6. Adding and Removing SSRCs The set of SSRCs present in a single RTP session can vary over time due to changes in the number of endpoints in the session or due to changes in the number or type of RTP streams being sent. Every endpoint in an RTP session will have at least one SSRC that it uses for RTCP reporting, and for sending media if desired. It can also have additional SSRCs, for sending extra media sources or for additional RTCP reporting. If the set of media sources being sent changes, then the set of SSRCs being sent will change. Changes in the media format or clock rate might also require changes in the set of SSRCs used. An endpoint can also have more SSRCs than it has active RTP streams, and send RTCP relating to SSRCs that are not currently sending RTP data packets so that its peers are aware of the SSRCs, and have the associated context (e.g., clock synchronization and an SDES CNAME) in place to be able to play out media as soon as they becomes active. In the following, we describe some considerations around adding and removing RTP streams and their associated SSRCs.
6.1. Adding RTP Streams When an endpoint joins an RTP session, it can have zero, one, or more RTP streams it will send, or that it is prepared to send. If it has no RTP stream it plans to send, it still needs an SSRC that will be used to send RTCP feedback. If it will send one or more RTP streams, it will need the corresponding number of SSRC values. The SSRCs used by an endpoint are made known to other endpoints in the RTP session by sending RTP and RTCP packets. SSRCs can also be signaled using non-RTP means (e.g., [RFC5576]). Unless restricted by signaling, an endpoint can, at any time, send an additional RTP stream, identified by a new SSRC (this might be associated with a signaling event, but that is outside the scope of this memo). This makes the new SSRC visible to the other endpoints in the session, since they share the single SSRC space inherent in the definition of an RTP session. An endpoint that has never sent an RTP stream will have an SSRC that it uses for RTCP reporting. If that endpoint wants to start sending an RTP stream, it is RECOMMENDED that it use its existing SSRC for that stream, since otherwise the participant count in the RTP session will be unnecessarily increased, leading to a longer RTCP reporting interval and larger RTCP reports due to cross reporting. If the endpoint wants to start sending more than one RTP stream, it will need to generate a new SSRC for the second and any subsequent RTP streams. An endpoint that has previously stopped sending an RTP stream, and that wants to start sending a new RTP stream, cannot generally reuse the existing SSRC, and often needs to generate a new SSRC, because an SSRC cannot change media type (e.g., audio to video) or RTP timestamp clock rate [RFC7160] and because the SSRC might be associated with a particular semantic by the application (note: an RTP stream can pause and restart using the same SSRC, provided RTCP is sent for that SSRC during the pause; these rules only apply to new RTP streams reusing an existing SSRC). 6.2. Removing RTP Streams An SSRC is removed from an RTP session in one of two ways. When an endpoint stops sending RTP and RTCP packets using an SSRC, then that SSRC will eventually time out as described in Section 6.3.5 of [RFC3550]. Alternatively, an SSRC can be explicitly removed from use by sending an RTCP BYE packet as described in Section 6.3.7 of [RFC3550]. It is RECOMMENDED that SSRCs be removed from use by sending an RTCP BYE packet. Note that [RFC3550] requires that the RTCP BYE SHOULD be the last RTP/RTCP packet sent in the RTP session
for an SSRC. If an endpoint needs to restart an RTP stream after sending an RTCP BYE for its SSRC, it needs to generate a new SSRC value for that stream. The finality of sending RTCP BYE means that endpoints need to consider if the ceasing of transmission of an RTP stream is temporary or permanent. Temporary suspension of media transmission using a particular RTP stream (SSRC) needs to maintain that SSRC as an active participant, by continuing RTCP transmission for it. That way the media sending can be resumed immediately, knowing that the context is in place. When permanently halting transmission, a participant needs to send an RTCP BYE to allow the other participants to use the RTCP bandwidth resources and clean up their state databases. An endpoint that ceases transmission of all its RTP streams but remains in the RTP session MUST maintain at least one SSRC that is to be used for RTCP reporting and feedback (i.e., it cannot send a BYE for all SSRCs, but needs to retain at least one active SSRC). As some Feedback packets can be bound to media type, there might be a need to maintain one SSRC per media type within an RTP session. An alternative can be to create a new SSRC to use for RTCP reporting and feedback. However, to avoid the perception that an endpoint drops completely out of an RTP session, such a new SSRC ought to be established first -- before terminating all the existing SSRCs. 7. RTCP Considerations for Streams with Disparate Rates An RTP session has a single set of parameters that configure the session bandwidth. These are the RTCP sender and receiver fractions (e.g., the SDP "b=RR:" and "b=RS:" lines [RFC3556]) and the parameters of the RTP/AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its secure extension, RTP/SAVPF [RFC5124]) is used. As a consequence, the base RTCP reporting interval, before randomization, will be the same for every sending SSRC in an RTP session. Similarly, every receiving SSRC in an RTP session will have the same base reporting interval, although this can differ from the reporting interval chosen by sending SSRCs. This uniform RTCP reporting interval for all SSRCs can result in RTCP reports being sent more often, or too seldom, than is considered desirable for an RTP stream. For example, consider a scenario in which an audio flow sending at tens of kilobits per second is multiplexed into an RTP session with a multi-megabit high-quality video flow. If the session bandwidth is configured based on the video sending rate, and the default RTCP bandwidth fraction of 5% of the session bandwidth is used, it is likely that the RTCP bandwidth will exceed the audio sending rate. If the reduced minimum RTCP interval described in Section 6.2 of [RFC3550] is then used in the session, as appropriate for video where
rapid feedback on damaged I-frames is wanted, the uniform reporting interval for all senders could mean that audio sources are expected to send RTCP packets more often than they send audio data packets. This bandwidth mismatch can be reduced by careful tuning of the RTCP parameters, especially trr_int when the RTP/AVPF profile is used, but cannot be avoided entirely as it is inherent in the design of the RTCP timing rules, and affects all RTP sessions that contain flows with greatly mismatched bandwidth. Different media rates or desired RTCP behaviors can also occur with SSRCs carrying the same media type. A common case in multiparty conferencing is when a small number of video streams are shown in high resolution, while the others are shown as low-resolution thumbnails, with the choice of which is shown in high resolution being voice-activity controlled. Here the differences are both in actual media rate and in choices for what feedback messages might be needed. Other examples of differences that can exist are due to the intended usage of a media source. A media source carrying the video of the speaker in a conference is different from a document camera. Basic parameters that can differ in this case are frame-rate, acceptable end-to-end delay, and the Signal-to-Noise Ratio (SNR) fidelity of the image. These differences affect not only the needed bitrates, but also possible transmission behaviors, usable repair mechanisms, what feedback messages the control and repair requires, the transmission requirements on those feedback messages, and monitoring of the RTP stream delivery. Other similar scenarios can also exist. Sending multiple media types in a single RTP session causes that session to contain more SSRCs than if each media type was sent in a separate RTP session. For example, if two participants each send an audio and a video RTP stream in a single RTP session, that session will comprise four SSRCs; but if separate RTP sessions had been used for audio and video, each of those two RTP sessions would comprise only two SSRCs. Hence, sending multiple RTP streams in an RTP session increases the amount of cross reporting between the SSRCs, as each SSRC reports on all other SSRCs in the session. This increases the size of the RTCP reports, causing them to be sent less often than would be the case if separate RTP sessions where used for a given RTCP bandwidth. Finally, when an RTP session contains multiple media types, it is important to note that the RTCP reception quality reports, feedback messages, and extended report blocks used might not be applicable to all media types. Endpoints will need to consider the media type of each SSRC, and only send or process reports and feedback that apply to that particular SSRC and its media type. Signaling solutions
might have shortcomings when it comes to indicating that a particular set of RTCP reports or feedback messages only apply to a particular media type within an RTP session. From an RTCP perspective, therefore, it can be seen that there are advantages to using separate RTP sessions for each media source, rather than sending multiple media sources in a single RTP session. However, these are frequently offset by the need to reduce port use, to ease NAT/firewall traversal, achieved by combining media sources into a single RTP session. The following sections consider some of the issues with using RTCP in sessions with multiple media sources in more detail. 7.1. Timing Out SSRCs Various issues have been identified with timing out SSRC values when sending multiple RTP streams in an RTP session. 7.1.1. Problems with the RTP/AVPF T_rr_interval Parameter The RTP/AVPF profile includes a method to prevent regular RTCP reports from being sent too often. This mechanism is described in Section 3.5.3 of [RFC4585]; it is controlled by the T_rr_interval parameter. It works as follows. When a regular RTCP report is sent, a new random value, T_rr_current_interval, is generated, drawn evenly in the range 0.5 to 1.5 times T_rr_interval. If a regular RTCP packet is to be sent earlier than T_rr_current_interval seconds after the previous regular RTCP packet, and there are no feedback messages to be sent, then that regular RTCP packet is suppressed and the next regular RTCP packet is scheduled. The T_rr_current_interval is recalculated each time a regular RTCP packet is sent. The benefit of suppression is that it avoids wasting bandwidth when there is nothing requiring frequent RTCP transmissions, but still allows utilization of the configured bandwidth when feedback is needed. Unfortunately, this suppression mechanism skews the distribution of the RTCP sending intervals compared to the regular RTCP reporting intervals. The standard RTCP timing rules, including reconsideration and the compensation factor, result in the intervals between sending RTCP packets having a distribution that is skewed towards the upper end of the range [0.5/1.21828, 1.5/1.21828]*Td, where Td is the deterministic calculated RTCP reporting interval. With Td = 5 s, this distribution covers the range [2.052 s, 6.156 s]. In comparison, the RTP/AVPF suppression rules act in an interval that is 0.5 to 1.5 times T_rr_interval; for T_rr_interval = 5s, this is [2.5 s, 7.5 s].
The effect of this is that the time between consecutive RTCP packets when using T_rr_interval suppression can become large. The maximum time interval between sending one regular RTCP packet and the next, when T_rr_interval is being used, occurs when T_rr_current_interval takes its maximum value and a regular RTCP packet is suppressed at the end of the suppression period, then the next regular RTCP packet is scheduled after its largest possible reporting interval. Taking the worst case of the two intervals gives a maximum time between two RTCP reports of 1.5*T_rr_interval + 1.5/1.21828*Td. This behavior can be surprising when Td and T_rr_interval have the same value. That is, when T_rr_interval is configured to match the regular RTCP reporting interval. In this case, one might expect that regular RTCP packets are sent according to their usual schedule, but feedback packets can be sent early. However, the above-mentioned issue results in the RTCP packets actually being sent in the range [0.5*Td, 2.731*Td] with a highly non-uniform distribution, rather than the range [0.41*Td, 1.23*Td]. This is perhaps unexpected, but is not a problem in itself. However, when coupled with packet loss, it raises the issue of premature timeout. 7.1.2. Avoiding Premature Timeout In RTP/AVP [RFC3550] the timeout behavior is simple; it is 5 times Td, where Td is calculated with a Tmin value of 5 seconds. In other words, if the configured RTCP bandwidth allows for an average RTCP reporting interval shorter than 5 seconds, the timeout is 25 seconds of no activity from the SSRC (RTP or RTCP); otherwise, the timeout is 5 average reporting intervals. RTP/AVPF [RFC4585] introduces different timeout behaviors depending on the value of T_rr_interval. When T_rr_interval is 0, it uses the same timeout calculation as RTP/AVP. However, when T_rr_interval is non-zero, it replaces Tmin in the timeout calculation, most likely to speed up detection of timed out SSRCs. However, using a non-zero T_rr_interval has two consequences for RTP behavior. First, due to suppression, the number of RTP and RTCP packets sent by an SSRC that is not an active RTP sender can become very low, because of the issue discussed in Section 7.1.1. As the RTCP packet interval can be as long as 2.73*Td, during a 5*Td time period, an endpoint might in fact transmit only a single RTCP packet. The long intervals result in fewer RTCP packets, to a point where a single RTCP packet loss can sometimes result in timing out an SSRC. Second, the RTP/AVPF changes to the timeout rules reduce robustness to misconfiguration. It is common to use RTP/AVPF configured such that RTCP packets can be sent frequently to allow rapid feedback;
however, this makes timeouts very sensitive to T_rr_interval. For example, if two SSRCs are configured, one with T_rr_interval = 0.1 s and the other with T_rr_interval = 0.6 s, then this small difference will result in the SSRC with the shorter T_rr_interval timing out the other if it stops sending RTP packets, since the other RTCP reporting interval is more than five times its own. When RTP/AVP is used, or RTP/AVPF with T_rr_interval = 0, this is a non-issue, as the timeout period will be 25 s, and differences between configured RTCP bandwidth can only cause premature timeouts when the reporting intervals are greater than 5 s and differ by a factor of five. To limit the scope for such problematic misconfiguration, we define an update to the RTP/AVPF timeout rules in Section 7.1.4. 7.1.3. Interoperability between RTP/AVP and RTP/AVPF If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their secure variants) are combined within a single RTP session, and the RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly below 5 seconds, there is a risk that the RTP/AVPF endpoints will prematurely time out the SSRCs of the RTP/AVP endpoints, due to their different RTCP timeout rules. Conversely, if the RTP/AVPF endpoints use a T_rr_interval that is significantly larger than 5 seconds, there is a risk that the RTP/AVP endpoints will time out the SSRCs of the RTP/AVPF endpoints. Mixing endpoints using two different RTP profiles within a single RTP session is NOT RECOMMENDED. However, if mixed RTP profiles are used, and the RTP/AVPF endpoints are not updated to follow Section 7.1.4 of this memo, then the RTP/AVPF session SHOULD be configured to use T_rr_interval = 4 seconds to avoid premature timeouts. The choice of T_rr_interval = 4 seconds for interoperability might appear strange. Intuitively, this value ought to be 5 seconds, to make both the RTP/AVP and RTP/AVPF use the same timeout period. However, the behavior outlined in Section 7.1.1 shows that actual RTP/AVPF reporting intervals can be longer than expected. Setting T_rr_interval = 4 seconds gives actual RTCP intervals near to those expected by RTP/AVP, ensuring interoperability.
7.1.4. Updated SSRC Timeout Rules To ensure interoperability and avoid premature timeouts, all SSRCs in an RTP session MUST use the same timeout behavior. However, previous specifications are inconsistent in this regard. To avoid interoperability issues, this memo updates the timeout rules as follows: o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, the timeout interval SHALL be calculated using a multiplier of five times the deterministic RTCP reporting interval. That is, the timeout interval SHALL be 5*Td. o For the RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF profiles, calculation of Td, for the purpose of calculating the participant timeout only, SHALL be done using a Tmin value of 5 seconds and not the reduced minimal interval, even if the reduced minimum interval is used to calculate RTCP packet transmission intervals. This changes the behavior for the RTP/AVPF or RTP/SAVPF profiles when T_rr_interval != 0. Specifically, the first paragraph of Section 3.5.4 of [RFC4585] is updated to use Tmin instead of T_rr_interval in the timeout calculation for RTP/AVPF entities. 7.2. Tuning RTCP Transmissions This subsection discusses what tuning can be done to reduce the downsides of the shared RTCP packet intervals. First, what possibilities exist for the RTP/AVP [RFC3551] profile are listed followed by what additional tools are provided by RTP/AVPF [RFC4585]. 7.2.1. RTP/AVP and RTP/SAVP When using the RTP/AVP or RTP/SAVP profiles, the options for tuning the RTCP reporting intervals are limited to the RTCP sender and receiver bandwidth, and whether the minimum RTCP interval is scaled according to the bandwidth. As the scheduling algorithm includes both randomization and reconsideration, one cannot simply calculate the expected average transmission interval using the formula for Td given in Section 6.3.1 of [RFC3550]. However, by considering the inputs to that expression, and the randomization and reconsideration rules, we can begin to understand the behavior of the RTCP transmission interval.
Let's start with some basic observations: a. Unless the scaled minimum RTCP interval is used, Td prior to randomization and reconsideration can never be less than Tmin. The default value of Tmin is 5 seconds. b. If the scaled minimum RTCP interval is used, Td can become as low as 360 divided by RTP Session bandwidth in kilobits per second. In SDP, the RTP session bandwidth is signaled using a "b=AS" line. An RTP Session bandwidth of 72 kbps results in Tmin being 5 seconds. An RTP session bandwidth of 360 kbps of course gives a Tmin of 1 second, and to achieve a Tmin equal to once every frame for a 25 frame-per-second video stream requires an RTP session bandwidth of 9 Mbps. Use of the RTP/AVPF or RTP/SAVPF profile allows more frequent RTCP reports for the same bandwidth, as discussed below. c. The value of Td scales with the number of SSRCs and the average size of the RTCP reports to keep the overall RTCP bandwidth constant. d. The actual transmission interval for a Td value is in the range [0.5*Td/1.21828, 1.5*Td/1.21828], and the distribution is skewed, due to reconsideration, with the majority of the probability mass being above Td. This means, for example, that for Td = 5 s, the actual transmission interval will be distributed in the range [2.052 s, 6.156 s], and tending towards the upper half of the interval. Note that Tmin parameter limits the value of Td before randomization and reconsideration are applied, so the actual transmission interval will cover a range extending below Tmin. Given the above, we can calculate the number of SSRCs, n, that an RTP session with 5% of the session bandwidth assigned to RTCP can support while maintaining Td equal to Tmin. This will tell us how many RTP streams we can report on, keeping the RTCP overhead within acceptable bounds. We make two assumptions that simplify the calculation: that all SSRCs are senders, and that they all send compound RTCP packets comprising an SR packet with n-1 report blocks, followed by an SDES packet containing a 16 octet CNAME value [RFC7022] (such RTCP packets will vary in size between 54 and 798 octets depending on n, up to the maximum of 31 report blocks that can be included in an SR packet). If we put this packet size, and a 5% RTCP bandwidth fraction into the RTCP interval calculation in Section 6.3.1 of [RFC3550], and calculate the value of n needed to give Td = Tmin for the scaled minimum interval, we find n=9 SSRCs can be supported (irrespective of the interval, due to the way the reporting interval scales with the session bandwidth). We see that to support more SSRCs without changing the scaled minimum interval, we need to increase the RTCP
bandwidth fraction from 5%; changing the session bandwidth to a higher value would reduce the Tmin. However, if using the default 5% allocation of RTCP bandwidth, an increase will result in more SSRCs being supported given a fixed Td target. Based on the above, when using the RTP/AVP profile or the RTP/SAVP profile, the key limitation for rapid RTCP reporting in small unicast sessions is going to be the Tmin value. The RTP session bandwidth configured in RTCP has to be sufficiently high to reach the reporting goals the application has following the rules for the scaled minimal RTCP interval. 7.2.2. RTP/AVPF and RTP/SAVPF When using RTP/AVPF or RTP/SAVPF, we have a powerful additional tool for tuning RTCP transmissions: the T_rr_interval parameter. Use of this parameter allows short RTCP reporting intervals; alternatively it gives the ability to sent frequent RTCP feedback without sending frequent regular RTCP reports. The use of the RTP/AVPF or RTP/SAVPF profile with T_rr_interval set to a value greater than zero but smaller than Tmin allows more frequent RTCP feedback than the RTP/AVP or RTP/SAVP profiles, for a given RTCP bandwidth. This happens because Tmin is set to zero after the transmission of the initial RTCP report, causing the reporting interval for later packet to be determined by the usual RTCP bandwidth-based calculation, with Tmin=0, and the T_rr_interval. This has the effect that we are no longer restricted by the minimal interval (whether the default 5-second minimum or the reduced minimum interval). Rather, the RTCP bandwidth and the T_rr_interval are the governing factors, allowing faster feedback. Applications that care about rapid regular RTCP feedback ought to consider using the RTP/ AVPF or RTP/SAVPF profile, even if they don't use the feedback features of that profile. The use of the RTP/AVPF or RTP/SAVPF profile allows RTCP feedback packets to be sent frequently, without also requiring regular RTCP reports to be sent frequently, since T_rr_interval limits the rate at which regular RTCP packets can be sent, while still permitting RTCP feedback packets to be sent. Applications that can use feedback packets for some RTP streams, e.g., video streams, but don't want frequent regular reporting for other RTP streams, can configure the T_rr_interval to a value so that the regular reporting for both audio and video is at a level that is considered acceptable for the audio. They could then use feedback packets, which will include RTCP SR/RR packets unless reduced size RTCP feedback packets [RFC5506] are used,
for the video reporting. This allows the available RTCP bandwidth to be devoted on the feedback that provides the most utility for the application. Using T_rr_interval still requires one to determine suitable values for the RTCP bandwidth value. Indeed, it might make this choice even more important, as this is more likely to affect the RTCP behavior and performance than when using the RTP/AVP or RTP/SAVP profile, as there are fewer limitations affecting the RTCP transmission. When T_rr_interval is non-zero, there are configurations that need to be avoided. If the RTCP bandwidth chosen is such that the Td value is smaller than, but close to, T_rr_interval, then the actual regular RTCP packet transmission interval can become very large, as discussed in Section 7.1.1. Therefore, for configuration where one intends to have Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted at values less than 1/4th of T_rr_interval, which results in the range becoming [0.5*T_rr_interval, 1.81*T_rr_interval]. With the RTP/AVPF or RTP/SAVPF profiles, using T_rr_interval = 0 has utility and results in a behavior where the RTCP transmission is only limited by the bandwidth, i.e., no Tmin limitations at all. This allows more frequent regular RTCP reporting than can be achieved using the RTP/AVP profile. Many configurations of RTCP will not consume all the bandwidth that they have been configured to use, but this configuration will consume what it has been given. Note that the same behavior will be achieved as long as T_rr_interval is smaller than 1/3 of Td as that prevents T_rr_interval from affecting the transmission. There exists no method for using different regular RTCP reporting intervals depending on the media type or individual RTP stream, other than using a separate RTP session for each type or stream. 8. Security Considerations When using the secure RTP protocol (RTP/SAVP) [RFC3711], or the secure variant of the feedback profile (RTP/SAVPF) [RFC5124], the cryptographic context of a compound secure RTCP packet is the SSRC of the sender of the first RTCP (sub-)packet. This could matter in some cases, especially for keying mechanisms such as MIKEY [RFC3830] that allow use of per-SSRC keying. Otherwise, the standard security considerations of RTP apply; sending multiple RTP streams from a single endpoint in a single RTP session does not appear to have different security consequences than sending the same number of RTP streams spread across different RTP sessions.
9. References 9.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <http://www.rfc-editor.org/info/rfc2119>. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <http://www.rfc-editor.org/info/rfc3550>. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004, <http://www.rfc-editor.org/info/rfc3711>. [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, DOI 10.17487/RFC4585, July 2006, <http://www.rfc-editor.org/info/rfc4585>. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <http://www.rfc-editor.org/info/rfc5124>. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, DOI 10.17487/RFC5506, April 2009, <http://www.rfc-editor.org/info/rfc5506>. 9.2. Informative References [CLUE-FRAME] Duckworth, M., Ed., Pepperell, A., and S. Wenger, "Framework for Telepresence Multi-Streams", Work in Progress, draft-ietf-clue-framework-25, January 2016. [MULTI-RTP] Westerlund, M., Perkins, C., and J. Lennox, "Sending Multiple Types of Media in a Single RTP Session", Work in Progress, draft-ietf-avtcore-multi-media-rtp-session-13, December 2015.
[MULTI-STREAM-OPT] Lennox, J., Westerlund, M., Wu, Q., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session: Grouping RTCP Reception Statistics and Other Feedback", Work in Progress, draft-ietf-avtcore-rtp-multi- stream-optimisation-12, March 2016. [RFC3390] Allman, M., Floyd, S., and C. Partridge, "Increasing TCP's Initial Window", RFC 3390, DOI 10.17487/RFC3390, October 2002, <http://www.rfc-editor.org/info/rfc3390>. [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, <http://www.rfc-editor.org/info/rfc3551>. [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556, DOI 10.17487/RFC3556, July 2003, <http://www.rfc-editor.org/info/rfc3556>. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, DOI 10.17487/RFC3830, August 2004, <http://www.rfc-editor.org/info/rfc3830>. [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, DOI 10.17487/RFC4588, July 2006, <http://www.rfc-editor.org/info/rfc4588>. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, February 2008, <http://www.rfc-editor.org/info/rfc5104>. [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, <http://www.rfc-editor.org/info/rfc5576>. [RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, DOI 10.17487/RFC6190, May 2011, <http://www.rfc-editor.org/info/rfc6190>.
[RFC6928] Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis, "Increasing TCP's Initial Window", RFC 6928, DOI 10.17487/RFC6928, April 2013, <http://www.rfc-editor.org/info/rfc6928>. [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, September 2013, <http://www.rfc-editor.org/info/rfc7022>. [RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple Clock Rates in an RTP Session", RFC 7160, DOI 10.17487/RFC7160, April 2014, <http://www.rfc-editor.org/info/rfc7160>. [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, DOI 10.17487/RFC7667, November 2015, <http://www.rfc-editor.org/info/rfc7667>. [SDP-BUNDLE] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", Work in Progress, draft-ietf-mmusic-sdp-bundle-negotiation-36, October 2016. [Sim88] Westerlund, M., "SIMULATION RESULTS FOR MULTI-STREAM", IETF 88 Proceedings, November 2013, <https://www.ietf.org/proceedings/88/slides/ slides-88-avtcore-0.pdf>. [Sim92] Westerlund, M., Lennox, J., Perkins, C., and Q. Wu, "Changes in RTP Multi-stream", IETF 92 Proceedings, March 2015, <https://www.ietf.org/proceedings/92/slides/ slides-92-avtcore-0.pdf>.
Acknowledgments The authors like to thank Harald Alvestrand and everyone else who has been involved in the development of this document. Authors' Addresses Jonathan Lennox Vidyo, Inc. 433 Hackensack Avenue Seventh Floor Hackensack, NJ 07601 United States of America Email: firstname.lastname@example.org Magnus Westerlund Ericsson Farogatan 2 SE-164 80 Kista Sweden Phone: +46 10 714 82 87 Email: email@example.com Qin Wu Huawei 101 Software Avenue, Yuhua District Nanjing, Jiangsu 210012 China Email: firstname.lastname@example.org Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: email@example.com