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Wenger 5 Intended status: Informational Vidyo 6 Expires: October 24, 2013 April 22, 2013 8 RTP Topologies 9 draft-ietf-avtcore-rtp-topologies-update-00 11 Abstract 13 This document discusses point to point and multi-endpoint topologies 14 used in Real-time Transport Protocol (RTP)-based environments. In 15 particular, centralized topologies commonly employed in the video 16 conferencing industry are mapped to the RTP terminology. 18 This document is updated with additional topologies and are intended 19 to replace RFC 5117. 21 Status of This Memo 23 This Internet-Draft is submitted in full conformance with the 24 provisions of BCP 78 and BCP 79. 26 Internet-Drafts are working documents of the Internet Engineering 27 Task Force (IETF). Note that other groups may also distribute 28 working documents as Internet-Drafts. The list of current Internet- 29 Drafts is at http://datatracker.ietf.org/drafts/current/. 31 Internet-Drafts are draft documents valid for a maximum of six months 32 and may be updated, replaced, or obsoleted by other documents at any 33 time. It is inappropriate to use Internet-Drafts as reference 34 material or to cite them other than as "work in progress." 36 This Internet-Draft will expire on October 24, 2013. 38 Copyright Notice 40 Copyright (c) 2013 IETF Trust and the persons identified as the 41 document authors. All rights reserved. 43 This document is subject to BCP 78 and the IETF Trust's Legal 44 Provisions Relating to IETF Documents 45 (http://trustee.ietf.org/license-info) in effect on the date of 46 publication of this document. Please review these documents 47 carefully, as they describe your rights and restrictions with respect 48 to this document. Code Components extracted from this document must 49 include Simplified BSD License text as described in Section 4.e of 50 the Trust Legal Provisions and are provided without warranty as 51 described in the Simplified BSD License. 53 Table of Contents 55 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 56 2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 57 2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 3 58 3. Topologies . . . . . . . . . . . . . . . . . . . . . . . . . 4 59 3.1. Point to Point . . . . . . . . . . . . . . . . . . . . . 4 60 3.2. Point to Point via Middlebox . . . . . . . . . . . . . . 5 61 3.2.1. Translators . . . . . . . . . . . . . . . . . . . . . 5 62 3.2.2. Back to Back RTP sessions . . . . . . . . . . . . . . 8 63 3.3. Point to Multipoint Using Multicast . . . . . . . . . . . 9 64 3.3.1. Any Source Multicast (ASM) . . . . . . . . . . . . . 9 65 3.3.2. Source Specific Multicast (SSM) . . . . . . . . . . . 11 66 3.3.3. SSM with Local Unicast Resources . . . . . . . . . . 12 67 3.4. Point to Multipoint Using Mesh . . . . . . . . . . . . . 14 68 3.5. Point to Multipoint Using the RFC 3550 Translator . . . . 15 69 3.5.1. Relay - Transport Translator . . . . . . . . . . . . 15 70 3.5.2. Media Translator . . . . . . . . . . . . . . . . . . 16 71 3.6. Point to Multipoint Using the RFC 3550 Mixer Model . . . 16 72 3.6.1. Media Mixing . . . . . . . . . . . . . . . . . . . . 18 73 3.6.2. Media Switching . . . . . . . . . . . . . . . . . . . 21 74 3.7. Source Projecting Middlebox . . . . . . . . . . . . . . . 23 75 3.8. Point to Multipoint Using Video Switching MCUs . . . . . 25 76 3.9. Point to Multipoint Using RTCP-Terminating MCU . . . . . 27 77 3.10. De-composite Endpoint . . . . . . . . . . . . . . . . . . 28 78 3.11. Non-Symmetric Mixer/Translators . . . . . . . . . . . . . 29 79 3.12. Combining Topologies . . . . . . . . . . . . . . . . . . 30 80 4. Comparing Topologies . . . . . . . . . . . . . . . . . . . . 30 81 4.1. Topology Properties . . . . . . . . . . . . . . . . . . . 31 82 4.1.1. All to All Media Transmission . . . . . . . . . . . . 31 83 4.1.2. Transport or Media Interoperability . . . . . . . . . 31 84 4.1.3. Per Domain Bit-Rate Adaptation . . . . . . . . . . . 31 85 4.1.4. Aggregation of Media . . . . . . . . . . . . . . . . 32 86 4.1.5. View of All Session Participants . . . . . . . . . . 32 87 4.1.6. Loop Detection . . . . . . . . . . . . . . . . . . . 32 88 4.2. Comparison of Topologies . . . . . . . . . . . . . . . . 33 89 5. Security Considerations . . . . . . . . . . . . . . . . . . . 33 90 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35 91 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 35 92 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 35 93 8.1. Normative References . . . . . . . . . . . . . . . . . . 35 94 8.2. Informative References . . . . . . . . . . . . . . . . . 36 95 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 37 97 1. Introduction 99 Real-time Transport Protocol (RTP) [RFC3550] topologies describe 100 methods for interconnecting RTP entities and their processing 101 behavior of RTP and RTCP. This document tries to address past and 102 existing confusion, especially with respect to terms not defined in 103 RTP but in common use in the conversational communication industry, 104 such as MCU. In doing so, this memo provides a common information 105 basis for future discussion and specification work. It attempts to 106 clarify and explain sections of the Real-time Transport Protocol 107 (RTP) spec [RFC3550] in an informal way. It is not intended to 108 update or change what is normatively specified within RFC 3550. 110 When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was 111 developed the main emphasis lay in the efficient support of point to 112 point and small multipoint scenarios without centralized multipoint 113 control. However, in practice, many small multipoint conferences 114 operate utilizing devices known as Multipoint Control Units (MCUs). 115 MCUs may implement Mixer or Translator (in RTP [RFC3550] terminology) 116 functionality and signalling support. They may also contain 117 additional application functionality. This document focuses on the 118 media transport aspects of the MCU that can be realized using RTP, as 119 discussed below. Further considered are the properties of Mixers and 120 Translators, and how some types of deployed MCUs deviate from these 121 properties. 123 2. Definitions 125 2.1. Glossary 127 ASM: Any Source Multicast 129 AVPF: The Extended RTP Profile for RTCP-based Feedback 131 CSRC: Contributing Source 133 Link: The data transport to the next IP hop 135 MCU: Multipoint Control Unit 137 Path: The concatenation of multiple links, resulting in an end-to- 138 end data transfer. 140 PtM: Point to Multipoint 142 PtP: Point to Point 144 SSM: Source-Specific Multicast 145 SSRC: Synchronization Source 147 3. Topologies 149 This subsection defines several topologies that are relevant for 150 codec control but also RTP usage in other contexts. The section 151 starts with point to point cases, without and with middleboxes. Then 152 follows a number of different methods for establishing point to 153 multipoint communication. These are structure around the most 154 fundamental enabler, i.e. multicast, a mesh of connections, 155 translators, mixers and source projection middlebox, to finally 156 discuss MCUs. The section ends by discussing de-composed endpoints, 157 asymmetric middlebox behaviors and combining topologies. 159 The topologies may be referenced in other documents by a shortcut 160 name, indicated by the prefix "Topo-". 162 For each of the RTP-defined topologies, we discuss how RTP, RTCP, and 163 the carried media are handled. With respect to RTCP, we also discuss 164 the handling of RTCP feedback messages as defined in [RFC4585] and 165 [RFC5104]. Any important differences between the two will be 166 illuminated in the discussion. At this stage we don't intended to 167 discuss in detail how each and every feedback messages should be 168 treated in the various topologies. 170 3.1. Point to Point 172 Shortcut name: Topo-Point-to-Point 174 The Point to Point (PtP) topology (Figure 1) consists of two 175 endpoints, communicating using unicast. Both RTP and RTCP traffic 176 are conveyed endpoint-to-endpoint, using unicast traffic only (even 177 if, in exotic cases, this unicast traffic happens to be conveyed over 178 an IP-multicast address). 180 +---+ +---+ 181 | A |<------->| B | 182 +---+ +---+ 184 Figure 1: Point to Point 186 The main property of this topology is that A sends to B, and only B, 187 while B sends to A, and only A. This avoids all complexities of 188 handling multiple endpoints and combining the requirements from them. 189 Note that an endpoint can still use multiple RTP Synchronization 190 Sources (SSRCs) in an RTP session. The number of RTP sessions in use 191 between A and B can also be of any number. 193 RTCP feedback messages for the indicated SSRCs are communicated 194 directly between the endpoints. Therefore, this topology poses 195 minimal (if any) issues for any feedback messages. For RTP sessions 196 which use multiple SSRC per endpoint it can be relevant to implement 197 support for cross reporting suppression as defined in "Real-Time 198 Transport Protocol (RTP) Considerations for Endpoints Sending 199 Multiple Media Streams" [I-D.lennox-avtcore-rtp-multi-stream]. 201 3.2. Point to Point via Middlebox 203 This section discusses cases where two endpoints communicate but have 204 one or more middlebox involved in the RTP session. 206 3.2.1. Translators 208 Shortcut name: Topo-PtP-Translator 210 Two main categories of Translators can be distinguished; Transport 211 Translators and Media translators. Both Translator types share 212 common attributes that separate them from Mixers. For each media 213 stream that the Translator receives, it generates an individual 214 stream in the other domain. A translator keeps the SSRC for a stream 215 across the translation, whereas a Mixer can select a single media 216 stream, or send out multiple mixed media streams, but always under 217 its own SSRC, possibly using the CSRC field to indicate the source(s) 218 of the content. Mixers are more common in point to multipoint cases 219 than in PtP. The reason is that in PtP use cases the primary focus 220 is interoperability, such as transcoding to a codec the receiver 221 supports, which can be done by a media translator. 223 As specified in Section 7.1 of [RFC3550], the SSRC space is common 224 for all participants in the RTP session, independent of on which side 225 of the Translator the session resides. Therefore, it is the 226 responsibility of the participants to run SSRC collision detection, 227 and the SSRC is thus a field the Translator cannot change. Any SDES 228 information associated with a SSRC or CSRC also needs to be forwarded 229 between the domains for any SSRC/CSRC used in the different domains. 231 A Translator commonly does not use an SSRC of its own, and is not 232 visible as an active participant in the session. One reason to have 233 its own SSRC is when a Translator acts as a quality monitor that 234 sends RTCP reports and therefore is required to have an SSRC. 235 Another example is the case when a Translator is prepared to use RTCP 236 feedback messages. This may, for example, occur in a translator 237 configured to detect packet loss of important video packets and wants 238 to trigger repair by the media sender, by sending feedback messages. 239 While such feedback could use the SSRC of the target for the 240 translator, but this in turn would require translation of the targets 241 RTCP reports to make them consistent. It may be simpler to expose an 242 additional SSRC in the session, the only concern are endpoints 243 failing to support the full RTP specification, thus having issues 244 with multiple SSRCs reporting on the RTP streams sent by that 245 endpoint. 247 In general, a Translator implementation should consider which RTCP 248 feedback messages or codec-control messages it needs to understand in 249 relation to the functionality of the Translator itself. This is 250 completely in line with the requirement to also translate RTCP 251 messages between the domains. 253 3.2.1.1. Transport Relay/Anchoring 255 There exist a number of different types of middleboxes that might be 256 inserted between two RTP endpoints on the transport level, e.g. 257 perform changes on the IP/UDP headers, and are, therefore, basic 258 transport translators. These middleboxes come in many variations 259 including NAT [RFC3022] traversal by pinning the media path to a 260 public address domain relay, network topologies where the media flow 261 is required to pass a particular point for audit by employing 262 relaying, or preserving privacy by hiding each peers transport 263 addresses to the other party. Other protocols or functionalities 264 that provide this behavior are TURN [RFC5766] servers, Session Border 265 Gateways and Media Processing Nodes with media anchoring 266 functionalities. 268 +---+ +---+ +---+ 269 | A |<------>| T |<------->| B | 270 +---+ +---+ +---+ 272 Figure 2: Point to Point with Translator 274 What is common for these functions is that they are normally 275 transparent on RTP level, i.e. they perform no changes on any RTP or 276 RTCP packet fields, only on the lower layers. However, they may 277 effect the path the RTP and RTCP packets are routed between the 278 endpoints in the RTP session, and thereby only indirectly affect the 279 RTP session. For this reason, one could believe that transport 280 translator type middleboxes do not need to included in this document. 281 However, this topology can raise additional requirements the RTP 282 implementation and its interactions with the signalling solution. 283 Both in signalling and in certain RTCP field other network addresses 284 than those of the relay can occur, due to that B has different 285 network address than the relay (T). However, implementation not 286 capable of this will neither not work when endpoints are subject to 287 NAT. 289 3.2.1.2. Transport Translator 291 Transport Translators (Topo-Trn-Translator) do not modify the media 292 stream itself, but are concerned with transport parameters. 293 Transport parameters, in the sense of this section, comprise the 294 transport addresses (to bridge different domains such unicast to 295 multicast) and the media packetization to allow other transport 296 protocols to be interconnected to a session (in gateways). Of the 297 transport Translators, this memo is primarily interested in those 298 that use RTP on both sides, and this is assumed henceforth. 299 Translators that bridge between different protocol worlds need to be 300 concerned about the mapping of the SSRC/CSRC (Contributing Source) 301 concept to the non-RTP protocol. When designing a Translator to a 302 non-RTP-based media transport, one crucial factor lies in how to 303 handle different sources and their identities. This problem space is 304 not discussed henceforth. 306 The most basic transport translators that operate below RTP level was 307 already discussed in Section 3.2.1.1. 309 3.2.1.3. Media Translator 311 Media Translators (Topo-Media-Translator), in contrast, modify the 312 media stream itself. This process is commonly known as transcoding. 313 The modification of the media stream can be as small as removing 314 parts of the stream, and it can go all the way to a full transcoding 315 (down to the sample level or equivalent) utilizing a different media 316 codec. Media Translators are commonly used to connect entities 317 without a common interoperability point in the media encoding. 319 Stand-alone Media Translators are rare. Most commonly, a combination 320 of Transport and Media Translators are used to translate both the 321 media stream and the transport aspects of a stream between two 322 transport domains (or clouds). 324 When media translation occurs, the Translator's task regarding 325 handling of RTCP traffic becomes substantially more complex. In this 326 case, the Translator needs to rewrite B's RTCP Receiver Report before 327 forwarding them to A. The rewriting is needed as the stream received 328 by B is not the same stream as the other participants receive. For 329 example, the number of packets transmitted to B may be lower than 330 what A sends, due to the different media format and data rate. 331 Therefore, if the Receiver Reports were forwarded without changes, 332 the extended highest sequence number would indicate that B were 333 substantially behind in reception, while it most likely it would not 334 be. Therefore, the Translator must translate that number to a 335 corresponding sequence number for the stream the Translator received. 336 Similar arguments can be made for most other fields in the RTCP 337 Receiver Reports. 339 A media Translator may in some cases act on behalf of the "real" 340 source and respond to RTCP feedback messages. This may occur, for 341 example, when a receiver requests a bandwidth reduction, and the 342 media Translator has not detected any congestion or other reasons for 343 bandwidth reduction between the media source and itself. In that 344 case, it is sensible that the media Translator reacts to the codec 345 control messages itself, for example, by transcoding to a lower media 346 rate. 348 A variant of translator behaviour worth pointing out is the one 349 depicted in Figure 3 of an endpoint A sends a media flow to B. On 350 the path there is a device T that on A's behalf does something with 351 the media streams, for example adds an RTP session with FEC 352 information for A's media streams. In this case, T needs to bind the 353 new FEC streams to A's media stream, for example by using the same 354 CNAME as A. 356 +------+ +------+ +------+ 357 | | | | | | 358 | A |------->| T |-------->| B | 359 | | | |---FEC-->| | 360 +------+ +------+ +------+ 362 Figure 3: When De-composition is a Translator 364 This type of functionality where T does something with the media 365 stream on behalf of A is covered under the media translator 366 definition. 368 3.2.2. Back to Back RTP sessions 370 There exist middleboxes that interconnect two endpoints through 371 themselves not by being part of a common RTP session. Instead they 372 establish two different RTP sessions, one between A and the middlebox 373 (MB) and another between the MB and B. 375 |<--Session A-->| |<--Session B-->| 376 +------+ +------+ +------+ 377 | A |------->| MB |-------->| B | 378 +------+ +------+ +------+ 380 Figure 4: When De-composition is a Translator 382 The MB acts as a application level gateway and bridges the two RTP 383 session. This bridging can be as basic as forwarding the RTP 384 payloads between the sessions, or more complex including media 385 transcoding. The difference with the single RTP session context is 386 the handling of the SSRCs and the other session related identifiers, 387 such as CNAMEs. With two different RTP sessions these can be freely 388 changed and it becomes the MB task to maintain the right relations. 390 The signalling or other above-RTP level functionalities referencing 391 RTP media streams may be what is most impacted by using two RTP 392 sessions and changing identifiers. The structure with two RTP 393 sessions also puts a congestion control requirement on the middlebox, 394 because it becomes fully responsible for the media stream it sources 395 into each of the sessions. 397 This can be solved locally or by bridging also statistics from the 398 receiving endpoint. However, from an implementation point this 399 requires the implementation to support dealing with a number of 400 inconsistencies. First, packet loss must be detected for an RTP flow 401 sent from A to the MB, and that loss must be reported through a 402 skipped sequence number in the flow from the MB to B. This coupling 403 and the resulting inconsistencies is conceptually easier to handle 404 when considering the two flows as belonging to a single RTP session. 406 3.3. Point to Multipoint Using Multicast 408 Multicast is a IP layer functionality that is available in some 409 networks. Two main flavors can be distinguished: Any Source 410 Multicast (ASM) where any multicast group participant can send to the 411 group address and expect the packet to reach all group participants; 412 and Source Specific Multicast (SSM), where only a particular IP host 413 sends to the multicast group. Both these models are discussed below 414 in their respective section. 416 3.3.1. Any Source Multicast (ASM) 418 Shortcut name: Topo-ASM (was Topo-Multicast) 419 +-----+ 420 +---+ / \ +---+ 421 | A |----/ \---| B | 422 +---+ / Multi- \ +---+ 423 + Cast + 424 +---+ \ Network / +---+ 425 | C |----\ /---| D | 426 +---+ \ / +---+ 427 +-----+ 429 Figure 5: Point to Multipoint Using Multicast 431 Point to Multipoint (PtM) is defined here as using a multicast 432 topology as a transmission model, in which traffic from any 433 participant reaches all the other participants, except for cases such 434 as: 436 o packet loss, or 438 o when a participant does not wish to receive the traffic for a 439 specific multicast group and, therefore, has not subscribed to the 440 IP-multicast group in question. This scenario can occur, for 441 example, where a multi-media session is distributed using two or 442 more multicast groups and a participant is subscribed only to a 443 subset of these sessions. 445 In the above context, "traffic" encompasses both RTP and RTCP 446 traffic. The number of participants can vary between one and many, 447 as RTP and RTCP scale to very large multicast groups (the theoretical 448 limit of the number of participants in a single RTP session is in the 449 range of billions). The above can be realized using Any Source 450 Multicast (ASM). 452 For feedback usage, it is useful to define a "small multicast group" 453 as a group where the number of participants is so low (and other 454 factors such as the connectivity is so good) that it allows the 455 participants to use early or immediate feedback, as defined in AVPF 456 [RFC4585]. Even when the environment would allow for the use of a 457 small multicast group, some applications may still want to use the 458 more limited options for RTCP feedback available to large multicast 459 groups, for example when there is a likelyhood that the threshold of 460 the small multicast group (in terms of participants) may be exceeded 461 during the lifetime of a session. 463 RTCP feedback messages in multicast reach, like media, every 464 subscriber (subject to packet losses and multicast group 465 subscription). Therefore, the feedback suppression mechanism 466 discussed in [RFC4585] is typically required. Each individual node 467 needs to process every feedback message it receives, not to determine 468 if it is affected or if the feedback message applies only to some 469 other participant, but also to derive timing restriction for the 470 sending of its own feedback messages, if any. 472 3.3.2. Source Specific Multicast (SSM) 474 In Any Source Multicast, any of the participants can send to all the 475 other participants, by sending a packet to the multicast group. In 476 contrast, Source Specific Multicast [RFC4607] refers to scenarios 477 where only a single source (Distribution Source) can send to the 478 multicast group, creating a topology that looks like the one below: 480 +--------+ +-----+ 481 |Media | | | Source-specific 482 |Sender 1|<----->| D S | Multicast 483 +--------+ | I O | +--+----------------> R(1) 484 | S U | | | | 485 +--------+ | T R | | +-----------> R(2) | 486 |Media |<----->| R C |->+ | : | | 487 |Sender 2| | I E | | +------> R(n-1) | | 488 +--------+ | B | | | | | | 489 : | U | +--+--> R(n) | | | 490 : | T +-| | | | | 491 : | I | |<---------+ | | | 492 +--------+ | O |F|<---------------+ | | 493 |Media | | N |T|<--------------------+ | 494 |Sender M|<----->| | |<-------------------------+ 495 +--------+ +-----+ RTCP Unicast 497 FT = Feedback Target 498 Transport from the Feedback Target to the Distribution 499 Source is via unicast or multicast RTCP if they are not 500 co-located. 502 Figure 6: Point to Multipoint using Source Specific Multicast 504 In the SSM topology (Figure 6) a number of RTP sources (1 to M) are 505 allowed to send media to the SSM group. These send media to a 506 dedicated distribution source, which then forwards the media streams 507 to the multicast group on behalf of the original senders. The media 508 streams reach the Receivers (R(1) to R(n)). The Receivers' RTCP 509 cannot be sent to the multicast group, as the SSM multicast group by 510 definition has only a single source. To support RTCP, an RTP 511 extension for SSM [RFC5760] was defined. It uses unicast 512 transmission to send RTCP from each of the receivers to one or more 513 Feedback Targets (FT). The feedback targets relay the RTCP 514 unmodified, or provide summary of the participants RTCP reports 515 towards the whole group by forwarding the RTCP traffic to the 516 distribution source. Figure 6 only shows a single feedback target 517 integrated in the distribution source, but for scalability the FT can 518 be many and have responsibility for sub-groups of the receivers. For 519 summary reports, however, there must be a single feedback aggregating 520 all the summaries to a common message to the whole receiver group. 522 The RTP extension for SSM specifies how feedback (both reception 523 information and specific feedback events) are handled. The more 524 general problems associated with the use of multicast, where everyone 525 receives what the distribution source sends needs to be accounted 526 for. 528 The result of this is some common behaviours for RTP multicast: 530 1. Multicast applications often use a group of RTP sessions, not 531 one. Each endpoint needs to be a member of most or all of these 532 RTP sessions in order to perform well. 534 2. Within each RTP session, the number of media sinks is likely to 535 be much larger than the number of RTP sources. 537 3. Multicast applications need signalling functions to identify the 538 relationships between RTP sessions. 540 4. Multicast applications need signalling functions to identify the 541 relationships between SSRCs in different RTP sessions. 543 All multicast configurations share a signalling requirement: all of 544 the participants need to have the same RTP and payload type 545 configuration. Otherwise, A could, for example, be using payload 546 type 97 to identify the video codec H.264, while B would identify it 547 as MPEG-2. 549 Security solutions for this type of group communications are also 550 challenging. First, the key-management and the security protocol 551 must support group communication. Source authentication becomes more 552 difficult and requires special solutions. For more discussion on 553 this please review Options for Securing RTP Sessions 554 [I-D.ietf-avtcore-rtp-security-options]. 556 3.3.3. SSM with Local Unicast Resources 558 [RFC6285] "Unicast-Based Rapid Acquisition of Multicast RTP Sessions" 559 results in additional extensions to SSM Topology. 561 ----------- -------------- 562 | |------------------------------------>| | 563 | |.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.->| | 564 | | | | 565 | Multicast | ---------------- | | 566 | Source | | Retransmission | | | 567 | |-------->| Server (RS) | | | 568 | |.-.-.-.->| | | | 569 | | | ------------ | | | 570 ----------- | | Feedback | |<.=.=.=.=.| | 571 | | Target (FT)| |<~~~~~~~~~| RTP Receiver | 572 PRIMARY MULTICAST | ------------ | | (RTP_Rx) | 573 RTP SESSION with | | | | 574 UNICAST FEEDBACK | | | | 575 | | | | 576 - - - - - - - - - - - |- - - - - - - - |- - - - - |- - - - - - - |- - 577 | | | | 578 UNICAST BURST | ------------ | | | 579 (or RETRANSMISSION) | | Burst/ | |<~~~~~~~~>| | 580 RTP SESSION | | Retrans. | |.........>| | 581 | |Source (BRS)| |<.=.=.=.=>| | 582 | ------------ | | | 583 | | | | 584 ---------------- -------------- 586 -------> Multicast RTP Flow 587 .-.-.-.> Multicast RTCP Flow 588 .=.=.=.> Unicast RTCP Reports 589 ~~~~~~~> Unicast RTCP Feedback Messages 590 .......> Unicast RTP Flow 592 Figure 7 594 The Rapid acquisition extension allows an endpoint joining an SSM 595 multicast session to request media starting with the last sync-point 596 (from where media can be decoded without prior packets) to be sent at 597 high speed until such time where, after decoding of these bursted 598 media packets, the correct media timing is established, i.e. media 599 packets are received within adequate buffer intervals for this 600 application. This is accomplished by first establishing an unicast 601 PtP RTP session between the BRS (Figure 7) and the RTP Receiver. 602 That session is used to transmit cached packets from the multicast 603 group at higher then nominal speed so to synchronize the receiver to 604 the ongoing multicast packet flow. Once the RTP receiver and its 605 decoder have caught up with the multicast session's current delivery, 606 the receiver switches over to receiving from the multicast group 607 directly. The (still existing) PtP RTP session can be used as a 608 repair channel, i.e. for RTP Retransmission traffic of those packets 609 that were not received from the multicast group. 611 3.4. Point to Multipoint Using Mesh 613 Shortcut name: Topo-Mesh 615 +---+ +---+ 616 | A |<---->| B | 617 +---+ +---+ 618 ^ ^ 619 \ / 620 \ / 621 v v 622 +---+ 623 | C | 624 +---+ 626 Figure 8: Point to Multi-Point using Mesh 628 Based on the RTP session definition, it is clearly possible to have a 629 joint RTP session over multiple unicast transport flows like the 630 above three endpoint joint session. In this case, A needs to send 631 its' media streams and RTCP packets to both B and C over their 632 respective transport flows. As long as all participants do the same, 633 everyone will have a joint view of the RTP session. 635 This doesn't create any additional requirements beyond the need to 636 have multiple transport flows associated with a single RTP session. 637 Note that an endpoint may use a single local port to receive all 638 these transport flows, or it might have separate local reception 639 ports for each of the endpoints. 641 An alternative structure for establishing the above topology is to 642 use independent RTP sessions between each pair of peers, i.e. three 643 different RTP sessions. In some scenarios, the same RTP media stream 644 is being sent from each sending endpoint. In others, some form of 645 local adaptation takes place in one or more of the RTP media streams, 646 rendering them non-identical. From a topologies viewpoint, a 647 difference exists in the behaviours around RTCP. For example, when a 648 single RTP session spans all three endpoints and their connecting 649 flows, a RTCP bandwidth is calculated and used for this single one 650 joint session. In contrast, when there are multiple independent RTP 651 sessions, each has its local RTCP bandwidth allocation. Also, when 652 multiple sessions are used, endpoints not directly involved in these 653 sessions do not have any awareness of the conditions occurring in 654 sessions not involving that endpoint. For example, in case of the 655 three endpoint configuration above, endpoint A has no awareness of 656 the conditions occurring in the session between endpoints B and C 657 (whereas, if a single RTP session were used, it would have such 658 awareness). Loop detection is also affected. With independent RTP 659 sessions, the SSRC/CSRC can't be used to determine when a endpoint 660 receives its own media stream or a mixed media stream including its 661 own media stream a condition known as a loop. The identification of 662 loops and, in most cases, its avoidance, has to be achieved by other 663 means, for example through signaling, or the use of an RTP external 664 name space binding SSRC/CSRC among any communicating RTP sessions in 665 the mesh. 667 3.5. Point to Multipoint Using the RFC 3550 Translator 669 This section discusses some additional usages related to point to 670 multipoint of Translators compared to the point to point only cases 671 in Section 3.2.1. 673 3.5.1. Relay - Transport Translator 675 Shortcut name: Topo-PtM-Trn-Translator 677 This section discusses Transport Translator only usages to enable 678 multipoint sessions. 680 +-----+ 681 +---+ / \ +------------+ +---+ 682 | A |<---/ \ | |<---->| B | 683 +---+ / Multi- \ | | +---+ 684 + Cast +->| Translator | 685 +---+ \ Network / | | +---+ 686 | C |<---\ / | |<---->| D | 687 +---+ \ / +------------+ +---+ 688 +-----+ 690 Figure 9: Point to Multipoint Using Multicast 692 Figure 9 depicts an example of a Transport Translator performing at 693 least IP address translation. It allows the (non-multicast-capable) 694 participants B and D to take part in an any source multicast session 695 by having the Translator forward their unicast traffic to the 696 multicast addresses in use, and vice versa. It must also forward B's 697 traffic to D, and vice versa, to provide each of B and D with a 698 complete view of the session. 700 +---+ +------------+ +---+ 701 | A |<---->| |<---->| B | 702 +---+ | | +---+ 703 | Translator | 704 +---+ | | +---+ 705 | C |<---->| |<---->| D | 706 +---+ +------------+ +---+ 708 Figure 10: RTP Translator (Relay) with Only Unicast Paths 710 Another Translator scenario is depicted in Figure 10. Herein, the 711 Translator connects multiple users of a conference through unicast. 712 This can be implemented using a very simple transport Translator, 713 which in this document is called a relay. The relay forwards all 714 traffic it receives, both RTP and RTCP, to all other participants. 715 In doing so, a multicast network is emulated without relying on a 716 multicast-capable network infrastructure. 718 For RTCP feedback this results in a similar set of considerations 719 those described in the ASM RTP topology. It also puts some 720 additional signalling requirements onto the session establishment; 721 for example, a common configuration of RTP payload types is required. 723 3.5.2. Media Translator 725 In the context of multipoint communications a Media Translator is not 726 providing new mechanisms to establish a multipoint session. It is 727 much more an enabler or facilitator that ensures one or some sub-set 728 of session participants can participate in the session. 730 If B in Figure 9 were behind a limited network path, the Translator 731 may perform media transcoding to allow the traffic received from the 732 other participants to reach B without overloading the path. This 733 transcoding can help the other participants in the Multicast part of 734 the session, by not requiring the quality transmitted by A to be 735 lowered to the nitrates that B is actually capable of receiving. 737 3.6. Point to Multipoint Using the RFC 3550 Mixer Model 739 Shortcut name: Topo-Mixer 741 A Mixer is a middlebox that aggregates multiple RTP streams, which 742 are part of a session, by generating a new RTP stream and, in most 743 cases, by manipulation of the media data. One common application for 744 a Mixer is to allow a participant to receive a session with a reduced 745 amount of resources. 747 +-----+ 748 +---+ / \ +-----------+ +---+ 749 | A |<---/ \ | |<---->| B | 750 +---+ / Multi- \ | | +---+ 751 + Cast +->| Mixer | 752 +---+ \ Network / | | +---+ 753 | C |<---\ / | |<---->| D | 754 +---+ \ / +-----------+ +---+ 755 +-----+ 757 Figure 11: Point to Multipoint Using the RFC 3550 Mixer Model 759 A Mixer can be viewed as a device terminating the media streams 760 received from other session participants. Using the media data from 761 the received media streams, a Mixer generates a media stream that is 762 sent to the session participant. 764 The content that the Mixer provides is the mixed aggregate of what 765 the Mixer receives over the PtP or PtM paths, which are part of the 766 same conference session. 768 The Mixer is the content source, as it mixes the content (often in 769 the uncompressed domain) and then encodes it for transmission to a 770 participant. The CSRC Count (CC) and CSRC fields in the RTP header 771 can be used to indicate the contributors of to the newly generated 772 stream. The SSRCs of the to-be-mixed streams on the Mixer input 773 appear as the CSRCs at the Mixer output. That output stream uses a 774 unique SSRC that identifies the Mixer's stream. The CSRC should be 775 forwarded between the different conference participants to allow for 776 loop detection and identification of sources that are part of the 777 global session. Note that Section 7.1 of RFC 3550 requires the SSRC 778 space to be shared between domains for these reasons. This also 779 implies that any SDES information normally needs to be forwarded 780 across the mixer. 782 The Mixer is responsible for generating RTCP packets in accordance 783 with its role. It is a receiver and should therefore send receiver 784 reports for the media streams it receives. In its role as a media 785 sender, it should also generate sender reports for those media 786 streams it sends. As specified in Section 7.3 of RFC 3550, a Mixer 787 must not forward RTCP unaltered between the two domains. 789 The Mixer depicted in Figure 11 is involved in three domains that 790 need to be separated: the any source multicast network (including 791 participants A and C), participant B, and participant D. Assuming 792 all four participants in the conference are interested in receiving 793 content from each other participant, the Mixer produces different 794 mixed streams for B and D, as the one to B may contain content 795 received from D, and vice versa. However, the Mixer may only need 796 one SSRC per media type in each domain that is the receiving entity 797 and transmitter of mixed content. 799 In the multicast domain, a Mixer still needs to provide a mixed view 800 of the other domains. This makes the Mixer simpler to implement and 801 avoids any issues with advanced RTCP handling or loop detection, 802 which would be problematic if the Mixer were providing non-symmetric 803 behavior. Please see Section 3.11 for more discussion on this topic. 804 However, the mixing operation in each domain could potentially be 805 different. 807 A Mixer is responsible for receiving RTCP feedback messages and 808 handling them appropriately. The definition of "appropriate" depends 809 on the message itself and the context. In some cases, the reception 810 of a codec-control message by the Mixer may result in the generation 811 and transmission of RTCP feedback messages by the Mixer to the 812 participants in the other domain(s). In other cases, a message is 813 handled by the Mixer itself and therefore not forwarded to any other 814 domain. 816 When replacing the multicast network in Figure 11 (to the left of the 817 Mixer) with individual unicast paths as depicted in Figure 12, the 818 Mixer model is very similar to the one discussed in Section 3.9 819 below. Please see the discussion in Section 3.9 about the 820 differences between these two models. 822 +---+ +------------+ +---+ 823 | A |<---->| |<---->| B | 824 +---+ | | +---+ 825 | Mixer | 826 +---+ | | +---+ 827 | C |<---->| |<---->| D | 828 +---+ +------------+ +---+ 830 Figure 12: RTP Mixer with Only Unicast Paths 832 Lets now discuss in more detail different mixing operations that a 833 mixer can perform and how they can affect the RTP and RTCP. 835 3.6.1. Media Mixing 837 The media mixing mixer is likely the one that most think of when they 838 hear the term "mixer". Its basic pattern of operation is that it 839 receives media streams from (typically several) participants. Of 840 those, it selects (either through static configuration or by dynamic, 841 content dependent means such as voice activation) the stream(s) to be 842 included in a media domain mix. Then it creates a single outgoing 843 stream from this mix. 845 The most commonly deployed media mixer is probably the audio mixer, 846 used in voice conferencing, where the output consists of a mixture of 847 all the input streams; this needs minimal signalling to be 848 successfully set up. Audio mixing is relatively straightforward and 849 commonly possible for a reasonable number of participants. Lets 850 assume that you want to mix N streams from different participants. 851 The mixer needs to decode those N streams, typically into the sample 852 domain. Then it needs to produce N or N+1 mixes, the reasons that 853 different mixes are needed being that each contributing source get a 854 mix of all other sources except its own, as this would result in an 855 echo. When N is lower than the number of all participants one may 856 produce a Mix of all N streams for the group that are currently not 857 included in the mix, thus N+1 mixes. These audio streams are then 858 encoded again, RTP packetized and sent out. In many cases, audio 859 level normalization is also required before the actual mixing 860 process. 862 Video can't really be "mixed" and produce something particularly 863 useful for the users, however creating an composition out of the 864 contributed video streams is possible and known as "tiling". For 865 example the reconstructed, appropriately scaled down videos can be 866 spatially arranged in a set of tiles, each tile containing the video 867 from a participant. Tiles can be of different sizes, so that, for 868 example, a particularly important participant, or the loudest 869 speaker, is being shown on in larger tile than other participants. A 870 self-picture can be included in the tiling, which can either be 871 locally produced or be a feedback from a received and reconstructed 872 video image (allowing for confidence monitoring, the participant sees 873 himself/herself just as other participants see him/her). The tiling 874 normally operates on reconstructed video in the sample domain. The 875 tiled image is encoded, packetized, and sent by the mixer. It is 876 possible that a middlebox with media mixing duties contains only a 877 single mixer of the aforementioned type, in which case all 878 participants necessarily see the same tiled video, even if it is 879 being sent over different RTP streams. More common, however, are 880 mixing arrangement where an individual mixer is available for each 881 outgoing port of the middlebox, allowing individual compositions for 882 each participant. 884 One problem with media mixing is that it consumes both large amount 885 of media processing (for the actual mixing process in the 886 uncompressed domain) and encoding resources (for the encoding of the 887 mixed signal). Another problem is the quality degradation created by 888 decoding and re-encoding the media that is encapsulated in the RTP 889 media stream, which is the result of the lossy nature of most, if not 890 all, commonly used media codecs. A third problem is the latency 891 introduced by the media mixing, which can be substantial and 892 annoyingly noticeable in case of video. The advantage of media 893 mixing is that it is quite simplistic for the clients to handle the 894 single media stream (which includes the mixed aggregate of many 895 sources), as they don't need to handle multiple decodings, local 896 mixing and composition. In fact, mixers were introduced in pre-RTP 897 times so that legacy, single stream receiving endpoints can 898 successfully participate in what a user would recognize as a 899 multiparty session. 901 +-A---------+ +-MIXER----------------------+ 902 | +-RTP1----| |-RTP1------+ +-----+ | 903 | | +-Audio-| |-Audio---+ | +---+ | | | 904 | | | AA1|--------->|---------+-+-|DEC|->| | | 905 | | | |<---------|MA1 <----+ | +---+ | | | 906 | | | | |(BA1+CA1)|\| +---+ | | | 907 | | +-------| |---------+ +-|ENC|<-| B+C | | 908 | +---------| |-----------+ +---+ | | | 909 +-----------+ | | | | 910 | | M | | 911 +-B---------+ | | E | | 912 | +-RTP2----| |-RTP2------+ | D | | 913 | | +-Audio-| |-Audio---+ | +---+ | I | | 914 | | | BA1|--------->|---------+-+-|DEC|->| A | | 915 | | | |<---------|MA2 <----+ | +---+ | | | 916 | | +-------| |(BA1+CA1)|\| +---+ | | | 917 | +---------| |---------+ +-|ENC|<-| A+C | | 918 +-----------+ |-----------+ +---+ | | | 919 | | M | | 920 +-C---------+ | | I | | 921 | +-RTP3----| |-RTP3------+ | X | | 922 | | +-Audio-| |-Audio---+ | +---+ | E | | 923 | | | CA1|--------->|---------+-+-|DEC|->| R | | 924 | | | |<---------|MA3 <----+ | +---+ | | | 925 | | +-------| |(BA1+CA1)|\| +---+ | | | 926 | +---------| |---------+ +-|ENC|<-| A+B | | 927 +-----------+ |-----------+ +---+ +-----+ | 928 +----------------------------+ 930 Figure 13: Session and SSRC details for Media Mixer 932 From an RTP perspective media mixing can be very straightforward as 933 can be seen in Figure 13. The mixer presents one SSRC towards the 934 receiving client, e.g. MA1 to Peer A; the associated stream of which 935 is the media mix of the other participants. As, in this example, 936 each peer receives a different version produced by the mixer, there 937 is no actual relation between the different RTP sessions in the 938 actual media or the transport level information. There are, however, 939 common relationships between RTP1-RTP3 namely SSRC space and identity 940 information. When A receives the MA1 stream which is a combination 941 of BA1 and CA1 streams, the mixer may include CSRC information in the 942 MA1 stream to identify the contributing source BA1 and CA1, allowing 943 the receiver to identify the contributing sources even if this were 944 not possible through the media itself or other signaling means. 946 The CSRC has, in turn, utility in RTP extensions, like the Mixer to 947 Client audio levels RTP header extension [RFC6465]. If the SSRC from 948 endpoint to mixer leg are used as CSRC in another RTP session, then 949 RTP1, RTP2 and RTP3 become one joint session as they have a common 950 SSRC space. At this stage, the mixer also need to consider which 951 RTCP information it needs to expose in the different legs. In the 952 above scenario, commonly, a mixer would expose nothing more than the 953 Source Description (SDES) information and RTCP BYE for CSRC leaving 954 the session. The main goal would be to enable the correct binding 955 against the application logic and other information sources. This 956 also enables loop detection in the RTP session. 958 3.6.2. Media Switching 960 Media switching mixers are commonly used in such limited 961 functionality scenarios where no, or only very limited, concurrent 962 presentation of multiple sources is required by the application. An 963 RTP Mixer based on media switching avoids the media decoding and 964 encoding cycle in the mixer, as it conceptually forwards the encoded 965 media stream as it was being sent to the mixer, but not the 966 decryption and re-encryption cycle as it rewrites RTP headers. 967 Forwarding media (in contrast to reconstructing-mixing-encoding 968 media) reduces the amount of computational resources needed in the 969 mixer and increases the media quality (both in terms of fidelity and 970 reduced latency) per transmitted bit. 972 A media switching mixer maintains a pool of SSRCs representing 973 conceptual or functional streams the mixer can produce. These 974 streams are created by selecting media from one of RTP media streams 975 received by the mixer and forwarded to the peer using the mixer's own 976 SSRCs. The mixer can switch between available sources if that is 977 required by the concept for the source, like currently active 978 speaker. Note that the mixer, in most cases, still need to perform a 979 certain amount of media processing, as many media formats do not 980 allow to "tune" into the stream at arbitrary points of their 981 bitstream. 983 To achieve a coherent RTP media stream from the mixer's SSRC, the 984 mixer needs to rewrite the incoming RTP packet's header. First the 985 SSRC field must be set to the value of the Mixer's SSRC. Secondly, 986 the sequence number must be the next in the sequence of outgoing 987 packets it sent. Thirdly the RTP timestamp value needs to be 988 adjusted using an offset that changes each time one switch media 989 source. Finally depending on the negotiation the RTP payload type 990 value representing this particular RTP payload configuration may have 991 to be changed if the different endpoint mixer legs have not arrived 992 on the same numbering for a given configuration. This also requires 993 that the different end-points do support a common set of codecs, 994 otherwise media transcoding for codec compatibility is still 995 required. 997 Lets consider the operation of media switching mixer that supports a 998 video conference with six participants (A-F) where the two latest 999 speakers in the conference are shown to each participants. Thus the 1000 mixer has two SSRCs sending video to each peer, and each peer is 1001 capable of locally handling two video streams simultaneously. 1003 +-A---------+ +-MIXER----------------------+ 1004 | +-RTP1----| |-RTP1------+ +-----+ | 1005 | | +-Video-| |-Video---+ | | | | 1006 | | | AV1|------------>|---------+-+------->| S | | 1007 | | | |<------------|MV1 <----+-+-BV1----| W | | 1008 | | | |<------------|MV2 <----+-+-EV1----| I | | 1009 | | +-------| |---------+ | | T | | 1010 | +---------| |-----------+ | C | | 1011 +-----------+ | | H | | 1012 | | | | 1013 +-B---------+ | | M | | 1014 | +-RTP2----| |-RTP2------+ | A | | 1015 | | +-Video-| |-Video---+ | | T | | 1016 | | | BV1|------------>|---------+-+------->| R | | 1017 | | | |<------------|MV3 <----+-+-AV1----| I | | 1018 | | | |<------------|MV4 <----+-+-EV1----| X | | 1019 | | +-------| |---------+ | | | | 1020 | +---------| |-----------+ | | | 1021 +-----------+ | | | | 1022 : : : : 1023 : : : : 1024 +-F---------+ | | | | 1025 | +-RTP6----| |-RTP6------+ | | | 1026 | | +-Video-| |-Video---+ | | | | 1027 | | | CV1|------------>|---------+-+------->| | | 1028 | | | |<------------|MV11 <---+-+-AV1----| | | 1029 | | | |<------------|MV12 <---+-+-EV1----| | | 1030 | | +-------| |---------+ | | | | 1031 | +---------| |-----------+ +-----+ | 1032 +-----------+ +----------------------------+ 1034 Figure 14: Media Switching RTP Mixer 1036 The Media Switching RTP mixer can, similarly to the Media Mixing 1037 Mixer, reduce the bit-rate required for media transmission towards 1038 the different peers by selecting and forwarding only a sub-set of RTP 1039 media streams it receives from the conference participants. In many 1040 practical cases, the link capacities of either direction between 1041 peers and mixer are the same, which effectively limits the subset to 1042 a single media stream. 1044 To ensure that a media receiver can correctly decode the RTP media 1045 stream after a switch, a state saving (frame-based) codec needs to 1046 start its decoding from independent refresh points or similar points 1047 in the bitstream. For some codecs, for example frame based speech 1048 and audio codecs, this is easily achieved by starting the decoding at 1049 RTP packet boundaries (proper packetization on the encoder side 1050 assumed), as each packet boundary provides a refresh point. For 1051 other (mostly video-) codecs, refresh points are less common in the 1052 bitstream or may not be present at all without an explicit request to 1053 the respective encoder. For this purpose there exists the Full Intra 1054 Request [RFC5104] RTCP codec control message. 1056 Also in this type of mixer one could consider to terminate the RTP 1057 sessions fully between the different end-point and mixer legs. The 1058 same arguments and considerations as discussed in Section 3.9 need to 1059 be taken into consideration and apply here. 1061 3.7. Source Projecting Middlebox 1063 Another method for handling media in the RTP mixer is to project all 1064 potential RTP sources (SSRCs) into a per end-point independent RTP 1065 session. The middlebox can select which of the potential sources 1066 that are currently actively transmitting media, despite that the 1067 middlebox, in another RTP session, may receive media from that end- 1068 point. This is similar to the media switching Mixer but has some 1069 important differences in RTP details. 1071 +-A---------+ +-Middlebox-----------------+ 1072 | +-RTP1----| |-RTP1------+ +-----+ | 1073 | | +-Video-| |-Video---+ | | | | 1074 | | | AV1|------------>|---------+-+------>| | | 1075 | | | |<------------|BV1 <----+-+-------| S | | 1076 | | | |<------------|CV1 <----+-+-------| W | | 1077 | | | |<------------|DV1 <----+-+-------| I | | 1078 | | | |<------------|EV1 <----+-+-------| T | | 1079 | | | |<------------|FV1 <----+-+-------| C | | 1080 | | +-------| |---------+ | | H | | 1081 | +---------| |-----------+ | | | 1082 +-----------+ | | M | | 1083 | | A | | 1084 +-B---------+ | | T | | 1085 | +-RTP2----| |-RTP2------+ | R | | 1086 | | +-Video-| |-Video---+ | | I | | 1087 | | | BV1|------------>|---------+-+------>| X | | 1088 | | | |<------------|AV1 <----+-+-------| | | 1089 | | | |<------------|CV1 <----+-+-------| | | 1090 | | | | : : : |: : : : : : : : :| | | 1091 | | | |<------------|FV1 <----+-+-------| | | 1092 | | +-------| |---------+ | | | | 1093 | +---------| |-----------+ | | | 1094 +-----------+ | | | | 1095 : : : : 1096 : : : : 1097 +-F---------+ | | | | 1098 | +-RTP6----| |-RTP6------+ | | | 1099 | | +-Video-| |-Video---+ | | | | 1100 | | | FV1|------------>|---------+-+------>| | | 1101 | | | |<------------|AV1 <----+-+-------| | | 1102 | | | | : : : |: : : : : : : : :| | | 1103 | | | |<------------|EV1 <----+-+-------| | | 1104 | | +-------| |---------+ | | | | 1105 | +---------| |-----------+ +-----+ | 1106 +-----------+ +---------------------------+ 1108 Figure 15: Media Projecting Middlebox 1110 In the six participant conference depicted above in (Figure 15) one 1111 can see that end-point A is aware of five incoming SSRCs, BV1-FV1. 1112 If this middlebox intends to have a similar behaviour as in 1113 Section 3.6.2 where the mixer provides the end-points with the two 1114 latest speaking end-points, then only two out of these five SSRCs 1115 need concurrently transmit media to A. As the middlebox selects the 1116 source in the different RTP sessions that transmit media to the end- 1117 points, each RTP media stream requires some rewriting of RTP header 1118 fields when being projected from one session into another. In 1119 particular, the sequence number needs to be consecutively incremented 1120 based on the packet actually being transmitted in each RTP session. 1121 Therefore, the RTP sequence number offset will change each time a 1122 source is turned on in a RTP session. The timestamp (possibly 1123 offset) stays the same. 1125 As the RTP sessions are independent, the SSRC numbers used can also 1126 be handled independently, thereby bypassing the requirement for SSRC 1127 collision detection and avoidance. On the other hand, tools such as 1128 remapping tables between the RTP sessions are required. For example, 1129 the stream that is being sent by endpoint B to the middlebox (BV1) 1130 may use an SSRC value of 12345678. When that media stream is sent to 1131 endpoint F by the middlebox, it can use any SSRC value, e.g. 1132 87654321. As a result, each endpoint may have a different view of 1133 the application usage of a particular SSRC. Any RTP level identity 1134 information, such as SDES items also needs to update the SSRC 1135 referenced, if the included SDES items are intended to be global. 1136 Thus the application must not use SSRC as references to RTP media 1137 streams when communicating with other peers directly. This also 1138 affects loop detection which will fail to work, as there is no common 1139 namespace and identities across the different legs in the 1140 communication session on RTP level. Instead this responsibility 1141 falls onto higher layers. 1143 The middlebox is also responsible to receive any RTCP codec control 1144 requests coming from an end-point, and decide if it can act on the 1145 request locally or needs to translate the request into the RTP 1146 session that contains the media source. Both end-points and the 1147 middlebox need to implement conference related codec control 1148 functionalities to provide a good experience. Commonly used are Full 1149 Intra Request to request from the media source to provide switching 1150 points between the sources, and Temporary Maximum Media Bit-rate 1151 Request (TMMBR) to enable the middlebox to aggregate congestion 1152 control responses towards the media source so to enable it to adjust 1153 its bit-rate (obviously only in case the limitation is not in the 1154 source to middlebox link). 1156 This version of the middlebox also puts different requirements on the 1157 end-point when it comes to decoder instances and handling of the RTP 1158 media streams providing media. As each projected SSRC can, at any 1159 time, provide media, the end-point either needs to be able to handle 1160 as many decoder instances as the middlebox received, or have 1161 efficient switching of decoder contexts in a more limited set of 1162 actual decoder instances to cope with the switches. The application 1163 also gets more responsibility to update how the media provided is to 1164 be presented to the user. 1166 Note, this could potentially be seen as a media translator which 1167 include an on/off logic as part of its media translation. The main 1168 difference would be a common global SSRC space in the case of the 1169 Media Translator and the mapped one used in the above. It also has 1170 mixer aspects, as the streams it provides are not basically 1171 translated version, but instead they have conceptual property 1172 assigned to them. Thus this topology appears to be some hybrid 1173 between the translator and mixer model. 1175 3.8. Point to Multipoint Using Video Switching MCUs 1177 Shortcut name: Topo-Video-switch-MCU 1179 +---+ +------------+ +---+ 1180 | A |------| Multipoint |------| B | 1181 +---+ | Control | +---+ 1182 | Unit | 1183 +---+ | (MCU) | +---+ 1184 | C |------| |------| D | 1185 +---+ +------------+ +---+ 1187 Figure 16: Point to Multipoint Using a Video Switching MCU 1189 This PtM topology was common before, although the RTCP-terminating 1190 MCUs, as discussed in the next section, where perhaps even more 1191 common. This topology, as well as the following one, was a result of 1192 lack of wide availability of IP multicast technologies, as well as 1193 the simplicity of content switching when compared to content mixing. 1194 The technology is commonly implemented in what is known as "Video 1195 Switching MCUs". 1197 A video switching MCU forwards to a participant a single media 1198 stream, selected from the available streams. The criteria for 1199 selection are often based on voice activity in the audio-visual 1200 conference, but other conference management mechanisms (like 1201 presentation mode or explicit floor control) are known to exist as 1202 well. 1204 The video switching MCU may also perform media translation to modify 1205 the content in bit-rate, encoding, or resolution. However, it still 1206 may indicate the original sender of the content through the SSRC. In 1207 this case, the values of the CC and CSRC fields are retained. 1209 If not terminating RTP, the RTCP Sender Reports are forwarded for the 1210 currently selected sender. All RTCP Receiver Reports are freely 1211 forwarded between the participants. In addition, the MCU may also 1212 originate RTCP control traffic in order to control the session and/or 1213 report on status from its viewpoint. 1215 The video switching MCU has most of the attributes of a Translator. 1216 However, its stream selection is a mixing behavior. This behavior 1217 has some RTP and RTCP issues associated with it. The suppression of 1218 all but one media stream results in most participants seeing only a 1219 subset of the sent media streams at any given time, often a single 1220 stream per conference. Therefore, RTCP Receiver Reports only report 1221 on these streams. Consequently, the media senders that are not 1222 currently forwarded receive a view of the session that indicates 1223 their media streams disappear somewhere en route. This makes the use 1224 of RTCP for congestion control, or any type of quality reporting, 1225 very problematic. 1227 To avoid the aforementioned issues, the MCU needs to implement two 1228 features. First, it needs to act as a Mixer (see Section 3.6) and 1229 forward the selected media stream under its own SSRC and with the 1230 appropriate CSRC values. Second, the MCU needs to modify the RTCP 1231 RRs it forwards between the domains. As a result, it is recommended 1232 that one implement a centralized video switching conference using a 1233 Mixer according to RFC 3550, instead of the shortcut implementation 1234 described here. 1236 3.9. Point to Multipoint Using RTCP-Terminating MCU 1238 Shortcut name: Topo-RTCP-terminating-MCU 1240 +---+ +------------+ +---+ 1241 | A |<---->| Multipoint |<---->| B | 1242 +---+ | Control | +---+ 1243 | Unit | 1244 +---+ | (MCU) | +---+ 1245 | C |<---->| |<---->| D | 1246 +---+ +------------+ +---+ 1248 Figure 17: Point to Multipoint Using Content Modifying MCUs 1250 In this PtM scenario, each participant runs an RTP point-to-point 1251 session between itself and the MCU. This is a very commonly deployed 1252 topology in multipoint video conferencing. The content that the MCU 1253 provides to each participant is either: 1255 a. a selection of the content received from the other participants, 1256 or 1258 b. the mixed aggregate of what the MCU receives from the other PtP 1259 paths, which are part of the same conference session. 1261 In case a), the MCU may modify the content in bit-rate, encoding, or 1262 resolution. No explicit RTP mechanism is used to establish the 1263 relationship between the original media sender and the version the 1264 MCU sends. In other words, the outgoing sessions typically use a 1265 different SSRC, and may well use a different payload type (PT), even 1266 if this different PT happens to be mapped to the same media type. 1267 This is a result of the individually negotiated session for each 1268 participant. 1270 In case b), the MCU is the content source as it mixes the content and 1271 then encodes it for transmission to a participant. According to RTP 1272 [RFC3550], the SSRC of the contributors are to be signalled using the 1273 CSRC/CC mechanism. In practice, today, most deployed MCUs do not 1274 implement this feature. Instead, the identification of the 1275 participants whose content is included in the Mixer's output is not 1276 indicated through any explicit RTP mechanism. That is, most deployed 1277 MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby 1278 indicating no available CSRC information, even if they could identify 1279 the content sources as suggested in RTP. 1281 The main feature that sets this topology apart from what RFC 3550 1282 describes is the breaking of the common RTP session across the 1283 centralized device, such as the MCU. This results in the loss of 1284 explicit RTP-level indication of all participants. If one were using 1285 the mechanisms available in RTP and RTCP to signal this explicitly, 1286 the topology would follow the approach of an RTP Mixer. The lack of 1287 explicit indication has at least the following potential problems: 1289 1. Loop detection cannot be performed on the RTP level. When 1290 carelessly connecting two misconfigured MCUs, a loop could be 1291 generated. 1293 2. There is no information about active media senders available in 1294 the RTP packet. As this information is missing, receivers cannot 1295 use it. It also deprives the client of information related to 1296 currently active senders in a machine-usable way, thus preventing 1297 clients from indicating currently active speakers in user 1298 interfaces, etc. 1300 Note that deployed MCUs (and endpoints) rely on signalling layer 1301 mechanisms for the identification of the contributing sources, for 1302 example, a SIP conferencing package [RFC4575]. This alleviates, to 1303 some extent, the aforementioned issues resulting from ignoring RTP's 1304 CSRC mechanism. 1306 As a result of the shortcomings of this topology, it is recommended 1307 to instead implement the Mixer concept as specified by RFC 3550. 1309 3.10. De-composite Endpoint 1311 The implementation of an application may desire to send a subset of 1312 the application's data to each of multiple devices, each with its own 1313 network address. A very basic use case for this would be to separate 1314 audio and video processing for a particular endpoint, like a 1315 conference room, into one device handling the audio and another 1316 handling the video, being interconnected by some control functions 1317 allowing them to behave as a single endpoint in all aspects except 1318 for transport Figure 18. 1320 Which decomposition scheme is possible is highly dependent on the RTP 1321 session usage. It is not really feasible to decompose one logical 1322 end-point into two different transport nodes in one RTP session. A 1323 third party monitor would report such an attempt as two entities 1324 being two different end-points with a CNAME collision. As a result, 1325 a fully RTP conformant de-composited endpoint is one where the 1326 different decomposed parts use separate RTP sessions to send and/or 1327 receive media streams intended for them. 1329 +---------------------+ 1330 | Endpoint A | 1331 | Local Area Network | 1332 | +------------+ | 1333 | +->| Audio |<+-RTP---\ 1334 | | +------------+ | \ +------+ 1335 | | +------------+ | +-->| | 1336 | +->| Video |<+-RTP-------->| B | 1337 | | +------------+ | +-->| | 1338 | | +------------+ | / +------+ 1339 | +->| Control |<+-SIP---/ 1340 | +------------+ | 1341 +---------------------+ 1343 Figure 18: De-composite End-Point 1345 In the above usage, let us assume that the different RTP sessions are 1346 used for audio and video. The audio and video parts, however, use a 1347 common CNAME and also have a common clock to ensure that 1348 synchronization and clock drift handling works, despite the 1349 decomposition. Also, the RTCP handling works correctly as long as 1350 only one part of the de-composite is part of each RTP session. That 1351 way any differences in the path between A's audio entity and B and 1352 A's video and B are related to different SSRCs in different RTP 1353 sessions. 1355 The requirement that can be derived from the above usage is that the 1356 transport flows for each RTP session might be under common control, 1357 but still are addressed to what looks like different endpoints (based 1358 on addresses and ports). This geometry cannot be accomplished using 1359 one RTP session, so in this case, multiple RTP sessions are needed. 1361 3.11. Non-Symmetric Mixer/Translators 1363 Shortcut name: Topo-Asymmetric 1365 It is theoretically possible to construct an MCU that is a Mixer in 1366 one direction and a Translator in another. The main reason to 1367 consider this would be to allow topologies similar to Figure 11, 1368 where the Mixer does not need to mix in the direction from B or D 1369 towards the multicast domains with A and C. Instead, the media 1370 streams from B and D are forwarded without changes. Avoiding this 1371 mixing would save media processing resources that perform the mixing 1372 in cases where it isn't needed. However, there would still be a need 1373 to mix B's stream towards D. Only in the direction B -> multicast 1374 domain or D -> multicast domain would it be possible to work as a 1375 Translator. In all other directions, it would function as a Mixer. 1377 The Mixer/Translator would still need to process and change the RTCP 1378 before forwarding it in the directions of B or D to the multicast 1379 domain. One issue is that A and C do not know about the mixed-media 1380 stream the Mixer sends to either B or D. Therefore, any reports 1381 related to these streams must be removed. Also, receiver reports 1382 related to A and C's media stream would be missing. To avoid A and C 1383 thinking that B and D aren't receiving A and C at all, the Mixer 1384 needs to insert locally generated reports reflecting the situation 1385 for the streams from A and C into B and D's Sender Reports. In the 1386 opposite direction, the Receiver Reports from A and C about B's and 1387 D's stream also need to be aggregated into the Mixer's Receiver 1388 Reports sent to B and D. Since B and D only have the Mixer as source 1389 for the stream, all RTCP from A and C must be suppressed by the 1390 Mixer. 1392 This topology is so problematic and it is so easy to get the RTCP 1393 processing wrong, that it is not recommended to implement this 1394 topology. 1396 3.12. Combining Topologies 1398 Topologies can be combined and linked to each other using Mixers or 1399 Translators. However, care must be taken in handling the SSRC/CSRC 1400 space. A Mixer does not forward RTCP from sources in other domains, 1401 but instead generates its own RTCP packets for each domain it mixes 1402 into, including the necessary Source Description (SDES) information 1403 for both the CSRCs and the SSRCs. Thus, in a mixed domain, the only 1404 SSRCs seen will be the ones present in the domain, while there can be 1405 CSRCs from all the domains connected together with a combination of 1406 Mixers and Translators. The combined SSRC and CSRC space is common 1407 over any Translator or Mixer. This is important to facilitate loop 1408 detection, something that is likely to be even more important in 1409 combined topologies due to the mixed behavior between the domains. 1410 Any hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, 1411 requires considerable thought on how RTCP is dealt with. 1413 4. Comparing Topologies 1415 The topologies discussed in Section 3 have different properties. 1416 This section first lists these properties and maps the different 1417 topologies to them. Please note that even if a certain property is 1418 supported within a particular topology concept, the necessary 1419 functionality may, in many cases, be optional to implement. 1421 Note: This section has not yet been updated with the new additions of 1422 topologies. 1424 4.1. Topology Properties 1426 4.1.1. All to All Media Transmission 1428 Multicast, at least Any Source Multicast (ASM), provides the 1429 functionality that everyone may send to, or receive from, everyone 1430 else within the session. MCUs, Mixers, and Translators may all 1431 provide that functionality at least on some basic level. However, 1432 there are some differences in which type of reachability they 1433 provide. 1435 The transport Translator function called "relay", in Section 3.5, is 1436 the one that provides the emulation of ASM that is closest to true 1437 IP-multicast-based, all to all transmission. Media Translators, 1438 Mixers, and the MCU variants do not provide a fully meshed forwarding 1439 on the transport level; instead, they only allow limited forwarding 1440 of content from the other session participants. 1442 The "all to all media transmission" requires that any media 1443 transmitting entity considers the path to the least capable receiver. 1444 Otherwise, the media transmissions may overload that path. 1445 Therefore, a media sender needs to monitor the path from itself to 1446 any of the participants, to detect the currently least capable 1447 receiver, and adapt its sending rate accordingly. As multiple 1448 participants may send simultaneously, the available resources may 1449 vary. RTCP's Receiver Reports help performing this monitoring, at 1450 least on a medium time scale. 1452 The transmission of RTCP automatically adapts to any changes in the 1453 number of participants due to the transmission algorithm, defined in 1454 the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] 1455 (when applicable). That way, the resources utilized for RTCP stay 1456 within the bounds configured for the session. 1458 4.1.2. Transport or Media Interoperability 1460 Translators, Mixers, and RTCP-terminating MCU all allow changing the 1461 media encoding or the transport to other properties of the other 1462 domain, thereby providing extended interoperability in cases where 1463 the participants lack a common set of media codecs and/or transport 1464 protocols. 1466 4.1.3. Per Domain Bit-Rate Adaptation 1467 Participants are most likely to be connected to each other with a 1468 heterogeneous set of paths. This makes congestion control in a Point 1469 to Multipoint set problematic. For the ASM and "relay" scenario, 1470 each individual sender has to adapt to the receiver with the least 1471 capable path. This is no longer necessary when Media Translators, 1472 Mixers, or MCUs are involved, as each participant only needs to adapt 1473 to the slowest path within its own domain. The Translator, Mixer, or 1474 MCU topologies all require their respective outgoing streams to 1475 adjust the bit-rate, packet-rate, etc., to adapt to the least capable 1476 path in each of the other domains. That way one can avoid lowering 1477 the quality to the least-capable participant in all the domains at 1478 the cost (complexity, delay, equipment) of the Mixer or Translator. 1480 4.1.4. Aggregation of Media 1482 In the all to all media property mentioned above and provided by ASM, 1483 all simultaneous media transmissions share the available bit-rate. 1484 For participants with limited reception capabilities, this may result 1485 in a situation where even a minimal acceptable media quality cannot 1486 be accomplished. This is the result of multiple media streams 1487 needing to share the available resources. The solution to this 1488 problem is to provide for a Mixer or MCU to aggregate the multiple 1489 streams into a single one. This aggregation can be performed 1490 according to different methods. Mixing or selection are two common 1491 methods. 1493 4.1.5. View of All Session Participants 1495 The RTP protocol includes functionality to identify the session 1496 participants through the use of the SSRC and CSRC fields. In 1497 addition, it is capable of carrying some further identity information 1498 about these participants using the RTCP Source Descriptors (SDES). 1499 To maintain this functionality, it is necessary that RTCP is handled 1500 correctly in domain bridging function. This is specified for 1501 Translators and Mixers. The MCU described in Section 3.8 does not 1502 entirely fulfill this. The one described in Section 3.9 does not 1503 support this at all. 1505 4.1.6. Loop Detection 1507 In complex topologies with multiple interconnected domains, it is 1508 possible to form media loops. RTP and RTCP support detecting such 1509 loops, as long as the SSRC and CSRC identities are correctly set in 1510 forwarded packets. It is likely that loop detection works for the 1511 MCU, described in Section 3.8, at least as long as it forwards the 1512 RTCP between the participants. However, the MCU in Section 3.9 will 1513 definitely break the loop detection mechanism. 1515 4.2. Comparison of Topologies 1517 The table below attempts to summarize the properties of the different 1518 topologies. The legend to the topology abbreviations are: Topo- 1519 Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-Translator 1520 (TTrn), Topo-Media-Translator (including Transport Translator) 1521 (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY), Topo-Video-switch- 1522 MCU (MCUs), and Topo-RTCP-terminating-MCU (MCUt). In the table 1523 below, Y indicates Yes or full support, N indicates No support, (Y) 1524 indicates partial support, and N/A indicates not applicable. 1526 Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt 1527 ------------------------------------------------------------------ 1528 All to All media N Y Y Y (Y) (Y) (Y) (Y) 1529 Interoperability N/A N Y Y Y Y N Y 1530 Per Domain Adaptation N/A N N Y Y Y N Y 1531 Aggregation of media N N N N Y (Y) Y Y 1532 Full Session View Y Y Y Y Y Y (Y) N 1533 Loop Detection Y Y Y Y Y Y (Y) N 1535 Please note that the Media Translator also includes the transport 1536 Translator functionality. 1538 5. Security Considerations 1540 The use of Mixers and Translators has impact on security and the 1541 security functions used. The primary issue is that both Mixers and 1542 Translators modify packets, thus preventing the use of integrity and 1543 source authentication, unless they are trusted devices that take part 1544 in the security context, e.g., the device can send Secure Realtime 1545 Transport Protocol (SRTP) and Secure Realtime Transport Control 1546 Protocol (SRTCP) [RFC3711] packets to session endpoints. If 1547 encryption is employed, the media Translator and Mixer need to be 1548 able to decrypt the media to perform its function. A transport 1549 Translator may be used without access to the encrypted payload in 1550 cases where it translates parts that are not included in the 1551 encryption and integrity protection, for example, IP address and UDP 1552 port numbers in a media stream using SRTP [RFC3711]. However, in 1553 general, the Translator or Mixer needs to be part of the signalling 1554 context and get the necessary security associations (e.g., SRTP 1555 crypto contexts) established with its RTP session participants. 1557 Including the Mixer and Translator in the security context allows the 1558 entity, if subverted or misbehaving, to perform a number of very 1559 serious attacks as it has full access. It can perform all the 1560 attacks possible (see RFC 3550 and any applicable profiles) as if the 1561 media session were not protected at all, while giving the impression 1562 to the session participants that they are protected. 1564 Transport Translators have no interactions with cryptography that 1565 works above the transport layer, such as SRTP, since that sort of 1566 Translator leaves the RTP header and payload unaltered. Media 1567 Translators, on the other hand, have strong interactions with 1568 cryptography, since they alter the RTP payload. A media Translator 1569 in a session that uses cryptographic protection needs to perform 1570 cryptographic processing to both inbound and outbound packets. 1572 A media Translator may need to use different cryptographic keys for 1573 the inbound and outbound processing. For SRTP, different keys are 1574 required, because an RFC 3550 media Translator leaves the SSRC 1575 unchanged during its packet processing, and SRTP key sharing is only 1576 allowed when distinct SSRCs can be used to protect distinct packet 1577 streams. 1579 When the media Translator uses different keys to process inbound and 1580 outbound packets, each session participant needs to be provided with 1581 the appropriate key, depending on whether they are listening to the 1582 Translator or the original source. (Note that there is an 1583 architectural difference between RTP media translation, in which 1584 participants can rely on the RTP Payload Type field of a packet to 1585 determine appropriate processing, and cryptographically protected 1586 media translation, in which participants must use information that is 1587 not carried in the packet.) 1589 When using security mechanisms with Translators and Mixers, it is 1590 possible that the Translator or Mixer could create different security 1591 associations for the different domains they are working in. Doing so 1592 has some implications: 1594 First, it might weaken security if the Mixer/Translator accepts a 1595 weaker algorithm or key in one domain than in another. Therefore, 1596 care should be taken that appropriately strong security parameters 1597 are negotiated in all domains. In many cases, "appropriate" 1598 translates to "similar" strength. If a key management system does 1599 allow the negotiation of security parameters resulting in a different 1600 strength of the security, then this system should notify the 1601 participants in the other domains about this. 1603 Second, the number of crypto contexts (keys and security related 1604 state) needed (for example, in SRTP [RFC3711]) may vary between 1605 Mixers and Translators. A Mixer normally needs to represent only a 1606 single SSRC per domain and therefore needs to create only one 1607 security association (SRTP crypto context) per domain. In contrast, 1608 a Translator needs one security association per participant it 1609 translates towards, in the opposite domain. Considering Figure 9, 1610 the Translator needs two security associations towards the multicast 1611 domain, one for B and one for D. It may be forced to maintain a set 1612 of totally independent security associations between itself and B and 1613 D respectively, so as to avoid two-time pad occurrences. These 1614 contexts must also be capable of handling all the sources present in 1615 the other domains. Hence, using completely independent security 1616 associations (for certain keying mechanisms) may force a Translator 1617 to handle N*DM keys and related state; where N is the total number of 1618 SSRCs used over all domains and DM is the total number of domains. 1620 There exist a number of different mechanisms to provide keys to the 1621 different participants. One example is the choice between group keys 1622 and unique keys per SSRC. The appropriate keying model is impacted 1623 by the topologies one intends to use. The final security properties 1624 are dependent on both the topologies in use and the keying 1625 mechanisms' properties, and need to be considered by the application. 1626 Exactly which mechanisms are used is outside of the scope of this 1627 document. Please review RTP Security Options 1628 [I-D.ietf-avtcore-rtp-security-options] to get a better understanding 1629 of most of the available options. 1631 6. IANA Considerations 1633 This document makes no request of IANA. 1635 Note to RFC Editor: this section may be removed on publication as an 1636 RFC. 1638 7. Acknowledgements 1640 The authors would like to thank Bo Burman, Umesh Chandra, Roni Even, 1641 Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their 1642 help in reviewing this document. 1644 8. References 1646 8.1. Normative References 1648 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1649 Jacobson, "RTP: A Transport Protocol for Real-Time 1650 Applications", STD 64, RFC 3550, July 2003. 1652 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1653 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1654 RFC 3711, March 2004. 1656 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session 1657 Initiation Protocol (SIP) Event Package for Conference 1658 State", RFC 4575, August 2006. 1660 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 1661 "Extended RTP Profile for Real-time Transport Control 1662 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 1663 2006. 1665 8.2. Informative References 1667 [I-D.ietf-avtcore-rtp-security-options] 1668 Westerlund, M. and C. Perkins, "Options for Securing RTP 1669 Sessions", draft-ietf-avtcore-rtp-security-options-02 1670 (work in progress), February 2013. 1672 [I-D.lennox-avtcore-rtp-multi-stream] 1673 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "RTP 1674 Considerations for Endpoints Sending Multiple Media 1675 Streams", draft-lennox-avtcore-rtp-multi-stream-02 (work 1676 in progress), February 2013. 1678 [RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network 1679 Address Translator (Traditional NAT)", RFC 3022, January 1680 2001. 1682 [RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for 1683 IP", RFC 4607, August 2006. 1685 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 1686 "Codec Control Messages in the RTP Audio-Visual Profile 1687 with Feedback (AVPF)", RFC 5104, February 2008. 1689 [RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control 1690 Protocol (RTCP) Extensions for Single-Source Multicast 1691 Sessions with Unicast Feedback", RFC 5760, February 2010. 1693 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 1694 Relays around NAT (TURN): Relay Extensions to Session 1695 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 1697 [RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T., and Z. Vax, 1698 "Unicast-Based Rapid Acquisition of Multicast RTP 1699 Sessions", RFC 6285, June 2011. 1701 [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time 1702 Transport Protocol (RTP) Header Extension for Mixer-to- 1703 Client Audio Level Indication", RFC 6465, December 2011. 1705 Authors' Addresses 1707 Magnus Westerlund 1708 Ericsson 1709 Farogatan 6 1710 SE-164 80 Kista 1711 Sweden 1713 Phone: +46 10 714 82 87 1714 Email: magnus.westerlund@ericsson.com 1716 Stephan Wenger 1717 Vidyo 1718 433 Hackensack Ave 1719 Hackensack, NJ 07601 1720 USA 1722 Email: stewe@stewe.org