Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
draft-camarillo-sipping-transc-b2bua-01.txt
February 7, 2004
Expires: August, 2004
The Session Initiation Protocol Conference Bridge Transcoding Model
STATUS OF THIS MEMO
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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 Caller's Invocation ................................. 3
2.1 Unsuccessful Session Establishment .................. 5
3 Callee's Invocation ................................. 6
4 Security Considerations ............................. 7
5 Contributors ........................................ 7
6 OPEN ISSUES ......................................... 7
7 Authors' Addresses .................................. 8
8 Bibliography ........................................ 9
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1 Introduction
The framework for transcoding with SIP [1] (draft-ietf-sipping-
transc-framework) describes how two SIP UAs 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. Using the
conference bridge model is one way to perform such invocation.
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 between both UAs and is
identified by a URI.
2 Caller's Invocation
Figure 1 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. The URI in the Request-URI of this
INVITE contains a list parameter, as defined in [2] (draft-
camarillo-sipping-uri-list-01), with a pointer to a URI list. This
URI list contains a single URI: the callee's URI, as shown below:
INVITE sip:transcoder@example.com;list=cid:cn35t8@example.com SIP/2.0
Via: SIP/2.0/TCP client.chicago.example.com
;branch=z9hG4bKhjhs8ass83
Max-Forwards: 70
To: "Transcoder" <sip:transcoder@example.com>
From: Caller <sip:caller@chicago.example.com>;tag=32331
Call-ID: d432fa84b4c76e66710
CSeq: 1 INVITE
Contact: <sip:caller@client.chicago.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Conten-Type: multipart/mixed;boundary="boundary1"
Content-Length: xxx
--boundary1
Content-Type: application/sdp
Content-Length: xxx
v=0
o=caller 2890844526 2890842807 IN IP4 chicago.example.com
s=Example Subject
c=IN IP4 192.0.0.1
t=0 0
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m=audio 20000 RTP/AVP 0
--boundary1
Content-Type: application/resource-lists+xml
Content-Length: 367
Content-ID: <cn35t8@example.com>
<?xml version="1.0" encoding="UTF-8"?>
<resource-lists xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<list name="ad-hoc-1">
<entry name="1" uri="sip:callee@example2.com" />
</list>
</resource-lists>
--boundary1--
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 1: Successful invocation of a transcoder by the caller
On reception of the INVITE, the transcoder generates a new INVITE
towards the callee. The transcoder acts as a B2BUA, so, 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
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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).
The advantage of this message flow is that, for both user agents, is
indentical to the flow for establishing a regular session (i.e.,
without transcoder) between them. Additionaly, the only difference in
the message contents is that the caller needs to use a list parameter
in the Request-URI of the initial INVITE.
2.1 Unsuccessful Session Establishment
Figure 2 shows a similar message flow as the one in Figure 1.
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 2: Unsuccessful session establishment
The problem with this flow is that 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, it is
recommended that the caller uses the reliable provisional responses
[3] SIP extension.
Figure 3 shows the resulting message flow when the caller requires
the use of the reliable provisional responses [3] SIP extension. The
repection of the 183 (Session Progress) reliable provisional response
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informs the caller that the transcoder was contacted susccessfully.
So, the 404 (Not Found) response indicates that the callee could not
be reached.
A T B
| | |
|--------(1) INVITE SDP A------>| |
| | |
|<-(2) 183 S. Prog. SDP on hold-| |
| |-----(3) INVITE SDP TB---->|
| | |
|-----------(4) PRACK---------->| |
| | |
|<----------(5) 200 OK----------| |
| | |
| |<----(6) 404 Not Found-----|
| | |
| |---------(7) ACK---------->|
|<-------(8) 404 Not Found------| |
| | |
|-------------(9) ACK---------->| |
| | |
Figure 3: Invocation using reliable provisional responses
3 Callee's Invocation
If a UA receives an INVITE with an offer that is not acceptable, it
can only invoke a transcoder if the caller supports the Replaces [4]
extension. This support is indicated by the Supported header field in
the INVITE.
If the caller (A) does not support Replaces, the callee (B) can
always reject the session and attempt to establish a new session with
A following the procedures in Section 2. This way, B would act as a
caller and, consequently, it would follow the procedures for caller's
invocation of transcoders.
Assuming that the caller (A) supports Replaces, the callee (B)
follows the steps shown in Figure 4 to invoke a transcoder. The
callee sends a 183 (Session Progress) response (2) to the caller.
This response carries a tag in the To header field. The caller needs
to receive this To tag so that this early dialog can be replaced
later in (5). So, the callee SHOULD use the reliable provisional
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responses [3] SIP extension. The SDP in the 183 (Session Progress)
response may put the media streams on hold. If the caller did not
support this extension, the callee MAY send a 200 (OK) putting the
media streams on hold.
OPEN ISSUE: can we use 0.0.0.0 instead of hold here?
After returning a response with a To tag to the caller, the callee
sends an INVITE (2) to the Transcoder. The URI in the Request-URI of
this INVITE contains a list parameter, as defined in [2] (draft-
camarillo-sipping-uri-list-01), with a pointer to a URI list. This
URI list contains a single URI: the URI received in the Contact
header field of the initial INVITE (1) with an escaped Replaces
header field, as shown in the following example:
sip:caller@client.chicago.example.com?Replaces=40d432fa84b4c76e66710;
;from-tag=32331
;to-tag=12dr45
We recommend the use of the reliable provisional responses between
the callee and the transcoder so that the callee is able to
distinguish between problems with the transcoder and problems with
the caller, as we described in Section 2.1.
When A receives this INVITE (5), it replaces the original dialog (1)
with this new dialog. The caller sends a CANCEL (10) to cancel the
original dialog (1) and receives a 487 (Request Terminated) response
(11) from the callee.
4 Security Considerations
TBD.
5 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.
6 OPEN ISSUES
In SIP, the Route header field is used to traverse proxies, but is
seems that using it for traversing B2BUAs would be stretching its
semantics too much.
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A T B
| | |
|----------------------(1) INVITE SDP A----------------------->|
| | |
|<-------------(2) 183 Session Progress SDP on hold------------|
| | |
|--------------------------(3) PRACK-------------------------->|
| | |
|<-------------------------(4) 200 OK--------------------------|
| | |
| |<----(5) INVITE SDP TB--------|
| | |
| |-(6) Session Progress SDP TB->|
| | |
| |<---------(7) PRACK-----------|
| | |
| |----------(8) 200 OK--------->|
| | |
|<------(9) INVITE SDP TA-------| |
| | |
|-------(10) 200 OK SDP A------>| |
| | |
|<-----------(11) ACK-----------| |
| |---------(12) 200 OK--------->|
| | |
| |<----------(13) ACK-----------|
| | |
|-------------------------(14) CANCEL------------------------->|
| | |
|<------------------------(15) 200 OK--------------------------|
| | |
|<------------------(16) 407 Request Terminated----------------+
| | |
|---------------------------(17) ACK-------------------------->|
| | |
| ***************************** | **************************** |
|** Media **|** Media **|
| ***************************** | **************************** |
Figure 4: Callee's invocation of a transcoder
7 Authors' Addresses
Gonzalo Camarillo
Ericsson
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Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
8 Bibliography
[1] G. Camarillo, "Framework for transcoding with the session
initiation protocol," Internet Draft draft-camarillo-sipping-transc-
framework-00, Internet Engineering Task Force, Aug. 2003. Work in
progress.
[2] G. Camarillo, "Providing a session initiation protocol (SIP)
application server with a list of URIs," Internet Draft draft-
camarillo-sipping-uri-list-00, Internet Engineering Task Force, Nov.
2003. Work in progress.
[3] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
responses in session initiation protocol (SIP)," RFC 3262, Internet
Engineering Task Force, June 2002.
[4] B. Biggs, R. W. Dean, and R. Mahy, "The session inititation
protocol (SIP) Engineering Task Force, Aug. 2003. Work in progress.
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