SIP Bridge Transcoding: The Session Initiation Protocol (SIP) Conference Bridge Transcoding Model
How a SIP call can be routed through a transcoding server that sits between the two callers like a conference bridge.
Camarillo, G., "The Session Initiation Protocol (SIP) Conference Bridge Transcoding Model", RFC 5370, DOI 10.17487/RFC5370, October 2008, <https://www.rfc-editor.org/info/rfc5370>.
@misc{rfc5370,
series = {Request for Comments},
number = 5370,
howpublished = {RFC 5370},
publisher = {RFC Editor},
doi = {10.17487/RFC5370},
url = {https://www.rfc-editor.org/info/rfc5370},
author = {G. Camarillo},
title = {{The Session Initiation Protocol (SIP) Conference Bridge Transcoding Model}},
pagetotal = 11,
year = 2008,
month = oct,
} RFC text
Contents
Network Working Group G. Camarillo
Request for Comments: 5370 Ericsson
Category: Standards Track October 2008
The Session Initiation Protocol (SIP)
Conference Bridge Transcoding Model
Status of This Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
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 ....................................................2
2. Terminology .....................................................3
3. Caller's Invocation .............................................3
3.1. Procedures at the User Agent ...............................3
3.2. Procedures at the Transcoder ...............................3
3.3. Example ....................................................4
3.4. Unsuccessful Session Establishment .........................6
4. Callee's Invocation .............................................7
5. Security Considerations .........................................7
6. Contributors ....................................................8
7. References ......................................................8
7.1. Normative References .......................................8
7.2. Informative References .....................................9
1. Introduction
RFC 5369 [RFC5369] describes how two SIP [RFC3261] UAs (User Agents)
can discover incompatibilities 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 [RFC4117] 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
Sections 3 and 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 [RFC2119], 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 RFC 5366 [RFC5366] 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 [RFC5363]. 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 [RFC5369],
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 [RFC5366].
3.2. Procedures at the Transcoder
On receiving an INVITE with a URI-list body, the transcoder follows
the procedures in [RFC5366] 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 [RFC3323] and [RFC3325]) 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 transcoder 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 [RFC5366],
the caller A adds a body part whose disposition type is recipient-
list [RFC5363].
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
[RFC4566] 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
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
[RFC4244] 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) to the INVITE request from the caller,
and it would 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
[RFC4575] at the transcoder. Although 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 of
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 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 [RFC4117] instead.
5. Security Considerations
Transcoders implementing this specification behave as a URI-list
service as described in [RFC5366]. Therefore, the security
considerations for URI-list services discussed in [RFC5363] 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 [RFC3850] or TLS [RFC5246], 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 [RFC4474].
The requirement in [RFC5363] to use opt-in lists (e.g., using RFC
5360 [RFC5360]) 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. 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.
7. References
7.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC3261] 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.
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session
Initiation Protocol (SIP)", RFC 3323, November 2002.
[RFC3325] 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.
[RFC3850] Ramsdell, B., Ed., "Secure/Multipurpose Internet Mail
Extensions (S/MIME) Version 3.1 Certificate Handling", RFC
3850, July 2004.
[RFC4117] 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.
[RFC5369] Camarillo, G., "Framework for Transcoding with the Session
Initiation Protocol", RFC 5369, October 2008.
[RFC5363] Camarillo, G. and A.B. Roach, "Framework and Security
Considerations for Session Initiation Protocol (SIP) URI-
List Services", RFC 5363, October 2008.
[RFC5366] Camarillo, G. and A. Johnston, "Conference Establishment
Using Request-Contained Lists in the Session Initiation
Protocol (SIP)", RFC 5366, October 2008.
[RFC4244] Barnes, M., Ed., "An Extension to the Session Initiation
Protocol (SIP) for Request History Information", RFC 4244,
November 2005.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
7.2. Informative References
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, August 2006.
[RFC5360] Rosenberg, J., "A Framework for Consent-Based
Communications in the Session Initiation Protocol (SIP)",
RFC 5360, October 2008.
Author's Address
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
EMail: Gonzalo.Camarillo@ericsson.com
Full Copyright Statement
Copyright (C) The IETF Trust (2008).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS
OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Intellectual Property
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.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
ietf-ipr@ietf.org.
AI ELI5
On this page
§1The Conference Bridge Model
RFC 5369 describes how two SIP user agents (UAs) can find incompatibilities that stop them from setting up a session, such as no shared codec or media type. They then need a transcoding service. The transcoding framework offers two ways to invoke one: the 3pcc (third-party call control) model of RFC 4117 and the conference bridge model, which this document specifies.
In this model, a transcoding server offering a service such as speech-to-text acts as a B2BUA (Back-to-Back User Agent) between the two UAs and is identified by a URI. Both UAs exchange signalling and media with the transcoder T, and exchange no traffic directly with each other.
Instead of two people talking straight to each other, both talk to a helper in the middle who changes what is said into a form the other can understand.
§3Caller's Invocation
Caller A must do two things: set up a session with the transcoder T, and give T the URI of callee B so T can send an INVITE to B. A does both in one INVITE. Following RFC 5366, A adds a body part with disposition type recipient-list, holding a URI-list with a single URI, B's.
On receiving this, the transcoder follows RFC 5366 to send an INVITE towards that URI. It acts as a B2BUA, not as a proxy, so the new INVITE is a separate transaction. The transcoder MUST build the outgoing From header field from the incoming one, subject to the privacy requirements in the incoming request; the tag parameter is excepted. The session description in the outgoing INVITE depends on the service. A transcoder fixing audio codec incompatibilities would list the audio codecs it supports.
When the transcoder gets a final response to its INVITE, it sends a final response to the incoming INVITE that SHOULD carry the same status code. If the URI-list has more than one URI, the transcoder SHOULD return a 488 (Max 1 URI allowed in URI-list) response. The model is a two-party conference server, so this document covers two-party sessions; multi-party sessions use INVITEs with several URIs as in RFC 5366.
You call the helper and say "please connect me to Sam". The helper then calls Sam itself and passes Sam's answer back to you.
§3.3Message Flow Example and Failure Case
In the successful flow, A sends INVITE (1) to T. T answers with 183 Session Progress (2) and sends its own INVITE (3) to B. B replies 200 OK (4), T sends ACK (5) to B, then T sends 200 OK (6) to A and A sends ACK (7). Media then flows between A and T and between T and B.
The example INVITE has two body parts: an SDP session description and an application/resource-lists+xml body whose single entry is B's URI. It also carries Require: recipient-list-invite.
In the unsuccessful flow, B returns a non-2xx final response, such as 603 Decline, and T sends a non-2xx response to A as well. This creates an ambiguity: if the provisional response (2) is lost, A cannot tell whether the transcoder or the callee rejected the call. Using the History-Info header field between the transcoder and the caller resolves it.
The RFC notes the alternative of making the transcoder a pure conference bridge, which answers 200 OK at once while the caller subscribes to the conference event package. That would also remove the ambiguity, but it is more complex, needs more messages and adds session setup delay, so it was not chosen.
If the person you wanted says no, the helper tells you no. But if one message gets lost you may not know who said no, so a note about the call's history clears it up.
- (1) INVITE SDP ACarries B's URI in a recipient-list body.
- (2) 183 Session Progress
- (3) INVITE SDP TB
- (4) 200 OK SDP B
- (5) ACK
- (6) 200 OK SDP TA
- (7) ACK
§4Callee's Invocation
A callee can also bring in the transcoder. If UA B receives an INVITE with a session description it cannot accept, it can redirect the request to the transcoder with a 302 (Moved Temporarily) response. The Contact header field holds the transcoder's URI plus a ?body= parameter containing a recipient-list body with B's URI. That body needs escaping, for example of carriage returns and line feeds.
A then sends a new INVITE to T and the flow continues as in the caller case. Because the encoding is quite complex, the RFC says it is simpler for callees to invoke transcoding with the 3pcc model of RFC 4117.
If the person you call cannot understand you, they can say "talk to my helper instead" and give you the helper's address.
§5Security Considerations
A transcoder following this document behaves as a URI-list service as in RFC 5366, so the security considerations of RFC 5363 for URI-list services apply. List integrity and unsolicited requests matter most. User agents SHOULD integrity protect URI-lists with mechanisms such as S/MIME or TLS, which can also give confidentiality. Transcoders MUST authenticate and authorize users and MAY include the original sender's identity in outgoing requests using the SIP identity mechanism.
RFC 5363 asks for opt-in lists, but this service does not need them. It causes no amplification, since only one INVITE is generated per incoming INVITE. It also does not translate to a URI the caller does not know, because the caller puts the callee's URI in the INVITE, and the caller's identity is in the transcoder's INVITE.
The helper must check who you are before helping. It also only makes one call for each call it gets, so it cannot be used to flood someone with calls.