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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: April 21, 2016                                    S. Nandakumar
                                                               M. Zanaty
                                                                   Cisco
                                                        October 19, 2015


                Using Simulcast in SDP and RTP Sessions
                   draft-ietf-mmusic-sdp-simulcast-03

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 discusses the
   best way of accomplishing simulcast in RTP and how to signal it in
   SDP.  A solution is defined by making an extension to SDP, and using
   RTP/RTCP identification methods to relate RTP streams belonging to
   the same media source.  The SDP extension consists of a new media
   level SDP attribute that expresses capability to send and/or receive
   simulcast RTP streams.  RTP/RTCP identification using either payload
   types or a separately defined method for RTP stream configuration are
   defined.

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 April 21, 2016.

Copyright Notice

   Copyright (c) 2015 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 Capability  . . . . . . . . . . . . . . . . . .   9
       6.1.1.  Declarative Use . . . . . . . . . . . . . . . . . . .  11
       6.1.2.  Offer/Answer Use  . . . . . . . . . . . . . . . . . .  12
     6.2.  Relating Simulcast Streams  . . . . . . . . . . . . . . .  14
     6.3.  Signaling Examples  . . . . . . . . . . . . . . . . . . .  14
       6.3.1.  Unified Plan Client . . . . . . . . . . . . . . . . .  14
       6.3.2.  Multi-Source Client . . . . . . . . . . . . . . . . .  16
   7.  Network Aspects . . . . . . . . . . . . . . . . . . . . . . .  18
   8.  Limitations . . . . . . . . . . . . . . . . . . . . . . . . .  18
     8.1.  Single RTP Session  . . . . . . . . . . . . . . . . . . .  18
     8.2.  SDP Format Identification . . . . . . . . . . . . . . . .  19
     8.3.  RID Identification  . . . . . . . . . . . . . . . . . . .  19
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  20
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  20
   11. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  20
   12. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  20
   13. References  . . . . . . . . . . . . . . . . . . . . . . . . .  20
     13.1.  Normative References . . . . . . . . . . . . . . . . . .  20
     13.2.  Informative References . . . . . . . . . . . . . . . . .  21
   Appendix A.  Changes From Earlier Versions  . . . . . . . . . . .  23
     A.1.  Modifications Between WG Version -02 and  -03 . . . . . .  23
     A.2.  Modifications Between WG Version -01 and  -02 . . . . . .  24
     A.3.  Modifications Between WG Version -00 and  -01 . . . . . .  24
     A.4.  Modifications Between Individual Version -00 and WG
           Version -00 . . . . . . . . . . . . . . . . . . . . . . .  24



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   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  24

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.  From an RTP perspective, simulcast is a specific application
   of the aspects discussed in RTP Multiplexing Guidelines
   [I-D.ietf-avtcore-multiplex-guidelines].

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

2.  Definitions

2.1.  Terminology

   This document makes use of the terminology defined in RTP Taxonomy
   [I-D.ietf-avtext-rtp-grouping-taxonomy], RTP Topology [RFC5117] and
   RTP Topologies Update [I-D.ietf-avtcore-rtp-topologies-update].  In
   addition, the following terms are used:

   RTP Mixer:  An RTP middle node, defined in [RFC5117] (Section 3.4:
      Topo-Mixer), further elaborated and extended with other topologies
      in [I-D.ietf-avtcore-rtp-topologies-update] (Section 3.6 to 3.9).

   RTP Switch:  A common short term for the terms "switching RTP mixer",
      "source projecting middlebox", and "video switching MCU" as
      discussed in [I-D.ietf-avtcore-rtp-topologies-update].




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   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
      [I-D.ietf-avtext-rtp-grouping-taxonomy].  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",
   "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 targets 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:

   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.



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      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.

   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



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   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 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.

   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



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   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:

   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:

      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.




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      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 "Unified Plan" 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
      streams 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
      [I-D.ietf-avtext-rtp-grouping-taxonomy] 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 can be expressed in
      two alternative ways, with complementing drawbacks and benefits:





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      *  Through existing SDP formats (corresponding to RTP payload
         types), enabling the use of simulcast with a minimum set of
         additions to existing SDP specifications.

      *  Through use of a separately specified RTP-level identification
         mechanism [I-D.pthatcher-mmusic-rid], which complements and
         effectively extends the available simulcast stream
         identification and configuration possibilities provided by
         using 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).

6.1.  Simulcast Capability

   Simulcast capability is expressed as a new media level SDP attribute,
   "a=simulcast".  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 ABNF [RFC5234] for this
   attribute is:

sc-attr     = "a=simulcast:" 1*2( WSP sc-str-list ) [WSP sc-pause-list]
sc-str-list = sc-dir WSP sc-id-type "=" sc-alt-list *( ";" sc-alt-list )
sc-pause-list = "paused=" sc-alt-list
sc-dir      = "send" / "recv"
sc-id-type  = "pt" / "rid" / token
sc-alt-list = sc-id *( "," sc-id )
sc-id       = fmt / rid-identifier / token
; WSP defined in [RFC5234]
; fmt, token defined in [RFC4566]
; rid-identifier defined in [I-D.pthatcher-mmusic-rid]


                       Figure 1: ABNF for Simulcast

   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.

   Two simulcast stream identification methods are defined; "pt" using
   RTP payload type (SDP format), and "rid" using an additional RTP-
   level identification mechanism [I-D.pthatcher-mmusic-rid].  Different



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   identification methods MUST NOT be used for different directions on a
   single "a=simulcast" line.  Implementations that support both
   identification methods MAY include one "a=simulcast" line for each
   identification method for the same "m="-line.  Multiple "a=simulcast"
   lines with the same identification method MUST NOT be used for a
   single "m="-line.

   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.pthatcher-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,
   there MUST NOT be any capability restriction in what alternative
   formats can be used across different simulcast streams, like
   requiring all simulcast streams to use the same codec format
   alternative.  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, 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



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   simulcast format (if there are such) in the SDP.  The used simulcast
   stream identification SHOULD be the codec format most relevant to the
   media description, if possible to identify, for example the audio
   codec rather than the DTMF.  What codec format to choose in the case
   of switching between multiple equally "important" formats is left
   open, but it is assumed that in the presence of such strong relation
   it does not matter which is chosen.

   If RTP stream pause/resume [I-D.ietf-avtext-rtp-stream-pause] is
   supported, the optional "paused=" parameter MAY be used in
   conjunction with "rid" simulcast stream identification to specify
   that a certain 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".  Initially paused simulcast streams MUST NOT be used
   with "pt" identification.  Initially paused simulcast streams are
   resumed as described by the RTP pause/resume specification.

   An initially paused simulcast stream in "send" direction MUST be
   considered equivalent to an unsolicited locally paused stream, and be
   handled accordingly.

   An initially paused simulcast stream in "recv" direction 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 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.

6.1.1.  Declarative Use

   When used as a declarative media description, a=simulcast "recv"
   direction formats indicates 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 "send" direction
   requests the end point to send a specified set of RTP streams as
   simulcast streams.



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   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
   stream.

      Editor's note: The RID identification mechanism currently lacks a
      declarative use definition.  As declarative use may also not
      follow unified plan with a single media source per '"m="-line, it
      is uncertain if declarative can be defined for the mechanism in
      its current shape.

6.1.2.  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" not listing a
   direction or 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
   directionality of the unidirectional direction parameters; "send"
   becomes "recv" and vice versa, and include it in the answer.  Note
   that, like all other use of SDP format tags ("pt:") for the send
   direction in Offer/Answer, format tags related to the simulcast
   stream identification send direction in an offer are placeholders
   that refer to information in the offer SDP, and the actual formats
   that will be used on the wire (including RTP Payload Format numbers)
   depends on information included in the SDP 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.





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   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
   answer.

   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
   streams 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 offerer that is capable of using both simulcast stream
   identification methods MAY include one "a=simulcast" line per
   identification method in the offer.  Note that it is in general not
   expected that the "pt" identification method will provide feature
   parity with the "rid" method, and the different "a=simulcast" lines
   can therefore express different use of simulcast functionality.
   However, for some configurations the different identification methods
   can be equivalent.

   An answerer receiving an offer listing both simulcast stream
   identification methods MUST choose only one and remove the other from
   the answer.  An answerer not supporting a simulcast stream
   identification method in the offer MUST remove the non-supported
   "a=simulcast" line from the answer, possibly falling back to not
   using simulcast at all.

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




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      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".

6.2.  Relating Simulcast Streams

   As long as there is only a single media source per SDP media
   description, simulcast RTP streams can be related on RTP level
   through the RTP payload type and (optionally) RID
   [I-D.pthatcher-mmusic-rid], as specified in the SDP "a=simulcast"
   attribute (Section 6.1) parameters.  When using BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation] with multiple SDP media
   descriptions to specify a single RTP session, there is an
   identification mechanism that allows relating RTP streams back to
   individual media descriptions, after which the above RTP payload type
   and RID relations can be used.

   BUNDLE's MID is an RTCP source description (SDES) item.  To ensure
   rapid initial reception, required to correctly process the RTP
   streams, it is also defined as an RTP header extension [RFC5285].

6.3.  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 2: Four-party Mixer-based Conference

6.3.1.  Unified Plan Client

   Alice is calling in to the mixer with a simulcast-enabled Unified
   Plan 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].  Media formats (RTP payload
   types) are used as simulcast stream identification.  Alice's Offer:







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   v=0
   o=alice 2362969037 2362969040 IN IP4 192.0.2.156
   s=Simulcast Enabled Unified Plan 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=simulcast: send pt=97;98 recv pt=97


                  Figure 3: Unified Plan 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.

   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.

   v=0
   o=server 823479283 1209384938 IN IP4 192.0.2.2
   s=Answer to Simulcast Enabled Unified Plan 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=simulcast: recv pt=97;98 send pt=97


                  Figure 4: Unified Plan Simulcast Answer




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

6.3.2.  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.  RID is used as simulcast stream identification, reducing the
   number of media formats needed.  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].  RID is used as simulcast
   stream identification.  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, 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.

















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   v=0
   o=fred 238947129 823479223 IN IP4 192.0.2.125
   s=Offer from Simulcast Enabled Multi-Source Client
   t=0 0
   c=IN IP4 192.0.2.125
   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-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:rid
   a=rtcp-fb:* ccm pause nowait
   a=simulcast: send rid=1;2;4,3 paused=2,3,4

   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-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:rid
   a=rtcp-fb:* ccm pause nowait
   a=simulcast: send rid=5;6;7 paused=6,7


               Figure 5: 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 that supports
      QoS as good as possible.

   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].

8.  Limitations

   The chosen approach has a few limitations that are described in this
   section.  Some relate to the use of a single RTP session for all
   simulcast formats of a media source, while others relate to the two
   different simulcast stream identification methods.

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
   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



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   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.

8.2.  SDP Format Identification

   The limitations in this section come from and thus apply only when
   using SDP format (RTP payload type) as simulcast stream
   identification method.

   The available RTP payload type number space may not be sufficient
   when many different media formats and/or simulcast streams are used
   in the SDP.  This can be particularly prominent when BUNDLE is used,
   and for any technology that adds to the number of required RTP
   payload types in a multiplicative way, such as for example adding RTP
   retransmission [RFC4588] and Forward Error Correction [RFC5109].
   Flexible FEC Scheme [I-D.ietf-payload-flexible-fec-scheme] can be
   used for RTP retransmissions and would avoid the double consumption
   of the PT space that RTP Retransmission [RFC4588] causes.

   Only existing SDP attributes and parameters can be used to define
   codec configuration for a simulcast format.  Any codec that does not
   define a sufficient set of codec parameters in "a=fmtp", or can make
   use of other SDP attributes, may not be capable of expressing the
   desired simulcast format dimensions (Section 3.1) with necessary
   precision, or not at all.  One example of this is the ability to
   separate simulcast formats by bandwidth for codecs lacking a codec-
   specific bandwidth parameter, since the SDP "b="-line covers all RTP
   payload types listed on an "m="-line.

   A simulcast stream signaled as initially paused is not possible to
   resume by a remote peer, because it cannot know which target SSRC to
   use in the RESUME message [I-D.ietf-avtext-rtp-stream-pause].

8.3.  RID Identification

   The limitations in this section come from and thus apply only when
   using RID as simulcast stream identification method.

   Use of the additional "a=rid"-line in SDP and the corresponding RID
   RTCP SDES item and RTP header extension requires some additional




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   implementation complexity, and incurs some extra bandwidth cost to
   carry the RID RTCP SDES item and RTP header extension.

9.  IANA Considerations

   This document requests to register a new SDP attribute, simulcast.

   Formal registrations to be written.

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.

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, and Peter
   Thatcher, from Google, contributed significantly to subsequent
   versions.

12.  Acknowledgements

13.  References

13.1.  Normative References

   [I-D.ietf-avtext-rtp-stream-pause]
              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.




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   [I-D.pthatcher-mmusic-rid]
              Thatcher, P., Zanaty, M., Nandakumar, S., Burman, B.,
              Roach, A., and B. Campen, "RTP Payload Format
              Constraints", draft-pthatcher-mmusic-rid-02 (work in
              progress), October 2015.

   [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>.

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, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-09 (work in progress),
              September 2015.




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   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-10 (work in progress),
              July 2015.

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, "A Taxonomy of Semantics and Mechanisms for
              Real-Time Transport Protocol (RTP) Sources", draft-ietf-
              avtext-rtp-grouping-taxonomy-08 (work in progress), July
              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-23 (work in progress), July 2015.

   [I-D.ietf-payload-flexible-fec-scheme]
              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.

   [I-D.ietf-payload-vp8]
              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,
              <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>.



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   [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>.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              DOI 10.17487/RFC5117, January 2008,
              <http://www.rfc-editor.org/info/rfc5117>.

   [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>.

   [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>.

Appendix A.  Changes From Earlier Versions

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

A.1.  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.



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   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.2.  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.3.  Modifications Between WG Version -00 and -01

   o  No changes.  Only preventing expiry.

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

   o  Added this appendix.

Authors' Addresses

   Bo Burman
   Ericsson
   Kistavagen 25
   SE-164 80 Stockholm
   Sweden

   Email: bo.burman@ericsson.com






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   Magnus Westerlund
   Ericsson
   Farogatan 2
   SE-164 80 Stockholm
   Sweden

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


   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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