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