Internet Engineering Task Force Gonzalo Camarillo
Internet draft Jan H÷ller
G÷ran AP Eriksson
Ericsson
June 2000
Expires March 2000
<draft-camarillo-sip-sdp-00.txt>
SDP media alignment in SIP
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 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 media stream, encoded
in different formats during the session, to different ports and host
interfaces.
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 callerÆs session description
corresponds to the nth media stream in the calleeÆs description."
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SDP media alignment in SIP
This way of performing the media alignment implies that all the
media from one single media stream has to be sent to the same port
number. This does not suit fine systems that handle different codecs
in different port numbers. Nor does it suit hosts supporting
multiple interfaces that handle different codecs on different
interface types.
2. 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 [3]. 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) [4],
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).
-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. Examining the payload of every
packet entering the UMTS network would require too much processing
power. It would not a viable solution. Demultiplexing of media
streams should be done at the port level.
So, when a session is negotiated using SIP, a means for providing
different port numbers for different codecs is needed. This
procedure has to be backwards compatible with appendix B of [1].
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SDP media alignment in SIP
Systems using certain architectures have also this requirement in
order to be able to use different audio libraries. Discussions about
this can be found in [5].
3. SIP, SDP and multiple host interfaces
Two typical scenarios apply where a single media stream may be
received or sent on different host interfaces. The first is when the
cellular device is equipped with several access technologies. The
second is the case where media format transcoding may be needed.
Cellular terminals are traditionally equipped with only one access
type, the cellular access. Cellular terminals may also be further
equipped with other higher speed wireless access technologies of
short range nature such as Wireless LAN and Bluetooth.
Multiple accesses on the terminal implies that multiple IP addresses
will be in use for a single terminal.
Applications based on SIP may rely on the fact that these additional
interfaces may also be available for use, but of a more sporadic
nature. The availability and applicability may for instance be based
on coverage or simply by the userÆs preferences.
In a session invitation, the caller (on a mobile SIP terminal) may
be willing to receive a single media stream that could be delivered
on one of the available interfaces. A typical case could be where a
stream of high quality audio can be supported over a high bit rate
interface such as WLAN, and a low rate codec is used over the
cellular access. Depending on the particular situation, the caller
may offer both codec types, whereas the callee may select only one
of the codec types, e.g. depending on what he supports.
If the caller moves out of coverage of the high bit rate access, an
ordinary re-invitation may move the media stream to be transported
over the low bit rate cellular access.
The second scenario concerns the need for a transcoding service.
Speech codecs have been developed basically to be robust against the
high bit error and frame error rate associated with wireless
accesses. One such codec is GSM. Other encoding formats developed
for wired accesses with negligible bit errors do not operate
properly over a lossy link.
As fixed terminals may not support the wireless codecs, while
cellular terminals can generally not support wired codecs, there is
a need for a transcoding function between the two.
The transcoding function is typically placed in another host
connected to the internet. This host is placed on the other side of
the cellular access as viewed from the cellular terminal.
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SDP media alignment in SIP
To make use of the transcoding function, the cellular terminal may
send an INVITE with an SDP part displaying support for the cellular
codec as well as support for the codecs supported by the transcoder.
In this situation, there is one media stream that depending on which
codec is selected, should be directed to different host interfaces.
The callee then makes a selection of codec and sends the media to
the appropriate host interface. The transcoder will also be
configured to forward the transcoded media stream to the cellular
terminal.
4. Alignment of SDP in SIP to support multiple port numbers
A new "flow identification" media attribute is defined. It is used
for identifying media streams within a session. It provides a means
for aligning a number of flows 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 example illustrates its usage.
5. Example of flow identification attribute
In the following example John uses a traditional access such as an
ethernet while Laura has a UMTS/SIP terminal. The caller John sends
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
m=audio 20002 RTP/AVP 0 8
a=fid:2
The callee Laura is on a UMTS/SIP terminal. She configures the
necessary radio bearers and implements the TFTs:
For the first media stream with flow identification equal to 1
(fid:1):
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).
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SDP media alignment in SIP
For the second media stream with flow identification equal to 2
(fid:2):
All the incoming IP packets with destination port UDP 30004 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 30006 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:
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
a=fid:2
m=audio 30006 RTP/AVP 8
a=fid:2
In another example, the callee Laura might want to receive single
flows in different IP addresses depending on the codec. It would
return an SDP description as follows.
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
c=IN IP4 123.123.123.123
a=fid:1
m=audio 30004 RTP/AVP 0
a=fid:2
m=audio 30006 RTP/AVP 8
c=IN IP4 123.123.123.123
a=fid:2
6. 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.
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SDP media alignment in SIP
7. Backward compatibility
Upon the reception of an SDP description with the fid attribute, the
callee must add also this attribute to the response. If the callee
does not understand the fid attribute it will not include fid in the
response. In this case the callee will proceed as it is described in
appendix B of [1]. Thus, a flow will be created for every "m" line
in the session description. The caller will have to issue a re-
INVITE in order to remove the undesirable media streams. RFC2543bis
prevents the caller to send an updated session description in an ACK
request since a complete session description was sent already in the
INVITE request. Until the re-INVITE is issued, the communication
between the terminals is possible because they have enough
information for receiving and sending media.
It would be possible to use different approaches that would resolve
these issues (demultiplexing of media stream on port level) and
provide better backward compatibility. One solution might be to add
additional codecs supported in different IP addresses or port number
in a new a attribute. This way, a callee that did not understand the
attribute would ignore it but still create the correct number of
media streams.
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=addcodec:8 222.222.222.222 30002
m=audio 30004 RTP/AVP 0
a=addcodec:8 222.222.222.222 30006
However, we believe that the fid attribute provides a more general
mechanism to bundle different media lines into a single flow. This
mechanism can be used by other applications different than the one
described in this draft to overcome different problems. We have
chosen generality over a specific solution.
The backward compatibility problem might not exist in a future
depending on the format chosen for SDPng. This draft is also
intended to serve as input for the SDPng work. Not on the syntax
level but on the semantics of the fid attribute.
8. Open issues
How should RTCP be handled?
9. Acronyms
BNF Backus-Naur Form
EDGE Enhanced Data rates for GSM and TDMA/136 Evolution
GSM Global System for Mobile communication
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SDP media alignment in SIP
IP Internet Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
TFT Traffic Flow Template
UMTS Universal Mobile Telecommunication System
WLAN Wireless Local Area Network
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.
[3] L. Westberg/M. Lindqvist, "Realtime Traffic over Cellular Access
Networks", draft-westberg-realtime-cellular-02.txt, IETF; May 2000.
Work in progress.
[4] 3G TS 23.060 v3.2.1 General Packet Radio Service Description.
[5] ftp://ftp.isi.edu/confctrl/confctrl.mail ,thread with subject
"SDP media alignment in SIP".
11. Authors' Addresses
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
Phone: +358 9 299 3371
Fax: +358 9 299 3118
Email: Gonzalo.Camarillo@ericsson.com
Jan H÷ller
Ericsson Research
S-16480 Stockholm
Sweden
Phone: +46 8 58532845
Fax: +46 8 4047020
Email: Jan.Holler@era.ericsson.se
G÷ran AP Eriksson
Ericsson Research
S-16480 Stockholm
Sweden
Phone: +46 8 58531762
Fax: +46 8 4047020
Email: Goran.AP.Eriksson@era.ericsson.se
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