SIPPING Working Group                                       G. Camarillo
Internet-Draft                                                  Ericsson
Expires: June 2, 2006                                  November 29, 2005


  The Session Initiation Protocol (SIP) Conference Bridge Transcoding
                                 Model
                 draft-ietf-sipping-transc-conf-01.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.  Procedures at the User Agent . . . . . . . . . . . . . . .  4
     3.2.  Procedures at the Transcoder . . . . . . . . . . . . . . .  4
     3.3.  Example  . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.4.  Unsuccessful Session Establishment . . . . . . . . . . . .  7
   4.  Callee's Invocation  . . . . . . . . . . . . . . . . . . . . .  8
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . .  9
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  9
   7.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . .  9
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 10
     8.2.  Informational References . . . . . . . . . . . . . . . . . 11
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 12
   Intellectual Property and Copyright Statements . . . . . . . . . . 13

































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1.  Introduction

   The Framework for Transcoding with SIP [9] describes how two SIP [3]
   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 [8] 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 [1] and indicate requirement levels for
   compliant implementations.


3.  Caller's Invocation

   User agent A needs to perform two operations to invoke transcoding
   services from T for a session between user agent A and user agent B.
   User agent A needs to establish a session with T and provide T with
   user agent B's URI so that T can generate an INVITE towards user
   agent B.

3.1.  Procedures at the User Agent

   User agent A uses the procedures for Conference Establishment Using
   Request-Contained Lists in SIP [11] to provide T with B's URI using
   the same INVITE that establishes the session between A and T. That
   is, user agent A adds to the INVITE a body part whose disposition
   type is recipient-list [10].  This body part consists of a URI-list
   that MUST contain a single URI: user agent B's URI.

3.2.  Procedures at the Transcoder

   On receiving an INVITE with a URI-list body, the transcoder follows
   the procedures in [11] to generate an INVITE request towards the URI
   contained in the URI-list body.  Note that the transcoder acts as a
   B2BUA, not as a proxy.

   Additionally, the transcoder MUST generate the From header field of
   the outgoing INVITE request using the same value as the From header
   field included in the incoming INVITE request, subject to the privacy
   requirements (see [5] and [6]) expressed in the incoming INVITE
   request.  Note that this does not apply to the "tag" parameter.

   The session description the transcoder includes in the outgoing
   INVITE request depends on the type of transcoding service that
   particular transcoder provides.  For example, a transcoder resolving
   audio codec incompatibilities would generate a session description
   listing the audio codecs the transcoder supports.

   When the transcoder receives a final response for the outgoing INVITE
   requests, it generates a new final response for the incoming INVITE
   request.  This new final response SHOULD have the same status code as



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   the one received in the response for the outgoing INVITE request.

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

3.3.  Example

   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 [11], the
   caller A adds a body part whose disposition type is recipient-list
   [10].


   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

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













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   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: 556

   --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
   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



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   response (6) has the same status code as the one received in the
   response from the callee (4).

3.4.  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) 603 Decline-------|
   |                           |                           |
   |                           |---------(5) ACK---------->|
   |<----(6) 603 Decline-------|                           |
   |                           |                           |
   |---------(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 603 (Decline)
   response means that the initial INVITE (1) was rejected by the
   transcoder or that the INVITE generated by the transcoder (4) was
   rejected by the callee.  The use of the "History-Info" header field
   [12] between the transcoder and the caller resolves the previous
   ambiguity.

   Callers that do not support the "History-Info" header field can,
   alternatively, require the use of the reliable provisional responses
   [4] SIP extension.  If the caller receives a response reporting a
   reachability problem, the caller can also send an OPTIONS request to
   the transcoder to check whether or not the transcoder is reachable.
   If the transcoder is reachable, the party that could not be reached
   was the callee.

   Note that this ambiguity problem could also have been resolved by
   having transcoders act as a pure conference bridge.  The transcoder
   would respond with a 200 (OK) the INVITE request from the caller and
   generate an outgoing INVITE request towards the callee.  The caller



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   would get information about the result of the latter INVITE request
   by subscribing to the conference event package [15] at the
   transcoder.  Nevertheless, while this flow would have resolved the
   ambiguity problem without requiring support for the "History-Info"
   header field, it is more complex, requires a higher number on
   messages, and introduces higher session setup delays.  That is why it
   was not chosen to implement transcoding services.


4.  Callee's Invocation

   If a UA receives an INVITE with a session description 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 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.


   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



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   recursion on the 302 (Moved Temporarily) response.  Any proxy in the
   path between A and B may perform such a recursion.


5.  Security Considerations

   Transcoders implementing this specification behave as a URI-list
   service as described in [11].  Therefore, the security considerations
   for URI-list services discussed in [10] apply here as well.

   In particular, the requirements related to list integrity and
   unsolicited requests are important for transcoding services.  User
   agents SHOULD integrity protect URI-lists using mechanisms such as
   S/MIME [7] or TLS [2], which can also provide URI-list
   confidentiality if needed.  Additionally, transcoders MUST
   authenticate and authorize users and MAY provide information about
   the identity of the original sender of the request in their outgoing
   requests by using the SIP identity mechanism [13].

   The requirement in [10] to use opt-in lists (e.g., using the
   Framework for Consent-Based Communications in SIP [16]) deserves
   special discussion.  The type of URI-list service implemented by
   transcoders following this specification does not produce
   amplification (only one INVITE request is generated by the transcoder
   on receiving an INVITE request from a user agent) and does not
   involve a translation to a URI that may be otherwise unknown to the
   caller (the caller places the callee's URI in the body of its initial
   INVITE request).  Additionally, the identity of the caller is present
   in the INVITE request generated by the transcoder.  Therefore, there
   is no requirement for transcoders implementing this specification to
   use opt-in lists.


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





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8.1.  Normative References

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

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

   [3]   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.

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

   [5]   Peterson, J., "A Privacy Mechanism for the Session Initiation
         Protocol (SIP)", RFC 3323, November 2002.

   [6]   Jennings, C., Peterson, J., and M. Watson, "Private Extensions
         to the Session Initiation Protocol (SIP) for Asserted Identity
         within Trusted Networks", RFC 3325, November 2002.

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

   [8]   Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk,
         "Transcoding Services Invocation in the Session Initiation
         Protocol (SIP) Using Third Party Call Control (3pcc)",
         RFC 4117, June 2005.

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

   [10]  Camarillo, G. and A. Roach, "Framework and Security
         Considerations for Session Initiation Protocol (SIP)  Uniform
         Resource Identifier (URI)-List Services",
         draft-ietf-sipping-uri-services-04 (work in progress),
         October 2005.

   [11]  Camarillo, G. and A. Johnston, "Conference Establishment Using
         Request-Contained Lists in the Session  Initiation Protocol
         (SIP)", draft-ietf-sipping-uri-list-conferencing-04 (work in
         progress), October 2005.




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   [12]  Barnes, M., "An Extension to the Session Initiation Protocol
         for Request History  Information",
         draft-ietf-sip-history-info-06 (work in progress),
         January 2005.

   [13]  Peterson, J. and C. Jennings, "Enhancements for Authenticated
         Identity Management in the Session Initiation  Protocol (SIP)",
         draft-ietf-sip-identity-06 (work in progress), October 2005.

8.2.  Informational References

   [14]  Handley, M., "SDP: Session Description Protocol",
         draft-ietf-mmusic-sdp-new-25 (work in progress), July 2005.

   [15]  Rosenberg, J., "A Session Initiation Protocol (SIP) Event
         Package for Conference State",
         draft-ietf-sipping-conference-package-12 (work in progress),
         July 2005.

   [16]  Rosenberg, J., "A Framework for Consent-Based Communications in
         the Session Initiation  Protocol (SIP)",
         draft-ietf-sipping-consent-framework-03 (work in progress),
         October 2005.




























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Author's Address

   Gonzalo Camarillo
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: Gonzalo.Camarillo@ericsson.com










































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