Internet Engineering Task Force Gonzalo Camarillo
Internet draft Jan Holler
Goran AP Eriksson
Ericsson
June 2001
Expires December 2001
<draft-ietf-mmusic-fid-02.txt>
Grouping of m lines in SDP
Status of this Memo
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Abstract
This document defines two SDP attributes: "groupe" and "mid". They
allow to group together several "m" lines for two different
purposes: for lip synchronization and for receiving media from a
single flow (several media streams), encoded in different formats
during a particular session, in different ports and host interfaces.
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Grouping of m lines in SDP
TABLE OF CONTENTS
1 Media stream identification attribute........................2
2 Groupe attribute.............................................2
3 Lip Synchronization (LS).....................................3
4 Flow Identification (FID)....................................3
4.1 SIP and cellular access......................................3
4.2 DTMF tones...................................................4
5 Media flow definition........................................4
6 FID semantics................................................4
7 Interactions of "groupe" with other media level attributes...5
8 Usage of the "groupe" attribute in SIP.......................6
8.1 Backward compatibility.......................................6
8.2 Caller does not support fid..................................6
8.3 Callee does not support fid..................................6
9 Acknoledgements..............................................7
10 References..................................................7
11 Authors³ Addresses..........................................7
1. Media stream identification attribute
A new "media stream identification" media attribute is defined. It
is used for identifying media streams within a session description.
Its formatting in SDP is described by the following BNF:
mid-attribute = "a=mid:" identification-tag
identification-tag = token
The identification tag is unique within the SDP session description.
2. Group attribute
A new "group" session level attribute is defined. It is used for
grouping together different media streams. Its formatting in SDP is
described by the following BNF:
groupe-attribute = "a=groupe:" semantics space
2*(space identification-tag)
semantics = "LS" | "FID"
This document defines two standard semantics: LS (Lip
Synchronization) and FID (Flow Identification). If in the future it
was needed to standardize further semantics they would need to be
defined in a standards track document. However, defining new
semantics apart from LS and FID is discouraged. Instead, it is
RECOMMENDED to use other session description mechanisms such as
SDPng [1].
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There might be several "a=groupe" lines in a session description.
"a=groupe" lines that contain identification-tags that are not
present in the session description are simply ignored. The
application acts as if the "a=groupe" line did not exist.
3. Lip Synchronization (LS)
The play out of media streams that are grouped together using LS
semantics have to be synchronized. Synchronization is typically
performed using RTCP, which provides enough information to map time
stamps from the different streams into a wall clock.
The following example shows a session description where the audio
and the video stream have to be synchronized.
v=0
o=Laura 289083124 289083124 IN IP4 first.example.com
t=0 0
c=IN IP4 131.160.1.112
a=groupe:LS 1 2
m=audio 30000 RTP/AVP 0
a=mid:1
m=video 30002 RTP/AVP 31
a=mid:2
4. Flow Identification (FID)
The RTSP RFC [2] defines a media stream as "a single media instance,
e.g., an audio stream or a video stream as well as a single
whiteboard or shared application group. When using RTP, a stream
consists of all RTP and RTCP packets created by a source within an
RTP session".
This definition assumes that a single audio (or video) stream maps
into an RTP session. The RTP RFC [3] defines an RTP session as
follows: "For each participant, the session is defined by a
particular pair of destination transport addresses (one network
address plus a port pair for RTP and RTCP)".
However, there are situations where a single media instance, (e.g.,
an audio stream or a video stream) is sent using more than one RTP
session. Two examples (among many others) of this kind of situation
are cellular systems using SIP [4] and systems receiving DTMF tones
on a different host than the voice.
4.1 SIP and cellular access
Systems using a cellular access and SIP as a signalling protocol
need to receive media over the air. During a session the media can
be encoded using different codecs. The encoded media has to traverse
the radio interface. The radio interface is generally characterized
by being bit error prone and associated with relatively high packet
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Grouping of m lines in SDP
transfer delays. In addition, radio interface resources in a
cellular environment are scarce and thus expensive, which calls for
special measures in providing a highly efficient transport [5]. In
order to get an appropriate speech quality in combination with an
efficient transport, precise knowledge of codec properties are
required so that a proper radio bearer for the RTP session can be
configured before transferring the media. These radio bearers are
dedicated bearers per media type, i.e. codec.
Cellular systems typically configure different radio bearers on
different port numbers. Therefore, incoming media has to have
different destination port numbers for the different possible codecs
in order to be routed properly to the correct radio bearer. Thus,
this is an example in which several RTP sessions are used to carry a
single media instance (the encoded speech from the sender).
4.2 DTMF tones
Some voice sessions include DTMF tones. Sometimes the voice handling
is performed by a different host than the DTMF handling. [6]
contains several examples of how application servers in the network
gather DTMF tones for the user while the user receives the encoded
speech on his user agent. In this situations it is necessary to
establish two RTP sessions: one for the voice and the other for the
DTMF tones. Both RTP sessions are logically part of the same media
instance.
5. Media flow definition
The previous examples show that the definition of a media stream in
[2] has to be updated. It cannot be assumed that a single media
instance maps into a single RTP session. Therefore, we introduce the
definition of a media flow:
Media flow consists of a single media instance, e.g., an audio
stream or a video stream as well as a single whiteboard or shared
application group. When using RTP, a media flow comprises one or
more RTP sessions.
For instance, in a two party call where the voice exchanged can be
encoded using GSM or PCM, the receiver wants to receive GSM on a
port number and PCM on a different port number. Two RTP sessions
will be established, one carrying GSM and the other carrying PCM.
At any particular moment just one codec is in use. Therefore, at any
moment one of the RTP sessions will not transport any voice. Here
the systems are dealing with a single media flow, but two RTP
sessions.
6. FID semantics
Several "m" lines grouped together using FID semantics form a media
flow. A media agent handling a media flow that comprises several "m"
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Grouping of m lines in SDP
lines sends media to different destinations (IP address/port number)
depending on the codec used at any moment. If several "m" lines
contain the codec used media is sent to different destinations in
parallel.
For instance, a SIP user agent receives an INVITE with the following
body:
v=0
o=Laura 289083124 289083124 IN IP4 second.example.com
t=0 0
c=IN IP4 131.160.1.112
a=groupe:FID 1 2 3
m=audio 30000 RTP/AVP 0
a=mid:1
m=audio 30002 RTP/AVP 8
a=mid:2
m=audio 30004 RTP/AVP 0 8
a=mid:3
At a particular point of time, if the media agent is sending PCM u-
law (payload 0) it sends RTP packets to ports 30000 and 30004 (first
and third "m" lines). If it is sending PCM A-law (payload 8) it
sends RTP packets to ports 30002 and 30004 (second and third "m"
lines).
Note that if several "m" lines with the same fid value contain the
same codec the media agent MUST send media over several RTP sessions
at the same time.
7 Interactions of "groupe" with other media level attributes
Media level attributes affect a media stream defined by an "m" line.
The presence of "groupe" does not modify this behavior.
For instance, a SIP user agent receives an INVITE with the following
body:
v=0
o=Laura 289083124 289083124 IN IP4 third.example.com
t=0 0
c=IN IP4 131.160.1.112
a=groupe:FID 1 2
m=audio 30000 RTP/AVP 0
a=mid:1
m=audio 30002 RTP/AVP 8
a=recvonly
a=mid:2
The media agent knows that at a certain moment it can send either
PCM u-law to port number 30000 or PCM A-law to port number 30002.
However, the media agent also knows that the other end will only
send PCM u-law (payload 0).
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Grouping of m lines in SDP
Note that the "groupe" attribute used with FID semantics allows to
express uni-directional codecs for a bi-directional media flow, as
it is shown in the example above.
8. Usage of the "groupe" attribute in SIP
SIP [4] is an application layer protocol for establishing,
terminating and modifying multimedia sessions. SIP carries session
descriptions in the bodies of the SIP messages but is independent
from the protocol used for describing sessions. SDP [7] is one of
the protocols that can be used for this purpose.
Appendix B of [4] describes the usage of SDP in relation to SIP. It
states: "The caller and callee align their media description so that
the nth media stream ("m=" line) in the caller³s session description
corresponds to the nth media stream in the callee³s description."
The presence of the "groupe" attribute in an SDP session description
does not modify this behavior.
8.1 Backward compatibility
This document does not define any SIP "Require" header. Therefore,
if one of the SIP user agents does not understand the "groupe"
attribute the standard SDP fall back mechanism is used.
A system that understands the "groupe" attribute MUST add an "mid"
attribute to every "m" line in any SDP session description that it
generates.
8.2 Caller does not support "groupe"
This situation does not represent a problem. The SDP in the INVITE
will not contain any "mid" attribute. The callee knows that the
caller does not support "groupe".
8.3 Callee does not support "groupe"
The callee will ignore the "groupe" attribute, since it does not
understand it. For LS semantics, the callee might decide to perform
or to not perform synchronization between media streams.
For FID semantics, the callee will consider that the session
comprises several media streams.
Different implementations would behave in different ways.
In the case of audio and different "m" lines for different codecs an
implementation might decide to act as a mixer with the different
incoming RTP sessions, which is the correct behavior.
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Grouping of m lines in SDP
An implementation might also decide to refuse the request (e.g. 488
Not acceptable here or 606 Not Acceptable) because it contains
several "m" lines. In this case, the callee does not support the
type of session that the caller wanted to establish. In case the
caller is willing to establish a simpler session anyway, he should
re-try the request without "groupe" attribute and only one "m" line
per flow.
9. Acknowledgments
The authors would like to thank Jonathan Rosenberg, Adam Roach and
Orit Levin for their feedback on this document.
10. References
[1] D. Kutscher/J. Ott/C. Bormann, "Session Description and
Capability Negotiation", draft-ietf-mmusic-sdpng-00.txt, IETF; April
2001. Work in progress.
[2] H. Schulzrinne/A. Rao/R. Lanphier, "Real Time Streaming Protocol
(RTSP)", RFC 2326, IETF; April 1998.
[3] H. Schulzrinne/S. Casner/R. Frederick/V. Jacobson, "RTP: A
Transport Protocol for Real-Time Applications", RFC 1889, IETF;
January 1996.
[4] M. Handley/H. Schulzrinne/E. Schooler/J. Rosenberg, "SIP:
Session Initiation Protocol", RFC 2543, IETF; Mach 1999.
[5] L. Westberg/M. Lindqvist, "Realtime Traffic over Cellular Access
Networks", draft-westberg-realtime-cellular-03.txt, IETF; November
2000. Work in progress.
[6] J. Rosemberg/P.Mataga/H.Schulzrinne, "An Applcation Server
Component Architecture for SIP", draft-rosenberg-sip-app-components-
00.txt, IETF; November 2000. Work in progress.
[7] M. Handley/V. Jacobson, "SDP: Session Description Protocol", RFC
2327, IETF; April 1998.
11. Authors³ Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
Phone: +358 9 299 3371
Fax: +358 9 299 3052
Email: Gonzalo.Camarillo@ericsson.com
Jan Holler
Ericsson Research
Camarillo/Holler/Eriksson 7
Grouping of m lines in SDP
S-16480 Stockholm
Sweden
Phone: +46 8 58532845
Fax: +46 8 4047020
Email: Jan.Holler@era.ericsson.se
Goran AP Eriksson
Ericsson Research
S-16480 Stockholm
Sweden
Phone: +46 8 58531762
Fax: +46 8 4047020
Email: Goran.AP.Eriksson@era.ericsson.se
Camarillo/Holler/Eriksson 8
Datatracker
draft-ietf-mmusic-fid-02
Internet-Draft
