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Versions: 00

Network Working Group                                        A. B. Roach
Internet-Draft                                                   Mozilla
Intended status: Informational                                 J. Uberti
Expires: January 16, 2014                                         Google
                                                              M. Thomson
                                                               Microsoft
                                                           July 15, 2013


     A Unified Plan for Using SDP with Large Numbers of Media Flows
                   draft-roach-mmusic-unified-plan-00

Abstract

   A recurrent theme in emerging real-time communications use cases,
   such as RTCWEB, has been the need to handle very large numbers of
   media flows.  Unfortunately, naive uses of SDP do not handle this
   case particularly well.  This document describes a modest set of
   extensions to SDP which allow it to cleanly handle arbitrary numbers
   of flows while still retaining a large degree of backward
   compatibility with existing and non-RTCWEB endpoints.

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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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 January 16, 2014.

Copyright Notice

   Copyright (c) 2013 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
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Design Goals  . . . . . . . . . . . . . . . . . . . . . .   4
       1.1.1.  Support for a large number of arbitrary sources . . .   4
       1.1.2.  Support for fine-grained receiver control of sources    5
       1.1.3.  Glareless addition and removal of sources . . . . . .   5
       1.1.4.  Interworking with other devices . . . . . . . . . . .   5
       1.1.5.  Avoidance of excessive port allocation  . . . . . . .   6
       1.1.6.  Simple binding of MediaStreamTrack to SDP . . . . . .   6
       1.1.7.  Support for RTX, FEC, simulcast, layered coding . . .   6
     1.2.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   6
     1.3.  Syntax Conventions  . . . . . . . . . . . . . . . . . . .   7
   2.  Solution Overview . . . . . . . . . . . . . . . . . . . . . .   7
   3.  Detailed Description  . . . . . . . . . . . . . . . . . . . .   8
     3.1.  Bundle-Only M-Lines . . . . . . . . . . . . . . . . . . .   8
     3.2.  Correlation . . . . . . . . . . . . . . . . . . . . . . .  12
       3.2.1.  Correlating RTP Sources with m-lines  . . . . . . . .  12
         3.2.1.1.  RTP Header Extension Correlation  . . . . . . . .  13
         3.2.1.2.  Payload Type Correlation  . . . . . . . . . . . .  14
       3.2.2.  Correlating Media Stream Tracks with m-lines  . . . .  16
       3.2.3.  Correlating Media Stream Tracks with RTP Sources  . .  16
     3.3.  Handling of Simulcast, Forward Error Correction, and
           Retransmission Streams  . . . . . . . . . . . . . . . . .  16
     3.4.  Glare Minimization  . . . . . . . . . . . . . . . . . . .  18
       3.4.1.  Adding a Stream . . . . . . . . . . . . . . . . . . .  19
       3.4.2.  Changing a Stream . . . . . . . . . . . . . . . . . .  19
       3.4.3.  Removing a Stream . . . . . . . . . . . . . . . . . .  20
     3.5.  Negotiation of Stream Ordinality  . . . . . . . . . . . .  20
     3.6.  Compatibility with Legacy uses  . . . . . . . . . . . . .  22
   4.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  23
     4.1.  Simple example with one audio and one video . . . . . . .  23
     4.2.  Multiple Videos . . . . . . . . . . . . . . . . . . . . .  26
     4.3.  Many Videos . . . . . . . . . . . . . . . . . . . . . . .  28
     4.4.  Multiple Videos with Simulcast  . . . . . . . . . . . . .  30
     4.5.  Video with Simulcast and RTX  . . . . . . . . . . . . . .  31
     4.6.  Video with Simulcast and FEC  . . . . . . . . . . . . . .  32
     4.7.  Video with Layered Coding . . . . . . . . . . . . . . . .  33
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  35
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  35
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  35
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  35



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     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  35
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  36
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  37

1.  Introduction

   A recurrent theme in new RTC technologies has been the need to
   cleanly handle very large numbers of media flows.  For instance, a
   videoconferencing application might have a main display plus
   thumbnails for 10 or more other speakers all displayed at the same
   time.  If each video source is encoded in multiple resolutions (e.g.,
   simulcast or layered coding) and also has FEC or RTX, this could
   easily add up to 30 or more independent RTP flows.

   This document focuses on the WebRTC use cases, and uses its
   terminology to discuss key concepts.  The approach described herein,
   however, is not intended to be WebRTC specific, and should be
   generalize to other SDP-using applications.

   The standard way of encoding this information in SDP is to have each
   RTP flow (i.e., SSRC) appear on its own m-line.  For instance, the
   SDP for two cameras with audio from a device with a public IP address
   could look something like:

   v=0
   o=- 20518 0 IN IP4 203.0.113.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7

   m=audio 54400 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 52595
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=candidate:0 1 UDP 2113667327 203.0.113.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 203.0.113.1 54401 typ host

   m=video 55400 RTP/SAVPF 96 97
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 56036
   a=rtpmap:96 H264/90000



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   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv
   a=candidate:0 1 UDP 2113667327 203.0.113.1 55400 typ host
   a=candidate:1 2 UDP 2113667326 203.0.113.1 55401 typ host

   m=video 56400 RTP/SAVPF 96 97
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 21909
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv
   a=candidate:0 1 UDP 2113667327 203.0.113.1 56400 typ host
   a=candidate:1 2 UDP 2113667326 203.0.113.1 56401 typ host


   Unfortunately, as the number of independent media sources starts to
   increase, the scaling properties of this approach become problematic.
   In particular, SDP currently requires that each m-line have its own
   transport parameters (port, ICE candidates, etc.), which can get
   expensive.  For instance, the [RFC5245] pacing algorithm requires
   that new STUN transactions be started no more frequently than 20 ms;
   with 30 RTP flows, which would add 600 ms of latency for candidate
   gathering alone.  Moreover, having 30 persistent flows might lead to
   excessive consumption of NAT binding resources.

   This document specifies a small number of modest extensions to SDP
   which are intended to reduce the transport impact of using a large
   number of flows.  The general design philosophy is to maintain the
   existing SDP negotiation model (inventing as few new mechanisms as
   possible) while simply reducing the consumption of network resources.

1.1.  Design Goals

   The mechanism described in this document is meant to address the
   following goals:

1.1.1.  Support for a large number of arbitrary sources

   In cases such as a video conference, there may be dozens or hundreds
   of participants, each with their own audio and video sources.  A
   participant may even want to browse conferences before joining one,
   meaning that there may be cases where there are many such conferences
   displayed simultaneously.

   In these conferences, participants may have varying capabilities and
   therefore video resolutions.  In addition, depending on conference



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   policy, user preference, and the desired UI, participants may be
   displayed in various layouts, including:

   o  A single large main speaker with thumbnails for other participants

   o  Multiple medium-sized main speakers, with or without thumbnails

   o  Large slides + medium speaker, without thumbnails

   These layouts can change dynamically, depending on the conference
   content and the preferences of the receiver.  As such, there are not
   well-defined 'roles', that could be used to group sources into
   specific 'large' or 'thumbnail' categories.  As such, the requirement
   we attempt to satisfy is support for sending and receiving up to
   hundreds of simultaneous, heterogeneous sources.

1.1.2.  Support for fine-grained receiver control of sources

   Since there may be large numbers of sources, which can be displayed
   in different layouts, it is imperative that the receiver can easily
   control which sources are received, and what resolution or quality is
   desired for each (for both audio and video).  The receiver should
   also be able to prioritize the source it requests, so that if system
   limits or bandwidth force a reduction in quality, the sources chosen
   by the receiver as important will receive the best quality.  These
   details must be exposed to the application via the API.

1.1.3.  Glareless addition and removal of sources

   Sources may come and go frequently, as is the case in a conference
   where various participants are presenting, or an interaction between
   multiple distributed conference servers.  Because of this, it is
   desirable that sources can be added to SDP in a way that avoids
   signaling glare.

1.1.4.  Interworking with other devices

   When interacting with devices that do not apply all of the techniques
   described in this document, it must be possible to degrade gracefully
   to a usable basic experience.  At a minimum, this basic experience
   should support setting up one audio stream and more than one video
   stream with existing videoconferencing equipment designed to
   establish a small number of simultaneous audio and video flows.  For
   the remainder of this document, we will call these devices "legacy
   devices," although it should be understood that statements about
   legacy devices apply equally to future devices that elect not to use
   the techniques described in this document.




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1.1.5.  Avoidance of excessive port allocation

   When there are dozens or hundreds of streams, it is desirable to
   avoid creating dozens or hundreds of transports, as empirical data
   shows a clear inverse relationship between number of transports (NAT
   bindings) and call success rate.  While BUNDLE helps avoid creating
   large numbers of transports, it is also desirable to avoid creating
   large numbers of ports during call setup.

1.1.6.  Simple binding of MediaStreamTrack to SDP

   In WebRTC, each media source is identified by a MediaStreamTrack
   object.  In order to ensure that the MSTs created by the sender show
   up at the receiver, each MST's id attribute needs to be reflected in
   SDP.

1.1.7.  Support for RTX, FEC, simulcast, layered coding

   For robust applications, techniques like RTX and FEC are used to
   protect media, and simulcast/layered coding can be used to provide
   support to heterogeneous receivers.  It needs to be possible to
   support these techniques, allow the recipient to optionally use or
   not use them on a source-by-source basis; and for simulcast/layered
   scenarios, to control which simulcast streams or layers are received.

1.2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHOULD", "SHOULD NOT",
   "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be
   interpreted as described in [RFC2119].

   This draft uses the API and terminology described in [webrtc-api].

   5-tuple: A collection of the following values: source IP address,
   source transport port, destination IP address, destination transport
   port and transport protocol.

   Transport-Flow: An transport 5 Tuple representing the UDP source and
   destination IP address and port over which RTP is flowing.

   m-line: An SDP [RFC4566] media description identifier that starts
   with an "m=" field and conveys the following values: media type,
   transport port, transport protocol and media format descriptions.

   Offer: An [RFC3264] SDP message generated by the participant who
   wishes to initiate a multimedia communication session.  An Offer
   describes the participant's capabilities for engaging in a multimedia
   session.



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   Answer: An [RFC3264] SDP message generated by the participant in
   response to an Offer.  An Answer describes the participant's
   capabilities in continuing with the multimedia session with in the
   constraints of the Offer.

   This draft avoids using terms that implementors do not have a clear
   idea of exactly what they are - for example RTP Session.

1.3.  Syntax Conventions

   The SDP examples given in this document deviate from actual on-the-
   wire SDP notation in several ways.  This is done to facilitate
   readability and to conform to the restrictions imposed by the RFC
   formatting rules.  These deviations are as follows:

   o  Any line that is indented (compared to the initial line in the SDP
      block) is a continuation of the preceding line.  The line break
      and indent are to be interpreted as a single space character.

   o  Empty lines in any SDP example are inserted to make functional
      divisions in the SDP clearer, and are not actually part of the SDP
      syntax.

   o  Excepting the above two conventions, line endings are to be
      interpreted as <CR><LF> pairs (that is, an ASCII 13 followed by an
      ASCII 10).

   o  Any text starting with the string "//" to the end of the line is
      inserted for the benefit of the reader, and is not actually part
      of the SDP syntax.

2.  Solution Overview

   At a high level, the solution described in this document can be
   summarized as follows:

   1.  Each media stream track is represented by its own unique m-line.
       This is a strict one-to-one mapping; a single media stream track
       cannot be spread across several m-lines, nor may a single m-line
       represent multiple media stream tracks.  Note that this requires
       a modification to the way simulcast is currently defined by the
       individual draft [I-D.westerlund-avtcore-rtp-simulcast].  This
       does not preclude "application level" simulcasting; i.e., the
       creation of multiple media stream tracks from a single source.

   2.  Each m-line is marked with an a=ssrc attribute to correlate it
       with its RTP packets.  Absent any other signaled extension,
       multiple SSRCs in a single m-line are interpreted as alternate



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       sources for the same media stream track: although senders can
       switch between the SSRCs as frequently as desired, only one
       should be sent at any given time.

   3.  Each m-line contains an MSID value to correlate it with a Media
       Stream ID and the Media Stream Track ID.

   4.  To minimize port allocation during a call, we rely on the BUNDLE
       [I-D.ietf-mmusic-sdp-bundle-negotiation] mechanism.

   5.  To reduce port allocation during call set-up, applications can
       mark less-critical media stream tracks in such a way that they
       will not require any port allocation, with the resulting property
       that such streams only work in the presence of the BUNDLE
       mechanism.

   6.  To address glare, we define a procedure via which partial offer/
       answer exchanges may take place.  These exchanges operate on a
       single m-line at a time, rather than an entire SDP body.  These
       operations are defined in a way that can completely avoid glare
       for stream additions and removals, and which reduces the chance
       of glare for changes to active streams.  This approach requires
       all m-lines to contain an a=mid attribute.

   7.  All sources in a single bundle are required to contain identical
       attributes except for those that apply directly to a media stream
       track (such as label, msid, and resolution).  See those
       attributes marked "IDENTICAL" in
       [I-D.nandakumar-mmusic-sdp-mux-attributes] for details.

   8.  RTP and RTCP streams are demultiplexed strictly based on their
       SSRC.  However, to handle legacy cases and signaling/media races,
       correlation of streams to m-sections can use other mechanisms, as
       described in Section 3.2.

3.  Detailed Description

3.1.  Bundle-Only M-Lines

   Even with the use of BUNDLE, it is expensive to allocate ICE
   candidates for a large number of m-lines.  An offer can contain
   "bundle-only" m-lines which will be negotiated only by endpoints
   which implement this specification and ignored by other endpoints.

      OPEN ISSUE: While it's probably pretty clear that this behavior
      will be controlled, in WebRTC, via a constraint, the "default"
      behavior -- that is, whether a line is "bundle-only" when there is
      no constraint present -- needs to be settled.  This is a balancing



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      act between maximizing interoperation with legacy equipment by
      default or minimizing port use during call setup by default.

   In order to offer such an m-line, the offerer does two things:

   o  Sets the port in the m-line to 0.  This indicates to old endpoints
      that the m-line is not to be negotiated.

   o  Adds an a=bundle-only line.  This indicates to new endpoints that
      the m-line is to be negotiated if (and only if) bundling is used.

   An example offer that uses this feature looks like this:

   v=0
   o=- 20518 0 IN IP4 203.0.113.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=group:BUNDLE S1 S2 S3
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7

   m=audio 54400 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 20970
   a=mid:1
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=ssrc:53280
   a=candidate:0 1 UDP 2113667327 203.0.113.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 203.0.113.1 54401 typ host

   m=video 0 RTP/SAVPF 96 97
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 1714
   a=mid:2
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=ssrc:49152
   a=bundle-only



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   m=video 0 RTP/SAVPF 96 97
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 57067
   a=mid:3
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=ssrc:32768
   a=bundle-only


   An old endpoint simply rejects the bundle-only m-lines by responding
   with a 0 port.  (This isn't a normative statement, just a description
   of the way the older endpoints are expected to act.)

   v=0
   o=- 20518 0 IN IP4 203.0.113.1
   s=
   t=0 0
   c=IN IP4 203.0.113.2
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7

   m=audio 55400 RTP/SAVPF 0 96
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=candidate:0 1 UDP 2113667327 203.0.113.2 55400 typ host
   a=candidate:1 2 UDP 2113667326 203.0.113.2 55401 typ host

   m=video 0 RTP/SAVPF 96 97
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv

   m=video 0 RTP/SAVPF 96 97
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv





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   A new endpoint accepts the m-lines (both bundle-only and regular) by
   offering m-lines with a valid port, though this port may be
   duplicated as specified in Section 6 of
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  For instance:

   v=0
   o=- 20518 0 IN IP4 203.0.113.2
   s=
   t=0 0
   c=IN IP4 203.0.113.2
   a=group:BUNDLE B1 B2 B3
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7

   m=audio 55400 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 24860
   a=mid:1
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=ssrc:35987
   a=candidate:0 1 UDP 2113667327 203.0.113.2 55400 typ host

   m=video 55400 RTP/SAVPF 96 97
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 49811
   a=mid:B2
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=ssrc:9587
   a=bundle-only

   m=video 55400 RTP/SAVPF 96 97
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 9307
   a=mid:3
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:97 VP8/90000
   a=sendrecv



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   a=rtcp-mux
   a=ssrc:21389
   a=bundle-only


   Endpoints MUST NOT accept bundle-only m-lines if they are not part of
   an accepted bundle group.

3.2.  Correlation

   The system under consideration has three constructs the need to be
   mutually correlated for proper functioning: m-lines, media stream
   tracks, and RTP sources.  These correlations are described in the
   following sections.

3.2.1.  Correlating RTP Sources with m-lines

   Sending several media streams over a single transport 5-tuple can
   pose challenges in the form of stream identification and correlation.
   This proposal maintains the use of SSRC as the single demultiplexing
   point for multiple streams sent between a transport 5-tuple.

   Nominally, this correlation is performed by including a=ssrc
   attributes in the SDP.  Under ideal circumstances, the use of a=ssrc
   in the SDP exchanged between endpoints is sufficient to correlate a
   demultiplexed stream to its m-line.  However, at least three
   unrelated situations can arise that make correlation using an
   alternate mechanism advantageous.

   During call establishment, circumstances may arise under which an
   endpoint can send an offer for a new stream, and begin receiving that
   media stream prior to receiving the SDP that correlates its SSRC to
   the m-line.  For such cases, the endpoint will not know how to handle
   the media, and will most probably be forced to discard it.  This can
   lead to media stream "clipping," which has a strongly negative impact
   on user experience.  For audio streams, an the "hello" of the
   answering party can be lost; for video streams, the initial I-frame
   can be lost, leading to corrupted or missing video until another
   I-frame is sent.

   In the rare circumstance that a SSRC change for an existing media
   source is required, then any party that has changed its SSRC needs to
   inform the remote participants of the updated mapping, e.g.  via a
   new SDP offer.  Since any media sent with the new SSRC cannot be
   rendered until the new offer/answer exchange takes place, the
   clipping concern mentioned above exists here as well.





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   A different problem can arise when interoperating with legacy
   equipment.  A number of circumstances can lead to the inability of a
   legacy endpoint to include SSRC information in its SDP.  For example,
   in a system that decomposes signaling and media into different
   network devices, the protocol used to communicate between the boxes
   frequently will not include SSRC information, making it impossible to
   include in the SDP.  If these devices choose to implement bundling,
   correlation of media streams to m-lines requires an alternate
   correlator.

   These cases (and possibly other similar situations) can be
   ameliorated by using information in the media stream itself as a
   correlator to the SDP offer.  If a packet arrives with an SSRC that
   is not yet associated with an m-line, we would ideally have some
   means of correlating it prior to the arrival of the answer.

   The authors reiterate and emphasize that this technique is used
   solely for the purposes of correlation of an RTP stream to an SDP
   m-line after that stream has already been demultiplexed.
   Demultiplexing of multiple streams on a single transport address
   continues to be based on SSRC values.

3.2.1.1.  RTP Header Extension Correlation

   The preferred mechanism for such correlation is a new RTP header
   extension [RFC5285] that can be used near the beginning of an RTP
   stream to correlate RTP packets for which SSRC mapping information is
   not available.  We propose that WebRTC implementations MUST implement
   this mechanism.  We expect and that all other users of the BUNDLE
   extension SHOULD make use of it.

   Although additional specification for this mechanism would be
   required for interoperability, the thumbnail sketch of such
   correlation is described below.

   An implementation making use of this mechanism for local correlation
   includes an a=extmap attribute in the m-lines for which it wishes to
   use the mechanism.  This attribute includes a mapping from the RTP
   header ID to the URL, as well as a 16-bit identifier (expressed as an
   integer) used for correlation; one such m-line would look like this:

   m=audio 55400 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 7582  // NEW
   a=mid:1
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20



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   a=sendrecv
   a=rtcp-mux
   a=ssrc:35987
   a=candidate:0 1 UDP 2113667327 203.0.113.2 55400 typ host


   The remote endpoint, if it supports this extension, MUST include an
   RTP header extension in several (on the order of 3 to 10) of the
   initial RTP packets in the stream.  The value of this header
   extension will contain the correlator from the extmap line (in the
   above example, 7582).

3.2.1.2.  Payload Type Correlation

   To support implementations that cannot implement the RTP header
   extension described in Section 3.2.1.1 but which wish to use the
   BUNDLE mechanism, we allow an alternate (but less-preferred) means of
   correlation using payload type.  This approach takes advantage of the
   fact that the offer contains payload types chosen by its creator,
   which will be present in any RTP received from the remote party.  If
   these payload types are unique, then they can be used to reliably
   correlate incoming RTP streams to their m= lines.

   Because of its inherent limitations, it is advisable to use other
   correlation techniques than PT multiplexing if at all possible.  In
   order to accomplish this, we propose, for WebRTC, that use of this
   technique be controlled by an additional constraint passed to
   createOffer by the Web application.

   If this constraint is set, the browser MUST behave as described in
   this section.  If the constraint is not set, the browser MUST use
   identical PTs for the same codec values within each m-line bundle.

   When such a constraint is present, implementations attempt to
   entirely exhaust the dynamic payload type numbering space before re-
   using a payload type within the scope of a local transport address.
   If such a constraint is present and the payload type space would
   ordinarily be exhausted within the scope of a local transport
   address, the implementation MAY (at its discretion) take any of the
   following actions:

   1.  Bind to multiple local transport addresses (using different
       BUNDLE groups) for the purpose of keeping the {payload type,
       transport address} combination unique.

   2.  Signal a failure to the application.





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      OPEN ISSUE: The above text specifically calls out "dynamic payload
      type numbering space," which consists of payload types 96 through
      127.  This is the most conservative range of payload types
      possible, with the greatest chance of exhaustion in normal use.
      In practice, it may make more sense to use a different range.  The
      canonical description of payload type allocation strategies for
      RTP/AVP and its related profiles is given in section 3 of
      [RFC3551].  Roughly summarized: all values from 0 to 127 can be
      dynamically bound to codecs; codes from 96 to 127 should be
      preferred, followed by previously unassigned values, followed by
      statically assigned values.  This is, however, modified by
      [RFC5761], which effectively eliminates payload types 64 through
      95.  Given these constraints, reasonable proposals (in order of
      most conservative to most aggressive) would include:

      1.  The dynamic range (96-127), for 32 usable payload types.  This
          is meant to accommodate the most naive implementation
          possible, which is only capable of dynamically binding payload
          types in the dynamic range.  Although not supported by current
          specifications, such limitations are suspected to exist in
          some modern RTP libraries.

      2.  The dynamic range (96-127), followed by the contiguous
          unassigned range (35-63), for 61 usable payload types.  This
          approach is intended to accommodate those implementations that
          do not support dynamic binding for payload types for which an
          "audio/video" type is registered in the IANA registry.

      3.  The dynamic range (96-127) followed by all unassigned payload
          types (20-24, 27, 29, 30, and 35-63), for 69 usable payload
          types.  This approach is intended to accommodate those
          implementations that are incapable of re-binding statically
          assigned payload types, while making use of all other
          available values.

      4.  The dynamic range (96-127) followed by all unassigned payload
          types (20-24, 27, 29, 30, and 35-63), followed by the
          statically assigned payload types (0-19, 25, 26, 28, and
          31-34) for 96 usable payload types.  This approach is most
          consistent with current IETF specifications, but is expected
          to cause interoperability issues with existing implementations
          (including libraries currently in use in early WebRTC
          implementations).

   Note that the presence or absence of the aforementioned flag does not
   affect how incoming streams are correlated: if the RTP header
   extension for correlation is present, it is used in preference to the
   payload type.  Conversely, if the flag is absent, and the RTP



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   contains no such header, then the payload type may be used for
   correlation inasmuch as a media line can be unambiguously identified.
   Of course, if the SSRC information has been made available in SDP
   prior to a need for stream correlation, then it can also be used for
   this purpose.

3.2.2.  Correlating Media Stream Tracks with m-lines

   Media Stream Tracks IDs are correlated with M-Lines directly by
   including an MSID in each m-line.  The MSID also provides the Media
   Stream ID.  (Note the format of the MSID used here is slightly
   different than what was proposed in the current MSID draft as that
   draft assumed multiple tracks in a single m-line and this proposal
   moves to a solution where there is a one to one relation between the
   Track and MSID.  This work assumes the MSID draft will be updated to
   match the syntax used user which simply provides the value of the
   MediaStream ID and MediaStreamTrack ID on an "a=msid" line.  )

3.2.3.  Correlating Media Stream Tracks with RTP Sources

   Media Stream Tracks are correlated with RTP sources transitively
   through the RTP-Source <=> M-Line <=> Media-Stream-Track
   relationship.  Since the Media-Stream-Track <=> M-Line binding is
   established in the SDP offer, and the M-Line <=> RTP-Source binding
   can be handled as described in Section 3.2.1, none of the previously
   identified issues arise.

3.3.  Handling of Simulcast, Forward Error Correction, and
      Retransmission Streams

   Simulcast refers to taking a single capture (e.g., a camera), and
   encoding it multiple times at different resolutions and / or frame
   rates.  For example, a device with a single HD camera may send one
   version of the video at full HD resolution, and a second version
   encoded at a low resolution.  This would allow a video conferencing
   bridge to be able to send the high resolution copy to some
   destination and low resolution copy to other destinations without
   having to recode the video at the conference bridge.

   Forward Error Correction (FEC) and Retransmission (RTX) streams are
   techniques that can provide stream robustness in the face of packet
   loss.  These approaches frequently make use of different payload
   types and different SSRC values than the stream to which they apply.

   In cases where a media source needs to correspond to more than one
   RTP flow, e.g.  RTX, FEC, or simulcast, the a=ssrc-group [RFC5576]
   concept is used to create a grouping of SSRCs for a single media
   stream track.  Each SSRC is declared using a=ssrc attributes, the



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   same MSID is shared between the SSRCs, and the a=ssrc-group attribute
   defines the behavior of the grouped SSRCs.

   These groupings are used to perform demux of the incoming RTP streams
   and associate them (by SSRC) with their primary flows (modulo the
   behavior described in Section 3.2.1, if applicable).  This
   multiplexing of RTX and FEC in a single RTP session is already well-
   defined; RTX SSRC-multiplexing behavior is defined in [RFC4588], and
   FEC SSRC-multiplexing behavior is defined in [RFC5956].

   Note that both RTC and FEC also include SDP expressions that use
   different m= lines for the correction streams (cf.  [RFC4588],
   section 8.7 and [RFC5956], section 4.2).  These formats intend for
   correlation of streams to be based on transport addresses, which is
   inapplicable for bundled media streams.  Our specific proposal is:
   (1) bundling implementations will never generate such a format; and
   (2) bundling implementations MAY choose to accept SDP in such a
   format or MAY simply reject the repair streams and proceed as if the
   indicated repair format is not supported.

   For multi-resolution simulcast, we can create a similar ssrc-group,
   and adapt the imageattr attribute defined in [RFC6236] for the a=ssrc
   line attribute to indicate the send resolution for a given simulcast
   stream.  (This will be added to
   [I-D.westerlund-avtcore-rtp-simulcast], as outlined in Section 2,
   bullet 1).  In the example below, the SDP advertises a simulcast of a
   camera source at two different resolutions, as well as a screen-share
   source that supports RTX; a=ssrc-group is used to correlate the
   different SSRCs as part of a single media source.

   Note that a characteristic of this approach is that it does not allow
   for independently setting attributes for simulcast, FEC, and RTX
   streams aside from those in fmtp.  In particular, attributes such as
   ptime and framerate are shared between the streams that are grouped
   together for a simulcast group.

   m=video 62537 RTP/SAVPF 96       // main video
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 15955
   a=mid:1
   a=rtpmap:96 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=ssrc:29154 imageattr:96 [x=1280,y=720]
   a=ssrc:47182 imageattr:96 [x=640,y=360]
   a=ssrc-group:SIMULCAST 29154 47182

   m=video 0 RTP/SAVPF 96 97          // slide video



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   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 26267
   a=mid:2
   a=rtpmap:96 VP8/90000
   a=rtpmap:97 rtx/90000
   a=sendrecv
   a=rtcp-mux
   a=fmtp:97 apt=96;rtx-time=3000
   a=ssrc:45982
   a=ssrc:9827
   a=ssrc-group:FID 45982 9827        // FID provides SSRC correlation
   a=bundle-only


   Providing explicit resolutions on a per-SSRC basis for SIMULCAST
   groupings allows an intermediary (such as a Media Translator
   [RFC5117]) to be able to select an appropriate SIMULCAST layer
   without inspecting the media stream, which could otherwise require
   decrypting and possibly partially decoding media packets.

3.4.  Glare Minimization

   To allow for guaranteed glareless addition and removal of streams,
   and to provide for a reduced chance of glare in stream attribute
   changes, we propose a technique that allows for m-lines to be changed
   independently of each other.

   The proposal for doing so is performed using "partial offers" and
   "partial answers."  Using this technique has two key prerequisites:
   (1) all offer/answer exchanges in the session have contained "a=mid"
   attributes [RFC5888] for each m-line, and (2) both sides are known to
   support the partial offer/answer technique (either because they are
   part of a single domain of control, or because use of this technique
   has been explicitly signaled).

   The use of a partial SDP body will be explicitly signaled, e.g.,
   using a different MIME type for SIP, or using a different "type" for
   the WebRTC API.

   The authors recognize that further formal definition would be
   required to describe this technique.  These are left as future study
   for the appropriate venues, such as the W3C WebRTC WG and the SIPCORE
   WG.  As a thumbnail sketch: For WebRTC, we envision that we would add
   a new constraint to createOffer, requesting that a partial offer be
   generated (if possible).  The resulting RTCSessionDescription would
   contain only the m-lines that have changed since the most recent
   offer/answer exchange, and would have a type of "partialOffer."  When
   createAnswer is called after receipt of a partialOffer, it would



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   create a partialAnswer, containing only the m-lines referenced in the
   partial offer, that can be provided to the remote party.

3.4.1.  Adding a Stream

   To add a stream glarelessly, a party creates a "partial offer"
   consisting of an m-line and all of its attributes.  This m-line
   contains an mid that has not yet been used in the session.  To reduce
   the chance of collision to effectively zero, this mid MUST contain at
   least 32 characters chosen randomly from full set of 79 characters
   allowed in a token.  It then sends this partial offer to the remote
   party and awaits a partial answer.

   Upon receipt of a partial offer, an implementation examines the mid
   in it.  If the mid does not match any existing mid in the session,
   then it represents a new media stream.  Assuming the recipient does
   not have an outstanding, unanswered partial offer that also adds a
   stream, this new m-line is simply appended to the end of the existing
   session description, the SDP version is incremented by one, and a
   partial answer is created.  This partial answer consists of an m-line
   and its attributes, and has an mid matching the one from the partial
   offer.

   If the recipient of a partial offer that contains a new mid has also
   sent a partial offer adding a new stream to the session, then
   ambiguity can arise regarding the canonical ordering of m-lines
   within the session.  In this situation, both partial offer/answer
   exchanges are allowed to complete independently (as no fundamental
   data glare has occurred).  However, the order in which they are
   appended to the session description is synchronized by performing a
   lexical comparison between each m-lines mid attribute: the m-line
   with the lexically smaller mid attribute is appended first, while the
   other m-line is appended after it.

3.4.2.  Changing a Stream

   Partial offers may also be generated for modification of an existing
   stream.  In this case, the mid in the partial offer will match an
   existing mid in the session description.












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   Upon receipt of a partial offer, an implementation examines the mid
   in it.  If the mid matches any existing mid in the session, then it
   represents a modification to that m-line.  Assuming the recipient
   does not have an outstanding, unanswered partial offer that also
   modifies that exact same stream, this m-line is treated as an
   independent renegotiation of that stream (only).  The SDP version is
   incremented by one, and a partial answer is created.  This partial
   answer consists of an m-line and its attributes, and has an mid
   matching the one from the partial offer.

   If the recipient of a partial offer that contains an existing mid has
   also sent a partial offer to change that exact same stream, and
   neither the received nor the sent partial offer contains an
   "a=inactive" attribute, then a legitimate glare condition has arisen.
   Normal glare recovery procedures -- e.g., using a tie-breaker token
   or a back-off timer -- must be engaged to resolve the conflict.

3.4.3.  Removing a Stream

   To remove a stream in a way that eliminates the chance of glare, an
   implementation generates a new partial offer, with an mid matching
   the m-line it wants to remove.  This partial offer contains an
   a=inactive attribute, indicating that the stream is being
   deactivated.

   If the recipient of a partial offer that contains an existing mid has
   also sent a partial offer to change that exact same stream, and
   either one of the received or the sent partial offer contains an
   "a=inactive" attribute, then a the stream is deactivated.  At this
   point, both partial offers are discarded, the corresponding m-line in
   the session is modified by changing any a=sendonly, a=recvonly, or
   a=sendrecv attribute to a=inactive (or, if no such attributes are
   present, an a=inactive attribute is added), and a partial answer is
   generated representing this single change.

3.5.  Negotiation of Stream Ordinality

   Within advanced applications, circumstances can easily arise in which
   the party creating the offer does not know ahead of time the number
   of streams the remote party will desire.  For example, in a meet-me
   videoconference application that sends a separate stream for each
   participant, a client creating an offer to send to the conference
   focus does not necessarily know how many video streams to indicate in
   its SDP.  Although this can be potentially be solved in an
   application-specific way (e.g., by always offering the maximum number
   of streams known to be supported by the application), this is not
   always desirable or even possible.




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   To address this situation, a three-way handshake can be employed.


                Calling Party                    Called Party
                      |                                |
     Calling party    |--- Offer (1 video, 1 audio) -->|
     creates offer    |                                |
     with audio and   |                                |
     video. Since     |                                |
     it does not      |                                |
     know how many    |                                |
     streams, it      |                                |
     "guesses" one    |                                |
     of each.         |                                |
                      |                                |
                      |<-- Answer (1 video, 1 audio) --|  Called party
   [Call starts now]  |<-- Offer (8 video, 1 audio) ---|  desires eight
                      |                                |  video streams.
                      |                                |  So it creates
                      |                                |  an answer for
                      |                                |  the "one of
                      |                                |  each" offer
                      |                                |  and an offer
                      |                                |  for the total
                      |                                |  number of
                      |                                |  streams it
                      |                                |  wants.
                      |                                |
     Calling party    |-- Answer (8 video, 1 audio) -->|
     answers for      |                                |
     all eight        |                                |
     video streams.   |                                |
                      |                                |


   The first leg of this handshake consists of an offer sent by the
   calling party.  This offer contains at least one m-line for each type
   of media the offerer wishes to use in the session.  The authors draw
   special attention to the clause "at least" in the preceding sentence:
   offerers can use external knowledge, hinting, or simple guesses to
   offer additional m-lines.

   Upon receipt of such an offer, the called party examines the number
   of streams of each media type being requested.  If the number of
   streams is equal to or greater than the number of total streams that
   the called party desires at this time, it simply forms an answer to
   complete the offer/answer exchange [RFC3264], and the call is set up.




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   On the other hand, if the called party determines that more streams
   are necessary than are indicated in the initial offer, it responds by
   first creating an answer with the same number of streams as were
   present in the initial offer.  It additionally creates a new answer
   that contains the number of streams it desires.  This answer/offer
   pair is sent to the calling party, in a single message if supported
   by the signaling protocol (as will frequently be the case for
   WebRTC), or in two consecutive messages in a way that guarantees in-
   order delivery.

   When the calling party receives this answer, it establishes the
   session, and all of the streams that were negotiated in this first
   offer/answer exchange.  So, within a single signaling round trip, the
   initial set of streams (consisting of those the calling party
   included in its initial offer) are established.

   When the calling party receives the subsequent offer, it comprises
   the beginning of a completely new RFC 3264 offer/answer exchange
   [RFC3264].  The calling party creates an answer that fully describes
   all of the streams in the session, and sends it to the called party.
   Consequently, within 1.5 round trips, the entire call is set up and
   all associated streams can be sent and received.

   Of particular note is the fact that this model does not deviate from
   normal RFC3264 offer/answer handling, even when three-way handshaking
   is necessary.

3.6.  Compatibility with Legacy uses

   Due to the fact that this approach re-uses existing SDP constructs
   for indicating parameters in a media section, it remains compatible
   with legacy clients.  Of particular note is the handling of "bundle-
   only" media sections, described in Section 3.1.  Offers generated by
   an RTCWEB client and sent to a legacy client will simply negotiate
   those media the RTCWEB client did not use the "bundle-only" extension
   with.  This allows RTCWEB clients to select which media streams are
   important for interoperability with legacy clients (by not making
   them bundle-only), and which ones are not.  Offers generated by
   legacy clients will simply omit any bundle-related attributes, and
   the RTCWEB client will be able to process the SDP otherwise
   identically to the SDP received from RTCWEB clients: each m-line
   represents a different media stream, and contains a description of
   that stream in a syntax identical to the syntax used between RTCWEB
   clients.

   With the bundle-only approach, only those streams that are "important
   for interoperability" will require allocation of ports and ICE
   exchanges.  By doing so, working with non-multiplexing clients is



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   enabled without requiring excess resource allocation for those
   streams that are not critical for proper user experience.

4.  Examples

   In all of these examples, there are many lines that are wrapped due
   to column width limitation.  It should be understood these lines are
   not wrapped in the real SDP.

   The convention used for IP addresses in this drafts is that private
   IP behind a NAT come from 192.0.2.0/24, the public side of a NAT
   comes from 198.51.100.0/24 and the TURN servers have addresses from
   203.0.113.0/24.  Typically the offer has an IP ending in .1 and the
   answer has an IP ending in .2.

   The examples do not include all the parts of SDP that are used in
   RTCWeb (See [I-D.ietf-rtcweb-rtp-usage]) as that makes the example
   unwieldy to read but instead focuses on showing the parts that are
   key for the multiplexing.

4.1.  Simple example with one audio and one video

   The following SDP shows an offer that offers one audio stream and one
   video steam with both a STUN and TURN address.  It also shows unique
   payload across the audio and video m=lines for the Answerer that does
   not support BUNDLE semantics.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:074c6550
   a=ice-pwd:a28a397a4c3f31747d1ee3474af08a068
   a=fingerprint:sha-1
           99:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:70:9d:1f:66:79:a8:07
   a=group:BUNDLE m1 m2

   m=audio 56600 RTP/SAVPF 0 109
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 33424
   a=mid:m1
   a=ssrc:53280
   a=rtpmap:0 PCMU/8000
   a=rtpmap:109 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux



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   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 56602 RTP/SAVPF 99 120
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 35969
   a=mid:m2
   a=ssrc:49843
   a=rtpmap:99 H264/90000
   a=fmtp:99 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:120 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=candidate:3 1 UDP 2113667327 192.0.2.1 54402 typ host
   a=candidate:4 2 UDP 2113667326 192.0.2.1 54403 typ host
   a=candidate:3 1 UDP 694302207 198.51.100.1 55502 typ srflx raddr
           192.0.2.1 rport 54402
   a=candidate:4 2 UDP 169430220 198.51.100.1 55503 typ srflx raddr
           192.0.2.1 rport 54403
   a=candidate:3 1 UDP 73545215 203.0.113.1 56602 typ relay raddr
           192.0.2.1 rport 54402
   a=candidate:4 2 UDP 51989708 203.0.113.1 56603 typ relay raddr
           192.0.2.1 rport 54403


   The following shows an answer to the above offer from a device that
   does not support bundle or rtcp-mux.

   v=0
   o=- 16833 0 IN IP4 198.51.100.2
   s=
   t=0 0
   c=IN IP4 203.0.113.2
   a=ice-ufrag:c300d85b
   a=ice-pwd:de4e99bd291c325921d5d47efbabd9a2
   a=fingerprint:sha-1
           91:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:70:9d:1f:66:79:a8:03

   m=audio 60600 RTP/SAVPF 109
   a=msid:ma ta



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   a=rtpmap:109 opus/48000
   a=ptime:20
   a=sendrecv
   a=candidate:0 1 UDP 2113667327 192.0.2.2 60400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.2 60401 typ host
   a=candidate:0 1 UDP 1694302207 198.51.100.2 60500 typ srflx raddr
           192.0.2.2 rport 60400
   a=candidate:1 2 UDP 1694302206 198.51.100.2 60501 typ srflx raddr
           192.0.2.2 rport 60401
   a=candidate:0 1 UDP 73545215 203.0.113.2 60600 typ relay raddr
           192.0.2.1 rport 60400
   a=candidate:1 2 UDP 51989708 203.0.113.2 60601 typ relay raddr
           192.0.2.1 rport 60401

   m=video 60602 RTP/SAVPF 99
   a=msid:ma tb
   a=rtpmap:99 H264/90000
   a=fmtp:99 profile-level-id=4d0028;packetization-mode=1
   a=sendrecv
   a=candidate:2 1 UDP 2113667327 192.0.2.2 60402 typ host
   a=candidate:3 2 UDP 2113667326 192.0.2.2 60403 typ host
   a=candidate:2 1 UDP 694302207 198.51.100.2 60502 typ srflx raddr
           192.0.2.2 rport 60402
   a=candidate:3 2 UDP 169430220 198.51.100.2 60503 typ srflx raddr
           192.0.2.2 rport 60403
   a=candidate:2 1 UDP 73545215 203.0.113.2 60602 typ relay raddr
           192.0.2.2 rport 60402
   a=candidate:3 2 UDP 51989708 203.0.113.2 60603 typ relay raddr
           192.0.2.2 rport 60403


   The following shows answer to the above offer from a device that does
   support bundle.

   v=0
   o=- 16833 0 IN IP4 198.51.100.2
   s=
   t=0 0
   c=IN IP4 203.0.113.2
   a=ice-ufrag:c300d85b
   a=ice-pwd:de4e99bd291c325921d5d47efbabd9a2
   a=fingerprint:sha-1
           91:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:70:9d:1f:66:79:a8:03
   a=group:BUNDLE m1 m2

   m=audio 60600 RTP/SAVPF 109
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 39829



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   a=mid:m1
   a=ssrc:35856
   a=rtpmap:109 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.2 60400 typ host
   a=candidate:0 1 UDP 1694302207 198.51.100.2 60500 typ srflx raddr
           192.0.2.2 rport 60400
   a=candidate:0 1 UDP 73545215 203.0.113.2  60600 typ relay raddr
           192.0.2.1 rport 60400

   m=video 60600 RTP/SAVPF 99
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 45163
   a=mid:m2
   a=ssrc:2638
   a=rtpmap:99 H264/90000
   a=fmtp:99 profile-level-id=4d0028;packetization-mode=1
   a=sendrecv
   a=rtcp-mux
   a=candidate:3 1 UDP 2113667327 192.0.2.2 60400 typ host
   a=candidate:3 1 UDP 694302207 198.51.100.2 60500 typ srflx raddr
           192.0.2.2 rport 60400
   a=candidate:3 1 UDP 73545215 203.0.113.2  60600 typ relay raddr
           192.0.2.2 rport 60400


4.2.  Multiple Videos

   Simple example showing an offer with one audio stream and two video
   streams.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m1 m2 m3

   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 47434
   a=mid:m1



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   a=ssrc:32385
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 56602 RTP/SAVPF 96 98
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 22705
   a=mid:m2
   a=ssrc:43985
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:98 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=candidate:2 1 UDP 2113667327 192.0.2.1 54402 typ host
   a=candidate:3 2 UDP 2113667326 192.0.2.1 54403 typ host
   a=candidate:2 1 UDP 694302207 198.51.100.1 55502 typ srflx raddr
           192.0.2.1 rport 54402
   a=candidate:3 2 UDP 169430220 198.51.100.1 55503 typ srflx raddr
           192.0.2.1 rport 54403
   a=candidate:2 1 UDP 73545215 203.0.113.1 56602 typ relay raddr
           192.0.2.1 rport 54402
   a=candidate:3 2 UDP 51989708 203.0.113.1 56603 typ relay raddr
           192.0.2.1 rport 54403
   a=ssrc:11111 cname:45:5f:fe:cb:81:e9

   m=video 56604 RTP/SAVPF 96 98
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 64870
   a=mid:m3
   a=ssrc:54269
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:98 VP8/90000
   a=sendrecv



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   a=rtcp-mux
   a=candidate:4 1 UDP 2113667327 192.0.2.1 54404 typ host
   a=candidate:5 2 UDP 2113667326 192.0.2.1 54405 typ host
   a=candidate:4 1 UDP 694302207 198.51.100.1 55504 typ srflx raddr
           192.0.2.1 rport 54404
   a=candidate:5 2 UDP 169430220 198.51.100.1 55505 typ srflx raddr
           192.0.2.1 rport 54405
   a=candidate:4 1 UDP 73545215 203.0.113.1 56604 typ relay raddr
           192.0.2.1 rport 54404
   a=candidate:5 2 UDP 51989708 203.0.113.1 56605 typ relay raddr
           192.0.2.1 rport 54405
   a=ssrc:22222 cname:45:5f:fe:cb:81:e9


4.3.  Many Videos

   This section adds three video streams and one audio.  The video
   streams are sent in such a way that they they are only accepted if
   the far side supports bundle using the "bundle only" approach
   described in Section 3.1.  The video streams also use the same
   payload types so it will not be possible to demux the video streams
   from each other without using the SSRC values.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m0 m1 m2 m3

   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 6614
   a=mid:m0
   a=ssrc:12359
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=ssrc:12359 cname:45:5f:fe:cb:81:e9
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr



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           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 0 RTP/SAVPF 96 98
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 24147
   a=mid:m1
   a=ssrc:26989
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:98 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=bundle-only
   a=ssrc:26989 cname:45:5f:fe:cb:81:e9

   m=video 0 RTP/SAVPF 96 98
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 33989
   a=mid:m2
   a=ssrc:32986
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:98 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=bundle-only
   a=ssrc:32986 cname:45:5f:fe:cb:81:e9

   m=video 0 RTP/SAVPF 96 98
   a=msid:ma td
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 61408
   a=mid:m3
   a=ssrc:46986
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1
   a=rtpmap:98 VP8/90000
   a=sendrecv
   a=rtcp-mux
   a=bundle-only
   a=ssrc:46986 cname:45:5f:fe:cb:81:e9





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4.4.  Multiple Videos with Simulcast

   This section shows an offer with with audio and two video each of
   which can send it two resolutions as described in Section 3.3.  One
   video stream supports VP8, while the other supports H.264.  All the
   video is bundle-only.  Note that the use of different codec-specific
   parameters causes two different payload types to be used.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m0 m1 m2

   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 31727
   a=mid:m0
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 0 RTP/SAVPF 96 100
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 41664
   b=AS:1756
   a=mid:m1
   a=rtpmap:96 VP8/90000
   a=ssrc-group:SIMULCAST 58949 28506
   a=ssrc:58949 imageattr:96 [x=1280,y=720]
   a=ssrc:28506 imageattr:96 [x=640,y=480]



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   a=sendrecv
   a=rtcp-mux
   a=bundle-only

   m=video 0 RTP/SAVPF 96 100
   a=msid:ma tc
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 14460
   b=AS:1756
   a=mid:m2
   a=rtpmap:96 H264/90000
   a=fmtp:96 profile-level-id=4d0028;packetization-mode=1;max-fr=30
   a=rtpmap:100 H264/90000
   a=fmtp:100 profile-level-id=4d0028;packetization-mode=1;max-fr=15
   a=ssrc-group:SIMULCAST 18875 54986
   a=ssrc:18875
   a=ssrc:54986
   a=sendrecv
   a=rtcp-mux
   a=bundle-only


4.5.  Video with Simulcast and RTX

   This section shows an SDP offer that has an audio and a single video
   stream.  The video stream that is simulcast at two resolutions and
   has [RFC4588] style re-transmission flows.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m0 m1

   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 42123
   a=mid:m0
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host



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   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 0 RTP/SAVPF 96 101
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 60725
   b=AS:2500
   a=mid:m1
   a=rtpmap:96 VP8/90000
   a=rtpmap:101 rtx/90000
   a=fmtp:101 apt=96;rtx-time=3000
   a=ssrc-group:SIMULCAST 78909 43567
   a=ssrc-group:FID 78909 56789
   a=ssrc-group:FID 43567 13098
   a=ssrc:78909
   a=ssrc:43567
   a=ssrc:13098
   a=ssrc:56789
   a=sendrecv
   a=rtcp-mux
   a=bundle-only



4.6.  Video with Simulcast and FEC

   This section shows an SDP offer that has an audio and a single video
   stream.  The video stream that is simulcast at two resolutions and
   has [RFC5956] style FEC flows.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m0 m1




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   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 42123
   a=mid:m0
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 0 RTP/SAVPF 96 101
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 60725
   b=AS:2500
   a=mid:m1
   a=rtpmap:96 VP8/90000
   a=rtpmap:101 1d-interleaved-parityfec/90000
   a=fmtp:96 max-fr=30;max-fs=8040
   a=fmtp:101 L=5; D=10; repair-window=200000
   a=ssrc-group:SIMULCAST 56780 34511
   a=ssrc-group:FEC-FR 56780 48675
   a=ssrc-group:FEC-FR 34511 21567
   a=ssrc:56780
   a=ssrc:34511
   a=ssrc:21567
   a=ssrc:48675
   a=sendrecv
   a=rtcp-mux
   a=bundle-only



4.7.  Video with Layered Coding








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   This section shows an SDP offer that has an audio and a single video
   stream.  The video stream that is layered coding at 3 different
   resolutions based on [RFC5583].  The video m=lines shows 3 streams
   with last stream (payload 100) dependent on streams with payload 96
   and 97 for decoding.

   v=0
   o=- 20518 0 IN IP4 198.51.100.1
   s=
   t=0 0
   c=IN IP4 203.0.113.1
   a=ice-ufrag:F7gI
   a=ice-pwd:x9cml/YzichV2+XlhiMu8g
   a=fingerprint:sha-1
           42:89:c5:c6:55:9d:6e:c8:e8:83:55:2a:39:f9:b6:eb:e9:a3:a9:e7
   a=group:BUNDLE m0 m1

   m=audio 56600 RTP/SAVPF 0 96
   a=msid:ma ta
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 42123
   a=mid:m0
   a=rtpmap:0 PCMU/8000
   a=rtpmap:96 opus/48000
   a=ptime:20
   a=sendrecv
   a=rtcp-mux
   a=candidate:0 1 UDP 2113667327 192.0.2.1 54400 typ host
   a=candidate:1 2 UDP 2113667326 192.0.2.1 54401 typ host
   a=candidate:0 1 UDP 694302207 198.51.100.1 55500 typ srflx raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 169430220 198.51.100.1 55501 typ srflx raddr
           192.0.2.1 rport 54401
   a=candidate:0 1 UDP 73545215 203.0.113.1 56600 typ relay raddr
           192.0.2.1 rport 54400
   a=candidate:1 2 UDP 51989708 203.0.113.1 56601 typ relay raddr
           192.0.2.1 rport 54401

   m=video 0 RTP/SAVPF 96 97 100
   a=msid:ma tb
   a=extmap:1 urn:ietf:params:rtp-hdrext:stream-correlator 60725
   b=AS:2500
   a=mid:m1
   a=rtpmap:96 H264/90000
   a=fmtp:96 max-fr=30;max-fs=8040
   a=rtpmap:97 H264/90000
   a=fmtp:97 max-fr=15;max-fs=1200
   a=rtpmap:100 H264-SVC/90000
   a=fmtp:100 max-fr=30;max-fs=8040



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   a=depend:100 lay m1:96,97;
   a=ssrc:48970
   a=ssrc:90898
   a=ssrc:66997
   a=sendrecv
   a=rtcp-mux
   a=bundle-only



5.  Security Considerations

   TBD

6.  IANA Considerations

   TBD

7.  Acknowledgements

   Thanks to Cullen Jennings and Suhas Nandakumar for their assistance
   in generating the SDP examples in this document.

   Some of the material in this document was taken from
   [I-D.jennings-rtcweb-plan].

8.  References

8.1.  Normative References

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Multiplexing Negotiation Using Session Description
              Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp-
              bundle-negotiation-03 (work in progress), February 2013.

   [I-D.jennings-mmusic-media-req]
              Jennings, C., Uberti, J., and E. Rescorla, "Requirements
              from various WG for MMUSIC", draft-jennings-mmusic-media-
              req-00 (work in progress), February 2013.

   [I-D.nandakumar-mmusic-sdp-mux-attributes]
              Nandakumar, S., "A Framework for SDP Attributes when
              Multiplexing", draft-nandakumar-mmusic-sdp-mux-
              attributes-02 (work in progress), April 2013.

   [I-D.westerlund-avtcore-rtp-simulcast]




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              Westerlund, M., Lindqvist, M., and F. Jansson, "Using
              Simulcast in RTP Sessions", draft-westerlund-avtcore-rtp-
              simulcast-02 (work in progress), February 2013.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June
              2002.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

8.2.  Informative References

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-06 (work in progress),
              February 2013.

   [I-D.jennings-rtcweb-plan]
              Jennings, C., "Proposed Plan for Usage of SDP and RTP",
              draft-jennings-rtcweb-plan-01 (work in progress), February
              2013.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              January 2008.



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   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)", RFC
              5583, July 2009.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5956]  Begen, A., "Forward Error Correction Grouping Semantics in
              the Session Description Protocol", RFC 5956, September
              2010.

   [iana.rtp-pt]
              IANA, "RTP Payload types (PT) for standard audio and video
              encodings", July 2013.

              Available at http://www.iana.org/assignments/rtp-
              parameters/rtp-parameters.xhtml#rtp-parameters-1

   [webrtc-api]
              Bergkvist, Burnett, Jennings, Narayanan, , "WebRTC 1.0:
              Real-time Communication Between Browsers", October 2011.

              Available at http://dev.w3.org/2011/webrtc/editor/
              webrtc.html

Authors' Addresses

   Adam Roach
   Mozilla
   Dallas, TX
   US

   Phone: +1 650 903 0800 x863
   Email: adam@nostrum.com









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   Justin Uberti
   Google
   747 6th St. S
   Kirkland, WA  98033
   USA

   Email: justin@uberti.name


   Martin Thomson
   Microsoft
   3210 Porter Drive
   Palo Alto, CA  94304
   US

   Phone: +1 650 353 1925
   Email: martin.thomson@skype.net

































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