Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
draft-camarillo-sipping-transc-framework-00.txt
August 28, 2003
Expires: February, 2004
Framework for Transcoding with the Session Initiation Protocol
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Abstract
This document defines a framework for transcoding with SIP. This
framework includes how to discover the need of transcoding services
in a session and how to invoke those transcoding services. Two models
for transcoding services invocation are discussed; the conference
bridge model and the third party call control model. Both models meet
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 Discovery of the Need for Transcoding Services ...... 3
3 Transcoding Services Invocation ..................... 4
3.1 Third Party Call Control Transcoding Model .......... 5
3.2 Conference Bridge Transcoding Model ................. 5
4 Security Considerations ............................. 8
5 Contributors ........................................ 8
6 Authors' Addresses .................................. 8
7 Bibliography ........................................ 8
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1 Introduction
Two user agents involved in a SIP [1] dialog may find it impossible
to establish a media session due to a variety of incompatibilities.
Assuming that both user agents understand the same session
description format (e.g., SDP), incompatibilities can be found at the
user agent level and at the user level. At the user agent level, both
terminals may not support any common codec or may not support common
media types (e.g., a text-only terminal and an audio-only terminal).
At the user level, a deaf person will not be able to understand what
it is said over an audio stream.
In order to make communications possible in the presence of
incompatibilities, user agents need to introduce intermediaries that
provide transcoding services to a session. From the SIP point of
view, the introduction of a transcoder is done in the same way to
resolve both user level and user agent level incompatibilities.
Therefore, the invocation mechanisms described in this document are
generally applicable to any type of incompatibility related to how
the information that needs to be communicated is encoded.
Furthermore, although this framework focuses on
transcoding, the mechanisms described are applicable to
media manipulation in general. It would be possible to use
them, for example, to invoke a server that simply increased
the volume of an audio stream.
This document does not describe media server discovery. That is an
orthogonal problem that one can address using user agent provisioning
or other methods.
The remainder of this document is organized as follows. Section 2
deals with the discovery of the need of transcoding services for a
particular session.
Section 3.2 introduces the conference bridge transcoding invocation
model, and Section 3.1 introduces the third party call control model.
Both models meet the requirements regarding transcoding services
invocation in RFC3351 [2] to support deaf, hard of hearing and
speech-impaired individuals.
2 Discovery of the Need for Transcoding Services
According to the one-party consent model defined in RFC 3238 [3],
services that involve media manipulation invocation are best invoked
by one of the end-points involved in the communication, as opposed to
being invoked by an intermediary in the network. Following this
principle, one of the end-points should be the one detecting that
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transcoding is needed for a particular session.
In order to decide whether or not transcoding is needed, a user agent
needs to know the capabilities of the remote user agent. A user agent
acting as an offerer typically obtains this knowledge by downloading
a presence document that includes media capabilities (e.g., Bob is
available on a terminal that only supports audio) or by getting an
SDP description of media capabilities as defined in RFC 3264 [4].
Presence documents are typically received in a NOTIFY request as a
result of a subscription. SDP media capabilities descriptions are
typically received in a 200 (OK) response to an OPTIONS request or in
a 488 (Not Acceptable Here) response to an INVITE.
It is recommended that an offerer does not invoke transcoding
services before making sure that the answerer does not support the
capabilities needed for the session. Making wrong assumptions about
the answerer's capabilities can lead to situations where two
transcoders are introduced (one by the offerer and one by the
answerer) in a session that would not need any transcoding services
at all.
An example of the situation above is a call between two GSM
phones (without using transcoding-free operation). Both
phones use a GSM codec, but the speech is converted from
GSM to PCM by the originating MSC and from PCM back to GSM
by the terminating MSC.
Note that transcoding services can be symmetric (e.g., speech-to-text
plus text-to-speech) or asymmetric (e.g., a one-way speech-to-text
transcoding for a hearing impaired user that can talk).
3 Transcoding Services Invocation
Once the need for transcoding for a particular session has been
identified as described in Section 2, one of the user agents needs to
invoke transcoding services.
As stated previously, transcoder location is outside the scope of
this document. Therefore, we assume that the user agent invoking
transcoding services knows the URI of a server that provides them.
Invoking transcoding services from a server (T) for a session between
two user agents (A and B) involves establishing two media sessions;
one between A and T and another between T and B. How to invoke T's
services (i.e., how to establish both A-T and T-B sessions) depends
on how we model the transcoding service. We have considered two
models for invoking a transcoding service. The first is to use third
party call control [5], also referred to as 3pcc. The second is to
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use a (dial-in and possibly dial-out) conference bridge that
negotiates the appropriate media parameters on each individual leg
(i.e., A-T and T-B).
Section 3.1 analyzes the applicability of the third party call
control model and Section 3.2 analyzes the applicability of the
conference bridge transcoding invocation model.
3.1 Third Party Call Control Transcoding Model
In the 3pcc transcoding model, defined in (draft-camarillo-sipping-
transc-3pcc), the user agent invoking the transcoding service has a
signalling relationship with the transcoder and another signalling
relationship with the remote user agent. There is no signalling
relationship between the transcoder and the remote user agent, as
shown in Figure 1.
This model is suitable for advanced end points that are able to
perform third party call control. It allows end-points to invoke
transcoding services on a stream basis. That is, the media streams
that need transcoding are routed through the transcoder while the
streams that do not need it are sent directly between the end points.
This model also allows to invoke one transcoder for the sending
direction and a different one for the receiving direction of the same
stream.
Invoking a transcoder in the middle of an ongoing session is also
quite simple. This is useful when session changes occur (e.g., an
audio session is upgraded to an audio/video session) and the end-
points cannot cope with the changes (e.g., they had common audio
codecs but no common video codecs).
The privacy level that is achieved using 3pcc is high, since the
transcoder does no see the signalling between both end-points. In
this model, the transcoder only has access to the information that is
strictly needed to perform its function.
3.2 Conference Bridge Transcoding Model
In a centralized conference, there are a number of media streams
between the conference server and each participant of a conference.
For a given media type (e.g., audio) the conference server sends over
each individual stream the media received over the rest of the
streams, typically performing some mixing. If the capabilities of all
the end-points participating in the conference are not the same, the
conference server may have to send audio to different participants
using different audio codecs.
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+-------+
| |
| T |**
| | **
+-------+ **
^ * **
| * **
| * **
SIP * **
| * **
| * **
v * **
+-------+ +-------+
| | | |
| A |<-----SIP----->| B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
Figure 1: Third Party Call Control Model
Consequently, we can model a transcoding service as a two-party
conference server that may change not only the codec in use, but also
the format of the media (e.g., audio to text).
Using this model, T behaves as a B2BUA and the whole A-T-B session is
established as described in (draft-camarillo-sipping-transc-b2bua).
Figure 2 shows the signalling relationships between the end-points
and the transcoder.
In the conferencing bridge model, the end-point invoking the
transcoder is generally involved in less signalling exchanges than in
the 3pcc model. This may be an important feature for end-poing using
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+-------+
| |**
| T | **
| |\ **
+-------+ \\ **
^ * \\ **
| * \\ **
| * SIP **
SIP * \\ **
| * \\ **
| * \\ **
v * \ **
+-------+ +-------+
| | | |
| A | | B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
Figure 2: Conference Bridge Control Model
low bandwidth or high-delay access links (e.g., some wireless
accesses).
However, this model is less flexible than the 3pcc model. It is not
possible to use different transcoders for different streams or for
different directions of a stream.
Invoking a transcoder in the middle of an ongoing session or changing
from one transcoder to another requires the remote end-point to
support the Replaces [6] extension. At present, not many user agents
support it.
Simple end-points that cannot perform 3pcc and thus cannot use the
3pcc model, of course, need to use the conference bridge model.
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4 Security Considerations
This document does not introduce any new security considerations.
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 Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
7 Bibliography
[1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. R. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
initiation protocol," RFC 3261, Internet Engineering Task Force, June
2002.
[2] N. Charlton, M. Gasson, G. Gybels, M. Spanner, and A. van Wijk,
"User requirements for the session initiation protocol (SIP) in
support of deaf, hard of hearing and speech-impaired individuals,"
RFC 3351, Internet Engineering Task Force, Aug. 2002.
[3] S. Floyd and L. Daigle, "IAB architectural and policy
considerations for open pluggable edge services," RFC 3238, Internet
Engineering Task Force, Jan. 2002.
[4] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
session description protocol (SDP)," RFC 3264, Internet Engineering
Task Force, June 2002.
[5] J. Rosenberg, J. L. Peterson, H. Schulzrinne, and G. Camarillo,
"Best current practices for third party call control in the session
initiation protocol," internet draft, Internet Engineering Task
Force, July 2003. Work in progress.
[6] 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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