| Rfc | 5369 |
|---|
| Title | Framework for Transcoding with the Session Initiation Protocol
(SIP) |
|---|
| Author | G. Camarillo |
|---|
| Date | October 2008 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Network Working Group G. Camarillo
Request for Comments: 5369 Ericsson
Category: Informational October 2008
Framework for Transcoding with the Session Initiation Protocol (SIP)
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Abstract
This document defines a framework for transcoding with SIP. This
framework includes how to discover the need for 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.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Discovery of the Need for Transcoding Services . . . . . . . . 2
3. Transcoding Services Invocation . . . . . . . . . . . . . . . . 4
3.1. Third-Party Call Control Transcoding Model . . . . . . . . 4
3.2. Conference Bridge Transcoding Model . . . . . . . . . . . . 6
4. Security Considerations . . . . . . . . . . . . . . . . . . . . 7
5. Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 8
6. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8
6.1. Normative References . . . . . . . . . . . . . . . . . . . 8
6.2. Informative References . . . . . . . . . . . . . . . . . . 9
1. Introduction
Two user agents involved in a SIP [RFC3261] 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 [RFC4566]),
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 increases 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 for 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 Sections 3.1 and 3.2, respectively. Both
models meet the requirements regarding transcoding services
invocation in RFC 3351 [RFC3351], which 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
[RFC3238], services that involve media manipulation invocation are
best invoked by one of the endpoints involved in the communication,
as opposed to being invoked by an intermediary in the network.
Following this principle, one of the endpoints should be the one
detecting that 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 [RFC3264] 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 [RFC3264].
Presence documents are typically received in a NOTIFY request
[RFC3265] 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.
In the absence of presence information, routing logic that involves
parallel forking to several user agents may make it difficult (or
impossible) for the caller to know which user agent will answer the
next call attempt. For example, a call attempt may reach the user's
voicemail while the next one may reach a SIP phone where the user is
available. If both terminating user agents have different
capabilities, the caller cannot know, even after the first call
attempt, whether or not transcoding will be necessary for the
session. This is a well-known SIP problem that is referred to as
HERFP (Heterogeneous Error Response Forking Problem). Resolving
HERFP is outside the scope of this document.
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
(Global System for Mobile Communications) phones (without using
transcoding-free operation). Both phones use a GSM codec, but the
speech is converted from GSM to PCM (Pulse Code Modulation) by the
originating MSC (Mobile Switching Center) 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 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 [RFC3725], also referred to as 3pcc. The
second is to use a (dial-in and 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 [RFC4117], 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 endpoints that are able to
perform third party call control. It allows endpoints 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 endpoints.
This model also allows invoking 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
endpoints 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 not see the signalling between both endpoints. 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 endpoints 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 [RFC5370].
Figure 2 shows the signalling relationships between the endpoints and
the transcoder.
+-------+
| |**
| T | **
| |\ **
+-------+ \\ **
^ * \\ **
| * \\ **
| * SIP **
SIP * \\ **
| * \\ **
| * \\ **
v * \ **
+-------+ +-------+
| | | |
| A | | B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
Figure 2: Conference Bridge Model
In the conferencing bridge model, the endpoint invoking the
transcoder is generally involved in less signalling exchanges than in
the 3pcc model. This may be an important feature for endpoints 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 endpoint to
support the Replaces [RFC3891] extension. At present, not many user
agents support it.
Simple endpoints 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.
In a session, a transcoder has access to at least some of the media
exchanged between the endpoints. In order to avoid rogue transcoders
getting access to those media, it is recommended that endpoints
authenticate the transcoder. TLS [RFC5246] and S/MIME [RFC3850] can
be used for this purpose.
To achieve a higher degree of privacy, endpoints 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 endpoints.
The fact that transcoders need to access media exchanged between the
endpoints implies that endpoints cannot use end-to-end media security
mechanisms. Media encryption would not allow the transcoder to
access the media, and media integrity protection would not allow the
transcoder to modify the media (which is obviously necessary to
perform the transcoding function). Nevertheless, endpoints can still
use media security between the transcoder and themselves.
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. References
6.1. Normative References
[RFC3238] Floyd, S. and L. Daigle, "IAB Architectural and Policy
Considerations for Open Pluggable Edge Services",
RFC 3238, January 2002.
[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.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3265] Roach, A.B., "Session Initiation Protocol (SIP)-Specific
Event Notification", RFC 3265, June 2002.
[RFC3351] 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.
[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and G.
Camarillo, "Best Current Practices for Third Party Call
Control (3pcc) in the Session Initiation Protocol (SIP)",
BCP 85, RFC 3725, April 2004.
[RFC3850] Ramsdell, B., "Secure/Multipurpose Internet Mail
Extensions (S/MIME) Version 3.1 Certificate Handling",
RFC 3850, July 2004.
[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891,
September 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.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5370] Camarillo, G., "The Session Initiation Protocol (SIP)
Conference Bridge Transcoding Model", RFC 5370,
October 2008.
6.2. Informative References
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
Author's Address
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
EMail: Gonzalo.Camarillo@ericsson.com
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