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2 CLUE WG R. Even
3 Internet-Draft Huawei Technologies
4 Intended status: Standards Track J. Lennox
5 Expires: April 27, 2017 Vidyo
6 October 24, 2016
8 Mapping RTP streams to CLUE Media Captures
9 draft-ietf-clue-rtp-mapping-09.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 April 27, 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 5. MCC Constituent CaptureID definition . . . . . . . . . . . . 5
57 5.1. RTCP CaptureID SDES Item . . . . . . . . . . . . . . . . 6
58 5.2. RTP Header Extension . . . . . . . . . . . . . . . . . . 6
59 6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 7
60 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
61 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
62 9. Security Considerations . . . . . . . . . . . . . . . . . . . 8
63 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
64 10.1. Normative References . . . . . . . . . . . . . . . . . . 10
65 10.2. Informative References . . . . . . . . . . . . . . . . . 10
66 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 13
68 1. Introduction
70 Telepresence systems can send and receive multiple media streams.
71 The CLUE framework [I-D.ietf-clue-framework] defines Media Captures
72 (MC) as a source of Media, such as from one or more Capture Devices.
73 A Media Capture may also be constructed from other Media streams. A
74 middle box can express conceptual Media Captures that it constructs
75 from Media streams it receives. A Multiple Content Capture (MCC) is
76 a special Media Capture composed of multiple Media Captures.
78 SIP offer answer [RFC3264] uses SDP [RFC4566] to describe the
79 RTP[RFC3550] media streams. Each RTP stream has a unique SSRC within
80 its RTP session. The content of the RTP stream is created by an
81 encoder in the endpoint. This may be an original content from a
82 camera or a content created by an intermediary device like an MCU
83 (Multipoint Control Unit).
85 This document makes recommendations, for the CLUE architecture, about
86 how RTP and RTCP streams should be encoded and transmitted, and how
87 their relation to CLUE Media Captures should be communicated. The
88 proposed solution supports multiple RTP topologies [RFC7667].
90 With regards to the media (audio, video and timed text), systems that
91 support CLUE use RTP for the media, SDP for codec and media transport
92 negotiation (CLUE individual encodings) and the CLUE protocol for
93 Media Capture description and selection. In order to associate the
94 media in the different protocols there are three mapping that need to
95 be specified:
97 1. CLUE individual encodings to SDP
98 2. RTP streams to SDP (this is not a CLUE specific mapping)
100 3. RTP streams to MC to map the received RTP steam to the current MC
101 in the MCC.
103 2. Terminology
105 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
106 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
107 document are to be interpreted as described in RFC2119[RFC2119] and
108 indicate requirement levels for compliant RTP implementations.
110 The definitions from the CLUE framework document
111 [I-D.ietf-clue-framework] section 3 are used by this document as
112 well.
114 3. RTP topologies for CLUE
116 The typical RTP topologies used by CLUE Telepresence systems specify
117 different behaviors for RTP and RTCP distribution. A number of RTP
118 topologies are described in [RFC7667]. For CLUE telepresence, the
119 relevant topologies include Point-to-Point, as well as Media-Mixing
120 mixers, Media- Switching mixers, and Selective Forwarding Middleboxs.
122 In the Point-to-Point topology, one peer communicates directly with a
123 single peer over unicast. There can be one or more RTP sessions,
124 each sent on a separate 5-tuple, and having a separate SSRC space,
125 with each RTP session carrying multiple RTP streams identified by
126 their SSRC. All SSRCs are recognized by the peers based on the
127 information in the RTCP SDES report that includes the CNAME and SSRC
128 of the sent RTP streams. There are different Point-to-Point use
129 cases as specified in CLUE use case [RFC7205]. In some cases, a CLUE
130 session which, at a high-level, is point-to-point may nonetheless
131 have an RTP stream which is best described by one of the mixer
132 topologies. For example, a CLUE endpoint can produce composite or
133 switched captures for use by a receiving system with fewer displays
134 than the sender has cameras. The Media Capture may be described
135 using MCC.
137 For the Media Mixer topology [RFC7667], the peers communicate only
138 with the mixer. The mixer provides mixed or composited media
139 streams, using its own SSRC for the sent streams. If needed by CLUE
140 endpoint, the conference roster information including conference
141 participants, endpoints, media and media-id (SSRC) can be determined
142 using the conference event package [RFC4575] element.
144 In the Media-Switching Mixer topology [RFC7667], the peer to mixer
145 communication is unicast with mixer RTCP feedback. It is
146 conceptually similar to a compositing mixer as described in the
147 previous paragraph, except that rather than compositing or mixing
148 multiple sources, the mixer provides one or more conceptual sources
149 selecting one source at a time from the original sources.
151 In the Selective Forwarding Middlebox (SFM) [RFC7667] topology, the
152 peer to middlebox communication is unicast with RTCP feedback. Every
153 potential sender in the conference has a source which may be
154 "projected" by the SFM into every other RTP session in the
155 conference; thus, even though the SFM establishes a separate RTP
156 session with each endpoint, every original source is maintained with
157 an independent SSRC to every receiver, maintaining separate decoding
158 state and its original RTCP SDES information.
160 4. Mapping CLUE Capture Encodings to RTP streams
162 The different topologies described in Section 3 create different SSRC
163 distribution models and RTP stream multiplexing points.
165 Most video conferencing systems today can separate multiple RTP
166 sources by placing them into RTP sessions using the SDP description,
167 the video conferencing application can also have some knowledge about
168 the usage of each RTP session. For example, video conferencing
169 applications that have main and slides video sources can send each
170 media source in a separate RTP session identified by the content
171 attribute [RFC4796]. Demultipexing at the media receiver is
172 straightforward if the multiplexing point is at the UDP transport
173 level, where each RTP stream uses a separate RTP session. This will
174 also be true for mapping the RTP streams to Media Captures Encodings
175 if each Media Capture Encodings uses a separate RTP session, and the
176 consumer can identify it based on the receiving RTP port. In this
177 case, SDP only needs to label the RTP session with an identifier that
178 can be used to identify the Media Capture in the CLUE description.
179 The SDP label attribute serves as this identifier. In this case, the
180 mapping does not change even if the RTP session is switched using
181 same or different SSRC.
183 The sending of each RTP stream in a separate RTP session is supported
184 by CLUE endpoints but for scaling reasons, CLUE endpoints that
185 support sending of several RTP streams in a single or multiple RTP
186 sessions MUST support [I-D.ietf-mmusic-sdp-bundle-negotiation]. When
187 sending multiple RTP streams in a single RTP session, the mapping of
188 RTP streams to Captures Encodings needs to be considered.
190 MCCs bring another mapping issue, in that an MCC represents multiple
191 Media Captures that can be sent as part of this MCC if configured by
192 the consumer. When receiving an RTP stream which is mapped to the
193 MCC, the consumer needs to know which original MC it is in order to
194 get the MC parameters from the advertisement. If a consumer
195 requested a MCC, the original MC does not have a capture encoding, so
196 it cannot be associated with an m-line using a label as described in
197 CLUE signaling [I-D.ietf-clue-signaling]. This is important, for
198 example, to get correct scaling information for the original MC,
199 which may be different for the various MCs that are contributing to
200 the MCC.
202 5. MCC Constituent CaptureID definition
204 For a MCC which can represent multiple switched MCs there is a need
205 to know which MC is represented in the current RTP stream at any
206 given time. This requires a mapping from the MCC RTP stream to the
207 constituent MC. In order to address this mapping this document
208 defines an RTP header extension and SDES item that includes the
209 captureID of the original MC, allowing the consumer to use the
210 original source MC's attributes like the spatial information.
212 This mapping temporarily associates the SSRC of the MCC stream with
213 the captureID of the single original MC that is currently switched
214 into the MCC. This mapping cannot be used for the composed case
215 where more than one original MC is composed into the MCC
216 simultaneously.
218 If there is only one MC in the MCC then the media provider MUST send
219 the captureID of the current constituent MC in the RTP header and as
220 a RTCP SDES message. When the media provider switches the MC it
221 sends within an MCC, it MUST send the captureID value for the MC just
222 switched into the MCC.
224 If there is more than one MC composed into the MCC then the media
225 provider MUST NOT send any of the MCs' captureIDs using this
226 mechanism. However, if an MCC is sending contributing source (CSRC)
227 information in the RTP header for a composed capture, it MAY send the
228 captureID values in the RTCP SDES packets giving source information
229 for these CSRC values.
231 If the media provider sends the captureID of a single MC switched
232 into an MCC, then later sends a composed stream of multiple MCs in
233 the same MCC, it MUST send the special value "-", a single dash
234 character, as the captureID RTP header and SDES message. The single
235 dash character indicates there is no applicable value for the MCC
236 constituent CaptureID. The media consumer interprets this as meaning
237 any previous captureID value in this RTP header no longer applies.
238 As [I-D.ietf-clue-data-model-schema] defines the captureID syntax as
239 "xs:ID", the single dash character is not a legal captureID value, so
240 there is no possibility of confusing it with an actual captureID.
242 5.1. RTCP CaptureID SDES Item
244 This document specifies a new RTCP SDES item.
246 0 1 2 3
247 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
248 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
249 | CaptureId=XX | length | CaptureID |
250 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
251 | .... |
252 +-+-+-+-+-+-+-+-+
254 This CaptureIDis a variable-length UTF-8 string corresponding either
255 to a CaptureID negotiated in the CLUE protocol, or the single
256 character "-".
258 This SDES message MUST be sent in a compound RTCP packet unless
259 support for Reduced-size RTCP has been negotiated as specified in RFC
260 5506 [RFC5506].
262 5.2. RTP Header Extension
264 The CaptureIDis also carried in an RTP header extension [RFC5285],
265 using the mechanism defined in [RFC7941].
267 Support is negotiated within SDP using the URN "urn:ietf:params:rtp-
268 hdrext:CaptureID".
270 Packets tagged by the sender with the CaptureID then contain a header
271 extension as shown below
273 0 1 2 3
274 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
275 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
276 | ID | len-1 | CaptureID ... |
277 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
278 One-byte header extension form (0xBEDE)
280 0 1 2 3
281 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
282 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
283 | ID | Len | CaptureID |
284 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
285 | .... |
286 +-+-+-+-+-+-+-+-+
287 Two-byte header extension form (0x100x)
289 The CaptureID is sent in a header extension because for switched
290 captures, receivers need to know which original MC corresponds to the
291 media being sent for an MCC, in order to correctly apply geometric
292 adjustments to the received media.
294 As discussed in [RFC7941], there is no need to send the CaptureId
295 header extension with all RTP packets. Senders MAY choose to send it
296 only when a new MC is sent. If such a mode is being used, the header
297 extension SHOULD be sent in the first few RTP packets to reduce the
298 risk of losing it due to packet loss. See [RFC7941] for more
299 discussion of this.
301 6. Examples
303 In this partial advertisement the Media Provider advertises a
304 composed capture VC7 made by a big picture representing the current
305 speaker (VC3) and two picture-in-picture boxes representing the
306 previous speakers (the previous one -VC5- and the oldest one -VC6).
308
310 CS1
311 true
312
313 VC3
314 VC5
315 VC6
316
317 3
318 false
319 big picture of the current speaker
320 pips about previous speakers
321 1
322 it
323 static
324 individual
325
327 In this case the media provider will send capture IDs VC3, VC5 or VC6
328 as an RTP header extension and RTCP SDES message for the RTP stream
329 associated with the MC.
331 7. Acknowledgements
333 The authors would like to thanks Allyn Romanow and Paul Witty for
334 contributing text to this work.
336 8. IANA Considerations
338 This document defines a new extension URI in the RTP SDES Compact
339 Header Extensions subregistry of the Real-Time Transport Protocol
340 (RTP) Parameters registry, according to the following data:
342 Extension URI: urn:ietf:params:rtp-hdrext:CaptureId
344 Description: CLUE CaptureId
346 Contact: roni.even@mail01.huawei.com
348 Reference: RFC XXXX
350 The IANA is requested to register one new RTCP SDES items in the
351 "RTCP SDES Item Types" registry, as follows:
353 Value Abbrev Name Reference
354 TBA CCID CLUE CaptureId [RFCXXXX]
356 9. Security Considerations
358 The security considerations of the RTP specification, the RTP/SAVPF
359 profile, and the various RTP/RTCP extensions and RTP payload formats
360 that form the complete protocol suite described in this memo apply.
361 It is not believed there are any new security considerations
362 resulting from the combination of these various protocol extensions.
364 The Extended Secure RTP Profile for Real-time Transport Control
365 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
366 handling of fundamental issues by offering confidentiality, integrity
367 and partial source authentication. CLUE endpoints MUST support RTP/
368 SAVPF and DTLS-SRTP keying [RFC5764].
370 RTCP packets convey a Canonical Name (CNAME) identifier that is used
371 to associate RTP packet streams that need to be synchronised across
372 related RTP sessions. Inappropriate choice of CNAME values can be a
373 privacy concern, since long-term persistent CNAME identifiers can be
374 used to track users across multiple calls. CLUE endpoint MUST
375 generate short-term persistent RTCP CNAMES, as specified in RFC7022
376 [RFC7022], resulting in untraceable CNAME values that alleviate this
377 risk.
379 Some potential denial of service attacks exist if the RTCP reporting
380 interval is configured to an inappropriate value. This could be done
381 by configuring the RTCP bandwidth fraction to an excessively large or
382 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some
383 similar mechanism, or by choosing an excessively large or small value
384 for the RTP/AVPF minimal receiver report interval (if using SDP, this
385 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as
386 follows:
388 1. the RTCP bandwidth could be configured to make the regular
389 reporting interval so large that effective congestion control
390 cannot be maintained, potentially leading to denial of service
391 due to congestion caused by the media traffic;
393 2. the RTCP interval could be configured to a very small value,
394 causing endpoints to generate high rate RTCP traffic, potentially
395 leading to denial of service due to the non-congestion controlled
396 RTCP traffic; and
398 3. RTCP parameters could be configured differently for each
399 endpoint, with some of the endpoints using a large reporting
400 interval and some using a smaller interval, leading to denial of
401 service due to premature participant timeouts due to mismatched
402 timeout periods which are based on the reporting interval (this
403 is a particular concern if endpoints use a small but non-zero
404 value for the RTP/AVPF minimal receiver report interval (trr-int)
405 [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]).
407 Premature participant timeout can be avoided by using the fixed (non-
408 reduced) minimum interval when calculating the participant timeout
409 ([I-D.ietf-avtcore-rtp-multi-stream]). To address the other
410 concerns, endpoints SHOULD ignore parameters that configure the RTCP
411 reporting interval to be significantly longer than the default five
412 second interval specified in [RFC3550] (unless the media data rate is
413 so low that the longer reporting interval roughly corresponds to 5%
414 of the media data rate), or that configure the RTCP reporting
415 interval small enough that the RTCP bandwidth would exceed the media
416 bandwidth.
418 The guidelines in [RFC6562] apply when using variable bit rate (VBR)
419 audio codecs such as Opus. The use of the encryption of the header
420 extensions are RECOMMENDED, unless there are known reasons, like RTP
421 middleboxes performing voice activity based source selection or third
422 party monitoring that will greatly benefit from the information, and
423 this has been expressed using API or signalling. If further evidence
424 are produced to show that information leakage is significant from
425 audio level indications, then use of encryption needs to be mandated
426 at that time.
428 In multi-party communication scenarios using RTP Middleboxes, a lot
429 of trust is placed on these middleboxes to preserve the sessions
430 security. The middlebox SHOULD maintain the confidentiality,
431 integrity and perform source authentication. The middlebox MAY
432 perform checks that prevents any endpoint participating in a
433 conference to impersonate another. Some additional security
434 considerations regarding multi-party topologies can be found in
435 [RFC7667]
437 10. References
439 10.1. Normative References
441 [I-D.ietf-clue-data-model-schema]
442 Presta, R. and S. Romano, "An XML Schema for the CLUE data
443 model", draft-ietf-clue-data-model-schema-17 (work in
444 progress), August 2016.
446 [I-D.ietf-clue-framework]
447 Duckworth, M., Pepperell, A., and S. Wenger, "Framework
448 for Telepresence Multi-Streams", draft-ietf-clue-
449 framework-25 (work in progress), January 2016.
451 [I-D.ietf-mmusic-sdp-bundle-negotiation]
452 Holmberg, C., Alvestrand, H., and C. Jennings,
453 "Negotiating Media Multiplexing Using the Session
454 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
455 negotiation-34 (work in progress), October 2016.
457 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
458 Requirement Levels", BCP 14, RFC 2119,
459 DOI 10.17487/RFC2119, March 1997,
460 .
462 [RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
463 Header Extension for the RTP Control Protocol (RTCP)
464 Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
465 August 2016, .
467 10.2. Informative References
469 [I-D.ietf-avtcore-rtp-multi-stream]
470 Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
471 "Sending Multiple Media Streams in a Single RTP Session",
472 draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
473 December 2015.
475 [I-D.ietf-clue-signaling]
476 Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
477 Signaling", draft-ietf-clue-signaling-09 (work in
478 progress), March 2016.
480 [I-D.ietf-mmusic-sdp-simulcast]
481 Westerlund, M., Nandakumar, S., and M. Zanaty, "Using
482 Simulcast in SDP and RTP Sessions", draft-ietf-mmusic-sdp-
483 simulcast-05 (work in progress), June 2016.
485 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
486 with Session Description Protocol (SDP)", RFC 3264,
487 DOI 10.17487/RFC3264, June 2002,
488 .
490 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
491 Jacobson, "RTP: A Transport Protocol for Real-Time
492 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
493 July 2003, .
495 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
496 Modifiers for RTP Control Protocol (RTCP) Bandwidth",
497 RFC 3556, DOI 10.17487/RFC3556, July 2003,
498 .
500 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
501 Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
502 July 2006, .
504 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
505 Session Initiation Protocol (SIP) Event Package for
506 Conference State", RFC 4575, DOI 10.17487/RFC4575, August
507 2006, .
509 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
510 "Extended RTP Profile for Real-time Transport Control
511 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
512 DOI 10.17487/RFC4585, July 2006,
513 .
515 [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description
516 Protocol (SDP) Content Attribute", RFC 4796,
517 DOI 10.17487/RFC4796, February 2007,
518 .
520 [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
521 "Codec Control Messages in the RTP Audio-Visual Profile
522 with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
523 February 2008, .
525 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
526 Real-time Transport Control Protocol (RTCP)-Based Feedback
527 (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
528 2008, .
530 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
531 Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
532 2008, .
534 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
535 Real-Time Transport Control Protocol (RTCP): Opportunities
536 and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
537 2009, .
539 [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific
540 Media Attributes in the Session Description Protocol
541 (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009,
542 .
544 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
545 Security (DTLS) Extension to Establish Keys for the Secure
546 Real-time Transport Protocol (SRTP)", RFC 5764,
547 DOI 10.17487/RFC5764, May 2010,
548 .
550 [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image
551 Attributes in the Session Description Protocol (SDP)",
552 RFC 6236, DOI 10.17487/RFC6236, May 2011,
553 .
555 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
556 Variable Bit Rate Audio with Secure RTP", RFC 6562,
557 DOI 10.17487/RFC6562, March 2012,
558 .
560 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
561 "Guidelines for Choosing RTP Control Protocol (RTCP)
562 Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
563 September 2013, .
565 [RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
566 "Use Cases for Telepresence Multistreams", RFC 7205,
567 DOI 10.17487/RFC7205, April 2014,
568 .
570 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
571 DOI 10.17487/RFC7667, November 2015,
572 .
574 Authors' Addresses
576 Roni Even
577 Huawei Technologies
578 Tel Aviv
579 Israel
581 Email: roni.even@mail01.huawei.com
583 Jonathan Lennox
584 Vidyo, Inc.
585 433 Hackensack Avenue
586 Seventh Floor
587 Hackensack, NJ 07601
588 US
590 Email: jonathan@vidyo.com