Internet Engineering Task Force SIP WG
Internet Draft G. Camarillo
Ericsson
E. Burger
Brooktrout
H. Schulzrinne
Columbia University
A. van Wijk
Viataal
draft-ietf-sipping-transc-3pcc-02.txt
September 17, 2004
Expires: March, 2005
Transcoding Services Invocation in the Session Initiation
Protocol (SIP) Using Third Party Call Control (3pcc)
STATUS OF THIS MEMO
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Abstract
This document describes how to invoke transcoding services using SIP
and third party call control. 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 General Overview .................................... 3
3 Third Party Call Control Flows ...................... 3
3.1 Terminology ......................................... 4
3.2 Callee's Invocation ................................. 4
3.3 Caller's Invocation ................................. 9
3.4 Receiving the Original Stream ....................... 9
3.5 Transcoding Services in Parallel .................... 11
3.6 Transcoding Services in Serial ...................... 15
4 Security Considerations ............................. 17
5 IANA Considerations ................................. 17
6 Authors' Addresses .................................. 17
7 Normative References ................................ 17
8 Informative References .............................. 18
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1 Introduction
The framework for transcoding with SIP [4] describes how two SIP [1]
UAs (User Agents) 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. 3pcc (third party call control) [2] is one way to
perform such invocation.
2 General Overview
In the 3pcc model for transcoding invocation, a transcoding server
that provides a particular transcoding service (e.g., speech-to-text)
is identified by a URI. A UA that wishes to invoke that service sends
an INVITE request to that URI establishing a number of media streams.
The way the transcoder manipulates and manages the contents of those
media streams (e.g., the text received over the text stream is
transformed into speech and sent over the audio stream) is service
specific.
All the call flows in this document use SDP. The same call
flows could be used with another session description
protocol that provided similar session description
capabilities.
3 Third Party Call Control Flows
Given two UAs (A and B) and a transcoding server (T), 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 callee (B), invokes the transcoding services.
Section 3.2 deals with callee invocation and Section 3.3 deals with
caller invocation.
In all our 3pcc flows we have followed a general principle; a 200
(OK) response from the transcoding service has to be received before
contacting the callee. This tries to ensure that the transcoding
service will be available when the callee accepts the session.
Still, the transcoding service does not know the exact type of
transcoding it will be performing until the callee accepts the
session. So, there are always chances of failing to provide
transcoding services after the callee has accepted the session. A
system with tough requirements could use preconditions to avoid this
situation. When preconditions are used, the callee is not alerted
until everything is ready for the session.
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3.1 Terminology
All the flows 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.
3.2 Callee's Invocation
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.
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.
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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's invocation of a transcoding service
(1) INVITE SDP A
m=audio 20000 RTP/AVP 0
c=IN IP4 A.example.com
(2) INVITE SDP A+B
m=audio 20000 RTP/AVP 0
c=IN IP4 A.example.com
m=text 40000 RTP/AVP 96
c=IN IP4 B.example.com
a=rtpmap:96 t140/1000
(3) 200 OK SDP TA+TB
m=audio 30000 RTP/AVP 0
c=IN IP4 T.example.com
m=text 30002 RTP/AVP 96
c=IN IP4 T.example.com
a=rtpmap:96 t140/1000
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(5) 200 OK SDP TA
m=audio 30000 RTP/AVP 0
c=IN IP4 T.example.com
Four media streams (i.e., two bi-directional streams) have been
established at this point:
1. Audio from A to T.example.com:30000
2. Text from T to B.example.com:40000
3. Text from B to T.example.com:30002
4. Audio from T to A.example.com:20000
When either A or B decide to terminate the session, they send a BYE
indicating that the session is over.
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.
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.
This scenario (Figure 2) 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). On the other hand, if T happens to return the
same SDP TA+TB in (8) as in (3), B can skip messages (9), (10), and
(11). So, 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:
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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's invocation after initial INVITE without SDP
(2) INVITE SDP A+B
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
m=text 40000 RTP/AVP 96
c=IN IP4 B.example.com
a=rtpmap:96 t140/1000
(3) 200 OK SDP TA+TB
m=audio 30000 RTP/AVP 0
c=IN IP4 T.example.com
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m=text 30002 RTP/AVP 96
c=IN IP4 T.example.com
a=rtpmap:96 t140/1000
(5) 200 OK SDP TA
m=audio 30000 RTP/AVP 0
c=IN IP4 T.example.com
(6) ACK SDP A
m=audio 20000 RTP/AVP 0
c=IN IP4 A.example.com
(8) 200 OK SDP TA+TB
m=audio 30004 RTP/AVP 0
c=IN IP4 T.example.com
m=text 30006 RTP/AVP 96
c=IN IP4 T.example.com
a=rtpmap:96 t140/1000
(9) INVITE SDP TA
m=audio 30004 RTP/AVP 0
c=IN IP4 T.example.com
(10) 200 OK SDP A
m=audio 20002 RTP/AVP 0
c=IN IP4 A.example.com
(12) ACK SDP A+B
m=audio 20002 RTP/AVP 0
c=IN IP4 A.example.com
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m=text 40000 RTP/AVP 96
c=IN IP4 B.example.com
a=rtpmap:96 t140/1000
Four media streams (i.e., two bi-directional streams) have been
established at this point:
1. Audio from A to T.example.com:30004
2. Text from T to B.example.com:40000
3. Text from B to T.example.com:30006
4. Audio from T to A.example.com:20002
3.3 Caller's 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. The call flow we provide here is slightly different than the ones
in [2]. In [2], the controller establishes a session between two user
agents, which are 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 [2].
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.
3.4 Receiving the Original Stream
Sometimes, as pointed out in the requirements for SIP in support of
deaf, hard of hearing, and speech-impaired individuals [5], 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
conversion). There are various possible solutions for this problem.
One solution consists of using the SDP group attribute with FID
semantics [3]. FID allows requesting that a stream is sent to two
different transport addresses in parallel, as shown below:
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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's invocation of a transcoding service
a=group:FID 1 2
m=audio 20000 RTP/AVP 0
c=IN IP4 A.example.com
a=mid:1
m=audio 30000 RTP/AVP 0
c=IN IP4 T.example.com
a=mid:2
The problem with this solution is that the majority of the SIP user
agents do not support FID. Moreover, only a small fraction of the few
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UAs that do support FID, support sending simultaneous copies of the
same media stream at the same time. In addition, FID forces both
copies of the stream to use the same codec.
So, we recommend that T (instead of one of the user agent) replicates
the media stream. The transcoder T receiving the following session
description performs speech-to-text and text-to-speech conversions
between the first audio stream and the text stream. In addition, T
copies the first audio stream to the second audio stream and sends it
to A.
m=audio 40000 RTP/AVP 0
c=IN IP4 B.example.com
m=audio 20000 RTP/AVP 0
c=IN IP4 A.example.com
a=recvonly
m=text 20002 RTP/AVP 96
c=IN IP4 A.example.com
a=rtpmap:96 t140/1000
3.5 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,
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.example.com
a=rtpmap:96 t140/1000
a=sendonly
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
a=recvonly
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(2) INVITE SDP AT2
m=text 20002 RTP/AVP 96
c=IN IP4 A.example.com
a=rtpmap:96 t140/1000
a=recvonly
m=audio 20000 RTP/AVP 0
c=IN IP4 0.0.0.0
a=sendonly
(3) 200 OK SDP T1A+T1B
m=text 30000 RTP/AVP 96
c=IN IP4 T1.example.com
a=rtpmap:96 t140/1000
a=recvonly
m=audio 30002 RTP/AVP 0
c=IN IP4 T1.example.com
a=sendonly
(5) 200 OK SDP T2A+T2B
m=text 40000 RTP/AVP 96
c=IN IP4 T2.example.com
a=rtpmap:96 t140/1000
a=sendonly
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.example.com
a=recvonly
(7) INVITE SDP T1B+T2B
m=audio 30002 RTP/AVP 0
c=IN IP4 T1.example.com
a=sendonly
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.example.com
a=recvonly
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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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(8) 200 OK SDP BT1+BT2
m=audio 50000 RTP/AVP 0
c=IN IP4 B.example.com
a=recvonly
m=audio 50002 RTP/AVP 0
c=IN IP4 B.example.com
a=sendonly
(11) 200 OK SDP T1A+T1B
m=text 30000 RTP/AVP 96
c=IN IP4 T1.example.com
a=rtpmap:96 t140/1000
a=recvonly
m=audio 30002 RTP/AVP 0
c=IN IP4 T1.example.com
a=sendonly
(12) 200 OK SDP T2A+T2B
m=text 40000 RTP/AVP 96
c=IN IP4 T2.example.com
a=rtpmap:96 t140/1000
a=sendonly
m=audio 40002 RTP/AVP 0
c=IN IP4 T2.example.com
a=recvonly
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.example.com
a=rtpmap:96 t140/1000
a=sendonly
m=audio 50000 RTP/AVP 0
c=IN IP4 B.example.com
a=recvonly
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(17) ACK SDP AT2+BT2
m=text 20002 RTP/AVP 96
c=IN IP4 A.example.com
a=rtpmap:96 t140/1000
a=recvonly
m=audio 50002 RTP/AVP 0
c=IN IP4 B.example.com
a=sendonly
Four media streams have been established at this point:
1. Text from A to T1.example.com:30000
2. Audio from T1 to B.example.com:50000
3. Audio from B to T2.example.com:40002
4. Text from T2 to A.example.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.
3.6 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.
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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
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4 Security Considerations
This document describes how to use third party call control to invoke
transcoding services. It does not introduce new security
considerations besides the ones discussed in [2].
5 IANA Considerations
This document has no actions for IANA.
6 Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
Eric Burger
Brooktrout Technology, Inc.
18 Keewaydin Way
Salem, NH 03079
USA
electronic mail: eburger@ieee.org
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
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands
electronic mail: a.vwijk@viataal.nl
7 Normative References
[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.
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Internet Draft 3pcc Transcoding in SIP September 17, 2004
[2] J. Rosenberg, J. Peterson, H. Schulzrinne, and G. Camarillo,
"Best current practices for third party call control (3pcc) in the
session initiation protocol (SIP)," RFC 3725, Internet Engineering
Task Force, Apr. 2004.
[3] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
"Grouping of media lines in the session description protocol (SDP),"
RFC 3388, Internet Engineering Task Force, Dec. 2002.
8 Informative References
[4] 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.
[5] 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.
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such proprietary rights by implementers or users of this
specification can be obtained from the IETF on-line IPR repository at
http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at ietf-
ipr@ietf.org.
G. Camarillo et. al. [Page 18]
Internet Draft 3pcc Transcoding in SIP September 17, 2004
Disclaimer of Validity
This document and the information contained herein are provided on an
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INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
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Copyright Statement
Copyright (C) The Internet Society (2004). This document is subject
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Acknowledgment
Funding for the RFC Editor function is currently provided by the
Internet Society.
G. Camarillo et. al. [Page 19]
