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RFC 7667

 
 
 

RTP Topologies

Part 2 of 3, p. 17 to 39
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3.4.  Point to Multipoint Using Mesh

   Shortcut name: Topo-Mesh

                             +---+      +---+
                             | A |<---->| B |
                             +---+      +---+
                               ^         ^
                                \       /
                                 \     /
                                  v   v
                                  +---+
                                  | C |
                                  +---+

                 Figure 8: Point to Multipoint Using Mesh

   Based on the RTP session definition, it is clearly possible to have a
   joint RTP session involving three or more endpoints over multiple
   unicast transport flows, like the joint three-endpoint session
   depicted above.  In this case, A needs to send its RTP streams and
   RTCP packets to both B and C over their respective transport flows.
   As long as all endpoints do the same, everyone will have a joint view
   of the RTP session.

   This topology does not create any additional requirements beyond the
   need to have multiple transport flows associated with a single RTP
   session.  Note that an endpoint may use a single local port to
   receive all these transport flows (in which case the sending port, IP
   address, or SSRC can be used to demultiplex), or it might have
   separate local reception ports for each of the endpoints.

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         +-A--------------------+
         |+---+                 |
         ||CAM|                 |                 +-B-----------+
         |+---+     +-UDP1------|                 |-UDP1------+ |
         |  |       | +-RTP1----|                 |-RTP1----+ | |
         |  V       | | +-Video-|                 |-Video-+ | | |
         |+----+    | | |       |<----------------|BV1    | | | |
         ||ENC |----+-+-+--->AV1|---------------->|       | | | |
         |+----+    | | +-------|                 |-------+ | | |
         |  |       | +---------|                 |---------+ | |
         |  |       +-----------|                 |-----------+ |
         |  |                   |                 +-------------+
         |  |                   |
         |  |                   |                 +-C-----------+
         |  |       +-UDP2------|                 |-UDP2------+ |
         |  |       | +-RTP1----|                 |-RTP1----+ | |
         |  |       | | +-Video-|                 |-Video-+ | | |
         |  +-------+-+-+--->AV1|---------------->|       | | | |
         |          | | |       |<----------------|CV1    | | | |
         |          | | +-------|                 |-------+ | | |
         |          | +---------|                 |---------+ | |
         |          +-----------|                 |-----------+ |
         +----------------------+                 +-------------+

          Figure 9: A Multi-Unicast Mesh with a Joint RTP Session

   Figure 9 depicts endpoint A's view of using a common RTP session when
   establishing the mesh as shown in Figure 8.  There is only one RTP
   session (RTP1) but two transport flows (UDP1 and UDP2).  The Media
   Source (CAM) is encoded and transmitted over the SSRC (AV1) across
   both transport layers.  However, as this is a joint RTP session, the
   two streams must be the same.  Thus, a congestion control adaptation
   needed for the paths A to B and A to C needs to use the most
   restricting path's properties.

   An alternative structure for establishing the above topology is to
   use independent RTP sessions between each pair of peers, i.e., three
   different RTP sessions.  In some scenarios, the same RTP stream may
   be sent from the transmitting endpoint; however, it also supports
   local adaptation taking place in one or more of the RTP streams,
   rendering them non-identical.

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          +-A----------------------+              +-B-----------+
          |+---+                   |              |             |
          ||MIC|       +-UDP1------|              |-UDP1------+ |
          |+---+       | +-RTP1----|              |-RTP1----+ | |
          | |  +----+  | | +-Audio-|              |-Audio-+ | | |
          | +->|ENC1|--+-+-+--->AA1|------------->|       | | | |
          | |  +----+  | | |       |<-------------|BA1    | | | |
          | |          | | +-------|              |-------+ | | |
          | |          | +---------|              |---------+ | |
          | |          +-----------|              |-----------+ |
          | |          ------------|              |-------------|
          | |                      |              |-------------+
          | |                      |
          | |                      |              +-C-----------+
          | |                      |              |             |
          | |          +-UDP2------|              |-UDP2------+ |
          | |          | +-RTP2----|              |-RTP2----+ | |
          | |  +----+  | | +-Audio-|              |-Audio-+ | | |
          | +->|ENC2|--+-+-+--->AA2|------------->|       | | | |
          |    +----+  | | |       |<-------------|CA1    | | | |
          |            | | +-------|              |-------+ | | |
          |            | +---------|              |---------+ | |
          |            +-----------|              |-----------+ |
          +------------------------+              +-------------+

      Figure 10: A Multi-Unicast Mesh with an Independent RTP Session

   Let's review the topology when independent RTP sessions are used from
   A's perspective in Figure 10 by considering both how the media is
   handled and how the RTP sessions are set up in Figure 10.  A's
   microphone is captured and the audio is fed into two different
   encoder instances, each with a different independent RTP session,
   i.e., RTP1 and RTP2, respectively.  The SSRCs (AA1 and AA2) in each
   RTP session are completely independent, and the media bitrate
   produced by the encoders can also be tuned differently to address any
   congestion control requirements differing for the paths A to B
   compared to A to C.

   From a topologies viewpoint, an important difference exists in the
   behavior around RTCP.  First, when a single RTP session spans all
   three endpoints A, B, and C, and their connecting RTP streams, a
   common RTCP bandwidth is calculated and used for this single joint
   session.  In contrast, when there are multiple independent RTP
   sessions, each RTP session has its local RTCP bandwidth allocation.

   Further, when multiple sessions are used, endpoints not directly
   involved in a session do not have any awareness of the conditions in
   those sessions.  For example, in the case of the three-endpoint

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   configuration in Figure 8, endpoint A has no awareness of the
   conditions occurring in the session between endpoints B and C
   (whereas if a single RTP session were used, it would have such
   awareness).

   Loop detection is also affected.  With independent RTP sessions, the
   SSRC/CSRC cannot be used to determine when an endpoint receives its
   own media stream, or a mixed media stream including its own media
   stream (a condition known as a loop).  The identification of loops
   and, in most cases, their avoidance, has to be achieved by other
   means, for example, through signaling or the use of an RTP external
   namespace binding SSRC/CSRC among any communicating RTP sessions in
   the mesh.

3.5.  Point to Multipoint Using the RFC 3550 Translator

   This section discusses some additional usages related to point to
   multipoint of translators compared to the point-to-point cases in
   Section 3.2.1.

3.5.1.  Relay - Transport Translator

   Shortcut name: Topo-PtM-Trn-Translator

   This section discusses Transport Translator-only usages to enable
   multipoint sessions.

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

              Figure 11: Point to Multipoint Using Multicast

   Figure 11 depicts an example of a Transport Translator performing at
   least IP address translation.  It allows the (non-multicast-capable)
   endpoints B and D to take part in an Any-Source Multicast session
   involving endpoints A and C, 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
   both B and D with a complete view of the session.

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

         Figure 12: RTP Translator (Relay) with Only Unicast Paths

   Another translator scenario is depicted in Figure 12.  The translator
   in this case connects multiple endpoints 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 endpoints.  In doing so, a
   multicast network is emulated without relying on a multicast-capable
   network infrastructure.

   For RTCP feedback, this results in a similar set of considerations to
   those described in the ASM RTP topology.  It also puts some
   additional signaling requirements onto the session establishment; for
   example, a common configuration of RTP payload types is required.

   Transport Translators and relays should always consider implementing
   source address filtering, to prevent attackers from using the
   listening ports on the translator to inject traffic.  The translator
   can, however, go one step further, especially if explicit SSRC
   signaling is used, to prevent endpoints from sending SSRCs other than
   its own (that are, for example, used by other participants in the
   session).  This can improve the security properties of the session,
   despite the use of group keys that on a cryptographic level allows
   anyone to impersonate another in the same RTP session.

   A translator that doesn't change the RTP/RTCP packet content can be
   operated without requiring it to have access to the security contexts
   used to protect the RTP/RTCP traffic between the participants.

3.5.2.  Media Translator

   In the context of multipoint communications, a Media Translator is
   not providing new mechanisms to establish a multipoint session.  It
   is more of an enabler, or facilitator, that ensures a given endpoint
   or a defined subset of endpoints can participate in the session.

   If endpoint B in Figure 11 were behind a limited network path, the
   translator may perform media transcoding to allow the traffic
   received from the other endpoints to reach B without overloading the
   path.  This transcoding can help the other endpoints in the multicast

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   part of the session, by not requiring the quality transmitted by A to
   be lowered to the bitrates that B is actually capable of receiving
   (and vice versa).

3.6.  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 generating one or more new RTP streams and, in
   most cases, by manipulating the media data.  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 13: Point to Multipoint Using the RFC 3550 Mixer Model

   A mixer can be viewed as a device terminating the RTP streams
   received from other endpoints in the same RTP session.  Using the
   media data carried in the received RTP streams, a mixer generates
   derived RTP streams that are sent to the receiving endpoints.

   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 Communication Session.

   The mixer creates the Media Source and the source RTP stream just
   like an endpoint, as it mixes the content (often in the uncompressed
   domain) and then encodes and packetizes it for transmission to a
   receiving endpoint.  The CSRC Count (CC) and CSRC fields in the RTP
   header can be used to indicate the contributors to the newly
   generated RTP 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 should be forwarded between the different endpoints to allow for
   loop detection and identification of sources that are part of the
   Communication Session.  Note that Section 7.1 of RFC 3550 requires

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   the SSRC space to be shared between domains for these reasons.  This
   also implies that any SDES information normally needs to be forwarded
   across the mixer.

   The mixer is responsible for generating RTCP packets in accordance
   with its role.  It is an RTP receiver and should therefore send RTCP
   receiver reports for the RTP streams it receives and terminates.  In
   its role as an RTP sender, it should also generate RTCP sender
   reports for those RTP streams it sends.  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 13 is involved in three domains that
   need to be separated: the Any-Source Multicast network (including
   endpoints A and C), endpoint B, and endpoint D.  Assuming all four
   endpoints in the conference are interested in receiving content from
   all other endpoints, the mixer produces different mixed RTP streams
   for B and D, as the one to B may contain content received from D, and
   vice versa.  However, the mixer may only need one SSRC per media type
   in each domain where it 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 3.11 for more discussion on this topic.
   The mixing operation, however, in each domain could potentially be
   different.

   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 by the mixer may result in the generation
   and transmission of RTCP feedback messages by the mixer to the
   endpoints in the other domain(s).  In other cases, a message is
   handled by the mixer locally and therefore not forwarded to any other
   domain.

   When replacing the multicast network in Figure 13 (to the left of the
   mixer) with individual unicast paths as depicted in Figure 14, the
   mixer model is very similar to the one discussed in Section 3.9
   below.  Please see the discussion in Section 3.9 about the
   differences between these two models.

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

               Figure 14: RTP Mixer with Only Unicast Paths

   We now discuss in more detail the different mixing operations that a
   mixer can perform and how they can affect RTP and RTCP behavior.

3.6.1.  Media-Mixing Mixer

   The Media-Mixing Mixer is likely the one that most think of when they
   hear the term "mixer".  Its basic mode of operation is that it
   receives RTP streams from several endpoints and selects the stream(s)
   to be included in a media-domain mix.  The selection can be through
   static configuration or by dynamic, content-dependent means such as
   voice activation.  The mixer then creates a single outgoing RTP
   stream from this mix.

   The most commonly deployed Media-Mixing Mixer is probably the audio
   mixer, used in voice conferencing, where the output consists of a
   mixture of all the input audio signals; this needs minimal signaling
   to be successfully set up.  From a signal processing viewpoint, audio
   mixing is relatively straightforward and commonly possible for a
   reasonable number of endpoints.  Assume, for example, that one wants
   to mix N streams from N different endpoints.  The mixer needs to
   decode those N streams, typically into the sample domain, and then
   produce N or N+1 mixes.  Different mixes are needed so that each
   endpoint gets a mix of all other sources except its own, as this
   would result in an echo.  When N is lower than the number of all
   endpoints, one may produce a mix of all N streams for the group that
   are currently not included in the mix; thus, N+1 mixes.  These audio
   streams are then encoded again, RTP packetized, and sent out.  In
   many cases, audio level normalization, noise suppression, and similar
   signal processing steps are also required or desirable before the
   actual mixing process commences.

   In video, the term "mixing" has a different interpretation than
   audio.  It is commonly used to refer to the process of spatially
   combining contributed video streams, which is also known as "tiling".
   The reconstructed, appropriately scaled down videos can be spatially
   arranged in a set of tiles, with each tile containing the video from
   an endpoint (typically showing a human participant).  Tiles can be of
   different sizes so that, for example, a particularly important

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   participant, or the loudest speaker, is being shown in a larger tile
   than other participants.  A self-view picture can be included in the
   tiling, which can be either locally produced or feedback from a
   mixer-received and reconstructed video image.  Such remote loopback
   allows for confidence monitoring, i.e., it enables the participant to
   see himself/herself in the same quality as other participants see
   him/her.  The tiling normally operates on reconstructed video in the
   sample domain.  The tiled image is encoded, packetized, and sent by
   the mixer to the receiving endpoints.  It is possible that a
   middlebox with media mixing duties contains only a single mixer of
   the aforementioned type, in which case all participants necessarily
   see the same tiled video, even if it is being sent over different RTP
   streams.  More common, however, are mixing arrangements where an
   individual mixer is available for each outgoing port of the
   middlebox, allowing individual compositions for each receiving
   endpoint (a feature commonly referred to as personalized layout).

   One problem with media mixing is that it consumes both large amounts
   of media processing resources (for the decoding and mixing process in
   the uncompressed domain) and encoding resources (for the encoding of
   the mixed signal).  Another problem is the quality degradation
   created by decoding and re-encoding the media, which is the result of
   the lossy nature of the most commonly used media codecs.  A third
   problem is the latency introduced by the media mixing, which can be
   substantial and annoyingly noticeable in case of video, or in case of
   audio if that mixed audio is lip-synchronized with high-latency
   video.  The advantage of media mixing is that it is straightforward
   for the endpoints to handle the single media stream (which includes
   the mixed aggregate of many sources), as they don't need to handle
   multiple decodings, local mixing, and composition.  In fact, mixers
   were introduced in pre-RTP times so that legacy, single stream
   receiving endpoints (that, in some protocol environments, actually
   didn't need to be aware of the multipoint nature of the conference)
   could successfully participate in what a user would recognize as a
   multiparty video conference.

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           +-A---------+          +-MIXER----------------------+
           | +-RTP1----|          |-RTP1------+        +-----+ |
           | | +-Audio-|          |-Audio---+ | +---+  |     | |
           | | |    AA1|--------->|---------+-+-|DEC|->|     | |
           | | |       |<---------|MA1 <----+ | +---+  |     | |
           | | |       |          |(BA1+CA1)|\| +---+  |     | |
           | | +-------|          |---------+ +-|ENC|<-| B+C | |
           | +---------|          |-----------+ +---+  |     | |
           +-----------+          |                    |     | |
                                  |                    |  M  | |
           +-B---------+          |                    |  E  | |
           | +-RTP2----|          |-RTP2------+        |  D  | |
           | | +-Audio-|          |-Audio---+ | +---+  |  I  | |
           | | |    BA1|--------->|---------+-+-|DEC|->|  A  | |
           | | |       |<---------|MA2 <----+ | +---+  |     | |
           | | +-------|          |(AA1+CA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|ENC|<-| A+C | |
           +-----------+          |-----------+ +---+  |     | |
                                  |                    |  M  | |
           +-C---------+          |                    |  I  | |
           | +-RTP3----|          |-RTP3------+        |  X  | |
           | | +-Audio-|          |-Audio---+ | +---+  |  E  | |
           | | |    CA1|--------->|---------+-+-|DEC|->|  R  | |
           | | |       |<---------|MA3 <----+ | +---+  |     | |
           | | +-------|          |(AA1+BA1)|\| +---+  |     | |
           | +---------|          |---------+ +-|ENC|<-| A+B | |
           +-----------+          |-----------+ +---+  +-----+ |
                                  +----------------------------+

            Figure 15: Session and SSRC Details for Media Mixer

   From an RTP perspective, media mixing can be a very simple process,
   as can be seen in Figure 15.  The mixer presents one SSRC towards the
   receiving endpoint, e.g., MA1 to Peer A, where the associated stream
   is the media mix of the other endpoints.  As each peer, in this
   example, receives a different version of a mix from the mixer, there
   is no actual relation between the different RTP sessions in terms of
   actual media or transport-level information.  There are, however,
   common relationships between RTP1-RTP3, namely SSRC space and
   identity information.  When A receives the MA1 stream, which is a
   combination of BA1 and CA1 streams, the mixer may include CSRC
   information in the MA1 stream to identify the Contributing Sources
   BA1 and CA1, allowing the receiver to identify the Contributing
   Sources even if this were not possible through the media itself or
   through other signaling means.

   The CSRC has, in turn, utility in RTP extensions, like the RTP header
   extension for Mixer-to-Client Audio Level Indication [RFC6465].  If

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   the SSRCs from the endpoint to mixer paths are used as CSRCs in
   another RTP session, then RTP1, RTP2, and RTP3 become one joint
   session as they have a common SSRC space.  At this stage, the mixer
   also needs to consider which RTCP information it needs to expose in
   the different paths.  In the above scenario, a mixer would normally
   expose nothing more than the SDES information and RTCP BYE for a CSRC
   leaving the session.  The main goal would be to enable the correct
   binding against the application logic and other information sources.
   This also enables loop detection in the RTP session.

3.6.2.  Media-Switching Mixer

   Media-Switching Mixers are used in limited functionality scenarios
   where no, or only very limited, concurrent presentation of multiple
   sources is required by the application and also in more complex
   multi-stream usages with receiver mixing or tiling, including
   combined with simulcast and/or scalability between source and mixer.
   An RTP mixer based on media switching avoids the media decoding and
   encoding operations in the mixer, as it conceptually forwards the
   encoded media stream as it was being sent to the mixer.  It does not
   avoid, however, the decryption and re-encryption cycle as it rewrites
   RTP headers.  Forwarding media (in contrast to reconstructing-mixing-
   encoding media) reduces the amount of computational resources needed
   in the mixer and increases the media quality (both in terms of
   fidelity and reduced latency).

   A Media-Switching Mixer maintains a pool of SSRCs representing
   conceptual or functional RTP streams that the mixer can produce.
   These RTP streams are created by selecting media from one of the RTP
   streams received by the mixer and forwarded to the peer using the
   mixer's own SSRCs.  The mixer can switch between available sources if
   that is required by the concept for the source, like the currently
   active speaker.  Note that the mixer, in most cases, still needs to
   perform a certain amount of media processing, as many media formats
   do not allow to "tune into" the stream at arbitrary points in their
   bitstream.

   To achieve a coherent RTP stream from the mixer's SSRC, the mixer
   needs to rewrite the incoming RTP packet's header.  First, the SSRC
   field must be set to the value of the mixer's SSRC.  Second, the
   sequence number must be the next in the sequence of outgoing packets
   it sent.  Third, the RTP timestamp value needs to be adjusted using
   an offset that changes each time one switches the Media Source.
   Finally, depending on the negotiation of the RTP payload type, the
   value representing this particular RTP payload configuration may have
   to be changed if the different endpoint-to-mixer paths have not
   arrived on the same numbering for a given configuration.  This also

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   requires that the different endpoints support a common set of codecs,
   otherwise media transcoding for codec compatibility would still be
   required.

   We now consider the operation of a Media-Switching Mixer that
   supports a video conference with six participating endpoints (A-F)
   where the two most recent speakers in the conference are shown to
   each receiving endpoint.  Thus, the mixer has two SSRCs sending video
   to each peer, and each peer is capable of locally handling two video
   streams simultaneously.

         +-A---------+             +-MIXER----------------------+
         | +-RTP1----|             |-RTP1------+        +-----+ |
         | | +-Video-|             |-Video---+ |        |     | |
         | | |    AV1|------------>|---------+-+------->|  S  | |
         | | |       |<------------|MV1 <----+-+-BV1----|  W  | |
         | | |       |<------------|MV2 <----+-+-EV1----|  I  | |
         | | +-------|             |---------+ |        |  T  | |
         | +---------|             |-----------+        |  C  | |
         +-----------+             |                    |  H  | |
                                   |                    |     | |
         +-B---------+             |                    |  M  | |
         | +-RTP2----|             |-RTP2------+        |  A  | |
         | | +-Video-|             |-Video---+ |        |  T  | |
         | | |    BV1|------------>|---------+-+------->|  R  | |
         | | |       |<------------|MV3 <----+-+-AV1----|  I  | |
         | | |       |<------------|MV4 <----+-+-EV1----|  X  | |
         | | +-------|             |---------+ |        |     | |
         | +---------|             |-----------+        |     | |
         +-----------+             |                    |     | |
                                   :                    :     : :
                                   :                    :     : :
         +-F---------+             |                    |     | |
         | +-RTP6----|             |-RTP6------+        |     | |
         | | +-Video-|             |-Video---+ |        |     | |
         | | |    FV1|------------>|---------+-+------->|     | |
         | | |       |<------------|MV11 <---+-+-AV1----|     | |
         | | |       |<------------|MV12 <---+-+-EV1----|     | |
         | | +-------|             |---------+ |        |     | |
         | +---------|             |-----------+        +-----+ |
         +-----------+             +----------------------------+


                   Figure 16: Media-Switching RTP Mixer

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   The Media-Switching Mixer can, similarly to the Media-Mixing Mixer,
   reduce the bitrate required for media transmission towards the
   different peers by selecting and forwarding only a subset of RTP
   streams it receives from the sending endpoints.  In case the mixer
   receives simulcast transmissions or a scalable encoding of the Media
   Source, the mixer has more degrees of freedom to select streams or
   subsets of streams to forward to a receiving endpoint, both based on
   transport or endpoint restrictions as well as application logic.

   To ensure that a media receiver in an endpoint can correctly decode
   the media in the RTP stream after a switch, a codec that uses
   temporal prediction needs to start its decoding from independent
   refresh points, or points in the bitstream offering similar
   functionality (like "dirty refresh points").  For some codecs, for
   example, frame-based speech and audio codecs, this is easily achieved
   by starting the decoding at RTP packet boundaries, as each packet
   boundary provides a refresh point (assuming proper packetization on
   the encoder side).  For other codecs, particularly in video, refresh
   points are less common in the bitstream or may not be present at all
   without an explicit request to the respective encoder.  The Full
   Intra Request [RFC5104] RTCP codec control message has been defined
   for this purpose.

   In this type of mixer, one could consider fully terminating the RTP
   sessions between the different endpoint and mixer paths.  The same
   arguments and considerations as discussed in Section 3.9 need to be
   taken into consideration and apply here.

3.7.  Selective Forwarding Middlebox

   Another method for handling media in the RTP mixer is to "project",
   or make available, all potential RTP sources (SSRCs) into a per-
   endpoint, independent RTP session.  The middlebox can select which of
   the potential sources that are currently actively transmitting media
   will be sent to each of the endpoints.  This is similar to the Media-
   Switching Mixer but has some important differences in RTP details.

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          +-A---------+             +-Middlebox-----------------+
          | +-RTP1----|             |-RTP1------+       +-----+ |
          | | +-Video-|             |-Video---+ |       |     | |
          | | |    AV1|------------>|---------+-+------>|     | |
          | | |       |<------------|BV1 <----+-+-------|  S  | |
          | | |       |<------------|CV1 <----+-+-------|  W  | |
          | | |       |<------------|DV1 <----+-+-------|  I  | |
          | | |       |<------------|EV1 <----+-+-------|  T  | |
          | | |       |<------------|FV1 <----+-+-------|  C  | |
          | | +-------|             |---------+ |       |  H  | |
          | +---------|             |-----------+       |     | |
          +-----------+             |                   |  M  | |
                                    |                   |  A  | |
          +-B---------+             |                   |  T  | |
          | +-RTP2----|             |-RTP2------+       |  R  | |
          | | +-Video-|             |-Video---+ |       |  I  | |
          | | |    BV1|------------>|---------+-+------>|  X  | |
          | | |       |<------------|AV1 <----+-+-------|     | |
          | | |       |<------------|CV1 <----+-+-------|     | |
          | | |       | :    :    : |: :  : : : : :  : :|     | |
          | | |       |<------------|FV1 <----+-+-------|     | |
          | | +-------|             |---------+ |       |     | |
          | +---------|             |-----------+       |     | |
          +-----------+             |                   |     | |
                                    :                   :     : :
                                    :                   :     : :
          +-F---------+             |                   |     | |
          | +-RTP6----|             |-RTP6------+       |     | |
          | | +-Video-|             |-Video---+ |       |     | |
          | | |    FV1|------------>|---------+-+------>|     | |
          | | |       |<------------|AV1 <----+-+-------|     | |
          | | |       | :    :    : |: :  : : : : :  : :|     | |
          | | |       |<------------|EV1 <----+-+-------|     | |
          | | +-------|             |---------+ |       |     | |
          | +---------|             |-----------+       +-----+ |
          +-----------+             +---------------------------+

                 Figure 17: Selective Forwarding Middlebox

   In the six endpoint conference depicted above (in Figure 17), one can
   see that endpoint A is aware of five incoming SSRCs, BV1-FV1.  If
   this middlebox intends to have a similar behavior as in Section 3.6.2
   where the mixer provides the endpoints with the two latest speaking
   endpoints, then only two out of these five SSRCs need concurrently
   transmit media to A.  As the middlebox selects the source in the
   different RTP sessions that transmit media to the endpoints, each RTP
   stream requires the rewriting of certain RTP header fields when being
   projected from one session into another.  In particular, the sequence

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   number needs to be consecutively incremented based on the packet
   actually being transmitted in each RTP session.  Therefore, the RTP
   sequence number offset will change each time a source is turned on in
   an RTP session.  The timestamp (possibly offset) stays the same.

   The RTP sessions can be considered independent, resulting in that the
   SSRC numbers used can also be handled independently.  This simplifies
   the SSRC collision detection and avoidance but requires tools such as
   remapping tables between the RTP sessions.  Using independent RTP
   sessions is not required, as it is possible for the switching
   behavior to also perform with a common SSRC space.  However, in this
   case, collision detection and handling becomes a different problem.
   It is up to the implementation to use a single common SSRC space or
   separate ones.

   Using separate SSRC spaces has some implications.  For example, the
   RTP stream that is being sent by endpoint B to the middlebox (BV1)
   may use an SSRC value of 12345678.  When that RTP stream is sent to
   endpoint F by the middlebox, it can use any SSRC value, e.g.,
   87654321.  As a result, each endpoint may have a different view of
   the application usage of a particular SSRC.  Any RTP-level identity
   information, such as SDES items, also needs to update the SSRC
   referenced, if the included SDES items are intended to be global.
   Thus, the application must not use SSRC as references to RTP streams
   when communicating with other peers directly.  This also affects loop
   detection, which will fail to work as there is no common namespace
   and identities across the different legs in the Communication Session
   on the RTP level.  Instead, this responsibility falls onto higher
   layers.

   The middlebox is also responsible for receiving any RTCP codec
   control requests coming from an endpoint and deciding if it can act
   on the request locally or needs to translate the request into the RTP
   session/transport leg that contains the Media Source.  Both endpoints
   and the middlebox need to implement conference-related codec control
   functionalities to provide a good experience.  Commonly used are Full
   Intra Request to request from the Media Source that switching points
   be provided between the sources and Temporary Maximum Media Bitrate
   Request (TMMBR) to enable the middlebox to aggregate congestion
   control responses towards the Media Source so to enable it to adjust
   its bitrate (obviously, only in case the limitation is not in the
   source to middlebox link).

   The Selective Forwarding Middlebox has been introduced in recently
   developed videoconferencing systems in conjunction with, and to
   capitalize on, scalable video coding as well as simulcasting.  An
   example of scalable video coding is Annex G of H.264, but other
   codecs, including H.264 AVC and VP8, also exhibit scalability, albeit

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   only in the temporal dimension.  In both scalable coding and
   simulcast cases, the video signal is represented by a set of two or
   more bitstreams, providing a corresponding number of distinct
   fidelity points.  The middlebox selects which parts of a scalable
   bitstream (or which bitstream, in the case of simulcasting) to
   forward to each of the receiving endpoints.  The decision may be
   driven by a number of factors, such as available bitrate, desired
   layout, etc.  Contrary to transcoding MCUs, SFMs have extremely low
   delay and provide features that are typically associated with high-
   end systems (personalized layout, error localization) without any
   signal processing at the middlebox.  They are also capable of scaling
   to a large number of concurrent users, and--due to their very low
   delay--can also be cascaded.

   This version of the middlebox also puts different requirements on the
   endpoint when it comes to decoder instances and handling of the RTP
   streams providing media.  As each projected SSRC can, at any time,
   provide media, the endpoint either needs to be able to handle as many
   decoder instances as the middlebox received, or have efficient
   switching of decoder contexts in a more limited set of actual decoder
   instances to cope with the switches.  The application also gets more
   responsibility to update how the media provided is to be presented to
   the user.

   Note that this topology could potentially be seen as a Media
   Translator that includes an on/off logic as part of its media
   translation.  The topology has the property that all SSRCs present in
   the session are visible to an endpoint.  It also has mixer aspects,
   as the streams it provides are not basically translated versions, but
   instead they have conceptual property assigned to them and can be
   both turned on/off as well as fully or partially delivered.  Thus,
   this topology appears to be some hybrid between the translator and
   mixer model.

   The differences between a Selective Forwarding Middlebox and a
   Switching-Media Mixer (Section 3.6.2) are minor, and they share most
   properties.  The above requirement on having a large number of
   decoding instances or requiring efficient switching of decoder
   contexts, are one point of difference.  The other is how the
   identification is performed, where the mixer uses CSRC to provide
   information on what is included in a particular RTP stream that
   represents a particular concept.  Selective forwarding gets the
   source information through the SSRC and instead uses other mechanisms
   to indicate the streams intended usage, if needed.

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3.8.  Point to Multipoint Using Video-Switching MCUs

   Shortcut name: Topo-Video-switch-MCU

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

        Figure 18: Point to Multipoint Using a Video-Switching MCU

   This PtM topology was popular in early implementations of multipoint
   videoconferencing systems due to its simplicity, and the
   corresponding middlebox design has been known as a "video-switching
   MCU".  The more complex RTCP-terminating MCUs, discussed in the next
   section, became the norm, however, when technology allowed
   implementations at acceptable costs.

   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
   well.

   The video-switching MCU may also perform media translation to modify
   the content in bitrate, 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 endpoints.  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 RTP stream results in most participants seeing only a
   subset of the sent RTP streams at any given time, often a single RTP
   stream per conference.  Therefore, RTCP receiver reports only report
   on these RTP streams.  Consequently, the endpoints emitting RTP
   streams that are not currently forwarded receive a view of the
   session that indicates their RTP streams disappear somewhere en

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   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 3.6) and
   forward the selected RTP 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.

3.9.  Point to Multipoint Using RTCP-Terminating MCU

   Shortcut name: Topo-RTCP-terminating-MCU

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

        Figure 19: Point to Multipoint Using Content Modifying MCUs

   In this PtM scenario, each endpoint 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 endpoints or

   b.  the mixed aggregate of what the MCU receives from the other PtP
       paths, which are part of the same Communication Session.

   In case (a), the MCU may modify the content in terms of bitrate,
   encoding format, or resolution.  No explicit RTP mechanism is used to
   establish the relationship between the original RTP stream of the
   media being sent and the RTP stream the MCU sends.  In other words,
   the outgoing RTP streams 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 RTP session for each endpoint.

   In case (b), the MCU is the Media Source and generates the Source RTP
   Stream as it mixes the received content and then encodes and
   packetizes it for transmission to an endpoint.  According to RTP

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   [RFC3550], the SSRC of the contributors are to be signaled using the
   CSRC/CC mechanism.  In practice, today, most deployed MCUs do not
   implement this feature.  Instead, the identification of the endpoints
   whose content is included in the mixer's output is not indicated
   through any explicit RTP mechanism.  That is, most deployed MCUs set
   the CC field in the RTP header to zero, thereby indicating no
   available CSRC information, even if they could identify the original
   sending endpoints 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
       generated.

   2.  There is no information about active media senders available in
       the RTP packet.  As this information is missing, receivers cannot
       use it.  It also 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 many/most deployed MCUs (and video conferencing endpoints)
   rely on signaling-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.

3.10.  Split Component Terminal

   Shortcut name: Topo-Split-Terminal

   In some applications, for example, in some telepresence systems,
   terminals may not be integrated into a single functional unit but
   composed of more than one subunits.  For example, a telepresence room
   terminal employing multiple cameras and monitors may consist of
   multiple video conferencing subunits, each capable of handling a
   single camera and monitor.  Another example would be a video
   conferencing terminal in which audio is handled by one subunit, and
   video by another.  Each of these subunits uses its own physical
   network interface (for example: Ethernet jack) and network address.

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   The various (media processing) subunits need (logically and
   physically) to be interconnected by control functionality, but their
   media plane functionality may be split.  These types of terminals are
   referred to as split component terminals.  Historically, the earliest
   split component terminals were perhaps the independent audio and
   video conference software tools used over the MBONE in the late
   1990s.

   An example for such a split component terminal is depicted in
   Figure 20.  Within split component terminal A, at least audio and
   video subunits are addressed by their own network addresses.  In some
   of these systems, the control stack subunit may also have its own
   network address.

   From an RTP viewpoint, each of the subunits terminates RTP and acts
   as an endpoint in the sense that each subunit includes its own,
   independent RTP stack.  However, as the subunits are semantically
   part of the same terminal, it is appropriate that this semantic
   relationship is expressed in RTCP protocol elements, namely in the
   CNAME.

               +---------------------+
               | Endpoint A          |
               | Local Area Network  |
               |      +------------+ |
               |   +->| Audio      |<+-RTP---\
               |   |  +------------+ |        \    +------+
               |   |  +------------+ |         +-->|      |
               |   +->| Video      |<+-RTP-------->|  B   |
               |   |  +------------+ |         +-->|      |
               |   |  +------------+ |        /    +------+
               |   +->| Control    |<+-SIP---/
               |      +------------+ |
               +---------------------+

                    Figure 20: Split Component Terminal

   It is further sensible that the subunits share a common clock from
   which RTP and RTCP clocks are derived, to facilitate synchronization
   and avoid clock drift.

   To indicate that audio and video Source Streams generated by
   different subunits share a common clock, and can be synchronized, the
   RTP streams generated from those Source Streams need to include the
   same CNAME in their RTCP SDES packets.  The use of a common CNAME for
   RTP flows carried in different transport-layer flows is entirely
   normal for RTP and RTCP senders, and fully compliant RTP endpoints,
   middleboxes, and other tools should have no problem with this.

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   However, outside of the split component terminal scenario (and
   perhaps a multihomed endpoint scenario, which is not further
   discussed herein), the use of a common CNAME in RTP streams sent from
   separate endpoints (as opposed to a common CNAME for RTP streams sent
   on different transport-layer flows between two endpoints) is rare.
   It has been reported that at least some third-party tools like some
   network monitors do not handle gracefully endpoints that use a common
   CNAME across multiple transport-layer flows: they report an error
   condition in which two separate endpoints are using the same CNAME.
   Depending on the sophistication of the support staff, such erroneous
   reports can lead to support issues.

   The aforementioned support issue can sometimes be avoided if each of
   the subunits of a split component terminal is configured to use a
   different CNAME, with the synchronization between the RTP streams
   being indicated by some non-RTP signaling channel rather than using a
   common CNAME sent in RTCP.  This complicates the signaling,
   especially in cases where there are multiple SSRCs in use with
   complex synchronization requirements, as is the same in many current
   telepresence systems.  Unless one uses RTCP terminating topologies
   such as Topo-RTCP-terminating-MCU, sessions involving more than one
   video subunit with a common CNAME are close to unavoidable.

   The different RTP streams comprising a split terminal system can form
   a single RTP session or they can form multiple RTP sessions,
   depending on the visibility of their SSRC values in RTCP reports.  If
   the receiver of the RTP streams sent by the split terminal sends
   reports relating to all of the RTP flows (i.e., to each SSRC) in each
   RTCP report, then a single RTP session is formed.  Alternatively, if
   the receiver of the RTP streams sent by the split terminal does not
   send cross-reports in RTCP, then the audio and video form separate
   RTP sessions.

   For example, in Figure 20, B will send RTCP reports to each of the
   subunits of A.  If the RTCP packets that B sends to the audio subunit
   of A include reports on the reception quality of the video as well as
   the audio, and similarly if the RTCP packets that B sends to the
   video subunit of A include reports on the reception quality of the
   audio as well as video, then a single RTP session is formed.
   However, if the RTCP packets B sends to the audio subunit of A only
   report on the received audio, and the RTCP packets B sends to the
   video subunit of A only report on the received video, then there are
   two separate RTP sessions.

   Forming a single RTP session across the RTP streams sent by the
   different subunits of a split terminal gives each subunit visibility
   into reception quality of RTP streams sent by the other subunits.

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   This information can help diagnose reception quality problems, but at
   the cost of increased RTCP bandwidth use.

   RTP streams sent by the subunits of a split terminal need to use the
   same CNAME in their RTCP packets if they are to be synchronized,
   irrespective of whether a single RTP session is formed or not.

3.11.  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 13,
   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 RTP 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 media 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.  Therefore, any reports
   related to these streams must be removed.  Also, receiver reports
   related to A's and C's RTP streams 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 locally generated reports reflecting the situation
   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 streams 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 for implementation.

3.12.  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 does not forward RTCP from sources in other domains,
   but instead generates its own RTCP packets for each domain it mixes
   into, including the necessary SDES information for both the CSRCs and

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   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.  It 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.



(page 39 continued on part 3)

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