Network Working Group                                Magnus Westerlund
INTERNET-DRAFT                                                Ericsson
Expires: April 2008                                     Stephan Wenger
Intended Status: Informational                                   Nokia

                                                      26 October, 2007

                              RTP Topologies

Status of this Memo

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Copyright Notice

   Copyright (C) The IETF Trust (2007).


   This document discusses multi-endpoint topologies used in Real-time
   Transport Protocol (RTP)-based environments.  In particular,
   centralized topologies commonly employed in the video conferencing
   industry are mapped to the RTP terminology.

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1. Introduction....................................................3
2. Definitions.....................................................3
   2.1. Glossary...................................................3
   2.2. Indicating Requirement leves...............................3
3. Topologies......................................................4
   3.1. Point to Point.............................................4
   3.2. Point to Multi-point using Multicast.......................5
   3.3. Point to Multipoint using the RFC 3550 translator..........6
   3.4. Point to Multipoint using the RFC 3550 mixer model.........9
   3.5. Point to Multipoint using video switching MCU.............12
   3.6. Point to Multipoint using RTCP-terminating MCU............13
   3.7. Non-Symmetric Mixer/Translators...........................14
   3.8. Combining Topologies......................................15
4. Comparing Topologies...........................................15
   4.1. Topology Proporties.......................................16
      4.1.1. All to All media transmission........................16
      4.1.2. Transport or Media Interoperability..................16
      4.1.3. Per Domain Bit-rate Adaptation.......................16
      4.1.4. Aggregation of Media.................................17
      4.1.5. View of all session participants.....................17
      4.1.6. Loop Detection.......................................17
   4.2. Comparision of topologies.................................18
5. Security Considerations........................................18
6. Acknowledgements...............................................20
7. IANA Considerations............................................20
8. References.....................................................21
   8.1. Normative References......................................21
   8.2. Informative References....................................21
9. Authors' Addresses.............................................22

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1. Introduction

   When working on the Codec Control Messages [CCM] considerable
   confusion was noticed in the community with respect to terms such
   as Multipoint Control Unit (MCU), mixer, and translator, and their
   usage in various topologies.  This document tries to address this
   confusion by providing a common information basis for future
   discussion and specification work. It attempts to clarify and
   explain sections of the Real-time Transport Protocol (RTP) spec
   [RFC3550] in an informal way. It is not intended to update or
   change what is normatively specified within RFC 3550.

   When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was
   developed the main emphasis lay 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 may implement mixers and translators (in RTP
   [RFC3550] terminology), but also signalling support. They may also
   contain additional application functionality.  This document
   focuses on the media transport aspects of the MCU that can be
   realized using RTP, as discussed below. Further considered are the
   properties of mixers and translators, and how some types of
   deployed MCUs deviate from these properties.

2. Definitions

2.1. Glossary

   ASM    - Any Source Multicast
   AVPF   - The Extended RTP Profile for RTCP-based Feedback
   CSRC   - Contributing Source
   Link   - The data transport to the next IP hop
   MCU    - Multipoint Control Unit
   Path   - The concatenation of multiple links, resulting in a end-
             to-end data transfer.
   PtM    - Point to Multipoint
   PtP    - Point to Point
   SSM    - Source-Specific Multicast
   SSRC   - Synchronization Source

2.2. Indicating Requirement leves

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL

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   "OPTIONAL" in this document are to be interpreted as described in
   RFC 2119 [RFC2119].

   The RFC 2119 language is used in this document to highlight those
   important requirements and/or resulting solutions that are
   necessary to address the issues raised in this document.

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 deployed system
   models as used in many H.323 [H323] video conferences, where both
   the media streams and the RTP Control Protocol (RTCP) control
   traffic terminate at the MCU.  In these two cases, the media sender
   does not receive the (unmodified or translator-modified) receiver
   reports from all sources (which it needs to interprete based on
   Synchronization Source (SSRC) values), and therefore has no full
   information about all the endpoint's situation as reported in RTCP-
   Receiver Reports (RR)s. More topologies can be constructed by
   combining any of the models; see Section 3.8.

   The topologies may be referenced in other documents by a shortcut
   name, indicated by the prefix "Topo-".

   For each of the RTP defined topologies, we discuss how RTP, RTCP,
   and the carried media are handled.  With respect to RTCP, we also
   introduce the handling of RTCP feedback message as defined in
   [RFC4585] and [CCM]. Any important differences between the two will
   be illuminated in the discussion.

3.1. Point to Point

   Shortcut name: Topo-Point-to-Point

   The Point to Point (PtP) topology (Figure 1) consists of two end-
   points, communicating using unicast.  Both RTP and RTCP traffic are
   conveyed endpoint-to-endpoint, using unicast traffic only (even if-
   --in exotic cases---this unicast traffic happens to be conveyed
   over an IP-multicast address).

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      +---+         +---+
      | 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.  Note that an endpoint can still use multiple RTP
   Synchronization Sources (SSRCs) in an RTP session.

   RTCP feedback messages for the indicated SSRCs are communicated
   directly between the endpoints. Therefore, this topology poses
   minimal (if any) issues for any feedback messages.

3.2. Point to Multi-point using Multicast

   Shortcut name: Topo-Multicast

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

   Figure 2 - Point to Multipoint using Multicast

   Point to Multipoint (PtM) is defined here as using a 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, or
      o a participant does not wish to receive the traffic for a
        specific multicast group, and therefore has not subscribed to
        the IP multicast group in question.  This is for the cases
        where a multi-media session is distributed using two or more
        multicast groups.

   In the above context, "traffic" encompasses both RTP and RTCP
   traffic.  The number of participants can vary between one and many-
   --as RTP and RTCP scales to very large multicast groups (the

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   theoretical limit of the number of participants in a single RTP
   session is approximately two billion).  The above can be realized
   using Any Source Multicast (ASM).  Source-Specific Multicast (SSM)
   may be also be used with RTP. However, then only the designated
   source may reach all receivers.  Please review [RTCP-SSM] for how
   RTCP can be made to work in combination with SSM.

   This draft is primarily interested in that subset of multicast
   sessions wherein the number of participants in the multicast group
   is so low that it allows the participants to use early or immediate
   feedback, as defined in AVPF [RFC4585].  This document refers to
   those groups as "small multicast groups".

   RTCP feedback messages in multicast will, like media, reach
   everyone (subject to packet losses and multicast group
   subscription). Therefore, the feedback suppression mechanism
   discussed in [RFC4585] is required. Each individual node needs to
   process every feedback message it receives, to determine if it is
   affected, or if the feedback message applies only to some other

3.3. Point to Multipoint using the RFC 3550 translator

   Shortcut name: Topo-Translator

   Two main categories of Translators can be distinguished.

   Transport Translators (Topo-Trn-Translator) 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 (in gateways).  Of the transport
   translators this memo is primarily interested in those which use
   RTP on both sides, and this is assumed henceforth.  Translators
   that bridge between different protocol worlds need to be concerned
   about the mapping of the SSRC/CSRC (Contributing Source) concept to
   the non-RTP protocol. When designing a translator to a non-RTP-
   based media transport one crucial factor consists in how to handle
   different sources and their identity.  This problem space is not
   discussed henceforth.

   Media Translators (Topo-Media-Translator), 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 (down to the sample level or equivalent) utilizing a
   different media codec.   Media translators are commonly used to
   connect entities without a common interoperability point.

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   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 separate them
   from mixers.  For each media stream that the translator receives,
   it generates an individual stream in the other domain.  A
   translator always keeps the SSRC for a stream across the
   translation, where a mixer can select a media stream, or send them
   out mixed, allways under its own SSRC, using the CSRC field to
   indicate the source(s) of the content.

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

      +---+     /       \     +------------+      +---+
      | 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) participants B and D to take part in a multicast 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 network path, the translator may perform
   media transcoding to allow the traffic received from the other
   participants to reach B without overloading the path.

   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

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   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.

   As specified in Section 7.1 of [RFC3550], the SSRC space is common
   for all participants in the session, independent of which side they
   are of the translator. Therefore, it is the responsibility of the
   participants to run SSRC collision detection, and the SSRC is a
   field the translator should not change.

      +---+      +------------+      +---+
      | A |<---->|            |<---->| B |
      +---+      |            |      +---+
                 | Translator |
      +---+      |            |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 4 - RTP Translator (relay) with only unicast paths

   Another translator scenario is depicted in Figure 4.  Herein, the
   translator connects multiple users of a conference through unicast.
   This can be implemented using a very simple transport translator,
   which in this document is 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 infrastructure.

   A translator normally does not use an SSRC of its own, and is not
   visible as an active participant in the session. One exception can
   be conceived when it acts as a quality monitor that sends RTCP
   reports, and therefore is required to have an SSRC.  Another
   example is the case when a translator is prepared to use RTCP
   feedback messages. This may, for example, occur when it suffers
   packet loss of important video packets and wants to trigger repair
   by the media sender, by sending feedback messages.  To be able to
   do this it needs to have a unique SSRC.

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   A media translator may in some cases act on behalf of the "real"
   source and respond to RTCP feedback messages.  This may occur, for
   example, when a receiver requests a bandwidth reduction and the
   media translator has not detected any congestion or other reasons
   for bandwidth reduction between the media source and itself.  In
   that case, it is sensible that the media translator reacts to the
   codec control messages itself, for example by transcoding to a
   lower media rate.  If it were not reacting, the media quality in
   the media sender's domain may suffer, as a result of the media
   sender adjusting its media rate (and quality) according to the
   needs of the slow past-translator endpoint, at the expense of the
   rate and quality of all other session participants.

   In general, a translator implementation should consider which RTCP
   feedback messages or codec control messages it needs to understand
   in relation to the functionality of the translator itself. This is
   completely in line with the requirement to translate also RTCP
   messages between the domains.
3.4. Point to Multipoint using the RFC 3550 mixer model

   Shortcut name: Topo-Mixer

   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

      +---+     /       \     +-----------+      +---+
      | 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.

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   The content that the mixer provides is the mixed aggregate of what
   the mixer receives over the PtP or PtM paths, 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 unique SSRC
   that identifies the Mixer's stream.  The CSRC are forwarded between
   the two domains to allow for loop detection and identification of
   sources that are part of the global session. Note that Section 7.1
   of RFC 3550 requires the SSRC space to be shared between domains
   for these reasons.

   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. In its role as a media
   sender, it should also generate sender reports for those media
   streams sent.  As specified in Section 7.3 of RFC 3550, a mixer
   must not forward RTCP unaltered between the two domains.

   The mixer depicted in

   Figure 5 is involved in three domains that need 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 content received from D and vice versa. However, the mixer
   only needs one SSRC in each domain that is the receiving entity and
   transmitter of mixed content.

   In the multicast domain, a mixer still needs to provide a mixed
   view of the other domains. This makes the mixer simpler to
   implement and avoids any issues with advanced RTCP handling or loop
   detection, which would be problematic if the mixer were providing
   non-symmetric behavior. Please see Section 3.7 for more discussion
   on this topic.

   A mixer is responsible for receiving RTCP feedback messages and
   handling 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 RTCP feedback 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 domain.

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   When replacing the multicast network in

   Figure 5 (to the left of the mixer) with individual unicast paths
   as depicted in Figure 6, the mixer model is very similar to the one
   discussed in section 3.6 below. Please see the discussion in 3.6
   about the differences between these two models.

      +---+      +------------+      +---+
      | A |<---->|            |<---->| B |
      +---+      |            |      +---+
                 |   Mixer    |
      +---+      |            |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 6 - RTP Mixer with only unicast paths

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3.5. Point to Multipoint using video switching MCU

   Shortcut name: Topo-Video-switch-MCU

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

   Figure 7 - Point to Multipoint using relaying MCU

   This PtM topology is still deployed today, although the RTCP-
   terminating MCUs, as discussed in the next section, are perhaps
   more common.  This topology, as well as the following one, reflect
   today's lack of wide availability of IP multicast technologies, 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 switching 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
   presentation mode or explicit floor control) are known to exist as

   The video switching MCU may also perform media translation, to
   modify the content in bit-rate, encoding, or resolution.  However
   it still may indicate the original sender of the content through
   the SSRC.  In this case the values of the CC and CSRC fields are

   If not terminating RTP, the RTCP Sender Reports are forwarded for
   the currently selected sender. All RTCP receiver reports are freely
   forwarded 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 behavior. This behavior
   has some RTP and RTCP issues associated with it.  The suppression
   of all but one media stream results in most participants seeing
   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.  Consequently, the media senders that

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   are not currently forwarded receive a view of the session that
   indicates their media streams disappear somewhere en route.  This
   makes the use of RTCP for congestion control or any type of quality
   reporting very problematic.

   To avoid the aforementioned issues, the MCU needs to implement two
   features. First it needs to act as a mixer (see section 3.4) and
   forward the selected media stream under its own SSRC and with the
   appropriate CSRC values. Second, the MCU needs to modify the RTCP
   RRs it forwards between the domains.  As a result, it is
   RECOMMENDED that one implement a centralized video switching
   conference using a Mixer according to RFC 3550, instead of the
   shortcut implementation described here.

3.6. Point to Multipoint using RTCP-terminating MCU

   Shortcut name: Topo-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.  This is a very commonly
   deployed topology in multipoint video conferencing. 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 paths, which are part of the same conference session.

   In case a) the MCU may modify the content in bit-rate, encoding, or
   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 sessions typically uses a
   different SSRC, and may well use a different payload type (PT),
   even if this different PT happens to be mapped to the same media
   type. This is a result of the session to each participant is
   negotiated individually.

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   In case b) the MCU is the content source as it mixes the content
   and then encodes it for transmission to a participant. According to
   RTP [RFC3550], the SSRC of the contributors are to be signalled
   using the CSRC/CC mechanism.  In practice, today, most deployed
   MCUs do not implement this feature.  Instead, the identification of
   the participants whose content is included in the mixer's output is
   not indicated through any explicit RTP mechanism.  That is, most
   deployed MCUs set the CSRC Count (CC) field in the RTP header to
   zero, thereby indicating no available CSRC information, even if
   they could identify the content sources as suggested in RTP.

   The main feature that sets this topology apart from what RFC 3550
   describes is the breaking of the common RTP session across the
   centralized device, such as the MCU. This results in the loss of
   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
     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 deprives the client of information
        related to currently active senders in a machine-usable way,
        thus preventing clients from indicating currently active
        speakers in user interfaces, etc.

   Note, that deployed MCUs (and endpoints) rely on signalling layer
   mechanisms for the identification of the contributing sources --
   for example, a SIP conferencing package [RFC4575].  This alleviates
   to some extent the aforementioned issues resulting from ignoring
   RTP's CSRC mechanism.

   As a result of the shortcomings of this topology it is RECOMMENDED
   to instead implement the Mixer concept as specified by RFC 3550.

3.7. Non-Symmetric Mixer/Translators

   Shortcut name: Topo-Asymmetric

   It is theoretically possible to construct an MCU that is a mixer in
   one direction and a translator in another. The main reason to
   consider this would be to allow topologies similar to Figure 5,
   where the mixer does not need to mix in the direction from B or D
   towards the multicast domains with A and C. Instead, the media

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   streams from B and D are forwarded without changes. Avoiding this
   mixing would save media processing resources that perform the
   mixing in cases where it isn't needed. However, there would still
   be a need to mix B's stream towards D. Only in the direction B ->
   multicast domain or D -> multicast domain would it be possible to
   work as a translator. In all other directions it would function as
   a mixer.

   The mixer/translator would still need to process and change the
   RTCP before forwarding it in the directions of B or D to the
   multicast domain. One issue is that A and C does not know about the
   mixed media stream the mixer sends to either B or D.  Thus, any
   reports related to these streams must be removed.  Also receiver
   reports related to A and C media stream would be missing.  To avoid
   A and C thinking that B and D aren't receiving A and C at all, the
   mixer needs to insert its receiver reports for the streams from A
   and C into B's and D's Sender Reports.  In the opposite direction
   the receiver reports from A and C about B's and D's stream also
   need to be aggregated into the mixer's receiver reports sent to B
   and D. This as B and D only has the mixer as source for the stream,
   all RTCP from A and C must be suppressed by the mixer.

   This topology is so problematic and it is so easy to get the RTCP
   processing wrong, that it is NOT RECOMMENDED to implement this

3.8. Combining Topologies

   Topologies can be combined and linked to each other using mixers or
   translators. However, care must be taken in handling the SSRC/CSRC
   space. A mixer will not forward RTCP from sources in other domains,
   but will instead generate its own RTCP packets for each domain it
   mixes into, including the necessary Source Description (SDES)
   information for both the CSRCs and the SSRCs. Thus, in a mixed
   domain the only SSRCs seen will be the ones present in the domain,
   while there can be CSRCs from all the domains connected together
   with a combination of mixers and translators. The combined SSRC and
   CSRC space is common over any translator or mixer. This is
   important to facilitate loop detection -- something that is likely
   to be even more important in combined topologies due to the mixed
   behavior between the domains. Any hybrid, like the Topo-Video-
   switch-MCU or Topo-Asymmetric, requires considerable thought on how
   RTCP is dealt with.

4. Comparing Topologies

   The topologies discussed in section 3 have different properties.
   This section first lists these properties and then maps the

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   different topologies to them.  Please note that even if a certain
   property is supported within a particular topology concept, the
   necessary functionality may in many cases be optional to implement.

4.1. Topology Properties

4.1.1. All to All media transmission

   Multicast, at least Any Source Multicast (ASM), provides the
   functionality that everyone may send to, or receive from, everyone
   else within the session. MCUs, Mixers and Translators may all
   provide that functionality at least on some basic level. However
   there are some differences in what type of reachability they

   The transport translator function called "relay" in Section 3.3 is
   the one that provides the emulation of ASM that is closest to true
   IP-multicast-based, all-to-all transmission.  Media Translators,
   Mixers and the MCU variants do not provide a fully meshed
   forwarding on the transport level, instead they only allow limited
   forwarding of content from the other session participants.

   The "all to all media transmission" requires that any media
   transmitting entity considers the path to the least capable
   receiver. Otherwise, the media transmissions may overload that
   path. Therefore, a media sender needs to monitor the path from
   itself to any of the participants, to detect the least capable
   receiver at this time instance, and adapt its sending rate
   accordingly. As multiple participants may send simultaneously, the
   available resources may vary. RTCP's Receiver Reports help
   performing this monitoring, at least on a medium time scale.

   The transmission of RTCP automatically adapts to any changes in the
   number of participants due to the transmission algorithm defined in
   the RTP specification [RFC3550], and the extensions in AVPF
   [RFC4585] (when applicable). That way, the resources utilized for
   RTCP stay within the bounds configured for the session.

4.1.2. Transport or Media Interoperability

   Translators, Mixers and RTCP-terminating MCU all allow changing the
   media encoding or the transport to other properties of the other
   domain, thereby providing extended interoperability in cases where
   the participants lack a common set of media codecs and/or transport

4.1.3. Per Domain Bit-rate Adaptation

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   Participants are most likely to be connected to each other with a
   heterogenous set of paths. This makes congestion control in a point
   to multi-point set problematic. For the ASM and "relay" scenario,
   each individual sender has to adapt to the receiver with the least
   capable path. This is no longer necessary when Media Translators,
   Mixers or MCUs are involved, as each participant only needs to
   adapt to the slowest path within its own domain. The Translator,
   Mixer or MCU topologies all require their respective outgoing
   streams to adjust the bit-rate, packet rate, etc, to adapt to the
   least capable path in each of the other domains.  That way one can
   avoid lowering the quality to least capable participant in all the
   domains, at the cost (complexity, delay, equipment) of the Mixer or

4.1.4. Aggregation of Media

   In the all-to-all-media property mentioned above and provided by
   ASM, all simultaneous media transmissions share the available bit-
   rate. For participants with limited reception capabilities this may
   result in a situation where even a minimal acceptable media quality
   cannot be accomplished. This is the result of multiple media
   streams need to share the available resources. The solution to this
   problem is to provide for a mixer or MCU to aggregate the multiple
   streams into a single one. This aggregation can be performed
   according to different methods.   Mixing or selection are two
   common methods.

4.1.5. View of all session participants

   The RTP protocol includes functionality to identify the session
   participants through the use of the SSRC and CSRC fields. In
   addition, it is capable of carrying some further identity
   information about these participants using the RTCP Source
   Descriptors (SDES). To maintain this functionality, it is necessary
   that RTCP is handled correctly in domain bridging function. This is
   specified for translators and mixers. The MCU described in Section
   3.5 does not fully fulfill this. The one described in Section 3.6
   does not support this at all.

4.1.6. Loop Detection

   In complex topologies with multiple domains interconnected, it is
   possible to form media loops. RTP and RTCP support detecting such
   loops, as long as the SSRC and CSRC identities are correctly set in
   forwarded packets. It is likely that loop detection works for the
   MCU described in Section 3.5, at least as long as it forwards the
   RTCP between the participants. However, the MCU in section 3.6 will
   definitely break the loop detection mechanism.

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4.2. Comparision of topologies

   The table below attempts to summarize the properties of the
   different topologies. The legend to the topology abbrevations are:
   Topo-Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-
   Translator (TTrn), Topo-Media-Translator (including Transport
   Translator) (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY),
   Topo-Video-switch-MCU  (MCUs), and Topo-RTCP-terminating-MCU
   (MCUt). In the below table Y indicates Y or full support, N
   indicates No support, (Y) indicates partial support. N/A is not

   Property               PtP  Multic TTrn MTrn Mixer ASY MCUs MCUt
   All to All media        N    Y      Y    Y   (Y)   (Y) (Y)  (Y)
   Interoperability        N/A  N      Y    Y    Y     Y   N    Y
   Per Domain Adaptation   N/A  N      N    Y    Y     Y   N    Y
   Aggregation of media    N    N      N    N    Y    (Y)  Y    Y
   Full Session View       Y    Y      Y    Y    Y     Y  (Y)   N
   Loop Detection          Y    Y      Y    Y    Y     Y  (Y)   N

   Please note that the Media Translator also includes the transport
   translator functionality.

5. Security Considerations

   The use of mixers and translators has impact on security and the
   security functions used. The primary issue is that both mixers and
   translators modify packets, thus preventing the use of integrity
   and source authentication, unless they are trusted devices that
   take part in the security context, e.g. the device can send SRTP
   and SRTCP [RFC3711] packets to session endpoints. If encryption is
   employed, the media translator and mixer needs to be able to
   decrypt the media to perform its function. A transport translator
   may be used without access to the encrypted payload in cases where
   it translates parts that are not included in the encryption and
   integrity protection, for example, IP address and UDP port numbers
   in a media stream using SRTP [RFC3711]. However, in general the
   translator or mixer needs to be part of the signalling context and
   get the necessary security associations (e.g. SRTP crypto contexts)
   established with its RTP session participants.

   Including the mixer and translator in the security context allows
   the entity, if subverted or misbehaving, to perform a number of
   very serious attacks as it has full access. It can perform all the
   attacks possible---see RFC 3550 and any applicable profiles---as if

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   the media session were not protected at all, while giving the
   impression to the session participants that they are protected.

   Transport translators have no interactions with cryptography that
   works above the transport layer, such as SRTP, since that sort of
   translator leaves the RTP header and payload unaltered.  Media
   translators, on the other hand, have strong interactions with
   cryptography, since they alter the RTP payload.  A media translator
   in a session that uses cryptographic protection needs to perform
   cryptographic processing to both inbound and outbound packets.

   A media translator may need to use different cryptographic keys for
   the inbound and outbound processing.  For SRTP, different keys are
   required, because an RFC 3550 media translator leaves the SSRC
   unchanged during its packet processing, and SRTP key sharing is
   only allowed when distinct SSRCs can be used to protect distinct
   packet streams.

   When the media translator uses different keys to process inbound
   and outbound packets, each session participant needs to be provided
   with the appropriate key, depending on whether they are listening
   to the translator or the original source.  (Note that there is an
   architectural difference between RTP media translation, in which
   participants can rely on the RTP Payload Type field of a packet to
   determine appropriate processing, and cryptographically protected
   media translation, in which participants must use information that
   is not carried in the packet.)

   When using security mechanisms with translators and mixers, it is
   possible that the translator or mixer creates different security
   associations for the different domains they are working in. Doing
   so has some implications.

   First, it might weaken security if the mixer/translator accepts in
   one domain a weaker algorithm or key than in another. Therefore,
   care should be taken that appropriatly strong security parameters
   are negotiated in all domains.  In many cases, "appropriate"
   translates to "similar" strength. If a key management system does
   allow the negotiation of security parameters resulting in a
   different strength of the security, then this system SHOULD notify
   the participants in the other domains about this.

   Second, the number of crypto contexts (keys, security related
   state) needed (for example in SRTP [RFC3711]) may vary between
   mixers and translators. A mixer normally needs to represent only a
   single SSRC per domain, and therefore needs to create only one
   security association (SRTP crypto context) per domain.  In
   contrast, a translator needs one security association per
   participant it translates towards, in the opposite domain.
   Considering Figure 3, the translator needs two security

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   associations towards the multicast domain, one for B and one for D.
   It may be forced to maintain a set of totally independent security
   associations between itself and B and D respectively, so to avoid
   two-time pad. These contexts must also be capable of handling all
   the sources present in the other domains. Hence, using completely
   independent security associations (for certain keying mechanisms)
   may force a translator to handle N*DM keys and related state; where
   N is the total number of SSRCs used over all domains and DM is the
   total number of domains.

   There exist a number of different mechanisms to provide keys to the
   different participants.  One example is the choice between group
   keys and unique keys per SSRC. The appropriate keying model is
   impacted by the topologies one intends to use. The final security
   properties are dependent on both the topologies in use and the
   keying mechanisms' properties, and need to be considered by the
   application. Exactly what mechanisms are used is outside of the
   scope of this document.

6. Acknowledgements

   The authors would like to thank Bo Burman, Umesh Chandra, Roni
   Even, Keith Lantz, Ladan Gharai, Geoff Hunt and Mark Baugher for
   their help in reviewing this document.

7. IANA Considerations

   This document specifies no actions for IANA.

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8. References

8.1. 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.
   [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
             Norrman, "The Secure Real-time Transport Protocol (SRTP)",
             RFC 3711, March 2004.
   [RFC4575] J. Rosenberg, H. Schulzrinne, O. Levin, "A Session
             Initiation Protocol (SIP) Event Package for Conference
             State", RFC 4575, August 2006
   [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
             "Extended RTP Profile for Real-time Transport Control
             Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July

8.2. Informative References

   [CCM]    Wenger, S., Chandra, U., Westerlund, M., Burman, B.,
             "Codec Control Messages in the Audio-Visual Profile with
             Feedback (AVPF)", Internet Draft, Work in Progress,
             draft-ietf-avt-avpf-ccm-08.txt>, July 2007
   [H323]   ITU-T Recommendation H.323, "Packet-based multimedia
             communications systems", June 2006.
   [RTCP-SSM]J. Ott, J. Chesterfield, E. Schooler, "RTCP Extensions
             for Single-Source Multicast Sessions with Unicast
             Feedback," draft-ietf-avt-rtcpssm-13, work in progress,
             March 2007.

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9. Authors' Addresses

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

   Phone: +46 8 7190000

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

   Phone: +358-50-486-0637

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