Internet Engineering Task Force Gonzalo Camarillo
Internet draft Jan Holler
Goran AP Eriksson
Ericsson
December 2000
Expires June 2001
<draft-ietf-mmusic-fid-00.txt>
SDP media alignment in SIP
Status of this Memo
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Abstract
This document defines an SDP media attribute. This attribute is
intended to be used in conjunction with SIP in order to align
different media streams belonging to a session. The use of this
attribute allows sending media from a single flow (several media
streams), encoded in different formats during the session, to
different ports and host interfaces.
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SDP media alignment in SIP
TABLE OF CONTENTS
1 Introduction.................................................2
2 Media flow definition........................................2
2.1 SIP and cellular access......................................3
2.2 DTMF tones...................................................4
3 Flow identification attribute................................4
4 Examples of flow identification attribute....................4
4.1 UMTS/SIP terminal............................................4
4.2 Application Server Components................................5
5 Media-level versus session-level attribute...................7
6 Backward compatibility.......................................7
6.1 Caller does not support fid..................................7
6.2 Callee does not support fid..................................7
7 Behavior of UAs..............................................8
8 Acronyms.....................................................9
9 Acknoledgements..............................................9
10 References...................................................9
11 Authors³ Addresses..........................................10
1. Introduction
SIP [1] 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 [2] is one of
the protocols that can be used for this purpose.
Appendix B of [1] 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."
This way of performing the media alignment is not efficient when a
single flow comprises several media streams. This is a common
situation when AP (Application Sever) components [3] are employed.
It is also common for systems that handle different codecs on
different port numbers (or on different interfaces).
2. Media flow definition
The RTSP RFC [4] 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 [5] 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)".
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SDP media alignment in SIP
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 and systems receiving DTMF tones on a
different host than the voice. Both examples are described in later
sections.
We introduce the definition of 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 flow (one audio stream) and
two RTP sessions.
2.1 SIP and cellular access
Systems using a cellular access (such as UMTS or EDGE) 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 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 [6]. 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.
In UMTS, for instance, when the RTP packets shall be delivered over
the air interface, a packet filtering function routes the packets to
the proper radio bearer towards the UMTS/SIP terminal. The packet
filtering function operates using a Traffic Flow Template (TFT) [7],
which is established when configuring the radio bearer. The TFT
hence specifies the profile of the data that should be carried by
the radio bearer. A TFT can contain the following data:
-Source Address and Subnet Mask.
-Protocol Number (IPv4) / Next Header (IPv6).
-Destination Port Range.
-Source Port Range.
-IPSec Security Parameter Index (SPI).
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SDP media alignment in SIP
-Type of Service (TOS) (IPv4) / Traffic class (IPv6) and Mask.
-Flow Label (IPv6).
It is worth noticing that just certain combinations of these
parameters are allowed.
The media has to have different destination port numbers for the
different possible codecs in order to be filtered and routed
properly to the correct radio bearer. Therefore, several RTP
sessions are used for a single media flow.
2.2 DTMF tones
Some voice sessions include DTMF tones. Sometimes the voice handling
is performed by a different host than the DTMF handling (e.g.
section 5.4, figures 3 and 4 of [3]). 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 flow.
3. Flow identification attribute
A new "flow identification" media attribute is defined. It is used
for identifying media flows within a session. It provides a means
for aligning a number of flows (rather than a number of media
streams) within a session between members participating in the
session. Its formatting in SDP is described by the following BNF:
fid-attribute = "a=fid:" identification-tag
identification-tag = token
The identification tag is unique within the SDP session description.
The following examples illustrate its usage.
4. Examples of flow identification attribute
4.1 UMTS/SIP terminal
In the following example John uses a traditional access such as an
ethernet while Laura has a UMTS/SIP terminal. The caller John sends
an INVITE with the following session description to the callee
Laura.
v=0
o=John 289085535 289085535 IN IP4 first.example.com
t=0 0
c=IN IP4 111.111.111.111
m=audio 20000 RTP/AVP 0 8
a=fid:1
Camarillo/Holler/Eriksson 4
SDP media alignment in SIP
The callee Laura is on a UMTS/SIP terminal. She configures the
necessary radio bearers and implements the TFTs:
All the incoming IP packets with destination port UDP 30000 will be
carried by the radio access bearer configured for G-711 u-law
(payload type 0).
All the incoming IP packets with destination port UDP 30002 will be
carried by the radio access bearer configured for G-711 A-law
(payload type 8).
Accordingly, the following SDP is returned to the caller in a 200 OK
response:
v=0
o=Laura 289083124 289083124 IN IP4 second.example.com
t=0 0
c=IN IP4 222.222.222.222
m=audio 30000 RTP/AVP 0
a=fid:1
m=audio 30002 RTP/AVP 8
a=fid:1
The ACK carries the definitive SDP from the caller:
v=0
o=John 289085535 289085535 IN IP4 first.example.com
t=0 0
c=IN IP4 111.111.111.111
m=audio 20000 RTP/AVP 0
a=fid:1
m=audio 20002 RTP/AVP 8
a=fid:1
With the current way of performing SDP media alignment in SIP the
callee would have accepted the call and immediately after re-INVITEd
the caller with the new SDP. The fid attribute saves many RTTs.
Besides saving bandwidth and RTTs the fid attribute provides a means
for describing a logical relationship between media streams that
belong to the same flow.
4.2 Application Server Components
Section 5.4 of "An Application Server Architecture for SIP" [3]
contains two examples (figures 3 and 4) where DTMF tones are
received by a different host than the voice stream. In both
situations using the fid attribute to perform media alignment would
save a tremendous amount of messages exchanged and reduce the global
session establishment time.
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SDP media alignment in SIP
Let us take figure 4. A UAC sends an INVITE with just a voice
stream. There are two ASs in the path that want to receive DTMF
tones.
Three steps are needed in order to set the session up:
1) A session is established between the UAC and the callee. This
involves three messages from the callerÆs point of view (INVITE-
200 OK-ACK).
2) The session is modified by A (one of the ASs that wants to
receive DTMF tones). It adds an "m" line to the session
description indicating that it wants to receive DTMF tones. This
involves three more messages from the callerÆs point of view
(INVITE-200 OK-ACK)
3) The session is modified once more by B (the other AS that also
wants to receive DTMF tones). It adds another "m" line indicating
that it wants to receive DTMF tones. This involves three more
messages from the callerÆs point of view (INVITE-200 OK-ACK).
Caller A B Callee
| | | |
|(1) SIP INV | | |
|-------------->|(2) SIP INV | |
| |--------------->|(3) SIP INV |
| | |---------------->|
| | |(4) 200 OK |
| |(5) 200 OK |<----------------|
|(6) 200 OK |<---------------| |
|<--------------| | |
|(7) SIP ACK | | |
|-------------->|(8) SIP ACK | |
| |--------------->|(9) SIP ACK |
| | |---------------->|
|(10) SIP INV | | |
|<--------------| | |
|(11) 200 OK | | |
|-------------->| | |
|(12) SIP ACK | | |
|<--------------| | |
| | | |
| |(13) SIP INV | |
|(14) SIP INV |<---------------| |
|<--------------| | |
|(15) 200 OK | | |
|-------------->|(16) 200 OK | |
| |--------------->| |
| |(17) SIP ACK | |
|(18) SIP ACK |<---------------| |
|<--------------| | |
| | | |
Figure 4 of "An AS Component Architecture for SIP" [3]
Camarillo/Holler/Eriksson 6
SDP media alignment in SIP
The whole session is not correctly set up until the end of this
sequence of messages. If the caller is using a low-rate access this
can take a long time.
The use of the fid attribute would reduce these nine messages that
the caller sees to just three (INVITE-200 OK-ACK). B would add an
"m" line to the 200 OK from the callee with the same fid value as
the voice stream. Then A would add another "m" line, again with the
same fid value than the two previous "m" lines.
As a result, the caller receives a 200 OK indicating that just one
flow is established, but also that all the DTMF tones should be sent
to A and B. For a low-rate access the establishment time has been
reduced a lot.
Note that the caller sends an updated SDP in the ACK with the local
RTP ports for all the "m" lines received in the 200 OK.
5. Media-level versus session-level attribute
Syntactically fid is a media-level attribute. It provides
information about a media stream defined by an "m" line.
Semantically fid would be defined as a session-level attribute since
it provides flow hierarchy inside a session description.
6. Backward compatibility
A system that understands the fid attribute MUST add it to any SDP
session description that it generates.
If a response to a request that included the fid attribute also
includes it media alignment is performed based on the fid attribute
rather than on matching of nth lines.
6.1 Caller does not support fid
This situation does not represent a problem. The SDP in the INVITE
will not contain any fid attribute and the callee will use the "nth-
line" method to perform media alignment.
The callee will need a re-INVITE in order to receive the proper
media encoding on the proper interface.
6.2 Callee does not support fid
The callee will ignore the fid attribute. It 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.
Camarillo/Holler/Eriksson 7
SDP media alignment in SIP
If an implementation decides to refuse the request (e.g. 488 Not
acceptable here or 606 Not Acceptable) the caller should re-try the
request without the fid attribute and only one "m" line per flow.
Note that even re-INVITEs without the fid attribute adding new "m"
lines would probably fail in this situation because the callee does
not support multiple "m" lines. Therefore, this problem is related
to UAs that do not handle multiple "m" lines rather than to the fid
attribute.
7. Behavior of UAs
UAs supporting the fid attribute can add new "m" lines belonging to
an existing flow (identified by a fid value) in re-INVITEs and 200
OK responses. UAs MUST NOT add "m" lines to existing flows in ACKs
since it would be impossible to receive the remote RTP/RTCP port for
the new "m" line.
A UA handling a media flow that comprises several "m" 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 UA receives the following 200 OK:
v=0
o=Laura 289083124 289083124 IN IP4 second.example.com
t=0 0
c=IN IP4 222.222.222.222
m=audio 30000 RTP/AVP 0
a=fid:1
m=audio 30002 RTP/AVP 8
a=fid:1
m=audio 30004 RTP/AVP 0 8
a=fid:1
At a particular point of time, if it 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 UA MUST send several RTP sessions in parallel. A UA
that sends an INVITE with a single "m" is willing to send one RTP
session at a time, but upon reception of a 200 OK might be asked to
send more than one RTP session in parallel. If the UA is not willing
to do so (e.g. due to bandwidth constraints) it should BYE the
session.
In order to avoid this situation UAS should follow certain
guidelines. If it is essential for the UAS that the UAC sends
several RTP sessions in parallel (e.g. two APs need to gather DTMF
Camarillo/Holler/Eriksson 8
SDP media alignment in SIP
tones) the UAS should use the fid attribute in the 200 OK to include
these "m" lines.
In this situation, if the UAC does not support sending RTP sessions
in parallel the UAS is not willing to accept the session. Thus, when
the UAC BYEs the session the result is the one expected (session
terminated).
If re-INVITEs had been used (instead of the fid attribute) the UAS
would have sent a BYE when the first re-INVITE had failed. Thus, the
result is the same as when using the fid attribute.
On the other hand, if it is desirable but not essential for the UAS
that the UAC sends several RTP sessions in parallel, the UAS should
use re-INVITEs to add new "m" lines. If a re-INVITE fails, the UAS
would continue with the session with a single RTP session at a time.
8. Acronyms
AP Application Server
BNF Backus-Naur Form
DTMF Dual Tone Multi Frequency
EDGE Enhanced Data rates for GSM and TDMA/136 Evolution
GSM Global System for Mobile communication
IP Internet Protocol
PCM Pulse Code Modulation
RFC Request For Comments
RTCP RTP Control Protocol
RTP Real-time Transport Protocol
RTSP Real-Time Streaming Protocol
RTT Round Trip Time
SDP Session Description Protocol
SIP Session Initiation Protocol
TFT Traffic Flow Template
UA User Agent
UAC User Agent Client
UAS User Agent Server
UMTS Universal Mobile Telecommunication System
WLAN Wireless Local Area Network
9. Acknowledgments
The authors would like to thank Jonathan Rosenberg and Adam Roach
for their feedback on this document.
10. References
[1] M. Handley/H. Schulzrinne/E. Schooler/J. Rosenberg, "SIP:
Session Initiation Protocol", RFC 2543, IETF; Mach 1999.
[2] M. Handley/V. Jacobson, "SDP: Session Description Protocol", RFC
2327, IETF; April 1998.
Camarillo/Holler/Eriksson 9
SDP media alignment in SIP
[3] J. Rosemberg/P.Mataga/H.Schulzrinne, "An Applcation Server
Component Architecture for SIP", draft-rosenberg-sip-app-components-
00.txt, IETF; November 2000.
[4] H. Schulzrinne/A. Rao/R. Lanphier, "Real Time Streaming Protocol
(RTSP)", RFC 2326, IETF; April 1998.
[5] H. Schulzrinne/S. Casner/R. Frederick/V. Jacobson, "RTP: A
Transport Protocol for Real-Time Applications", RFC 1889, IETF;
January 1996.
[6] L. Westberg/M. Lindqvist, "Realtime Traffic over Cellular Access
Networks", draft-westberg-realtime-cellular-03.txt, IETF; November
2000. Work in progress.
[7] 3G TS 23.060 v3.2.1 General Packet Radio Service Description.
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
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 10
Datatracker
draft-ietf-mmusic-fid-00
Internet-Draft
