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Versions: 00 01                                                         
Internet Engineering Task Force                                   SIP WG
Internet Draft                                              G. Camarillo
                                                                Ericsson
draft-camarillo-sipping-transc-b2bua-01.txt
February 7, 2004
Expires: August, 2004


  The Session Initiation Protocol Conference Bridge Transcoding Model

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.


Abstract

   This document describes how to invoke transcoding services using the
   conference bridge model. This way of invocation meets the
   requirements for SIP regarding transcoding services invocation to
   support deaf, hard of hearing and speech-impaired individuals.













G. Camarillo                                                  [Page 1]


Internet Draft                    SIP                   February 7, 2004





                           Table of Contents



   1          Introduction ........................................    3
   2          Caller's Invocation .................................    3
   2.1        Unsuccessful Session Establishment ..................    5
   3          Callee's Invocation .................................    6
   4          Security Considerations .............................    7
   5          Contributors ........................................    7
   6          OPEN ISSUES .........................................    7
   7          Authors' Addresses ..................................    8
   8          Bibliography ........................................    9



































G. Camarillo                                                  [Page 2]


Internet Draft                    SIP                   February 7, 2004


1 Introduction

   The framework for transcoding with SIP [1] (draft-ietf-sipping-
   transc-framework) describes how two SIP UAs can discover
   imcompatibilities that prevent them from establishing a session
   (e.g., lack of support for a common codec or for a common media
   type). When such incompatibilities are found, the UAs need to invoke
   transcoding services to successfully establish the session. Using the
   conference bridge model is one way to perform such invocation.

   In the conference bridge model for transcoding invocation, a
   transcoding server that provides a particular transcoding service
   (e.g., speech-to-text) behaves as a B2BUA between both UAs and is
   identified by a URI.

2 Caller's Invocation

   Figure 1 shows the message flow for the caller's invocation of a
   transcoder T. The caller (A) sends an INVITE (1) to the transcoder
   (T) to establish the session A-T. The URI in the Request-URI of this
   INVITE contains a list parameter, as defined in [2] (draft-
   camarillo-sipping-uri-list-01), with a pointer to a URI list. This
   URI list contains a single URI: the callee's URI, as shown below:


   INVITE sip:transcoder@example.com;list=cid:cn35t8@example.com SIP/2.0
   Via: SIP/2.0/TCP client.chicago.example.com
       ;branch=z9hG4bKhjhs8ass83
   Max-Forwards: 70
   To: "Transcoder" <sip:transcoder@example.com>
   From: Caller <sip:caller@chicago.example.com>;tag=32331
   Call-ID: d432fa84b4c76e66710
   CSeq: 1 INVITE
   Contact: <sip:caller@client.chicago.example.com>
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY
   Conten-Type: multipart/mixed;boundary="boundary1"
   Content-Length: xxx

   --boundary1
   Content-Type: application/sdp
   Content-Length: xxx

   v=0
   o=caller 2890844526 2890842807 IN IP4 chicago.example.com
   s=Example Subject
   c=IN IP4 192.0.0.1
   t=0 0



G. Camarillo                                                  [Page 3]


Internet Draft                    SIP                   February 7, 2004


   m=audio 20000 RTP/AVP 0

   --boundary1
   Content-Type: application/resource-lists+xml
   Content-Length: 367
   Content-ID: <cn35t8@example.com>

   <?xml version="1.0" encoding="UTF-8"?>
   <resource-lists xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
     <list name="ad-hoc-1">
       <entry name="1" uri="sip:callee@example2.com" />
     </list>
   </resource-lists>
   --boundary1--





    A                           T                           B
    |                           |                           |
    |-----(1) INVITE SDP A----->|                           |
    |                           |                           |
    |<-(2) 183 Session Progress-|                           |
    |                           |-----(3) INVITE SDP TB---->|
    |                           |                           |
    |                           |<-----(4) 200 OK SDP B-----|
    |                           |                           |
    |                           |---------(5) ACK---------->|
    |<----(6) 200 OK SDP TA-----|                           |
    |                           |                           |
    |---------(7) ACK---------->|                           |
    |                           |                           |
    | ************************* | ************************* |
    |**        Media          **|**        Media          **|
    | ************************* | ************************* |
    |                           |                           |


   Figure 1: Successful invocation of a transcoder by the caller



   On reception of the INVITE, the transcoder generates a new INVITE
   towards the callee. The transcoder acts as a B2BUA, so, this new
   INVITE (3) belongs to a different transaction than the INVITE (1)
   received by the transcoder.

   When the transcoder receives a final response (4) from the callee, it



G. Camarillo                                                  [Page 4]


Internet Draft                    SIP                   February 7, 2004


   generates a new final response (6) for INVITE (1). This new final
   response (6) has the same status code as the one received in the
   response from the callee (4).

   The advantage of this message flow is that, for both user agents, is
   indentical to the flow for establishing a regular session (i.e.,
   without transcoder) between them. Additionaly, the only difference in
   the message contents is that the caller needs to use a list parameter
   in the Request-URI of the initial INVITE.

2.1 Unsuccessful Session Establishment

   Figure 2 shows a similar message flow as the one in Figure 1.
   Nevertheless, this time the callee generates a non-2xx final response
   (4). Consequently, the transcoder generates a non-2xx final response
   (6) towards the caller as well.



    A                           T                           B
    |                           |                           |
    |-----(1) INVITE SDP A----->|                           |
    |                           |                           |
    |<-(2) 183 Session Progress-|                           |
    |                           |-----(3) INVITE SDP TB---->|
    |                           |                           |
    |                           |<----(4) 404 Not Found-----|
    |                           |                           |
    |                           |---------(5) ACK---------->|
    |<----(6) 404 Not Found-----|                           |
    |                           |                           |
    |---------(7) ACK---------->|                           |
    |                           |                           |


   Figure 2: Unsuccessful session establishment



   The problem with this flow is that the caller does not know whether
   the 404 (Not Found) response means that the initial INVITE (1) did
   not reach the transcoder or that the INVITE generated by the
   transcoder (4) did not reach the callee. To resolve this, it is
   recommended that the caller uses the reliable provisional responses
   [3] SIP extension.

   Figure 3 shows the resulting message flow when the caller requires
   the use of the reliable provisional responses [3] SIP extension. The
   repection of the 183 (Session Progress) reliable provisional response



G. Camarillo                                                  [Page 5]


Internet Draft                    SIP                   February 7, 2004


   informs the caller that the transcoder was contacted susccessfully.
   So, the 404 (Not Found) response indicates that the callee could not
   be reached.



    A                               T                           B
    |                               |                           |
    |--------(1) INVITE SDP A------>|                           |
    |                               |                           |
    |<-(2) 183 S. Prog. SDP on hold-|                           |
    |                               |-----(3) INVITE SDP TB---->|
    |                               |                           |
    |-----------(4) PRACK---------->|                           |
    |                               |                           |
    |<----------(5) 200 OK----------|                           |
    |                               |                           |
    |                               |<----(6) 404 Not Found-----|
    |                               |                           |
    |                               |---------(7) ACK---------->|
    |<-------(8) 404 Not Found------|                           |
    |                               |                           |
    |-------------(9) ACK---------->|                           |
    |                               |                           |


   Figure 3: Invocation using reliable provisional responses



3 Callee's Invocation

   If a UA receives an INVITE with an offer that is not acceptable, it
   can only invoke a transcoder if the caller supports the Replaces [4]
   extension. This support is indicated by the Supported header field in
   the INVITE.

   If the caller (A) does not support Replaces, the callee (B) can
   always reject the session and attempt to establish a new session with
   A following the procedures in Section 2. This way, B would act as a
   caller and, consequently, it would follow the procedures for caller's
   invocation of transcoders.

   Assuming that the caller (A) supports Replaces, the callee (B)
   follows the steps shown in Figure 4 to invoke a transcoder. The
   callee sends a 183 (Session Progress) response (2) to the caller.
   This response carries a tag in the To header field. The caller needs
   to receive this To tag so that this early dialog can be replaced
   later in (5). So, the callee SHOULD use the reliable provisional



G. Camarillo                                                  [Page 6]


Internet Draft                    SIP                   February 7, 2004


   responses [3] SIP extension. The SDP in the 183 (Session Progress)
   response may put the media streams on hold. If the caller did not
   support this extension, the callee MAY send a 200 (OK) putting the
   media streams on hold.

   OPEN ISSUE: can we use 0.0.0.0 instead of hold here?

   After returning a response with a To tag to the caller, the callee
   sends an INVITE (2) to the Transcoder. The URI in the Request-URI of
   this INVITE contains a list parameter, as defined in [2] (draft-
   camarillo-sipping-uri-list-01), with a pointer to a URI list. This
   URI list contains a single URI: the URI received in the Contact
   header field of the initial INVITE (1) with an escaped Replaces
   header field, as shown in the following example:


   sip:caller@client.chicago.example.com?Replaces=40d432fa84b4c76e66710;
                   ;from-tag=32331
                   ;to-tag=12dr45



   We recommend the use of the reliable provisional responses between
   the callee and the transcoder so that the callee is able to
   distinguish between problems with the transcoder and problems with
   the caller, as we described in Section 2.1.


   When A receives this INVITE (5), it replaces the original dialog (1)
   with this new dialog. The caller sends a CANCEL (10) to cancel the
   original dialog (1) and receives a 487 (Request Terminated) response
   (11) from the callee.

4 Security Considerations

   TBD.

5 Contributors

   This document is the result of discussions amongst the conferencing
   design team. The members of this team include Eric Burger, Henning
   Schulzrinne and Arnoud van Wijk.

6 OPEN ISSUES

   In SIP, the Route header field is used to traverse proxies, but is
   seems that using it for traversing B2BUAs would be stretching its
   semantics too much.



G. Camarillo                                                  [Page 7]


Internet Draft                    SIP                   February 7, 2004




    A                               T                              B
    |                               |                              |
    |----------------------(1) INVITE SDP A----------------------->|
    |                               |                              |
    |<-------------(2) 183 Session Progress SDP on hold------------|
    |                               |                              |
    |--------------------------(3) PRACK-------------------------->|
    |                               |                              |
    |<-------------------------(4) 200 OK--------------------------|
    |                               |                              |
    |                               |<----(5) INVITE SDP TB--------|
    |                               |                              |
    |                               |-(6) Session Progress SDP TB->|
    |                               |                              |
    |                               |<---------(7) PRACK-----------|
    |                               |                              |
    |                               |----------(8) 200 OK--------->|
    |                               |                              |
    |<------(9) INVITE SDP TA-------|                              |
    |                               |                              |
    |-------(10) 200 OK SDP A------>|                              |
    |                               |                              |
    |<-----------(11) ACK-----------|                              |
    |                               |---------(12) 200 OK--------->|
    |                               |                              |
    |                               |<----------(13) ACK-----------|
    |                               |                              |
    |-------------------------(14) CANCEL------------------------->|
    |                               |                              |
    |<------------------------(15) 200 OK--------------------------|
    |                               |                              |
    |<------------------(16) 407 Request Terminated----------------+
    |                              |                               |
    |---------------------------(17) ACK-------------------------->|
    |                               |                              |
    | ***************************** | **************************** |
    |**          Media            **|**            Media         **|
    | ***************************** | **************************** |


   Figure 4: Callee's invocation of a transcoder


7 Authors' Addresses

   Gonzalo Camarillo
   Ericsson



G. Camarillo                                                  [Page 8]


Internet Draft                    SIP                   February 7, 2004


   Advanced Signalling Research Lab.
   FIN-02420 Jorvas
   Finland
   electronic mail:  Gonzalo.Camarillo@ericsson.com

8 Bibliography

   [1] G. Camarillo, "Framework for transcoding with the session
   initiation protocol," Internet Draft draft-camarillo-sipping-transc-
   framework-00, Internet Engineering Task Force, Aug. 2003.  Work in
   progress.

   [2] G. Camarillo, "Providing a session initiation protocol (SIP)
   application server with a list of URIs," Internet Draft draft-
   camarillo-sipping-uri-list-00, Internet Engineering Task Force, Nov.
   2003.  Work in progress.

   [3] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
   responses in session initiation protocol (SIP)," RFC 3262, Internet
   Engineering Task Force, June 2002.

   [4] B. Biggs, R. W. Dean, and R. Mahy, "The session inititation
   protocol (SIP) Engineering Task Force, Aug. 2003.  Work in progress.



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G. Camarillo                                                  [Page 9]


Internet Draft                    SIP                   February 7, 2004


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G. Camarillo                                                 [Page 10]