Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
E. Burger
SnowShore Networks
H. Schulzrinne
Columbia University
A. van Wijk
Ericsson
draft-camarillo-sip-deaf-00.txt
September 9, 2002
Expires: March, 2003
Transcoding Services Invocation in the Session Initiation Protocol
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
This document describes how to use SIP third party call control to
invoke transcoding services that involve media manipulations by a
media server. In particular, this document describes how to meet the
requirements for the Session Initiation Protocol in support of deaf,
hard of hearing and speech-impaired individuals using third party
call control.
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Table of Contents
1 Introduction ........................................ 3
2 Conference Bridge Transcoding Model ................. 4
3 3PCC Transcoding Model .............................. 5
3.1 Called Party Invocation ............................. 5
3.2 Caller invocation ................................... 11
4 Receiving the original stream ....................... 11
5 Transcoding services in parallel .................... 13
6 Transcoding services in serial ...................... 18
7 Authors' Addresses .................................. 18
8 Bibliography ........................................ 19
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1 Introduction
A key requirement for SIP [1] to support deaf, hard of hearing and
speech-impaired individuals [2] is to be able to introduce
intermediaries that provide transcoding services to a session. Common
examples of transcoding services are speech-to-text and speech-to-SL
(Sign Language), a video format. One can envision other transcoding
scenarios, such as a hearing-impaired user that wishes to still see
and hear the original media (voice and video), yet also see a text
translation and sign language feed.
Transcoding can be symmetric, as in a speech-to-text plus text-to-
speech. This would be the case for a TTY user communicating with a
speaking and hearing user. Transcoding can also be asymmetric, as in
a one-way speech-to-text transcoding. This would be the case for a
hearing impaired user that can still talk.
Note that the mechanisms described in this document are not specific
to the support of deaf, hard of hearing and speech-impaired
individuals. If they are used to invoke other types of transcoding
services, such as video-to-audio or image-to-audio, they can be used
to support blind individuals as well. In fact, this document
describes a general mechanism for invoking any service that involves
media manipulations by a media server.
One example of this is wireless communications. Many network
topologies require transcoding between different media formats. For
example, 3GPP handsets may use GSM AMR2, an advanced audio codec,
while older handsets may use GSM FR, SVC, or other codecs. Existing
wireless topologies transcode all media to a common format, thus
introducing a number of transcoding steps that is often unnecessary.
Thus there is a need to transcode between these codecs on an ad hoc,
or on demand, basis.
This document does not describe media server discovery. That is an
orthogonal problem that one can address using user agent
provisioning, registrars, or other methods.
There are two models for invoking a transcoding service. The first is
to use an RTP mixer (conference bridge) that negotiates the
appropriate media parameters on each individual leg. The second is to
use third party call control [3], also referred to as 3pcc, to invoke
the transcoding service.
As the conference bridge model is straightforward and standard as is,
this document will only describe how it works. The following sections
of this document describe how to use 3pcc to invoke transcoding
services.
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All the examples provided in this document use the Session
Description Protocol (SDP) [4]. However, other session description
formats can be used with the same call flows, as long as they provide
enough functionality.
2 Conference Bridge Transcoding Model
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
possible models, namely, the conferencing server model and a model
specific to transcoding services.
A conference server typically establishes an audio stream with each
participant of a conference. The server sends over each individual
stream the media received over the rest of the streams, typically
performing some mixing. The conference server may have to send audio
to different participants using different audio codecs. Each of these
audio streams is typically established though an INVITE request from
each participant to the conference server that carries a session
description.
We can think of a transcoding service as a two-party conference
server that not only changes the codec in use, but the format of the
media as well (e.g., audio to text). Using this model, the whole A-
T-B session would be established in the same way as a conference. T
would receive two INVITEs; one INVITE with A's session description
and another with B's session description.
This approach has the advantage of being general, since the
transcoding service is modeled as a particular case of a conference.
However, in this model T is involved in two INVITE transactions to
set up the session. If we model T as a transcoding service rather
than a special case of a conferencing server, a single INVITE
transaction would provide T with both A's and B's session
descriptions. In order to provide in a single session description
information about media streams that belong to different entities (A
and B), the session description format in use should provide a means
to define how these streams should be mapped. For instance, in a
session description with two audio streams and one text stream, a
possible mapping would be the following; the information received
over the first audio stream should be sent over the text stream and
over the second audio stream, and the incoming text should be sent
only over the first audio stream. SDP [4] can convey this information
using the source and sink attributes .
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Using the conference bridge model provides a clean way to invoke
transcoding services for conferences where many users and transcoding
instances are involved. Although this is an interesting situation,
our expectation is that the vast majority of transcoding for a given
session will occur at a single media server between two users. In
this case, 3pcc is a much more efficient method than the conference
bridge method. The conference bridge method has the overhead of
multiple, seemingly uncorrelated call sessions that happen to
terminate at the conference bridge. The following section details the
protocol mechanics of using 3pcc for providing transcoding services.
3 3PCC Transcoding Model
As stated above, the invocation of a transcoding service consists of
establishing two sessions; A-T and T-B. How these sessions are
established depends on which party, the caller (A) or the called
party (B), invokes the transcoding services.
All the figures in this document follow the naming convention below:
SDP A: A session description generated by A. It contains, among
other things, the transport address/es (IP address and port
number) where A wants to receive media for each particular
stream.
SDP B: A session description generated by B. It contains, among
other things, the transport address/es where B wants to
receive media for each particular stream.
SDP A+B: A session description that contains, among other
things, the transport address/es where A wants to receive
media and the transport address/es where B wants to receive
media.
SDP TA: A session description generated by T and intended for A.
It contains, among other things, the transport address/es
where T wants to receive media from A.
SDP TB: A session description generated by T and intended for B.
It contains, among other things, the transport address/es
where T wants to receive media from B.
SDP TA+TB: A session description generated by T that contains,
among other things, the transport address/es where T wants
to receive media from A and the transport address/es where
T wants to receive media from B.
3.1 Called Party Invocation
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In this scenario B receives an INVITE from A, and B decides to
introduce T in the session. Figure 1 shows the call flow for this
scenario.
A T B
| | |
|--------------------(1) INVITE SDP A-------------------->|
| | |
| |<---(2) INVITE SDP A+B------|
| | |
| |---(3) 200 OK SDP TA+TB---->|
| | |
| |<---------(4) ACK-----------|
| | |
|<-------------------(5) 200 OK SDP TA--------------------|
| | |
|------------------------(6) ACK------------------------->|
| | |
| ************************** | ************************** |
|* MEDIA *|* MEDIA *|
| ************************** | ************************** |
| | |
Figure 1: Callee invocation of a transcoding service
In figure 1 A can both hear and speak and B is a deaf user with a
speech impairment. A proposes to establish a session that consists of
an audio stream (1). B wants to send and receive only text, so it
invokes a transcoding service T that will perform both speech-to-text
and text-to-speech conversions (2). The session descriptions of
figure 1 are partially shown below.
(1) INVITE SDP A
m=audio 20000 RTP/AVP 0
c=IN IP4 A.domain.com
(2) INVITE SDP A+B
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m=audio 20000 RTP/AVP 0
c=IN IP4 A.domain.com
a=source:1
a=sink:2
m=text 40000 RTP/AVP 96
c=IN IP4 B.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
(3) 200 OK SDP TA+TB
m=audio 30000 RTP/AVP 0
c=IN IP4 T.domain.com
a=source:1
a=sink:2
m=text 30002 RTP/AVP 96
c=IN IP4 T.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
(5) 200 OK SDP TA
m=audio 30000 RTP/AVP 0
c=IN IP4 T.domain.com
Four media streams (i.e., two bi-directional streams) have been
established at this point:
1. Audio from A to T.domain.com:30000
2. Text from T to B.domain.com:40000
3. Text from B to T.domain.com:30002
4. Audio from T to A.domain.com:20000
When either A or B decide to terminate the session, B will send a BYE
to T indicating that the session is over.
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If the first INVITE (1) received by B is empty (no session
description), the call flow is slightly different. Figure 2 shows the
messages involved.
A T B
| | |
|----------------------(1) INVITE------------------------>|
| | |
| |<-----(2) INVITE SDP B------|
| | |
| |---(3) 200 OK SDP TA+TB---->|
| | |
| |<---------(4) ACK-----------|
| | |
|<-------------------(5) 200 OK SDP TA--------------------|
| | |
|-----------------------(6) ACK SDP A-------------------->|
| | |
| |<-------(7) INVITE----------|
| | |
| |---(8) 200 OK SDP TA+TB---->|
| | |
|<-----------------(9) INVITE SDP TA----------------------|
| | |
|------------------(10) 200 OK SDP A--------------------->|
| | |
|<-----------------------(11) ACK-------------------------|
| | |
| |<-----(12) ACK SDP A+B------|
| | |
| ************************** | ************************** |
|* MEDIA *|* MEDIA *|
| ************************** | ************************** |
Figure 2: Callee invocation after initial INVITE without SDP
B may have different reasons for invoking T before knowing A's
session description. B may want to hide its capabilities, and
therefore it wants to return a session description with all the
codecs B supports plus all the codecs T supports. Or T may provide
recording services (besides transcoding), and B wants T to record the
conversation, regardless of whether or not transcoding is needed.
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This scenario is a bit more complex than the previous one. In INVITE
(2), B still does not have SDP A, so it cannot provide T with that
information. When B finally receives SDP A in (6), it has to send it
to T. B sends an empty INVITE to T (7) and gets a 200 OK with SDP
TA+TB (8). In general, this SDP TA+TB can be different than the one
that was sent in (3). That is why B needs to send the updated SDP TA
to A in (9). A then sends a possibly updated SDP A (10) and B sends
it to T in (12). However, if T happens to return the same SDP TA+TB
in (8) as in (3), B can skip messages (9), (10) and (11). Therefore,
implementors of transcoding services are encouraged to return the
same session description in (8) as in (3) in this type of scenario.
The session descriptions of this flow are shown below:
(2) INVITE SDP A+B
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
a=source:1
a=sink:2
m=text 40000 RTP/AVP 96
c=IN IP4 B.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
(3) 200 OK SDP TA+TB
m=audio 30000 RTP/AVP 0
c=IN IP4 T.domain.com
a=source:1
a=sink:2
m=text 30002 RTP/AVP 96
c=IN IP4 T.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
(5) 200 OK SDP TA
m=audio 30000 RTP/AVP 0
c=IN IP4 T.domain.com
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(6) ACK SDP A
m=audio 20000 RTP/AVP 0
c=IN IP4 A.domain.com
(8) 200 OK SDP TA+TB
m=audio 30004 RTP/AVP 0
c=IN IP4 T.domain.com
a=source:1
a=sink:2
m=text 30006 RTP/AVP 96
c=IN IP4 T.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
(9) INVITE SDP TA
m=audio 30004 RTP/AVP 0
c=IN IP4 T.domain.com
(10) 200 OK SDP A
m=audio 20002 RTP/AVP 0
c=IN IP4 A.domain.com
(12) ACK SDP A+B
m=audio 20002 RTP/AVP 0
c=IN IP4 A.domain.com
a=source:1
a=sink:2
m=text 40000 RTP/AVP 96
c=IN IP4 B.domain.com
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a=rtpmap:96 t140/1000
a=source:2
a=sink:1
Four media streams (i.e., two bi-directional streams) have been
established at this point:
1. Audio from A to T.domain.com:30004
2. Text from T to B.domain.com:40000
3. Text from B to T.domain.com:30006
4. Audio from T to A.domain.com:20002
3.2 Caller invocation
In this scenario A wishes to establish a session with B using a
transcoding service. A uses 3pcc to set up the session between T and
B. A may have different reasons for invoking T's services before
knowing B's session description. A may have contacted B right before
(i.e., INVITE-488 Not Acceptable Here-ACK)and noticed that they do
not have any codecs in common. T may also provide recording services,
which B wants to invoke.
The call flow we provide here is slightly different than the ones in
[3]. In [3], the controller establishes a session between two user
agents, being the user agents the ones deciding the characteristics
of the streams. Here, A wants to establish a session between T and B,
but A wants to decide how many and which types of streams are
established. That is why A sends its session description in the first
INVITE (1) to T, as opposed to the media-less initial INVITE
recommended by [3]. Figure 3 shows the call flow for this scenario.
We do not include the session descriptions of this flow, since they
are very similar to the ones in Figure 2. In this flow, if T returns
the same SDP TA+TB in (8) as in (2), messages (9), (10) and (11) can
be skipped.
4 Receiving the original stream
Sometimes, as pointed out in the requirements for SIP in support of
deaf, hard of hearing and speech-impaired individuals [2], a user
wants to receive both the original stream (e.g., audio) and the
transcoded stream (e.g., the output of the speech-to-text
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A T B
| | |
|-------(1) INVITE SDP A---->| |
| | |
|<----(2) 200 OK SDP TA+TB---| |
| | |
|----------(3) ACK---------->| |
| | |
|--------------------(4) INVITE SDP TA------------------->|
| | |
|<--------------------(5) 200 OK SDP B--------------------|
| | |
|-------------------------(6) ACK------------------------>|
| | |
|--------(7) INVITE--------->| |
| | |
|<---(8) 200 OK SDP TA+TB --| |
| | |
|--------------------(9) INVITE SDP TA------------------->|
| | |
|<-------------------(10) 200 OK SDP B--------------------|
| | |
|-------------------------(11) ACK----------------------->|
| | |
|------(12) ACK SDP A+B----->| |
| | |
| ************************** | ************************** |
|* MEDIA *|* MEDIA *|
| ************************** | ************************** |
| | |
Figure 3: Caller invocation of a transcoding service
conversion). There are various possible solutions for this problem.
One solution consists of using the SDP group attribute with FID
semantics [5]. FID allows requesting that a stream is sent to two
different transport addresses in parallel, as shown below:
a=group:FID 1 2
m=audio 20000 RTP/AVP 0
c=IN IP4 A.domain.com
a=mid:1
m=audio 30000 RTP/AVP 0
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c=IN IP4 T.domain.com
a=mid:2
The problem with this solution is that the majority of the SIP user
agents do not support FID. And even if FID is supported, many user
agents do not support sending simultaneous copies of the same media
stream at the same time.
Therefore, for user agents that do not support FID, requesting T to
replicate the stream will always work. The following session
description requests T to perform speech-to-text and text-to-speech
conversions between the first audio stream and the text stream. In
addition, it requests T to copy of the first audio stream to the
second audio stream and send it to A.
m=audio 40000 RTP/AVP 0
c=IN IP4 B.domain.com
a=source:1
a=sink:2
m=audio 20000 RTP/AVP 0
c=IN IP4 A.domain.com
a=recvonly
a=sink:1
m=text 20002 RTP/AVP 96
c=IN IP4 A.domain.com
a=rtpmap:96 t140/1000
a=source:2
a=sink:1
5 Transcoding services in parallel
Transcoding services sometimes consist of human relays (e.g., a
person performing speech-to-text and text-to-speech conversions for a
session). If the same person is involved in both conversions (i.e.,
from A to B and from B to A), he or she has access to all the
conversation. In order to provide some degree of privacy, sometimes
two different persons are allocated to do the job (i.e., one person
handles A->B and the other B->A). This type of disposition is also
useful for automated transcoding services, where one machine converts
text to synthetic speech (text-to-speech) and a different machine
performs voice recognition (speech-to-text).
The scenario just described involves four different sessions; A-T1,
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T1-B, B-T2 and T2-A. Figure 4 shows the call flow where A invokes T1
and T2.
(1) INVITE SDP AT1
m=text 20000 RTP/AVP 96
c=IN IP4 A.domain.com
a=rtpmap:96 t140/1000
a=sendonly
a=source:1
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
a=recvonly
a=sink:1
(2) INVITE SDP AT2
m=text 20002 RTP/AVP 96
c=IN IP4 A.domain.com
a=rtpmap:96 t140/1000
a=recvonly
a=sink:1
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
a=sendonly
a=source:1
(3) 200 OK SDP T1A+T1B
m=text 30000 RTP/AVP 96
c=IN IP4 T1.domain.com
a=rtpmap:96 t140/1000
a=recvonly
a=source:1
m=audio 30002 RTP/AVP 0
c=IN IP4 T1.domain.com
a=sendonly
a=sink:1
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(5) 200 OK SDP T2A+T2B
m=text 40000 RTP/AVP 96
c=IN IP4 T2.domain.com
a=rtpmap:96 t140/1000
a=sendonly
a=sink:1
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.domain.com
a=recvonly
a=source:1
(7) INVITE SDP T1B+T2B
m=audio 30002 RTP/AVP 0
c=IN IP4 T1.domain.com
a=sendonly
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.domain.com
a=recvonly
(8) 200 OK SDP BT1+BT2
m=audio 50000 RTP/AVP 0
c=IN IP4 B.domain.com
a=recvonly
m=audio 50002 RTP/AVP 0
c=IN IP4 B.domain.com
a=sendonly
(11) 200 OK SDP T1A+T1B
m=text 30000 RTP/AVP 96
c=IN IP4 T1.domain.com
a=rtpmap:96 t140/1000
a=recvonly
a=source:1
m=audio 30002 RTP/AVP 0
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A T1 T2 B
| | | |
|----(1) INVITE SDP AT1--->| | |
| | | |
|----------------(2) INVITE SDP AT2-------------->| |
| | | |
|<-(3) 200 OK SDP T1A+T1B--| | |
| | | |
|---------(4) ACK--------->| | |
| | | |
|<---------------(5) 200 OK SDP T2A+T2B-----------| |
| | | |
|----------------------(6) ACK------------------->| |
| | | |
|-----------------------(7) INVITE SDP T1B+T2B----------------->|
| | | |
|<----------------------(8) 200 OK SDP BT1+BT2------------------|
| | | |
|------(9) INVITE--------->| | |
| | | |
|-------------------(10) INVITE------------------>| |
| | | |
|<-(11) 200 OK SDP T1A+T1B-| | |
| | | |
|<------------(12) 200 OK SDP T2A+T2B-------------| |
| | | |
|------------------(13) INVITE SDP T1B+T2B--------------------->|
| | | |
|<-----------------(14) 200 OK SDP BT1+BT2----------------------|
| | | |
|--------------------------(15) ACK---------------------------->|
| | | |
|---(16) ACK SDP AT1+BT1-->| | |
| | | |
|------------(17) ACK SDP AT2+BT2---------------->| |
| | | |
| ************************ | ********************************** |
|* MEDIA *|* MEDIA *|
| ************************ | ********************************** |
| | | |
| *********************************************** ***********
|* MEDIA *|* MEDIA *|
| *********************************************** | *********** |
| | | |
Figure 4: Transcoding services in parallel
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c=IN IP4 T1.domain.com
a=sendonly
a=sink:1
(12) 200 OK SDP T2A+T2B
m=text 40000 RTP/AVP 96
c=IN IP4 T2.domain.com
a=rtpmap:96 t140/1000
a=sendonly
a=sink:1
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.domain.com
a=recvonly
a=source:1
Since T1 have returned the same SDP in (11) as in (3) and T2 has
returned the same SDP in (12) as in (5), messages (13), (14) and (15)
can be skipped.
(16) ACK SDP AT1+BT1
m=text 20000 RTP/AVP 96
c=IN IP4 A.domain.com
a=rtpmap:96 t140/1000
a=sendonly
a=source:1
m=audio 50000 RTP/AVP 0
c=IN IP4 B.domain.com
a=recvonly
a=sink:1
(17) ACK SDP AT2+BT2
m=text 20002 RTP/AVP 96
c=IN IP4 A.domain.com
a=rtpmap:96 t140/1000
a=recvonly
a=sink:1
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m=audio 50002 RTP/AVP 0
c=IN IP4 B.domain.com
a=sendonly
a=source:1
Four media streams have been established at this point:
1. Text from A to T1.domain.com:30000
2. Audio from T1 to B.domain.com:50000
3. Audio from B to T2.domain.com:40002
4. Text from T2 to A.domain.com:20002
Note that B, the user agent server, needs to support two media
streams; one sendonly and the other recvonly. At present, some user
agents, although they support a single sendrecv media stream, they do
not support a different media line per direction. Implementers are
encouraged to build support for this feature.
6 Transcoding services in serial
In a distributed environment, a complex transcoding service (e.g.,
English text to Spanish speech) is often provided by several servers.
For example, one server performs English text to Spanish text
translation, and its output is feed into a server that performs
text-to-speech conversion. The flow in figure 5 shows how A invokes
T1 and T2.
7 Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
Eric W. Burger
SnowShore Networks, Inc.
Chelmsford, MA
USA
electronic mail: eburger@snowshore.com
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Henning Schulzrinne
Dept. of Computer Science
Columbia University 1214 Amsterdam Avenue, MC 0401
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Arnoud van Wijk
Ericsson EuroLab Netherlands BV
P.O. Box 8
5120 AA Rijen
The Netherlands
electronic mail: Arnoud.van.Wijk@eln.ericsson.se
8 Bibliography
[1] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. 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 et al. , "User requirements for the session
initiation protocol (SIP) in support of deaf, hard of hearing and
speech-impaired individuals," Internet Draft, Internet Engineering
Task Force, Apr. 2002. Work in progress.
[3] J. Rosenberg, J. 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, June 2002. Work in progress.
[4] M. Handley and V. Jacobson, "SDP: session description protocol,"
RFC 2327, Internet Engineering Task Force, Apr. 1998.
[5] G. Camarillo, J. Holler, G. Eriksson, and H. Schulzrinne,
"Grouping of m lines in SDP," Internet Draft, Internet Engineering
Task Force, Feb. 2002. Work in progress.
Full Copyright Statement
Copyright (c) The Internet Society (2002). All Rights Reserved.
This document and translations of it may be copied and furnished to
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or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
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G. Camarillo et. al. [Page 19]
Internet Draft SIP September 9, 2002
A T1 T2 B
| | | |
|----(1) INVITE SDP A-----> | | |
| | | |
|<-(2) 200 OK SDP T1A+T1T2- | | |
| | | |
|----------(3) ACK--------> | | |
| | | |
|-----------(4) INVITE SDP T1T2------------------>| |
| | | |
|<-----------(5) 200 OK SDP T2T1+T2B--------------| |
| | | |
|---------------------(6) ACK-------------------->| |
| | | |
|---------------------------(7) INVITE SDP T2B----------------->|
| | | |
|<--------------------------(8) 200 OK SDP B--------------------|
| | | |
|--------------------------------(9) ACK----------------------->|
| | | |
|---(10) INVITE-----------> | | |
| | | |
|------------------(11) INVITE------------------->| |
| | | |
|<-(12) 200 OK SDP T1A+T1T2-| | |
| | | |
|<-------------(13) 200 OK SDP T2T1+T2B-----------| |
| | | |
|---(14) ACK SDP T1T2+B---> | | |
| | | |
|-----------------------(15) INVITE SDP T2B-------------------->|
| | | |
|<----------------------(16) 200 OK SDP B-----------------------|
| | | |
|----------------(17) ACK SDP T1T2+B------------->| |
| | | |
|----------------------------(18) ACK-------------------------->|
| | | |
| ************************* | ******************* *********** |
|* MEDIA *|* MEDIA *|* MEDIA *|
| ************************* | ******************* | *********** |
| | | |
Figure 5: Transcoding services in serial
included on all such copies and derivative works. However, this
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the copyright notice or references to the Internet Society or other
G. Camarillo et. al. [Page 20]
Internet Draft SIP September 9, 2002
copyrights defined in the Internet Standards process must be
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G. Camarillo et. al. [Page 21]