Mapping RTP streams to CLUE media captures
draft-ietf-clue-rtp-mapping-05
The information below is for an old version of the document.
Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8849.
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Authors | Roni Even , Jonathan Lennox | ||
Last updated | 2015-10-18 | ||
RFC stream | Internet Engineering Task Force (IETF) | ||
Formats | |||
Reviews |
GENART Last Call review
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by Vijay Gurbani
Ready w/nits
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Additional resources | Mailing list discussion | ||
Stream | WG state | WG Document | |
Document shepherd | Daniel C. Burnett | ||
IESG | IESG state | Became RFC 8849 (Proposed Standard) | |
Consensus boilerplate | Unknown | ||
Telechat date | (None) | ||
Responsible AD | (None) | ||
Send notices to | (None) |
draft-ietf-clue-rtp-mapping-05
quot;, i.e. was added to the conference after the receiver requested the MCC. Media-6: Whenever a given source is assigned to a switched capture, it must be immediately possible for a receiver to determine the MCC it corresponds to, and thus that any previous source is no longer being mapped to that switched capture. Media-7: It must be possible for a receiver to identify the original capture(s) that are currently being mapped to an MCC, and correlate it with both the Clue advertisement and out-of-band (non-Clue) information such as rosters. Media-8: It must be possible for a source to move among MCCs without requiring a refresh of decoder state (e.g., for video, a fresh I-frame), when this is unnecessary. However, it must also be possible for a receiver to indicate when a refresh of decoder state is in fact necessary. Media-9: If a given source is being sent on the same transport flow for more than one reason (e.g. if it corresponds to more than one switched capture at once, or to a static capture), it should be possible for a sender to send only one copy of the source. Media-10: On the network, media flows should, as much as possible, look and behave like currently-defined usages of existing protocols; established semantics of existing protocols must not be redefined. Media-11: The solution should seek to minimize the processing burden for boxes that distribute media to decoding hardware. Media-12: If multiple sources from a single synchronization context are being sent simultaneously, it must be possible for a receiver to associate and synchronize them properly, even for sources that are are mapped to switched captures. Even & Lennox Expires April 20, 2016 [Page 8] Internet-Draft RTP mapping to CLUE October 2015 4.3. Static Mapping Static mapping is widely used in current MCU implementations. It is also common for a point to point symmetric use case when both endpoints have the same capabilities. For capture encodings with static SSRCs, it is most straightforward to indicate this mapping outside the media stream, in the CLUE or SDP signaling. When using SSRC multiplexing [I-D.ietf-mmusic-sdp-bundle-negotiation] defines the use of the SDP mid attribute value to associate between the received RTP stream and the SDP m-line. The mid is carried as an RTP header extension and RTCP SDES message defined in [I-D.ietf-mmusic-sdp-bundle-negotiation] . 4.4. Dynamic mapping Dynamic mapping by tagging each media packet with the SDP mid value. This means that a receiver immediately knows how to interpret received media, even when an unknown SSRC is seen. As long as the media carries a known mid, it can be assumed that this media stream will replace the stream currently being received with that mid. This gives significant advantages to switching latency, as a switch between sources can be achieved without any form of negotiation with the receiver. However, the disadvantage in using a mid in the stream that it introduces additional processing costs for every media packet, as mid are scoped only within one hop (i.e., within a cascaded conference a mid that is used from the source to the first MCU is not meaningful between two MCUs, or between an MCU and a receiver), and so they may need to be added or modified at every stage. An additional issue with putting mid in the RTP packets comes from cases where a non-bundle aware endpoint is being switched by an MCU to a bundle endpoint. In this case, we may require up to an additional 12 bytes in the RTP header, which may push a media packet over the MTU. However, as the MTU on either side of the switch may not match, it is possible that this could happen even without adding extra data into the RTP packet. The 12 additional bytes per packet could also be a significant bandwidth increase in the case of very low bandwidth audio codecs. 4.5. Recommendations The recommendation is that CLUE endpoint using SSRC multiplexing MUST support [[I-D.ietf-mmusic-sdp-bundle-negotiation] and use the SDP mid attribute for mapping. Even & Lennox Expires April 20, 2016 [Page 9] Internet-Draft RTP mapping to CLUE October 2015 5. Application to CLUE Media Requirements The requirement section Section 4.2 offers a number of requirements that are believed to be necessary for a CLUE RTP mapping. The solutions described in this document are believed to meet these requirements, though some of them are only possible for some of the topologies. (Since the requirements are generally of the form "it must be possible for a sender to do something", this is adequate; a sender which wishes to perform that action needs to choose a topology which allows the behavior it wants. In this section we address only those requirements where the topologies or the association mechanisms treat the requirements differently. Media-4: It must be possible for an original source to move among switched captures (i.e. at one time be sent for one switched capture, and at a later time be sent for another one). This applies naturally for static sources with a Switched Mixer. For dynamic sources with a Selective Forwarding middlebox, this just requires the mid in the header extension element to be updated appropriately. Media-6: Whenever a given source is transmitted for a switched capture, it must be immediately possible for a receiver to determine the switched capture it corresponds to, and thus that any previous source is no longer being mapped to that switched capture. For a Switched Mixer, this applies naturally. For a Selective Forwarding middlebox, this is done based on the mid. Media-7: It must be possible for a receiver to identify the original source that is currently being mapped to a switched capture, and correlate it with out-of-band (non-Clue) information such as rosters. For a Switched Mixer, this is done based on the CSRC, if the mixer is providing CSRCs; For a Selective Forwarding middlebox, this is done based on the SSRC. For MCC which can represent multiple switched MCs there is a need to know which MC represents the current RTP stream, requires a mapping from an RTP stream to an MC. In order to address this mapping this document defines an RTP header extension that includes the CaptureID in order to map to the original MC allowing the consumer to use the MC attributes like the spatial information. Even & Lennox Expires April 20, 2016 [Page 10] Internet-Draft RTP mapping to CLUE October 2015 Media-8: It must be possible for a source to move among switched captures without requiring a refresh of decoder state (e.g., for video, a fresh I-frame), when this is unnecessary. However, it must also be possible for a receiver to indicate when a refresh of decoder state is in fact necessary. This can be done by a Selective Forwarding middlebox, but not by a Switching Mixer. The last requirement can be accomplished through an FIR message [RFC5104], though potentially a faster mechanism (not requiring a round-trip time from the receiver) would be preferable. Media-9: If a given source is being sent on the same transport flow to satisfy more than one capture (e.g. if it corresponds to more than one switched capture at once, or to a static capture as well as a switched capture), it should be possible for a sender to send only one copy of the source. For a Selective Forwarding middlebox, this may be a problem since an encoding can be used by a single MC, it will require using the same SDP label for multiple MC (example middle camera and active speaker MC) this can also be done for an environment with a hybrid of mixer topologies and static and dynamic captures. It is not possible for static captures from a Switched Mixer. Media-12: If multiple sources from a single synchronization context are being sent simultaneously, it must be possible for a receiver to associate and synchronize them properly, even for sources that are mapped to switched captures. For a Mixed or Switched Mixer topology, receivers will see only a single synchronization context (CNAME), corresponding to the mixer. For a Selective Forwarding middlebox, separate projecting sources keep separate synchronization contexts based on their original CNAMEs, thus allowing independent synchronization of sources from independent rooms without needing global synchronization. In hybrid cases, however (e.g. if audio is mixed), all sources which need to be synchronized with the mixed audio must get the same CNAME (and thus a mixer-provided timebase) as the mixed audio. 6. CaptureID definition For mapping an RTP stream to a specific MC in the MCC the CLUE captureId is used. The media sender MUST send for MCC the captureID in the RTP header and as a RTCP SDES message. Even & Lennox Expires April 20, 2016 [Page 11] Internet-Draft RTP mapping to CLUE October 2015 6.1. RTCP CaptureId SDES Item This document specifies a new RTCP SDES message 0 1 2 3 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | CaptureId = XXX | length |CaptureId +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | .... This CaptureID is the same as in the CLUE MC and is also used in the RTP header extension. This SDES message MAY be sent in a compound RTCP packet based on the application need. 6.2. RTP Header Extension The CaptureId is carried within the RTP header extension field, using [RFC5285] two bytes header extension. Support is negotiated within the SDP, i.e. a=extmap:1 urn:ietf:params:rtp-hdrext:CaptureId Packets tagged by the sender with the CapturId then contain a header extension as shown below 0 1 2 3 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | ID | Len-1 | CaptureId +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | CaptureId .. | +-+-+-+-+-+-+-+-+ There is no need to send the CaptureId header extension with all RTP packets. Senders MAY choose to send it only when a new MC is sent. If such a mode is being used, the header extension SHOULD be sent in the first few RTP packets to reduce the risk of losing it due to packet loss. 7. Examples TBD Even & Lennox Expires April 20, 2016 [Page 12] Internet-Draft RTP mapping to CLUE October 2015 8. Acknowledgements The authors would like to thanks Allyn Romanow and Paul Witty for contributing text to this work. 9. IANA Considerations This document defines a new extension URI in the RTP Compact Header Extensions subregistry of the Real-Time Transport Protocol (RTP) Parameters registry, according to the following data: Extension URI: urn:ietf:params:rtp-hdrext:CaptureId Description: CLUE CaptureId Contact: roni.even@mail01.huawei.com Reference: RFC XXXX The IANA is requested to register one new RTCP SDES items in the "RTCP SDES Item Types" registry, as follows: Value Abbrev Name Reference TBA CCID CLUE CaptureId [RFCXXXX] 10. Security Considerations The security considerations of the RTP specification, the RTP/SAVPF profile, and the various RTP/RTCP extensions and RTP payload formats that form the complete protocol suite described in this memo apply. It is not believed there are any new security considerations resulting from the combination of these various protocol extensions. The Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides handling of fundamental issues by offering confidentiality, integrity and partial source authentication. A mandatory to support media security solution is created by combining this secured RTP profile and DTLS-SRTP keying [RFC5764] RTCP packets convey a Canonical Name (CNAME) identifier that is used to associate RTP packet streams that need to be synchronised across related RTP sessions. Inappropriate choice of CNAME values can be a privacy concern, since long-term persistent CNAME identifiers can be used to track users across multiple calls. This memo mandates generation of short-term persistent RTCP CNAMES, as specified in RFC7022 [RFC7022], resulting in untraceable CNAME values that alleviate this risk. Even & Lennox Expires April 20, 2016 [Page 13] Internet-Draft RTP mapping to CLUE October 2015 Some potential denial of service attacks exist if the RTCP reporting interval is configured to an inappropriate value. This could be done by configuring the RTCP bandwidth fraction to an excessively large or small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some similar mechanism, or by choosing an excessively large or small value for the RTP/AVPF minimal receiver report interval (if using SDP, this is the "a=rtcp-fb:... trr-int" parameter) [RFC4585] The risks are as follows: 1. the RTCP bandwidth could be configured to make the regular reporting interval so large that effective congestion control cannot be maintained, potentially leading to denial of service due to congestion caused by the media traffic; 2. the RTCP interval could be configured to a very small value, causing endpoints to generate high rate RTCP traffic, potentially leading to denial of service due to the non-congestion controlled RTCP traffic; and 3. RTCP parameters could be configured differently for each endpoint, with some of the endpoints using a large reporting interval and some using a smaller interval, leading to denial of service due to premature participant timeouts due to mismatched timeout periods which are based on the reporting interval (this is a particular concern if endpoints use a small but non-zero value for the RTP/AVPF minimal receiver report interval (trr-int) [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]). Premature participant timeout can be avoided by using the fixed (non- reduced) minimum interval when calculating the participant timeout ([I-D.ietf-avtcore-rtp-multi-stream]). To address the other concerns, endpoints SHOULD ignore parameters that configure the RTCP reporting interval to be significantly longer than the default five second interval specified in [RFC3550] (unless the media data rate is so low that the longer reporting interval roughly corresponds to 5% of the media data rate), or that configure the RTCP reporting interval small enough that the RTCP bandwidth would exceed the media bandwidth. The guidelines in [RFC6562] apply when using variable bit rate (VBR) audio codecs such as Opus. The use of the encryption of the header extensions are RECOMMENDED, unless there are known reasons, like RTP middleboxes performing voice activity based source selection or third party monitoring that will greatly benefit from the information, and this has been expressed using API or signalling. If further evidence are produced to show that information leakage is significant from audio level indications, then use of encryption needs to be mandated at that time. Even & Lennox Expires April 20, 2016 [Page 14] Internet-Draft RTP mapping to CLUE October 2015 In multi-party communication scenarios using RTP Middleboxes, a lot of trust is placed on these middleboxes to preserve the sessions security. The middlebox needs to maintain the confidentiality, integrity and perform source authentication. The middlebox can perform checks that prevents any endpoint participating in a conference to impersonate another. Some additional security considerations regarding multi-party topologies can be found in [I-D.ietf-avtcore-rtp-topologies-update] 11. References 11.1. Normative References [I-D.ietf-clue-framework] Duckworth, M., Pepperell, A., and S. Wenger, "Framework for Telepresence Multi-Streams", draft-ietf-clue- framework-23 (work in progress), September 2015. [I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H., and C. Jennings, "Negotiating Media Multiplexing Using the Session Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle- negotiation-23 (work in progress), July 2015. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <http://www.rfc-editor.org/info/rfc2119>. 11.2. Informative References [I-D.ietf-avtcore-rtp-multi-stream] Lennox, J., Westerlund, M., Wu, W., and C. Perkins, "Sending Multiple Media Streams in a Single RTP Session", draft-ietf-avtcore-rtp-multi-stream-09 (work in progress), September 2015. [I-D.ietf-avtcore-rtp-topologies-update] Westerlund, M. and S. Wenger, "RTP Topologies", draft- ietf-avtcore-rtp-topologies-update-10 (work in progress), July 2015. [I-D.ietf-clue-signaling] Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE Signaling", draft-ietf-clue-signaling-06 (work in progress), August 2015. Even & Lennox Expires April 20, 2016 [Page 15] Internet-Draft RTP mapping to CLUE October 2015 [I-D.ietf-mmusic-sdp-simulcast] Westerlund, M., Nandakumar, S., and M. Zanaty, "Using Simulcast in SDP and RTP Sessions", draft-ietf-mmusic-sdp- simulcast-02 (work in progress), October 2015. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002, <http://www.rfc-editor.org/info/rfc3264>. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, <http://www.rfc-editor.org/info/rfc3550>. [RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556, DOI 10.17487/RFC3556, July 2003, <http://www.rfc-editor.org/info/rfc3556>. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006, <http://www.rfc-editor.org/info/rfc4566>. [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A Session Initiation Protocol (SIP) Event Package for Conference State", RFC 4575, DOI 10.17487/RFC4575, August 2006, <http://www.rfc-editor.org/info/rfc4575>. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, DOI 10.17487/RFC4585, July 2006, <http://www.rfc-editor.org/info/rfc4585>. [RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description Protocol (SDP) Content Attribute", RFC 4796, DOI 10.17487/RFC4796, February 2007, <http://www.rfc-editor.org/info/rfc4796>. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104, February 2008, <http://www.rfc-editor.org/info/rfc5104>. [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, DOI 10.17487/RFC5117, January 2008, <http://www.rfc-editor.org/info/rfc5117>. Even & Lennox Expires April 20, 2016 [Page 16] Internet-Draft RTP mapping to CLUE October 2015 [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <http://www.rfc-editor.org/info/rfc5124>. [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 2008, <http://www.rfc-editor.org/info/rfc5285>. [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009, <http://www.rfc-editor.org/info/rfc5576>. [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)", RFC 5764, DOI 10.17487/RFC5764, May 2010, <http://www.rfc-editor.org/info/rfc5764>. [RFC6236] Johansson, I. and K. Jung, "Negotiation of Generic Image Attributes in the Session Description Protocol (SDP)", RFC 6236, DOI 10.17487/RFC6236, May 2011, <http://www.rfc-editor.org/info/rfc6236>. [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of Variable Bit Rate Audio with Secure RTP", RFC 6562, DOI 10.17487/RFC6562, March 2012, <http://www.rfc-editor.org/info/rfc6562>. [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, September 2013, <http://www.rfc-editor.org/info/rfc7022>. [RFC7205] Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed., "Use Cases for Telepresence Multistreams", RFC 7205, DOI 10.17487/RFC7205, April 2014, <http://www.rfc-editor.org/info/rfc7205>. Authors' Addresses Roni Even Huawei Technologies Tel Aviv Israel Email: roni.even@mail01.huawei.com Even & Lennox Expires April 20, 2016 [Page 17] Internet-Draft RTP mapping to CLUE October 2015 Jonathan Lennox Vidyo, Inc. 433 Hackensack Avenue Seventh Floor Hackensack, NJ 07601 US Email: jonathan@vidyo.com Even & Lennox Expires April 20, 2016 [Page 18]