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Versions: (draft-burman-mmusic-sdp-simulcast) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14

Network Working Group                                          B. Burman
Internet-Draft                                             M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: December 31, 2016                                 S. Nandakumar
                                                               M. Zanaty
                                                                   Cisco
                                                           June 29, 2016


                Using Simulcast in SDP and RTP Sessions
                   draft-ietf-mmusic-sdp-simulcast-05

Abstract

   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
   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 December 31, 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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   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
   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
     6.3.  Offer/Answer Use  . . . . . . . . . . . . . . . . . . . .  13
       6.3.1.  Generating the Initial SDP Offer  . . . . . . . . . .  13
       6.3.2.  Creating the SDP Answer . . . . . . . . . . . . . . .  13
       6.3.3.  Offerer Processing the SDP Answer . . . . . . . . . .  14
       6.3.4.  Modifying the Session . . . . . . . . . . . . . . . .  15
     6.4.  Declarative Use . . . . . . . . . . . . . . . . . . . . .  15
     6.5.  Relating Simulcast Streams  . . . . . . . . . . . . . . .  15
     6.6.  Signaling Examples  . . . . . . . . . . . . . . . . . . .  16
       6.6.1.  Single-Source Client  . . . . . . . . . . . . . . . .  16
       6.6.2.  Multi-Source Client . . . . . . . . . . . . . . . . .  18
   7.  Network Aspects . . . . . . . . . . . . . . . . . . . . . . .  21
     7.1.  Bitrate Adaptation  . . . . . . . . . . . . . . . . . . .  21
   8.  Limitation  . . . . . . . . . . . . . . . . . . . . . . . . .  22
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  22
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  23
   11. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  23
   12. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  23
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  23
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  23
     13.2.  Informative References . . . . . . . . . . . . . . . . .  25
   Appendix A.  Changes From Earlier Versions  . . . . . . . . . . .  26
     A.1.  Modifications Between WG Version -04 and  -05 . . . . . .  26
     A.2.  Modifications Between WG Version -03 and  -04 . . . . . .  27



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     A.3.  Modifications Between WG Version -02 and  -03 . . . . . .  27
     A.4.  Modifications Between WG Version -01 and  -02 . . . . . .  28
     A.5.  Modifications Between WG Version -00 and  -01 . . . . . .  28
     A.6.  Modifications Between Individual Version -00 and WG
           Version -00 . . . . . . . . . . . . . . . . . . . . . . .  28
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  28

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
   done.

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

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
      3.9).






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   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
      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 in offer/
      answer [RFC3264], any one of the negotiated alternative formats
      can be used in a single RTP stream at a given point in time, but
      not more than one (based on RTP timestamp).  What format is used
      can change dynamically from one RTP packet to another.

   Simulcast Stream Identifier (SCID):  The identification value used to
      refer to individual simulcast streams, identical to the "rid-id"
      identification value for an RTP Constraint [I-D.ietf-mmusic-rid]
      and the corresponding content of "RtpStreamId" RTCP SDES Item
      [I-D.ietf-avtext-rid].

2.2.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

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 target the RTP Mixer type of
   topology.

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




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   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
      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.




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   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
   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 with best possible
   resolution by creating a simulcast of RTP streams with 360p and 720p
   resolution for each sent video media source.

   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.





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   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),
   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
         description.

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

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




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      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.

      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
      streams:

      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
         DTMF.

   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.





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   o  An offer or answer can contain multiple, alternative simulcast
      stream formats in the same fashion as multiple, alternative
      formats 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.

   o  The codec configuration for a simulcast stream is expressed
      through use of separately specified RTP payload format constraints
      [I-D.ietf-mmusic-rid] with an associated RTP-level identification
      mechanism [I-D.ietf-avtext-rid] to identify which RTP payload
      format constraints an RTP stream adheres to.  This complements and
      effectively extends simulcast stream identification and
      configuration possibilities that could be provided by using only
      SDP formats as identifier.

   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 6.5) provides the definition of the RTP/RTCP mechanisms
   used.  The section is concluded with a number of examples.

6.1.  Simulcast Attribute

   This document defines a new SDP media-level "a=simulcast" attribute
   with the following ABNF [RFC5234] syntax:

   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
   ; SP defined in [RFC5234]
   ; rid-identifier defined in [I-D.ietf-mmusic-rid]


                       Figure 1: ABNF for Simulcast





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   The "a=simulcast" attribute has a parameter in the form of one or two
   simulcast stream descriptions, each consisting of a direction ("send"
   or "recv"), followed by a list of one or more simulcast streams.  The
   simulcast stream identification (SCID) MUST be have the form of an
   RTP stream identifier, as described by RTP Payload Format Constraints
   [I-D.ietf-mmusic-rid].

   In the list of simulcast streams, each SCID is separated by a
   semicolon (";").  Each simulcast stream can in turn be offered in one
   or more alternative formats, where each alternative SCID is separated
   by a comma (",").  Each simulcast stream can also be specified as
   initially paused [RFC7728], indicated by prepending a "~" to the
   SCID.  In case there are alternative SCID, pause can be specified
   individually for each SCID.  The reason to allow separate initial
   pause states for each SCID is that pause capability can be specified
   individually for each RTP payload type referenced by an SCID.  Since
   pause capability specified via the "a=rtcp-fb" attribute and SCID
   specified by "a=rid" can refer to common payload types, it is
   unfeasible to pause streams with SCID where any of the related RTP
   payload type(s) do not have pause capability.

   Examples:

   a=simulcast:send 1,2,3;~4,~5 recv 6;~7,~8
   a=simulcast:recv 1;4,5 send 6;7


                       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 in three different alternative formats (SCID 1,
   2, 3), and the second simulcast stream in send direction can be sent
   in two different alternative formats (SCID 4, 5).  Both of the second
   stream alternative formats in send direction are offered as initially
   paused.  The first simulcast stream in receive direction has no
   alternative formats (SCID 6).  The second simulcast stream in receive
   direction has two alternative formats (SCID 7, 8) 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, for convenience, even though
   directionality is reversed.  This example answer has removed all



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   offered alternative formats for the first simulcast stream (keeping
   only SCID 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 6.6.

6.2.  Simulcast Capability

   Simulcast capability is expressed through a new media level SDP
   attribute, "a=simulcast" (Section 6.1).  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.

   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 alternate simulcast stream formats (separated
   by commas).  The different simulcast stream formats MUST be
   identified through the RTP payload format constraints
   [I-D.ietf-mmusic-rid] RTP-level identification mechanism
   [I-D.ietf-avtext-rid], here denoted SCID.  Simulcast streams using
   undefined SCID MUST NOT be used as valid simulcast streams by an RTP
   stream receiver.

   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.
   Attribute parameters are grouped by direction and consist of a
   listing of SCID to be used.  The direction for an SCID MUST be
   aligned with the direction specified for the corresponding RTP stream
   identifier on the "a=rid" line.

   The number of (non-alternative, see above) SCID in the list sets a
   limit to the number of supported simulcast streams in that direction.
   The order of the listed SCID in the "send" direction suggests a
   proposed order of preference, in decreasing order: the SCID listed
   first is the most preferred and subsequent streams have progressively
   lower preference.  The order of the listed SCID 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.

   SCID that have explicit dependencies [RFC5583] [I-D.ietf-mmusic-rid]
   to other SCID (even in the same media description) MAY be used.





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   Alternative SCID MAY be specified as part of the attribute parameters
   by expressing each simulcast stream as a comma-separated list of
   alternative SCID.  In this case, it is not possible to align what
   alternative SCID that are used between different simulcast streams,
   like requiring all simulcast streams to use SCID alternatives
   referring to the same codec format.  The order of the SCID
   alternatives within a simulcast stream is significant; the SCID
   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,
   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 defined as SCID and 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.

   If RTP stream pause/resume [RFC7728] is supported, any SCID 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 where "a=simulcast" is included.
   If the simulcast stream is specified as a list of alternative SCID,
   each of which can be individually prefixed by the paused indication.
   All RTP payload types related to such initially paused simulcast
   stream MUST be listed in the SDP as pause/resume capable as specified
   by [RFC7728], e.g. by using the "*" wildcard format for "a=rtcp-fb".

   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 SCID value in an RtpStreamId RTCP SDES item
   [I-D.ietf-avtext-rid].

   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



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   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
   receiving SSRC owns the restriction when TMMBR/TMMBN are used for
   pause/resume signaling since the RTP receiver's SSRC in send
   direction is sometimes not yet known.

   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, either as
   different SDP formats with differing "a=rtpmap" and/or "a=fmtp"
   lines, as differently defined RTP Constraints, or both.  If this
   difference is not required, RTP duplication [RFC7104] procedures
   SHOULD be considered instead of simulcast.

6.3.  Offer/Answer Use

      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".

6.3.1.  Generating the Initial SDP Offer

   An offerer wanting to use simulcast SHALL include the "a=simulcast"
   attribute in the offer.  An offerer listing a set of receive
   simulcast streams and/or alternative formats as SCID in the offer
   MUST be prepared to receive RTP streams for any of those simulcast
   streams and/or alternative formats from the answerer.

6.3.2.  Creating the SDP Answer

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




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   An answerer that receives an offer with simulcast containing an
   "a=simulcast" attribute listing alternative SCID MAY keep all the
   alternative SCID in the answer, but it MAY also choose to remove any
   non-desirable alternative SCID in the answer.  The answerer MUST NOT
   add any alternative SCID that were not present in the offer.  The
   answerer MUST be prepared to receive any of the receive direction
   SCID alternatives, and MAY send any of the send direction
   alternatives that are kept in the answer.

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

   An RTP stream pause/resume capable answerer that receives an offer
   with RTP stream pause/resume capability MAY mark any SCID that refer
   to pause/resume capable formats as initially paused in the answer.

   An answerer that receives indication in an offer of an SCID being
   initially paused , SHOULD mark that SCID as initially paused also in
   the answer, regardless of direction, unless it has good reason for
   the SCID not being initially paused.  One such reason could for
   example be that the answerer would otherwise initially not receive
   any media of that type at all.

6.3.3.  Offerer Processing the SDP Answer

   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 SCID in a specified direction MUST NOT
   use simulcast in that direction.

   An offerer that receives an answer where some SCID alternatives are
   kept MUST be prepared to receive any of the kept send direction SCID
   alternatives, and MAY send any of the kept receive direction SCID
   alternatives.

   An offerer that receives an answer where some of the SCID 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 SCID.

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



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   An offerer with RTP stream pause/resume capability that receives an
   answer where some SCID are marked as initially paused, SHOULD
   initially pause those RTP streams regardless if they were marked as
   initially paused also in the offer, unless it has good reason for
   those RTP streams not being initially paused.  One such reason could
   for example be that the answerer would otherwise initially not
   receive any media of that type at all.

6.3.4.  Modifying the Session

   Offers and answers inside an existing session follow the rules for
   initial session negotiation, with the additional restriction that any
   SCID marked as initially paused in such offer or answer MUST already
   be paused, thus a new offer/answer MUST NOT replace use of RTP stream
   pause/resume [RFC7728] in the session.  Session modification
   restrictions in section 6.5 of "a=rid" [I-D.ietf-mmusic-rid] also
   apply.

6.4.  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 alternative 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 stream, which is aligned with the semantics of listing
   multiple formats on the "m=" line.

   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.

6.5.  Relating Simulcast Streams

   Simulcast RTP streams MUST be related on RTP level through RID
   [I-D.ietf-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



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   [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.

   RTP streams MUST only use a single alternative SCID at a time (based
   on RTP timestamps), 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.

6.6.  Signaling Examples

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

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

                Figure 3: Four-party Mixer-based Conference

6.6.1.  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:














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   v=0
   o=alice 2362969037 2362969040 IN IP4 192.0.2.156
   s=Simulcast Enabled Client
   t=0 0
   c=IN IP4 192.0.2.156
   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 send
   a=rid:2 pt=98 send
   a=rid:3 pt=97 recv
   a=simulcast:send 1;2 recv 3


                  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 "a=fmtp" and "a=imageattr" parameters
   indicates that sent simulcast streams can differ in video resolution.

   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.




















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   v=0
   o=server 823479283 1209384938 IN IP4 192.0.2.2
   s=Answer to Simulcast Enabled Client
   t=0 0
   c=IN IP4 192.0.2.43
   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 recv
   a=rid:2 pt=98 recv
   a=rid:3 pt=97 send
   a=simulcast:recv 1;2 send 3


                 Figure 5: Single-Source Simulcast Answer

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

6.6.2.  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 [RFC7741] 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
   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



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   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 [I-D.ietf-mmusic-sdp-bundle-negotiation] and RID
   [I-D.ietf-mmusic-rid] recommends using the RTP header extension
   [RFC5285] for carrying these RTP stream identification fields, which
   is consequently also included in the SDP.  Note also that for RID,
   the corresponding SDES attribute is named RtpStreamId
   [I-D.ietf-avtext-rid].










































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   v=0
   o=fred 238947129 823479223 IN IP6 2001:db8::c000:27d
   s=Offer from Simulcast Enabled Multi-Source Client
   t=0 0
   c=IN IP6 2001:db8::c000:27d
   a=group:BUNDLE foo bar zen

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

   m=video 49600 RTP/AVPF 100 101 103
   a=mid:bar
   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; \
       mst-mode=NI-TC
   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-fps=60;depend=2
   a=rid:2 send pt=101;max-width=1280;max-height=720;max-fps=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:RtpStreamId
   a=rtcp-fb:* ccm pause nowait
   a=simulcast:send 1;2;~4,3

   m=video 49602 RTP/AVPF 96 104
   a=mid:zen
   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-fps=30
   a=rid:6 send pt=96;max-fs=614400;max-fps=15
   a=rid:7 send pt=96;max-fs=230400;max-fps=30
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:RtpStreamId
   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
      support.

   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.  Limitation

   The chosen approach has a limitation that relates to the use of a
   single RTP session for all simulcast formats of a media source, which
   comes from sending all simulcast streams related to a media source
   under the same SDP media description.

   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 also 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 media-level SDP attribute,
   "simulcast", in the "att-field (media level only)" registry within
   the SDP parameters registry, according to the procedures of [RFC4566]
   and [I-D.ietf-mmusic-sdp-mux-attributes].

   Contact name, email:  IETF, contacted via mmusic@ietf.org, or a
      successor address designated by IESG

   Attribute name:  simulcast

   Long-form attribute name:  Simulcast stream description

   Charset dependent:  No

   Attribute value:  See Section 6.1 of RFC XXXX.

   Purpose:  Signals simulcast capability for a set of RTP streams

   MUX category:  NORMAL

   Note to RFC Editor: Please replace "RFC XXXX" with the assigned
   number of this RFC.






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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.

   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.ietf-avtext-rid].  There are no
   additional security concerns related to their use in this
   specification.

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

   [I-D.ietf-avtext-rid]
              Roach, A., Nandakumar, S., and P. Thatcher, "RTP Stream
              Identifier Source Description (SDES)", draft-ietf-avtext-
              rid-04 (work in progress), June 2016.








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   [I-D.ietf-mmusic-rid]
              Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
              Roach, A., and B. Campen, "RTP Payload Format
              Constraints", draft-ietf-mmusic-rid-05 (work in progress),
              March 2016.

   [I-D.ietf-mmusic-sdp-mux-attributes]
              Nandakumar, S., "A Framework for SDP Attributes when
              Multiplexing", draft-ietf-mmusic-sdp-mux-attributes-13
              (work in progress), June 2016.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [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, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <http://www.rfc-editor.org/info/rfc5109>.

   [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234,
              DOI 10.17487/RFC5234, January 2008,
              <http://www.rfc-editor.org/info/rfc5234>.

   [RFC7104]  Begen, A., Cai, Y., and H. Ou, "Duplication Grouping
              Semantics in the Session Description Protocol", RFC 7104,
              DOI 10.17487/RFC7104, January 2014,
              <http://www.rfc-editor.org/info/rfc7104>.

   [RFC7728]  Burman, B., Akram, A., Even, R., and M. Westerlund, "RTP
              Stream Pause and Resume", RFC 7728, DOI 10.17487/RFC7728,
              February 2016, <http://www.rfc-editor.org/info/rfc7728>.

   [RFC7741]  Westin, P., Lundin, H., Glover, M., Uberti, J., and F.
              Galligan, "RTP Payload Format for VP8 Video", RFC 7741,
              DOI 10.17487/RFC7741, March 2016,
              <http://www.rfc-editor.org/info/rfc7741>.




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13.2.  Informative References

   [I-D.ietf-avtcore-multiplex-guidelines]
              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.

   [I-D.ietf-avtcore-rtp-multi-stream]
              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.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-31 (work in progress), June 2016.

   [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,
              <http://www.rfc-editor.org/info/rfc2198>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <http://www.rfc-editor.org/info/rfc3264>.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389,
              September 2002, <http://www.rfc-editor.org/info/rfc3389>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,
              <http://www.rfc-editor.org/info/rfc4588>.

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              DOI 10.17487/RFC4733, December 2006,
              <http://www.rfc-editor.org/info/rfc4733>.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <http://www.rfc-editor.org/info/rfc5285>.



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   [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,
              <http://www.rfc-editor.org/info/rfc5576>.

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)",
              RFC 5583, DOI 10.17487/RFC5583, July 2009,
              <http://www.rfc-editor.org/info/rfc5583>.

   [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,
              <http://www.rfc-editor.org/info/rfc6184>.

   [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,
              <http://www.rfc-editor.org/info/rfc6190>.

   [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,
              <http://www.rfc-editor.org/info/rfc6236>.

   [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,
              <http://www.rfc-editor.org/info/rfc7656>.

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,
              <http://www.rfc-editor.org/info/rfc7667>.

Appendix A.  Changes From Earlier Versions

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

A.1.  Modifications Between WG Version -04 and -05

   o  Aligned with recent changes in draft-ietf-mmusic-rid and draft-
      ietf-avtext-rid.

   o  Modified the SDP offer/answer section to follow the generally
      accepted structure, also adding a brief text on modifying the
      session that is aligned with draft-ietf-mmusic-rid.




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   o  Improved text around simulcast stream identification (as opposed
      to the simulcast stream itself) to consistently use the acronym
      SCID and defined that in the Terminology section.

   o  Changed references for RTP-level pause/resume and VP8 payload
      format that are now published as RFC.

   o  Improved IANA registration text.

   o  Removed unused reference to draft-ietf-payload-flexible-fec-
      scheme.

   o  Editorial improvements and corrections.

A.2.  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.

A.3.  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.



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

A.4.  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
      stream.

   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.5.  Modifications Between WG Version -00 and -01

   o  No changes.  Only preventing expiry.

A.6.  Modifications Between Individual Version -00 and WG Version -00

   o  Added this appendix.

Authors' Addresses

   Bo Burman
   Ericsson
   Gronlandsgatan 31
   SE-164 80 Stockholm
   Sweden

   Email: bo.burman@ericsson.com


   Magnus Westerlund
   Ericsson
   Farogatan 2
   SE-164 80 Stockholm
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com



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   Suhas Nandakumar
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: snandaku@cisco.com


   Mo Zanaty
   Cisco
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: mzanaty@cisco.com



































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