RFC8849: Mapping RTP Streams to Controlling Multiple Streams for Telepresence (CLUE) Media Captures

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Internet Engineering Task Force (IETF)                           R. Even
Request for Comments: 8849                                              
Category: Standards Track                                      J. Lennox
ISSN: 2070-1721                                              8x8 / Jitsi
                                                            January 2021

  Mapping RTP Streams to Controlling Multiple Streams for Telepresence
                         (CLUE) Media Captures


   This document describes how the Real-time Transport Protocol (RTP) is
   used in the context of the Controlling Multiple Streams for
   Telepresence (CLUE) protocol.  It also describes the mechanisms and
   recommended practice for mapping RTP media streams, as defined in the
   Session Description Protocol (SDP), to CLUE Media Captures and
   defines a new RTP header extension (CaptureID).

Status of This Memo

   This is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   described in the Simplified BSD License.

Table of Contents

   1.  Introduction
   2.  Terminology
   3.  RTP Topologies for CLUE
   4.  Mapping CLUE Capture Encodings to RTP Streams
   5.  MCC Constituent CaptureID Definition
     5.1.  RTCP CaptureID SDES Item
     5.2.  RTP Header Extension
   6.  Examples
   7.  Communication Security
   8.  IANA Considerations
   9.  Security Considerations
   10. References
     10.1.  Normative References
     10.2.  Informative References
   Authors' Addresses

1.  Introduction

   Telepresence systems can send and receive multiple media streams.
   The CLUE Framework [RFC8845] defines Media Captures (MCs) as a source
   of Media, from one or more Capture Devices.  A Media Capture may also
   be constructed from other Media streams.  A middlebox can express
   conceptual Media Captures that it constructs from Media streams it
   receives.  A Multiple Content Capture (MCC) is a special Media
   Capture composed of multiple Media Captures.

   SIP Offer/Answer [RFC3264] uses SDP [RFC4566] to describe the RTP
   media streams [RFC3550].  Each RTP stream has a unique
   Synchronization Source (SSRC) within its RTP session.  The content of
   the RTP stream is created by an encoder in the endpoint.  This may be
   an original content from a camera or a content created by an
   intermediary device like a Multipoint Control Unit (MCU).

   This document makes recommendations for the CLUE architecture about
   how RTP and RTP Control Protocol (RTCP) streams should be encoded and
   transmitted and how their relation to CLUE Media Captures should be
   communicated.  The proposed solution supports multiple RTP topologies

   With regards to the media (audio, video, and timed text), systems
   that support CLUE use RTP for the media, SDP for codec and media
   transport negotiation (CLUE individual encodings), and the CLUE
   protocol for Media Capture description and selection.  In order to
   associate the media in the different protocols, there are three
   mappings that need to be specified:

   1.  CLUE individual encodings to SDP

   2.  RTP streams to SDP (this is not a CLUE-specific mapping)

   3.  RTP streams to MC to map the received RTP stream to the current
       MC in the MCC.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "OPTIONAL" in this document are to be interpreted as described in
   BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.

   Definitions from the CLUE Framework (see Section 3 of [RFC8845]) are
   used by this document as well.

3.  RTP Topologies for CLUE

   The typical RTP topologies used by CLUE telepresence systems specify
   different behaviors for RTP and RTCP distribution.  A number of RTP
   topologies are described in [RFC7667].  For CLUE telepresence, the
   relevant topologies include Point-to-Point, as well as Media-Mixing
   Mixers, Media-Switching Mixers, and Selective Forwarding Middleboxes.

   In the Point-to-Point topology, one peer communicates directly with a
   single peer over unicast.  There can be one or more RTP sessions,
   each sent on a separate 5-tuple, that have a separate SSRC space,
   with each RTP session carrying multiple RTP streams identified by
   their SSRC.  All SSRCs are recognized by the peers based on the
   information in the RTCP Source description (SDES) report that
   includes the Canonical Name (CNAME) and SSRC of the sent RTP streams.
   There are different Point-to-Point use cases as specified in the CLUE
   use case [RFC7205].  In some cases, a CLUE session that, at a high
   level, is Point-to-Point may nonetheless have an RTP stream that is
   best described by one of the mixer topologies.  For example, a CLUE
   endpoint can produce composite or switched captures for use by a
   receiving system with fewer displays than the sender has cameras.
   The Media Capture may be described using an MCC.

   For the media mixer topology [RFC7667], the peers communicate only
   with the mixer.  The mixer provides mixed or composited media
   streams, using its own SSRC for the sent streams.  If needed by the
   CLUE endpoint, the conference roster information including conference
   participants, endpoints, media, and media-id (SSRC) can be determined
   using the conference event package [RFC4575] element.

   Media-Switching Mixers and Selective Forwarding Middleboxes behave as
   described in [RFC7667].

4.  Mapping CLUE Capture Encodings to RTP Streams

   The different topologies described in Section 3 create different SSRC
   distribution models and RTP stream multiplexing points.

   Most video conferencing systems today can separate multiple RTP
   sources by placing them into RTP sessions using the SDP description;
   the video conferencing application can also have some knowledge about
   the purpose of each RTP session.  For example, video conferencing
   applications that have a primary video source and a slides video
   source can send each media source in a separate RTP session with a
   content attribute [RFC4796], enabling different application behavior
   for each received RTP media source.  Demultiplexing is
   straightforward because each Media Capture is sent as a single RTP
   stream, with each RTP stream being sent in a separate RTP session, on
   a distinct UDP 5-tuple.  This will also be true for mapping the RTP
   streams to Capture Encodings, if each Capture Encoding uses a
   separate RTP session and the consumer can identify it based on the
   receiving RTP port.  In this case, SDP only needs to label the RTP
   session with an identifier that can be used to identify the Media
   Capture in the CLUE description.  The SDP label attribute serves as
   this identifier.

   Each Capture Encoding MUST be sent as a separate RTP stream.  CLUE
   endpoints MUST support sending each such RTP stream in a separate RTP
   session signaled by an SDP "m=" line.  They MAY also support sending
   some or all of the RTP streams in a single RTP session, using the
   mechanism described in [RFC8843] to relate RTP streams to SDP "m="

   MCCs bring another mapping issue, in that an MCC represents multiple
   Media Captures that can be sent as part of the MCC if configured by
   the consumer.  When receiving an RTP stream that is mapped to the
   MCC, the consumer needs to know which original MC it is in order to
   get the MC parameters from the advertisement.  If a consumer
   requested a MCC, the original MC does not have a Capture Encoding, so
   it cannot be associated with an "m=" line using a label as described
   in "CLUE Signaling" [RFC8848].  It is important, for example, to get
   correct scaling information for the original MC, which may be
   different for the various MCs that are contributing to the MCC.

5.  MCC Constituent CaptureID Definition

   For an MCC that can represent multiple switched MCs, there is a need
   to know which MC is represented in the current RTP stream at any
   given time.  This requires a mapping from the SSRC of the RTP stream
   conveying a particular MCC to the constituent MC.  In order to
   address this mapping, this document defines an RTP header extension
   and SDES item that includes the captureID of the original MC,
   allowing the consumer to use the MC's original source attributes like
   the spatial information.

   This mapping temporarily associates the SSRC of the RTP stream
   conveying a particular MCC with the captureID of the single original
   MC that is currently switched into the MCC.  This mapping cannot be
   used for a composed case where more than one original MC is composed
   into the MCC simultaneously.

   If there is only one MC in the MCC, then the media provider MUST send
   the captureID of the current constituent MC in the RTP header
   extension and as an RTCP CaptureID SDES item.  When the media
   provider switches the MC it sends within an MCC, it MUST send the
   captureID value for the MC that just switched into the MCC in an RTP
   header extension and as an RTCP CaptureID SDES item as specified in

   If there is more than one MC composed into the MCC, then the media
   provider MUST NOT send any of the MCs' captureIDs using this
   mechanism.  However, if an MCC is sending Contributing Source (CSRC)
   information in the RTP header for a composed capture, it MAY send the
   captureID values in the RTCP SDES packets giving source information
   for the SSRC values sent as CSRCs.

   If the media provider sends the captureID of a single MC switched
   into an MCC, then later sends one composed stream of multiple MCs in
   the same MCC, it MUST send the special value "-", a single-dash
   character, as the captureID RTP header extension and RTCP CaptureID
   SDES item.  The single-dash character indicates there is no
   applicable value for the MCC constituent CaptureID.  The media
   consumer interprets this as meaning that any previous CaptureID value
   associated with this SSRC no longer applies.  As [RFC8846] defines
   the captureID syntax as "xs:ID", the single-dash character is not a
   legal captureID value, so there is no possibility of confusing it
   with an actual captureID.

5.1.  RTCP CaptureID SDES Item

   This document specifies a new RTCP SDES item.

    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
   |   CaptId=14   |     length    | CaptureID                     |
   |   ....        |

   This CaptureID is a variable-length UTF-8 string corresponding to
   either a CaptureID negotiated in the CLUE protocol or the single
   character "-".

   This SDES item MUST be sent in an SDES packet within a compound RTCP
   packet unless support for Reduced-Size RTCP has been negotiated as
   specified in RFC 5506 [RFC5506], in which case it can be sent as an
   SDES packet in a noncompound RTCP packet.

5.2.  RTP Header Extension

   The CaptureID is also carried in an RTP header extension [RFC8285],
   using the mechanism defined in [RFC7941].

   Support is negotiated within SDP using the URN "urn:ietf:params:rtp-

   The CaptureID is sent in an RTP header extension because for switched
   captures, receivers need to know which original MC corresponds to the
   media being sent for an MCC, in order to correctly apply geometric
   adjustments to the received media.

   As discussed in [RFC7941], there is no need to send the CaptId 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.  See [RFC7941] for further

6.  Examples

   In this partial advertisement, the media provider advertises a
   composed capture VC7 made of a big picture representing the current
   speaker (VC3) and two picture-in-picture boxes representing the
   previous speakers (the previous one -- VC5 -- and the oldest one --

        xsi:type="ns2:videoCaptureType" captureID="VC7"
            <ns2:description lang="en">big picture of the current
              speaker pips about previous speakers</ns2:description>

   In this case, the media provider will send capture IDs VC3, VC5, or
   VC6 as an RTP header extension and RTCP SDES message for the RTP
   stream associated with the MC.

   Note that this is part of the full advertisement message example from
   the CLUE data model example [RFC8846] and is not a valid XML

7.  Communication Security

   CLUE endpoints MUST support RTP/SAVPF profiles and the Secure Real-
   time Transport Protocol (SRTP) [RFC3711].  CLUE endpoints MUST
   support DTLS [RFC6347] and DTLS-SRTP [RFC5763] [RFC5764] for SRTP

   All media channels SHOULD be secure via SRTP and the RTP/SAVPF
   profile unless the RTP media and its associated RTCP are secure by
   other means (see [RFC7201] and [RFC7202]).

   All CLUE implementations MUST support DTLS 1.2 with the
   TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
   curve [FIPS186].  The DTLS-SRTP protection profile
   SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
   Implementations MUST favor cipher suites that support Perfect Forward
   Secrecy (PFS) over non-PFS cipher suites and SHOULD favor
   Authenticated Encryption with Associated Data (AEAD) over non-AEAD
   cipher suites.  Encrypted SRTP Header extensions [RFC6904] MUST be

   Implementations SHOULD implement DTLS 1.2 with the
   TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
   Implementations MUST favor cipher suites that support Perfect Forward
   Secrecy (PFS) over non- PFS cipher suites and SHOULD favor
   Authenticated Encryption with Associated Data (AEAD) over non-AEAD
   cipher suites.

   NULL Protection profiles MUST NOT be used for RTP or RTCP.

   CLUE endpoints MUST generate short-term persistent RTCP CNAMEs, as
   specified in [RFC7022], and thus can't be used for long-term tracking
   of the users.

8.  IANA Considerations

   This document defines a new extension URI in the "RTP SDES 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:sdes:CaptId

   Description:  CLUE CaptId

   Contact:  Roni Even <ron.even.tlv@gmail.com>

   Reference:  RFC 8849

   The IANA has registered one new RTCP SDES items in the "RTCP SDES
   Item Types" registry, as follows:

   | Value | Abbrev | Name        | Reference |
   | 14    | CCID   | CLUE CaptId | RFC 8849  |

                     Table 1

9.  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 believed that there are no 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 (RTP/SAVPF)" document [RFC5124]
   provides the handling of fundamental issues by offering
   confidentiality, integrity, and partial source authentication.  A
   mandatory-to-implement and use media security solution is created by
   combining this secured RTP profile and DTLS-SRTP keying [RFC5764] as
   defined in the communication security section of this memo
   (Section 7).

   RTCP packets convey a CNAME identifier that is used to associate RTP
   packet streams that need to be synchronized 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.  The communication security
   section of this memo (Section 7) mandates the generation of short-
   term persistent RTCP CNAMEs, as specified in [RFC7022], so they can't
   be used for long-term tracking of the users.

   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, which
       potentially leads 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, which are due to
       mismatched timeout periods that 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 [RFC8108]).

   Premature participant timeout can be avoided by using the fixed (non-
   reduced) minimum interval when calculating the participant timeout
   [RFC8108].  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.

   Encryption of the header extensions is 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
   signaling.  If further evidence is produced to show that information
   leakage is significant from audio level indications, then the use of
   encryption needs to be mandated at that time.

   In multi-party communication scenarios using RTP middleboxes, the
   middleboxes are REQUIRED, by this protocol, to not weaken the
   sessions' security.  The middlebox SHOULD maintain confidentiality,
   maintain integrity, and perform source authentication.  The middlebox
   MAY perform checks that prevent any endpoint participating in a
   conference to impersonate another.  Some additional security
   considerations regarding multi-party topologies can be found in

   The CaptureID is created as part of the CLUE protocol.  The CaptId
   SDES item is used to convey the same CaptureID value in the SDES
   item.  When sending the SDES item, the security considerations
   specified in Section 6 of [RFC7941] and in the communication security
   section of this memo (see Section 7) are applicable.  Note that since
   the CaptureID is also carried in CLUE protocol messages, it is
   RECOMMENDED that this SDES item use at least similar protection
   profiles as the CLUE protocol messages carried in the CLUE data

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
              2010, <https://www.rfc-editor.org/info/rfc5763>.

   [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,

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <https://www.rfc-editor.org/info/rfc6347>.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904,
              DOI 10.17487/RFC6904, April 2013,

   [RFC7941]  Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
              Header Extension for the RTP Control Protocol (RTCP)
              Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
              August 2016, <https://www.rfc-editor.org/info/rfc7941>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8843]  Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", RFC 8843,
              DOI 10.17487/RFC8843, January 2021,

   [RFC8845]  Duckworth, M., Ed., Pepperell, A., and S. Wenger,
              "Framework for Telepresence Multi-Streams", RFC 8845,
              DOI 10.17487/RFC8845, January 2021,

   [RFC8846]  Presta, R. and S P. Romano, "An XML Schema for the
              Controlling Multiple Streams for Telepresence (CLUE) Data
              Model", RFC 8846, DOI 10.17487/RFC8846, January 2021,

10.2.  Informative References

   [FIPS186]  National Institute of Standards and Technology (NIST),
              "Digital Signature Standard (DSS)", FIPS, PUB 186-4,
              DOI 10.6028/NIST.FIPS.186-4, July 2013,

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [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, <https://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,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <https://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, <https://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,

   [RFC4796]  Hautakorpi, J. and G. Camarillo, "The Session Description
              Protocol (SDP) Content Attribute", RFC 4796,
              DOI 10.17487/RFC4796, February 2007,

   [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, <https://www.rfc-editor.org/info/rfc5124>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <https://www.rfc-editor.org/info/rfc5506>.

   [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,

   [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, <https://www.rfc-editor.org/info/rfc7022>.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,

   [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
              2014, <https://www.rfc-editor.org/info/rfc7202>.

   [RFC7205]  Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
              "Use Cases for Telepresence Multistreams", RFC 7205,
              DOI 10.17487/RFC7205, April 2014,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

   [RFC8108]  Lennox, J., Westerlund, M., Wu, Q., and C. Perkins,
              "Sending Multiple RTP Streams in a Single RTP Session",
              RFC 8108, DOI 10.17487/RFC8108, March 2017,

   [RFC8285]  Singer, D., Desineni, H., and R. Even, Ed., "A General
              Mechanism for RTP Header Extensions", RFC 8285,
              DOI 10.17487/RFC8285, October 2017,

   [RFC8848]  Hanton, R., Kyzivat, P., Xiao, L., and C. Groves, "Session
              Signaling for Controlling Multiple Streams for
              Telepresence (CLUE)", RFC 8848, DOI 10.17487/RFC8848,
              January 2021, <https://www.rfc-editor.org/info/rfc8848>.


   The authors would like to thank Allyn Romanow and Paul Witty for
   contributing text to this work.  Magnus Westerlund helped draft the
   security section.

Authors' Addresses

   Roni Even
   Tel Aviv

   Email: ron.even.tlv@gmail.com

   Jonathan Lennox
   8x8, Inc. / Jitsi
   Jersey City, NJ 07302
   United States of America

   Email: jonathan.lennox@8x8.com