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2 CLUE WG R. Even
3 Internet-Draft Huawei Technologies
4 Intended status: Standards Track J. Lennox
5 Expires: August 31, 2017 Vidyo
6 February 27, 2017
8 Mapping RTP streams to CLUE Media Captures
9 draft-ietf-clue-rtp-mapping-14.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 31, 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.Encrypted SRTP
329 Header extensions [RFC6904] MUST be supported.
331 Implementations SHOULD implement DTLS 1.2 with the
332 TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
333 Implementations MUST favor cipher suites which support PFS over non-
334 PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites.
336 NULL Protection profiles MUST NOT be used for RTP or RTCP.
338 CLUE endpoint MUST generate short-term persistent RTCP CNAMES, as
339 specified in [RFC7022], and thus can't be used for long term tracking
340 of the users.
342 8. Acknowledgments
344 The authors would like to thanks Allyn Romanow and Paul Witty for
345 contributing text to this work. Magnus Westerlund helped drafting
346 the security section.
348 9. IANA Considerations
350 This document defines a new extension URI in the RTP SDES Compact
351 Header Extensions subregistry of the Real-Time Transport Protocol
352 (RTP) Parameters registry, according to the following data:
354 Extension URI: urn:ietf:params:rtp-hdrext:sdes:CaptId
356 Description: CLUE CaptId
358 Contact: ron.even.tlv@gmail.com
360 Reference: RFC XXXX
362 The IANA is requested to register one new RTCP SDES items in the
363 "RTCP SDES Item Types" registry, as follows:
365 Value Abbrev Name Reference
366 TBA CCID CLUE CaptId [RFCXXXX]
368 Note to the RFC Editor: Please replace RFCXXXX with this RFC number.
370 10. Security Considerations
372 The security considerations of the RTP specification, the RTP/SAVPF
373 profile, and the various RTP/RTCP extensions and RTP payload formats
374 that form the complete protocol suite described in this memo apply.
375 It is not believed there are any new security considerations
376 resulting from the combination of these various protocol extensions.
378 The Extended Secure RTP Profile for Real-time Transport Control
379 Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
380 handling of fundamental issues by offering confidentiality, integrity
381 and partial source authentication. A mandatory to implement and use
382 media security solution is created by combining this secured RTP
383 profile and DTLS-SRTP keying [RFC5764] as defined in the
384 communication security section of this memo Section 7
386 RTCP packets convey a Canonical Name (CNAME) identifier that is used
387 to associate RTP packet streams that need to be synchronised across
388 related RTP sessions. Inappropriate choice of CNAME values can be a
389 privacy concern, since long-term persistent CNAME identifiers can be
390 used to track users across multiple calls. The communication
391 security section of this memo Section 7 mandates generation of short-
392 term persistent RTCP CNAMES, as specified in [RFC7022] so they can't
393 be used for long term tracking of the users.
395 Some potential denial of service attacks exist if the RTCP reporting
396 interval is configured to an inappropriate value. This could be done
397 by configuring the RTCP bandwidth fraction to an excessively large or
398 small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some
399 similar mechanism, or by choosing an excessively large or small value
400 for the RTP/AVPF minimal receiver report interval (if using SDP, this
401 is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as
402 follows:
404 1. the RTCP bandwidth could be configured to make the regular
405 reporting interval so large that effective congestion control
406 cannot be maintained, potentially leading to denial of service
407 due to congestion caused by the media traffic;
409 2. the RTCP interval could be configured to a very small value,
410 causing endpoints to generate high rate RTCP traffic, potentially
411 leading to denial of service due to the non-congestion controlled
412 RTCP traffic; and
414 3. RTCP parameters could be configured differently for each
415 endpoint, with some of the endpoints using a large reporting
416 interval and some using a smaller interval, leading to denial of
417 service due to premature participant timeouts due to mismatched
418 timeout periods which are based on the reporting interval (this
419 is a particular concern if endpoints use a small but non-zero
420 value for the RTP/AVPF minimal receiver report interval (trr-int)
421 [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]).
423 Premature participant timeout can be avoided by using the fixed (non-
424 reduced) minimum interval when calculating the participant timeout
425 ([I-D.ietf-avtcore-rtp-multi-stream]). To address the other
426 concerns, endpoints SHOULD ignore parameters that configure the RTCP
427 reporting interval to be significantly longer than the default five
428 second interval specified in [RFC3550] (unless the media data rate is
429 so low that the longer reporting interval roughly corresponds to 5%
430 of the media data rate), or that configure the RTCP reporting
431 interval small enough that the RTCP bandwidth would exceed the media
432 bandwidth.
434 The guidelines in [RFC6562] apply when using variable bit rate (VBR)
435 audio codecs such as Opus.
437 The use of the encryption of the header extensions are RECOMMENDED,
438 unless there are known reasons, like RTP middleboxes performing voice
439 activity based source selection or third party monitoring that will
440 greatly benefit from the information, and this has been expressed
441 using API or signalling. If further evidence are produced to show
442 that information leakage is significant from audio level indications,
443 then use of encryption needs to be mandated at that time.
445 In multi-party communication scenarios using RTP Middleboxes; this
446 middleboxes are REQUIRED, by this protocol, to not weaken the
447 sessions' security. The middlebox SHOULD maintain the
448 confidentiality, integrity and perform source authentication. The
449 middlebox MAY perform checks that prevents any endpoint participating
450 in a conference to impersonate another. Some additional security
451 considerations regarding multi-party topologies can be found in
452 [RFC7667]
454 The CaptureID is created as part of the CLUE protocol. The CaptId
455 SDES item is used to convey the same CaptureID value in the SDES
456 item. When sending the SDES item the security consideration
457 specified in the security section of [RFC7941] and in the
458 communication security section of this memo Section 7 are applicable.
459 Note that since the CaptureID is carried also in CLUE protocol
460 messages it is RECOMMENDED that this SDES item use at least similar
461 protection profiles as the CLUE protocol messages carried in the CLUE
462 data channel. .
464 11. References
466 11.1. Normative References
468 [I-D.ietf-clue-data-model-schema]
469 Presta, R. and S. Romano, "An XML Schema for the CLUE data
470 model", draft-ietf-clue-data-model-schema-17 (work in
471 progress), August 2016.
473 [I-D.ietf-clue-framework]
474 Duckworth, M., Pepperell, A., and S. Wenger, "Framework
475 for Telepresence Multi-Streams", draft-ietf-clue-
476 framework-25 (work in progress), January 2016.
478 [I-D.ietf-mmusic-sdp-bundle-negotiation]
479 Holmberg, C., Alvestrand, H., and C. Jennings,
480 "Negotiating Media Multiplexing Using the Session
481 Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
482 negotiation-36 (work in progress), October 2016.
484 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
485 Requirement Levels", BCP 14, RFC 2119,
486 DOI 10.17487/RFC2119, March 1997,
487 .
489 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
490 Norrman, "The Secure Real-time Transport Protocol (SRTP)",
491 RFC 3711, DOI 10.17487/RFC3711, March 2004,
492 .
494 [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
495 for Establishing a Secure Real-time Transport Protocol
496 (SRTP) Security Context Using Datagram Transport Layer
497 Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
498 2010, .
500 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
501 Security (DTLS) Extension to Establish Keys for the Secure
502 Real-time Transport Protocol (SRTP)", RFC 5764,
503 DOI 10.17487/RFC5764, May 2010,
504 .
506 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
507 Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
508 January 2012, .
510 [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure
511 Real-time Transport Protocol (SRTP)", RFC 6904,
512 DOI 10.17487/RFC6904, April 2013,
513 .
515 [RFC7941] Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
516 Header Extension for the RTP Control Protocol (RTCP)
517 Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
518 August 2016, .
520 11.2. Informative References
522 [FIPS186] National Institute of Standards and Technology, "Digital
523 Signature Standard", FIPS PUB 186-4, July 2013.
525 [I-D.ietf-avtcore-rtp-multi-stream]
526 Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
527 "Sending Multiple Media Streams in a Single RTP Session",
528 draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
529 December 2015.
531 [I-D.ietf-clue-signaling]
532 Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
533 Signaling", draft-ietf-clue-signaling-10 (work in
534 progress), January 2017.
536 [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
537 with Session Description Protocol (SDP)", RFC 3264,
538 DOI 10.17487/RFC3264, June 2002,
539 .
541 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
542 Jacobson, "RTP: A Transport Protocol for Real-Time
543 Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
544 July 2003, .
546 [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
547 Modifiers for RTP Control Protocol (RTCP) Bandwidth",
548 RFC 3556, DOI 10.17487/RFC3556, July 2003,
549 .
551 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
552 Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
553 July 2006, .
555 [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
556 Session Initiation Protocol (SIP) Event Package for
557 Conference State", RFC 4575, DOI 10.17487/RFC4575, August
558 2006, .
560 [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
561 "Extended RTP Profile for Real-time Transport Control
562 Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
563 DOI 10.17487/RFC4585, July 2006,
564 .
566 [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description
567 Protocol (SDP) Content Attribute", RFC 4796,
568 DOI 10.17487/RFC4796, February 2007,
569 .
571 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
572 Real-time Transport Control Protocol (RTCP)-Based Feedback
573 (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
574 2008, .
576 [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
577 Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
578 2008, .
580 [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
581 Real-Time Transport Control Protocol (RTCP): Opportunities
582 and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
583 2009, .
585 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
586 Variable Bit Rate Audio with Secure RTP", RFC 6562,
587 DOI 10.17487/RFC6562, March 2012,
588 .
590 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
591 "Guidelines for Choosing RTP Control Protocol (RTCP)
592 Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
593 September 2013, .
595 [RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP
596 Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
597 .
599 [RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP
600 Framework: Why RTP Does Not Mandate a Single Media
601 Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
602 2014, .
604 [RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
605 "Use Cases for Telepresence Multistreams", RFC 7205,
606 DOI 10.17487/RFC7205, April 2014,
607 .
609 [RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
610 DOI 10.17487/RFC7667, November 2015,
611 .
613 Authors' Addresses
614 Roni Even
615 Huawei Technologies
616 Tel Aviv
617 Israel
619 Email: roni.even@huawei.com
621 Jonathan Lennox
622 Vidyo, Inc.
623 433 Hackensack Avenue
624 Seventh Floor
625 Hackensack, NJ 07601
626 US
628 Email: jonathan@vidyo.com