One document matched: draft-ietf-sipping-transc-conf-03.txt
Differences from draft-ietf-sipping-transc-conf-02.txt
SIPPING Working Group G. Camarillo
Internet-Draft Ericsson
Expires: November 23, 2006 May 22, 2006
The Session Initiation Protocol (SIP) Conference Bridge Transcoding
Model
draft-ietf-sipping-transc-conf-03.txt
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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.
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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 . . . . . . . . . . . . . . . . . . . 8
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
7. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 9
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . . 10
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 12
Intellectual Property and Copyright Statements . . . . . . . . . . 13
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1. Introduction
The Framework for Transcoding with SIP [8] 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 [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 [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 [10] 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 [9]. This body part consists of a URI-list
that contains a single URI: user agent B's URI.
Note that, as described in the transcoding framework [8], 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 [10].
3.2. Procedures at the Transcoder
On receiving an INVITE with a URI-list body, the transcoder follows
the procedures in [10] 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 [4] and [5]) 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
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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 [10], the
caller A adds a body part whose disposition type is recipient-list
[9].
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
[13] 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
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
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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
[11] between the transcoder and the caller resolves the previous
ambiguity.
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 [14] 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.
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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 the syntax resulting from encoding a 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 [7] instead.
5. Security Considerations
Transcoders implementing this specification behave as a URI-list
service as described in [10]. Therefore, the security considerations
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for URI-list services discussed in [9] 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 [6] 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 [12].
The requirement in [9] to use opt-in lists (e.g., using the Framework
for Consent-Based Communications in SIP [15]) 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
[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.,
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Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[4] Peterson, J., "A Privacy Mechanism for the Session Initiation
Protocol (SIP)", RFC 3323, November 2002.
[5] 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.
[6] Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
(S/MIME) Version 3.1 Certificate Handling", RFC 3850,
July 2004.
[7] 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.
[8] Camarillo, G., "Framework for Transcoding with the Session
Initiation Protocol",
draft-camarillo-sipping-transc-framework-00 (work in progress),
August 2003.
[9] 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-05 (work in progress),
January 2006.
[10] Camarillo, G. and A. Johnston, "Conference Establishment Using
Request-Contained Lists in the Session Initiation Protocol
(SIP)", draft-ietf-sipping-uri-list-conferencing-05 (work in
progress), February 2006.
[11] Barnes, M., "An Extension to the Session Initiation Protocol
for Request History Information",
draft-ietf-sip-history-info-06 (work in progress),
January 2005.
[12] 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. Informative References
[13] Handley, M., "SDP: Session Description Protocol",
draft-ietf-mmusic-sdp-new-26 (work in progress), January 2006.
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[14] Rosenberg, J., "A Session Initiation Protocol (SIP) Event
Package for Conference State",
draft-ietf-sipping-conference-package-12 (work in progress),
July 2005.
[15] Rosenberg, J., "A Framework for Consent-Based Communications in
the Session Initiation Protocol (SIP)",
draft-ietf-sipping-consent-framework-04 (work in progress),
March 2006.
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Author's Address
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: Gonzalo.Camarillo@ericsson.com
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