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Network Working Group                                       J. Rosenberg
Request for Comments: 3261                                   dynamicsoft
Obsoletes: 2543                                           H. Schulzrinne
Category: Standards Track                                    Columbia U.
                                                            G. Camarillo
                                                                Ericsson
                                                             A. Johnston
                                                                WorldCom
                                                             J. Peterson
                                                                 Neustar
                                                               R. Sparks
                                                             dynamicsoft
                                                              M. Handley
                                                                    ICIR
                                                             E. Schooler
                                                                    AT&T
                                                               June 2002

                    SIP: Session Initiation Protocol

Status of this Memo

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

Copyright Notice

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

Abstract

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

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



Rosenberg, et. al.          Standards Track                     [Page 1]

RFC 3261            SIP: Session Initiation Protocol           June 2002


Table of Contents

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



Rosenberg, et. al.          Standards Track                     [Page 2]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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



Rosenberg, et. al.          Standards Track                     [Page 3]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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



Rosenberg, et. al.          Standards Track                     [Page 4]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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



Rosenberg, et. al.          Standards Track                     [Page 5]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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



Rosenberg, et. al.          Standards Track                     [Page 6]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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



Rosenberg, et. al.          Standards Track                     [Page 7]

RFC 3261            SIP: Session Initiation Protocol           June 2002


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

1 Introduction

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



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

2 Overview of SIP Functionality

   SIP is an application-layer control protocol that can establish,
   modify, and terminate multimedia sessions (conferences) such as
   Internet telephony calls.  SIP can also invite participants to
   already existing sessions, such as multicast conferences.  Media can
   be added to (and removed from) an existing session.  SIP
   transparently supports name mapping and redirection services, which
   supports personal mobility [27] - users can maintain a single
   externally visible identifier regardless of their network location.

   SIP supports five facets of establishing and terminating multimedia
   communications:

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

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

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

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

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

   SIP is not a vertically integrated communications system.  SIP is
   rather a component that can be used with other IETF protocols to
   build a complete multimedia architecture.  Typically, these
   architectures will include protocols such as the Real-time Transport
   Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
   providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
   2326 [29]) for controlling delivery of streaming media, the Media



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   Gateway Control Protocol (MEGACO) (RFC 3015 [30]) for controlling
   gateways to the Public Switched Telephone Network (PSTN), and the
   Session Description Protocol (SDP) (RFC 2327 [1]) for describing
   multimedia sessions.  Therefore, SIP should be used in conjunction
   with other protocols in order to provide complete services to the
   users.  However, the basic functionality and operation of SIP does
   not depend on any of these protocols.

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

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

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

   SIP works with both IPv4 and IPv6.

3 Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
   RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
   described in BCP 14, RFC 2119 [2] and indicate requirement levels for
   compliant SIP implementations.

4 Overview of Operation

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





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

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

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

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








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

         Figure 1: SIP session setup example with SIP trapezoid

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

      (Alice's SDP not shown)

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






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

   To contains a display name (Bob) and a SIP or SIPS URI
   (sip:bob@biloxi.com) towards which the request was originally
   directed.  Display names are described in RFC 2822 [3].

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

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

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

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

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

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

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

   The complete set of SIP header fields is defined in Section 20.

   The details of the session, such as the type of media, codec, or
   sampling rate, are not described using SIP.  Rather, the body of a
   SIP message contains a description of the session, encoded in some
   other protocol format.  One such format is the Session Description
   Protocol (SDP) (RFC 2327 [1]).  This SDP message (not shown in the




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

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

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

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




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

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

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

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

      (Bob's SDP not shown)

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



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

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

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

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

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






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

   Section 24.2 describes the messages shown in Figure 1 in full.

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

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

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




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

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

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

   The complete set of SIP message details for this registration example
   is in Section 24.1.

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

5 Structure of the Protocol

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

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

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






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   The second layer is the transport layer.  It defines how a client
   sends requests and receives responses and how a server receives
   requests and sends responses over the network.  All SIP elements
   contain a transport layer.  The transport layer is described in
   Section 18.

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

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

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




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

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

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

6 Definitions

   The following terms have special significance for SIP.

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

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






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

      Call Leg: Another name for a dialog [31]; no longer used in this
         specification.

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

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

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

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

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

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

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

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

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



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

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

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

      Informational Response: Same as a provisional response.

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

      Invitation: An INVITE request.

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

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

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

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





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

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

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

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

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

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

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

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

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






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

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

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

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

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

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

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

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

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

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



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

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

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

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

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

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

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




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

      URL-encoded: A character string encoded according to RFC 2396,
         Section 2.4 [5].

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

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

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

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

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

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

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

7 SIP Messages

   SIP is a text-based protocol and uses the UTF-8 charset (RFC 2279
   [7]).



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

   Both Request (section 7.1) and Response (section 7.2) messages use
   the basic format of RFC 2822 [3], even though the syntax differs in
   character set and syntax specifics.  (SIP allows header fields that
   would not be valid RFC 2822 header fields, for example.)  Both types
   of messages consist of a start-line, one or more header fields, an
   empty line indicating the end of the header fields, and an optional
   message-body.

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

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

   Except for the above difference in character sets, much of SIP's
   message and header field syntax is identical to HTTP/1.1.  Rather
   than repeating the syntax and semantics here, we use [HX.Y] to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [8]).

   However, SIP is not an extension of HTTP.

7.1 Requests

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

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

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

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





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      Request-URI: The Request-URI is a SIP or SIPS URI as described in
           Section 19.1 or a general URI (RFC 2396 [5]).  It indicates
           the user or service to which this request is being addressed.
           The Request-URI MUST NOT contain unescaped spaces or control
           characters and MUST NOT be enclosed in "<>".

           SIP elements MAY support Request-URIs with schemes other than
           "sip" and "sips", for example the "tel" URI scheme of RFC
           2806 [9].  SIP elements MAY translate non-SIP URIs using any
           mechanism at their disposal, resulting in SIP URI, SIPS URI,
           or some other scheme.

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

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

7.2 Responses

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

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

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

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

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





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

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

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

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

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

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

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

   Section 21 defines these classes and describes the individual codes.

7.3 Header Fields

   SIP header fields are similar to HTTP header fields in both syntax
   and semantics.  In particular, SIP header fields follow the [H4.2]
   definitions of syntax for the message-header and the rules for
   extending header fields over multiple lines.  However, the latter is
   specified in HTTP with implicit whitespace and folding.  This
   specification conforms to RFC 2234 [10] and uses only explicit
   whitespace and folding as an integral part of the grammar.

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

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

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



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7.3.1 Header Field Format

   Header fields follow the same generic header format as that given in
   Section 2.2 of RFC 2822 [3].  Each header field consists of a field
   name followed by a colon (":") and the field value.

      field-name: field-value

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

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

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

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

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

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



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   field rows with these names MAY be present in a message, but since
   their grammar does not follow the general form listed in Section 7.3,
   they MUST NOT be combined into a single header field row.

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

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

      Route: <sip:alice@atlanta.com>
      Subject: Lunch
      Route: <sip:bob@biloxi.com>
      Route: <sip:carol@chicago.com>

      Route: <sip:alice@atlanta.com>, <sip:bob@biloxi.com>
      Route: <sip:carol@chicago.com>
      Subject: Lunch

      Subject: Lunch
      Route: <sip:alice@atlanta.com>, <sip:bob@biloxi.com>,
             <sip:carol@chicago.com>

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

      Route: <sip:alice@atlanta.com>
      Route: <sip:bob@biloxi.com>
      Route: <sip:carol@chicago.com>

      Route: <sip:bob@biloxi.com>
      Route: <sip:alice@atlanta.com>
      Route: <sip:carol@chicago.com>

      Route: <sip:alice@atlanta.com>,<sip:carol@chicago.com>,
             <sip:bob@biloxi.com>

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

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






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

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

      Contact: <sip:alice@atlanta.com>;expires=3600

   is equivalent to

      CONTACT: <sip:alice@atlanta.com>;ExPiReS=3600

   and

      Content-Disposition: session;handling=optional

   is equivalent to

      content-disposition: Session;HANDLING=OPTIONAL

   The following two header fields are not equivalent:

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

7.3.2 Header Field Classification

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

7.3.3 Compact Form

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



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

7.4 Bodies

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

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

7.4.1 Message Body Type

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

   The "multipart" MIME type defined in RFC 2046 [11] MAY be used within
   the body of the message.  Implementations that send requests
   containing multipart message bodies MUST send a session description
   as a non-multipart message body if the remote implementation requests
   this through an Accept header field that does not contain multipart.

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

7.4.2 Message Body Length

   The body length in bytes is provided by the Content-Length header
   field.  Section 20.14 describes the necessary contents of this header
   field in detail.

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







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7.5 Framing SIP Messages

   Unlike HTTP, SIP implementations can use UDP or other unreliable
   datagram protocols.  Each such datagram carries one request or
   response.  See Section 18 on constraints on usage of unreliable
   transports.

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

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

8 General User Agent Behavior

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

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

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

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

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




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8.1 UAC Behavior

   This section covers UAC behavior outside of a dialog.

8.1.1 Generating the Request

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

   Examples of requests sent outside of a dialog include an INVITE to
   establish a session (Section 13) and an OPTIONS to query for
   capabilities (Section 11).

8.1.1.1 Request-URI

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

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

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






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8.1.1.2 To

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

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

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

   For further information on the To header field, see Section 20.39.
   The following is an example of a valid To header field:

      To: Carol <sip:carol@chicago.com>




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8.1.1.3 From

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

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

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

   The From field MUST contain a new "tag" parameter, chosen by the UAC.
   See Section 19.3 for details on choosing a tag.

   For further information on the From header field, see Section 20.20.
   Examples:

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

8.1.1.4 Call-ID

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

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



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

   Use of cryptographically random identifiers (RFC 1750 [12]) in the
   generation of Call-IDs is RECOMMENDED.  Implementations MAY use the
   form "localid@host".  Call-IDs are case-sensitive and are simply
   compared byte-by-byte.

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

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

   For further information on the Call-ID header field, see Section
   20.8.

   Example:

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

8.1.1.5 CSeq

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

   Section 12.2.1.1 discusses construction of the CSeq for requests
   within a dialog.

   Example:

      CSeq: 4711 INVITE










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8.1.1.6 Max-Forwards

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

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

8.1.1.7 Via

   The Via header field indicates the transport used for the transaction
   and identifies the location where the response is to be sent.  A Via
   header field value is added only after the transport that will be
   used to reach the next hop has been selected (which may involve the
   usage of the procedures in [4]).

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

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

      The uniqueness property of the branch ID parameter, to facilitate
      its use as a transaction ID, was not part of RFC 2543.

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



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

   The Via header maddr, ttl, and sent-by components will be set when
   the request is processed by the transport layer (Section 18).

   Via processing for proxies is described in Section 16.6 Item 8 and
   Section 16.7 Item 3.

8.1.1.8 Contact

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

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

   For further information on the Contact header field, see Section
   20.10.

8.1.1.9 Supported and Require

   If the UAC supports extensions to SIP that can be applied by the
   server to the response, the UAC SHOULD include a Supported header
   field in the request listing the option tags (Section 19.2) for those
   extensions.

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

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



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

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

8.1.1.10 Additional Message Components

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

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

8.1.2 Sending the Request

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

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



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

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

   The UAC SHOULD follow the procedures defined in [4] for stateful
   elements, trying each address until a server is contacted.  Each try
   constitutes a new transaction, and therefore each carries a different
   topmost Via header field value with a new branch parameter.
   Furthermore, the transport value in the Via header field is set to
   whatever transport was determined for the target server.

8.1.3 Processing Responses

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

8.1.3.1 Transaction Layer Errors

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

8.1.3.2 Unrecognized Responses

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





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8.1.3.3 Vias

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

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

8.1.3.4 Processing 3xx Responses

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

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

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

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






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

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

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

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

      sip:user@host?Subject=foo&Call-Info=<http://www.foo.com>

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

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

   Finally, once the new request has been constructed, it is sent using
   a new client transaction, and therefore MUST have a new branch ID in
   the top Via field as discussed in Section 8.1.1.7.






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

   In some instances, Contact header field values may be cached at UAC
   temporarily or permanently depending on the status code received and
   the presence of an expiration interval; see Sections 21.3.2 and
   21.3.3.

8.1.3.5 Processing 4xx Responses

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

   If a 401 (Unauthorized) or 407 (Proxy Authentication Required)
   response is received, the UAC SHOULD follow the authorization
   procedures of Section 22.2 and Section 22.3 to retry the request with
   credentials.

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

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

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

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

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



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

8.2 UAS Behavior

   When a request outside of a dialog is processed by a UAS, there is a
   set of processing rules that are followed, independent of the method.
   Section 12 gives guidance on how a UAS can tell whether a request is
   inside or outside of a dialog.

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

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

8.2.1 Method Inspection

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

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

8.2.2 Header Inspection

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

8.2.2.1 To and Request-URI

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



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

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

8.2.2.2 Merged Requests

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

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

8.2.2.3 Require

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

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



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

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

   Example:

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

      UAS->UAC:   SIP/2.0 420 Bad Extension
                  Unsupported: 100rel

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

8.2.3 Content Processing

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





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8.2.4 Applying Extensions

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

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

8.2.5 Processing the Request

   Assuming all of the checks in the previous subsections are passed,
   the UAS processing becomes method-specific.  Section 10 covers the
   REGISTER request, Section 11 covers the OPTIONS request, Section 13
   covers the INVITE request, and Section 15 covers the BYE request.

8.2.6 Generating the Response

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

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

8.2.6.1 Sending a Provisional Response

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





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

8.2.6.2 Headers and Tags

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

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

8.2.7 Stateless UAS Behavior

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

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





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   transaction state that might slow or completely halt call processing
   in a UAS, effectively creating a denial of service condition; for
   more information see Section 26.1.5.

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

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

      o  A stateless UAS MUST NOT retransmit responses.

      o  A stateless UAS MUST ignore ACK requests.

      o  A stateless UAS MUST ignore CANCEL requests.

      o  To header tags MUST be generated for responses in a stateless
         manner - in a manner that will generate the same tag for the
         same request consistently.  For information on tag construction
         see Section 19.3.

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

8.3 Redirect Servers

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

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

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

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



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

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

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

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

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

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

   A Contact response header field can contain any suitable URI
   indicating where the called party can be reached, not limited to SIP
   URIs.  For example, it could contain URIs for phones, fax, or irc (if
   they were defined) or a mailto:  (RFC 2368 [32]) URL.  Section 26.4.4
   discusses implications and limitations of redirecting a SIPS URI to a
   non-SIPS URI.

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




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

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

9 Canceling a Request

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

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

   CANCEL requests can be constructed and sent by both proxies and user
   agent clients.  Section 15 discusses under what conditions a UAC
   would CANCEL an INVITE request, and Section 16.10 discusses proxy
   usage of CANCEL.

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

9.1 Client Behavior

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

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





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

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

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

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

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

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

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

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




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   defined in Section 17.1.1.1), the client SHOULD then consider the
   original transaction cancelled and SHOULD destroy the client
   transaction handling the original request.

9.2 Server Behavior

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

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

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

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

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



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10 Registrations

10.1 Overview

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

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

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

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

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



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

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

10.2 Constructing the REGISTER Request

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

   Except as noted, the construction of the REGISTER request and the
   behavior of clients sending a REGISTER request is identical to the
   general UAC behavior described in Section 8.1 and Section 17.1.

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

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

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

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





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

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

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

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

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

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



























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

                      Figure 2: REGISTER example

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

      action: The "action" parameter from RFC 2543 has been deprecated.
           UACs SHOULD NOT use the "action" parameter.

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

10.2.1 Adding Bindings

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






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

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

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

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

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

10.2.1.1 Setting the Expiration Interval of Contact Addresses

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

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





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

10.2.1.2 Preferences among Contact Addresses

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

10.2.2 Removing Bindings

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

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

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

10.2.3 Fetching Bindings

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

10.2.4 Refreshing Bindings

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






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

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

10.2.5 Setting the Internal Clock

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

10.2.6 Discovering a Registrar

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

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

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

10.2.7 Transmitting a Request

   Once the REGISTER method has been constructed, and the destination of
   the message identified, UACs follow the procedures described in
   Section 8.1.2 to hand off the REGISTER to the transaction layer.



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

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

10.2.8 Error Responses

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

10.3 Processing REGISTER Requests

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

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

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

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

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






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

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

      2. To guarantee that the registrar supports any necessary
         extensions, the registrar MUST process the Require header field
         values as described for UASs in Section 8.2.2.

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

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

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

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










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

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

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

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

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

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

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



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

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

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

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

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

11 Querying for Capabilities

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

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




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

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

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

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

11.1 Construction of OPTIONS Request

   An OPTIONS request is constructed using the standard rules for a SIP
   request as discussed in Section 8.1.1.

   A Contact header field MAY be present in an OPTIONS.

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

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

   Example OPTIONS request:

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




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11.2 Processing of OPTIONS Request

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

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

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

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

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

   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).  If the types
   include one that can describe media capabilities, the UAS SHOULD
   include a body in the response for that purpose.  Details on the
   construction of such a body in the case of application/sdp are
   described in [13].









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   Example OPTIONS response generated by a UAS (corresponding to the
   request in Section 11.1):

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

      (SDP not shown)

12 Dialogs

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

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

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



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

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

12.1 Creation of a Dialog

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

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

12.1.1 UAS behavior

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



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

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

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

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

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

      This is to maintain backwards compatibility with RFC 2543, which
      did not mandate From tags.

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

12.1.2 UAC Behavior

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



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

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

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

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

      This is to maintain backwards compatibility with RFC 2543, which
      did not mandate To tags.

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

12.2 Requests within a Dialog

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

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




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   INVITE, the only target refresh request defined is re-INVITE (see
   Section 14).  Other extensions may define different target refresh
   requests for dialogs established in other ways.

      Note that an ACK is NOT a target refresh request.

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

12.2.1 UAC Behavior

12.2.1.1 Generating the Request

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

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

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

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





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

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

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

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

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

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

      <sip:proxy1>,<sip:proxy2>,<sip:proxy3;lr>,<sip:proxy4>

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

   METHOD sip:proxy1
   Route: <sip:proxy2>,<sip:proxy3;lr>,<sip:proxy4>,<sip:user@remoteua>

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




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

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

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

   The rest of the request is formed as described in Section 8.1.1.

   Once the request has been constructed, the address of the server is
   computed and the request is sent, using the same procedures for
   requests outside of a dialog (Section 8.1.2).

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

12.2.1.2 Processing the Responses

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

   The behavior of a UAC that receives a 3xx response for a request sent
   within a dialog is the same as if the request had been sent outside a
   dialog.  This behavior is described in Section 8.1.3.4.

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

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





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

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

12.2.2 UAS Behavior

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

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

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

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

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






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

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

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

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

12.3 Termination of a Dialog

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

13 Initiating a Session

13.1 Overview

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



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

   After possibly receiving one or more provisional responses, the UAC
   will get one or more 2xx responses or one non-2xx final response.
   Because of the protracted amount of time it can take to receive final
   responses to INVITE, the reliability mechanisms for INVITE
   transactions differ from those of other requests (like OPTIONS).
   Once it receives a final response, the UAC needs to send an ACK for
   every final response it receives.  The procedure for sending this ACK
   depends on the type of response.  For final responses between 300 and
   699, the ACK process