| Network Working Group | M. Westerlund |
| Internet-Draft | Ericsson |
| Obsoletes: 5117 (if approved) | S. Wenger |
| Intended status: Informational | Vidyo |
| Expires: October 24, 2013 | April 22, 2013 |
RTP Topologies
draft-ietf-avtcore-rtp-topologies-update-00
This document discusses point to point and multi-endpoint topologies used in Real-time Transport Protocol (RTP)-based environments. In particular, centralized topologies commonly employed in the video conferencing industry are mapped to the RTP terminology.
This document is updated with additional topologies and are intended to replace RFC 5117.
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Real-time Transport Protocol (RTP) [RFC3550] topologies describe methods for interconnecting RTP entities and their processing behavior of RTP and RTCP. This document tries to address past and existing confusion, especially with respect to terms not defined in RTP but in common use in the conversational communication industry, such as MCU. In doing so, this memo provides a common information basis for future discussion and specification work. It attempts to clarify and explain sections of the Real-time Transport Protocol (RTP) spec [RFC3550] in an informal way. It is not intended to update or change what is normatively specified within RFC 3550.
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was developed the main emphasis lay in the efficient support of point to point and small multipoint scenarios without centralized multipoint control. However, in practice, many small multipoint conferences operate utilizing devices known as Multipoint Control Units (MCUs). MCUs may implement Mixer or Translator (in RTP [RFC3550] terminology) functionality and signalling support. They may also contain additional application functionality. This document focuses on the media transport aspects of the MCU that can be realized using RTP, as discussed below. Further considered are the properties of Mixers and Translators, and how some types of deployed MCUs deviate from these properties.
This subsection defines several topologies that are relevant for codec control but also RTP usage in other contexts. The section starts with point to point cases, without and with middleboxes. Then follows a number of different methods for establishing point to multipoint communication. These are structure around the most fundamental enabler, i.e. multicast, a mesh of connections, translators, mixers and source projection middlebox, to finally discuss MCUs. The section ends by discussing de-composed endpoints, asymmetric middlebox behaviors and combining topologies.
The topologies may be referenced in other documents by a shortcut name, indicated by the prefix "Topo-".
For each of the RTP-defined topologies, we discuss how RTP, RTCP, and the carried media are handled. With respect to RTCP, we also discuss the handling of RTCP feedback messages as defined in [RFC4585] and [RFC5104]. Any important differences between the two will be illuminated in the discussion. At this stage we don't intended to discuss in detail how each and every feedback messages should be treated in the various topologies.
Shortcut name: Topo-Point-to-Point
The Point to Point (PtP) topology [fig-point-to-point] consists of two endpoints, communicating using unicast. Both RTP and RTCP traffic are conveyed endpoint-to-endpoint, using unicast traffic only (even if, in exotic cases, this unicast traffic happens to be conveyed over an IP-multicast address).
+---+ +---+ | A |<------->| B | +---+ +---+
Figure 1: Point to Point
The main property of this topology is that A sends to B, and only B, while B sends to A, and only A. This avoids all complexities of handling multiple endpoints and combining the requirements from them. Note that an endpoint can still use multiple RTP Synchronization Sources (SSRCs) in an RTP session. The number of RTP sessions in use between A and B can also be of any number.
RTCP feedback messages for the indicated SSRCs are communicated directly between the endpoints. Therefore, this topology poses minimal (if any) issues for any feedback messages. For RTP sessions which use multiple SSRC per endpoint it can be relevant to implement support for cross reporting suppression as defined in "Real-Time Transport Protocol (RTP) Considerations for Endpoints Sending Multiple Media Streams" [I-D.lennox-avtcore-rtp-multi-stream].
This section discusses cases where two endpoints communicate but have one or more middlebox involved in the RTP session.
Shortcut name: Topo-PtP-Translator
Two main categories of Translators can be distinguished; Transport Translators and Media translators. Both Translator types share common attributes that separate them from Mixers. For each media stream that the Translator receives, it generates an individual stream in the other domain. A translator keeps the SSRC for a stream across the translation, whereas a Mixer can select a single media stream, or send out multiple mixed media streams, but always under its own SSRC, possibly using the CSRC field to indicate the source(s) of the content. Mixers are more common in point to multipoint cases than in PtP. The reason is that in PtP use cases the primary focus is interoperability, such as transcoding to a codec the receiver supports, which can be done by a media translator.
As specified in Section 7.1 of [RFC3550], the SSRC space is common for all participants in the RTP session, independent of on which side of the Translator the session resides. Therefore, it is the responsibility of the participants to run SSRC collision detection, and the SSRC is thus a field the Translator cannot change. Any SDES information associated with a SSRC or CSRC also needs to be forwarded between the domains for any SSRC/CSRC used in the different domains.
A Translator commonly does not use an SSRC of its own, and is not visible as an active participant in the session. One reason to have its own SSRC is when a Translator acts as a quality monitor that sends RTCP reports and therefore is required to have an SSRC. Another example is the case when a Translator is prepared to use RTCP feedback messages. This may, for example, occur in a translator configured to detect packet loss of important video packets and wants to trigger repair by the media sender, by sending feedback messages. While such feedback could use the SSRC of the target for the translator, but this in turn would require translation of the targets RTCP reports to make them consistent. It may be simpler to expose an additional SSRC in the session, the only concern are endpoints failing to support the full RTP specification, thus having issues with multiple SSRCs reporting on the RTP streams sent by that endpoint.
In general, a Translator implementation should consider which RTCP feedback messages or codec-control messages it needs to understand in relation to the functionality of the Translator itself. This is completely in line with the requirement to also translate RTCP messages between the domains.
There exist a number of different types of middleboxes that might be inserted between two RTP endpoints on the transport level, e.g. perform changes on the IP/UDP headers, and are, therefore, basic transport translators. These middleboxes come in many variations including NAT [RFC3022] traversal by pinning the media path to a public address domain relay, network topologies where the media flow is required to pass a particular point for audit by employing relaying, or preserving privacy by hiding each peers transport addresses to the other party. Other protocols or functionalities that provide this behavior are TURN [RFC5766] servers, Session Border Gateways and Media Processing Nodes with media anchoring functionalities.
+---+ +---+ +---+ | A |<------>| T |<------->| B | +---+ +---+ +---+
Figure 2: Point to Point with Translator
What is common for these functions is that they are normally transparent on RTP level, i.e. they perform no changes on any RTP or RTCP packet fields, only on the lower layers. However, they may effect the path the RTP and RTCP packets are routed between the endpoints in the RTP session, and thereby only indirectly affect the RTP session. For this reason, one could believe that transport translator type middleboxes do not need to included in this document. However, this topology can raise additional requirements the RTP implementation and its interactions with the signalling solution. Both in signalling and in certain RTCP field other network addresses than those of the relay can occur, due to that B has different network address than the relay (T). However, implementation not capable of this will neither not work when endpoints are subject to NAT.
Transport Translators (Topo-Trn-Translator) do not modify the media stream itself, but are concerned with transport parameters. Transport parameters, in the sense of this section, comprise the transport addresses (to bridge different domains such unicast to multicast) and the media packetization to allow other transport protocols to be interconnected to a session (in gateways). Of the transport Translators, this memo is primarily interested in those that use RTP on both sides, and this is assumed henceforth. Translators that bridge between different protocol worlds need to be concerned about the mapping of the SSRC/CSRC (Contributing Source) concept to the non-RTP protocol. When designing a Translator to a non-RTP-based media transport, one crucial factor lies in how to handle different sources and their identities. This problem space is not discussed henceforth.
The most basic transport translators that operate below RTP level was already discussed in Section 3.2.1.1.
Media Translators (Topo-Media-Translator), in contrast, modify the media stream itself. This process is commonly known as transcoding. The modification of the media stream can be as small as removing parts of the stream, and it can go all the way to a full transcoding (down to the sample level or equivalent) utilizing a different media codec. Media Translators are commonly used to connect entities without a common interoperability point in the media encoding.
Stand-alone Media Translators are rare. Most commonly, a combination of Transport and Media Translators are used to translate both the media stream and the transport aspects of a stream between two transport domains (or clouds).
When media translation occurs, the Translator's task regarding handling of RTCP traffic becomes substantially more complex. In this case, the Translator needs to rewrite B's RTCP Receiver Report before forwarding them to A. The rewriting is needed as the stream received by B is not the same stream as the other participants receive. For example, the number of packets transmitted to B may be lower than what A sends, due to the different media format and data rate. Therefore, if the Receiver Reports were forwarded without changes, the extended highest sequence number would indicate that B were substantially behind in reception, while it most likely it would not be. Therefore, the Translator must translate that number to a corresponding sequence number for the stream the Translator received. Similar arguments can be made for most other fields in the RTCP Receiver Reports.
A media Translator may in some cases act on behalf of the "real" source and respond to RTCP feedback messages. This may occur, for example, when a receiver requests a bandwidth reduction, and the media Translator has not detected any congestion or other reasons for bandwidth reduction between the media source and itself. In that case, it is sensible that the media Translator reacts to the codec control messages itself, for example, by transcoding to a lower media rate.
A variant of translator behaviour worth pointing out is the one depicted in Figure 3 of an endpoint A sends a media flow to B. On the path there is a device T that on A's behalf does something with the media streams, for example adds an RTP session with FEC information for A's media streams. In this case, T needs to bind the new FEC streams to A's media stream, for example by using the same CNAME as A.
+------+ +------+ +------+ | | | | | | | A |------->| T |-------->| B | | | | |---FEC-->| | +------+ +------+ +------+
Figure 3: When De-composition is a Translator
This type of functionality where T does something with the media stream on behalf of A is covered under the media translator definition.
There exist middleboxes that interconnect two endpoints through themselves not by being part of a common RTP session. Instead they establish two different RTP sessions, one between A and the middlebox (MB) and another between the MB and B.
|<--Session A-->| |<--Session B-->| +------+ +------+ +------+ | A |------->| MB |-------->| B | +------+ +------+ +------+
Figure 4: When De-composition is a Translator
The MB acts as a application level gateway and bridges the two RTP session. This bridging can be as basic as forwarding the RTP payloads between the sessions, or more complex including media transcoding. The difference with the single RTP session context is the handling of the SSRCs and the other session related identifiers, such as CNAMEs. With two different RTP sessions these can be freely changed and it becomes the MB task to maintain the right relations.
The signalling or other above-RTP level functionalities referencing RTP media streams may be what is most impacted by using two RTP sessions and changing identifiers. The structure with two RTP sessions also puts a congestion control requirement on the middlebox, because it becomes fully responsible for the media stream it sources into each of the sessions.
This can be solved locally or by bridging also statistics from the receiving endpoint. However, from an implementation point this requires the implementation to support dealing with a number of inconsistencies. First, packet loss must be detected for an RTP flow sent from A to the MB, and that loss must be reported through a skipped sequence number in the flow from the MB to B. This coupling and the resulting inconsistencies is conceptually easier to handle when considering the two flows as belonging to a single RTP session.
Multicast is a IP layer functionality that is available in some networks. Two main flavors can be distinguished: Any Source Multicast (ASM) where any multicast group participant can send to the group address and expect the packet to reach all group participants; and Source Specific Multicast (SSM), where only a particular IP host sends to the multicast group. Both these models are discussed below in their respective section.
Shortcut name: Topo-ASM (was Topo-Multicast)
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 5: Point to Multipoint Using Multicast
Point to Multipoint (PtM) is defined here as using a multicast topology as a transmission model, in which traffic from any participant reaches all the other participants, except for cases such as:
In the above context, "traffic" encompasses both RTP and RTCP traffic. The number of participants can vary between one and many, as RTP and RTCP scale to very large multicast groups (the theoretical limit of the number of participants in a single RTP session is in the range of billions). The above can be realized using Any Source Multicast (ASM).
For feedback usage, it is useful to define a "small multicast group" as a group where the number of participants is so low (and other factors such as the connectivity is so good) that it allows the participants to use early or immediate feedback, as defined in AVPF [RFC4585]. Even when the environment would allow for the use of a small multicast group, some applications may still want to use the more limited options for RTCP feedback available to large multicast groups, for example when there is a likelyhood that the threshold of the small multicast group (in terms of participants) may be exceeded during the lifetime of a session.
RTCP feedback messages in multicast reach, like media, every subscriber (subject to packet losses and multicast group subscription). Therefore, the feedback suppression mechanism discussed in [RFC4585] is typically required. Each individual node needs to process every feedback message it receives, not to determine if it is affected or if the feedback message applies only to some other participant, but also to derive timing restriction for the sending of its own feedback messages, if any.
In Any Source Multicast, any of the participants can send to all the other participants, by sending a packet to the multicast group. In contrast, Source Specific Multicast [RFC4607] refers to scenarios where only a single source (Distribution Source) can send to the multicast group, creating a topology that looks like the one below:
+--------+ +-----+
|Media | | | Source-specific
|Sender 1|<----->| D S | Multicast
+--------+ | I O | +--+----------------> R(1)
| S U | | | |
+--------+ | T R | | +-----------> R(2) |
|Media |<----->| R C |->+ | : | |
|Sender 2| | I E | | +------> R(n-1) | |
+--------+ | B | | | | | |
: | U | +--+--> R(n) | | |
: | T +-| | | | |
: | I | |<---------+ | | |
+--------+ | O |F|<---------------+ | |
|Media | | N |T|<--------------------+ |
|Sender M|<----->| | |<-------------------------+
+--------+ +-----+ RTCP Unicast
FT = Feedback Target
Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not
co-located.
Figure 6: Point to Multipoint using Source Specific Multicast
In the SSM topology [fig-multipoint-ssm] a number of RTP sources (1 to M) are allowed to send media to the SSM group. These send media to a dedicated distribution source, which then forwards the media streams to the multicast group on behalf of the original senders. The media streams reach the Receivers (R(1) to R(n)). The Receivers' RTCP cannot be sent to the multicast group, as the SSM multicast group by definition has only a single source. To support RTCP, an RTP extension for SSM [RFC5760] was defined. It uses unicast transmission to send RTCP from each of the receivers to one or more Feedback Targets (FT). The feedback targets relay the RTCP unmodified, or provide summary of the participants RTCP reports towards the whole group by forwarding the RTCP traffic to the distribution source. Figure 6 only shows a single feedback target integrated in the distribution source, but for scalability the FT can be many and have responsibility for sub-groups of the receivers. For summary reports, however, there must be a single feedback aggregating all the summaries to a common message to the whole receiver group.
The RTP extension for SSM specifies how feedback (both reception information and specific feedback events) are handled. The more general problems associated with the use of multicast, where everyone receives what the distribution source sends needs to be accounted for.
The result of this is some common behaviours for RTP multicast:
All multicast configurations share a signalling requirement: all of the participants need to have the same RTP and payload type configuration. Otherwise, A could, for example, be using payload type 97 to identify the video codec H.264, while B would identify it as MPEG-2.
Security solutions for this type of group communications are also challenging. First, the key-management and the security protocol must support group communication. Source authentication becomes more difficult and requires special solutions. For more discussion on this please review Options for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options].
[RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions" results in additional extensions to SSM Topology.
----------- --------------
| |------------------------------------>| |
| |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| |
| | | |
| Multicast | ---------------- | |
| Source | | Retransmission | | |
| |-------->| Server (RS) | | |
| |.-.-.-.->| | | |
| | | ------------ | | |
----------- | | Feedback | |<.=.=.=.=.| |
| | Target (FT)| |<~~~~~~~~~| RTP Receiver |
PRIMARY MULTICAST | ------------ | | (RTP_Rx) |
RTP SESSION with | | | |
UNICAST FEEDBACK | | | |
| | | |
- - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- -
| | | |
UNICAST BURST | ------------ | | |
(or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| |
RTP SESSION | | Retrans. | |.........>| |
| |Source (BRS)| |<.=.=.=.=>| |
| ------------ | | |
| | | |
---------------- --------------
-------> Multicast RTP Flow
.-.-.-.> Multicast RTCP Flow
.=.=.=.> Unicast RTCP Reports
~~~~~~~> Unicast RTCP Feedback Messages
.......> Unicast RTP Flow
Figure 7
The Rapid acquisition extension allows an endpoint joining an SSM multicast session to request media starting with the last sync-point (from where media can be decoded without prior packets) to be sent at high speed until such time where, after decoding of these bursted media packets, the correct media timing is established, i.e. media packets are received within adequate buffer intervals for this application. This is accomplished by first establishing an unicast PtP RTP session between the BRS (Figure 7) and the RTP Receiver. That session is used to transmit cached packets from the multicast group at higher then nominal speed so to synchronize the receiver to the ongoing multicast packet flow. Once the RTP receiver and its decoder have caught up with the multicast session's current delivery, the receiver switches over to receiving from the multicast group directly. The (still existing) PtP RTP session can be used as a repair channel, i.e. for RTP Retransmission traffic of those packets that were not received from the multicast group.
Shortcut name: Topo-Mesh
+---+ +---+
| A |<---->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 8: Point to Multi-Point using Mesh
Based on the RTP session definition, it is clearly possible to have a joint RTP session over multiple unicast transport flows like the above three endpoint joint session. In this case, A needs to send its' media streams and RTCP packets to both B and C over their respective transport flows. As long as all participants do the same, everyone will have a joint view of the RTP session.
This doesn't 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, or it might have separate local reception ports for each of the endpoints.
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 media stream is being sent from each sending endpoint. In others, some form of local adaptation takes place in one or more of the RTP media streams, rendering them non-identical. From a topologies viewpoint, a difference exists in the behaviours around RTCP. For example, when a single RTP session spans all three endpoints and their connecting flows, a RTCP bandwidth is calculated and used for this single one joint session. In contrast, when there are multiple independent RTP sessions, each has its local RTCP bandwidth allocation. Also, when multiple sessions are used, endpoints not directly involved in these sessions do not have any awareness of the conditions occurring in sessions not involving that endpoint. For example, in case of the three endpoint configuration above, 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 can't be used to determine when a 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, its avoidance, has to be achieved by other means, for example through signaling, or the use of an RTP external name space binding SSRC/CSRC among any communicating RTP sessions in the mesh.
This section discusses some additional usages related to point to multipoint of Translators compared to the point to point only cases in Section 3.2.1.
Shortcut name: Topo-PtM-Trn-Translator
This section discusses Transport Translator only usages to enable multipoint sessions.
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 9: Point to Multipoint Using Multicast
Figure 9 depicts an example of a Transport Translator performing at least IP address translation. It allows the (non-multicast-capable) participants B and D to take part in an any source multicast session by having the Translator forward their unicast traffic to the multicast addresses in use, and vice versa. It must also forward B's traffic to D, and vice versa, to provide each of B and D with a complete view of the session.
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 10: RTP Translator (Relay) with Only Unicast Paths
Another Translator scenario is depicted in Figure 10. Herein, the Translator connects multiple users of a conference through unicast. This can be implemented using a very simple transport Translator, which in this document is called a relay. The relay forwards all traffic it receives, both RTP and RTCP, to all other participants. In doing so, a multicast network is emulated without relying on a multicast-capable network infrastructure.
For RTCP feedback this results in a similar set of considerations those described in the ASM RTP topology. It also puts some additional signalling requirements onto the session establishment; for example, a common configuration of RTP payload types is required.
In the context of multipoint communications a Media Translator is not providing new mechanisms to establish a multipoint session. It is much more an enabler or facilitator that ensures one or some sub-set of session participants can participate in the session.
If B in Figure 9 were behind a limited network path, the Translator may perform media transcoding to allow the traffic received from the other participants to reach B without overloading the path. This transcoding can help the other participants in the Multicast part of the session, by not requiring the quality transmitted by A to be lowered to the nitrates that B is actually capable of receiving.
Shortcut name: Topo-Mixer
A Mixer is a middlebox that aggregates multiple RTP streams, which are part of a session, by generating a new RTP stream and, in most cases, by manipulation of 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 11: Point to Multipoint Using the RFC 3550 Mixer Model
A Mixer can be viewed as a device terminating the media streams received from other session participants. Using the media data from the received media streams, a Mixer generates a media stream that is sent to the session participant.
The content that the Mixer provides is the mixed aggregate of what the Mixer receives over the PtP or PtM paths, which are part of the same conference session.
The Mixer is the content source, as it mixes the content (often in the uncompressed domain) and then encodes it for transmission to a participant. The CSRC Count (CC) and CSRC fields in the RTP header can be used to indicate the contributors of to the newly generated stream. The SSRCs of the to-be-mixed streams on the Mixer input appear as the CSRCs at the Mixer output. That output stream uses a unique SSRC that identifies the Mixer's stream. The CSRC should be forwarded between the different conference participants to allow for loop detection and identification of sources that are part of the global session. Note that Section 7.1 of RFC 3550 requires the SSRC space to be shared between domains for these reasons. 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 a receiver and should therefore send receiver reports for the media streams it receives. In its role as a media sender, it should also generate sender reports for those media 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 11 is involved in three domains that need to be separated: the any source multicast network (including participants A and C), participant B, and participant D. Assuming all four participants in the conference are interested in receiving content from each other participant, the Mixer produces different mixed 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 that is the receiving entity and transmitter of mixed content.
In the multicast domain, a Mixer still needs to provide a mixed view of the other domains. This makes the Mixer simpler to implement and avoids any issues with advanced RTCP handling or loop detection, which would be problematic if the Mixer were providing non-symmetric behavior. Please see Section 3.11 for more discussion on this topic. However, the mixing operation 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 participants in the other domain(s). In other cases, a message is handled by the Mixer itself and therefore not forwarded to any other domain.
When replacing the multicast network in Figure 11 (to the left of the Mixer) with individual unicast paths as depicted in Figure 12, 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 12: RTP Mixer with Only Unicast Paths
Lets now discuss in more detail different mixing operations that a mixer can perform and how they can affect the RTP and RTCP.
The media mixing mixer is likely the one that most think of when they hear the term "mixer". Its basic pattern of operation is that it receives media streams from (typically several) participants. Of those, it selects (either through static configuration or by dynamic, content dependent means such as voice activation) the stream(s) to be included in a media domain mix. Then it creates a single outgoing stream from this mix.
The most commonly deployed media mixer is probably the audio mixer, used in voice conferencing, where the output consists of a mixture of all the input streams; this needs minimal signalling to be successfully set up. Audio mixing is relatively straightforward and commonly possible for a reasonable number of participants. Lets assume that you want to mix N streams from different participants. The mixer needs to decode those N streams, typically into the sample domain. Then it needs to produce N or N+1 mixes, the reasons that different mixes are needed being that each contributing source get 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 participants 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 is also required before the actual mixing process.
Video can't really be "mixed" and produce something particularly useful for the users, however creating an composition out of the contributed video streams is possible and known as "tiling". For example the reconstructed, appropriately scaled down videos can be spatially arranged in a set of tiles, each tile containing the video from a participant. Tiles can be of different sizes, so that, for example, a particularly important participant, or the loudest speaker, is being shown on in larger tile than other participants. A self-picture can be included in the tiling, which can either be locally produced or be a feedback from a received and reconstructed video image (allowing for confidence monitoring, the participant sees himself/herself just 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. 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 arrangement where an individual mixer is available for each outgoing port of the middlebox, allowing individual compositions for each participant.
One problem with media mixing is that it consumes both large amount of media processing (for the actual 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 that is encapsulated in the RTP media stream, which is the result of the lossy nature of most, if not all, 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. The advantage of media mixing is that it is quite simplistic for the clients 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 can successfully participate in what a user would recognize as a multiparty session.
+-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 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+C | |
+-----------+ |-----------+ +---+ | | |
| | M | |
+-C---------+ | | I | |
| +-RTP3----| |-RTP3------+ | X | |
| | +-Audio-| |-Audio---+ | +---+ | E | |
| | | CA1|--------->|---------+-+-|DEC|->| R | |
| | | |<---------|MA3 <----+ | +---+ | | |
| | +-------| |(BA1+CA1)|\| +---+ | | |
| +---------| |---------+ +-|ENC|<-| A+B | |
+-----------+ |-----------+ +---+ +-----+ |
+----------------------------+
Figure 13: Session and SSRC details for Media Mixer
From an RTP perspective media mixing can be very straightforward as can be seen in Figure 13. The mixer presents one SSRC towards the receiving client, e.g. MA1 to Peer A; the associated stream of which is the media mix of the other participants. As, in this example, each peer receives a different version produced by the mixer, there is no actual relation between the different RTP sessions in the actual media or the 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 source BA1 and CA1, allowing the receiver to identify the contributing sources even if this were not possible through the media itself or other signaling means.
The CSRC has, in turn, utility in RTP extensions, like the Mixer to Client audio levels RTP header extension [RFC6465]. If the SSRC from endpoint to mixer leg are used as CSRC 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 need to consider which RTCP information it needs to expose in the different legs. In the above scenario, commonly, a mixer would expose nothing more than the Source Description (SDES) information and RTCP BYE for 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.
Media switching mixers are commonly used in such limited functionality scenarios where no, or only very limited, concurrent presentation of multiple sources is required by the application. An RTP Mixer based on media switching avoids the media decoding and encoding cycle in the mixer, as it conceptually forwards the encoded media stream as it was being sent to the mixer, but not 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) per transmitted bit.
A media switching mixer maintains a pool of SSRCs representing conceptual or functional streams the mixer can produce. These streams are created by selecting media from one of RTP media 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 currently active speaker. Note that the mixer, in most cases, still need to perform a certain amount of media processing, as many media formats do not allow to "tune" into the stream at arbitrary points of their bitstream.
To achieve a coherent RTP media 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. Secondly, the sequence number must be the next in the sequence of outgoing packets it sent. Thirdly the RTP timestamp value needs to be adjusted using an offset that changes each time one switch media source. Finally depending on the negotiation the RTP payload type value representing this particular RTP payload configuration may have to be changed if the different endpoint mixer legs have not arrived on the same numbering for a given configuration. This also requires that the different end-points do support a common set of codecs, otherwise media transcoding for codec compatibility is still required.
Lets consider the operation of media switching mixer that supports a video conference with six participants (A-F) where the two latest speakers in the conference are shown to each participants. 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---+ | | | |
| | | CV1|------------>|---------+-+------->| | |
| | | |<------------|MV11 <---+-+-AV1----| | |
| | | |<------------|MV12 <---+-+-EV1----| | |
| | +-------| |---------+ | | | |
| +---------| |-----------+ +-----+ |
+-----------+ +----------------------------+
Figure 14: Media Switching RTP Mixer
The Media Switching RTP mixer can, similarly to the Media Mixing Mixer, reduce the bit-rate required for media transmission towards the different peers by selecting and forwarding only a sub-set of RTP media streams it receives from the conference participants. In many practical cases, the link capacities of either direction between peers and mixer are the same, which effectively limits the subset to a single media stream.
To ensure that a media receiver can correctly decode the RTP media stream after a switch, a state saving (frame-based) codec needs to start its decoding from independent refresh points or similar points in the bitstream. For some codecs, for example frame based speech and audio codecs, this is easily achieved by starting the decoding at RTP packet boundaries (proper packetization on the encoder side assumed), as each packet boundary provides a refresh point. For other (mostly video-) codecs, refresh points are less common in the bitstream or may not be present at all without an explicit request to the respective encoder. For this purpose there exists the Full Intra Request [RFC5104] RTCP codec control message.
Also in this type of mixer one could consider to terminate the RTP sessions fully between the different end-point and mixer legs. The same arguments and considerations as discussed in Section 3.9 need to be taken into consideration and apply here.
Another method for handling media in the RTP mixer is to project all potential RTP sources (SSRCs) into a per end-point independent RTP session. The middlebox can select which of the potential sources that are currently actively transmitting media, despite that the middlebox, in another RTP session, may receive media from that end-point. 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 15: Media Projecting Middlebox
In the six participant conference depicted above in [fig-projecting] one can see that end-point A is aware of five incoming SSRCs, BV1-FV1. If this middlebox intends to have a similar behaviour as in Section 3.6.2 where the mixer provides the end-points with the two latest speaking end-points, 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 end-points, each RTP media stream requires some rewriting of 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 a RTP session. The timestamp (possibly offset) stays the same.
As the RTP sessions are independent, the SSRC numbers used can also be handled independently, thereby bypassing the requirement for SSRC collision detection and avoidance. On the other hand, tools such as remapping tables between the RTP sessions are required. For example, the stream that is being sent by endpoint B to the middlebox (BV1) may use an SSRC value of 12345678. When that media 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 media 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 RTP level. Instead this responsibility falls onto higher layers.
The middlebox is also responsible to receive any RTCP codec control requests coming from an end-point, and decide if it can act on the request locally or needs to translate the request into the RTP session that contains the media source. Both end-points 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 to provide switching points between the sources, and Temporary Maximum Media Bit-rate Request (TMMBR) to enable the middlebox to aggregate congestion control responses towards the media source so to enable it to adjust its bit-rate (obviously only in case the limitation is not in the source to middlebox link).
This version of the middlebox also puts different requirements on the end-point when it comes to decoder instances and handling of the RTP media streams providing media. As each projected SSRC can, at any time, provide media, the end-point 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, this could potentially be seen as a media translator which include an on/off logic as part of its media translation. The main difference would be a common global SSRC space in the case of the Media Translator and the mapped one used in the above. It also has mixer aspects, as the streams it provides are not basically translated version, but instead they have conceptual property assigned to them. Thus this topology appears to be some hybrid between the translator and mixer model.
Shortcut name: Topo-Video-switch-MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 16: Point to Multipoint Using a Video Switching MCU
This PtM topology was common before, although the RTCP-terminating MCUs, as discussed in the next section, where perhaps even more common. This topology, as well as the following one, was a result of lack of wide availability of IP multicast technologies, as well as the simplicity of content switching when compared to content mixing. The technology is commonly implemented in what is known as "Video Switching MCUs".
A video switching MCU forwards to a participant a single media stream, selected from the available streams. The criteria for selection are often based on voice activity in the audio-visual conference, but other conference management mechanisms (like presentation mode or explicit floor control) are known to exist as well.
The video switching MCU may also perform media translation to modify the content in bit-rate, encoding, or resolution. However, it still may indicate the original sender of the content through the SSRC. In this case, the values of the CC and CSRC fields are retained.
If not terminating RTP, the RTCP Sender Reports are forwarded for the currently selected sender. All RTCP Receiver Reports are freely forwarded between the participants. In addition, the MCU may also originate RTCP control traffic in order to control the session and/or report on status from its viewpoint.
The video switching MCU has most of the attributes of a Translator. However, its stream selection is a mixing behavior. This behavior has some RTP and RTCP issues associated with it. The suppression of all but one media stream results in most participants seeing only a subset of the sent media streams at any given time, often a single stream per conference. Therefore, RTCP Receiver Reports only report on these streams. Consequently, the media senders that are not currently forwarded receive a view of the session that indicates their media streams disappear somewhere en route. This makes the use of RTCP for congestion control, or any type of quality reporting, very problematic.
To avoid the aforementioned issues, the MCU needs to implement two features. First, it needs to act as a Mixer (see Section 3.6) and forward the selected media stream under its own SSRC and with the appropriate CSRC values. Second, the MCU needs to modify the RTCP RRs it forwards between the domains. As a result, it is recommended that one implement a centralized video switching conference using a Mixer according to RFC 3550, instead of the shortcut implementation described here.
Shortcut name: Topo-RTCP-terminating-MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 17: Point to Multipoint Using Content Modifying MCUs
In this PtM scenario, each participant runs an RTP point-to-point session between itself and the MCU. This is a very commonly deployed topology in multipoint video conferencing. The content that the MCU provides to each participant is either:
In case a), the MCU may modify the content in bit-rate, encoding, or resolution. No explicit RTP mechanism is used to establish the relationship between the original media sender and the version the MCU sends. In other words, the outgoing sessions typically use a different SSRC, and may well use a different payload type (PT), even if this different PT happens to be mapped to the same media type. This is a result of the individually negotiated session for each participant.
In case b), the MCU is the content source as it mixes the content and then encodes it for transmission to a participant. According to RTP [RFC3550], the SSRC of the contributors are to be signalled using the CSRC/CC mechanism. In practice, today, most deployed MCUs do not implement this feature. Instead, the identification of the participants whose content is included in the Mixer's output is not indicated through any explicit RTP mechanism. That is, most deployed MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby indicating no available CSRC information, even if they could identify the content sources as suggested in RTP.
The main feature that sets this topology apart from what RFC 3550 describes is the breaking of the common RTP session across the centralized device, such as the MCU. This results in the loss of explicit RTP-level indication of all participants. If one were using the mechanisms available in RTP and RTCP to signal this explicitly, the topology would follow the approach of an RTP Mixer. The lack of explicit indication has at least the following potential problems:
Note that deployed MCUs (and endpoints) rely on signalling layer mechanisms for the identification of the contributing sources, for example, a SIP conferencing package [RFC4575]. This alleviates, to some extent, the aforementioned issues resulting from ignoring RTP's CSRC mechanism.
As a result of the shortcomings of this topology, it is recommended to instead implement the Mixer concept as specified by RFC 3550.
The implementation of an application may desire to send a subset of the application's data to each of multiple devices, each with its own network address. A very basic use case for this would be to separate audio and video processing for a particular endpoint, like a conference room, into one device handling the audio and another handling the video, being interconnected by some control functions allowing them to behave as a single endpoint in all aspects except for transport Figure 18.
Which decomposition scheme is possible is highly dependent on the RTP session usage. It is not really feasible to decompose one logical end-point into two different transport nodes in one RTP session. A third party monitor would report such an attempt as two entities being two different end-points with a CNAME collision. As a result, a fully RTP conformant de-composited endpoint is one where the different decomposed parts use separate RTP sessions to send and/or receive media streams intended for them.
+---------------------+ | Endpoint A | | Local Area Network | | +------------+ | | +->| Audio |<+-RTP---\ | | +------------+ | \ +------+ | | +------------+ | +-->| | | +->| Video |<+-RTP-------->| B | | | +------------+ | +-->| | | | +------------+ | / +------+ | +->| Control |<+-SIP---/ | +------------+ | +---------------------+
Figure 18: De-composite End-Point
In the above usage, let us assume that the different RTP sessions are used for audio and video. The audio and video parts, however, use a common CNAME and also have a common clock to ensure that synchronization and clock drift handling works, despite the decomposition. Also, the RTCP handling works correctly as long as only one part of the de-composite is part of each RTP session. That way any differences in the path between A's audio entity and B and A's video and B are related to different SSRCs in different RTP sessions.
The requirement that can be derived from the above usage is that the transport flows for each RTP session might be under common control, but still are addressed to what looks like different endpoints (based on addresses and ports). This geometry cannot be accomplished using one RTP session, so in this case, multiple RTP sessions are needed.
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 11, where the Mixer does not need to mix in the direction from B or D towards the multicast domains with A and C. Instead, the media streams from B and D are forwarded without changes. Avoiding this mixing would save media processing resources that perform the mixing in cases where it isn't needed. However, there would still be a need to mix B's stream towards D. Only in the direction B -> multicast domain or D -> multicast domain would it be possible to work as a Translator. In all other directions, it would function as a Mixer.
The Mixer/Translator would still need to process and change the RTCP before forwarding it in the directions of B or D to the multicast domain. One issue is that A and C do not know about the mixed-media stream the Mixer sends to either B or D. Therefore, any reports related to these streams must be removed. Also, receiver reports related to A and C's media stream would be missing. To avoid A and C thinking that B and D aren't receiving A and C at all, the Mixer needs to insert locally generated reports reflecting the situation for the streams from A and C into B and D's Sender Reports. In the opposite direction, the Receiver Reports from A and C about B's and D's stream also need to be aggregated into the Mixer's Receiver Reports sent to B and D. Since B and D only have the Mixer as source for the stream, all RTCP from A and C must be suppressed by the Mixer.
This topology is so problematic and it is so easy to get the RTCP processing wrong, that it is not recommended to implement this topology.
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 Source Description (SDES) information for both the CSRCs and the SSRCs. Thus, in a mixed domain, the only SSRCs seen will be the ones present in the domain, while there can be CSRCs from all the domains connected together with a combination of Mixers and Translators. The combined SSRC and CSRC space is common over any Translator or Mixer. This is important to facilitate loop detection, something that is likely to be even more important in combined topologies due to the mixed behavior between the domains. Any hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, requires considerable thought on how RTCP is dealt with.
The topologies discussed in Section 3 have different properties. This section first lists these properties and maps the different topologies to them. Please note that even if a certain property is supported within a particular topology concept, the necessary functionality may, in many cases, be optional to implement.
Note: This section has not yet been updated with the new additions of topologies.
Multicast, at least Any Source Multicast (ASM), provides the functionality that everyone may send to, or receive from, everyone else within the session. MCUs, Mixers, and Translators may all provide that functionality at least on some basic level. However, there are some differences in which type of reachability they provide.
The transport Translator function called "relay", in Section 3.5, is the one that provides the emulation of ASM that is closest to true IP-multicast-based, all to all transmission. Media Translators, Mixers, and the MCU variants do not provide a fully meshed forwarding on the transport level; instead, they only allow limited forwarding of content from the other session participants.
The "all to all media transmission" requires that any media transmitting entity considers the path to the least capable receiver. Otherwise, the media transmissions may overload that path. Therefore, a media sender needs to monitor the path from itself to any of the participants, to detect the currently least capable receiver, and adapt its sending rate accordingly. As multiple participants may send simultaneously, the available resources may vary. RTCP's Receiver Reports help performing this monitoring, at least on a medium time scale.
The transmission of RTCP automatically adapts to any changes in the number of participants due to the transmission algorithm, defined in the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] (when applicable). That way, the resources utilized for RTCP stay within the bounds configured for the session.
Translators, Mixers, and RTCP-terminating MCU all allow changing the media encoding or the transport to other properties of the other domain, thereby providing extended interoperability in cases where the participants lack a common set of media codecs and/or transport protocols.
Participants are most likely to be connected to each other with a heterogeneous set of paths. This makes congestion control in a Point to Multipoint set problematic. For the ASM and "relay" scenario, each individual sender has to adapt to the receiver with the least capable path. This is no longer necessary when Media Translators, Mixers, or MCUs are involved, as each participant only needs to adapt to the slowest path within its own domain. The Translator, Mixer, or MCU topologies all require their respective outgoing streams to adjust the bit-rate, packet-rate, etc., to adapt to the least capable path in each of the other domains. That way one can avoid lowering the quality to the least-capable participant in all the domains at the cost (complexity, delay, equipment) of the Mixer or Translator.
In the all to all media property mentioned above and provided by ASM, all simultaneous media transmissions share the available bit-rate. For participants with limited reception capabilities, this may result in a situation where even a minimal acceptable media quality cannot be accomplished. This is the result of multiple media streams needing to share the available resources. The solution to this problem is to provide for a Mixer or MCU to aggregate the multiple streams into a single one. This aggregation can be performed according to different methods. Mixing or selection are two common methods.
The RTP protocol includes functionality to identify the session participants through the use of the SSRC and CSRC fields. In addition, it is capable of carrying some further identity information about these participants using the RTCP Source Descriptors (SDES). To maintain this functionality, it is necessary that RTCP is handled correctly in domain bridging function. This is specified for Translators and Mixers. The MCU described in Section 3.8 does not entirely fulfill this. The one described in Section 3.9 does not support this at all.
In complex topologies with multiple interconnected domains, it is possible to form media loops. RTP and RTCP support detecting such loops, as long as the SSRC and CSRC identities are correctly set in forwarded packets. It is likely that loop detection works for the MCU, described in Section 3.8, at least as long as it forwards the RTCP between the participants. However, the MCU in Section 3.9 will definitely break the loop detection mechanism.
The table below attempts to summarize the properties of the different topologies. The legend to the topology abbreviations are: Topo-Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-Translator (TTrn), Topo-Media-Translator (including Transport Translator) (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY), Topo-Video-switch-MCU (MCUs), and Topo-RTCP-terminating-MCU (MCUt). In the table below, Y indicates Yes or full support, N indicates No support, (Y) indicates partial support, and N/A indicates not applicable.
Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt ------------------------------------------------------------------ All to All media N Y Y Y (Y) (Y) (Y) (Y) Interoperability N/A N Y Y Y Y N Y Per Domain Adaptation N/A N N Y Y Y N Y Aggregation of media N N N N Y (Y) Y Y Full Session View Y Y Y Y Y Y (Y) N Loop Detection Y Y Y Y Y Y (Y) N
Please note that the Media Translator also includes the transport Translator functionality.
The use of Mixers and Translators has impact on security and the security functions used. The primary issue is that both Mixers and Translators modify packets, thus preventing the use of integrity and source authentication, unless they are trusted devices that take part in the security context, e.g., the device can send Secure Realtime Transport Protocol (SRTP) and Secure Realtime Transport Control Protocol (SRTCP) [RFC3711] packets to session endpoints. If encryption is employed, the media Translator and Mixer need to be able to decrypt the media to perform its function. A transport Translator may be used without access to the encrypted payload in cases where it translates parts that are not included in the encryption and integrity protection, for example, IP address and UDP port numbers in a media stream using SRTP [RFC3711]. However, in general, the Translator or Mixer needs to be part of the signalling context and get the necessary security associations (e.g., SRTP crypto contexts) established with its RTP session participants.
Including the Mixer and Translator in the security context allows the entity, if subverted or misbehaving, to perform a number of very serious attacks as it has full access. It can perform all the attacks possible (see RFC 3550 and any applicable profiles) as if the media session were not protected at all, while giving the impression to the session participants that they are protected.
Transport Translators have no interactions with cryptography that works above the transport layer, such as SRTP, since that sort of Translator leaves the RTP header and payload unaltered. Media Translators, on the other hand, have strong interactions with cryptography, since they alter the RTP payload. A media Translator in a session that uses cryptographic protection needs to perform cryptographic processing to both inbound and outbound packets.
A media Translator may need to use different cryptographic keys for the inbound and outbound processing. For SRTP, different keys are required, because an RFC 3550 media Translator leaves the SSRC unchanged during its packet processing, and SRTP key sharing is only allowed when distinct SSRCs can be used to protect distinct packet streams.
When the media Translator uses different keys to process inbound and outbound packets, each session participant needs to be provided with the appropriate key, depending on whether they are listening to the Translator or the original source. (Note that there is an architectural difference between RTP media translation, in which participants can rely on the RTP Payload Type field of a packet to determine appropriate processing, and cryptographically protected media translation, in which participants must use information that is not carried in the packet.)
When using security mechanisms with Translators and Mixers, it is possible that the Translator or Mixer could create different security associations for the different domains they are working in. Doing so has some implications:
First, it might weaken security if the Mixer/Translator accepts a weaker algorithm or key in one domain than in another. Therefore, care should be taken that appropriately strong security parameters are negotiated in all domains. In many cases, "appropriate" translates to "similar" strength. If a key management system does allow the negotiation of security parameters resulting in a different strength of the security, then this system should notify the participants in the other domains about this.
Second, the number of crypto contexts (keys and security related state) needed (for example, in SRTP [RFC3711]) may vary between Mixers and Translators. A Mixer normally needs to represent only a single SSRC per domain and therefore needs to create only one security association (SRTP crypto context) per domain. In contrast, a Translator needs one security association per participant it translates towards, in the opposite domain. Considering Figure 9, the Translator needs two security associations towards the multicast domain, one for B and one for D. It may be forced to maintain a set of totally independent security associations between itself and B and D respectively, so as to avoid two-time pad occurrences. These contexts must also be capable of handling all the sources present in the other domains. Hence, using completely independent security associations (for certain keying mechanisms) may force a Translator to handle N*DM keys and related state; where N is the total number of SSRCs used over all domains and DM is the total number of domains.
There exist a number of different mechanisms to provide keys to the different participants. One example is the choice between group keys and unique keys per SSRC. The appropriate keying model is impacted by the topologies one intends to use. The final security properties are dependent on both the topologies in use and the keying mechanisms' properties, and need to be considered by the application. Exactly which mechanisms are used is outside of the scope of this document. Please review RTP Security Options [I-D.ietf-avtcore-rtp-security-options] to get a better understanding of most of the available options.
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an RFC.
The authors would like to thank Bo Burman, Umesh Chandra, Roni Even, Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their help in reviewing this document.
| [RFC3550] | Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. |
| [RFC3711] | Baugher, M., McGrew, D., Naslund, M., Carrara, E. and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. |
| [RFC4575] | Rosenberg, J., Schulzrinne, H. and O. Levin, "A Session Initiation Protocol (SIP) Event Package for Conference State", RFC 4575, August 2006. |
| [RFC4585] | Ott, J., Wenger, S., Sato, N., Burmeister, C. and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006. |