One document matched: draft-ietf-sipping-transc-framework-02.txt
Differences from draft-ietf-sipping-transc-framework-01.txt
SIPPING Working Group G. Camarillo
Internet-Draft Ericsson
Expires: December 3, 2005 June 1, 2005
Framework for Transcoding with the Session Initiation Protocol (SIP)
draft-ietf-sipping-transc-framework-02.txt
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Copyright (C) The Internet Society (2005).
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 . . . . . . . . . . . 6
4. Security Considerations . . . . . . . . . . . . . . . . . . . 7
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
6. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 8
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8
7.1 Normative References . . . . . . . . . . . . . . . . . . . 8
7.2 Informational References . . . . . . . . . . . . . . . . . 9
Author's Address . . . . . . . . . . . . . . . . . . . . . . . 9
Intellectual Property and Copyright Statements . . . . . . . . 10
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1. Introduction
Two user agents involved in a SIP [3] 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 [11]), 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 understand
anything 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. So,
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 introduces the third party call
control and conference bridge transcoding invocation models, which
are further described in Section 3.1 and Section 3.2 respectively.
Both models meet the requirements regarding transcoding services
invocation in RFC3351 [6] 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 [2],
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
transcoding is needed for a particular session.
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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 [5] 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 we said earlier, transcoder location is outside the scope of this
document. So, 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 [7], also referred to as 3pcc. The second is to
use a (dial-in and dial-out) conference bridge that negotiates the
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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 [10], 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.
+-------+
| |
| T |**
| | **
+-------+ **
^ * **
| * **
| * **
SIP * **
| * **
| * **
v * **
+-------+ +-------+
| | | |
| A |<-----SIP----->| B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
Figure 1: Third party call control model
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.
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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.
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 (Back-to-Back User Agent) and
the whole A-T-B session is established as described in
[draft-ietf-sipping-transc-conf]. Figure 2 shows the signalling
relationships between the end-points and the transcoder.
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+-------+
| |**
| T | **
| |\ **
+-------+ \\ **
^ * \\ **
| * \\ **
| * SIP **
SIP * \\ **
| * \\ **
| * \\ **
v * \ **
+-------+ +-------+
| | | |
| A | | B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
Figure 2: Conference bridge model
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
low bandwidth or high-delay access links (e.g., some wireless
accesses).
On the other hand, 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 [9] 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.
4. Security Considerations
The specifications of the 3pcc and the conferencing transcoding
models discuss security issues directly related to the implementation
of those models. Additionally, there are some considerations that
apply to transcoding in general.
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In a session, a transcoder has access to at least some of the media
exchanged between the end points. In order to avoid rogue
transcoders getting access to those media, it is recommended that end
points authenticate the transcoder. TLS [1] and S/MIME [8] can be
used for this purpose.
To achieve a higher degree of privacy, end points following the 3pcc
transcoding model can use one transcoder in one direction and a
different one in the other direction. This way, no single transcoder
has access to all the media exchanged between the end points.
5. IANA Considerations
This document does not contain any IANA actions.
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
[1] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0",
RFC 2246, January 1999.
[2] Floyd, S. and L. Daigle, "IAB Architectural and Policy
Considerations for Open Pluggable Edge Services", RFC 3238,
January 2002.
[3] 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.
[4] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[6] Charlton, N., Gasson, M., Gybels, G., Spanner, M., 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, August 2002.
[7] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo,
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"Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)", BCP 85, RFC 3725,
April 2004.
[8] Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
(S/MIME) Version 3.1 Certificate Handling", RFC 3850,
July 2004.
[9] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.
[10] Camarillo, G., "Transcoding Services Invocation in the Session
Initiation Protocol (SIP) Using Third Party Call Control
(3pcc)", draft-ietf-sipping-transc-3pcc-02 (work in progress),
September 2004.
7.2 Informational References
[11] Handley, M., "SDP: Session Description Protocol",
draft-ietf-mmusic-sdp-new-24 (work in progress), February 2005.
Author's Address
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: Gonzalo.Camarillo@ericsson.com
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