URI parameter, and the
   request is a new dialog-forming request, the proxy needs to adjust
   the route set to ensure that subsequent requests in the dialog can be
   delivered over a valid flow to the UA instance identified by the flow
   token.

      Note: A simple approach to satisfy this requirement is for the
      proxy to add a Record-Route header field value that contains the
      flow-token, by copying the URI in the Route header minus the "ob"
      parameter.

   Next, whether the Route header field contained an "ob" URI parameter
   or not, the proxy removes the Route header field value and forwards
   the request over the 'logical flow' identified by the flow token,
   that is known to deliver data to the specific target UA instance.  If
   the flow token has been tampered with, the proxy SHOULD send a 403
   (Forbidden) response.  If the flow no longer exists, the proxy SHOULD
   send a 430 (Flow Failed) response to the request.

   Proxies that used the example algorithm described in <a href="#section-5.2">Section 5.2</a> to
   form a flow token follow the procedures below to determine the
   correct flow.  To decode the flow token, take the flow identifier in
   the user portion of the URI and base64 decode it, then verify the
   HMAC is correct by recomputing the HMAC and checking that it matches.
   If the HMAC is not correct, the request has been tampered with.

<span class="h4"><a class="selflink" id="section-5.3.2" href="#section-5.3.2">5.3.2</a>.  Processing Outgoing Requests</span>

   For mid-dialog requests to work with outbound UAs, the requests need
   to be forwarded over some valid flow to the appropriate UA instance.
   If the edge proxy receives an outgoing dialog-forming request, the
   edge proxy can use the presence of the "ob" URI parameter in the
   UAC's Contact URI (or topmost Route header field) to determine if the
   edge proxy needs to assist in mid-dialog request routing.




<span class="grey">Jennings, et al.            Standards Track                    [Page 24]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-25" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


      Implementation note: Specific procedures at the edge proxy to
      ensure that mid-dialog requests are routed over an existing flow
      are not part of this specification.  However, an approach such as
      having the edge proxy add a Record-Route header with a flow token
      is one way to ensure that mid-dialog requests are routed over the
      correct flow.

<span class="h3"><a class="selflink" id="section-5.4" href="#section-5.4">5.4</a>.  Edge Proxy Keep-Alive Handling</span>

   All edge proxies compliant with this specification MUST implement
   support for STUN NAT keep-alives on their SIP UDP ports as described
   in <a href="#section-8">Section 8</a>.

   When a server receives a double CRLF sequence between SIP messages on
   a connection-oriented transport such as TCP or SCTP, it MUST
   immediately respond with a single CRLF over the same connection.

   The last proxy to forward a successful registration response to a UA
   MAY include a Flow-Timer header field if the response contains the
   outbound option-tag in a Require header field value in the response.
   The reason a proxy would send a Flow-Timer is if it wishes to detect
   flow failures proactively and take appropriate action (e.g., log
   alarms, provide alternative treatment if incoming requests for the UA
   are received, etc.).  The server MUST wait for an amount of time
   larger than the Flow-Timer in order to have a grace period to account
   for transport delay.

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

   This specification updates the definition of a binding in <a href="./rfc3261#section-10">[RFC3261],
   Section&nbsp;10</a> and <a href="./rfc3327#section-5.3">[RFC3327], Section&nbsp;5.3</a>.

   Registrars that implement this specification MUST support the Path
   header mechanism [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>].

   When receiving a REGISTER request, the registrar MUST check from its
   Via header field if the registrar is the first hop or not.  If the
   registrar is not the first hop, it MUST examine the Path header of
   the request.  If the Path header field is missing or it exists but
   the first URI does not have an "ob" URI parameter, then outbound
   processing MUST NOT be applied to the registration.  In this case,
   the following processing applies: if the REGISTER request contains
   the reg-id and the outbound option tag in a Supported header field,
   then the registrar MUST respond to the REGISTER request with a 439
   (First Hop Lacks Outbound Support) response; otherwise, the registrar
   MUST ignore the "reg-id" parameter of the Contact header.  See
   <a href="#section-11.6">Section 11.6</a> for more information on the 439 response code.




<span class="grey">Jennings, et al.            Standards Track                    [Page 25]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-26" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   A Contact header field value with an instance-id media feature tag
   but no "reg-id" header field parameter is valid (this combination
   will result in the creation of a GRUU, as described in the GRUU
   specification [<a href="./rfc5627" title="&quot;Obtaining and Using Globally Routable User Agent URIs (GRUUs) in the Session Initiation Protocol (SIP)&quot;">RFC5627</a>]), but one with a reg-id but no instance-id is
   not valid.  If the registrar processes a Contact header field value
   with a reg-id but no instance-id, it simply ignores the reg-id
   parameter.

   A registration containing a "reg-id" header field parameter and a
   non-zero expiration is used to register a single UA instance over a
   single flow, and can also de-register any Contact header fields with
   zero expiration.  Therefore, if the Contact header field contains
   more than one header field value with a non-zero expiration and any
   of these header field values contain a "reg-id" Contact header field
   parameter, the entire registration SHOULD be rejected with a 400 (Bad
   Request) response.  The justification for recommending rejection
   versus making it mandatory is that the receiver is allowed by
   [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] to squelch (not respond to) excessively malformed or
   malicious messages.

   If the Contact header did not contain a "reg-id" Contact header field
   parameter or if that parameter was ignored (as described above), the
   registrar MUST NOT include the outbound option-tag in the Require
   header field of its response.

   The registrar MUST be prepared to receive, simultaneously for the
   same AOR, some registrations that use instance-id and reg-id and some
   registrations that do not.  The registrar MAY be configured with
   local policy to reject any registrations that do not include the
   instance-id and reg-id, or with Path header field values that do not
   contain the "ob" URI parameter.  If the Contact header field does not
   contain a "+sip.instance" Contact header field parameter, the
   registrar processes the request using the Contact binding rules in
   [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].

   When a "+sip.instance" Contact header field parameter and a "reg-id"
   Contact header field parameter are present in a Contact header field
   of a REGISTER request (after the Contact header validation as
   described above), the corresponding binding is between an AOR and the
   combination of the instance-id (from the "+sip.instance" Contact
   header parameter) and the value of "reg-id" Contact header field
   parameter parameter.  The registrar MUST store in the binding the
   Contact URI, all the Contact header field parameters, and any Path
   header field values.  (Even though the Contact URI is not used for
   binding comparisons, it is still needed by the authoritative proxy to
   form the target set.)  Provided that the UAC had included an outbound
   option-tag (defined in <a href="#section-11.4">Section 11.4</a>) in a Supported header field




<span class="grey">Jennings, et al.            Standards Track                    [Page 26]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-27" ></span>
<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


   value in the REGISTER request, the registrar MUST include the
   outbound option-tag in a Require header field value in its response
   to that REGISTER request.

   If the UAC has a direct flow with the registrar, the registrar MUST
   store enough information to uniquely identify the network flow over
   which the request arrived.  For common operating systems with TCP,
   this would typically be just the handle to the file descriptor where
   the handle would become invalid if the TCP session was closed.  For
   common operating systems with UDP this would typically be the file
   descriptor for the local socket that received the request, the local
   interface, and the IP address and port number of the remote side that
   sent the request.  The registrar MAY store this information by adding
   itself to the Path header field with an appropriate flow token.

   If the registrar receives a re-registration for a specific
   combination of AOR, and instance-id and reg-id values, the registrar
   MUST update any information that uniquely identifies the network flow
   over which the request arrived if that information has changed, and
   SHOULD update the time the binding was last updated.

   To be compliant with this specification, registrars that can receive
   SIP requests directly from a UAC without intervening edge proxies
   MUST implement the same keep-alive mechanisms as edge proxies
   (<a href="#section-5.4">Section 5.4</a>).  Registrars with a direct flow with a UA MAY include a
   Flow-Timer header in a 2xx class registration response that includes
   the outbound option-tag in the Require header.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Authoritative Proxy Procedures: Forwarding Requests</span>

   When a proxy uses the location service to look up a registration
   binding and then proxies a request to a particular contact, it
   selects a contact to use normally, with a few additional rules:

   o  The proxy MUST NOT populate the target set with more than one
      contact with the same AOR and instance-id at a time.

   o  If a request for a particular AOR and instance-id fails with a 430
      (Flow Failed) response, the proxy SHOULD replace the failed branch
      with another target (if one is available) with the same AOR and
      instance-id, but a different reg-id.

   o  If the proxy receives a final response from a branch other than a
      408 (Request Timeout) or a 430 (Flow Failed) response, the proxy
      MUST NOT forward the same request to another target representing
      the same AOR and instance-id.  The targeted instance has already
      provided its response.




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   The proxy uses the next-hop target of the message and the value of
   any stored Path header field vector in the registration binding to
   decide how to forward and populate the Route header in the request.
   If the proxy is co-located with the registrar and stored information
   about the flow to the UA that created the binding, then the proxy
   MUST send the request over the same 'logical flow' saved with the
   binding, since that flow is known to deliver data to the specific
   target UA instance's network flow that was saved with the binding.

      Implementation note: Typically this means that for TCP, the
      request is sent on the same TCP socket that received the REGISTER
      request.  For UDP, the request is sent from the same local IP
      address and port over which the registration was received, to the
      same IP address and port from which the REGISTER was received.

   If a proxy or registrar receives information from the network that
   indicates that no future messages will be delivered on a specific
   flow, then the proxy MUST invalidate all the bindings in the target
   set that use that flow (regardless of AOR).  Examples of this are a
   TCP socket closing or receiving a destination unreachable ICMP error
   on a UDP flow.  Similarly, if a proxy closes a file descriptor, it
   MUST invalidate all the bindings in the target set with flows that
   use that file descriptor.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  STUN Keep-Alive Processing</span>

   This section describes changes to the SIP transport layer that allow
   SIP and STUN [<a href="./rfc5389" title="&quot;Session Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>] Binding Requests to be mixed over the same
   flow.  This constitutes a new STUN usage.  The STUN messages are used
   to verify that connectivity is still available over a UDP flow, and
   to provide periodic keep-alives.  These STUN keep-alives are always
   sent to the next SIP hop.  STUN messages are not delivered end-to-
   end.

   The only STUN messages required by this usage are Binding Requests,
   Binding Responses, and Binding Error Responses.  The UAC sends
   Binding Requests over the same UDP flow that is used for sending SIP
   messages.  These Binding Requests do not require any STUN attributes.
   The corresponding Binding Responses do not require any STUN
   attributes except the XOR-MAPPED-ADDRESS.  The UAS, proxy, or
   registrar responds to a valid Binding Request with a Binding Response
   that MUST include the XOR-MAPPED-ADDRESS attribute.

   If a server compliant to this section receives SIP requests on a
   given interface and UDP port, it MUST also provide a limited version
   of a STUN server on the same interface and UDP port.





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      Note: It is easy to distinguish STUN and SIP packets sent over
      UDP, because the first octet of a STUN Binding method has a value
      of 0 or 1, while the first octet of a SIP message is never a 0 or
      1.

   Because sending and receiving binary STUN data on the same ports used
   for SIP is a significant and non-backwards compatible change to <a href="./rfc3261">RFC</a>
   <a href="./rfc3261">3261</a>, this section requires a number of checks before sending STUN
   messages to a SIP node.  If a SIP node sends STUN requests (for
   example, due to incorrect configuration) despite these warnings, the
   node could be blacklisted for UDP traffic.

   A SIP node MUST NOT send STUN requests over a flow unless it has an
   explicit indication that the target next-hop SIP server claims to
   support this specification.  UACs MUST NOT use an ambiguous
   configuration option such as "Work through NATs?" or "Do keep-
   alives?" to imply next-hop STUN support.  A UAC MAY use the presence
   of an "ob" URI parameter in the Path header in a registration
   response as an indication that its first edge proxy supports the
   keep-alives defined in this document.

      Note: Typically, a SIP node first sends a SIP request and waits to
      receive a 2xx class response over a flow to a new target
      destination, before sending any STUN messages.  When scheduled for
      the next NAT refresh, the SIP node sends a STUN request to the
      target.

   Once a flow is established, failure of a STUN request (including its
   retransmissions) is considered a failure of the underlying flow.  For
   SIP over UDP flows, if the XOR-MAPPED-ADDRESS returned over the flow
   changes, this indicates that the underlying connectivity has changed,
   and is considered a flow failure.

   The SIP keep-alive STUN usage requires no backwards compatibility
   with [<a href="./rfc3489" title="&quot;STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)&quot;">RFC3489</a>].

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Use with SigComp</span>

   When STUN is used together with SigComp [<a href="./rfc3320" title="&quot;Signaling Compression (SigComp)&quot;">RFC3320</a>] compressed SIP
   messages over the same flow, the STUN messages are simply sent
   uncompressed, "outside" of SigComp.  This is supported by
   multiplexing STUN messages with SigComp messages by checking the two
   topmost bits of the message.  These bits are always one for SigComp,
   or zero for STUN.

      Note: All SigComp messages contain a prefix (the five most
      significant bits of the first byte are set to one) that does not
      occur in UTF-8 [<a href="./rfc3629" title="&quot;UTF-8, a transformation format of ISO 10646&quot;">RFC3629</a>] encoded text messages, so for



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      applications that use this encoding (or ASCII encoding) it is
      possible to multiplex uncompressed application messages and
      SigComp messages on the same UDP port.  The most significant two
      bits of every STUN Binding method are both zeroes.  This, combined
      with the magic cookie, aids in differentiating STUN packets from
      other protocols when STUN is multiplexed with other protocols on
      the same port.

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  Example Message Flow</span>

   Below is an example message flow illustrating most of the concepts
   discussed in this specification.  In many cases, Via, Content-Length,
   and Max-Forwards headers are omitted for brevity and readability.

   In these examples, "EP1" and "EP2" are outbound proxies, and "Proxy"
   is the authoritativeProxy.

   The section is subdivided into independent calls flows; however, they
   are structured in sequential order of a hypothetical sequence of call
   flows.

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Subscription to Configuration Package</span>

   If the outbound proxy set is already configured on Bob's UA, then
   this subsection can be skipped.  Otherwise, if the outbound proxy set
   is learned through the configuration package, Bob's UA sends a
   SUBSCRIBE request for the UA profile configuration package
   [<a href="#ref-CONFIG-FMWK" title="&quot;A Framework for Session Initiation Protocol User Agent Profile Delivery&quot;">CONFIG-FMWK</a>].  This request is a poll (Expires is zero).  After
   receiving the NOTIFY request, Bob's UA fetches the external
   configuration using HTTPS (not shown) and obtains a configuration
   file that contains the outbound-proxy-set "sip:ep1.example.com;lr"
   and "sip:ep2.example.com;lr".

     [----example.com domain-------------------------]
     Bob         EP1   EP2     Proxy             Config
      |           |     |        |                  |
    1)|SUBSCRIBE-&gt;|     |        |                  |
    2)|           |---SUBSCRIBE Event: ua-profile -&gt;|
    3)|           |&lt;--200 OK -----------------------|
    4)|&lt;--200 OK--|     |        |                  |
    5)|           |&lt;--NOTIFY------------------------|
    6)|&lt;--NOTIFY--|     |        |                  |
    7)|---200 OK-&gt;|     |        |                  |
    8)|           |---200 OK ----------------------&gt;|
      |           |     |        |                  |

   In this example, the DNS server happens to be configured so that sip:
   example.com resolves to EP1 and EP2.



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   Example Message #1:

   SUBSCRIBE sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com
     SIP/2.0
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnlsdkdj2
   Max-Forwards: 70
   From: &lt;anonymous@example.com&gt;;tag=23324
   To: &lt;sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com&gt;
   Call-ID: nSz1TWN54x7My0GvpEBj
   CSeq: 1 SUBSCRIBE
   Event: ua-profile ;profile-type=device
    ;vendor="example.com";model="uPhone";version="1.1"
   Expires: 0
   Supported: path, outbound
   Accept: message/external-body, application/x-uPhone-config
   Contact: &lt;sip:192.0.2.2;transport=tcp;ob&gt;
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Content-Length: 0

   In message #2, EP1 adds the following Record-Route header:

   Record-Route:
    &lt;sip:GopIKSsn0oGLPXRdV9BAXpT3coNuiGKV@ep1.example.com;lr&gt;

   In message #5, the configuration server sends a NOTIFY with an
   external URL for Bob to fetch his configuration.  The NOTIFY has a
   Subscription-State header that ends the subscription.

   Message #5

   NOTIFY sip:192.0.2.2;transport=tcp;ob SIP/2.0
   Via: SIP/2.0/TCP 192.0.2.5;branch=z9hG4bKn81dd2
   Max-Forwards: 70
   To: &lt;anonymous@example.com&gt;;tag=23324
   From: &lt;sip:00000000-0000-1000-8000-AABBCCDDEEFF@example.com&gt;;tag=0983
   Call-ID: nSz1TWN54x7My0GvpEBj
   CSeq: 1 NOTIFY
   Route: &lt;sip:GopIKSsn0oGLPXRdV9BAXpT3coNuiGKV@ep1.example.com;lr&gt;
   Subscription-State: terminated;reason=timeout
   Event: ua-profile
   Content-Type: message/external-body; access-type="URL"
    ;expiration="Thu, 01 Jan 2009 09:00:00 UTC"
    ;URL="http://example.com/uPhone.cfg"
    ;size=9999;hash=10AB568E91245681AC1B
   Content-Length: 0






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   EP1 receives this NOTIFY request, strips off the Route header,
   extracts the flow-token, calculates the correct flow, and forwards
   the request (message #6) over that flow to Bob.

   Bob's UA fetches the configuration file and learns the outbound proxy
   set.

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

   Now that Bob's UA is configured with the outbound-proxy-set whether
   through configuration or using the configuration framework procedures
   of the previous section, Bob's UA sends REGISTER requests through
   each edge proxy in the set.  Once the registrations succeed, Bob's UA
   begins sending CRLF keep-alives about every 2 minutes.

     Bob         EP1   EP2     Proxy     Alice
      |           |     |        |         |
    9)|-REGISTER-&gt;|     |        |         |
   10)|           |---REGISTER--&gt;|         |
   11)|           |&lt;----200 OK---|         |
   12)|&lt;-200 OK---|     |        |         |
   13)|----REGISTER----&gt;|        |         |
   14)|           |     |--REG--&gt;|         |
   15)|           |     |&lt;-200---|         |
   16)|&lt;----200 OK------|        |         |
      |           |     |        |         |
      |  about 120 seconds later...        |
      |           |     |        |         |
   17)|--2CRLF---&gt;|     |        |         |
   18)|&lt;--CRLF----|     |        |         |
   19)|------2CRLF-----&gt;|        |         |
   20)|&lt;------CRLF------|        |         |
      |           |     |        |         |

   In message #9, Bob's UA sends its first registration through the
   first edge proxy in the outbound-proxy-set by including a loose
   route.  The UA includes an instance-id and reg-id in its Contact
   header field value.  Note the option-tags in the Supported header.













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   Message #9

   REGISTER sip:example.com SIP/2.0
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7
   Max-Forwards: 70
   From: Bob &lt;sip:bob@example.com&gt;;tag=7F94778B653B
   To: Bob &lt;sip:bob@example.com&gt;
   Call-ID: 16CB75F21C70
   CSeq: 1 REGISTER
   Supported: path, outbound
   Route: &lt;sip:ep1.example.com;lr&gt;
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=1
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Content-Length: 0

   Message #10 is similar.  EP1 removes the Route header field value,
   decrements Max-Forwards, and adds its Via header field value.  Since
   EP1 is the first edge proxy, it adds a Path header with a flow token
   and includes the "ob" parameter.

   Path: &lt;sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob&gt;

   Since the response to the REGISTER (message #11) contains the
   outbound option-tag in the Require header field, Bob's UA will know
   that the registrar used outbound binding rules.  The response also
   contains the currently active Contacts, and the Path for the current
   registration.

   Message #11

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP 192.0.2.15;branch=z9hG4bKnuiqisi
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnashds7
   From: Bob &lt;sip:bob@example.com&gt;;tag=7F94778B653B
   To: Bob &lt;sip:bob@example.com&gt;;tag=6AF99445E44A
   Call-ID: 16CB75F21C70
   CSeq: 1 REGISTER
   Supported: path, outbound
   Require: outbound
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=1;expires=3600
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Path: &lt;sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob&gt;
   Content-Length: 0

   The second registration through EP2 (message #13) is similar except
   that the Call-ID has changed, the reg-id is 2, and the Route header
   goes through EP2.




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   Message #13

   REGISTER sip:example.com SIP/2.0
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnqr9bym
   Max-Forwards: 70
   From: Bob &lt;sip:bob@example.com&gt;;tag=755285EABDE2
   To: Bob &lt;sip:bob@example.com&gt;
   Call-ID: E05133BD26DD
   CSeq: 1 REGISTER
   Supported: path, outbound
   Route: &lt;sip:ep2.example.com;lr&gt;
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=2
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Content-Length: 0

   Likewise in message #14, EP2 adds a Path header with flow token and
   "ob" parameter.

   Path: &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob&gt;

   Message #16 tells Bob's UA that outbound registration was successful,
   and shows both Contacts.  Note that only the Path corresponding to
   the current registration is returned.

   Message #16

   SIP/2.0 200 OK
   Via: SIP/2.0/TCP 192.0.2.2;branch=z9hG4bKnqr9bym
   From: Bob &lt;sip:bob@example.com&gt;;tag=755285EABDE2
   To: Bob &lt;sip:bob@example.com&gt;;tag=49A9AD0B3F6A
   Call-ID: E05133BD26DD
   Supported: path, outbound
   Require: outbound
   CSeq: 1 REGISTER
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=1;expires=3600
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=2;expires=3600
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"
   Path: &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob&gt;
   Content-Length: 0

<span class="h3"><a class="selflink" id="section-9.3" href="#section-9.3">9.3</a>.  Incoming Call and Proxy Crash</span>

   In this example, after registration, EP1 crashes and reboots.  Before
   Bob's UA notices that its flow to EP1 is no longer responding, Alice
   calls Bob.  Bob's authoritative proxy first tries the flow to EP1,





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   but EP1 no longer has a flow to Bob, so it responds with a 430 (Flow
   Failed) response.  The proxy removes the stale registration and tries
   the next binding for the same instance.

     Bob         EP1   EP2     Proxy     Alice
      |           |     |        |         |
      |    CRASH  X     |        |         |
      |        Reboot   |        |         |
      |           |     |        |         |
   21)|           |     |        |&lt;-INVITE-|
   22)|           |&lt;---INVITE----|         |
   23)|           |----430------&gt;|         |
   24)|           |     |&lt;-INVITE|         |
   25)|&lt;---INVITE-------|        |         |
   26)|----200 OK------&gt;|        |         |
   27)|           |     |200 OK-&gt;|         |
   28)|           |     |        |-200 OK-&gt;|
   29)|           |     |&lt;----------ACK----|
   30)|&lt;---ACK----------|        |         |
      |           |     |        |         |
   31)|           |     |&lt;----------BYE----|
   32)|&lt;---BYE----------|        |         |
   33)|----200 OK------&gt;|        |         |
   34)|           |     |--------200 OK---&gt;|
      |           |     |        |         |


   Message #21

   INVITE sip:bob@example.com SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE

   Bob's proxy rewrites the Request-URI to the Contact URI used in Bob's
   registration, and places the path for one of the registrations
   towards Bob's UA instance into a Route header field.  This Route goes
   through EP1.

   Message #22

   INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE
   Route: &lt;sip:VskztcQ/S8p4WPbOnHbuyh5iJvJIW3ib@ep1.example.com;lr;ob&gt;



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   Since EP1 just rebooted, it does not have the flow described in the
   flow token.  It returns a 430 (Flow Failed) response.

   Message #23

   SIP/2.0 430 Flow Failed
   To: Bob &lt;sip:bob@example.com&gt;
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE

   The proxy deletes the binding for this path and tries to forward the
   INVITE again, this time with the path through EP2.

   Message #24

   INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE
   Route: &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr;ob&gt;

   In message #25, EP2 needs to add a Record-Route header field value,
   so that any subsequent in-dialog messages from Alice's UA arrive at
   Bob's UA.  EP2 can determine it needs to Record-Route since the
   request is a dialog-forming request and the Route header contained a
   flow token and an "ob" parameter.  This Record-Route information is
   passed back to Alice's UA in the responses (messages #26, 27, and
   28).

   Message #25

   INVITE sip:bob@192.0.2.2;transport=tcp SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE
   Record-Route:
     &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr&gt;











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   Message #26

   SIP/2.0 200 OK
   To: Bob &lt;sip:bob@example.com&gt;;tag=skduk2
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 INVITE
   Record-Route:
     &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr&gt;

   At this point, both UAs have the correct route-set for the dialog.
   Any subsequent requests in this dialog will route correctly.  For
   example, the ACK request in message #29 is sent from Alice's UA
   directly to EP2.  The BYE request in message #31 uses the same route-
   set.

   Message #29

   ACK sip:bob@192.0.2.2;transport=tcp SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;;tag=skduk2
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 1 ACK
   Route: &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr&gt;

   Message #31

   BYE sip:bob@192.0.2.2;transport=tcp SIP/2.0
   To: Bob &lt;sip:bob@example.com&gt;;tag=skduk2
   From: Alice &lt;sip:alice@a.example&gt;;tag=02935
   Call-ID: klmvCxVWGp6MxJp2T2mb
   CSeq: 2 BYE
   Route: &lt;sip:wazHDLdIMtUg6r0I/oRZ15zx3zHE1w1Z@ep2.example.com;lr&gt;

<span class="h3"><a class="selflink" id="section-9.4" href="#section-9.4">9.4</a>.  Re-Registration</span>

   Somewhat later, Bob's UA sends keep-alives to both its edge proxies,
   but it discovers that the flow with EP1 failed.  Bob's UA re-
   registers through EP1 using the same reg-id and Call-ID it previously
   used.











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     Bob         EP1   EP2     Proxy     Alice
      |           |     |        |         |
   35)|------2CRLF-----&gt;|        |         |
   36)|&lt;------CRLF------|        |         |
   37)|--2CRLF-&gt;X |     |        |         |
      |           |     |        |         |
   38)|-REGISTER-&gt;|     |        |         |
   39)|           |---REGISTER--&gt;|         |
   40)|           |&lt;----200 OK---|         |
   41)|&lt;-200 OK---|     |        |         |
      |           |     |        |         |

   Message #38

   REGISTER sip:example.com SIP/2.0
   From: Bob &lt;sip:bob@example.com&gt;;tag=7F94778B653B
   To: Bob &lt;sip:bob@example.com&gt;
   Call-ID: 16CB75F21C70
   CSeq: 2 REGISTER
   Supported: path, outbound
   Route: &lt;sip:ep1.example.com;lr&gt;
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp&gt;;reg-id=1
    ;+sip.instance="&lt;urn:uuid:00000000-0000-1000-8000-AABBCCDDEEFF&gt;"

   In message #39, EP1 inserts a Path header with a new flow token:

   Path: &lt;sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr;ob&gt;

<span class="h3"><a class="selflink" id="section-9.5" href="#section-9.5">9.5</a>.  Outgoing Call</span>

   Finally, Bob makes an outgoing call to Alice.  Bob's UA includes an
   "ob" parameter in its Contact URI in message #42.  EP1 adds a Record-
   Route with a flow-token in message #43.  The route-set is returned to
   Bob in the response (messages #45, 46, and 47), and either Bob or
   Alice can send in-dialog requests.
















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     Bob         EP1   EP2     Proxy     Alice
      |           |     |        |         |
   42)|--INVITE--&gt;|     |        |         |
   43)|           |---INVITE----&gt;|         |
   44)|           |     |        |-INVITE-&gt;|
   45)|           |     |        |&lt;--200---|
   46)|           |&lt;----200 OK---|         |
   47)|&lt;-200 OK---|     |        |         |
   48)|--ACK-----&gt;|     |        |         |
   49)|           |-----ACK---------------&gt;|
      |           |     |        |         |
   50)|-- BYE----&gt;|     |        |         |
   51)|           |-----------BYE---------&gt;|
   52)|           |&lt;----------200 OK-------|
   53)|&lt;--200 OK--|     |        |         |
      |           |     |        |         |

   Message #42

   INVITE sip:alice@a.example SIP/2.0
   From: Bob &lt;sip:bob@example.com&gt;;tag=ldw22z
   To: Alice &lt;sip:alice@a.example&gt;
   Call-ID: 95KGsk2V/Eis9LcpBYy3
   CSeq: 1 INVITE
   Route: &lt;sip:ep1.example.com;lr&gt;
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp;ob&gt;

   In message #43, EP1 adds the following Record-Route header.

   Record-Route:
     &lt;sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr&gt;

   When EP1 receives the BYE (message #50) from Bob's UA, it can tell
   that the request is an "outgoing" request (since the source of the
   request matches the flow in the flow token) and simply deletes its
   Route header field value and forwards the request on to Alice's UA.

   Message #50

   BYE sip:alice@a.example SIP/2.0
   From: Bob &lt;sip:bob@example.com&gt;;tag=ldw22z
   To: Alice &lt;sip:alice@a.example&gt;;tag=plqus8
   Call-ID: 95KGsk2V/Eis9LcpBYy3
   CSeq: 2 BYE
   Route: &lt;sip:3yJEbr1GYZK9cPYk5Snocez6DzO7w+AX@ep1.example.com;lr&gt;
   Contact: &lt;sip:bob@192.0.2.2;transport=tcp;ob&gt;





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

   This specification defines a new header field "Flow-Timer", and new
   Contact header field parameters, "reg-id" and "+sip.instance".  The
   grammar includes the definitions from [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>].  Flow-Timer is an
   extension-header from the message-header in the [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] ABNF.

   The ABNF [<a href="./rfc5234" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">RFC5234</a>] is:

    Flow-Timer     = "Flow-Timer" HCOLON 1*DIGIT

    contact-params =/ c-p-reg / c-p-instance

    c-p-reg        = "reg-id" EQUAL 1*DIGIT ; 1 to (2^31 - 1)

    c-p-instance   =  "+sip.instance" EQUAL
                      DQUOTE "&lt;" instance-val "&gt;" DQUOTE

    instance-val   = 1*uric ; defined in <a href="./rfc3261">RFC 3261</a>

   The value of the reg-id MUST NOT be 0 and MUST be less than 2^31.

<span class="h2"><a class="selflink" id="section-11" href="#section-11">11</a>.  IANA Considerations</span>

<span class="h3"><a class="selflink" id="section-11.1" href="#section-11.1">11.1</a>.  Flow-Timer Header Field</span>

   This specification defines a new SIP header field "Flow-Timer" whose
   syntax is defined in <a href="#section-10">Section 10</a>.

     Header Name        compact    Reference
     -----------------  -------    ---------
     Flow-Timer                    [<a href="./rfc5626">RFC5626</a>]

<span class="h3"><a class="selflink" id="section-11.2" href="#section-11.2">11.2</a>.  "reg-id" Contact Header Field Parameter</span>

   This specification defines a new Contact header field parameter
   called reg-id in the "Header Field Parameters and Parameter Values"
   sub-registry as per the registry created by [<a href="./rfc3968" title="&quot;The Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the Session Initiation Protocol (SIP)&quot;">RFC3968</a>].  The syntax is
   defined in <a href="#section-10">Section 10</a>.  The required information is:

                                                  Predefined
   Header Field            Parameter Name         Values      Reference
   ----------------------  ---------------------  ----------  ---------
   Contact                 reg-id                 No          [<a href="./rfc5626">RFC5626</a>]







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<span class="h3"><a class="selflink" id="section-11.3" href="#section-11.3">11.3</a>.  SIP/SIPS URI Parameters</span>

   This specification augments the "SIP/SIPS URI Parameters" sub-
   registry as per the registry created by [<a href="./rfc3969" title="&quot;The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)&quot;">RFC3969</a>].  The required
   information is:

   Parameter Name     Predefined Values     Reference
   --------------     -----------------     ---------
   ob                 No                    [<a href="./rfc5626">RFC5626</a>]

<span class="h3"><a class="selflink" id="section-11.4" href="#section-11.4">11.4</a>.  SIP Option Tag</span>

   This specification registers a new SIP option tag, as per the
   guidelines in <a href="./rfc3261#section-27.1">Section&nbsp;27.1 of [RFC3261]</a>.

   Name:  outbound

   Description:  This option-tag is used to identify UAs and registrars
      that support extensions for Client-Initiated Connections.  A UA
      places this option in a Supported header to communicate its
      support for this extension.  A registrar places this option-tag in
      a Require header to indicate to the registering User Agent that
      the registrar used registrations using the binding rules defined
      in this extension.

<span class="h3"><a class="selflink" id="section-11.5" href="#section-11.5">11.5</a>.  430 (Flow Failed) Response Code</span>

   This document registers a new SIP response code (430 Flow Failed), as
   per the guidelines in <a href="./rfc3261#section-27.4">Section&nbsp;27.4 of [RFC3261]</a>.  This response code
   is used by an edge proxy to indicate to the Authoritative Proxy that
   a specific flow to a UA instance has failed.  Other flows to the same
   instance could still succeed.  The Authoritative Proxy SHOULD attempt
   to forward to another target (flow) with the same instance-id and
   AOR.  Endpoints should never receive a 430 response.  If an endpoint
   receives a 430 response, it should treat it as a 400 (Bad Request)
   per normal procedures, as in <a href="./rfc3261#section-8.1.3.2">Section&nbsp;8.1.3.2 of [RFC3261]</a>.  This
   response code is defined by the following information, which has been
   added to the method and response-code sub-registry under the SIP
   Parameters registry.

     Response Code                               Reference
     ------------------------------------------  ---------
     Request Failure 4xx
       430 Flow Failed                           [<a href="./rfc5626">RFC5626</a>]







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<span class="h3"><a class="selflink" id="section-11.6" href="#section-11.6">11.6</a>.  439 (First Hop Lacks Outbound Support) Response Code</span>

   This document registers a new SIP response code (439 First Hop Lacks
   Outbound Support), as per the guidelines in <a href="./rfc3261#section-27.4">Section&nbsp;27.4 of
   [RFC3261]</a>.  This response code is used by a registrar to indicate
   that it supports the 'outbound' feature described in this
   specification, but that the first outbound proxy that the user is
   attempting to register through does not.  Note that this response
   code is only appropriate in the case that the registering User Agent
   advertises support for outbound processing by including the outbound
   option tag in a Supported header field.  Proxies MUST NOT send a 439
   response to any requests that do not contain a "reg-id" parameter and
   an outbound option tag in a Supported header field.  This response
   code is defined by the following information, which has been added to
   the method and response-code sub-registry under the SIP Parameters
   registry.

     Response Code                               Reference
     ------------------------------------------  ---------
     Request Failure 4xx
       439 First Hop Lacks Outbound Support      [RFC&amp;rfc.number;]

<span class="h3"><a class="selflink" id="section-11.7" href="#section-11.7">11.7</a>.  Media Feature Tag</span>

   This section registers a new media feature tag, per the procedures
   defined in [<a href="./rfc2506" title="&quot;Media Feature Tag Registration Procedure&quot;">RFC2506</a>].  The tag is placed into the sip tree, which is
   defined in [<a href="./rfc3840" title="&quot;Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)&quot;">RFC3840</a>].

   Media feature tag name:  sip.instance

   ASN.1 Identifier:  23

   Summary of the media feature indicated by this tag:  This feature tag
      contains a string containing a URN that indicates a unique
      identifier associated with the UA instance registering the
      Contact.

   Values appropriate for use with this feature tag:  String (equality
      relationship).

   The feature tag is intended primarily for use in the following
      applications, protocols, services, or negotiation mechanisms:
      This feature tag is most useful in a communications application,
      for describing the capabilities of a device, such as a phone or
      PDA.

   Examples of typical use:  Routing a call to a specific device.




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   Related standards or documents:  <a href="./rfc5626">RFC 5626</a>

   Security Considerations:  This media feature tag can be used in ways
      which affect application behaviors.  For example, the SIP caller
      preferences extension [<a href="./rfc3841" title="&quot;Caller Preferences for the Session Initiation Protocol (SIP)&quot;">RFC3841</a>] allows for call routing decisions
      to be based on the values of these parameters.  Therefore, if an
      attacker can modify the values of this tag, they might be able to
      affect the behavior of applications.  As a result, applications
      that utilize this media feature tag SHOULD provide a means for
      ensuring its integrity.  Similarly, this feature tag should only
      be trusted as valid when it comes from the user or User Agent
      described by the tag.  As a result, protocols for conveying this
      feature tag SHOULD provide a mechanism for guaranteeing
      authenticity.

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

   One of the key security concerns in this work is making sure that an
   attacker cannot hijack the sessions of a valid user and cause all
   calls destined to that user to be sent to the attacker.  Note that
   the intent is not to prevent existing active attacks on SIP UDP and
   TCP traffic, but to ensure that no new attacks are added by
   introducing the outbound mechanism.

   The simple case is when there are no edge proxies.  In this case, the
   only time an entry can be added to the routing for a given AOR is
   when the registration succeeds.  SIP already protects against
   attackers being able to successfully register, and this scheme relies
   on that security.  Some implementers have considered the idea of just
   saving the instance-id without relating it to the AOR with which it
   registered.  This idea will not work because an attacker's UA can
   impersonate a valid user's instance-id and hijack that user's calls.

   The more complex case involves one or more edge proxies.  When a UA
   sends a REGISTER request through an edge proxy on to the registrar,
   the edge proxy inserts a Path header field value.  If the
   registration is successfully authenticated, the registrar stores the
   value of the Path header field.  Later, when the registrar forwards a
   request destined for the UA, it copies the stored value of the Path
   header field into the Route header field of the request and forwards
   the request to the edge proxy.

   The only time an edge proxy will route over a particular flow is when
   it has received a Route header that has the flow identifier
   information that it has created.  An incoming request would have
   gotten this information from the registrar.  The registrar will only
   save this information for a given AOR if the registration for the AOR
   has been successful; and the registration will only be successful if



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   the UA can correctly authenticate.  Even if an attacker has spoofed
   some bad information in the Path header sent to the registrar, the
   attacker will not be able to get the registrar to accept this
   information for an AOR that does not belong to the attacker.  The
   registrar will not hand out this bad information to others, and
   others will not be misled into contacting the attacker.

   The Security Considerations discussed in [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] and [<a href="./rfc3327" title="&quot;Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts&quot;">RFC3327</a>] are
   also relevant to this document.  For the security considerations of
   generating flow tokens, please also see <a href="#section-5.2">Section 5.2</a>.  A discussion of
   preventing the avalanche restart problem is in <a href="#section-4.5">Section 4.5</a>.

   This document does not change the mandatory-to-implement security
   mechanisms in SIP.  User Agents are already required to implement
   Digest authentication while support of TLS is recommended; proxy
   servers are already required to implement Digest and TLS.

<span class="h2"><a class="selflink" id="section-13" href="#section-13">13</a>.  Operational Notes on Transports</span>

   This entire section is non-normative.

   [<a id="ref-RFC3261">RFC3261</a>] requires proxies, registrars, and User Agents to implement
   both TCP and UDP but deployments can chose which transport protocols
   they want to use.  Deployments need to be careful in choosing what
   transports to use.  Many SIP features and extensions, such as large
   presence notification bodies, result in SIP requests that can be too
   large to be reasonably transported over UDP.  [<a href="./rfc3261" title="&quot;SIP: Session Initiation Protocol&quot;">RFC3261</a>] states that
   when a request is too large for UDP, the device sending the request
   attempts to switch over to TCP.  It is important to note that when
   using outbound, this will only work if the UA has formed both UDP and
   TCP outbound flows.  This specification allows the UA to do so, but
   in most cases it will probably make more sense for the UA to form a
   TCP outbound connection only, rather than forming both UDP and TCP
   flows.  One of the key reasons that many deployments choose not to
   use TCP has to do with the difficulty of building proxies that can
   maintain a very large number of active TCP connections.  Many
   deployments today use SIP in such a way that the messages are small
   enough that they work over UDP but they can not take advantage of all
   the functionality SIP offers.  Deployments that use only UDP outbound
   connections are going to fail with sufficiently large SIP messages.

<span class="h2"><a class="selflink" id="section-14" href="#section-14">14</a>.  Requirements</span>

   This specification was developed to meet the following requirements:

   1.  Must be able to detect that a UA supports these mechanisms.

   2.  Support UAs behind NATs.



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   3.  Support TLS to a UA without a stable DNS name or IP address.

   4.  Detect failure of a connection and be able to correct for this.

   5.  Support many UAs simultaneously rebooting.

   6.  Support a NAT rebooting or resetting.

   7.  Minimize initial startup load on a proxy.

   8.  Support architectures with edge proxies.

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

   Francois Audet acted as document shepherd for this document, tracking
   hundreds of comments and incorporating many grammatical fixes as well
   as prodding the editors to "get on with it".  Jonathan Rosenberg,
   Erkki Koivusalo, and Byron Campen provided many comments and useful
   text.  Dave Oran came up with the idea of using the most recent
   registration first in the proxy.  Alan Hawrylyshen co-authored the
   document that formed the initial text of this specification.
   Additionally, many of the concepts here originated at a connection
   reuse meeting at IETF 60 that included the authors, Jon Peterson,
   Jonathan Rosenberg, Alan Hawrylyshen, and Paul Kyzivat.  The TCP
   design team consisting of Chris Boulton, Scott Lawrence, Rajnish
   Jain, Vijay K. Gurbani, and Ganesh Jayadevan provided input and text.
   Nils Ohlmeier provided many fixes and initial implementation
   experience.  In addition, thanks to the following folks for useful
   comments: Francois Audet, Flemming Andreasen, Mike Hammer, Dan Wing,
   Srivatsa Srinivasan, Dale Worely, Juha Heinanen, Eric Rescorla,
   Lyndsay Campbell, Christer Holmberg, Kevin Johns, Jeroen van Bemmel,
   Derek MacDonald, Dean Willis, and Robert Sparks.

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

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

   [<a id="ref-RFC2119">RFC2119</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-RFC2141">RFC2141</a>]      Moats, R., "URN Syntax", <a href="./rfc2141">RFC 2141</a>, May 1997.

   [<a id="ref-RFC2506">RFC2506</a>]      Holtman, K., Mutz, A., and T. Hardie, "Media Feature
                  Tag Registration Procedure", <a href="https://www.rfc-editor.org/bcp/bcp31">BCP 31</a>, <a href="./rfc2506">RFC 2506</a>,
                  March 1999.






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   [<a id="ref-RFC3261">RFC3261</a>]      Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                  Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                  and E. Schooler, "SIP: Session Initiation Protocol",
                  <a href="./rfc3261">RFC 3261</a>, June 2002.

   [<a id="ref-RFC3263">RFC3263</a>]      Rosenberg, J. and H. Schulzrinne, "Session Initiation
                  Protocol (SIP): Locating SIP Servers", <a href="./rfc3263">RFC 3263</a>,
                  June 2002.

   [<a id="ref-RFC3327">RFC3327</a>]      Willis, D. and B. Hoeneisen, "Session Initiation
                  Protocol (SIP) Extension Header Field for Registering
                  Non-Adjacent Contacts", <a href="./rfc3327">RFC 3327</a>, December 2002.

   [<a id="ref-RFC3581">RFC3581</a>]      Rosenberg, J. and H. Schulzrinne, "An Extension to the
                  Session Initiation Protocol (SIP) for Symmetric
                  Response Routing", <a href="./rfc3581">RFC 3581</a>, August 2003.

   [<a id="ref-RFC3629">RFC3629</a>]      Yergeau, F., "UTF-8, a transformation format of ISO
                  10646", STD 63, <a href="./rfc3629">RFC 3629</a>, November 2003.

   [<a id="ref-RFC3840">RFC3840</a>]      Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                  "Indicating User Agent Capabilities in the Session
                  Initiation Protocol (SIP)", <a href="./rfc3840">RFC 3840</a>, August 2004.

   [<a id="ref-RFC3841">RFC3841</a>]      Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
                  "Caller Preferences for the Session Initiation
                  Protocol (SIP)", <a href="./rfc3841">RFC 3841</a>, August 2004.

   [<a id="ref-RFC3968">RFC3968</a>]      Camarillo, G., "The Internet Assigned Number Authority
                  (IANA) Header Field Parameter Registry for the Session
                  Initiation Protocol (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp98">BCP 98</a>, <a href="./rfc3968">RFC 3968</a>,
                  December 2004.

   [<a id="ref-RFC3969">RFC3969</a>]      Camarillo, G., "The Internet Assigned Number Authority
                  (IANA) Uniform Resource Identifier (URI) Parameter
                  Registry for the Session Initiation Protocol (SIP)",
                  <a href="https://www.rfc-editor.org/bcp/bcp99">BCP 99</a>, <a href="./rfc3969">RFC 3969</a>, December 2004.

   [<a id="ref-RFC4122">RFC4122</a>]      Leach, P., Mealling, M., and R. Salz, "A Universally
                  Unique IDentifier (UUID) URN Namespace", <a href="./rfc4122">RFC 4122</a>,
                  July 2005.

   [<a id="ref-RFC5234">RFC5234</a>]      Crocker, D. and P. Overell, "Augmented BNF for Syntax
                  Specifications: ABNF", STD 68, <a href="./rfc5234">RFC 5234</a>, January 2008.

   [<a id="ref-RFC5389">RFC5389</a>]      Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
                  "Session Traversal Utilities for NAT (STUN)",
                  <a href="./rfc5389">RFC 5389</a>, October 2008.



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<span class="grey"><a href="./rfc5626">RFC 5626</a>          Client-Initiated Connections in SIP       October 2009</span>


<span class="h3"><a class="selflink" id="section-16.2" href="#section-16.2">16.2</a>.  Informative References</span>

   [<a id="ref-CONFIG-FMWK">CONFIG-FMWK</a>]  Petrie, D. and S. Channabasappa, Ed., "A Framework for
                  Session Initiation Protocol User Agent Profile
                  Delivery", Work in Progress, February 2008.

   [<a id="ref-NAT-SCEN">NAT-SCEN</a>]     Boulton, C., Rosenberg, J., Camarillo, G., and F.
                  Audet, "Best Current Practices for NAT Traversal for
                  Client-Server SIP", Work in Progress, September 2008.

   [<a id="ref-RFC0768">RFC0768</a>]      Postel, J., "User Datagram Protocol", STD 6, <a href="./rfc768">RFC 768</a>,
                  August 1980.

   [<a id="ref-RFC0793">RFC0793</a>]      Postel, J., "Transmission Control Protocol", STD 7,
                  <a href="./rfc793">RFC 793</a>, September 1981.

   [<a id="ref-RFC1035">RFC1035</a>]      Mockapetris, P., "Domain names - implementation and
                  specification", STD 13, <a href="./rfc1035">RFC 1035</a>, November 1987.

   [<a id="ref-RFC2104">RFC2104</a>]      Krawczyk, H., Bellare, M., and R. Canetti, "HMAC:
                  Keyed-Hashing for Message Authentication", <a href="./rfc2104">RFC 2104</a>,
                  February 1997.

   [<a id="ref-RFC2131">RFC2131</a>]      Droms, R., "Dynamic Host Configuration Protocol",
                  <a href="./rfc2131">RFC 2131</a>, March 1997.

   [<a id="ref-RFC2782">RFC2782</a>]      Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR
                  for specifying the location of services (DNS SRV)",
                  <a href="./rfc2782">RFC 2782</a>, February 2000.

   [<a id="ref-RFC3320">RFC3320</a>]      Price, R., Bormann, C., Christoffersson, J., Hannu,
                  H., Liu, Z., and J. Rosenberg, "Signaling Compression
                  (SigComp)", <a href="./rfc3320">RFC 3320</a>, January 2003.

   [<a id="ref-RFC3489">RFC3489</a>]      Rosenberg, J., Weinberger, J., Huitema, C., and R.
                  Mahy, "STUN - Simple Traversal of User Datagram
                  Protocol (UDP) Through Network Address Translators
                  (NATs)", <a href="./rfc3489">RFC 3489</a>, March 2003.

   [<a id="ref-RFC3986">RFC3986</a>]      Berners-Lee, T., Fielding, R., and L. Masinter,
                  "Uniform Resource Identifier (URI): Generic Syntax",
                  STD 66, <a href="./rfc3986">RFC 3986</a>, January 2005.

   [<a id="ref-RFC4340">RFC4340</a>]      Kohler, E., Handley, M., and S. Floyd, "Datagram
                  Congestion Control Protocol (DCCP)", <a href="./rfc4340">RFC 4340</a>,
                  March 2006.





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   [<a id="ref-RFC4648">RFC4648</a>]      Josefsson, S., "The Base16, Base32, and Base64 Data
                  Encodings", <a href="./rfc4648">RFC 4648</a>, October 2006.

   [<a id="ref-RFC4960">RFC4960</a>]      Stewart, R., "Stream Control Transmission Protocol",
                  <a href="./rfc4960">RFC 4960</a>, September 2007.

   [<a id="ref-RFC5246">RFC5246</a>]      Dierks, T. and E. Rescorla, "The Transport Layer
                  Security (TLS) Protocol Version 1.2", <a href="./rfc5246">RFC 5246</a>,
                  August 2008.

   [<a id="ref-RFC5627">RFC5627</a>]      Rosenberg, J., "Obtaining and Using Globally Routable
                  User Agent URIs (GRUUs) in the Session Initiation
                  Protocol (SIP)", <a href="./rfc5627">RFC 5627</a>, October 2009.






































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<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  Default Flow Registration Backoff Times</span>

   The base-time used for the flow re-registration backoff times
   described in <a href="#section-4.5">Section 4.5</a> are configurable.  If the base-time-all-fail
   value is set to the default of 30 seconds and the base-time-not-
   failed value is set to the default of 90 seconds, the following table
   shows the resulting amount of time the UA will wait to retry
   registration.

     +-------------------+--------------------+---------------------+
     | # of reg failures | all flows unusable | &gt; 1 non-failed flow |
     +-------------------+--------------------+---------------------+
     | 0                 | 0 s                | 0 s                 |
     | 1                 | 30-60 s            | 90-180 s            |
     | 2                 | 1-2 min            | 3-6 min             |
     | 3                 | 2-4 min            | 6-12 min            |
     | 4                 | 4-8 min            | 12-24 min           |
     | 5                 | 8-16 min           | 15-30 min           |
     | 6 or more         | 15-30 min          | 15-30 min           |
     +-------------------+--------------------+---------------------+

<span class="h2"><a class="selflink" id="appendix-B" href="#appendix-B">Appendix B</a>.  ABNF</span>

   This appendix contains the ABNF defined earlier in this document.


      CRLF = CR LF
      double-CRLF = CR LF CR LF
      CR = %x0D
      LF = %x0A

      Flow-Timer     = "Flow-Timer" HCOLON 1*DIGIT

      contact-params =/ c-p-reg / c-p-instance

      c-p-reg        = "reg-id" EQUAL 1*DIGIT ; 1 to (2^31 - 1)

      c-p-instance   =  "+sip.instance" EQUAL
                        DQUOTE "&lt;" instance-val "&gt;" DQUOTE

      instance-val   = 1*uric ; defined in <a href="./rfc3261">RFC 3261</a>










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Authors' Addresses

   Cullen Jennings (editor)
   Cisco Systems
   170 West Tasman Drive
   Mailstop SJC-21/2
   San Jose, CA  95134
   USA

   Phone: +1 408 902-3341
   EMail: fluffy@cisco.com


   Rohan Mahy (editor)
   Unaffiliated

   EMail: rohan@ekabal.com


   Francois Audet (editor)
   Skype Labs

   EMail: francois.audet@skypelabs.com




























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