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Network Working Group M. Westerlund Request for Comments: 5117 Ericsson Category: Informational S. Wenger

                                                          January 2008
                           RTP Topologies

Status of This Memo

 This memo provides information for the Internet community.  It does
 not specify an Internet standard of any kind.  Distribution of this
 memo is unlimited.


 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.

Westerlund & Wenger Informational [Page 1] RFC 5117 RTP Topologies January 2008

Table of Contents

 1. Introduction ....................................................2
 2. Definitions .....................................................3
    2.1. Glossary ...................................................3
    2.2. Indicating Requirement Levels ..............................3
 3. Topologies ......................................................3
    3.1. Point to Point .............................................4
    3.2. Point to Multipoint 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 MCUs ............11
    3.6. Point to Multipoint Using RTCP-Terminating MCU ............12
    3.7. Non-Symmetric Mixer/Translators ...........................13
    3.8. Combining Topologies ......................................14
 4. Comparing Topologies ...........................................15
    4.1. Topology Properties .......................................15
         4.1.1. All to All Media Transmission ......................15
         4.1.2. Transport or Media Interoperability ................16
         4.1.3. Per Domain Bit-Rate Adaptation .....................16
         4.1.4. Aggregation of Media ...............................16
         4.1.5. View of All Session Participants ...................16
         4.1.6. Loop Detection .....................................17
    4.2. Comparison of Topologies ..................................17
 5. Security Considerations ........................................17
 6. Acknowledgements ...............................................19
 7. References .....................................................19
    7.1. Normative References ......................................19
    7.2. Informative References ....................................20

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 Mixer or Translator (in RTP [RFC3550] terminology)

Westerlund & Wenger Informational [Page 2] RFC 5117 RTP Topologies January 2008

 functionality and 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

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 an
          end-to-end data transfer.
 PtM    - Point to Multipoint
 PtP    - Point to Point
 SSM    - Source-Specific Multicast
 SSRC   - Synchronization Source

2.2. Indicating Requirement Levels

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 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 interpret based on Synchronization Source
 (SSRC) values) and therefore has no full information about all the
 endpoint's situation as reported in RTCP Receiver Reports (RRs).
 More topologies can be constructed by combining any of the models;
 see Section 3.8.

Westerlund & Wenger Informational [Page 3] RFC 5117 RTP Topologies January 2008

 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 messages 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
 endpoints, 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).
          +---+         +---+
          | 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.

Westerlund & Wenger Informational [Page 4] RFC 5117 RTP Topologies January 2008

3.2. Point to Multipoint 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
    o packet loss, or
    o when 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
 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 scale to very large multicast groups (the 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 document 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]

Westerlund & Wenger Informational [Page 5] RFC 5117 RTP Topologies January 2008

 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 participant.

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 that 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 lies in how to handle different sources and their
 identities.  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 it 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.
 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, always under
 its own SSRC, using the CSRC field to indicate the source(s) of the

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 The RTCP translation process can be trivial, for example, when
 Transport Translators just need to adjust IP addresses, or they can
 be quite complex as in the case of media Translators.  See Section
 7.2 of [RFC3550].
       +---+     /       \     +------------+      +---+
       | 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 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.

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 As specified in Section 7.1 of [RFC3550], the SSRC space is common
 for all participants in the session, independent of on 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 a Translator 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.
 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.

Westerlund & Wenger Informational [Page 8] RFC 5117 RTP Topologies January 2008

 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 also translate 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, which
 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 the 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 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 CSRC Count (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.

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 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
 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 Section 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 MCUs

 Shortcut name: Topo-Video-switch-MCU
       +---+      +------------+      +---+
       | A |------| Multipoint |------| B |
       +---+      |  Control   |      +---+
                  |   Unit     |
       +---+      |   (MCU)    |      +---+
       | C |------|            |------| D |
       +---+      +------------+      +---+
    Figure 7 - Point to Multipoint Using a Video Switching 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 retained.
 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 most of 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 are not
 currently forwarded receive a view of the session that indicates

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 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 MCUs
 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,
 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 use 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 individually negotiated session for each

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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 streams from

Westerlund & Wenger Informational [Page 13] RFC 5117 RTP Topologies January 2008

 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 do 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's 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 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.  Since B and D only
 have 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

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4. Comparing Topologies

 The topologies discussed in Section 3 have different properties.
 This section first lists these properties and then maps the 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 which 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 currently least capable
 receiver, 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.

Westerlund & Wenger Informational [Page 15] RFC 5117 RTP Topologies January 2008

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

 Participants are most likely to be connected to each other with a
 heterogeneous set of paths.  This makes congestion control in a Point
 to Multipoint 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 the least-capable participant in all the domains at
 the cost (complexity, delay, equipment) of the Mixer or Translator.

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
 needing 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

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
 entirely fulfill this.  The one described in Section 3.6 does not
 support this at all.

Westerlund & Wenger Informational [Page 16] RFC 5117 RTP Topologies January 2008

4.1.6. Loop Detection

 In complex topologies with multiple interconnected domains, 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.

4.2. Comparison of Topologies

 The table below attempts to summarize the properties of the different
 topologies.  The legend to the topology abbreviations 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 table below, Y indicates Yes or full support, N
 indicates No support, (Y) indicates partial support, and N/A
 indicates not applicable.
 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 Secure Realtime
 Transport Protocol (SRTP) and Secure Realtime Transport Control
 Protocol (SRTCP) [RFC3711] packets to session endpoints.  If
 encryption is employed, the media Translator and Mixer need 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

Westerlund & Wenger Informational [Page 17] RFC 5117 RTP Topologies January 2008

 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 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
 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 could create 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 a
 weaker algorithm or key in one domain than in another.  Therefore,
 care should be taken that appropriately strong security parameters
 are negotiated in all domains.  In many cases, "appropriate"

Westerlund & Wenger Informational [Page 18] RFC 5117 RTP Topologies January 2008

 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 and 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 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 as to avoid two-time pad occurrences.  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 which mechanisms are used is outside of the scope of this

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. References

7.1. 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.

Westerlund & Wenger Informational [Page 19] RFC 5117 RTP Topologies January 2008

 [RFC3711]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
             Norrman, "The Secure Real-time Transport Protocol
             (SRTP)", RFC 3711, March 2004.
 [RFC4575]   Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "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 2006.

7.2. Informative References

 [CCM]       Wenger, S., Chandra, U., Westerlund, M., Burman, B.,
             "Codec Control Messages in the RTP Audio-Visual Profile
             with Feedback (AVPF)", Work in Progress, 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," Work in Progress, March 2007.

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

Westerlund & Wenger Informational [Page 20] RFC 5117 RTP Topologies January 2008

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Westerlund & Wenger Informational [Page 21]

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