Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Standards Track August 17, 2011
Expires: February 18, 2012
SDP Grouping for Single RTP Sessions
draft-alvestrand-one-rtp-01
Abstract
This document describes an extension to the Session Description
Protocol (SDP) to describe RTP sessions where media of multiple top
level types, for example audio and video, are carried in the same RTP
session.
This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
consideration.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on February 18, 2012.
Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements for a solution . . . . . . . . . . . . . . . . . 3
3. SDP Grouping Framework Parameter . . . . . . . . . . . . . . . 3
4. Use in Offer/Answer . . . . . . . . . . . . . . . . . . . . . 4
5. Parameter combining . . . . . . . . . . . . . . . . . . . . . 5
6. Interaction with other extensions . . . . . . . . . . . . . . 5
7. RTCP bandwidth considerations . . . . . . . . . . . . . . . . 5
8. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
10. Security Considerations . . . . . . . . . . . . . . . . . . . 8
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 9
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9
12.1. Normative References . . . . . . . . . . . . . . . . . . 9
12.2. Informative References . . . . . . . . . . . . . . . . . 9
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 9
A.1. From draft-alvestrand-one-rtp-00 to -01 . . . . . . . . . 10
Appendix B. Other matters . . . . . . . . . . . . . . . . . . . . 10
B.1. RTCP session establishment . . . . . . . . . . . . . . . 10
B.2. Renegotiation . . . . . . . . . . . . . . . . . . . . . . 10
B.3. ICE negotiation sequence . . . . . . . . . . . . . . . . 11
B.4. SDP-inspecting intermediaries . . . . . . . . . . . . . . 11
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11
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1. Introduction
In the work with the RTCWEB specifications
[I-D.ietf-rtcweb-overview], a need was discovered for representing
within the SDP framework an SDP session consisting of a single RTP
session, where that single RTP session, mapped to a single transport
flow, contained multiple top-level data types.
This is advantageous for the use case where there is no desire for
different treatment by the network of the different flows in the
session, there exist other appropriate mechanisms (for instance based
on SSRC) to identify the flows in the session to the applications,
and where the handling of multiple RTP sessions would increase the
work required to establish the session (for instance by requiring
multiple ICE [RFC5245] negotiations, or handling of failure cases
where one RTP session is established and another is not).
This document describes how to represent such a session.
2. Requirements for a solution
The requirements for our representation are:
o It should be possible to represent an SDP session consisting of a
single RTP session, where that session carries both audio and
video.
o If this description is presented in an Offer in the offer/answer
model to an entity that does not understand it, the resulting
Answer should contain a valid description of an SDP session
consisting of one video RTP session and one audio RTP session.
3. SDP Grouping Framework Parameter
This document defines a new semantics extension called TOGETHER
within the SDP Grouping framework [RFC5888].
If this semantics extension is present in an SDP Session-level
a=group: line, the semantics are that the two or more m lines are
intended to be read as components of a single RTP session, creating a
single SSRC numbering space that can contain components of all the
types described in the referenced media sections.
The following properties of the media sections are REQUIRED:
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o The defined RTPMAP values of the section MUST NOT overlap
o The "proto" profile of the sections MUST be the same (e.g RTP/
AVPF)
The media sections MAY contain connection data (port numbers or ICE
parameters), but some of these may be ignored in processing (see next
section).
The reason for the requirement for systematic proto is that there are
many combinations that don't make sense (for instance "RTP/AVPF" in
one section and "RTP/SAVP" in another would make encryption and
availability of TMMBR depend on the outcome of negotiation, which
seems strange). The cases where combinations make sense (RTP/AVPF
with UDP/FEC for instance) also usually require that separate RTP
sessions be used. [[QUESTION IN DRAFT: Are there sensible
combinations?]]
4. Use in Offer/Answer
This extension MAY be included in a Offer; as specified in RFC 5888
section 9 when describing SIP usage, if it is not included in an
Offer, it MUST NOT be included in an answer.
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
session, and the parameters for the other sections MUST be ignored.
The following parameters are taken from the first section only:
o Port number from the m= line
o All media-level attributes defined in RFC 5245 section 15.1 - this
includes "candidate", "remote-candidates", "ice-mismatch", "ice-
ufrag", "ice-pwd"
The bandwidth of the "m" line is treated specially: The values for
all "m=" lines in the group are added together, and the resulting
value is taken to be the negotiated bandwidth value for the RTP
session.
The expected behaviour when the extension is present in an offer and
not understood is that the generated answer will not contain the
"a=group:TOGETHER" line, and that each sections' parameters will be
used.
If the answerer wishes to refuse an entire m= section, while
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accepting the RTP session otherwise, he MAY indicate this by setting
the port number of the relevant section to zero. In all other cases,
the port number of second and subsequent sections is to be ignored.
The answerer cannot refuse the first m= section and establish the
call. [[NOTE IN DRAFT: This could be relaxed by allowing it when ICE
is in use, or saying that the first non-zero port number is used.
Both lead to nonobvious edge cases. Is the extra complexity worth
it?]]
5. Parameter combining
The general approach taken when combining the sections is to treat
the parameters as if they were all contained in the same section.
However, a number of special considerations have to apply.
o For the "a=rtpmap" lines, their interpretation depends on the "m="
line they occur under, even after combination; "a=rtpmap:8 PCMA/
8000" needs to carry the info that it occured under an "m=audio"
line.
o For parameters that can only occur once in a section, such as the
port number from the m= line, the occurence of the parameter that
comes from the first section needs to take precedence.
o The "b=" (bandwidth) line needs to be considered specially, since
the values are to be added together.
o Media-level ICE attributes need to come from the first section
only, even though syntax would allow more occurences.
[[QUESTION IN DRAFT: Should we instead say that "all lines come from
the first section except a=rtpmap amd b= lines? Is there a case for
other parameters that need to be in second-and-subsequent sections?]]
6. Interaction with other extensions
If other extensions modify the bandwidth calculation algorithm, those
extensions will have to take into consideration how bandwidth from
multiple sections of the SDP description should be merged.
7. RTCP bandwidth considerations
A concern has been raised that when audio and video are combined, the
bandwidth of RTCP reports required for an audio stream may exceed the
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bandwidth of the audio stream itself, which seems a bit bizarre.
While not critical (overall RTCP bandwidth is still limited to 5% of
the total bandwidth), this warrants a little more study.
Considering a combined RTP session with one sender and one recipient,
four 1-Mbit/sec video flows and four 100-Kbit/sec audio flows flowing
in one direction.
The total bandwidth is 4.4 Mbit/sec, so if the RTCP share of the
bandwidth is 5% as recommended by [RFC3550] section 6.2, the RTCP
bandwidth limit is 220 Kbits/sec. Eight SSRCs need to be reported
on.
[CHECK: Does this mean there are 9 reporters (8 senders and 1
receiver), or 2 reporters (1 of each)? Assuming 9.]
Each report sender will have 24.4 Kbits/second of RTCP bandwidth at
its disposal. Assuming a packet size of 100 bytes (11 bytes per SSRC
reported on), the maximum RTCP rate allowed is 30 RTCP packets per
second, which is slightly slower than the typical audio heartbeat
flow of 50 packets per second (20 ms interval).
If this is deemed excessive, one can adopt the RTP/AVPF model of
5-second regular RTCP reports with additional availability of "on-
demand" RTCP packets. But the RTCP feedback interval also enters
into congestion control algorithms, which may complicate the picture.
8. Examples
The examples are taken from RFC 4317, "SDP Offer/Answer Examples".
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Offer
v=0
o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
s=
c=IN IP4 host.atlanta.example.com
t=0 0
a=group:TOGETHER foo bar
m=audio 49170 RTP/AVP 0 8 97
a=mid:foo
b=AS:200
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
m=video 51372 RTP/AVP 31 32
a=mid:bar
b=AS:1000
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
This is a request to have both audio and video sent over port 49170.
If this can't be done, audio will be sent over port 49170, and video
will be sent on port 51372. The total bandwidth, if combined, is
1200 Kbits/second; if separated, 200 Kbits goes to audio and 1000
Kbits goes to video.
Answer, from an entity that understands TOGETHER
v=0
o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
s=
c=IN IP4 host.biloxi.example.com
t=0 0
a=group:TOGETHER foo bar
m=audio 49174 RTP/AVP 0
a=mid:foo
b=AS:200
a=rtpmap:0 PCMU/8000
m=video 49170 RTP/AVP 32
a=mid:bar
b=AS:1000
a=rtpmap:32 MPV/90000
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Answer, from an entity that understands grouping, but does not
understand TOGETHER
v=0
o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
s=
c=IN IP4 host.biloxi.example.com
t=0 0
m=audio 49174 RTP/AVP 0
a=mid:foo
a=rtpmap:0 PCMU/8000
b=AS:200
m=video 49170 RTP/AVP 32
a=mid:bar
a=rtpmap:32 MPV/90000
b=AS:1000
Answer, from an entity that does not understand grouping
v=0
o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
s=
c=IN IP4 host.biloxi.example.com
t=0 0
m=audio 49174 RTP/AVP 0
a=rtpmap:0 PCMU/8000
b=AS:200
m=video 49170 RTP/AVP 32
a=rtpmap:32 MPV/90000
b=AS:1000
9. IANA Considerations
This document requests IANA to register the new SDP Grouping semantic
extension called TOGETHER.
10. Security Considerations
No new security issues have been raised specifically for this
extension.
Third-party interceptors that sniff negotiation but do not understand
the extension may end up listening to the wrong port number for some
of the media flows. This is not deemed greatly harmful.
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11. Acknowledgements
This draft is based on a discussion between a number of participants
at the Quebec City IETF, July 2011, about the issue of multiplexing
audio and video on a single network transport using RTP.
Thanks to (in alphabetical order) Christer Holmberg, Randell Jesup,
Paul Kyzivat, Muthu Arul Pozhi Perumal, Justin Uberti for reviews and
comments.
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
12.2. Informative References
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-00 (work
in progress), June 2011.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
Appendix A. Change log
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A.1. From draft-alvestrand-one-rtp-00 to -01
Added change log and "other matters" appendix.
Added section on parameter combining.
Added an example for an entity that does not understand grouping.
Added port-zero and no-codecs methods for refusing a section.
Appendix B. Other matters
This appendix is not normative.
A number of matters have been raised in discussion on this draft.
Some of the discussions have led to actual changes, others have
pointed to the need to identify other documents as authoritative on
these issues. This appendix serves to collect such matters.
B.1. RTCP session establishment
This document does not specify anything about RTCP sessions. Since a
purpose of the specification is to minimize the number of transport
connections, it's natural to use the "a=rtcp-mux" attribute defined
in [RFC5761], but this document does not specify anything about that
matter.
If the TOGETHER extension is successfully negotiated, there will be
only one RTCP session for the RTP session described by the TOGETHER
group.
Since the purpose of this extension is to have multiple SSRCs on one
RTP session, it may be more important than usual to get SSRC
metainformation such as CNAME quicly; this can be done by using the
recommendation in [RFC6051]to send an RTCP SR at once when starting
the flow, and the recipient can use the RTCP-SR-REQ defined in the
same document to request an RTCP SR if the initial one is lost.
B.2. Renegotiation
After the session has been established, there might occur a need to
change its parameters. At the moment, no issues that are different
from the issues for a session that does not use TOGETHER have been
identified.
If a renegotiation mechanism that uses SDP fragments, rather than
whole SDP descriptors, is developed, there might be a need for
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specifying that all the fragments included in the TOGETHER group are
sent together. This will have to wait for the development of such a
mechanism.
B.3. ICE negotiation sequence
When an SDP agent sends out an offer using TOGETHER and ICE, it must
choose one of two strategies:
o List ICE parameters for all sections. This means that candidate
gathering and opening of local ports must happen before the SDP
can be generated, and some of the ports get closed, unused, after
the answer has been received. Since the generation of candidates
may involve negotiation with a remote server such as a TURN
server, this may be an expensive proposition.
o List ICE parameters for only the first section. If the TOGETHER
extension is not understood, this will lead to only the first
section's RTP session being established. It is up to the
application whether or not to consider this to be a call failure.
The choice between these two options is left as a local matter.
The offerer may also choose to send out ICE probes on either one port
or all ports even before the answer comes back, in order to speed up
connection establishment; this too is left as a local matter.
B.4. SDP-inspecting intermediaries
There may exist intermediaries that inspect, but do not modify, the
SDP flow, and take some action based on what they parse there.
Such intermediaries are architecturally unsound in general (if they
really are needed, they should be explicit participants and be able
to refuse to carry the extension they don't understand; if not, their
working is of no concern to the callers), but they still exist.
In the RTCWEB use case, they may have more problems existing, since
the negotiation will normally be carried over encrypted HTTPS
connections. In other cases, where negotiation is done in the clear,
they may be more common.
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Author's Address
Harald Tveit Alvestrand
Google
Kungsbron 2
Stockholm, 11122
Sweden
Email: harald@alvestrand.no
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