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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