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Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: Informational                                  Ericsson
Expires: August 6, 2012                                 February 3, 2012

                    Multi-Stream Media Conferencing


   This memo describes a multimedia multi-party conferencing
   architecture based on use of multiple Real-Time Transport Protocol
   (RTP) streams.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 6, 2012.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Use Cases  . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Point to Point . . . . . . . . . . . . . . . . . . . . . .  5
     3.2.  RTP Mixer  . . . . . . . . . . . . . . . . . . . . . . . .  5
       3.2.1.  Incompatible Codecs  . . . . . . . . . . . . . . . . .  5
       3.2.2.  Low Quality End-Point  . . . . . . . . . . . . . . . .  6
       3.2.3.  Medium Quality End-Point . . . . . . . . . . . . . . .  7
       3.2.4.  Single Channel High Quality End-Point  . . . . . . . .  9
       3.2.5.  Dual Channel High Quality End-Point  . . . . . . . . . 10
       3.2.6.  Mixer Source and Sink Selection  . . . . . . . . . . . 11
       3.2.7.  Media Composition  . . . . . . . . . . . . . . . . . . 12
     3.3.  Multicast  . . . . . . . . . . . . . . . . . . . . . . . . 14
   4.  RTP Usage  . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     4.1.  Use of SSRC and CSRC . . . . . . . . . . . . . . . . . . . 15
     4.2.  Signaling Extensions . . . . . . . . . . . . . . . . . . . 16
     4.3.  Optimizations  . . . . . . . . . . . . . . . . . . . . . . 17
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 18
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 18
   7.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
   8.  Informative References . . . . . . . . . . . . . . . . . . . . 18
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20

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1.  Introduction

   Multimedia multi-party conferencing is being improved through the use
   of multiple media streams per media type.  At the same time, the
   number of different types of end-points that are capable of
   participating in a multimedia conference increases, and so does the
   number of access types that end-points may use to connect.  This memo
   describes a number of use cases relevant to that scenario.  The use
   cases aims to use as high media quality as possible, while making as
   efficient use of available resources as possible and accommodating a
   high degree of end-point and network diversity.

2.  Requirements

   The use cases in this memo should handle diversity in end-point
   capability such as media quality, processing power, and network
   interface.  The perceived end-user media quality is in turn impacted
   by media stream bitrate and also number of streams per media type, as
   well as end-to-end delay, which should also be taken into account.
   Diversity in network capacity, network quality, and user preferences
   based on location, device, etc, should also be handled.  These
   properties should be expected to change between conference sessions
   and even within the same session.

   Different types of conference Topologies must be supported, including
   centralized, multicast, and point-to-point.  It should also be
   possible to use cascaded, centralized conferences as well as
   combinations of unicast and multicast.  The relevant RTP base
   topologies are described in [RFC5117].

   Use cases including an RTP Mixer should avoid adding delay or
   reducing quality by forwarding streams as unmodified as possible,
   with reduced processing requirements in the RTP Mixer as added

   The conference use cases should strive for continuous presence,
   presenting media from as many participants as reasonably possible.
   At the same time, it should strive to use as high quality media as
   possible from each participant.  The bandwidth used for the
   conference should at the same time be as low as possible.  Those
   three requirements are in conflict and there is no generally
   accepted, optimum trade-off.

   The number of participants in the conference shall assumed to be
   unlimited, and it may thus not be possible to create a continuous
   presence experience with media from all participants being presented
   to all other participants, and only media from a limited sub-set of

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   participants may be presented to each individual participant.  This
   is seen as the hardest conferencing use case and most other use cases
   should also be covered by that use case.  How many other
   participants' media that are presented simultaneously on a certain
   end-point is allowed to vary and may depend on a number of

3.  Use Cases

   The sub-sections below describe multi-stream conferencing use cases
   relevant to each RTP base topology.  Each use case is constructed as
   to cover as many requirements as possible and must consequently
   include a number of different end-point types.

   In the figures below, the involved end-points are for clarity drawn
   as only sending or only receiving, but may in a real scenario of
   course have the possibility to both send and receive.

   The common figure legends for the end-point capability categories
   used in all use cases are:

   D: Dual channel, high quality, for example conference room client.
      Streams in figures are denoted by '||' or '='.  The term 'channel'
      is chosen as a generic term and may apply to any media.  A channel
      is related to a single media source in the sender and a media sink
      in the receiver.  For video media, the channel source is typically
      a camera and the sink a screen or window.  For audio, the channel
      source is typically a microphone and the sink a loudspeaker.  A
      dual channel sending end-point thus has two independent media
      sources that can be sent simultaneously to a receiving end-point.
      Nothing prevents those dual channels from using other qualities
      than 'high', but that is chosen as an example in this memo to
      simplify the description.

   H: High quality, single channel, for example meeting room client.
      Streams in figures are denoted by '|' or '-'.

   M: Medium quality, single channel, for example desktop client.
      Streams in figures are denoted by ':' or '..'.

   L: Low quality, single channel, for example mobile client.  Streams
      in figures are denoted by '''.

   It is of course possible to divide end-points into more categories,
   but the chosen ones should make it possible to highlight most of the
   relevant topics.

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   Note also that the use cases are kept as separated and clean-cut as
   possible to simplify the description.  Most use cases, especially the
   RTP Mixer ones (Section 3.2), could be combined into larger, more
   complex scenarios.

3.1.  Point to Point

   The point to point case is close to trivial, since it is assumed that
   capability exchange during setup will be able to negotiate the best
   quality between the two end-points.  When the number of channels
   differ between sender and receiver, the channel aspects of
   Section 3.2.5 apply.

3.2.  RTP Mixer

   Each link between the RTP Mixer and an end-point is in principle a
   Point to Point connection (Section 3.1) as described above.  One
   major difference from a Point to Point Connection is that the RTP
   Mixer should represent and act according to some combination of the
   wishes and needs of multiple end-points on the other side of the
   Mixer.  That may include handling of conflicting or partly
   conflicting requirements, and the way to resolve those is not
   generally defined but will typically depend on RTP Mixer design,
   configuration, applied policies, or some combination.

3.2.1.  Incompatible Codecs

   This use case is the only feasible one when end-points have
   incompatible codecs, and transcoding is in that case always necessary
   and cannot be avoided.  The incompatibility is not necessarily only
   due to different codec types, but may also be caused by limited codec
   capacity, or limitations in media stream transport between the end-
   points.  This use case is trivial in the sense that the Mixer is
   assumed to always be capable of creating a dedicated media mix to
   each receiving end-point.  It may be used as an overall strategy, or
   as part of other use cases.

   A special case of transcoding is when the sender is configured to use
   scalable encoding and receivers do not support scalability.
   Transcoding of scalable streams to non-scalable streams is often a
   less complex operation than transcoding in general.

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                               | Codec 1 |
                                | Mixer |
                               /         \ Trans-
                              /           \ coded
                             v             v stream
                        +---------+   +---------+
                        | Codec 1 |   | Codec 2 |
                        +---------+   +---------+

                 Figure 1: Transcoding Incompatible Codecs

3.2.2.  Low Quality End-Point

   A low quality end-point has per definition the lowest media quality
   in the conference, and as a sender it is assumed that all other end-
   points can receive and present the media without restrictions.  Some
   of the more capable end-points will have to choose how to present the
   received media in the best way, but it can always be presented.

   As a receiver, a low quality end-point is only capable of receiving
   streams from other low quality end-points (without transcoding).

   It is conceivable that a receiver of a certain quality category (not
   only low quality) can receive higher quality streams and reduce the
   quality locally such that it is feasible for presentation, but that
   will in general waste both bandwidth and processing resources.

                               | L |
                             +-------+    +---+---+
                             | Mixer |'''>|   D   |
                             +-------+    +---+---+
                            '    '    '
                           '     '     '
                          v      v      v
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+

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                       Figure 2: Low Quality Stream

3.2.3.  Medium Quality End-Point

   Similar to above, the medium quality sender media stream is assumed
   to be possible to receive without restrictions in all but the low
   quality end-point.  For the medium quality media stream to reach also
   the low quality end-point, there are three options that are described
   in the sub-sections below.

   When receiving, a medium quality end-point is capable of receiving
   other medium quality streams, as well as low quality streams (without
   transcoding).  Transcoding

   Transcode the stream when sent towards low quality receivers, as
   described above (Section 3.2.1).  This could sometimes be feasible
   quality-wise, especially if the quality difference between the medium
   quality and the low quality streams are large, making the reduced
   medium quality stream be relatively close quality-wise to un-encoded
   low quality media.  End-to-end delay will however always suffer.

                               | M |
                             +-------+    +---+---+
                             | Mixer |...>|   D   |
                             +-------+    +---+---+
                            '    :    :
                          T'     :     :
                          v      v      v
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+
                        T: Transcoded stream

                Figure 3: Medium Quality Stream Transcoding  Simulcast

   Encode both a medium quality and a low quality stream from the same
   un-encoded source data and simulcast them.  The RTP Mixer can,
   without having to transcode, forward the low quality stream towards
   the low quality end-points, and forward the medium quality stream
   towards all other end-points.

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                               | M |
                                ' :
                                v v
                             +-------+    +---+---+
                             | Mixer |...>|   D   |
                             +-------+    +---+---+
                            '    :    :
                           '     :     :
                          v      v      v
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+

                 Figure 4: Medium Quality Stream Simulcast  Scalable Coding

   As a variant of simulcast, if it is possible to use a scalable codec,
   create a scalable stream with one low quality sub-stream and one sub-
   stream that together with the low quality sub-stream can reconstruct
   a medium quality stream.  Similar but not identical to the above, the
   RTP Mixer can, without having to transcode, forward the low quality
   sub-stream towards the low quality end-points, and forward both the
   low quality and the medium quality sub-streams (jointly describing a
   medium quality stream) to all other end-points.

                               | M |
                             +-------+    +---+---+
                             | Mixer |...>|   D   |
                            '   ':   ':
                           '    ':    ':
                          v     vv     vv
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+

                 Figure 5: Medium Quality Scalable Stream

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3.2.4.  Single Channel High Quality End-Point

   This use case is very similar to the medium quality end-point case
   (Section 3.2.3) above.  The difference is that there are now two
   different (sample) categories of end-points that cannot receive the
   high quality stream instead of one category.  Simulcast and scalable
   streams must thus be extended to three versions or three sub-streams,

   As a receiver, all streams from other end-points can be received.
   The only exception is when multiple streams from a single end-point
   are used, such as from D.  Transcoding

   Similar to Medium Quality (Section, just that the
   transcoding needs to produce two different qualities instead of one.

                               | H |
                             +-------+    +---+---+
                             | Mixer |--->|   D   |
                             +-------+    +---+---+
                            '    :    |
                          T'    T:     |
                          v      v      v
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+
                        T: Transcoded stream

                 Figure 6: High Quality Stream Transcoding  Simulcast

   Similar to Medium Quality (Section, just that three instead
   of two simulcast streams need to be sent.

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                               | H |
                               ' : |
                               v v v
                             +-------+    +---+---+
                             | Mixer |--->|   D   |
                             +-------+    +---+---+
                            '    :    |
                           '     :     |
                          v      v      v
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+

                  Figure 7: High Quality Stream Simulcast  Scalable Coding

   Similar to Medium Quality (Section, just that three instead
   of two scalable layers are used.

                               | H |
                             | Mixer |...>|   D   |
                            '   ':   ':|
                           '    ':    ':|
                          v     vv     vvv
                        +---+  +---+  +---+
                        | L |  | M |  | H |
                        +---+  +---+  +---+

                  Figure 8: High Quality Scalable Stream

3.2.5.  Dual Channel High Quality End-Point

   Again, this use case is very similar to the one above
   (Section 3.2.4).  The major difference is that this end-point is
   capable of sending and receiving dual, high quality streams where
   each stream has to be treated in a similar way to the previous

   When using multiple inter-related media, such as video with

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   corresponding audio, those media streams need not only be
   synchronized time-wise, just as for single channel end-points, but
   their spatial relation need also be established.  For example, a left
   camera with an attached microphone and a right camera with an
   attached microphone.  In general it is likely always desirable to be
   able to relate streams from a multi-channel end-point in a defined
   way, representing related sub-parts of a larger scene, both intra-
   media and inter-media.  Description and signaling of stream relations
   is a complex problem in itself, which is currently work in progress
   in CLUE WG [I-D.ietf-clue-framework] and will not be elaborated
   further in this memo.

   Another major, additional, aspect to account for is that the RTP
   Mixer needs to choose how to map dual (or multiple) streams onto a
   single stream, when forwarding towards end-points that has fewer
   receive channels than the sender.  This problem is similar to
   choosing a limited set of participants from a potentially unlimited
   set, which is described below (Section 3.2.6).

   A dual-channel (or multi-channel) receiving end-point that is
   receiving fewer simultaneous channel streams from a sending end-point
   than the maximum possible this end-point can handle, will have to
   decide which one(s) of the available receive channels should be used
   for each received stream.  This decision can also be made by the RTP
   Mixer, if it knows the concept of multi-channel clients, has
   information about how many simultaneous channels the individual
   receiver supports, and knows how those channels should be related.

   There may of course be end-points that have capability for more than
   two simultaneous channels.  It is also possible to envision end-
   points where the number of receive channels differ from the number of
   send channels.

3.2.6.  Mixer Source and Sink Selection

   When a Mixer cannot forward all available streams to each client, it
   has to choose a small set out of a potentially very large set of
   streams in the conference.  Multiple strategies are possible for that
   choice.  The Mixer may use different strategies towards different
   receivers, depending on for example their capabilities or

   Another dimension of selection exist when the conference contains
   end-points with different number of channels (multiple media streams
   of the same media type).  On the Mixer receive side, it may be
   necessary to select a few streams from a multi-stream media.  On the
   Mixer send side, it may be necessary to select to which channel in a
   multi-stream capable receiver a certain stream should be sent.

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   The choice of source may be either algorithmic (pre-configured) or
   manual (user controlled from one or more end-points).  Speech
   activity is a commonly used algorithmic measure to choose which
   participants' media streams to forward, but it is not the only
   conceivable measure.  Which and how many streams to select can either
   be based on some algorithm included in the RTP Mixer (for example the
   N most active speakers), or it can be controlled by the conference
   owner, or even by the receiving users individually.

   The figure below depicts the case where a receiving end-point
   explicitly selects streams through signaling (*) to the Mixer.  Both
   the media senders and their streams are numbered for clarity, and the
   conceptual signaling message is contained in {...}.

                          +----+  +----+  +----+
                          | L1 |  | M2 |  | S3 |
                          +----+  +----+  +----+
                              '      :      |
                               1'   2:   3|
                                 v   v   v
                     +----+ 4  +-----------+  5 +----+
                     | M4 |...>|   Mixer   |<'''| L5 |
                     +----+    +-----------+    +----+
                                   ^   |1,3,4
                            {1,3,4}*   v
                                   | H |

                    Figure 9: Receiver Stream Selection

   To be able to make an informed choice on what streams to select, the
   user at the receiving end-point will need information about which
   conference participant correspond to which stream, and possibly also
   other meta-information about the streams and sending end-points.

3.2.7.  Media Composition

   When it is desirable that the RTP Mixer selects (Section 3.2.6) and
   forwards a larger number of simultaneous streams than what the
   receiving end-point can support, the Mixer has the option to make a
   composition of multiple streams onto fewer streams, possibly to only
   a single stream.  Which and how many streams to compose is typically
   based on the selection, as described in the previous section above.

   The composition operation is basically independent from selection.
   In general, Mixer composition requires transcoding.  A Mixer
   composition use case example is depicted below.  To simplify the

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   picture, only a single receiver is included.  Also, both the media
   senders and their streams are numbered for clarity.  In the below
   example, the Mixer has chosen to compose stream 1, 3 and 4 into the
   single stream sent to the receiving end-point.

                          +----+  +----+  +----+
                          | L1 |  | M2 |  | S3 |
                          +----+  +----+  +----+
                              '      :      |
                               1'   2:   3|
                                 v   v   v
                     +----+ 4  +-----------+  5 +----+
                     | M4 |...>|   Mixer   |<'''| L5 |
                     +----+    +-----------+    +----+
                                   | H |

                       Figure 10: Mixer Composition

   An alternative to make composition in the Mixer is to let the end-
   point do local composition by sending it multiple, un-composed,
   streams.  This could avoid transcoding at the cost of sending
   multiple streams, which is depicted below, where stream 2, 3 and 5
   are sent to the receiving client for local composition, as an

                          +----+  +----+  +----+
                          | L1 |  | M2 |  | S3 |
                          +----+  +----+  +----+
                              '      :      |
                               1'   2:   3|
                                 v   v   v
                     +----+ 4  +-----------+  5 +----+
                     | M4 |...>|   Mixer   |<'''| L5 |
                     +----+    +-----------+    +----+
                                 2: 3| 5'
                                  :  |  '
                                  v  v  v
                                  |  H  |

                       Figure 11: Local Composition

   When the senders offer streams of multiple qualities, either the

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   mixer or the local composition can select and combine media of
   different qualities.  Use of multiple qualities could help optimizing
   resource utilization for transport, decoding and rendering.  In the
   figure below, one high quality (3), one medium quality (4) and two
   low qualities (1 and 2) are selected for local composition, as an

                         +----+  +----+  +----+
                         | L1 |  | M2 |  | S3 |
                         +----+  +----+  +----+
                             '    ' :    ' : |
                              1' 2'2: 3'3:3|
                           4   v  v v  v v v
                    +----+...>+-------------+  5 +----+
                    | M4 | 4  |    Mixer    |<'''| L5 |
                    +----+'''>+-------------+    +----+
                              3|  4:  1'  2'
                                 |  : '  '
                                  v v v v
                                 |   H   |

                Figure 12: Multi Quality Local Composition

   This local composition scenario can be further enhanced by the Mixer
   providing different quality streams to the receiver, based on the
   Mixer selection algorithm.  One example could be to let the Mixer
   forward the stream from the most active speaker as a high quality
   stream, and forward the less active speakers as lower quality
   streams.  To use this in the local composition, the receiving end-
   point must know the streams' different roles, which requires a stream
   role agreement between the Mixer and the receiving end-point.  In the
   figure above, this can be achieved by tagging for example the
   leftmost stream from the mixer as having the "most active" role.  The
   role agreement can be made through signaling between Mixer and
   receiving end-point.

   When the receiving end-point supports reception and presentation of
   several channels (for example has several screens), it is possible to
   combine Mixer composition with local composition of multiple un-
   composed streams by sending one or more composed streams and one or
   more un-composed streams from the Mixer.

3.3.  Multicast

   In the multicast or multi-unicast case, each media stream from a
   single sender will reach multiple receivers unmodified.  This can be

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   achieved by multicast addressing, or by multi-unicast and RTP
   Translators [RFC5117].

   This use case is similar to when an RTP Mixer is neither performing
   composition (Section 3.2.7) nor source selection (Section 3.2.6), but
   is forwarding all streams (and qualities, if more than one) to all
   receivers.  The entire load of stream and quality selection for
   presentation is put on the receiving end-point.

   In the figure below, multi-unicast through the use of an RTP
   Translator is depicted since the figure becomes clearer than with a
   full mesh multicast.  The figure illustrates non-scalable streams,
   but it is of course also possible to multicast scalable streams.

                              | H |
                              ' : |
                              v v v
                        |  Translator   |...>|   D   |
                        ' : | ' : | ' : |
                       ' : |  ' : |  ' : |
                      v v v   v v v   v v v
                      +---+   +---+   +---+
                      | L |   | M |   | H |
                      +---+   +---+   +---+

              Figure 13: Multi-Unicast of Multiple Qualities

4.  RTP Usage

   This section discusses how RTP transport [RFC3550] could be used with
   the scenarios discussed in the previous section.  It complements,
   extends and partially presents an alternative solution to what is
   described in [I-D.lennox-clue-rtp-usage].

4.1.  Use of SSRC and CSRC

   It is assumed that each RTP media stream in the use cases in the
   previous section is identified by an SSRC.  There already exist
   methods to convey information about each media stream and sending
   end-point in a conference [RFC4575].  Those methods also provide
   means to correlate that information with the stream SSRC.  This
   information could be sufficient for a receiving end-point to make
   informed media stream selection decisions (Section 3.2.6).

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   When the RTP Mixer associates a role to a stream (Section 3.2.7), for
   example "most active speaker" or "leftmost channel", it is possible
   to associate that additional property to an SSRC belonging to the
   Mixer, while also keeping the original SSRC in the RTP packet as
   CSRC.  This way, it is possible to apply special treatment to
   received streams based on their SSRC without losing the ability to
   identify the original source, using existing RTP functionality.

                          +----+  +----+  +----+
                          | L1 |  | M2 |  | S3 |
                          +----+  +----+  +----+
                         SSRC1' SSRC2: SSRC3|
                               1'   2:   3|
                          SSRC4  v   v   v  SSRC5
                     +----+ 4  +-----------+  5 +----+
                     | M4 |...>|   Mixer   |<'''| L5 |
                     +----+    +-----------+    +----+
                             SSRC6 :2 5' SSRC7
                            (CSRC2)v   v(CSRC5)
                                   | H |

                    Figure 14: SSRC and CSRC from Mixer

   It is also possible in a receiving end-point to let each role (and
   Mixer SSRC) map towards a specific media decoder, since that Mixer
   SSRC would rarely (if ever) change during a session other than due to
   SSRC conflicts, while the CSRC would typically change every time a
   new stream is selected for that specific role, for example "active

   Note that when simulcast is used, different simulcast versions can
   typically use different SSRC.  When scalable coding is used,
   different layers can sometimes be sent within a single SSRC using a
   single Payload Type and thus cannot be distinguished on RTP level.
   Identification of different layers will then have to be codec
   specific.  Some scalable codecs can also send different layers on
   separate SSRC or using separate Payload Types.

4.2.  Signaling Extensions

   End-points and Mixers supporting multiple channels (Section 3.2.5)
   need to know how many simultaneous channels that can be accepted in a
   receiver and will be used from a sender.  Assuming that each channel
   is sent as a single SSRC, there should be signaling that limits the
   number of SSRC in an RTP session [I-D.westerlund-avtcore-max-ssrc].
   This maps well with the above suggested relation between SSRC and

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   media decoders, since the suggested limitation then also expresses
   the maximum simultaneously available decoding resources.

   When representing a specific media source in several different
   qualities and when using simulcast to transport (Section
   them rather than as scalable layers contained in a single stream,
   those separate streams need to be signaled as simulcast versions
   [I-D.westerlund-avtcore-rtp-simulcast], in order for the receiver to
   be able to apply correct selection logic (Section 3.2.6).

   When a conference is configured to let individual users at receiving
   end-points choose which streams to receive (Section 3.2.6),
   responsive selection signaling between end-point and RTP Mixer
   [I-D.westerlund-dispatch-stream-selection] is needed to initiate the
   stream selection.  This selection is applicable to the media streams
   included in a compositions also.

   Note that when simulcast is used, different simulcast versions can
   typically use different SSRC.  When scalable coding is used,
   different layers can sometimes be sent within a single SSRC using a
   single Payload Type and thus cannot use the parts of SDP signaling
   that relies on those identifiers.  Identification of different layers
   will then have to be codec specific.  Some scalable codecs can also
   send different layers on separate SSRC or using separate Payload

4.3.  Optimizations

   When it is desirable to minimize the number of UDP ports used by an
   end-point, for example to reduce the resources for NAT and firewall
   traversal, it should be possible to send all media streams from all
   RTP sessions on a single UDP port
   [I-D.westerlund-avtcore-transport-multiplexing].  This should
   preferably be done without losing any important RTP functionality.
   Transport resource priority and Quality of Service handling are
   typically performed based on 5-tuple (source and destination
   addresses and ports, and protocol), which together with desired
   differentiation of media stream priority can require use of more than
   one UDP port (5-tuple).

   In a conference use case with multiple sending end-points and where
   the receiving end-points do not make use of all available streams,
   there is a risk that some of the sent streams are not used by any
   receiver.  The probability for this increases when end-points provide
   multiple streams of different qualities.  The need for a certain
   stream can change very quickly, for example when the need is based on
   conditions of other streams such as speech activity.  To save
   bandwidth and processing resources in the sending end-point, it would

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   thus be desirable for an RTP Mixer to be able to quickly turn off or
   pause individual streams [I-D.westerlund-avtext-rtp-stream-pause]
   that are no longer used in any media mix sent to receiving end-
   points, and even more importantly be able to quickly resume needed
   streams when they are needed again.

   In use cases where multiple media streams (Section 3.2.5) are used in
   a single RTP session, when SDP is used as signaling protocol, and
   specifically when the number of streams depends on the SDP
   negotiation outcome (Section 4.2), the currently defined bandwidth
   signaling attribute is only capable of describing the maximum
   possible bandwidth usage for the most demanding alternative.  It
   would be desirable to express bandwidth requirements in a more
   precise way [I-D.westerlund-mmusic-sdp-bw-attribute].

   While any RTP stream relations such as for example spatial co-
   location of related audio and video streams should be possible to
   express in session signaling or other application signaling protocol,
   there may be times when it is desirable that RTP stream SSRC
   relations [I-D.westerlund-avtext-rtcp-sdes-srcname] such as simulcast
   alternatives or related FEC streams can be seen directly in the RTP
   or RTCP streams.  This would allow for processing of media and
   related streams in middle boxes, without the need to have access to
   all higher layer signaling.  Keeping protocol layer separation will
   enable some architectural freedom and may ease future extensions.

5.  Security Considerations

   Any security considerations relevant to this memo are described in
   the RFCs and drafts referenced in the RTP Usage section (Section 4).

6.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an

7.  Acknowledgements

8.  Informative References

              Romanow, A., Duckworth, M., Pepperell, A., and B. Baldino,

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              "Framework for Telepresence Multi-Streams",
              draft-ietf-clue-framework-02 (work in progress),
              January 2012.

              Lennox, J., Romanow, A., and P. Witty, "Real-Time
              Transport Protocol (RTP) Usage for Telepresence Sessions",
              draft-lennox-clue-rtp-usage-01 (work in progress),
              October 2011.

              Westerlund, M., Burman, B., and F. Jansson, "Multiple
              Synchronization sources (SSRC) in RTP Session Signaling",
              draft-westerlund-avtcore-max-ssrc-00 (work in progress),
              October 2011.

              Westerlund, M., Burman, B., Lindqvist, M., and F. Jansson,
              "Using Simulcast in RTP sessions",
              draft-westerlund-avtcore-rtp-simulcast-00 (work in
              progress), October 2011.

              Westerlund, M. and C. Perkins, "Multiple RTP Session on a
              Single Lower-Layer Transport",
              draft-westerlund-avtcore-transport-multiplexing-01 (work
              in progress), October 2011.

              Westerlund, M., Burman, B., and P. Sandgren, "RTCP SDES
              Item SRCNAME to Label Individual Sources",
              draft-westerlund-avtext-rtcp-sdes-srcname-00 (work in
              progress), October 2011.

              Akram, A., Burman, B., Grondal, D., and M. Westerlund,
              "RTP Media Stream Pause and Resume",
              draft-westerlund-avtext-rtp-stream-pause-00 (work in
              progress), October 2011.

              Grondal, D., Burman, B., and M. Westerlund, "Media Stream
              Selection (MESS)",
              draft-westerlund-dispatch-stream-selection-00 (work in
              progress), October 2011.

              Frankkila, T., Westerlund, M., and B. Burman, "Extensible

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              Bandwidth Attribute for SDP",
              draft-westerlund-mmusic-sdp-bw-attribute-00 (work in
              progress), October 2011.

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

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session
              Initiation Protocol (SIP) Event Package for Conference
              State", RFC 4575, August 2006.

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

Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

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

   Bo Burman
   Farogatan 6
   SE-164 80 Kista

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

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