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Versions: 00 draft-ietf-mmusic-sdp-simulcast

Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: Standards Track                                Ericsson
Expires: April 30, 2015                                    S. Nandakumar
                                                               M. Zanaty
                                                                   Cisco
                                                        October 27, 2014


                Using Simulcast in SDP and RTP Sessions
                  draft-burman-mmusic-sdp-simulcast-00

Abstract

   In some application scenarios it may be desirable to send multiple
   differently encoded versions of the same media source in independent
   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 a new media level
   SDP attribute that express capability to send and/or receive
   simulcast RTP streams.  One part of the RTP/RTCP identification
   method is included as a reference to a separate document, since it is
   useful also for other purposes.

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 30, 2015.

Copyright Notice

   Copyright (c) 2014 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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   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 Adaptation in Multicast/Broadcast  . . . . . . .   6
     3.4.  Receiver Media Source Preferences . . . . . . . . . . . .   7
   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   7
   5.  Proposed Solution Overview  . . . . . . . . . . . . . . . . .   8
   6.  Proposed Solution . . . . . . . . . . . . . . . . . . . . . .   9
     6.1.  Simulcast Capability  . . . . . . . . . . . . . . . . . .   9
       6.1.1.  Declarative Use . . . . . . . . . . . . . . . . . . .  11
       6.1.2.  Offer/Answer Use  . . . . . . . . . . . . . . . . . .  11
     6.2.  Relating Simulcast Versions . . . . . . . . . . . . . . .  12
     6.3.  Signaling Examples  . . . . . . . . . . . . . . . . . . .  13
       6.3.1.  Unified Plan Client . . . . . . . . . . . . . . . . .  13
       6.3.2.  Multi-Source Client . . . . . . . . . . . . . . . . .  14
   7.  Network Aspects . . . . . . . . . . . . . . . . . . . . . . .  16
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  17
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .  17
   10. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  18
   11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  18
   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .  18
     12.1.  Normative References . . . . . . . . . . . . . . . . . .  18
     12.2.  Informative References . . . . . . . . . . . . . . . . .  18
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  20

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



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   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 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].

   The purpose of this document is to describe a few scenarios where it
   is motivated to use simulcast, and propose a suitable solution for
   SDP signaling and performing RTP simulcast.

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].

   Simulcast version:  One encoded stream from the set of encoded
      streams that constitutes the simulcast for a single media source.

   Simulcast version alternative:  One encoded stream being encoded in
      one of possibly multiple alternative ways to create a simulcast
      version.





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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.
      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.

   A multicast/broadcast case where the receivers themselves selects the
   most appropriate simulcast version and tune in to the right media
   transport to receive that version is also considered (Section 3.3) .
   This enables large, heterogeneous receiver populations, when it comes
   to capabilities and the use of network path bandwidth resources.



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3.1.  Reaching a Diverse Set of Receivers

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

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

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

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

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



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   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
   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 Adaptation in Multicast/Broadcast

   When using broadcast or multicast technology to distribute real-time
   media streams to large populations of receivers, there can still be
   significant heterogeneity among the receiver population.  This can
   depend on several factors:

   Network Bandwidth:  The network paths to individual receivers will
      have variations in the bandwidth, thus putting different limits on
      the supported bit-rates that can be received.

   Endpoint Capabilities:  The end point's hardware and software can
      have varying capabilities in relation to screen resolution,
      decoding capabilities, and supported media codecs.



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   To handle these variations, a transmitter of real-time media may want
   to apply simulcast to a media source and provide it as a set of
   different encoded streams, enabling the receivers to select the best
   fit from this set themselves.  The end point capabilities will
   usually result in a single initial choice.  However, the network
   bandwidth can vary over time, which requires a client to continuously
   monitor its reception to determine if the received RTP streams still
   fit within the available bandwidth.  If not, another set of encoded
   streams from the ones offered in the simulcast will have to be
   chosen.

   When using IP multicast, the level of granularity that the receiver
   can select from is decided by its ability to choose different
   multicast addresses.  Thus, different simulcast versions need to be
   put on different media transports using different multicast
   addresses.  If these simulcast versions are described using SDP, they
   need to be part of different SDP media descriptions, as SDP binds to
   transport on media description level.

3.4.  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 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 versions
         that can be provided.

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

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

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

   REQ-4:  Distinguishing features.  It must be possible to have
      different simulcast versions 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.  Proposed Solution Overview

   The proposed solution consists of signaling simulcast capability and
   configurations in SDP [RFC4566]:

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





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

   o  Currently, a single media source per SDP media description is
      assumed, which makes the solution work in an Unified Plan
      [I-D.roach-mmusic-unified-plan] context (although different from
      what is currently defined there), both with and without BUNDLE
      grouping.

   o  The codec configuration for each simulcast version is expressed in
      terms of existing SDP formats (and typically RTP payload types).
      Some codecs may rely on codec configuration based on general
      attributes that apply for all formats within a media description,
      and which could thus not be used to separate different simulcast
      versions.  This memo makes no attempt to address such
      shortcomings, but if needed instead encourages that a separate,
      general mechanism is defined for that purpose.

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

6.  Proposed Solution

   This section further details the signaling solution outlined 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/sendrecv), the
   simulcast attribute defines a list of simulcast versions (separated
   by semicolons), each of which is a list of alternative RTP payload
   types (separated by commas) for that simulcast version.  The meaning
   of the attribute on SDP session level is undefined and MUST NOT be
   used.  There MUST be at most one "a=simulcast" attribute per media
   description.  The ABNF [RFC5234] for this attribute is:

   simulcast-attribute = "a=simulcast" 1*3( WSP sc-dir-list )
   sc-dir-list         = sc-dir WSP sc-fmt-list *( ";" sc-fmt-list )
   sc-dir              = "send" / "recv" / "sendrecv"
   sc-fmt-list         = sc-fmt *( "," sc-fmt )
   sc-fmt              = fmt
   ; WSP defined in [RFC5234]
   ; fmt defined in [RFC4566]


                       Figure 1: ABNF for Simulcast



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   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.

   Attribute parameters are grouped by direction and consist of a
   listing of SDP format tokens (usually corresponding to RTP payload
   types), which describe the simulcast versions to be used.  The number
   of (non-alternative, see below) formats in the list sets a limit to
   the number of supported simulcast versions in that direction.  The
   order of the listed simulcast versions in the "send" direction is not
   significant.  The order of the listed simulcast versions in the
   "recv" direction expresses a preference which simulcast versions that
   are preferred, with the leftmost being most preferred, if the number
   of actually sent simulcast versions have to be reduced for some
   reason.

   Formats that have explicit dependencies [RFC5583] to other formats
   (even in the same media description) MAY be listed as different
   simulcast versions.

   Alternative simulcast versions MAY be specified as part of the
   attribute parameters by expressing each simulcast version format as a
   comma-separated list of alternative values.  In this case, all
   combinations of those alternatives MUST be supported.  The order of
   the alternatives within a simulcast version is not significant; codec
   preference is expressed by format type ordering on the m-line, using
   regular SDP rules.

   A simulcast version 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 versions of the media source, but rather a
   specific way of generating the RTP stream of a single simulcast
   version with varying RTP payload type.  Instead, only a single codec
   format MUST be used per simulcast version or simulcast version
   alternative (if there are such).  The codec format SHOULD be the
   codec 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.

   Use of the redundant audio data [RFC2198] format could be seen as a
   form of simulcast for loss protection purposes, but is not considered



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   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 version on the "a=simulcast" line.

      Editor's note: Consider adding the possibility to put an RTP
      stream in "paused" state [I-D.ietf-avtext-rtp-stream-pause] from
      the beginning of the session, possibly starting it at a later
      point in time by applying RTP/RTCP level procedures from that
      specification.

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.  The "sendrecv" direction combines "send" and
   "recv" requirements, using the same format values for both.

   If simulcast version alternatives 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
   version.

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 formats 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.  If the offered direction
   is "sendrecv", the answerer MAY keep it, but MAY also change it to
   "send" or "recv" to indicate that it is only interested in simulcast
   for a single direction.  Note that, like all other use of SDP format
   tags for the send direction in Offer/Answer, format tags related to
   the simulcast send direction in an offer ("send" or "sendrecv") are
   placeholders that refer to information in the offer SDP, and the



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   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 versions and/or
   alternatives in the offer MUST be prepared to receive RTP streams for
   any of those simulcast versions and/or alternatives from the
   answerer.

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

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

   An offerer that receives an answer where some simulcast version
   alternatives 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.  This is similar to the case
   when the answer includes multiple formats on the m-line.

   An offerer that receives an answer where some of the simulcast
   versions 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 versions.

   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.

      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".

6.2.  Relating Simulcast Versions

   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, as specified in the SDP "a=simulcast"
   attribute (Section 6.1) parameters.  When using BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation] to multiplex multiple SDP
   media descriptions over a specify a single RTP session, there is an
   identification mechanism that allows relating RTP streams back to



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   individual media descriptions, after which the above RTP payload type
   relation can be used.

6.3.  Signaling Examples

   These examples are for a case of 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.  Alice's Offer:

   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 97;98 recv 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 versions can differ in video resolution and framerate.



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

   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 97;98 send 97


                  Figure 4: Unified Plan Simulcast Answer

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

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 a Unified
   Plan client, restricted to a single media source per media
   description.

   The first two simulcast versions for the first media source use
   different codecs, H264-SVC [RFC6190] and H264 [RFC6184].  These two
   simulcast versions 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 version for the first media
   source.

   The second media source is offered with three different simulcast
   versions.  All video streams of this second media source are loss
   protected by RTP retransmission [RFC4588].





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   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.  There are not so many RTP payload types in this example
   that there is any risk of running out of payload types, but for the
   sake of making an example, it is assumed that one of the payload
   types cannot be kept unique across all media descriptions.
   Therefore, the SDP makes use of the mechanism (work in progress) in
   BUNDLE that identifies which media description an RTP stream belongs
   to (a new RTCP SDES item and RTP header extension [RFC5285] type
   carrying the a=mid value).  That identification will make it possible
   to identify unambiguously also on RTP level which media source it is
   and thus what the related simulcast versions are, even though two
   separate RTP streams in the joint RTP session share RTP payload type.

   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/AVP 100 101 102 103
   a=mid:bar
   a=rtpmap:100 H264-SVC/90000
   a=rtpmap:101 H264/90000
   a=rtpmap:102 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:102 profile-level-id=42c00d; max-fs=900; max-mbps=27000
   a=fmtp:103 max-fs=900; max-fr=30
   a=imageattr:100 send [x=1280,y=720] recv [x=1280,y=720]
   a=imageattr:101 send [x=1280,y=720] recv [x=1280,y=720]
   a=imageattr:102 send [x=640,y=360] recv [x=640,y=360]
   a=imageattr:103 send [x=640,y=360] recv [x=640,y=360]
   a=depend:100 lay bar:101
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=simulcast sendrecv 100;101 send 103,102

   m=video 49602 RTP/AVP 96 103 97 104 105 106
   a=mid:zen
   a=rtpmap:96 VP8/90000
   a=fmtp:96 max-fs=3600; max-fr=30



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   a=rtpmap:104 rtx/90000
   a=fmtp:104 apt=96;rtx-time=200
   a=rtpmap:103 VP8/90000
   a=fmtp:103 max-fs=900; max-fr=30
   a=rtpmap:105 rtx/90000
   a=fmtp:105 apt=103;rtx-time=200
   a=rtpmap:97 VP8/90000
   a=fmtp:97 max-fs=240; max-fr=15
   a=rtpmap:106 rtx/90000
   a=fmtp:106 apt=97;rtx-time=200
   a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid
   a=simulcast send 97;96;103


               Figure 5: Fred's Multi-Source Simulcast Offer

      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 version, rather than applying
      equal treatment to all versions.  For example, lower bit-rate
      versions may be prioritized over higher bit-rate versions to
      minimize congestion or packet losses in the low bit-rate versions.
      Thus, there is a benefit to use a simulcast solution that supports
      QoS as good as possible.  By separating simulcast versions into
      different RTP sessions and send those RTP sessions over different
      media transports, a simulcast version can be prioritized by
      existing flow based QoS mechanisms.  When using unicast, QoS
      mechanisms based on individual packet marking are also feasible,
      which do not require separation of simulcast versions into
      different RTP sessions to apply different QoS.  The proposed
      solution can be extended to support this functionality with an



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      optional mid: prefix before the RTP payload types of a simulcast
      version, to describe simulcast across multiple media descriptions.

   NAT/FW Traversal:  Using multiple RTP sessions will incur more cost
      for NAT/FW traversal unless they can re-use the same transport
      flow, which can be achieved by either one of multiplexing multiple
      RTP sessions on a single lower layer transport
      [I-D.westerlund-avtcore-transport-multiplexing] or Multiplexing
      Negotiation Using SDP Port Numbers
      [I-D.ietf-mmusic-sdp-bundle-negotiation].  If flow based QoS with
      any differentiation is desirable, the cost for additional
      transport flows is likely necessary.

   Multicast:  Multiple RTP sessions will be required to enable
      combining simulcast with multicast.  Different simulcast versions
      have to be separated to different multicast groups to allow a
      multicast receiver to pick the version 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.  The proposed solution can be extended to support this
      functionality with an optional mid: prefix before the RTP payload
      types of a simulcast version, to describe simulcast across
      multiple media descriptions.

8.  IANA Considerations

   This document requests to register a new attribute, simulcast.

   Formal registrations to be written.

9.  Security Considerations

   The simulcast capability and 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 versions
   irrespective of their number.  There may however be a large number of
   unwanted RTP streams that will impact the share of the bandwidth
   allocated for the originally wanted RTP stream.

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





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

10.  Contributors

   Morgan Lindqvist and Fredrik Jansson, both from Ericsson, have
   contributed with important material to the first versions of this
   document.  Robert Hansen, from Cisco, contributed significantly to
   subsequent versions.

11.  Acknowledgements

12.  References

12.1.  Normative References

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

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

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

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

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

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

12.2.  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.







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   [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-05 (work in progress),
              July 2014.

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-04 (work in progress),
              August 2014.

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
              "A Taxonomy of Grouping Semantics and Mechanisms for Real-
              Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
              rtp-grouping-taxonomy-02 (work in progress), June 2014.

   [I-D.ietf-avtext-rtp-stream-pause]
              Akram, A., Even, R., and M. Westerlund, "RTP Stream Pause
              and Resume", draft-ietf-avtext-rtp-stream-pause-04 (work
              in progress), October 2014.

   [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-12 (work in progress), October 2014.

   [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-vp8-13 (work in progress), October 2014.

   [I-D.roach-mmusic-unified-plan]
              Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for
              Using SDP with Large Numbers of Media Flows", draft-roach-
              mmusic-unified-plan-00 (work in progress), July 2013.

   [I-D.westerlund-avtcore-transport-multiplexing]
              Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
              Sessions onto a Single Lower-Layer Transport", draft-
              westerlund-avtcore-transport-multiplexing-07 (work in
              progress), October 2013.

   [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,
              September 1997.



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

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

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

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

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

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

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

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

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

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

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

Authors' Addresses










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   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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


   Bo Burman
   Ericsson
   Kistavagen 25
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 13 11
   Email: bo.burman@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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