Network Working Group                                          B. Burman
Internet-Draft                                             M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: August 6, 2016                                    S. Nandakumar
                                                               M. Zanaty
                                                        February 3, 2016

                Using Simulcast in SDP and RTP Sessions


   In some application scenarios it may be desirable to send multiple
   differently encoded versions of the same media source in different
   RTP streams.  This is called simulcast.  This document describes how
   to accomplish simulcast in RTP and how to signal it in SDP.  The
   described solution uses an RTP/RTCP identification method to identify
   RTP streams belonging to the same media source, and makes an
   extension to SDP to relate those RTP streams as being different
   simulcast formats of that media source.  The SDP extension consists
   of a new media level SDP attribute that expresses capability to send
   and/or receive simulcast RTP streams.

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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   Drafts is at

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   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 August 6, 2016.

Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Definitions . . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.1.  Terminology . . . . . . . . . . . . . . . . . . . . . . .   3
     2.2.  Requirements Language . . . . . . . . . . . . . . . . . .   4
   3.  Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . .   4
     3.1.  Reaching a Diverse Set of Receivers . . . . . . . . . . .   5
     3.2.  Application Specific Media Source Handling  . . . . . . .   6
     3.3.  Receiver Media Source Preferences . . . . . . . . . . . .   7
   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   7
   5.  Overview  . . . . . . . . . . . . . . . . . . . . . . . . . .   8
   6.  Detailed Description  . . . . . . . . . . . . . . . . . . . .   9
     6.1.  Simulcast Attribute . . . . . . . . . . . . . . . . . . .   9
     6.2.  Simulcast Capability  . . . . . . . . . . . . . . . . . .  11
       6.2.1.  Declarative Use . . . . . . . . . . . . . . . . . . .  13
       6.2.2.  Offer/Answer Use  . . . . . . . . . . . . . . . . . .  13
     6.3.  Relating Simulcast Streams  . . . . . . . . . . . . . . .  15
     6.4.  Signaling Examples  . . . . . . . . . . . . . . . . . . .  15
       6.4.1.  Single-Source Client  . . . . . . . . . . . . . . . .  16
       6.4.2.  Multi-Source Client . . . . . . . . . . . . . . . . .  17
   7.  Network Aspects . . . . . . . . . . . . . . . . . . . . . . .  20
     7.1.  Bitrate Adaptation  . . . . . . . . . . . . . . . . . . .  20
   8.  Limitations . . . . . . . . . . . . . . . . . . . . . . . . .  21
     8.1.  Single RTP Session  . . . . . . . . . . . . . . . . . . .  21
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  21
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  21
   11. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  22
   12. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  22
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  22
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  22
     13.2.  Informative References . . . . . . . . . . . . . . . . .  23
   Appendix A.  Changes From Earlier Versions  . . . . . . . . . . .  25
     A.1.  Modifications Between WG Version -03 and  -04 . . . . . .  25
     A.2.  Modifications Between WG Version -02 and  -03 . . . . . .  26
     A.3.  Modifications Between WG Version -01 and  -02 . . . . . .  26
     A.4.  Modifications Between WG Version -00 and  -01 . . . . . .  26
     A.5.  Modifications Between Individual Version -00 and WG

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           Version -00 . . . . . . . . . . . . . . . . . . . . . . .  26
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  26

1.  Introduction

   Most of today's multiparty video conference solutions make use of
   centralized servers to reduce the bandwidth and CPU consumption in
   the endpoints.  Those servers receive RTP streams from each
   participant and send some suitable set of possibly modified RTP
   streams to the rest of the participants, which usually have
   heterogeneous capabilities (screen size, CPU, bandwidth, codec, etc).
   One of the biggest issues is how to perform RTP stream adaptation to
   different participants' constraints with the minimum possible impact
   on both video quality and server performance.

   Simulcast is defined in this memo as the act of simultaneously
   sending multiple different encoded streams of the same media source,
   e.g. the same video source encoded with different video encoder types
   or image resolutions.  This can be done in several ways and for
   different purposes.  This document focuses on the case where it is
   desirable to provide a media source as multiple encoded streams over
   RTP [RFC3550] towards an intermediary so that the intermediary can
   provide the wanted functionality by selecting which RTP stream(s) to
   forward to other participants in the session, and more specifically
   how the identification and grouping of the involved RTP streams are

   This document describes a few scenarios where it is motivated to use
   simulcast, and also defines the needed RTP/RTCP and SDP signaling for

2.  Definitions

2.1.  Terminology

   This document makes use of the terminology defined in RTP Taxonomy
   [RFC7656], and RTP Topologies [RFC7667].  In addition, the following
   terms are used:

   RTP Mixer:  An RTP middle node, defined in [RFC7667] (Section 3.6 to

   RTP Switch:  A common short term for the terms "switching RTP mixer",
      "source projecting middlebox", and "video switching MCU" as
      discussed in [RFC7667].

   Simulcast Stream:  One Encoded Stream or Dependent Stream from a set
      of concurrently transmitted Encoded Streams and optional Dependent

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      Streams, all sharing a common Media Source, as defined in
      [RFC7656].  Decoding a Dependent Stream also requires the related
      (Dependent and) Encoded Stream(s), but in the context of simulcast
      that is considered a property of the Dependent Stream constituting
      the simulcast stream.  For example, HD and thumbnail video
      simulcast versions of a single Media Source sent concurrently as
      separate RTP Streams.

   Simulcast Format:  Different formats of a simulcast stream serve the
      same purpose as alternative RTP payload types in non-simulcast
      SDP, to allow multiple alternative media formats for a given RTP
      Stream.  As for multiple RTP payload types on the m-line, any one
      of the alternative formats can be used at a given point in time,
      but not more than one (based on RTP timestamp), and what format is
      used can change dynamically from one RTP packet to another.  For
      example, if all participants in a group video call can decode
      H.264 and H.265 video, but only some can encode H.265, both H.264
      and H.265 can be kept as alternative formats, and the format may
      dynamically switch between H.264 and H.265 as different
      participants become active speaker.

2.2.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Use Cases

   Many use cases of simulcast as described in this document relate to a
   multi-party communication session where one or more central nodes are
   used to adapt the view of the communication session towards
   individual participants, and facilitate the media transport between
   participants.  Thus, these cases targets the RTP Mixer type of

   There are two principle approaches for an RTP Mixer to provide this
   adapted view of the communication session to each receiving

   o  Transcoding (decoding and re-encoding) received RTP streams with
      characteristics adapted to each receiving participant.  This often
      include mixing or composition of media sources from multiple
      participants into a mixed media source originated by the RTP
      Mixer.  The main advantage of this approach is that it achieves
      close to optimal adaptation to individual receiving participants.
      The main disadvantages are that it can be very computationally
      expensive to the RTP Mixer and typically also degrades media

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      Quality of Experience (QoE) such as end-to-end delay for the
      receiving participants.

   o  Switching a subset of all received RTP streams or sub-streams to
      each receiving participant, where the used subset is typically
      specific to each receiving participant.  The main advantages of
      this approach are that it is computationally cheap to the RTP
      Mixer and it has very limited impact on media QoE.  The main
      disadvantage is that it can be difficult to combine a subset of
      received RTP streams into a perfect fit to the resource situation
      of a receiving participant.

   The use of simulcast relates to the latter approach, where it is more
   important to reduce the load on the RTP Mixer and/or minimize QoE
   impact than to achieve an optimal adaptation of resource usage.

3.1.  Reaching a Diverse Set of Receivers

   The media sources provided by a sending participant potentially need
   to reach several receiving participants that differ in terms of
   available resources.  The receiver resources that typically differ
   include, but are not limited to:

   Codec:  This includes codec type (such as SDP MIME type) and can
      include codec configuration options (e.g.  SDP fmtp parameters).
      A couple of codec resources that differ only in codec
      configuration will be "different" if they are somehow not
      "compatible", like if they differ in video codec profile, or the
      transport packetization configuration.

   Sampling:  This relates to how the media source is sampled, in
      spatial as well as in temporal domain.  For video streams, spatial
      sampling affects image resolution and temporal sampling affects
      video frame rate.  For audio, spatial sampling relates to the
      number of audio channels and temporal sampling affects audio
      bandwidth.  This may be used to suit different rendering
      capabilities or needs at the receiving endpoints, as well as a
      method to achieve different transport capabilities, bitrates and
      eventually QoE by controlling the amount of source data.

   Bitrate:  This relates to the amount of bits spent per second to
      transmit the media source as an RTP stream, which typically also
      affects the Quality of Experience (QoE) for the receiving user.

   Letting the sending participant create a simulcast of a few
   differently configured RTP streams per media source can be a good
   tradeoff when using an RTP switch as middlebox, instead of sending a

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   single RTP stream and using an RTP mixer to create individual
   transcodings to each receiving participant.

   This requires that the receiving participants can be categorized in
   terms of available resources and that the sending participant can
   choose a matching configuration for a single RTP stream per category
   and media source.

   For example, assume for simplicity a set of receiving participants
   that differ only in that some have support to receive Codec A, and
   the others have support to receive Codec B.  Further assume that the
   sending participant can send both Codec A and B.  It can then reach
   all receivers by creating two simulcasted RTP streams from each media
   source; one for Codec A and one for Codec B.

   In another simple example, a set of receiving participants differ
   only in screen resolution; some are able to display video with at
   most 360p resolution and some support 720p resolution.  A sending
   participant can then reach all receivers by creating a simulcast of
   RTP streams with 360p and 720p resolution for each sent video media

   In more elaborate cases, the receiving participants differ both in
   available sampling and bitrate, and maybe also codec, and it is up to
   the RTP switch to find a good trade-off in which simulcasted stream
   to choose for each intended receiver.  It is also the responsibility
   of the RTP switch to negotiate a good fit of simulcast streams with
   the sending participant.

   The maximum number of simulcasted RTP streams that can be sent is
   mainly limited by the amount of processing and uplink network
   resources available to the sending participant.

3.2.  Application Specific Media Source Handling

   The application logic that controls the communication session may
   include special handling of some media sources.  It is for example
   commonly the case that the media from a sending participant is not
   sent back to itself.

   It is also common that a currently active speaker participant is
   shown in larger size or higher quality than other participants (the
   sampling or bitrate aspects of Section 3.1).  Not sending the active
   speaker media back to itself means there is some other participant's
   media that instead has to receive special handling towards the active
   speaker; typically the previous active speaker.  This way, the
   previously active speaker is needed both in larger size (to current
   active speaker) and in small size (to the rest of the participants),

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   which can be solved with a simulcast from the previously active
   speaker to the RTP switch.

3.3.  Receiver Media Source Preferences

   The application logic that controls the communication session may
   allow receiving participants to apply preferences to the
   characteristics of the RTP stream they receive, for example in terms
   of the aspects listed in Section 3.1.  Sending a simulcast of RTP
   streams is one way of accommodating receivers with conflicting or
   otherwise incompatible preferences.

4.  Requirements

   The following requirements need to be met to support the use cases in
   previous sections:

      Editor's note: Consider adding an explicit requirement that the
      solution supports use of simulcast even when using multiple codecs
      and multiple redundant RTP streams per defined codec (or something
      similar), since this is really an existing requirement and should
      also fully motivate the use of RID as identification mechanism.

   REQ-1:  Identification.  It must be possible to identify a set of
      simulcasted RTP streams as originating from the same media source:

      REQ-1.1:  In SDP signaling.

      REQ-1.2:  On RTP/RTCP level.

   REQ-2:  Transport usage.  The solution must work when using:

      REQ-2.1:  Legacy SDP with separate media transports per SDP media

      REQ-2.2:  Bundled [I-D.ietf-mmusic-sdp-bundle-negotiation] SDP
         media descriptions.

   REQ-3:  Capability negotiation.  It must be possible that:

      REQ-3.1:  Sender can express capability of sending simulcast.

      REQ-3.2:  Receiver can express capability of receiving simulcast.

      REQ-3.3:  Sender can express maximum number of simulcast streams
         that can be provided.

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      REQ-3.4:  Receiver can express maximum number of simulcast streams
         that can be received.

      REQ-3.5:  Sender can detail the characteristics of the simulcast
         streams that can be provided.

      REQ-3.6:  Receiver can detail the characteristics of the simulcast
         streams that it prefers to receive.

   REQ-4:  Distinguishing features.  It must be possible to have
      different simulcast streams use different codec parameters, as can
      be expressed by SDP format values and RTP payload types.

   REQ-5:  Compatibility.  It must be possible to use simulcast in
      combination with other RTP mechanisms that generate additional RTP

      REQ-5.1:  RTP Retransmission [RFC4588].

      REQ-5.2:  RTP Forward Error Correction [RFC5109].

      REQ-5.3:  Related payload types such as audio Comfort Noise and/or

   REQ-6:  Interoperability.  The solution must be possible to use in:

      REQ-6.1:  Interworking with non-simulcast legacy clients using a
         single media source per media type.

      REQ-6.2:  WebRTC environment with a single media source per SDP
         media description.

5.  Overview

   As an overview, the above requirements are met by signaling simulcast
   capability and configurations in SDP [RFC4566]:

   o  An offer or answer can contain a number of simulcast streams,
      separate for send and receive directions.

   o  An offer or answer can contain multiple, alternative simulcast
      stream formats in the same fashion as multiple, alternative codecs
      can be offered in a media description.

   o  A single media source per SDP media description is assumed, which
      is aligned with the concepts defined in [RFC7656] and will
      specifically work in a WebRTC context, both with and without
      BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation] grouping.

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   o  The codec configuration for a simulcast stream is expressed
      through use of a separately specified RTP-level identification
      mechanism [I-D.ietf-mmusic-rid][I-D.roach-avtext-rid], which
      complements and effectively extends the available simulcast stream
      identification and configuration possibilities that could be
      provided by using only SDP formats.

   o  It is possible, but not required to use source-specific signaling
      [RFC5576] with the proposed solution.

6.  Detailed Description

   This section further details the overview above (Section 5).  First,
   formal syntax is provided (Section 6.1), followed by the rest of the
   SDP attribute definition in Section 6.2.  Relating Simulcast Streams
   (Section 6.3) provides the definition of the RTP/RTCP mechanisms
   used.  The section is concluded with a number of examples.

6.1.  Simulcast Attribute

   Name:  simulcast

   Value:  sc-value

   Usage Level:  media

   Charset Dependent:  no

   Multiplex Category:  NORMAL

   Syntax [RFC5234]:

   sc-attr      = "a=simulcast:" sc-value
   sc-value     = sc-str-list [SP sc-str-list]
   sc-str-list  = sc-dir SP sc-alt-list *( ";" sc-alt-list )
   sc-dir       = "send" / "recv"
   sc-alt-list  = sc-id *( "," sc-id )
   sc-id-paused = "~"
   sc-id        = [sc-id-paused] rid-identifier / token
   ; SP defined in [RFC5234]
   ; token defined in [RFC4566]
   ; rid-identifier defined in [I-D.ietf-mmusic-rid]

                       Figure 1: ABNF for Simulcast

   The "a=simulcast" attribute has a parameter in the form of one or two
   simulcast stream descriptions, each consisting of a direction ("send"

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   or "recv"), followed by a list of one or more simulcast streams.
   Each simulcast stream in that list is separated by a semicolon (";").
   Each simulcast stream can in turn be offered in one or more
   alternative formats, where each simulcast stream alternative is
   separated by a comma (",").  The simulcast stream alternative MUST be
   described in the form of a RID, as described by
   [I-D.ietf-mmusic-rid].  Each simulcast stream can be initially paused
   [I-D.ietf-avtext-rtp-stream-pause], indicated by prepending a "~" to
   the simulcast stream.  In case there are simulcast stream
   alternatives, pause can be specified individually for each
   alternative.  The reason to allow separate initial pause states for
   each simulcast stream alternative is that pause capability can be
   specified individually for each RTP payload type referenced by a RID,
   which makes it infeasible to pause RID where any of the related RTP
   payload type(s) do not have pause capability.


   a=simulcast:send 1,2,3;~4,~5 recv 1;~2,~5
   a=simulcast:recv 1;4,5 send 1;2

                       Figure 2: Simulcast Examples

   Above are two examples of different "a=simulcast" lines.

   The first line is an example offer to send two simulcast streams and
   to receive two simulcast streams.  The first simulcast stream in send
   direction can be sent as three different alternatives (1, 2, 3), and
   the second simulcast stream in send direction can be sent as two
   different alternatives (4, 5).  All second stream send alternatives
   are offered as initially paused.  The first simulcast stream in
   receive direction has no alternatives (only 1).  The second simulcast
   stream in receive direction has two alternatives (2, 5) that are both
   offered as initially paused.

   The second line is an example answer to the first line, accepting to
   send and receive the two offered simulcast streams, however send and
   receive directions are specified in opposite order compared to the
   first line, which lets the answer keep the same order of simulcast
   streams in the SDP as in the offer, even though directionality is
   reversed.  This example answer has removed all offered alternatives
   for the first simulcast stream (keeping only 1), but kept alternative
   formats for the second simulcast stream in receive direction (4, 5).
   The answer accepts to send two simulcast streams, without
   alternatives.  The answer does not accept initial pause of any
   simulcast streams, in either direction.  More examples can be found
   in Section 6.4.

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6.2.  Simulcast Capability

   Simulcast capability is expressed as a new media level SDP attribute,
   "a=simulcast" (Section 6.1), with multiplex category
   [I-D.ietf-mmusic-sdp-mux-attributes] NORMAL.

   For each desired direction (send/recv), the simulcast attribute
   defines a list of simulcast streams (separated by semicolons), each
   of which is a list of simulcast formats (separated by commas).  The
   meaning of the attribute on SDP session level is undefined and MUST
   NOT be used.

   The meaning of including multiple "a=simulcast" lines in a single SDP
   media description is undefined and MUST NOT be used.  There are
   separate and independent sets of parameters for simulcast in send and
   receive directions.  When listing multiple directions, each direction
   MUST NOT occur more than once on the same line.

   The different simulcast streams MUST be identified through the RTP-
   level "RID" identification mechanism [I-D.ietf-mmusic-rid].

   Attribute parameters are grouped by direction and consist of a
   listing of simulcast stream identifications to be used.  The number
   of (non-alternative, see below) identifications in the list sets a
   limit to the number of supported simulcast streams in that direction.
   The order of the listed simulcast versions in the "send" direction
   suggests a proposed order of preference, in decreasing order: the
   stream listed first is the most preferred Section 3.1, and subsequent
   streams have progressively lower preference.  The order of the listed
   simulcast streams in the "recv" direction expresses a preference
   which simulcast streams that are preferred, with the leftmost being
   most preferred.  This can be of importance if the number of actually
   sent simulcast streams have to be reduced for some reason.

   Formats that have explicit dependencies [RFC5583]
   [I-D.ietf-mmusic-rid] to other formats (even in the same media
   description) MAY be listed as different simulcast streams.

   Alternative simulcast formats MAY be specified as part of the
   attribute parameters by expressing each simulcast stream as a comma-
   separated list of alternative format identifiers.  In this case, it
   is not possible to align what alternative formats that are used
   between different simulcast streams, like requiring all simulcast
   streams to use alternatives with the same codec format.  The order of
   the format alternatives within a simulcast stream is significant; the
   alternatives are listed from (left) most preferred to (right) least
   preferred.  For the use of simulcast, this overrides the normal codec
   preference as expressed by format type ordering on the "m="-line,

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   using regular SDP rules.  This is to enable a separation of general
   codec preferences and simulcast stream configuration preferences.

   A simulcast stream can use a codec defined such that the same RTP
   SSRC can change RTP payload type multiple times during a session,
   possibly even on a per-packet basis.  A typical example can be a
   speech codec that makes use of Comfort Noise [RFC3389] and/or DTMF
   [RFC4733] formats.  In those cases, such "related" formats MUST NOT
   be listed explicitly in the attribute parameters, since they are not
   strictly simulcast streams of the media source, but rather a specific
   way of generating the RTP stream of a single simulcast stream with
   varying RTP payload type.  Instead, only a single simulcast stream
   identification MUST be used per simulcast stream or alternative
   simulcast format (if there are such) in the SDP.

   If RTP stream pause/resume [I-D.ietf-avtext-rtp-stream-pause] is
   supported, any simulcast stream identification MAY be prefixed by a
   "~" character to indicate that the corresponding simulcast stream is
   initially paused already from start of the RTP session.  In this
   case, support for RTP stream pause/resume MUST also be included under
   the same "m="-line listing "a=simulcast".  If the simulcast stream is
   specified as a list of alternative formats, the indication is
   prepended to the first format of the list and applies to whatever
   alternative that is eventually chosen.  All RTP payload types related
   to such initially paused simulcast stream MUST be listed in the SDP
   as pause/resume capable as specified by

   An initially paused simulcast stream in "send" direction MUST be
   considered equivalent to an unsolicited locally paused stream, and be
   handled accordingly.  Initially paused simulcast streams are resumed
   as described by the RTP pause/resume specification.  An RTP stream
   receiver that wishes to resume an unsolicited locally paused stream
   needs to know the SSRC of that stream.  The SSRC of an initially
   paused simulcast stream can be obtained from an RTP stream sender
   RTCP Sender Report (SR) including both the desired SSRC as "SSRC of
   sender", and the stream RID identification as an RID RTCP SDES item.

   Including an initially paused simulcast stream in "recv" direction in
   an SDP towards an RTP sender, SHOULD cause the remote RTP sender to
   put the stream as unsolicited locally paused, unless there are other
   RTP stream receivers that do not mark the simulcast stream as
   initially paused.  The reason to require an initially paused "recv"
   stream to be considered locally paused by the remote RTP sender,
   instead of making it equivalent to implicitly sending a pause
   request, is because the pausing RTP sender cannot know which SSRC
   owns the restriction when TMMBR/TMMBN are used for pause/resume

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   signaling since the RTP receiver's SSRC in send direction is not
   known yet.

   Use of the redundant audio data [RFC2198] format could be seen as a
   form of simulcast for loss protection purposes, but is not considered
   conflicting with the mechanisms described in this memo and MAY
   therefore be used as any other format.  In this case the "red"
   format, rather than the carried formats, SHOULD be the one to list as
   a simulcast stream on the "a=simulcast" line.

   The media formats and corresponding characteristics of simulcast
   streams SHOULD be chosen such that they are different.  If this
   difference is not required, RTP duplication [RFC7104] procedures
   SHOULD be considered instead of simulcast.

6.2.1.  Declarative Use

   When used as a declarative media description, "a=simulcast" line
   "recv" direction formats indicate the configured end point's required
   capability to recognize and receive a specified set of RTP streams as
   simulcast streams.  In the same fashion, "a=simulcast" line "send"
   direction requests the end point to send a specified set of RTP
   streams as simulcast streams.

   If multiple simulcast formats are listed, it means that the
   configured end point MUST be prepared to receive any of the "recv"
   formats, and MAY send any of the "send" formats for that simulcast

      Editor's note: It may not be beneficial for declarative use to be
      limited to a single media source per "m=" line, as elaborated
      further in Section 8.

6.2.2.  Offer/Answer Use

   An offerer wanting to use simulcast SHALL include the "a=simulcast"
   attribute in the offer.  An offerer that receives an answer without
   "a=simulcast" MUST NOT use simulcast towards the answerer.  An
   offerer that receives an answer with "a=simulcast" without any
   simulcast stream identifications in a specified direction MUST NOT
   use simulcast in that direction.

   An answerer that does not understand the concept of simulcast will
   also not know the attribute and will remove it in the SDP answer, as
   defined in existing SDP Offer/Answer [RFC3264] procedures.

   An answerer that does understand the attribute and that wants to
   support simulcast in an indicated direction SHALL reverse

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   directionality of the unidirectional direction parameters; "send"
   becomes "recv" and vice versa, and include it in the answer.

   An offerer listing a set of receive simulcast streams and/or
   alternative formats in the offer MUST be prepared to receive RTP
   streams for any of those simulcast streams and/or alternative formats
   from the answerer.

   An answerer that receives an offer with simulcast containing an
   "a=simulcast" attribute listing alternative formats for simulcast
   streams MAY keep all the alternatives in the answer, but it MAY also
   choose to remove any non-desirable alternatives per simulcast stream
   in the answer.  The answerer MUST NOT add any alternatives that were
   not present in the offer.

   An answerer that receives an offer with simulcast that lists a number
   of simulcast streams, MAY reduce the number of simulcast streams in
   the answer, but MUST NOT add simulcast streams.

   An offerer that receives an answer where some simulcast formats are
   kept MUST be prepared to receive any of the kept send direction
   alternatives, and MAY send any of the kept receive direction
   alternatives from the answer.  Similarly, the answerer MUST be
   prepared to receive any of the kept receive direction alternatives,
   and MAY send any of the kept send direction alternatives in the

   The offerer and answerer MUST NOT send more than a single alternative
   format at a time (based on RTP timestamps) per simulcast stream, but
   MAY change format on a per-RTP packet basis.  This corresponds to the
   existing (non-simulcast) SDP offer/answer case when multiple formats
   are included on the "m=" line in the SDP answer.

   An offerer that receives an answer where some of the simulcast
   streams are removed MAY release the corresponding resources (codec,
   transport, etc) in its receive direction and MUST NOT send any RTP
   packets corresponding to the removed simulcast streams.

   Simulcast streams or formats using undefined simulcast stream
   identifications MUST NOT be used as valid simulcast streams by an RTP
   stream receiver.

   An answerer that receives an offer without RTP stream pause/resume
   capability MUST NOT mark any simulcast streams as initially paused in
   the answer.

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   An answerer that receives an offer with RTP stream pause/resume
   capability MAY mark any simulcast streams as initially paused in the

   An answerer that receives indication in an offer of a simulcast
   stream being initially paused , SHOULD mark that simulcast stream as
   initially paused also in the answer, regardless of direction, unless
   it has good reason for the stream not being initially paused.

   An offerer that offered some of its simulcast streams as initially
   paused and that receives an answer that does not indicate RTP stream
   pause/resume capability, MUST NOT intially pause any simulcast

   An offerer with RTP stream pause/resume capability that receives an
   answer where some simulcast streams are marked as initially paused,
   SHOULD initially pause them regardless if they were marked as
   initially paused also in the offer, unless it has good reason for
   those streams not being initially paused.

      Note: The inclusion of "a=simulcast" or the use of simulcast does
      not change any of the interpretation or Offer/Answer procedures
      for other SDP attributes, like "a=fmtp" or "a=rid".

6.3.  Relating Simulcast Streams

   Simulcast RTP streams MUST be related on RTP level through RID
   [I-D.roach-avtext-rid], as specified in the SDP "a=simulcast"
   attribute (Section 6.2) parameters.  This is sufficient as long as
   there is only a single media source per SDP media description.  When
   using BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], where multiple
   SDP media descriptions jointly specify a single RTP session, the SDES
   MID identification mechanism in BUNDLE allows relating RTP streams
   back to individual media descriptions, after which the above
   described RID relations can be used.  Use of the RTP header extension
   [RFC5285] for both MID and RID identifications can be important to
   ensure rapid initial reception, required to correctly interpret and
   process the RTP streams.  Implementers of this specification MUST
   support RTCP source description (SDES) item and SHOULD support RTP
   header extension method to signal RID on RTP level.

6.4.  Signaling Examples

   These examples describe a client to video conference service, using a
   centralized media topology with an RTP mixer.

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                    +---+      +-----------+      +---+
                    | A |<---->|           |<---->| B |
                    +---+      |           |      +---+
                               |   Mixer   |
                    +---+      |           |      +---+
                    | F |<---->|           |<---->| J |
                    +---+      +-----------+      +---+

                Figure 3: Four-party Mixer-based Conference

6.4.1.  Single-Source Client

   Alice is calling in to the mixer with a simulcast-enabled client
   capable of a single media source per media type.  The client can send
   a simulcast of 2 video resolutions and frame rates: HD 1280x720p
   30fps and thumbnail 320x180p 15fps.  This is defined below using the
   "imageattr" [RFC6236].  In this example, only the "pt" RID parameter
   is used, effectively achieving a 1:1 mapping between RID and media
   formats (RTP payload types), to describe simulcast stream formats.
   Alice's Offer:

   o=alice 2362969037 2362969040 IN IP4
   s=Simulcast Enabled Client
   t=0 0
   c=IN IP4
   m=audio 49200 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   m=video 49300 RTP/AVP 97 98
   a=rtpmap:97 H264/90000
   a=rtpmap:98 H264/90000
   a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
   a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
   a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
   a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
   a=rid:1 pt=97
   a=rid:2 pt=98
   a=simulcast:send 1;2 recv 1

                  Figure 4: Single-Source Simulcast Offer

   The only thing in the SDP that indicates simulcast capability is the
   line in the video media description containing the "simulcast"
   attribute.  The included format parameters indicates that sent
   simulcast streams can differ in video resolution.

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   The Answer from the server indicates that it too is simulcast
   capable.  Should it not have been simulcast capable, the
   "a=simulcast" line would not have been present and communication
   would have started with the media negotiated in the SDP.

   o=server 823479283 1209384938 IN IP4
   s=Answer to Simulcast Enabled Client
   t=0 0
   c=IN IP4
   m=audio 49672 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   m=video 49674 RTP/AVP 97 98
   a=rtpmap:97 H264/90000
   a=rtpmap:98 H264/90000
   a=fmtp:97 profile-level-id=42c01f; max-fs=3600; max-mbps=108000
   a=fmtp:98 profile-level-id=42c00b; max-fs=240; max-mbps=3600
   a=imageattr:97 send [x=1280,y=720] recv [x=1280,y=720]
   a=imageattr:98 send [x=320,y=180] recv [x=320,y=180]
   a=rid:1 pt=97
   a=rid:2 pt=98
   a=simulcast:recv 1;2 send 1

                 Figure 5: Single-Source Simulcast Answer

   Since the server is the simulcast media receiver, it reverses the
   direction of the "simulcast" attribute parameters.

6.4.2.  Multi-Source Client

   Fred is calling in to the same conference as in the example above
   with a two-camera, two-display system, thus capable of handling two
   separate media sources in each direction, where each media source is
   simulcast-enabled in the send direction.  Fred's client is restricted
   to a single media source per media description.

   The first two simulcast streams for the first media source use
   different codecs, H264-SVC [RFC6190] and H264 [RFC6184].  These two
   simulcast streams also have a temporal dependency.  Two different
   video codecs, VP8 [I-D.ietf-payload-vp8] and H264, are offered as
   alternatives for the third simulcast stream for the first media
   source.  Only the highest fidelity simulcast stream are sent from
   start, the lower fidelity streams being initially paused.

   The second media source is offered with three different simulcast
   streams.  All video streams of this second media source are loss

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   protected by RTP retransmission [RFC4588].  Also here, all but the
   highest fidelity simulcast stream are initially paused.

   Fred's client is also using BUNDLE to send all RTP streams from all
   media descriptions in the same RTP session on a single media
   transport.  Although using many different simulcast streams in this
   example, the use of RID as simulcast stream identification enables
   use of a low number of RTP payload types.  Note that the use of both
   BUNDLE and RID recommends using the RTP header extension [RFC5285]
   for carrying these fields, which is consequently also included in the

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   o=fred 238947129 823479223 IN IP4
   s=Offer from Simulcast Enabled Multi-Source Client
   t=0 0
   c=IN IP4
   a=group:BUNDLE foo bar zen

   m=audio 49200 RTP/AVP 99
   a=rtpmap:99 G722/8000

   m=video 49600 RTP/AVPF 100 101 103
   a=rtpmap:100 H264-SVC/90000
   a=rtpmap:101 H264/90000
   a=rtpmap:103 VP8/90000
   a=fmtp:100 profile-level-id=42400d; max-fs=3600; max-mbps=108000; \
   a=fmtp:101 profile-level-id=42c00d; max-fs=3600; max-mbps=54000
   a=fmtp:103 max-fs=900; max-fr=30
   a=rid:1 send pt=100;max-width=1280;max-height=720;max-fr=60;depend=2
   a=rid:2 send pt=101;max-width=1280;max-height=720;max-fr=30
   a=rid:3 send pt=101;max-width=640;max-height=360
   a=rid:4 send pt=103;max-width=640;max-height=360
   a=depend:100 lay bar:101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rid
   a=rtcp-fb:* ccm pause nowait
   a=simulcast:send 1;2;~4,3

   m=video 49602 RTP/AVPF 96 104
   a=rtpmap:96 VP8/90000
   a=fmtp:96 max-fs=3600; max-fr=30
   a=rtpmap:104 rtx/90000
   a=fmtp:104 apt=96;rtx-time=200
   a=rid:5 send pt=96;max-fs=921600;max-fr=30
   a=rid:6 send pt=96;max-fs=614400;max-fr=15
   a=rid:7 send pt=96;max-fs=230400;max-fr=30
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rid
   a=rtcp-fb:* ccm pause nowait
   a=simulcast:send 5;~6;~7

               Figure 6: Fred's Multi-Source Simulcast Offer

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      Note: Empty lines in the SDP above are added only for readability
      and would not be present in an actual SDP.

7.  Network Aspects

   Simulcast is in this memo defined as the act of sending multiple
   alternative encoded streams of the same underlying media source.
   When transmitting multiple independent streams that originate from
   the same source, it could potentially be done in several different
   ways using RTP.  A general discussion on considerations for use of
   the different RTP multiplexing alternatives can be found in
   Guidelines for Multiplexing in RTP
   [I-D.ietf-avtcore-multiplex-guidelines].  Discussion and
   clarification on how to handle multiple streams in an RTP session can
   be found in [I-D.ietf-avtcore-rtp-multi-stream].

   The network aspects that are relevant for simulcast are:

   Quality of Service:  When using simulcast it might be of interest to
      prioritize a particular simulcast stream, rather than applying
      equal treatment to all streams.  For example, lower bit-rate
      streams may be prioritized over higher bit-rate streams to
      minimize congestion or packet losses in the low bit-rate streams.
      Thus, there is a benefit to use a simulcast solution with good QoS

   NAT/FW Traversal:  Using multiple RTP sessions incurs more cost for
      NAT/FW traversal unless they can re-use the same transport flow,
      which can be achieved by Multiplexing Negotiation Using SDP Port
      Numbers [I-D.ietf-mmusic-sdp-bundle-negotiation].

7.1.  Bitrate Adaptation

   Use of multiple simulcast streams can require a significant amount of
   network resources.  If the amount of available network resources
   varies during an RTP session such that it does not match what is
   negotiated in SDP, the bitrate used by the different simulcast
   streams may have to be reduced dynamically.  What simulcast streams
   to prioritize when allocating available bitrate among the simulcast
   streams in such adaptation SHOULD be taken from the simulcast stream
   order on the "a=simulcast" line.  Simulcast streams that have pause/
   resume capability and that would be given such low bitrate by the
   adaptation process that they are considered not really useful can be
   temporarily paused until the limiting condition clears.

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8.  Limitations

   The chosen approach has a few limitations that are described in this
   section.  The only one currently described relates to the use of a
   single RTP session for all simulcast formats of a media source.

8.1.  Single RTP Session

   The limitations in this section come from sending all simulcast
   streams related to a media source under the same SDP media
   description, which also means they are sent in the same RTP session.

   It is not possible to use different simulcast streams on different
   media transports, limiting the possibilities to apply different QoS
   to different simulcast streams.  When using unicast, QoS mechanisms
   based on individual packet marking are feasible, since they do not
   require separation of simulcast streams into different RTP sessions
   to apply different QoS.

   It is not possible to separate different simulcast streams into
   different multicast groups to allow a multicast receiver to pick the
   stream it wants, rather than receive all of them.  In this case, the
   only reasonable implementation is to use different RTP sessions for
   each multicast group so that reporting and other RTCP functions
   operate as intended.

9.  IANA Considerations

   This document requests to register a new SDP attribute, simulcast, as
   defined in Section 6.1.

10.  Security Considerations

   The simulcast capability, configuration attributes, and parameters
   are vulnerable to attacks in signaling.

   A false inclusion of the "a=simulcast" attribute may result in
   simultaneous transmission of multiple RTP streams that would
   otherwise not be generated.  The impact is limited by the media
   description joint bandwidth, shared by all simulcast streams
   irrespective of their number.  There may however be a large number of
   unwanted RTP streams that will impact the share of bandwidth
   allocated for the originally wanted RTP stream.

   A hostile removal of the "a=simulcast" attribute will result in
   simulcast not being used.

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   Neither of the above will likely have any major consequences and can
   be mitigated by signaling that is at least integrity and source
   authenticated to prevent an attacker to change it.

   Security considerations related to the use of RID is covered in
   [I-D.ietf-mmusic-rid] and [I-D.roach-avtext-rid].  There are no
   additional security concerns related to its use in this

11.  Contributors

   Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
   contributed with important material to the first versions of this
   document.  Robert Hansen and Cullen Jennings, from Cisco, Peter
   Thatcher, from Google, and Adam Roach, from Mozilla, contributed
   significantly to subsequent versions.

12.  Acknowledgements

13.  References

13.1.  Normative References

              Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP
              Stream Pause and Resume", draft-ietf-avtext-rtp-stream-
              pause-10 (work in progress), September 2015.

              Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
              Roach, A., and B. Campen, "RTP Payload Format
              Constraints", draft-ietf-mmusic-rid-01 (work in progress),
              February 2016.

              Nandakumar, S., "A Framework for SDP Attributes when
              Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-12
              (work in progress), January 2016.

              Roach, A., Nandakumar, S., and P. Thatcher, "RTP Payload
              Format Constraints", draft-roach-avtext-rid-01 (work in
              progress), February 2016.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

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   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <>.

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <>.

   [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234,
              DOI 10.17487/RFC5234, January 2008,

   [RFC7104]  Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
              Semantics in the Session Description Protocol", RFC 7104,
              DOI 10.17487/RFC7104, January 2014,

13.2.  Informative References

              Westerlund, M., Perkins, C., and H. Alvestrand,
              "Guidelines for using the Multiplexing Features of RTP to
              Support Multiple Media Streams", draft-ietf-avtcore-
              multiplex-guidelines-03 (work in progress), October 2014.

              Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
              December 2015.

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-25 (work in progress), January 2016.

              Singh, V., Begen, A., Zanaty, M., and G. Mandyam, "RTP
              Payload Format for Flexible Forward Error Correction
              (FEC)", draft-ietf-payload-flexible-fec-scheme-01 (work in
              progress), October 2015.

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              Westin, P., Lundin, H., Glover, M., Uberti, J., and F.
              Galligan, "RTP Payload Format for VP8 Video", draft-ietf-
              payload-vp8-17 (work in progress), September 2015.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              DOI 10.17487/RFC2198, September 1997,

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
              September 2002, <>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              DOI 10.17487/RFC4733, December 2006,

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <>.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)",
              RFC 5583, DOI 10.17487/RFC5583, July 2009,

   [RFC6184]  Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP
              Payload Format for H.264 Video", RFC 6184,
              DOI 10.17487/RFC6184, May 2011,

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   [RFC6190]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding", RFC 6190,
              DOI 10.17487/RFC6190, May 2011,

   [RFC6236]  Johansson, I. and K. Jung, "Negotiation of Generic Image
              Attributes in the Session Description Protocol (SDP)",
              RFC 6236, DOI 10.17487/RFC6236, May 2011,

   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
              DOI 10.17487/RFC7656, November 2015,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

Appendix A.  Changes From Earlier Versions

   NOTE TO RFC EDITOR: Please remove this section prior to publication.

A.1.  Modifications Between WG Version -03 and -04

   o  Changed to only use RID identification, as was consensus during
      IETF 94.

   o  ABNF improvements.

   o  Clarified offer-answer rules for initially paused streams.

   o  Changed references for RTP topologies and RTP taxonomy documents
      that are now published as RFC.

   o  Added reference to the new RID draft in AVTEXT.

   o  Re-structured section 6 to provide an easy reference by the
      updated IANA section.

   o  Added a sub-section 7.1 with a discussion of bitrate adaptation.

   o  Editorial improvements.

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A.2.  Modifications Between WG Version -02 and -03

   o  Removed text on multicast / broadcast from use cases, since it is
      not supported by the solution.

   o  Removed explicit references to unified plan draft.

   o  Added possibility to initiate simulcast streams in paused mode.

   o  Enabled an offerer to offer multiple stream identification (pt or
      rid) methods and have the answerer choose which to use.

   o  Added a preference indication also in send direction offers.

   o  Added a section on limitations of the current proposal, including
      identification method specific limitations.

A.3.  Modifications Between WG Version -01 and -02

   o  Relying on the new RID solution for codec constraints and
      configuration identification.  This has resulted in changes in
      syntax to identify if pt or RID is used to describe the simulcast

   o  Renamed simulcast version and simulcast version alternative to
      simulcast stream and simulcast format respectively, and improved
      definitions for them.

   o  Clarification that it is possible to switch between simulcast
      version alternatives, but that only a single one be used at any
      point in time.

   o  Changed the definition so that ordering of simulcast formats for a
      specific simulcast stream do have a preference order.

A.4.  Modifications Between WG Version -00 and -01

   o  No changes.  Only preventing expiry.

A.5.  Modifications Between Individual Version -00 and WG Version -00

   o  Added this appendix.

Authors' Addresses

Burman, et al.           Expires August 6, 2016                [Page 26]

Internet-Draft                  Simulcast                  February 2016

   Bo Burman
   Kistavagen 25
   SE-164 80 Stockholm


   Magnus Westerlund
   Farogatan 2
   SE-164 80 Stockholm

   Phone: +46 10 714 82 87

   Suhas Nandakumar
   170 West Tasman Drive
   San Jose, CA  95134


   Mo Zanaty
   170 West Tasman Drive
   San Jose, CA  95134


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