Sending Multiple Media Streams in a Single RTP Session
Author(s): Jonathan Lennox, Magnus Westerlund, Colin Perkins, Wenson Wu
This document expands and clarifies the behavior of the Real-Time Transport Protocol (RTP) endpoints when they are sending multiple media streams in a single RTP session. In particular, issues involving Real-Time Transport Control Protocol (RTCP) messages are described....
AVTCORE J. Lennox Internet-Draft Vidyo Updates: 3550 (if approved) M. Westerlund Intended status: Standards Track Ericsson Expires: January 12, 2014 Q. Wu Huawei C. Perkins University of Glasgow July 11, 2013 Sending Multiple Media Streams in a Single RTP Session draft-ietf-avtcore-rtp-multi-stream-01 Abstract This document expands and clarifies the behavior of the Real-Time Transport Protocol (RTP) endpoints when they are sending multiple media streams in a single RTP session. In particular, issues involving Real-Time Transport Control Protocol (RTCP) messages are described. This document updates RFC 3550 in regards to handling of multiple SSRCs per endpoint in RTP sessions. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on January 12, 2014. Copyright Notice Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved. Lennox, et al. Expires January 12, 2014 [Page 1] Internet-Draft Multiple Media Streams in an RTP Session July 2013 This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Use Cases For Multi-Stream Endpoints . . . . . . . . . . . . 3 3.1. Multiple-Capturer Endpoints . . . . . . . . . . . . . . . 3 3.2. Multi-Media Sessions . . . . . . . . . . . . . . . . . . 3 3.3. Multi-Stream Mixers . . . . . . . . . . . . . . . . . . . 4 4. Multi-Stream Endpoint RTP Media Recommendations . . . . . . . 4 5. Multi-Stream Endpoint RTCP Recommendations . . . . . . . . . 4 5.1. RTCP Reporting Requirement . . . . . . . . . . . . . . . 5 5.2. Initial Reporting Interval . . . . . . . . . . . . . . . 5 5.3. Compound RTCP Packets . . . . . . . . . . . . . . . . . . 5 6. RTCP Considerations for Streams with Disparate Rates . . . . 7 6.1. Timing out SSRCs . . . . . . . . . . . . . . . . . . . . 8 6.2. Tuning RTCP transmissions . . . . . . . . . . . . . . . . 9 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 8. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 12 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 10.1. Normative References . . . . . . . . . . . . . . . . . . 12 10.2. Informative References . . . . . . . . . . . . . . . . . 13 Appendix A. Changes From Earlier Versions . . . . . . . . . . . 14 A.1. Changes From WG Draft -00 . . . . . . . . . . . . . . . . 14 A.2. Changes From Individual Draft -02 . . . . . . . . . . . . 14 A.3. Changes From Individual Draft -01 . . . . . . . . . . . . 14 A.4. Changes From Individual Draft -00 . . . . . . . . . . . . 14 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15 1. Introduction At the time The Real-Time Transport Protocol (RTP) [RFC 3550] was originally written, and for quite some time after, endpoints in RTP sessions typically only transmitted a single media stream per RTP session, where separate RTP sessions were typically used for each distinct media type. Lennox, et al. Expires January 12, 2014 [Page 2] Internet-Draft Multiple Media Streams in an RTP Session July 2013 Recently, however, a number of scenarios have emerged (discussed further in Section 3) in which endpoints wish to send multiple RTP media streams, distinguished by distinct RTP synchronization source (SSRC) identifiers, in a single RTP session. Although RTP's initial design did consider such scenarios, the specification was not consistently written with such use cases in mind. The specifications are thus somewhat unclear. The purpose of this document is to expand and clarify [RFC 3550]'s language for these use cases. The authors believe this does not result in any major normative changes to the RTP specification, however this document defines how the RTP specification is to be interpreted. In these cases, this document updates RFC 3550. The document starts with terminology and some use cases where multiple sources will occur. This is followed by some case studies to try to identify issues that exist and need considerations. This is followed by RTP and RTCP recommendations to resolve issues. Next are security considerations and remaining open issues. 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 send multiple streams in a single RTP session. 3.1. Multiple-Capturer Endpoints The most straightforward motivation for an endpoint to send multiple media streams in a session is the scenario where an endpoint has multiple capture devices of the same media type and characteristics. For example, telepresence endpoints, of the type described by the CLUE Telepresence Framework [I-D.ietf-clue-framework] is designed, often have multiple cameras or microphones covering various areas of a room. 3.2. Multi-Media Sessions Recent work has been done in RTP [I-D.ietf-avtcore-multi-media-rtp-session] and SDP Lennox, et al. Expires January 12, 2014 [Page 3] Internet-Draft Multiple Media Streams in an RTP Session July 2013 [I-D.ietf-mmusic-sdp-bundle-negotiation] to update RTP's historical assumption that media streams of different media types would always be sent on different RTP sessions. In this work, a single endpoint's audio and video media streams (for example) are instead sent in a single RTP session. 3.3. Multi-Stream Mixers There are several RTP topologies which can involve a central device that itself generates multiple media streams in a session. One 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. More complicated is the Source Projecting Mixer, see Section 3.6 of [I-D.ietf-avtcore-rtp-topologies-update]. This is a central box that receives media streams from several endpoints, and then selectively forwards modified versions of some of the streams toward the other endpoints it is connected to. Toward one destination, a separate media source appears in the session for every other source connected to the mixer, "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, in that it terminates RTCP reporting about the mixed streams, and it can re-write SSRCs, timestamps, and sequence numbers, as well as the contents of the RTP payloads, and can turn sources on and off at will without appearing to be generating packet loss. Each projected stream will typically preserve its original RTCP source description (SDES) information. 4. Multi-Stream Endpoint RTP Media Recommendations While an endpoint MUST (of course) stay within its share of the available session bandwidth, as determined by signalling and congestion control, this need not be applied independently or uniformly to each media stream. In particular, session bandwidth MAY be reallocated among an endpoint's media streams, for example by varying the bandwidth use of a variable-rate codec, or changing the codec used by the media stream, up to the constraints of the session's negotiated (or declared) codecs. This includes enabling or disabling media streams as more or less bandwidth becomes available. 5. Multi-Stream Endpoint RTCP Recommendations Lennox, et al. Expires January 12, 2014 [Page 4] Internet-Draft Multiple Media Streams in an RTP Session July 2013 This section contains a number of different RTCP clarifications or recommendations that enables more efficient and simpler behavior without loss of functionality. The RTP Control Protocol (RTCP) is defined in Section 6 of [RFC 3550], but it is largely documented in terms of "participants". In many cases, the specification's recommendations for "participants" are to be interpreted as applying to individual media streams, rather than to endpoints. This section describes several concrete cases where this applies. (tbd: rather than think in terms of media streams, it might be clearer to refer to SSRC values, where a participant with multiple active SSRC values counts as multiple participants, once per SSRC) 5.1. RTCP Reporting Requirement For each of an endpoint's media streams, whether or not it is currently sending media, SR/RR and SDES packets MUST be sent at least once per RTCP report interval. (For discussion of the content of SR or RR packets' reception statistic reports, see [I-D.ietf-avtcore-rtp-multi-stream-optimisation].) 5.2. Initial Reporting Interval When a new media stream is added to a unicast session, the sentence in [RFC 3550]'s Section 6.2 applies: "For unicast sessions ... the delay before sending the initial compound RTCP packet MAY be zero." This applies to individual media sources as well. Thus, endpoints MAY send an initial RTCP packet for an SSRC immediately upon adding it to a unicast session. This allowance also applies, as written, when initially joining a unicast session. However, in this case some caution needs to be exercised if the end-point or mixer has a large number of sources (SSRCs) as this can create a significant burst. How big an issue this depends on the number of source to send initial SR or RR and Session Description CNAME items for in relation to the RTCP bandwidth. (tbd: Maybe some recommendation here? The aim in restricting this to unicast sessions was to avoid this burst of traffic, which the usual RTCP timing and reconsideration rules will prevent) 5.3. Compound RTCP Packets Section 6.1 gives the following advice to RTP translators and mixers: Lennox, et al. Expires January 12, 2014 [Page 5] Internet-Draft Multiple Media Streams in an RTP Session July 2013 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. Note: To avoid confusion, an RTCP packet is an individual item, such as a Sender Report (SR), Receiver Report (RR), Source Description (SDES), Goodbye (BYE), Application Defined (APP), Feedback [RFC4585] or Extended Report (XR) [RFC3611] packet. A compound packet is the combination of two or more such RTCP packets where the first packet has to be an SR or an RR packet, and which contains a SDES packet containing an CNAME item. Thus the above results in compound RTCP packets that contain multiple SR or RR packets from different sources as well as any of the other packet types. There are no restrictions on the order in which the packets can occur within the compound packet, except the regular compound rule, i.e., starting with an SR or RR. This advice applies to multi-media-stream endpoints as well, with the same restrictions and considerations. (Note, however, that the last sentence does not apply to AVPF [RFC4585] or SAVPF [RFC5124] feedback packets if Reduced-Size RTCP [RFC5506] is in use.) Due to RTCP's randomization of reporting times, there is a fair bit of tolerance in precisely when an endpoint schedules RTCP to be sent. Thus, one potential way of implementing this recommendation would be to randomize all of an endpoint's sources together, with a single randomization schedule, so an MTU's worth of RTCP all comes out simultaneously. (tbd: Multiplexing RTCP packets from multiple different sources might require some adjustment to the calculation of RTCP's avg_rtcp_size, as the RTCP group interval is proportional to avg_rtcp_size times the group size). Lennox, et al. Expires January 12, 2014 [Page 6] Internet-Draft Multiple Media Streams in an RTP Session July 2013 6. RTCP Considerations for Streams with Disparate Rates It is possible for a single RTP session to carry streams of greatly differing bandwidth. There are two scenarios where this can occur. The first is when a single RTP session carries multiple flows of the same media type, but with very different quality; for example a video switching multi-point conference unit might send a full rate high- definition video stream of the active speaker but only thumbnails for the other participants, all sent in a single RTP session. The second scenarios occurs when audio and video flows are sent in a single RTP session, as discussed in [I-D.ietf-avtcore-multi-media-rtp-session]. An RTP session has a single set of parameters that configure the session bandwidth, the RTCP sender and receiver fractions (e.g., via the SDP "b=RR:" and "b=RS:" lines), 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 RTCP reporting interval will be the same for every SSRC in an RTP session. This uniform RTCP reporting interval can result in RTCP reports being sent more often than is considered desirable for a particular media type. For example, if an audio flow is multiplexed with a high quality video flow where the session bandwidth is configured to match the video bandwidth, this can result in the RTCP packets having a greater bandwidth allocation than the audio data rate. If the reduced minimum RTCP interval described in Section 6.2 of [RFC 3550] is used in the session, which might be appropriate for video where rapid feedback is wanted, the audio sources could be expected to send RTCP packets more often than they send audio data packets. This is most likely undesirable, and while the mismatch can be reduced through careful tuning of the RTCP parameters, particularly trr_int in RTP/AVPF sessions, it is inherent in the design of the RTCP timing rules, and affects all RTP sessions containing flows with mismatched bandwidth. Having multiple media types in one RTP session also results in more SSRCs being present in this RTP session. This increasing the amount of cross reporting between the SSRCs. From an RTCP perspective, two RTP sessions with half the number of SSRCs in each will be slightly more efficient. If someone needs either the higher efficiency due to the lesser number of SSRCs or the fact that one can't tailor RTCP usage per media type, they need to use independent RTP sessions. Lennox, et al. Expires January 12, 2014 [Page 7] Internet-Draft Multiple Media Streams in an RTP Session July 2013 When it comes to configuring RTCP the need for regular periodic reporting needs to be weighted against any feedback or control messages being sent. Applications using RTP/AVPF or RTP/SAVPF are RECOMMENDED to consider setting the trr-int parameter to a value suitable for the application's needs, thus potentially reducing the need for regular reporting and thus releasing more bandwidth for use for feedback or control. Another aspect of an RTP session with multiple media types is that the RTCP packets, RTCP Feedback Messages, or RTCP XR metrics used might not be applicable to all media types. Instead, all RTP/RTCP endpoints need to correlate the media type of the SSRC being referenced in a message or packet and only use those that apply to that particular SSRC and its media type. Signalling 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. 6.1. Timing out SSRCs All SSRCs used in an RTP session MUST use the same timeout behaviour to avoid premature timeouts. This will depend on the RTP profile and its configuration. The RTP specification provides several options that can influence the values used when calculating the time interval. To avoid interoperability issues when using this specification, this document makes several clarifications to the calculations. For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval = 0, the timeout interval SHALL be calculated using a multiplier of 5, i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be done using a Tmin value of 5 seconds, not the reduced minimal interval even if used to calculate RTCP packet transmission intervals. If using either the RTP/AVPF or RTP/SAVPF profiles with T_rr_interval != 0 then the calculation as specified in Section 3.5.4 of RFC4585 SHALL be used with a multiplier of 5, i.e. Tmin in the Td calculation is the T_rr_interval. Note: If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their secure variants) are combined in a single RTP session, and the RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly lower than 5 seconds, then there is a risk that the RTP/AVP endpoints will prematurely timeout the RTP/AVPF endpoints due to their different RTCP timeout intervals. Since an RTP session can only use a single RTP profile, this issue ought never occur. If such mixed RTP profiles are used, however, the RTP/AVPF session MUST NOT use a non-zero T_rr_interval that is smaller than 5 seconds. Lennox, et al. Expires January 12, 2014 [Page 8] Internet-Draft Multiple Media Streams in an RTP Session July 2013 (tbd: it has been suggested that a minimum non-zero T_rr_interval of 4 seconds is more appropriate, due to the nature of the timing rules). 6.2. Tuning RTCP transmissions This sub-section discusses what tuning can be done to reduce the downsides of the shared RTCP packet intervals. When using the RTP/AVP or RTP/SAVP profiles the tuning one can do is very limited. The controls one has are limited to the RTCP bandwidth values and whether the minimum RTCP interval is scaled according to the bandwidth. As the scheduling algorithm includes both random factors and reconsideration, one can't simply calculate the expected average transmission interval using the formula for Td. But it does indicate the important factors affecting the transmission interval, namely the RTCP bandwidth available for the role (Active Sender or Participant), the average RTCP packet size, and the number of SSRCs classified in the relevant role. Note that if the ratio of senders to total number of session participants is larger than the ratio of RTCP bandwidth for senders in relation to the total RTCP bandwidth, then senders and receivers are treated together. Let's start with some basic observations: a. Unless the scaled minimum RTCP interval is used, then Td prior to randomization and reconsideration can never be less than 5 seconds (assuming default Tmin of 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. In SDP the RTP session bandwidth is signalled using b=AS. 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 Hz video stream requires an RTP session bandwidth of 9 Mbps! (The use of the RTP/AVPF or RTP/SAVPF profile allows a smaller Tmin, and hence more frequent RTCP reports, as discussed below). c. Let's calculate the number (n) of SSRCs in the RTP session that 5% of the session bandwidth can support to yield a Td value equal to Tmin with minimal scaling. For this calculation we have to make two assumptions. The first is that we will consider most or all SSRC being senders, resulting in everyone sharing the available bandwidth. Secondly we will select an average RTCP packet size. This packet will consist of an SR, containing (n-1) report blocks up to 31 report blocks, and an SDES item with at least a CNAME (17 bytes in size) in it. Such a basic packet will Lennox, et al. Expires January 12, 2014 [Page 9] Internet-Draft Multiple Media Streams in an RTP Session July 2013 be 800 bytes for n>=32. With these parameters, and as the bandwidth goes up the time interval is proportionally decreased (due to minimal scaling), thus all the example bandwidths 72 kbps, 360 kbps and 9 Mbps all support 9 SSRCs. d. The actual transmission interval for a Td value is [0.5*Td/ 1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the interval is actually [2.052,6.156] and the distribution is not uniform, but rather exponentially-increasing. The probability for sending at time X, given it is within the interval, is probability of picking X in the interval times the probability to randomly picking a number that is <=X within the interval with an uniform probability distribution. This results in that the majority of the probability mass is above the Td value. To conclude, with RTP/AVP and RTP/SAVP the key limitation for small unicast sessions is going to be the Tmin value. Thus 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. When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional tool, the setting of the T_rr_interval which has several effects on the RTCP reporting. First of all as Tmin is set to 0 after the initial transmission, the regular reporting interval is instead determined by the regular bandwidth based calculation and the T_rr_interval. This has the effect that we are no longer restricted by the minimal interval or even the scaling rule for the minimal rule. Instead the RTCP bandwidth and the T_rr_interval are the governing factors. Now it also becomes important to separate between the application's need for regular reports and RTCP feedback packet types. In both regular RTCP mode, as in Early RTCP Mode, the usage of the T_rr_interval prevents regular RTCP packets, i.e. packets without any Feedback packets, to be sent more often than T_rr_interval. This value is a hard as no regular RTCP packet can be sent less than T_rr_interval after the previous regular packet packet. Lennox, et al. Expires January 12, 2014 [Page 10] Internet-Draft Multiple Media Streams in an RTP Session July 2013 So applications that have a use for feedback packets for some media streams, for example video streams, but don't want frequent regular reporting for audio, could 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, and can include other report information in addition to the feedback packet that needs to be sent. That way the available RTCP bandwidth can be focused for the use which provides the most utility for the application. Using T_rr_interval still requires one to determine suitable values for the RTCP bandwidth value, in fact it might make it even more important, as this is more likely to affect the RTCP behaviour and performance than when using RTP/AVP, as there are fewer limitations affecting the RTCP transmission. When using T_rr_interval, i.e. having it be non zero, there are configurations that have to be avoided. If the resulting Td value is smaller but close to T_rr_interval then the interval in which the actual regular RTCP packet transmission falls into becomes very large, from 0.5 times T_rr_interval up to 2.73 times the T_rr_interval. 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 that the range becomes [0.5*T_rr_interval, 1.81*T_rr_interval]. With RTP/AVPF, using a T_rr_interval of 0 or with another low value significantly lower than Td still has utility, and different behaviour compared to RTP/AVP. This avoids the Tmin limitations of RTP/AVP, thus allowing more frequent regular RTCP reporting. In fact this will result that the RTCP traffic becomes as high as the configured values. (tbd: a future version of this memo will include examples of how to choose RTCP parameters for common scenarios) There exists no method within the specification for using different regular RTCP reporting intervals depending on the media type or individual media stream. 7. Security Considerations Lennox, et al. Expires January 12, 2014 [Page 11] Internet-Draft Multiple Media Streams in an RTP Session July 2013 In the secure RTP protocol (SRTP) [RFC3711], the cryptographic context of a compound SRTCP 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] which use per-SSRC keying. Other than that, the standard security considerations of RTP apply; sending multiple media streams from a single endpoint does not appear to have different security consequences than sending the same number of streams. 8. Open Issues At this stage this document contains a number of open issues. The below list tries to summarize the issues: 1. Further clarifications on how to handle the RTCP scheduler when sending multiple sources in one compound packet. 2. How is the RTCP avg_rtcp_size be calculated when RTCP packets are routinely multiplexed among multiple RTCP senders? 3. Do we need to provide a recommendation for unicast session joiners with many sources to not use 0 initial minimal interval from bit-rate burst perspective? 9. IANA Considerations No IANA actions needed. 10. References 10.1. Normative References [I-D.ietf-avtcore-6222bis] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", draft-ietf-avtcore-6222bis-04 (work in progress), June 2013. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC2119, March 1997. [RFC 3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. Lennox, et al. Expires January 12, 2014 [Page 12] Internet-Draft Multiple Media Streams in an RTP Session July 2013 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC3711, March 2004. [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)", RFC4585, July 2006. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC5124, February 2008. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC5506, April 2009. 10.2. Informative References [I-D.ietf-avtcore-multi-media-rtp-session] Westerlund, M., Perkins, C., and J. Lennox, "Multiple Media Types in an RTP Session", draft-ietf-avtcore-multi- media-rtp-session-02 (work in progress), February 2013. [I-D.ietf-avtcore-rtp-multi-stream-optimisation] 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 ", draft-ietf-avtcore-rtp-multi-stream-optimisation-00 (work in progress), July 2013. [I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", draft- ietf-avtcore-rtp-topologies-update-00 (work in progress), April 2013. [I-D.ietf-clue-framework] Duckworth, M., Pepperell, A., and S. Wenger, "Framework for Telepresence Multi-Streams", draft-ietf-clue- framework-10 (work in progress), May 2013. [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Multiplexing Negotiation Using Session Description Protocol (SDP) Port Numbers", draft-ietf-mmusic-sdp- bundle-negotiation-04 (work in progress), June 2013. Lennox, et al. Expires January 12, 2014 [Page 13] Internet-Draft Multiple Media Streams in an RTP Session July 2013 [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC3611, November 2003. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC3830, August 2004. Appendix A. Changes From Earlier Versions Note to the RFC-Editor: please remove this section prior to publication as an RFC. A.1. Changes From WG Draft -00 o Split the Reporting Group Extension from this draft into draft- ietf-avtcore-rtp-multi-stream-optimization-00. o Added RTCP tuning considerations from draft-ietf-avtcore-multi- media-rtp-session-02. A.2. Changes From Individual Draft -02 o Resubmitted as working group draft. o Updated references. A.3. Changes From Individual Draft -01 o Merged with draft-wu-avtcore-multisrc-endpoint-adver. o Changed how Reporting Groups are indicated in RTCP, to make it clear which source(s) is the group's reporting sources. o Clarified the rules for when sources can be placed in the same reporting group. o Clarified that mixers and translators need to pass reporting group SDES information if they are forwarding RR and SR traffic from members of a reporting group. A.4. Changes From Individual Draft -00 o Added the Reporting Group semantic to explicitly indicate which sources come from a single endpoint, rather than leaving it implicit. Lennox, et al. Expires January 12, 2014 [Page 14] Internet-Draft Multiple Media Streams in an RTP Session July 2013 o Specified that Reporting Group semantics (as they now are) apply to AVPF and XR, as well as to RR/SR report blocks. o Added a description of the cascaded source-projecting mixer, along with a calculation of its RTCP overhead if reporting groups are not in use. o Gave some guidance on how the flexibility of RTCP randomization allows some freedom in RTCP multiplexing. o Clarified the language of several of the recommendations. o Added an open issue discussing how avg_rtcp_size ought to be calculated for multiplexed RTCP. o Added an open issue discussing how RTCP bandwidths are to be chosen for sessions where source bandwidths greatly differ. Authors' Addresses Jonathan Lennox Vidyo, Inc. 433 Hackensack Avenue Seventh Floor Hackensack, NJ 07601 US Email: email@example.com Magnus Westerlund Ericsson Farogatan 6 SE-164 80 Kista Sweden Phone: +46 10 714 82 87 Email: firstname.lastname@example.org Qin Wu Huawei 101 Software Avenue, Yuhua District Nanjing, Jiangsu 210012 China Email: email@example.com Lennox, et al. Expires January 12, 2014 [Page 15] Internet-Draft Multiple Media Streams in an RTP Session July 2013 Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: firstname.lastname@example.org Lennox, et al. Expires January 12, 2014 [Page 16]