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Mapping RTP streams to CLUE Media Captures

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.
Authors Roni Even , Jonathan Lennox
Last updated 2016-09-15 (Latest revision 2016-08-27)
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
Stream WG state WG Consensus: Waiting for Write-Up
Revised I-D Needed - Issue raised by WGLC
Document shepherd Paul Kyzivat
Shepherd write-up Show Last changed 2016-08-24
IESG IESG state Became RFC 8849 (Proposed Standard)
Consensus boilerplate Unknown
Telechat date (None)
Responsible AD Alissa Cooper
Send notices to "Paul Kyzivat" <>
CLUE WG                                                          R. Even
Internet-Draft                                       Huawei Technologies
Intended status: Standards Track                               J. Lennox
Expires: February 28, 2017                                         Vidyo
                                                         August 27, 2016

               Mapping RTP streams to CLUE Media Captures


   This document describes how the Real Time transport Protocol (RTP) is
   used in the context of the CLUE protocol.  It also describes the
   mechanisms and recommended practice for mapping RTP media streams
   defined in SDP to CLUE Media Captures.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on February 28, 2017.

Copyright Notice

   Copyright (c) 2016 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  RTP topologies for CLUE . . . . . . . . . . . . . . . . . . .   3
   4.  Mapping CLUE Capture Encodings to RTP streams . . . . . . . .   4
     4.1.  Review of RTP related documents relevant to CLUE work.  .   5
     4.2.  Recommendations . . . . . . . . . . . . . . . . . . . . .   6
   5.  CaptureID definition  . . . . . . . . . . . . . . . . . . . .   6
     5.1.  RTCP CaptureId SDES Item  . . . . . . . . . . . . . . . .   6
     5.2.  RTP Header Extension  . . . . . . . . . . . . . . . . . .   6
   6.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   9.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  10
     10.2.  Informative References . . . . . . . . . . . . . . . . .  10
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  12

1.  Introduction

   Telepresence systems can send and receive multiple media streams.
   The CLUE framework [I-D.ietf-clue-framework] defines Media Captures
   (MC) as a source of Media, such as from one or more Capture Devices.
   A Media Capture may also be constructed from other Media streams.  A
   middle box 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[RFC3550] media streams.  Each RTP stream has a unique 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 an MCU
   (Multipoint Control Unit).

   This document makes recommendations, for the CLUE architecture, about
   how RTP and 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 [RFC7667].

   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 mapping that need to
   be specified:

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   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 steam to the current MC
       in the MCC.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC2119[RFC2119] and
   indicate requirement levels for compliant RTP implementations.

   The definitions from the CLUE framework document
   [I-D.ietf-clue-framework] section 3 are used by this document as

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 telepresence, the
   relevant topologies include Point-to-Point, as well as Media-Mixing
   mixers, Media- Switching mixers, and Selective Forwarding Middleboxs.

   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, and having 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 SDES report that includes the CNAME and SSRC
   of the sent RTP streams.  There are different Point-to-Point use
   cases as specified in CLUE use case [RFC7205].  In some cases, a CLUE
   session which, at a high-level, is point-to-point may nonetheless
   have an RTP stream which 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 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.  The conference
   roster information including conference participants, endpoints,
   media and media-id (SSRC) can be determined using the conference
   event package [RFC4575] element.

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   In the Media-Switching Mixer topology [RFC7667], the peer to mixer
   communication is unicast with mixer RTCP feedback.  It is
   conceptually similar to a compositing mixer as described in the
   previous paragraph, except that rather than compositing or mixing
   multiple sources, the mixer provides one or more conceptual sources
   selecting one source at a time from the original sources.  The Mixer
   creates a conference-wide RTP session by sharing remote SSRC values
   as CSRCs to all conference participants, and forwarding RTCP reports.

   In the Selective Forwarding Middlebox (SFM) [RFC7667] topology, the
   peer to middlebox communication is unicast with RTCP feedback.  Every
   potential sender in the conference has a source which may be
   "projected" by the SFM into every other RTP session in the
   conference; thus, even though the SFM establishes a separate RTP
   session with each endpoint, every original source is maintained with
   an independent SSRC to every receiver, maintaining separate decoding
   state and its original RTCP SDES information.

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.
   For example, main and slides video sources are separated into
   separate RTP sessions based on the content attribute [RFC4796].  This
   solution is straightforward if the multiplexing point is at the UDP
   transport level, where each RTP stream uses a separate RTP session.
   This will also be true for mapping the RTP streams to Media Captures
   Encodings if each Media Capture Encodings 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.  In
   this case, the mapping does not change even if the RTP session is
   switched using same or different SSRC.

   Even though Session multiplexing is supported by CLUE, for scaling
   reasons, CLUE indicates that SSRC multiplexing in a single or
   multiple sessions using [I-D.ietf-mmusic-sdp-bundle-negotiation]may
   be used.  When SSRC multiplexing is used, the mapping of RTP streams
   to Captures Encodings needs to be considered.

   MCCs bring another mapping issue, in that an MCC represents multiple
   Media Captures that can be sent as part of this MCC if configured by
   the consumer.  When receiving an RTP stream which is mapped to the
   MCC, the consumer needs to know which original MC it is in order to

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   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 [I-D.ietf-clue-signaling].  This 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.

4.1.  Review of RTP related documents relevant to CLUE work.

   This section provides an overview of the RFCs and drafts that can be
   used in a CLUE system and as a base for a mapping solution.  This
   section is for information only; the normative behavior is given in
   the cited documents.  Tools for SSRC multiplexing support are defined
   for general conferencing applications; CLUE systems use the same

   When looking at the available tools based on current work in MMUSIC,
   AVTcore and AVText Working Groups for supporting SSRC multiplexing
   the following documents are considered to be relevant.

   Negotiating Media Multiplexing Using the Session Description Protocol
   in [I-D.ietf-mmusic-sdp-bundle-negotiation] defines a "bundle" SDP
   grouping extension that can be used with SDP Offer/Answer mechanism
   to negotiate the usage of a single 5-tuple for sending and receiving
   media associated with multiple SDP media descriptions ("m=").
   [I-D.ietf-mmusic-sdp-bundle-negotiation] specifies how to associate a
   received RTP stream with the m-line describing it.  The assumption in
   Bundle is that each SDP m-line represents a single media source.
   [I-D.ietf-mmusic-sdp-bundle-negotiation] specifies using the SDP mid
   value and sending it as RTCP SDES and an RTP header extension in
   order to be able to map the RTP stream to the SDP m-line.  This is
   relevant when there are multiple RTP streams with the same payload
   subtype number.

   SDP Source attribute [RFC5576] provides mechanisms to describe
   specific attributes of RTP sources based on their SSRC.

   Negotiation of generic image attributes in SDP [RFC6236] provides the
   means to negotiate the image size.  The image attribute can be used
   to offer different image parameters like size.  Offering multiple RTP
   streams with different resolutions is done using separate RTP session
   for each image option.  ([I-D.ietf-mmusic-sdp-bundle-negotiation]
   provides the support of a single RTP session but each image option
   will need a separate SDP m-line).

   The recommended support of the simulcast case is to use

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4.2.  Recommendations

   The recommendation is that CLUE endpoints using SSRC multiplexing
   MUST support [I-D.ietf-mmusic-sdp-bundle-negotiation].

5.  CaptureID definition

   For MCC which can represent multiple switched MCs there is a need to
   know which MC represents the current RTP stream.  This 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 original source MC attributes like the spatial
   information.  The media provider MUST send for MCC Capture Encoding
   the captureID of the current MC in the RTP header and as a RTCP SDES

5.1.  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 = XX |     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.

5.2.  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 CaptureId then contain a header
   extension as shown below

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  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
        | ....              |

   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.

6.  Examples

   In this partial advertisement the Media Provider advertises a
   composed capture VC7 made by 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 -VC6).

<ns2:mediaCapture xmlns:xsi=""
      xsi:type="ns2:videoCaptureType" captureID="VC7" mediaType="video">
          <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.

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7.  Acknowledgements

   The authors would like to thanks Allyn Romanow and Paul Witty for
   contributing text to this work.

8.  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


      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]

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 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.

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   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

   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

   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.

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   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

10.  References

10.1.  Normative References

              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-25 (work in progress), January 2016.

              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-32 (work in progress), August 2016.

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

10.2.  Informative References

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
              December 2015.

              Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
              Signaling", draft-ietf-clue-signaling-09 (work in
              progress), March 2016.

              Westerlund, M., Nandakumar, S., and M. Zanaty, "Using
              Simulcast in SDP and RTP Sessions", draft-ietf-mmusic-sdp-
              simulcast-05 (work in progress), June 2016.

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   [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, <>.

   [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, <>.

   [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, <>.

   [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,

   [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, <>.

   [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, <>.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <>.

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   [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,

   [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,

   [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,

   [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, <>.

   [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,

Authors' Addresses

   Roni Even
   Huawei Technologies
   Tel Aviv


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   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601


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