SIPPING Working Group                                       G. Camarillo
Internet-Draft                                                  Ericsson
Expires: December 3, 2005                                   June 1, 2005


  The Session Initiation Protocol (SIP) Conference Bridge Transcoding
                                 Model
                 draft-ietf-sipping-transc-conf-00.txt

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Copyright Notice

   Copyright (C) The Internet Society (2005).

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.







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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Caller's Invocation  . . . . . . . . . . . . . . . . . . . . .  4
     3.1   Unsuccessful Session Establishment . . . . . . . . . . . .  6
   4.  Callee's Invocation  . . . . . . . . . . . . . . . . . . . . .  6
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . .  7
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  8
   7.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . .  8
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . .  8
     8.1   Normative References . . . . . . . . . . . . . . . . . . .  8
     8.2   Informational References . . . . . . . . . . . . . . . . .  9
       Author's Address . . . . . . . . . . . . . . . . . . . . . . .  9
       Intellectual Property and Copyright Statements . . . . . . . . 10




































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1.  Introduction

   The Framework for Transcoding with SIP [6] describes how two SIP [4]
   UAs (User Agents) 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.  The transcoding framework introduces two models to
   invoke transcoding services: the 3pcc (third-party call control)
   model [7] and the conference bridge model.  This document specifies
   the conference bridge model.

   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 (Back-to-Back User Agent)
   between both UAs and is identified by a URI.  As shown in Figure 1,
   both UAs, A and B, exchange signalling and media with the transcoder
   T. The UAs do not exchange any traffic (signalling or media) directly
   between them.


          +-------+
          |       |**
          |   T   |  **
          |       |\   **
          +-------+ \\   **
            ^   *     \\   **
            |   *       \\   **
            |   *         SIP  **
           SIP  *           \\   **
            |   *             \\   **
            |   *               \\   **
            v   *                 \    **
          +-------+               +-------+
          |       |               |       |
          |   A   |               |   B   |
          |       |               |       |
          +-------+               +-------+


           <-SIP-> Signalling
           ******* Media

                     Figure 1: Conference bridge model

   Section 3 and Section 4 specify how the caller A or the callee B,
   respectively, can use the conference bridge model to invoke
   transcoding services from T.



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2.  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 [3] and indicate requirement levels for
   compliant implementations.

3.  Caller's Invocation

   A needs to perform two operations to invoke transcoding services from
   T for a session between A and B. A needs to establish a session with
   T and provide T with B's URI so that T can generate an INVITE towards
   B. A uses the procedures for Conference Establishment Using Request-
   Contained Lists in SIP [9] to provide T with B's URI using the same
   INVITE that establishes the session between A and T.

   Figure 2 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.  Following the procedures in [9], A
   adds a body part whose disposition type is recipient-list [8].  This
   body part consists of a URI-list that MUST contain a single URI: B's
   URI.

   If a trancoder receives a URI-list with more than one URI, it SHOULD
   return a 488 (Max 1 URI allowed in URI-list) response.


   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 2: Successful invocation of a transcoder by the caller



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   The following example shows an INVITE with two body parts: an SDP
   [10] session description and a URI-list.


   INVITE sip:transcoder@example.com SIP/2.0
   Via: SIP/2.0/TCP client.chicago.example.com
       ;branch=z9hG4bKhjhs8ass83
   Max-Forwards: 70
   To: Transcoder <sip:transcoder@example.org>
   From: A <sip:A@chicago.example.com>;tag=32331
   Call-ID: d432fa84b4c76e66710
   CSeq: 1 INVITE
   Contact: <sip:A@client.chicago.example.com>
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
        SUBSCRIBE, NOTIFY
   Allow-Events: dialog
   Accept: application/sdp, message/sipfrag
   Require: recipient-list-invite
   Conten-Type: multipart/mixed;boundary="boundary1"
   Content-Length: xxx

   --boundary1
   Content-Type: application/sdp

   v=0
   o=example 2890844526 2890842807 IN IP4 chicago.example.com
   s=-
   c=IN IP4 192.0.2.1
   t=0 0
   m=audio 20000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000

   --boundary1
   Content-Type: application/resource-lists+xml
   Content-Disposition: recipient-list

   <?xml version="1.0" encoding="UTF-8"?>
   <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"
                  xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
     <list>
       <entry uri="sip:B@example.org" />
     </list>
   </resource-lists>
   --boundary1--

   On receiving the INVITE, the transcoder generates a new INVITE
   towards the callee.  The transcoder acts as a B2BUA, not as a proxy.
   Therefore, this new INVITE (3) belongs to a different transaction



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   than the INVITE (1) received by the transcoder.

   When the transcoder receives a final response (4) from the callee, it
   generates a new final response (6) for INVITE (1).  This new final
   response (6) SHOULD have the same status code as the one received in
   the response from the callee (4).

3.1  Unsuccessful Session Establishment

   Figure 3 shows a similar message flow as the one in Figure 3.
   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 3: Unsuccessful session establishment

   The ambiguity in this flow is that, if the provisional response (2)
   gets lost, 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 ambiguity, the callee can either
   require the use of the reliable provisional responses [5] SIP
   extension or send an OPTIONS request to the transcoder to check
   whether it is reachable.

4.  Callee's Invocation

   If a UA receives an INVITE with an offer that is not acceptable, it
   can redirect it to the transcoder by using a 302 (Moved Temporarily)
   response.  The Contact header field of the 302 (Moved Temporarily)
   response contains the URI of the transcoder plus a "?body="
   parameter.  This parameter contains a recipient-list body with B's



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   URI.  Note that some escaping (e.g., for Carriage Returns and Line
   Feeds) is needed to encode a recipient-list body in such a parameter.
   Figure 4 shows the message flow for this scenario.

   <t>
      Please view in a fixed-width font such as Courier.

   A                           T                           B
   |                           |                           |
   |-------------------(1) INVITE SDP A------------------->|
   |                           |                           |
   |<--------------(2) 302 Moved Temporarily---------------|
   |                           |                           |
   |-----------------------(3) ACK------------------------>|
   |                           |                           |
   |-----(4) INVITE SDP A----->|                           |
   |                           |                           |
   |<-(5) 183 Session Progress-|                           |
   |                           |-----(6) INVITE SDP TB---->|
   |                           |                           |
   |                           |<-----(7) 200 OK SDP B-----|
   |                           |                           |
   |                           |---------(8) ACK---------->|
   |<----(9) 200 OK SDP TA-----|                           |
   |                           |                           |
   |--------(10) ACK---------->|                           |
   |                           |                           |
   | ************************* | ************************* |
   |**        Media          **|**        Media          **|
   | ************************* | ************************* |

              Figure 4: \{Callee's invocation of a transcoder

   Note that A does not necessarily need to be the one performing the
   recursion on the 302 (Moved Temporarily) response.  Any proxy in the
   path between A and B may perform such a recursion.

5.  Security Considerations

   TBD.

   Need to mention how consent applies to this work when consent is more
   mature.

   Need to mention TLS [1] and S/MIME [2].






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6.  IANA Considerations

   This document does not contain any IANA actions.

7.  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.

8.  References

8.1  Normative References

   [1]  Dierks, T. and C. Allen, "The TLS Protocol Version 1.0",
        RFC 2246, January 1999.

   [2]  Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
        (S/MIME) Version 3.1 Certificate Handling", RFC 3850, July 2004.

   [3]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", BCP 14, RFC 2119, March 1997.

   [4]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
        Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
        Session Initiation Protocol", RFC 3261, June 2002.

   [5]  Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
        Responses in Session Initiation Protocol (SIP)", RFC 3262,
        June 2002.

   [6]  Camarillo, G., "Framework for Transcoding with the Session
        Initiation Protocol",
        draft-camarillo-sipping-transc-framework-00 (work in progress),
        August 2003.

   [7]  Camarillo, G., "Transcoding Services Invocation in the Session
        Initiation Protocol (SIP)  Using Third Party Call Control
        (3pcc)", draft-ietf-sipping-transc-3pcc-02 (work in progress),
        September 2004.

   [8]  Camarillo, G. and A. Roach, "Requirements and Framework for
        Session Initiation Protocol (SIP)Uniform  Resource Identifier
        (URI)-List Services", draft-ietf-sipping-uri-services-02 (work
        in progress), December 2004.

   [9]  Camarillo, G. and A. Johnston, "Conference Establishment Using
        Request-Contained Lists in the Session  Initiation Protocol



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        (SIP)", draft-ietf-sipping-uri-list-conferencing-02 (work in
        progress), December 2004.

8.2  Informational References

   [10]  Handley, M., "SDP: Session Description Protocol",
         draft-ietf-mmusic-sdp-new-24 (work in progress), February 2005.


Author's Address

   Gonzalo Camarillo
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: Gonzalo.Camarillo@ericsson.com

































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