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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 CLUE WG R. Even 3 Internet-Draft Huawei Technologies 4 Intended status: Standards Track J. Lennox 5 Expires: February 28, 2017 Vidyo 6 August 27, 2016 8 Mapping RTP streams to CLUE Media Captures 9 draft-ietf-clue-rtp-mapping-08.txt 11 Abstract 13 This document describes how the Real Time transport Protocol (RTP) is 14 used in the context of the CLUE protocol. It also describes the 15 mechanisms and recommended practice for mapping RTP media streams 16 defined in SDP to CLUE Media Captures. 18 Status of This Memo 20 This Internet-Draft is submitted in full conformance with the 21 provisions of BCP 78 and BCP 79. 23 Internet-Drafts are working documents of the Internet Engineering 24 Task Force (IETF). Note that other groups may also distribute 25 working documents as Internet-Drafts. The list of current Internet- 26 Drafts is at http://datatracker.ietf.org/drafts/current/. 28 Internet-Drafts are draft documents valid for a maximum of six months 29 and may be updated, replaced, or obsoleted by other documents at any 30 time. It is inappropriate to use Internet-Drafts as reference 31 material or to cite them other than as "work in progress." 33 This Internet-Draft will expire on February 28, 2017. 35 Copyright Notice 37 Copyright (c) 2016 IETF Trust and the persons identified as the 38 document authors. All rights reserved. 40 This document is subject to BCP 78 and the IETF Trust's Legal 41 Provisions Relating to IETF Documents 42 (http://trustee.ietf.org/license-info) in effect on the date of 43 publication of this document. Please review these documents 44 carefully, as they describe your rights and restrictions with respect 45 to this document. Code Components extracted from this document must 46 include Simplified BSD License text as described in Section 4.e of 47 the Trust Legal Provisions and are provided without warranty as 48 described in the Simplified BSD License. 50 Table of Contents 52 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 53 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 54 3. RTP topologies for CLUE . . . . . . . . . . . . . . . . . . . 3 55 4. Mapping CLUE Capture Encodings to RTP streams . . . . . . . . 4 56 4.1. Review of RTP related documents relevant to CLUE work. . 5 57 4.2. Recommendations . . . . . . . . . . . . . . . . . . . . . 6 58 5. CaptureID definition . . . . . . . . . . . . . . . . . . . . 6 59 5.1. RTCP CaptureId SDES Item . . . . . . . . . . . . . . . . 6 60 5.2. RTP Header Extension . . . . . . . . . . . . . . . . . . 6 61 6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 7 62 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 63 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 64 9. Security Considerations . . . . . . . . . . . . . . . . . . . 8 65 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 10 66 10.1. Normative References . . . . . . . . . . . . . . . . . . 10 67 10.2. Informative References . . . . . . . . . . . . . . . . . 10 68 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 12 70 1. Introduction 72 Telepresence systems can send and receive multiple media streams. 73 The CLUE framework [I-D.ietf-clue-framework] defines Media Captures 74 (MC) as a source of Media, such as from one or more Capture Devices. 75 A Media Capture may also be constructed from other Media streams. A 76 middle box can express conceptual Media Captures that it constructs 77 from Media streams it receives. A Multiple Content Capture (MCC) is 78 a special Media Capture composed of multiple Media Captures. 80 SIP offer answer [RFC3264] uses SDP [RFC4566] to describe the 81 RTP[RFC3550] media streams. Each RTP stream has a unique SSRC within 82 its RTP session. The content of the RTP stream is created by an 83 encoder in the endpoint. This may be an original content from a 84 camera or a content created by an intermediary device like an MCU 85 (Multipoint Control Unit). 87 This document makes recommendations, for the CLUE architecture, about 88 how RTP and RTCP streams should be encoded and transmitted, and how 89 their relation to CLUE Media Captures should be communicated. The 90 proposed solution supports multiple RTP topologies [RFC7667]. 92 With regards to the media (audio, video and timed text), systems that 93 support CLUE use RTP for the media, SDP for codec and media transport 94 negotiation (CLUE individual encodings) and the CLUE protocol for 95 Media Capture description and selection. In order to associate the 96 media in the different protocols there are three mapping that need to 97 be specified: 99 1. CLUE individual encodings to SDP 101 2. RTP streams to SDP (this is not a CLUE specific mapping) 103 3. RTP streams to MC to map the received RTP steam to the current MC 104 in the MCC. 106 2. Terminology 108 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 109 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 110 document are to be interpreted as described in RFC2119[RFC2119] and 111 indicate requirement levels for compliant RTP implementations. 113 The definitions from the CLUE framework document 114 [I-D.ietf-clue-framework] section 3 are used by this document as 115 well. 117 3. RTP topologies for CLUE 119 The typical RTP topologies used by CLUE Telepresence systems specify 120 different behaviors for RTP and RTCP distribution. A number of RTP 121 topologies are described in [RFC7667]. For telepresence, the 122 relevant topologies include Point-to-Point, as well as Media-Mixing 123 mixers, Media- Switching mixers, and Selective Forwarding Middleboxs. 125 In the Point-to-Point topology, one peer communicates directly with a 126 single peer over unicast. There can be one or more RTP sessions, 127 each sent on a separate 5-tuple, and having a separate SSRC space, 128 with each RTP session carrying multiple RTP streams identified by 129 their SSRC. All SSRCs are recognized by the peers based on the 130 information in the RTCP SDES report that includes the CNAME and SSRC 131 of the sent RTP streams. There are different Point-to-Point use 132 cases as specified in CLUE use case [RFC7205]. In some cases, a CLUE 133 session which, at a high-level, is point-to-point may nonetheless 134 have an RTP stream which is best described by one of the mixer 135 topologies. For example, a CLUE endpoint can produce composite or 136 switched captures for use by a receiving system with fewer displays 137 than the sender has cameras. The Media Capture may be described 138 using MCC. 140 For the Media Mixer topology [RFC7667], the peers communicate only 141 with the mixer. The mixer provides mixed or composited media 142 streams, using its own SSRC for the sent streams. The conference 143 roster information including conference participants, endpoints, 144 media and media-id (SSRC) can be determined using the conference 145 event package [RFC4575] element. 147 In the Media-Switching Mixer topology [RFC7667], the peer to mixer 148 communication is unicast with mixer RTCP feedback. It is 149 conceptually similar to a compositing mixer as described in the 150 previous paragraph, except that rather than compositing or mixing 151 multiple sources, the mixer provides one or more conceptual sources 152 selecting one source at a time from the original sources. The Mixer 153 creates a conference-wide RTP session by sharing remote SSRC values 154 as CSRCs to all conference participants, and forwarding RTCP reports. 156 In the Selective Forwarding Middlebox (SFM) [RFC7667] topology, the 157 peer to middlebox communication is unicast with RTCP feedback. Every 158 potential sender in the conference has a source which may be 159 "projected" by the SFM into every other RTP session in the 160 conference; thus, even though the SFM establishes a separate RTP 161 session with each endpoint, every original source is maintained with 162 an independent SSRC to every receiver, maintaining separate decoding 163 state and its original RTCP SDES information. 165 4. Mapping CLUE Capture Encodings to RTP streams 167 The different topologies described in Section 3 create different SSRC 168 distribution models and RTP stream multiplexing points. 170 Most video conferencing systems today can separate multiple RTP 171 sources by placing them into RTP sessions using, the SDP description. 172 For example, main and slides video sources are separated into 173 separate RTP sessions based on the content attribute [RFC4796]. This 174 solution is straightforward if the multiplexing point is at the UDP 175 transport level, where each RTP stream uses a separate RTP session. 176 This will also be true for mapping the RTP streams to Media Captures 177 Encodings if each Media Capture Encodings uses a separate RTP 178 session, and the consumer can identify it based on the receiving RTP 179 port. In this case, SDP only needs to label the RTP session with an 180 identifier that can be used to identify the Media Capture in the CLUE 181 description. The SDP label attribute serves as this identifier. In 182 this case, the mapping does not change even if the RTP session is 183 switched using same or different SSRC. 185 Even though Session multiplexing is supported by CLUE, for scaling 186 reasons, CLUE indicates that SSRC multiplexing in a single or 187 multiple sessions using [I-D.ietf-mmusic-sdp-bundle-negotiation]may 188 be used. When SSRC multiplexing is used, the mapping of RTP streams 189 to Captures Encodings needs to be considered. 191 MCCs bring another mapping issue, in that an MCC represents multiple 192 Media Captures that can be sent as part of this MCC if configured by 193 the consumer. When receiving an RTP stream which is mapped to the 194 MCC, the consumer needs to know which original MC it is in order to 195 get the MC parameters from the advertisement. If a consumer 196 requested a MCC, the original MC does not have a capture encoding, so 197 it cannot be associated with an m-line using a label as described in 198 CLUE signaling [I-D.ietf-clue-signaling]. This is important, for 199 example, to get correct scaling information for the original MC, 200 which may be different for the various MCs that are contributing to 201 the MCC. 203 4.1. Review of RTP related documents relevant to CLUE work. 205 This section provides an overview of the RFCs and drafts that can be 206 used in a CLUE system and as a base for a mapping solution. This 207 section is for information only; the normative behavior is given in 208 the cited documents. Tools for SSRC multiplexing support are defined 209 for general conferencing applications; CLUE systems use the same 210 tools. 212 When looking at the available tools based on current work in MMUSIC, 213 AVTcore and AVText Working Groups for supporting SSRC multiplexing 214 the following documents are considered to be relevant. 216 Negotiating Media Multiplexing Using the Session Description Protocol 217 in [I-D.ietf-mmusic-sdp-bundle-negotiation] defines a "bundle" SDP 218 grouping extension that can be used with SDP Offer/Answer mechanism 219 to negotiate the usage of a single 5-tuple for sending and receiving 220 media associated with multiple SDP media descriptions ("m="). 221 [I-D.ietf-mmusic-sdp-bundle-negotiation] specifies how to associate a 222 received RTP stream with the m-line describing it. The assumption in 223 Bundle is that each SDP m-line represents a single media source. 224 [I-D.ietf-mmusic-sdp-bundle-negotiation] specifies using the SDP mid 225 value and sending it as RTCP SDES and an RTP header extension in 226 order to be able to map the RTP stream to the SDP m-line. This is 227 relevant when there are multiple RTP streams with the same payload 228 subtype number. 230 SDP Source attribute [RFC5576] provides mechanisms to describe 231 specific attributes of RTP sources based on their SSRC. 233 Negotiation of generic image attributes in SDP [RFC6236] provides the 234 means to negotiate the image size. The image attribute can be used 235 to offer different image parameters like size. Offering multiple RTP 236 streams with different resolutions is done using separate RTP session 237 for each image option. ([I-D.ietf-mmusic-sdp-bundle-negotiation] 238 provides the support of a single RTP session but each image option 239 will need a separate SDP m-line). 241 The recommended support of the simulcast case is to use 242 [I-D.ietf-mmusic-sdp-simulcast]. 244 4.2. Recommendations 246 The recommendation is that CLUE endpoints using SSRC multiplexing 247 MUST support [I-D.ietf-mmusic-sdp-bundle-negotiation]. 249 5. CaptureID definition 251 For MCC which can represent multiple switched MCs there is a need to 252 know which MC represents the current RTP stream. This requires a 253 mapping from an RTP stream to an MC. In order to address this 254 mapping this document defines an RTP header extension that includes 255 the CaptureID in order to map to the original MC allowing the 256 consumer to use the original source MC attributes like the spatial 257 information. The media provider MUST send for MCC Capture Encoding 258 the captureID of the current MC in the RTP header and as a RTCP SDES 259 message. 261 5.1. RTCP CaptureId SDES Item 263 This document specifies a new RTCP SDES message 265 0 1 2 3 266 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 267 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 268 | CaptureId = XX | length |CaptureId 269 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 270 | .... 272 This CaptureID is the same as in the CLUE MC and is also used in the 273 RTP header extension. 275 This SDES message MAY be sent in a compound RTCP packet based on the 276 application need. 278 5.2. RTP Header Extension 280 The CaptureId is carried within the RTP header extension field, using 281 [RFC5285] two bytes header extension. 283 Support is negotiated within the SDP, i.e. 285 a=extmap:1 urn:ietf:params:rtp-hdrext:CaptureId 287 Packets tagged by the sender with the CaptureId then contain a header 288 extension as shown below 290 0 1 2 3 291 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 292 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 293 | ID | Len-1 | CaptureId 294 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 295 | .... | 296 +-+-+-+-+-+-+-+-+ 298 There is no need to send the CaptureId header extension with all RTP 299 packets. Senders MAY choose to send it only when a new MC is sent. 300 If such a mode is being used, the header extension SHOULD be sent in 301 the first few RTP packets to reduce the risk of losing it due to 302 packet loss. 304 6. Examples 306 In this partial advertisement the Media Provider advertises a 307 composed capture VC7 made by a big picture representing the current 308 speaker (VC3) and two picture-in-picture boxes representing the 309 previous speakers (the previous one -VC5- and the oldest one -VC6). 311 313 CS1 314 true 315 316 VC3 317 VC5 318 VC6 319 320 3 321 false 322 big picture of the current speaker 323 pips about previous speakers 324 1 325 it 326 static 327 individual 328 330 In this case the media provider will send capture IDs VC3, VC5 or VC6 331 as an RTP header extension and RTCP SDES message for the RTP stream 332 associated with the MC. 334 7. Acknowledgements 336 The authors would like to thanks Allyn Romanow and Paul Witty for 337 contributing text to this work. 339 8. IANA Considerations 341 This document defines a new extension URI in the RTP Compact Header 342 Extensions subregistry of the Real-Time Transport Protocol (RTP) 343 Parameters registry, according to the following data: 345 Extension URI: urn:ietf:params:rtp-hdrext:CaptureId 347 Description: CLUE CaptureId 349 Contact: roni.even@mail01.huawei.com 351 Reference: RFC XXXX 353 The IANA is requested to register one new RTCP SDES items in the 354 "RTCP SDES Item Types" registry, as follows: 356 Value Abbrev Name Reference 357 TBA CCID CLUE CaptureId [RFCXXXX] 359 9. Security Considerations 361 The security considerations of the RTP specification, the RTP/SAVPF 362 profile, and the various RTP/RTCP extensions and RTP payload formats 363 that form the complete protocol suite described in this memo apply. 364 It is not believed there are any new security considerations 365 resulting from the combination of these various protocol extensions. 367 The Extended Secure RTP Profile for Real-time Transport Control 368 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides 369 handling of fundamental issues by offering confidentiality, integrity 370 and partial source authentication. A mandatory to support media 371 security solution is created by combining this secured RTP profile 372 and DTLS-SRTP keying [RFC5764] 374 RTCP packets convey a Canonical Name (CNAME) identifier that is used 375 to associate RTP packet streams that need to be synchronised across 376 related RTP sessions. Inappropriate choice of CNAME values can be a 377 privacy concern, since long-term persistent CNAME identifiers can be 378 used to track users across multiple calls. This memo mandates 379 generation of short-term persistent RTCP CNAMES, as specified in 380 RFC7022 [RFC7022], resulting in untraceable CNAME values that 381 alleviate this risk. 383 Some potential denial of service attacks exist if the RTCP reporting 384 interval is configured to an inappropriate value. This could be done 385 by configuring the RTCP bandwidth fraction to an excessively large or 386 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some 387 similar mechanism, or by choosing an excessively large or small value 388 for the RTP/AVPF minimal receiver report interval (if using SDP, this 389 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as 390 follows: 392 1. the RTCP bandwidth could be configured to make the regular 393 reporting interval so large that effective congestion control 394 cannot be maintained, potentially leading to denial of service 395 due to congestion caused by the media traffic; 397 2. the RTCP interval could be configured to a very small value, 398 causing endpoints to generate high rate RTCP traffic, potentially 399 leading to denial of service due to the non-congestion controlled 400 RTCP traffic; and 402 3. RTCP parameters could be configured differently for each 403 endpoint, with some of the endpoints using a large reporting 404 interval and some using a smaller interval, leading to denial of 405 service due to premature participant timeouts due to mismatched 406 timeout periods which are based on the reporting interval (this 407 is a particular concern if endpoints use a small but non-zero 408 value for the RTP/AVPF minimal receiver report interval (trr-int) 409 [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]). 411 Premature participant timeout can be avoided by using the fixed (non- 412 reduced) minimum interval when calculating the participant timeout 413 ([I-D.ietf-avtcore-rtp-multi-stream]). To address the other 414 concerns, endpoints SHOULD ignore parameters that configure the RTCP 415 reporting interval to be significantly longer than the default five 416 second interval specified in [RFC3550] (unless the media data rate is 417 so low that the longer reporting interval roughly corresponds to 5% 418 of the media data rate), or that configure the RTCP reporting 419 interval small enough that the RTCP bandwidth would exceed the media 420 bandwidth. 422 The guidelines in [RFC6562] apply when using variable bit rate (VBR) 423 audio codecs such as Opus. The use of the encryption of the header 424 extensions are RECOMMENDED, unless there are known reasons, like RTP 425 middleboxes performing voice activity based source selection or third 426 party monitoring that will greatly benefit from the information, and 427 this has been expressed using API or signalling. If further evidence 428 are produced to show that information leakage is significant from 429 audio level indications, then use of encryption needs to be mandated 430 at that time. 432 In multi-party communication scenarios using RTP Middleboxes, a lot 433 of trust is placed on these middleboxes to preserve the sessions 434 security. The middlebox needs to maintain the confidentiality, 435 integrity and perform source authentication. The middlebox can 436 perform checks that prevents any endpoint participating in a 437 conference to impersonate another. Some additional security 438 considerations regarding multi-party topologies can be found in 439 [RFC7667] 441 10. References 443 10.1. Normative References 445 [I-D.ietf-clue-framework] 446 Duckworth, M., Pepperell, A., and S. Wenger, "Framework 447 for Telepresence Multi-Streams", draft-ietf-clue- 448 framework-25 (work in progress), January 2016. 450 [I-D.ietf-mmusic-sdp-bundle-negotiation] 451 Holmberg, C., Alvestrand, H., and C. Jennings, 452 "Negotiating Media Multiplexing Using the Session 453 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- 454 negotiation-32 (work in progress), August 2016. 456 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 457 Requirement Levels", BCP 14, RFC 2119, 458 DOI 10.17487/RFC2119, March 1997, 459 . 461 10.2. Informative References 463 [I-D.ietf-avtcore-rtp-multi-stream] 464 Lennox, J., Westerlund, M., Wu, W., and C. Perkins, 465 "Sending Multiple Media Streams in a Single RTP Session", 466 draft-ietf-avtcore-rtp-multi-stream-11 (work in progress), 467 December 2015. 469 [I-D.ietf-clue-signaling] 470 Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE 471 Signaling", draft-ietf-clue-signaling-09 (work in 472 progress), March 2016. 474 [I-D.ietf-mmusic-sdp-simulcast] 475 Westerlund, M., Nandakumar, S., and M. Zanaty, "Using 476 Simulcast in SDP and RTP Sessions", draft-ietf-mmusic-sdp- 477 simulcast-05 (work in progress), June 2016. 479 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model 480 with Session Description Protocol (SDP)", RFC 3264, 481 DOI 10.17487/RFC3264, June 2002, 482 . 484 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 485 Jacobson, "RTP: A Transport Protocol for Real-Time 486 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 487 July 2003, . 489 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth 490 Modifiers for RTP Control Protocol (RTCP) Bandwidth", 491 RFC 3556, DOI 10.17487/RFC3556, July 2003, 492 . 494 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 495 Description Protocol", RFC 4566, DOI 10.17487/RFC4566, 496 July 2006, . 498 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A 499 Session Initiation Protocol (SIP) Event Package for 500 Conference State", RFC 4575, DOI 10.17487/RFC4575, August 501 2006, . 503 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, 504 "Extended RTP Profile for Real-time Transport Control 505 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, 506 DOI 10.17487/RFC4585, July 2006, 507 . 509 [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description 510 Protocol (SDP) Content Attribute", RFC 4796, 511 DOI 10.17487/RFC4796, February 2007, 512 . 514 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, 515 "Codec Control Messages in the RTP Audio-Visual Profile 516 with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, 517 February 2008, . 519 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for 520 Real-time Transport Control Protocol (RTCP)-Based Feedback 521 (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 522 2008, . 524 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP 525 Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 526 2008, . 528 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific 529 Media Attributes in the Session Description Protocol 530 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, 531 . 533 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 534 Security (DTLS) Extension to Establish Keys for the Secure 535 Real-time Transport Protocol (SRTP)", RFC 5764, 536 DOI 10.17487/RFC5764, May 2010, 537 . 539 [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image 540 Attributes in the Session Description Protocol (SDP)", 541 RFC 6236, DOI 10.17487/RFC6236, May 2011, 542 . 544 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 545 Variable Bit Rate Audio with Secure RTP", RFC 6562, 546 DOI 10.17487/RFC6562, March 2012, 547 . 549 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 550 "Guidelines for Choosing RTP Control Protocol (RTCP) 551 Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, 552 September 2013, . 554 [RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed., 555 "Use Cases for Telepresence Multistreams", RFC 7205, 556 DOI 10.17487/RFC7205, April 2014, 557 . 559 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, 560 DOI 10.17487/RFC7667, November 2015, 561 . 563 Authors' Addresses 565 Roni Even 566 Huawei Technologies 567 Tel Aviv 568 Israel 570 Email: roni.even@mail01.huawei.com 571 Jonathan Lennox 572 Vidyo, Inc. 573 433 Hackensack Avenue 574 Seventh Floor 575 Hackensack, NJ 07601 576 US 578 Email: jonathan@vidyo.com