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Versions: (draft-wenger-avt-topologies) 00 01 02 03 04 05 06 07 RFC 5117

Network Working Group                                Magnus Westerlund
INTERNET-DRAFT                                                Ericsson
Expires: February 2007                                  Stephan Wenger
                                                                 Nokia

                                                       August 28, 2006

                              RTP Topologies
                     draft-ietf-avt-topologies-00.txt>


Status of this Memo

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Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   This document disucsses multi-endpoint topologies commonly used in
   RTP based environments.  In particular, centralized topologies
   commonly employed in the video conferencing industry are mapped to
   the RTP terminology.








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TABLE OF CONTENTS

1. Introduction....................................................3
2. Definitions.....................................................3
  2.1. Glossary...................................................4
  2.2. Terminology................................................4
  2.3. Topologies.................................................5
     2.3.1. Point to Point........................................5
     2.3.2. Point to Multi-point using Multicast..................6
     2.3.3. Point to Multipoint using the RFC 3550 translator.....7
     2.3.4. Point to Multipoint using the RFC 3550 mixer model....9
     2.3.5. Point to Multipoint using video switching MCU........11
     2.3.6. Point to Multipoint using RTCP-terminating MCU.......12
     2.3.7. Combining Topologies.................................13
3. Acknowledgements...............................................13
4. References.....................................................14
  4.1. Normative references......................................14
  4.2. Informative references....................................14
5. Authors' Addresses.............................................14
6. List of Changes relative to previous drafts....................14
































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1.  Introduction

   When working on the Codec Control Messages [CCM], we noticed a
   considerable confusion in the community with respect to terms such as
   MCU, mixer, and translator.  In the process of writing, we became
   increasingly unsure of our own understanding, and therefore added
   what became the core of this draft to the CCM draft.  Later, it was
   found that this information has its own value, and was "outsourced"
   from the CCM draft into the present memo.

   It could be argued that this document clarifies and explains sections
   of the RTP spec [RFC3550], and is therefore of informational nature.
   In this case, the present memo may end up as an informational RFC.

   When the Audio-Visual Profile with Feedback (AVPF) [AVPF] was
   developed, the main emphasis lied in the efficient support of point-
   to-point and small multipoint scenarios without centralized
   multipoint control.  However, in practice, many small multipoint
   conferences operate utilizing devices known as Multipoint Control
   Units (MCUs).  MCUs comprise mixers and translators (in RTP [RFC3550]
   terminology), but also signalling support.  Long standing experience
   of the conversational video conferencing industry suggests that there
   is a need for a few additional feedback messages, to efficiently
   support MCU-based multipoint conferencing.  Some of the messages have
   applications beyond centralized multipoint, and this is indicated in
   the description of the message.

   Some of the messages defined here are forward only, in that they do
   not require an explicit acknowledgement.  Other messages require
   acknowledgement, leading to a two way communication model that could
   suggest to some to be useful for control purposes.  It is not the
   intention of this memo to open up the use of RTCP to generalized
   control protocol functionality.  All mentioned messages have
   relatively strict real-time constraints and are of transient nature,
   which make the use of more traditional control protocol means, such
   as SIP re-invites, undesirable.  Furthermore, all messages are of a
   very simple format that can be easily processed by an RTP/RTCP
   sender/receiver.  Finally, all messages infer only to the RTP stream
   they are related to, and not to any other property of a communication
   system.



2.  Definitions






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2.1.    Glossary

   ASM    - Asynchronous Multicast
   AVPF   - The Extended RTP Profile for RTCP-based Feedback
   MCU    - Multipoint Control Unit
   PtM    - Point to Multipoint
   PtP    - Point to Point


2.2.    Terminology

   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].


   Message:
          Codepoint defined by this specification, of one of the
          following types:

     Request:
             Message that requires Acknowledgement

     Acknowledgment:
             Message that answers a Request

     Command:
             Message that forces the receiver to an action

     Indication:
             Message that reports a situation

     Notification:
             See Indication.

    Note that, with the exception of "Notification", this terminology
    is in alignment with ITU-T Rec. H.245.

  Decoder Refresh Point:
           A bit string, packetised in one or more RTP packets, which
           completely resets the decoder to a known state. Typical
           examples of Decoder Refresh Points are H.261 Intra pictures
           and H.264 IDR pictures. However, there are also much more
           complex decoder refresh points.

           Typical examples for "hard" decoder refresh points are Intra
           pictures in H.261, H.263, MPEG 1, MPEG 2, and MPEG-4 part 2,
           and IDR pictures in H.264.  "Gradual" decoder refresh points
           may also be used; see for example [11].  While both "hard"
           and "gradual" decoder refresh points are acceptable in the



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           scope of this specification, in most cases the user
           experience will benefit from using a "hard" decoder refresh
           point.

           A decoder refresh point also contains all header information
           above the picture layer (or equivalent, depending on the
           video compression standard) that is conveyed in-band.  In
           H.264, for example, a decoder refresh point contains
           parameter set NAL units that generate parameter sets
           necessary for the decoding of the following slice/data
           partition NAL units (and that are not conveyed out of band).
           To the best of the author's knowledge, the term "Decoder
           Refresh Point" has been formally defined only in H.264; hence
           we are referring here to this video compression standard.

  Decoding:
           The operation of reconstructing the media stream.

  Rendering:
           The operation of presenting (parts of) the reconstructed
           media stream to the user.
  Stream thinning:
          The operation of removing some of the packets from a media
           stream.  Stream thinning, preferably, is performed in a media
           aware fashion implying that the media packets are removed in
           the order of their relevance to the reproductive quality.
           However even when employing media-aware stream thinning, most
           media streams quickly lose quality when subject to increasing
           levels of thinning.  Media-unaware stream thinning leads to
           even worse quality degradation.


2.3.    Topologies

   This subsection defines several basic topologies that are relevant
   for codec control. The first four relate to the RTP system model
   utilizing multicast and/or unicast, as envisioned in RFC 3550.  The
   last two topologies, in contrast, describe the widely deployed system
   model as used in most H.323 video conferences, where both the media
   streams and the RTCP control traffic terminate at the MCU.  More
   topologies can be constructed by combining any of the models, see
   Section 2.3.7.


2.3.1.      Point to Point

   The Point to Point (PtP) topology (Figure 1) consists of two end-
   points with unicast capabilities between them.  Both RTP and RTCP
   traffic are conveyed endpoint to endpoint using unicast traffic only




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   (even if this unicast traffic happens to be conveyed over an IP-
   multicast address).

      +---+         +---+
      | A |<------->| B |
      +---+         +---+

   Figure 1 - Point to Point

   The main property of this topology is that A sends to B and only B,
   while B sends to A and only A. This avoids all complexities of
   handling multiple endpoints and combining the requirements from them.
   Do note that an endpoint may still use multiple RTP Synchronization
   Sources (SSRCs) in an RTP session.


2.3.2.      Point to Multi-point using Multicast

                 +-----+
      +---+     /       \    +---+
      | A |----/         \---| B |
      +---+   /   Multi-  \  +---+
             +    Cast     +
      +---+   \  Network  /  +---+
      | C |----\         /---| D |
      +---+     \       /    +---+
                 +-----+

   Figure 2 - Point to Multipoint using Multicast

   We define Point to Multipoint (PtM) using multicast topology as a
   transmission model in which traffic from any participant reaches all
   the other participants, except for cases such as
     o packet loss occurs,
     o a participant participant does not wish to receive the traffic
       from a certain other participant, and therefore has not
       subscribed to the IP multicast group in question.

   In this sense, "traffic" encompasses both RTP and RTCP traffic.  The
   number of participants can be between one and many -- as RTP and RTCP
   scales to very large multicast groups (the theoretical limit of RTP
   is approximately two billion participants).

   This draft is primarily interested in the subset of multicast session
   where the number of participants in the multicast group allows the
   participants to use early or immediate feedback as defined in AVPF.
   This document refers to those groups as as "small multicast groups".






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2.3.3.      Point to Multipoint using the RFC 3550 translator

   Two main categories of Translators can be distinguished.

   Transport Translators do not modify the media stream itself, but are
   concerned with transport parameters.  Transport parameters, in the
   sense of this section, comprise the transport addresses to bridge
   different domains, and the media packetization to allow other
   transport protocols to be interconnected to a session (gateways).

   Media Translators, in contrast, modify the media stream itself.  This
   process is commonly known as transcoding.  The modification of the
   media stream can be as small as removing parts of the stream, and can
   go all the way to a full transcoding utilizing a different media
   codec.   Media translators are commonly used to connect entities
   without a common interoperability point.

   Stand-alone Media Translators are rare.  Most commonly, a combination
   of Transport and Media Translators are used to translate both the
   media stream and the transport aspects of a stream between two
   transport domains (or clouds).

   Both Translator types share common attributes that separates them
   from mixers.  For each media stream that the translator receives, it
   generates an individual stream in the other domain.  However, a
   translator maintains a complete view of all existing participants
   between both domains. Therefore, the SSRC space is shared across the
   two domains.

   The RTCP translation process can be trivial, for example when
   Transport translators just need to adjust IP addresses, and can be
   quite complex in the case of media translators.  See section 7.2 of
   [RFC 3550].


                 +-----+
      +---+     /       \     +------------+      +---+
      | A |<---/         \    |            |<---->| B |
      +---+   /   Multi-  \   |            |      +---+
             +    Cast     +->| Translator |
      +---+   \  Network  /   |            |      +---+
      | C |<---\         /    |            |<---->| D |
      +---+     \       /     +------------+      +---+
                 +-----+

   Figure 3 - Point to Multipoint using a Translator

   Figure 3 depicts an example of a Transport Translator performing at
   least IP address translation.  It allows the (non multicast capable)




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   participants B and D to take part in a multicasted session by having
   the translator forward their unicast traffic to the multicast
   addresses in use, and vice versa.  It must also forward B's traffic
   to D and vice versa, to provide each of B and D with a complete view
   of the session.

   If B were behind a limited link, the translator may perform media
   transcoding to allow the traffic received from the other participants
   to reach B without overloading the link.

   When in the example depicted in Figure 3 the translator acts only as
   a Transport Translator, then the RTCP traffic can simply be
   forwarded, similar to the media traffic.  However, when media
   translation occurs, the translator's task becomes substantially more
   complex even with respect to the RTCP traffic.  In this case, the
   translator needs to rewrite B's RTCP receiver report, before
   forwarding them to D and the multicast network.  The rewriting is
   needed as the stream received by B is not the same stream as the
   other participants receive. For example, the number of packets
   transmitted to B may be lower than what D receives, due to the
   different media format. Therefore, if the receiver reports were
   forwarded without changes, the extended highest sequence number would
   indicate that B were substantially behind in reception -- while it
   most likely it would not be. Therefore, the translator must translate
   that number to a corresponding sequence number for the stream the
   translator received.  Similar arguments can be made for most other
   fields in the RTCP receiver reports.

      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 4 - MCU with RTP Translator (relay) with only unicast links

   A common MCU scenario is the one depicted in Figure 4 - MCU with RTP
   Translator (relay) with only unicast links.  Herein, the MCU connects
   multiple users of a conference through unicast. This can be
   implemented using a very simple transport translator, which could be
   called a relay. The relay forwards all traffic it receives, both RTP
   and RTCP, to all other participants. In doing so, a multicast network
   is emulated without relying on a multicast capable network structure.








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2.3.4.      Point to Multipoint using the RFC 3550 mixer model

   A mixer is a middlebox that aggregates multiple RTP streams that are
   part of a session, by mixing the media data and generating a new RTP
   stream.  One common application for a mixer is to allow a participant
   to receive a session with a reduced amount of resources.

                 +-----+
      +---+     /       \     +-----------+      +---+
      | A |<---/         \    |           |<---->| B |
      +---+   /   Multi-  \   |           |      +---+
             +    Cast     +->|   Mixer   |
      +---+   \  Network  /   |           |      +---+
      | C |<---\         /    |           |<---->| D |
      +---+     \       /     +-----------+      +---+
                 +-----+

   Figure 5 - Point to Multipoint using RFC 3550 mixer model

   A mixer can be viewed as a device terminating the media streams
   received from other session participants.  Using the media data from
   the received media streams, a mixer generates a media stream that is
   sent to the session participant.

   The content that the mixer provides is the mixed aggregate of what
   the mixer receives from the PtP or PtM links, which are part of the
   same conference session.

   The mixer is the content source, as it mixes the content (often in
   the uncompressed domain) and then encodes it for transmission to a
   participant. The CC and CSRC fields in the RTP header are used to
   indicate the contributors of to the newly generated stream.  The
   SSRCs of the to-be-mixed streams on the mixer input appear as the
   CSRCs at the mixer output.  That output stream uses a new SSRC that
   identifies the Mixer.  The CSRC are forwarded between the two domains
   to allow for loop detection and identification of sources that are
   part of the global session.

   The mixer is responsible for generating RTCP packets in accordance
   with its role. It is a receiver and should therefore send reception
   reports for the media streams it receives. As a media sender itself
   it should also generate sender report for those media streams sent.
   The content of the SRs created by the mixer may or may not take into
   account the situation on its receiving side.  Similarly, the content
   of RRs created by the mixer may or may not be based on the situation
   on the mixer's sending side.  This is left open to the
   implementation.  As specified in Section 7.3 of RFC 3550, a mixer
   must not forward RTCP unaltered between the two domains.





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   The mixer depicted in Figure 5 has three domains that needs to be
   separated; the multicast network, participant B and participant D.
   The Mixer produces different mixed streams to B and D, as the one to
   B may contain D and vice versa. However the mixer does only need one
   SSRC in each domain that is the receiving entity and transmitter of
   mixed content.

   In the multicast domain the mixer does not provide a mixed view of
   the other domains and only forwards media from B and D into the
   multicast network using B's and D's SSRC.

   The mixer is responsible for receiving the codec control messages and
   handles them appropriately.  The definition of "appropriate" depends
   on the message itself and the context. In some cases, the reception
   of a codec control message may result in the generation and
   transmission of codec control messages by the mixer to the
   participants in the other domain. In other cases, a message is
   handled by the mixer itself and therefore not forwarded to any other
   domains.

   It should be noted that this form of mixing technology is not widely
   deployed.  Most multipoint video conferences used today employ one of
   the models discussed in the next sections.

   When replacing the multicast network in Figure 5 (to the left of the
   mixer) with individual unicast links as depicted in Figure 6, the
   mixer model is very similar to the one discussed in section 2.3.6
   below.

      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 6 - RTP Mixer with only unicast links















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2.3.5.      Point to Multipoint using video switching MCU

      +---+      +------------+      +---+
      | A |------| Multipoint |------| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |------|            |------| D |
      +---+      +------------+      +---+

   Figure 7 - Point to Multipoint using relaying MCU

   This PtM topology is, today, perhaps the most widely deployed one.
   It reflects today's lack of wide deployment of IP multicast
   technologies on IP networks and the Internet, as well as the
   simplicity of content switching when compared to content mixing.  The
   technology is commonly implemented in what is known as "Video
   Switching MCUs".

   A video switch MCU forwards to a participant a single media stream,
   selected from the available streams.  The criteria for selection are
   often based on voice activity in the audio-visual conference, but
   other conference management mechanisms (like explicit floor control)
   are known to exist as well.

   The video switching MCU may also perform media translation to modify
   the content in bit-rate, encoding, resolution; however it still
   indicates the original sender of the content through the SSRC.  The
   values of the CC and CSRC fields are retained.

   RTCP Sender Reports are forwarded for the currently selected sender.
   All RTCP receiver reports are freely forward between the
   participants. In addition, the MCU may also originate RTCP control
   traffic in order to control the session and/or report on status from
   its viewpoint.

   The video switching MCU has mostly the attributes of a translator.
   However its stream selection is a mixing behaviour. This behaviour
   has some RTP and RTCP issues associated with it. The suppression of
   all but one media stream results in that most participants see only a
   subset of the sent media streams at any given time; often a single
   stream per conference. Therefore, RTCP receiver reports only report
   on these streams.  In consequence, the media senders that are not
   currently forwarded receive a view of the session that indicates
   their media streams disappearing somewhere en route. This makes the
   use of RTCP for congestion control very problematic. To avoid these
   issues the MCU needs to modify the RTCP RRs.






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2.3.6.      Point to Multipoint using RTCP-terminating MCU

      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 8 - Point to Multipoint using content modifying MCU

   In this PtM scenario, each participant runs an RTP point-to-point
   session between itself and the MCU. The content that the MCU provides
   to each participant is either:

     a) A selection of the content received from the other participants,
        or

     b) The mixed aggregate of what the MCU receives from the other PtP
        links, which are part of the same conference session.

   In case a) the MCU may modify the content in bit-rate, encoding,
   resolution. No explicit RTP mechanism is used to establish the
   relationship between the original media sender and the version the
   MCU sends.  In other words, the outgoing session typically uses a
   different SSRC, and may well use a different PT, even if this
   different PT happens to be mapped to the same media type.  (This is
   the definition of this topology and distinguishes it from the
   topologies previously discussed).

   In case b) the MCU is the content source as it mixes the content and
   then encodes it for transmission to a participant. The participant's
   content that is included in the aggregated content is not indicated
   through any explicit RTP mechanism.  For example, regardless of the
   number of streams that are aggregated, in the MCU generated streams
   CC is zero and therefore no CSRC fields are present.

   The MCU is responsible for receiving the codec control messages and
   handle them appropriately. In some cases, the reception of a codec
   control message may result in the generation and transmission of
   codec control messages by the MCU to some or all of the other
   participants.

   An MCU may transparently relay some codec control messages and
   intercept, modify, and (when appropriate) generate codec control
   messages of its own and transmit them to the media senders.






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   The main feature that sets this topology apart from what RFC 3550
   describes, is the lack of an explicit RTP level indication of all
   participants. If one were using the mechanisms available in RTP and
   RTCP to signal this explicitly, the topology would follow the
   approach of an RTP mixer. The lack of explicit indication has at
   least the following potential problems:

    1) Loop detection cannot be performed on the RTP level.  When
        carelessly connecting two misconfigured MCUs, a loop could be
        generated.
    2) There is no information about active media senders available in
        the RTP packet.  As this information is missing, receivers
        cannot use it.  It also deprive the participant's clients
        information about who are actively sending in a machine usable
        way. Thus preventing clients from doing indication of currently
        active speakers in user interfaces, etc.



2.3.7.      Combining Topologies

   Topologies can be combined and linked to each other using mixers or
   translators. Care must however be taken to how the SSRC space is
   handled, mixers separate the SSRC space into two parts, while
   translators maintain the space across themselves. Any hybrid, like
   the video switching MCU, 2.3.5, requires considerable afterthought on
   how RTCP is dealt with.


3.  Acknowledgements

   The authors would like to thank N.N.





















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4.  References


4.1.    Normative references

   [AVPF]   draft-ietf-avt-rtcp-feedback-11.txt
   [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.
    [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            July 2003.


4.2.    Informative references


   Any 3GPP document can be downloaded from the 3GPP web server,
   "http://www.3gpp.org/", see specifications.


5.  Authors' Addresses

   Magnus Westerlund
   Ericsson Research
   Ericsson AB
   SE-164 80 Stockholm, SWEDEN

   Phone: +46 8 7190000
   EMail: magnus.westerlund@ericsson.com


   Stephan Wenger
   Nokia Corporation
   P.O. Box 100
   FIN-33721 Tampere
   FINLAND

   Phone: +358-50-486-0637
   EMail: stewe@stewe.org




6.  List of Changes relative to previous drafts






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INTERNET-DRAFT               RTP Topologies           August 28, 2006






RFC Editor Considerations

   The RFC editor is requested to replace all occurrences of XXXX with
   the RFC number this document receives.













































Wenger, et al.               Informational                  [Page 16]


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