Internet Engineering Task Force (IETF) J. Lennox Request for Comments: 8108 Vidyo Updates: 3550, 4585 M. Westerlund Category: Standards Track Ericsson ISSN: 2070-1721 Q. Wu Huawei C. Perkins University of Glasgow March 2017 Sending Multiple RTP Streams in a Single RTP Session Abstract This memo expands and clarifies the behavior of Real-time Transport Protocol (RTP) endpoints that use multiple synchronization sources (SSRCs). This occurs, for example, when an endpoint sends multiple RTP streams in a single RTP session. This memo updates RFC 3550 with regard to handling multiple SSRCs per endpoint in RTP sessions, with a particular focus on RTP Control Protocol (RTCP) behavior. It also updates RFC 4585 to change and clarify the calculation of the timeout of SSRCs and the inclusion of feedback messages. Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc8108.
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Table of Contents 1. Introduction ....................................................4 2. Terminology .....................................................4 3. Use Cases for Multi-Stream Endpoints ............................4 3.1. Endpoints with Multiple Capture Devices ....................4 3.2. Multiple Media Types in a Single RTP Session ...............5 3.3. Multiple Stream Mixers .....................................5 3.4. Multiple SSRCs for a Single Media Source ...................5 4. Use of RTP by Endpoints That Send Multiple Media Streams ........6 5. Use of RTCP by Endpoints That Send Multiple Media Streams .......6 5.1. RTCP Reporting Requirement .................................7 5.2. Initial Reporting Interval .................................7 5.3. Aggregation of Reports into Compound RTCP Packets ..........8 5.3.1. Maintaining AVG_RTCP_SIZE ...........................9 5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs ....10 5.4. Use of RTP/AVPF or RTP/SAVPF Feedback .....................13 5.4.1. Choice of SSRC for Feedback Packets ................13 5.4.2. Scheduling an RTCP Feedback Packet .................14 6. Adding and Removing SSRCs ......................................15 6.1. Adding RTP Streams ........................................16 6.2. Removing RTP Streams ......................................16 7. RTCP Considerations for Streams with Disparate Rates ...........17 7.1. Timing Out SSRCs ..........................................19 7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter ..........................................19 7.1.2. Avoiding Premature Timeout .........................20 7.1.3. Interoperability between RTP/AVP and RTP/AVPF ......21 7.1.4. Updated SSRC Timeout Rules .........................22 7.2. Tuning RTCP Transmissions .................................22 7.2.1. RTP/AVP and RTP/SAVP ...............................22 7.2.2. RTP/AVPF and RTP/SAVPF .............................24 8. Security Considerations ........................................25 9. References .....................................................26 9.1. Normative References ......................................26 9.2. Informative References ....................................26 Acknowledgments ...................................................29 Authors' Addresses ................................................29
1. Introduction At the time the Real-Time Transport Protocol (RTP) [RFC3550] was originally designed, and for quite some time after, endpoints in RTP sessions typically only transmitted a single media source and, thus, used a single RTP stream and synchronization source (SSRC) per RTP session, where separate RTP sessions were typically used for each distinct media type. Recently, however, a number of scenarios have emerged in which endpoints wish to send multiple RTP streams, distinguished by distinct RTP synchronization source (SSRC) identifiers, in a single RTP session. These are outlined in Section 3. Although the initial design of RTP did consider such scenarios, the specification was not consistently written with such use cases in mind; thus, the specification is somewhat unclear in places. This memo updates [RFC3550] to clarify behavior in use cases where endpoints use multiple SSRCs. It also updates [RFC4585] to resolve problems with regard to timeout of inactive SSRCs and to clarify behavior around inclusion of feedback messages. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119] and indicate requirement levels for compliant implementations. 3. Use Cases for Multi-Stream Endpoints This section discusses several use cases that have motivated the development of endpoints that sends RTP data using multiple SSRCs in a single RTP session. 3.1. Endpoints with Multiple Capture Devices The most straightforward motivation for an endpoint to send multiple simultaneous RTP streams in a single RTP session is when an endpoint has multiple capture devices and, hence, can generate multiple media sources, of the same media type and characteristics. For example, telepresence systems of the type described by the CLUE Telepresence Framework [CLUE-FRAME] often have multiple cameras or microphones covering various areas of a room and, hence, send several RTP streams of each type within a single RTP session.
3.2. Multiple Media Types in a Single RTP Session Recent work has updated RTP [MULTI-RTP] and Session Description Protocol (SDP) [SDP-BUNDLE] to remove the historical assumption in RTP that media sources of different media types would always be sent on different RTP sessions. In this work, a single endpoint's audio and video RTP streams (for example) are instead sent in a single RTP session to reduce the number of transport-layer flows used. 3.3. Multiple Stream Mixers There are several RTP topologies that can involve a central device that itself generates multiple RTP streams in a session. An example is a mixer providing centralized compositing for a multi-capture scenario like that described in Section 3.1. In this case, the centralized node is behaving much like a multi-capturer endpoint, generating several similar and related sources. A more complex example is the selective forwarding middlebox, described in Section 3.7 of [RFC7667]. This is a middlebox that receives RTP streams from several endpoints and then selectively forwards modified versions of some RTP streams toward the other endpoints to which it is connected. For each connected endpoint, a separate media source appears in the session for every other source connected to the middlebox, "projected" from the original streams, but at any given time many of them can appear to be inactive (and thus are receivers, not senders, in RTP). This sort of device is closer to being an RTP mixer than an RTP translator: it terminates RTCP reporting about the mixed streams; it can rewrite SSRCs, timestamps, and sequence numbers, as well as the contents of the RTP payloads; and it can turn sources on and off at will without appearing to generate packet loss. Each projected stream will typically preserve its original RTCP source description (SDES) information. 3.4. Multiple SSRCs for a Single Media Source There are also several cases where multiple SSRCs can be used to send data from a single media source within a single RTP session. These include, but are not limited to, transport robustness tools, such as the RTP retransmission payload format [RFC4588], that require one SSRC to be used for the media data and another SSRC for the repair data. Similarly, some layered media encoding schemes, for example, H.264 Scalable Video Coding (SVC) [RFC6190], can be used in a configuration where each layer is sent using a different SSRC within a single RTP session.
4. Use of RTP by Endpoints That Send Multiple Media Streams RTP is inherently a group communication protocol. Each endpoint in an RTP session will use one or more SSRCs, as will some types of RTP- level middlebox. Accordingly, unless restrictions on the number of SSRCs have been signaled, RTP endpoints can expect to receive RTP data packets sent using a number of different SSRCs, within a single RTP session. This can occur irrespective of whether the RTP session is running over a point-to-point connection or a multicast group, since middleboxes can be used to connect multiple transport connections together into a single RTP session (the RTP session is defined by the shared SSRC space, not by the transport connections). Furthermore, if RTP mixers are used, some SSRCs might only be visible in the contributing source (CSRC) list of an RTP packet and in RTCP, and might not appear directly as the SSRC of an RTP data packet. Every RTP endpoint will have an allocated share of the available session bandwidth, as determined by signaling and congestion control. The endpoint needs to keep its total media sending rate within this share. However, endpoints that send multiple RTP streams do not necessarily need to subdivide their share of the available bandwidth independently or uniformly to each RTP stream and its SSRCs. In particular, an endpoint can vary the bandwidth allocation to different streams depending on their needs, and it can dynamically change the bandwidth allocated to different SSRCs (for example, by using a variable-rate codec), provided the total sending rate does not exceed its allocated share. This includes enabling or disabling RTP streams, or their redundancy streams, as more or less bandwidth becomes available. 5. Use of RTCP by Endpoints That Send Multiple Media Streams RTCP is defined in Section 6 of [RFC3550]. The description of the protocol is phrased in terms of the behavior of "participants" in an RTP session, under the assumption that each endpoint is a participant with a single SSRC. However, for correct operation in cases where endpoints have multiple SSRC values, implementations MUST treat each SSRC as a separate participant in the RTP session, so that an endpoint that has multiple SSRCs counts as multiple participants.
5.1. RTCP Reporting Requirement An RTP endpoint that has multiple SSRCs MUST treat each SSRC as a separate participant in the RTP session. Each SSRC will maintain its own RTCP-related state information and, hence, will have its own RTCP reporting interval that determines when it sends RTCP reports. If the mechanism in [MULTI-STREAM-OPT] is not used, then each SSRC will send RTCP reports for all other SSRCs, including those co-located at the same endpoint. If the endpoint has some SSRCs that are sending data and some that are only receivers, then they will receive different shares of the RTCP bandwidth and calculate different base RTCP reporting intervals. Otherwise, all SSRCs at an endpoint will calculate the same base RTCP reporting interval. The actual reporting intervals for each SSRC are randomized in the usual way, but reports can be aggregated as described in Section 5.3. 5.2. Initial Reporting Interval When a participant joins a unicast session, the following text from Section 6.2 of [RFC3550] is relevant: "For unicast sessions... the delay before sending the initial compound RTCP packet MAY be zero." The basic assumption is that this also ought to apply in the case of multiple SSRCs. Caution has to be exercised, however, when an endpoint (or middlebox) with a large number of SSRCs joins a unicast session, since immediate transmission of many RTCP reports can create a significant burst of traffic, leading to transient congestion and packet loss due to queue overflows. To ensure that the initial burst of traffic generated by an RTP endpoint is no larger than would be generated by a TCP connection, an RTP endpoint MUST NOT send more than four compound RTCP packets with zero initial delay when it joins an RTP session, independent of the number of SSRCs used by the endpoint. Each of those initial compound RTCP packets MAY include aggregated reports from multiple SSRCs, provided the total compound RTCP packet size does not exceed the MTU, and the avg_rtcp_size is maintained as in Section 5.3.1. Aggregating reports from several SSRCs in the initial compound RTCP packets allows a substantial number of SSRCs to report immediately. Endpoints SHOULD prioritize reports on SSRCs that are likely to be most immediately useful, e.g., for SSRCs that are initially senders. An endpoint that needs to report on more SSRCs than will fit into the four compound RTCP reports that can be sent immediately MUST send the other reports later, following the usual RTCP timing rules including timer reconsideration. Those reports MAY be aggregated as described in Section 5.3.
Note: The above is chosen to match the TCP maximum initial window of four packets [RFC3390], not the larger TCP initial windows for which there is an ongoing experiment [RFC6928]. The reason for this is a desire to be conservative, since an RTP endpoint will also in many cases start sending RTP data packets at the same time as these initial RTCP packets are sent. 5.3. Aggregation of Reports into Compound RTCP Packets As outlined in Section 5.1, an endpoint with multiple SSRCs has to treat each SSRC as a separate participant when it comes to sending RTCP reports. This will lead to each SSRC sending a compound RTCP packet in each reporting interval. Since these packets are coming from the same endpoint, it might reasonably be expected that they can be aggregated to reduce overheads. Indeed, Section 6.1 of [RFC3550] allows RTP translators and mixers to aggregate packets in similar circumstances: It is RECOMMENDED that translators and mixers combine individual RTCP packets from the multiple sources they are forwarding into one compound packet whenever feasible in order to amortize the packet overhead (see Section 7). An example RTCP compound packet as might be produced by a mixer is shown in Fig. 1. If the overall length of a compound packet would exceed the MTU of the network path, it SHOULD be segmented into multiple shorter compound packets to be transmitted in separate packets of the underlying protocol. This does not impair the RTCP bandwidth estimation because each compound packet represents at least one distinct participant. Note that each of the compound packets MUST begin with an SR or RR packet. This allows RTP translators and mixers to generate compound RTCP packets that contain multiple Sender Report (SR) or Receiver Report (RR) packets from different SSRCs, as well as any of the other packet types. There are no restrictions on the order in which the RTCP packets can occur within the compound packet, except the regular rule that the compound RTCP packet starts with an SR or RR packet. Due to this rule, correctly implemented RTP endpoints will be able to handle compound RTCP packets that contain RTCP packets relating to multiple SSRCs. Accordingly, endpoints that use multiple SSRCs can aggregate the RTCP packets sent by their different SSRCs into compound RTCP packets, provided 1) the resulting compound RTCP packets begin with an SR or RR packet, 2) they maintain the average RTCP packet size as described in Section 5.3.1, and 3) they schedule packet transmission and manage aggregation as described in Section 5.3.2.
5.3.1. Maintaining AVG_RTCP_SIZE The RTCP scheduling algorithm in [RFC3550] works on a per-SSRC basis. Each SSRC sends a single compound RTCP packet in each RTCP reporting interval. When an endpoint uses multiple SSRCs, it is desirable to aggregate the compound RTCP packets sent by its SSRCs, reducing the overhead by forming a larger compound RTCP packet. This aggregation can be done as described in Section 5.3.2, provided the average RTCP packet size calculation is updated as follows. Participants in an RTP session update their estimate of the average RTCP packet size (avg_rtcp_size) each time they send or receive an RTCP packet (see Section 6.3.3 of [RFC3550]). When a compound RTCP packet that contains RTCP packets from several SSRCs is sent or received, the avg_rtcp_size estimate for each SSRC that is reported upon is updated using div_packet_size rather than the actual packet size: avg_rtcp_size = (1/16) * div_packet_size + (15/16) * avg_rtcp_size where div_packet_size is packet_size divided by the number of SSRCs reporting in that compound packet. The number of SSRCs reporting in a compound packet is determined by counting the number of different SSRCs that are the source of SR or RR RTCP packets within the compound RTCP packet. Non-compound RTCP packets (i.e., RTCP packets that do not contain an SR or RR packet [RFC5506]) are considered to report on a single SSRC. A participant that doesn't follow the above rule, and instead uses the full RTCP compound packet size to calculate avg_rtcp_size, will derive an RTCP reporting interval that is overly large by a factor that is proportional to the number of SSRCs aggregated into compound RTCP packets and the size of set of SSRCs being aggregated relative to the total number of participants. This increased RTCP reporting interval can cause premature timeouts if it is more than five times the interval chosen by the SSRCs that understand compound RTCP that aggregate reports from many SSRCs. A 1500-octet MTU can fit five typical-size reports into a compound RTCP packet, so this is a real concern if endpoints aggregate RTCP reports from multiple SSRCs. The issue raised in the previous paragraph is mitigated by the modification in timeout behavior specified in Section 7.1.2 of this memo. This mitigation is in place in those cases where the RTCP bandwidth is sufficiently high that an endpoint, using avg_rtcp_size calculated without taking into account the number of reporting SSRCs, can transmit more frequently than approximately every 5 seconds. Note, however, that the non-updated endpoint's RTCP reporting is still negatively impacted even if the premature timeouts of its SSRCs
are avoided. If compatibility with non-updated endpoints is a concern, the number of reports from different SSRCs aggregated into a single compound RTCP packet SHOULD either be limited to two reports or aggregation ought not be used at all. This will limit the non-updated endpoint's RTCP reporting interval to be no larger than twice the RTCP reporting interval that would be chosen by an endpoint following this specification. 5.3.2. Scheduling RTCP when Aggregating Multiple SSRCs This section revises and extends the behavior defined in Section 6.3 of [RFC3550], and in Section 3.5.3 of [RFC4585] if the RTP/AVPF profile or the RTP/SAVPF profile is used, regarding actions to take when scheduling and sending RTCP packets where multiple reporting SSRCs are aggregating their RTCP packets into the same compound RTCP packet. These changes to the RTCP scheduling rules are needed to maintain important RTCP timing properties, including the inter-packet distribution, and the behavior during flash joins and other changes in session membership. The variables tn, tp, tc, T, and Td used in the following are defined in Section 6.3 of [RFC3550]. The variables T_rr_interval and T_rr_last are defined in [RFC4585]. Each endpoint MUST schedule RTCP transmission independently for each of its SSRCs using the regular calculation of tn for the RTP profile being used. Each time the timer tn expires for an SSRC, the endpoint MUST perform RTCP timer reconsideration and, if applicable, suppression based on T_rr_interval. If the result indicates that a compound RTCP packet is to be sent by that SSRC, and the transmission is not an early RTCP packet [RFC4585], then the endpoint SHOULD try to aggregate RTCP packets of additional SSRCs that are scheduled in the future into the compound RTCP packet before it is sent. The reason to limit or not aggregate due to backwards compatibility reasons is discussed in Section 5.3.1. Aggregation proceeds as follows. The endpoint selects the SSRC that has the smallest tn value after the current time, tc, and prepares the RTCP packets that SSRC would send if its timer tn expired at tc. If those RTCP packets will fit into the compound RTCP packet that is being generated, taking into account the path MTU and the previously added RTCP packets, then they are added to the compound RTCP packet; otherwise, they are discarded. This process is repeated for each SSRC, in order of increasing tn, until the compound RTCP packet is full or all SSRCs have been aggregated. At that point, the compound RTCP packet is sent.
When the compound RTCP packet is sent, the endpoint MUST update tp, tn, and T_rr_last (if applicable) for each SSRC that was included. These variables are updated as follows: a. For the first SSRC that reported in the compound RTCP packet, set the effective transmission time, tt, of that SSRC to tc. b. For each additional SSRC that reported in the compound RTCP packet, calculate the transmission time that SSRC would have had if it had not been aggregated into the compound RTCP packet. This is derived by taking tn for that SSRC, then performing reconsideration and updating tn until tp + T <= tn. Once this is done, set the effective transmission time, tt, for that SSRC to the calculated value of tn. If the RTP/AVPF profile or the RTP/ SAVPF profile is being used, then suppression based on T_rr_interval MUST NOT be used in this calculation. c. Calculate average effective transmission time, tt_avg, for the compound RTCP packet based on the tt values for all SSRCs sent in the compound RTCP packet. Set tp for each of the SSRCs sent in the compound RTCP packet to tt_avg. If the RTP/AVPF profile or the RTP/SAVPF profile is being used, set T_tt_last for each SSRC sent in the compound RTCP packet to tt_avg. d. For each of the SSRCs sent in the compound RTCP packet, calculate new tn values based on the updated parameters and the usual RTCP timing rules and reschedule the timers. When using the RTP/AVPF profile or the RTP/SAVPF profile, the above mechanism only attempts to aggregate RTCP packets when the compound RTCP packet to be sent is not an early RTCP packet, and hence the algorithm in Section 3.5.3 of [RFC4585] will control RTCP scheduling. If T_rr_interval == 0, or if T_rr_interval != 0 and option 1, 2a, or 2b of the algorithm are chosen, then the above mechanism updates the necessary variables. However, if the transmission is suppressed per option 2c of the algorithm, then tp is updated to tc as aggregation has not taken place. Reverse reconsideration MUST be performed following Section 6.3.4 of [RFC3550]. In some cases, this can lead to the value of tp after reverse reconsideration being larger than tc. This is not a problem, and has the desired effect of proportionally pulling the tp value towards tc (as well as tn) as the reporting interval shrinks in direct proportion the reduced group size. The above algorithm has been shown in simulations [Sim88] [Sim92] to maintain the inter-RTCP packet transmission time distribution for each SSRC and to consume the same amount of bandwidth as
non-aggregated RTCP packets. With this algorithm, the actual transmission interval for an SSRC triggering an RTCP compound packet transmission is following the regular transmission rules. The value tp is set to somewhere in the interval [0, 1.5/1.21828*Td] ahead of tc. The actual value is the average of one instance of tc and the randomized transmission times of the additional SSRCs; thus, the lower range of the interval is more probable. This compensates for the bias that is otherwise introduced by picking the shortest tn value out of the N SSRCs included in aggregate. The algorithm also handles the cases where the number of SSRCs that can be included in an aggregated packet varies. An SSRC that previously was aggregated and fails to fit in a packet still has its own transmission scheduled according to normal rules. Thus, it will trigger a transmission in due time, or the SSRC will be included in another aggregate. The algorithm's behavior under SSRC group size changes is as follows: RTP sessions where the number of SSRCs is growing: When the group size is growing, Td grows in proportion to the number of new SSRCs in the group. When reconsideration is performed due to expiry of the tn timer, that SSRC will reconsider the transmission and with a certain probability reschedule the tn timer. This part of the reconsideration algorithm is only impacted by the above algorithm having tp values that were in the future instead of set to the time of the actual last transmission at the time of updating tp. RTP sessions where the number of SSRCs is shrinking: When the group shrinks, reverse reconsideration moves the tp and tn values towards tc proportionally to the number of SSRCs that leave the session compared to the total number of participants when they left. The setting of the tp value forward in time related to the tc could be believed to have negative effect. However, the reason for this setting is to compensate for bias caused by picking the shortest tn out of the N aggregated. This bias remains over a reduction in the number of SSRCs. The reverse reconsideration compensates the reduction independently of whether or not aggregation is being used. The negative effect that can occur on removing an SSRC is that the most favorable tn belonged to the removed SSRC. The impact of this is limited to delaying the transmission, in the worst case, one reporting interval. In conclusion, the investigations performed have found no significant negative impact on the scheduling algorithm.
5.4. Use of RTP/AVPF or RTP/SAVPF Feedback This section discusses the transmission of RTP/AVPF feedback packets when the transmitting endpoint has multiple SSRCs. The guidelines in this section also apply to endpoints using the RTP/SAVPF profile. 5.4.1. Choice of SSRC for Feedback Packets When an RTP/AVPF endpoint has multiple SSRCs, it can choose what SSRC to use as the source for the RTCP feedback packets it sends. Several factors can affect that choice: o RTCP feedback packets relating to a particular media type SHOULD be sent by an SSRC that receives that media type. For example, when audio and video are multiplexed onto a single RTP session, endpoints will use their audio SSRC to send feedback on the audio received from other participants. o RTCP feedback packets and RTCP codec control messages that are notifications or indications regarding RTP data processed by an endpoint MUST be sent from the SSRC used for that RTP data. This includes notifications that relate to a previously received request or command [RFC4585][RFC5104]. o If separate SSRCs are used to send and receive media, then the corresponding SSRC SHOULD be used for feedback, since they have differing RTCP bandwidth fractions. This can also affect the consideration of whether or not the SSRC can be used in immediate mode. o Some RTCP feedback packet types require consistency in the SSRC used. For example, if a Temporary Maximum Media Stream Bit Rate Request (TMMBR) limitation [RFC5104] is set by an SSRC, the same SSRC needs to be used to remove the limitation. o If several SSRCs are suitable for sending feedback, it might be desirable to use an SSRC that allows the sending of feedback as an early RTCP packet. When an RTCP feedback packet is sent as part of a compound RTCP packet that aggregates reports from multiple SSRCs, there is no requirement that the compound packet contain an SR or RR packet generated by the sender of the RTCP feedback packet. For reduced- size RTCP packets, aggregation of RTCP feedback packets from multiple sources is not limited further than Section 4.2.2 of [RFC5506].
5.4.2. Scheduling an RTCP Feedback Packet When an SSRC has a need to transmit a feedback packet in early mode, it MUST schedule that packet following the algorithm in Section 3.5 of [RFC4585] modified as follows: o To determine whether an RTP session is considered to be a point- to-point session or a multiparty session, an endpoint MUST count the number of distinct RTCP SDES CNAME values used by the SSRCs listed in the SSRC field of RTP data packets it receives and in the "SSRC of sender" field of RTCP SR, RR, RTPFB, or PSFB packets it receives. An RTP session is considered to be a multiparty session if more than one CNAME is used by those SSRCs, unless signaling indicates that the session is to be handled as point to point or RTCP reporting groups [MULTI-STREAM-OPT] are used. If RTCP reporting groups are used, an RTP session is considered to be a point-to-point session if the endpoint receives only a single reporting group and is considered to be a multiparty session if multiple reporting groups are received or a combination of reporting groups and SSRCs that are not part of a reporting group are received. Endpoints MUST NOT determine whether an RTP session is multiparty or point to point based on the type of connection (unicast or multicast) used, or on the number of SSRCs received. o When checking if there is already a scheduled compound RTCP packet containing feedback messages (Step 2 in Section 3.5.2 of [RFC4585]), that check MUST be done considering all local SSRCs. o If an SSRC is not allowed to send an early RTCP packet, then the feedback message MAY be queued for transmission as part of any early or regular scheduled transmission that can occur within the maximum useful lifetime of the feedback message (T_max_fb_delay). This modifies the behavior in item 4a in Section 3.5.2 of [RFC4585]. The first bullet point above specifies a rule to determine if an RTP session is to be considered a point-to-point session or a multiparty session. This rule is straightforward to implement, but is known to incorrectly classify some sessions as multiparty sessions. The known problems are as follows: Endpoint with multiple synchronization contexts: An endpoint that is part of a point-to-point session can have multiple synchronization contexts, for example, due to forwarding an external media source into an interactive real-time conversation. In this case, the classification will consider the peer as two endpoints, while the actual RTP/RTCP transmission will be under the control of one endpoint.
Selective Forwarding Middlebox: The Selective Forwarding Middlebox (SFM) as defined in Section 3.7 of [RFC7667] has control over the transmission and configurations between itself and each peer endpoint individually. It also fully controls the RTCP packets being forwarded between the individual legs. Thus, this type of middlebox can be compared to the RTP mixer, which uses its own SSRCs to mix or select the media it forwards, that will be classified as a point-to-point RTP session by the above rule. In the above cases, it is very reasonable to use RTCP reporting groups [MULTI-STREAM-OPT]. If that extension is used, an endpoint can indicate that the multitude of CNAMEs are in fact under a single endpoint or middlebox control by using only a single reporting group. The above rules will also classify some sessions where the endpoint is connected to an RTP mixer as being point to point. For example, the mixer could act as gateway to an RTP session based on Any Source Multicast for the discussed endpoint. However, this will, in most cases, be okay, as the RTP mixer provides separation between the two parts of the session. The responsibility falls on the mixer to act accordingly in each domain. Finally, we note that signaling mechanisms could be defined to override the rules when they would result in the wrong classification.