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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: May 2007                                       Stephan Wenger

                                                     November 30, 2006

                             RTP Topologies

Status of this Memo

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Copyright Notice

   Copyright (C) The Internet Society (2006).


   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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Status of this Memo................................................1
Copyright Notice...................................................1
TABLE OF CONTENTS..................................................2
1. Introduction....................................................3
2. Definitions.....................................................3
  2.1. Glossary...................................................3
3. Topologies......................................................4
  3.1. Point to Point.............................................4
  3.2. Point to Multi-point using Multicast.......................4
  3.3. Point to Multipoint using the RFC 3550 translator..........5
  3.4. Point to Multipoint using the RFC 3550 mixer model.........8
  3.5. Point to Multipoint using video switching MCU.............10
  3.6. Point to Multipoint using RTCP-terminating MCU............11
  3.7. Combining Topologies......................................13
4. Security Considerations........................................13
5. Acknowledgements...............................................15
6. IANA Considerations............................................15
7. References.....................................................16
  7.1. Normative references......................................16
  7.2. Informative references....................................16
8. Authors' Addresses.............................................16
RFC Editor Considerations.........................................17

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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.

   When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] 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.

2.  Definitions

2.1.    Glossary

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

2.2.    Indicating Requirement leves

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
   "OPTIONAL" in this document are to be interpreted as described in
   RFC 2119 [RFC2119].

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   The RFC 2119 language is used in this document to indicate
   important requirements on solutions to fulfil issues described 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 widely deployed
   system model as used in most H.323 [H323] 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 3.7.

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

3.1.    Point to Point

   Shortcut name: Topo-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 (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.

3.2.    Point to Multi-point using Multicast

   Shortcut name: Topo-Multicast

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      +---+     /       \    +---+
      | 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 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 this sense, "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 theoretical
   limit of the number of participants in a single RTP session is
   approximately two billion).

   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".

3.3.    Point to Multipoint using the RFC 3550 translator

   Shortcut name: Topo-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

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   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 [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 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.

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   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.

   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. Thus it is the responsibility of the
   participants to run SSRC collision detection, and the SSRC a field
   the translator should not change.

      +---+      +------------+      +---+
      | 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.  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.

   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.  However such
   a behavior should only be used in corner cases such as quality
   monitoring.  It is not envisioned that such a corner-case
   translator will emit, or respond to, codec control messages.

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   It also needs to be noted that a media translator, in some cases
   may act on behalf of the "real" source and respond to codec
   control 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
   for the media translator to react to the codec control messages,
   for example by transrating through a transcoding step.  If it were
   not reacting to the codec control message, the media quality in
   the media sender's domain may suffer. This is a result of having
   the media sender adjust the quality of the media being sent to all
   other session participants.

   The codec control messages memo [CCM] has a problem comparable to
   the ones listed above, in that translators have to rewrite RTCP to
   provide a view intended for the original domain, that corresponds
   to what has occurred in the translated domain.  In this case, the
   translator needs to decide on which messages it has to act on (and
   how to act on them), and which to pass through transparently.
   None of this can be specified, as the appropriate reaction depends
   on the situation, the translator's capabilities, and the codec
   control message received.

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

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   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. 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. 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.

   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 need to provide a
   mixed view of the other domains and will commonly only forward the
   media from B and D into the multicast network using B's and D's

   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

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   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
   3.6 below.

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

   Figure 6 - RTP Mixer with only unicast links

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, today, still deployed, 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

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   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, 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 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.

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 the most commonly
   deployed topology. The content that the MCU provides to each
   participant is either:

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     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. 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 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 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.

   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
    2) There is no information about active media senders available
        in the RTP packet.  As this information is missing, receivers

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        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.

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

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, 3.5, requires considerable afterthought
   on how RTCP is dealt with. But do note that the SSRC uniquenss
   always needs to be global across the different domains.

4.  Security Considerations

   The usage of mixers and translators do have impact on security and
   the security functions used. The primary issue is that both mixers
   and translators do modify packets, thus preventing the usage 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 packets to session endpoints. If
   encryption is employed, the media translator and mixers will 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 they translate 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

   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 was not protected at all, while giving the
   impression to the session participants that they are protected
   against them.

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   In the translator case, RFC 3550 specifies that the SSRC is kept
   'intact'.  If the translator uses the same SRTP master key as the
   source it is translating, the source and translator encrypt two
   different plaintexts (the original payload and the translated
   payload) with the same keystream. Unless the translator is a no-
   op, this results in a two-time pad that can compromise the
   security of the stream. Therefore, in cases where keystream reuse
   is a danger translators MUST use unique keys for each translation.
   This is the case with counter-mode encryption but is not the case
   with CBC and some other modes.

   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 as follows.  First it might weaken
   security if the mixer/translator accepts in one domain any weaker
   algorithms or keys than used in the other domains. Therefore, care
   should be taken that appropriate (which often translate in
   similar) strong security parameters are negotiated in all domains.
   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 toward 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
   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*D keys and
   related state; wherein N is the total of number of SSRCs part of
   the joint SSRC space over all domains, and N is the total number
   of domains.

   There exist a number of different ways 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.

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5.  Acknowledgements

   The authors would like to thank Bo Burman, Umesh Chandra and Mark
   Baugher for their help in reviewing this document.

6.  IANA Considerations

   This document specifies no actions for IANA.

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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.
   [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
            K. Norrman, "The Secure Real-time Transport Protocol
            (SRTP)", RFC 3711, March 2004.
   [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 Audio-Visual Profile with
            Feedback (AVPF)", Internet Draft, Work in Progress,
            draft-ietf-avt-avpf-ccm-02.txt>, October 2006
   [H323]   ITU-T Recommendation H.323, "Packet-based multimedia
            communications systems", June 2006.

8.  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

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

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