SIPPING Working Group G. Camarillo Internet-Draft Ericsson Expires:
July 20,November 23, 2006 January 16,May 22, 2006 The Session Initiation Protocol (SIP) Conference Bridge Transcoding Model draft-ietf-sipping-transc-conf-02.txtdraft-ietf-sipping-transc-conf-03.txt Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. 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. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on July 20,November 23, 2006. Copyright Notice Copyright (C) The Internet Society (2006). 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. 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 . . . . . . . . . . . . . . . . . . . 98 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 109 8.1. Normative References . . . . . . . . . . . . . . . . . . . 109 8.2. InformationalInformative References . . . . . . . . . . . . . . . . . 11. 10 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 12 Intellectual Property and Copyright Statements . . . . . . . . . . 13 1. Introduction The Framework for Transcoding with SIP  describes how two SIP  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  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. 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  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  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 .. This body part consists of a URI-list that MUST containcontains a single URI: user agent B's URI. Note that, as described in the transcoding framework , the transcoding model described in this document is modeled as a two- party conference server. Consequently, this document focuses on two-party sessions that need transcoding. Multi-party sessions can be established using INVITE requests with multiple URIs in their bodies, as specified in . 3.2. Procedures at the Transcoder On receiving an INVITE with a URI-list body, the transcoder follows the procedures in  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  and )) 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 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 ,, the caller A adds a body part whose disposition type is recipient-list .. 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  session description and a URI-list. INVITE sip:email@example.com SIP/2.0 Via: SIP/2.0/TCP client.chicago.example.com ;branch=z9hG4bKhjhs8ass83 Max-Forwards: 70 To: Transcoder <sip:firstname.lastname@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 Content-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 50000 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 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  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  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 would get information about the result of the latter INVITE request by subscribing to the conference event package  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 recursion onthe 302 (Moved Temporarily) response. Any proxy in the path between A and B may perform suchsyntax resulting from encoding a recursion.body into a URI as described earlier is quite complex. It is actually simpler for callees to invoke transcoding services using the 3pcc transcoding model  instead. 5. Security Considerations Transcoders implementing this specification behave as a URI-list service as described in .. Therefore, the security considerations for URI-list services discussed in  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  or TLS , 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 .. The requirement in  to use opt-in lists (e.g., using the Framework for Consent-Based Communications in SIP )) 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 8.1. Normative References  Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.  Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC 2246, January 1999.  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.  Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional Responses in Session Initiation Protocol (SIP)", RFC 3262, June 2002. Peterson, J., "A Privacy Mechanism for the Session Initiation Protocol (SIP)", RFC 3323, November 2002.  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.  Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions (S/MIME) Version 3.1 Certificate Handling", RFC 3850, July 2004.  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.  Camarillo, G., "Framework for Transcoding with the Session Initiation Protocol", draft-camarillo-sipping-transc-framework-00 (work in progress), August 2003.  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-04draft-ietf-sipping-uri-services-05 (work in progress), October 2005. January 2006.  Camarillo, G. and A. Johnston, "Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)", draft-ietf-sipping-uri-list-conferencing-04draft-ietf-sipping-uri-list-conferencing-05 (work in progress), October 2005. February 2006.  Barnes, M., "An Extension to the Session Initiation Protocol for Request History Information", draft-ietf-sip-history-info-06 (work in progress), January 2005.  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. InformationalInformative References  Handley, M., "SDP: Session Description Protocol", draft-ietf-mmusic-sdp-new-25draft-ietf-mmusic-sdp-new-26 (work in progress), July 2005. January 2006.  Rosenberg, J., "A Session Initiation Protocol (SIP) Event Package for Conference State", draft-ietf-sipping-conference-package-12 (work in progress), July 2005.  Rosenberg, J., "A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)", draft-ietf-sipping-consent-framework-03draft-ietf-sipping-consent-framework-04 (work in progress), October 2005.March 2006. Author's Address Gonzalo Camarillo Ericsson Hirsalantie 11 Jorvas 02420 Finland Email: Gonzalo.Camarillo@ericsson.com Intellectual Property Statement The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79. Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr. 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