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2 CLUE WG R. Even
3 Internet-Draft Huawei Technologies
4 Intended status: Standards Track J. Lennox
5 Expires: August 17, 2017 Vidyo
6 February 13, 2017
8 Mapping RTP streams to CLUE Media Captures
9 draft-ietf-clue-rtp-mapping-13.txt
11 Abstract
13 This document describes how the Real Time transport Protocol (RTP) is
14 used in the context of the CLUE protocol (ControLling mUltiple
15 streams for tElepresence). It also describes the mechanisms and
16 recommended practice for mapping RTP media streams defined in Session
17 Description Protocol (SDP) to CLUE Media Captures and defines a new
18 RTP header extension (CaptureId).
20 Status of This Memo
22 This Internet-Draft is submitted in full conformance with the
23 provisions of BCP 78 and BCP 79.
25 Internet-Drafts are working documents of the Internet Engineering
26 Task Force (IETF). Note that other groups may also distribute
27 working documents as Internet-Drafts. The list of current Internet-
28 Drafts is at http://datatracker.ietf.org/drafts/current/.
30 Internet-Drafts are draft documents valid for a maximum of six months
31 and may be updated, replaced, or obsoleted by other documents at any
32 time. It is inappropriate to use Internet-Drafts as reference
33 material or to cite them other than as "work in progress."
35 This Internet-Draft will expire on August 17, 2017.
37 Copyright Notice
39 Copyright (c) 2017 IETF Trust and the persons identified as the
40 document authors. All rights reserved.
42 This document is subject to BCP 78 and the IETF Trust's Legal
43 Provisions Relating to IETF Documents
44 (http://trustee.ietf.org/license-info) in effect on the date of
45 publication of this document. Please review these documents
46 carefully, as they describe your rights and restrictions with respect
47 to this document. Code Components extracted from this document must
48 include Simplified BSD License text as described in Section 4.e of
49 the Trust Legal Provisions and are provided without warranty as
50 described in the Simplified BSD License.
52 Table of Contents
54 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
55 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
56 3. RTP topologies for CLUE . . . . . . . . . . . . . . . . . . . 3
57 4. Mapping CLUE Capture Encodings to RTP streams . . . . . . . . 4
58 5. MCC Constituent CaptureID definition . . . . . . . . . . . . 5
59 5.1. RTCP CaptureID SDES Item . . . . . . . . . . . . . . . . 5
60 5.2. RTP Header Extension . . . . . . . . . . . . . . . . . . 6
61 6. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 6
62 7. Communication Security . . . . . . . . . . . . . . . . . . . 7
63 8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 8
64 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
65 10. Security Considerations . . . . . . . . . . . . . . . . . . . 8
66 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 10
67 11.1. Normative References . . . . . . . . . . . . . . . . . . 10
68 11.2. Informative References . . . . . . . . . . . . . . . . . 11
69 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 13
71 1. Introduction
73 Telepresence systems can send and receive multiple media streams.
74 The CLUE framework [I-D.ietf-clue-framework] defines Media Captures
75 (MC) as a source of Media, from one or more Capture Devices. A Media
76 Capture may also be constructed from other Media streams. A middle
77 box can express conceptual Media Captures that it constructs from
78 Media streams it receives. A Multiple Content Capture (MCC) is a
79 special Media Capture composed of multiple Media Captures.
81 SIP Offer/Answer [RFC3264] uses SDP [RFC4566] to describe the
82 RTP[RFC3550] media streams. Each RTP stream has a unique
83 Synchronization Source (SSRC) within its RTP session. The content of
84 the RTP stream is created by an encoder in the endpoint. This may be
85 an original content from a camera or a content created by an
86 intermediary device like an MCU (Multipoint Control Unit).
88 This document makes recommendations for the CLUE architecture about
89 how RTP and RTCP streams should be encoded and transmitted, and how
90 their relation to CLUE Media Captures should be communicated. The
91 proposed solution supports multiple RTP topologies [RFC7667].
93 With regards to the media (audio, video and timed text), systems that
94 support CLUE use RTP for the media, SDP for codec and media transport
95 negotiation (CLUE individual encodings) and the CLUE protocol for
96 Media Capture description and selection. In order to associate the
97 media in the different protocols there are three mapping that need to
98 be specified:
100 1. CLUE individual encodings to SDP
102 2. RTP streams to SDP (this is not a CLUE specific mapping)
104 3. RTP streams to MC to map the received RTP steam to the current MC
105 in the MCC.
107 2. Terminology
109 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
110 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
111 document are to be interpreted as described in RFC2119[RFC2119] and
112 indicate requirement levels for RTP processing in compliant CLUE
113 implementations.
115 The definitions from the CLUE framework document
116 [I-D.ietf-clue-framework] section 3 are used by this document as
117 well.
119 3. RTP topologies for CLUE
121 The typical RTP topologies used by CLUE Telepresence systems specify
122 different behaviors for RTP and RTCP distribution. A number of RTP
123 topologies are described in [RFC7667]. For CLUE telepresence, the
124 relevant topologies include Point-to-Point, as well as Media-Mixing
125 mixers, Media- Switching mixers, and Selective Forwarding Middleboxs.
127 In the Point-to-Point topology, one peer communicates directly with a
128 single peer over unicast. There can be one or more RTP sessions,
129 each sent on a separate 5-tuple, and having a separate SSRC space,
130 with each RTP session carrying multiple RTP streams identified by
131 their SSRC. All SSRCs are recognized by the peers based on the
132 information in the RTCP Source description (SDES) report that
133 includes the CNAME and SSRC of the sent RTP streams. There are
134 different Point-to-Point use cases as specified in CLUE use case
135 [RFC7205]. In some cases, a CLUE session which, at a high-level, is
136 point-to-point may nonetheless have an RTP stream which is best
137 described by one of the mixer topologies. For example, a CLUE
138 endpoint can produce composite or switched captures for use by a
139 receiving system with fewer displays than the sender has cameras.
140 The Media Capture may be described using an MCC.
142 For the Media Mixer topology [RFC7667], the peers communicate only
143 with the mixer. The mixer provides mixed or composited media
144 streams, using its own SSRC for the sent streams. If needed by CLUE
145 endpoint, the conference roster information including conference
146 participants, endpoints, media and media-id (SSRC) can be determined
147 using the conference event package [RFC4575] element.
149 Media-switching mixers and Selective Forwarding Middleboxes behave as
150 described in [RFC7667]
152 4. Mapping CLUE Capture Encodings to RTP streams
154 The different topologies described in Section 3 create different SSRC
155 distribution models and RTP stream multiplexing points.
157 Most video conferencing systems today can separate multiple RTP
158 sources by placing them into RTP sessions using the SDP description;
159 the video conferencing application can also have some knowledge about
160 the purpose of each RTP session. For example, video conferencing
161 applications that have a primary video source and a slides video
162 source can send each media source in a separate RTP session with a
163 content attribute [RFC4796] enabling different application behavior
164 for each received RTP media source. Demultiplexing is
165 straightforward because each media capture is sent as a single RTP
166 stream, with each RTP stream being sent in a separate RTP session, on
167 a distinct UDP 5-tuple. This will also be true for mapping the RTP
168 streams to Media Captures Encodings if each Media Capture Encodings
169 uses a separate RTP session, and the consumer can identify it based
170 on the receiving RTP port. In this case, SDP only needs to label the
171 RTP session with an identifier that can be used to identify the Media
172 Capture in the CLUE description. The SDP label attribute serves as
173 this identifier.
175 Each Capture Encoding MUST be sent as a separate RTP stream. CLUE
176 endpoints MUST support sending each such RTP stream in a separate RTP
177 session signalled by an SDP m= line. They MAY also support sending
178 some or all of the RTP streams in a single RTP session, using the
179 mechanism described in [I-D.ietf-mmusic-sdp-bundle-negotiation] to
180 relate RTP streams to SDP m= lines.
182 MCCs bring another mapping issue, in that an MCC represents multiple
183 Media Captures that can be sent as part of this MCC if configured by
184 the consumer. When receiving an RTP stream which is mapped to the
185 MCC, the consumer needs to know which original MC it is in order to
186 get the MC parameters from the advertisement. If a consumer
187 requested a MCC, the original MC does not have a capture encoding, so
188 it cannot be associated with an m-line using a label as described in
189 CLUE signaling [I-D.ietf-clue-signaling]. This is important, for
190 example, to get correct scaling information for the original MC,
191 which may be different for the various MCs that are contributing to
192 the MCC.
194 5. MCC Constituent CaptureID definition
196 For a MCC which can represent multiple switched MCs there is a need
197 to know which MC is represented in the current RTP stream at any
198 given time. This requires a mapping from the SSRC of the RTP stream
199 conveying a particular MCC to the constituent MC. In order to
200 address this mapping this document defines an RTP header extension
201 and SDES item that includes the captureID of the original MC,
202 allowing the consumer to use the original source MC's attributes like
203 the spatial information.
205 This mapping temporarily associates the SSRC of the RTP stream
206 conveying a particular MCC with the captureID of the single original
207 MC that is currently switched into the MCC. This mapping cannot be
208 used for the composed case where more than one original MC is
209 composed into the MCC simultaneously.
211 If there is only one MC in the MCC then the media provider MUST send
212 the captureID of the current constituent MC in the RTP Header
213 Extension and as a RTCP CaptureID SDES item. When the media provider
214 switches the MC it sends within an MCC, it MUST send the captureID
215 value for the MC just switched into the MCC in an RTP Header
216 Extension and as a RTCP CaptureID SDES item as specified in [RFC7941]
218 If there is more than one MC composed into the MCC then the media
219 provider MUST NOT send any of the MCs' captureIDs using this
220 mechanism. However, if an MCC is sending contributing source (CSRC)
221 information in the RTP header for a composed capture, it MAY send the
222 captureID values in the RTCP SDES packets giving source information
223 for the SSRC values sent as contributing sources (CSRCs).
225 If the media provider sends the captureID of a single MC switched
226 into an MCC, then later sends one composed stream of multiple MCs in
227 the same MCC, it MUST send the special value "-", a single dash
228 character, as the captureID RTP Header Extension and RTCP CaptureID
229 SDES item. The single dash character indicates there is no
230 applicable value for the MCC constituent CaptureID. The media
231 consumer interprets this as meaning that any previous CaptureID value
232 associated with this SSRC no longer applies. As
233 [I-D.ietf-clue-data-model-schema] defines the captureID syntax as
234 "xs:ID", the single dash character is not a legal captureID value, so
235 there is no possibility of confusing it with an actual captureID.
237 5.1. RTCP CaptureID SDES Item
239 This document specifies a new RTCP SDES item.
241 0 1 2 3
242 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
243 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
244 | CaptId=TBA | length | CaptureID |
245 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
246 | .... |
247 +-+-+-+-+-+-+-+-+
249 Note to the RFC Editor: Please replace TBA with the value assigned by
250 IANA.
252 This CaptureID is a variable-length UTF-8 string corresponding either
253 to a CaptureID negotiated in the CLUE protocol, or the single
254 character "-".
256 This SDES item MUST be sent in an SDES packet within a compound RTCP
257 packet unless support for Reduced-size RTCP has been negotiated as
258 specified in RFC 5506 [RFC5506], in which case it can be sent as an
259 SDES packet in a non-compound RTCP packet.
261 5.2. RTP Header Extension
263 The CaptureID is also carried in an RTP header extension [RFC5285],
264 using the mechanism defined in [RFC7941].
266 Support is negotiated within SDP using the URN "urn:ietf:params:rtp-
267 hdrext:sdes:CaptureID".
269 The CaptureID is sent in a RTP Header Extension because for switched
270 captures, receivers need to know which original MC corresponds to the
271 media being sent for an MCC, in order to correctly apply geometric
272 adjustments to the received media.
274 As discussed in [RFC7941], there is no need to send the CaptId Header
275 Extension with all RTP packets. Senders MAY choose to send it only
276 when a new MC is sent. If such a mode is being used, the header
277 extension SHOULD be sent in the first few RTP packets to reduce the
278 risk of losing it due to packet loss. See [RFC7941] for more
279 discussion of this.
281 6. Examples
283 In this partial advertisement the Media Provider advertises a
284 composed capture VC7 made of a big picture representing the current
285 speaker (VC3) and two picture-in-picture boxes representing the
286 previous speakers (the previous one -VC5- and the oldest one -VC6).
288
290 CS1
291 true
292
293 VC3
294 VC5
295 VC6
296
297 3
298 false
299 big picture of the current speaker
300 pips about previous speakers
301 1
302 it
303 static
304 individual
305
307 In this case the media provider will send capture IDs VC3, VC5 or VC6
308 as an RTP header extension and RTCP SDES message for the RTP stream
309 associated with the MC.
311 Note that this is part of the full advertisement message example from
312 CLUE data model[I-D.ietf-clue-data-model-schema] example and is not a
313 valid xml document.
315 7. Communication Security
317 CLUE endpoints MUST support RTP/SAVPF profile and SRTP [RFC3711].
318 CLUE endpoints MUST support DTLS [RFC6347] and DTLS-SRTP [RFC5763]
319 [RFC5764] for SRTP keying.
321 All media channels SHOULD be secure via SRTP and the RTP/SAVPF
322 profile unless the RTP media and its associated RTCP are secure by
323 other means (see [RFC7201] [RFC7202]).
325 All CLUE implementations MUST implement DTLS 1.0, with the cipher
326 suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA with the the P-256 curve
327 [FIPS186]. The DTLS-SRTP protection profile
328 SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
330 Implementations SHOULD implement DTLS 1.2 with the
331 TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
332 Implementations MUST favor cipher suites which support PFS over non-
333 PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites.
335 NULL Protection profiles MUST NOT be used for RTP or RTCP.
337 CLUE endpoint MUST generate short-term persistent RTCP CNAMES, as
338 specified in [RFC7022], resulting in untraceable CNAME values.
340 8. Acknowledgments
342 The authors would like to thanks Allyn Romanow and Paul Witty for
343 contributing text to this work. Magnus Westerlund helped drafting
344 the security section.
346 9. IANA Considerations
348 This document defines a new extension URI in the RTP SDES Compact
349 Header Extensions subregistry of the Real-Time Transport Protocol
350 (RTP) Parameters registry, according to the following data:
352 Extension URI: urn:ietf:params:rtp-hdrext:sdes:CaptId
354 Description: CLUE CaptId
356 Contact: ron.even.tlv@gmail.com
358 Reference: RFC XXXX
360 The IANA is requested to register one new RTCP SDES items in the
361 "RTCP SDES Item Types" registry, as follows:
363 Value Abbrev Name Reference
364 TBA CCID CLUE CaptId [RFCXXXX]
366 Note to the RFC Editor: Please replace RFCXXXX with this RFC number.
368 10. Security Considerations
370 The security considerations of the RTP specification, the RTP/SAVPF
371 profile, and the various RTP/RTCP extensions and RTP payload formats
372 that form the complete protocol suite described in this memo apply.
373 It is not believed there are any new security considerations
374 resulting from the combination of these various protocol extensions.
376 The Extended Secure RTP Profile for Real-time Transport Control
377 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
378 handling of fundamental issues by offering confidentiality, integrity
379 and partial source authentication. A mandatory to implement and use
380 media security solution is created by combining this secured RTP
381 profile and DTLS-SRTP keying [RFC5764] as defined in the
382 communication security section of this memo Section 7
383 RTCP packets convey a Canonical Name (CNAME) identifier that is used
384 to associate RTP packet streams that need to be synchronised across
385 related RTP sessions. Inappropriate choice of CNAME values can be a
386 privacy concern, since long-term persistent CNAME identifiers can be
387 used to track users across multiple calls. The communication
388 security section of this memo Section 7 mandates generation of short-
389 term persistent RTCP CNAMES, as specified in [RFC7022], resulting in
390 untraceable CNAME values that alleviate this risk.
392 Some potential denial of service attacks exist if the RTCP reporting
393 interval is configured to an inappropriate value. This could be done
394 by configuring the RTCP bandwidth fraction to an excessively large or
395 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some
396 similar mechanism, or by choosing an excessively large or small value
397 for the RTP/AVPF minimal receiver report interval (if using SDP, this
398 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as
399 follows:
401 1. the RTCP bandwidth could be configured to make the regular
402 reporting interval so large that effective congestion control
403 cannot be maintained, potentially leading to denial of service
404 due to congestion caused by the media traffic;
406 2. the RTCP interval could be configured to a very small value,
407 causing endpoints to generate high rate RTCP traffic, potentially
408 leading to denial of service due to the non-congestion controlled
409 RTCP traffic; and
411 3. RTCP parameters could be configured differently for each
412 endpoint, with some of the endpoints using a large reporting
413 interval and some using a smaller interval, leading to denial of
414 service due to premature participant timeouts due to mismatched
415 timeout periods which are based on the reporting interval (this
416 is a particular concern if endpoints use a small but non-zero
417 value for the RTP/AVPF minimal receiver report interval (trr-int)
418 [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]).
420 Premature participant timeout can be avoided by using the fixed (non-
421 reduced) minimum interval when calculating the participant timeout
422 ([I-D.ietf-avtcore-rtp-multi-stream]). To address the other
423 concerns, endpoints SHOULD ignore parameters that configure the RTCP
424 reporting interval to be significantly longer than the default five
425 second interval specified in [RFC3550] (unless the media data rate is
426 so low that the longer reporting interval roughly corresponds to 5%
427 of the media data rate), or that configure the RTCP reporting
428 interval small enough that the RTCP bandwidth would exceed the media
429 bandwidth.
431 The guidelines in [RFC6562] apply when using variable bit rate (VBR)
432 audio codecs such as Opus.
434 The use of the encryption of the header extensions are RECOMMENDED,
435 unless there are known reasons, like RTP middleboxes performing voice
436 activity based source selection or third party monitoring that will
437 greatly benefit from the information, and this has been expressed
438 using API or signalling. If further evidence are produced to show
439 that information leakage is significant from audio level indications,
440 then use of encryption needs to be mandated at that time.
442 In multi-party communication scenarios using RTP Middleboxes; this
443 middleboxes are REQUIRED, by this protocol, to not weaken the
444 sessions' security. The middlebox SHOULD maintain the
445 confidentiality, integrity and perform source authentication. The
446 middlebox MAY perform checks that prevents any endpoint participating
447 in a conference to impersonate another. Some additional security
448 considerations regarding multi-party topologies can be found in
449 [RFC7667]
451 The CaptureID is created as part of the CLUE protocol. The CaptId
452 SDES item is used to convey the same CaptureID value in the SDES
453 item. When sending the SDES item the security consideration
454 specified in the security section of [RFC7941] and in the
455 communication security section of this memo Section 7 are applicable.
456 Note that since the CaptureID is carried also in CLUE protocol
457 messages it is RECOMMENDED that this SDES item use at least similar
458 protection profiles as the CLUE protocol messages carried in the CLUE
459 data channel. .
461 11. References
463 11.1. Normative References
465 [I-D.ietf-clue-data-model-schema]
466 Presta, R. and S. Romano, "An XML Schema for the CLUE data
467 model", draft-ietf-clue-data-model-schema-17 (work in
468 progress), August 2016.
470 [I-D.ietf-clue-framework]
471 Duckworth, M., Pepperell, A., and S. Wenger, "Framework
472 for Telepresence Multi-Streams", draft-ietf-clue-
473 framework-25 (work in progress), January 2016.
475 [I-D.ietf-mmusic-sdp-bundle-negotiation]
476 Holmberg, C., Alvestrand, H., and C. Jennings,
477 "Negotiating Media Multiplexing Using the Session
478 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
479 negotiation-36 (work in progress), October 2016.
481 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
482 Requirement Levels", BCP 14, RFC 2119,
483 DOI 10.17487/RFC2119, March 1997,
484 .
486 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
487 Norrman, "The Secure Real-time Transport Protocol (SRTP)",
488 RFC 3711, DOI 10.17487/RFC3711, March 2004,
489 .
491 [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
492 for Establishing a Secure Real-time Transport Protocol
493 (SRTP) Security Context Using Datagram Transport Layer
494 Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
495 2010, .
497 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
498 Security (DTLS) Extension to Establish Keys for the Secure
499 Real-time Transport Protocol (SRTP)", RFC 5764,
500 DOI 10.17487/RFC5764, May 2010,
501 .
503 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
504 Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
505 January 2012, .
507 [RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
508 Header Extension for the RTP Control Protocol (RTCP)
509 Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
510 August 2016, .
512 11.2. Informative References
514 [FIPS186] National Institute of Standards and Technology, "Digital
515 Signature Standard", FIPS PUB 186-4, July 2013.
517 [I-D.ietf-avtcore-rtp-multi-stream]
518 Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
519 "Sending Multiple Media Streams in a Single RTP Session",
520 draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
521 December 2015.
523 [I-D.ietf-clue-signaling]
524 Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
525 Signaling", draft-ietf-clue-signaling-10 (work in
526 progress), January 2017.
528 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
529 with Session Description Protocol (SDP)", RFC 3264,
530 DOI 10.17487/RFC3264, June 2002,
531 .
533 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
534 Jacobson, "RTP: A Transport Protocol for Real-Time
535 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
536 July 2003, .
538 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
539 Modifiers for RTP Control Protocol (RTCP) Bandwidth",
540 RFC 3556, DOI 10.17487/RFC3556, July 2003,
541 .
543 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
544 Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
545 July 2006, .
547 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
548 Session Initiation Protocol (SIP) Event Package for
549 Conference State", RFC 4575, DOI 10.17487/RFC4575, August
550 2006, .
552 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
553 "Extended RTP Profile for Real-time Transport Control
554 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
555 DOI 10.17487/RFC4585, July 2006,
556 .
558 [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description
559 Protocol (SDP) Content Attribute", RFC 4796,
560 DOI 10.17487/RFC4796, February 2007,
561 .
563 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
564 Real-time Transport Control Protocol (RTCP)-Based Feedback
565 (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
566 2008, .
568 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
569 Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
570 2008, .
572 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
573 Real-Time Transport Control Protocol (RTCP): Opportunities
574 and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
575 2009, .
577 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
578 Variable Bit Rate Audio with Secure RTP", RFC 6562,
579 DOI 10.17487/RFC6562, March 2012,
580 .
582 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
583 "Guidelines for Choosing RTP Control Protocol (RTCP)
584 Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
585 September 2013, .
587 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
588 Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
589 .
591 [RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP
592 Framework: Why RTP Does Not Mandate a Single Media
593 Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
594 2014, .
596 [RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
597 "Use Cases for Telepresence Multistreams", RFC 7205,
598 DOI 10.17487/RFC7205, April 2014,
599 .
601 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
602 DOI 10.17487/RFC7667, November 2015,
603 .
605 Authors' Addresses
607 Roni Even
608 Huawei Technologies
609 Tel Aviv
610 Israel
612 Email: roni.even@huawei.com
613 Jonathan Lennox
614 Vidyo, Inc.
615 433 Hackensack Avenue
616 Seventh Floor
617 Hackensack, NJ 07601
618 US
620 Email: jonathan@vidyo.com