| Network Working Group | H. T. Alvestrand |
| Internet-Draft | |
| Intended status: Standards Track | August 11, 2011 |
| Expires: February 12, 2012 |
SDP Grouping for Single RTP Sessions
draft-alvestrand-one-rtp-00
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.
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].
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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.
The requirements for our representation are:
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:
The media sections MAY contain connection data (port numbers or ICE parameters), but some of these may be ignored in processing (see next section).
This extension MAY be included in an Offer; if it is not included in an Offer, it MUST NOT be included in an answer. [Note: I believe this to be true for all extensions within the Grouping framework. Check.]
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:
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 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.
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 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 RFC 3550 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.
The examples are taken from RFC 4317, "SDP Offer/Answer Examples".
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
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
This document requests IANA to register the new SDP Grouping semantic extension called TOGETHER.
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.
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.
| [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. |
| [I-D.ietf-rtcweb-overview] | Alvestrand, H, "Overview: Real Time Protocols for Brower-based Applications", Internet-Draft draft-ietf-rtcweb-overview-02, September 2011. |