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In: MissingRef
CLUE WG                                                          R. Even
Internet-Draft                                       Huawei Technologies
Intended status: Standards Track                               J. Lennox
Expires: April 20, 2016                                            Vidyo
                                                        October 18, 2015


               Mapping RTP streams to CLUE media captures
                   draft-ietf-clue-rtp-mapping-05.txt

Abstract

   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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   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 April 20, 2016.

Copyright Notice

   Copyright (c) 2015 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
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   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 . . . . . . . .   5
     4.1.  Review of RTP related documents relevant to CLUE work.  .   6
     4.2.  Requirements of a solution  . . . . . . . . . . . . . . .   7
     4.3.  Static Mapping  . . . . . . . . . . . . . . . . . . . . .   9
     4.4.  Dynamic mapping . . . . . . . . . . . . . . . . . . . . .   9
     4.5.  Recommendations . . . . . . . . . . . . . . . . . . . . .   9
   5.  Application to CLUE Media Requirements  . . . . . . . . . . .  10
   6.  CaptureID definition  . . . . . . . . . . . . . . . . . . . .  11
     6.1.  RTCP CaptureId SDES Item  . . . . . . . . . . . . . . . .  12
     6.2.  RTP Header Extension  . . . . . . . . . . . . . . . . . .  12
   7.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  12
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  13
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  15
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  15
     11.2.  Informative References . . . . . . . . . . . . . . . . .  15
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  17

1.  Introduction

   Telepresence systems can send and receive multiple media streams.
   The CLUE framework [I-D.ietf-clue-framework] defines media captures
   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.

   This document makes recommendations, for the telepresence
   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.

   With regards to the media (audio and video), systems that support
   CLUE use RTP for the media, SDP for codec and media transport



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   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:

   1.  CLUE individual encodings to SDP

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

   3.  RTP media 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",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC2119[RFC2119] and
   indicate requirement levels for compliant RTP implementations.

3.  RTP topologies for CLUE

   The typical RTP topologies used by Telepresence systems specify
   different behaviors for RTP and RTCP distribution.  A number of RTP
   topologies are described in [I-D.ietf-avtcore-rtp-topologies-update].
   For telepresence, the relevant topologies include point-to-point, as
   well as media 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, and
   each RTP session can carry multiple RTP streams identified by their
   SSRC.  All SSRCs will be recognized by the peers based on the
   information in the RTCP SDES report that will include the CNAME and
   SSRC of the sent RTP streams.  There are different point to point use
   cases as specified in CLUE use case [RFC7205].  There may be a
   difference between the symmetric and asymmetric use cases.  While in
   the symmetric use case the typical mapping will be from a Media
   capture device to a render device (e.g. camera to monitor) in the
   asymmetric case the render device may receive different capture
   information (RTP stream from different cameras) if it has fewer
   rendering devices (monitors).  In some cases, a CLUE session which,
   at a high-level, is point-to-point may nonetheless have RTP 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.





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   For the Media Mixer topology
   [I-D.ietf-avtcore-rtp-topologies-update], the peers communicate only
   with the mixer.  The mixer provides mixed or composited media
   streams, using its own SSRC for the sent streams.  There are two
   cases here.  In the first case the mixer may have separate RTP
   sessions with each peer (similar to the point to point topology)
   terminating the RTCP sessions on the mixer; this is known as Topo-
   RTCP-Terminating MCU in [I-D.ietf-avtcore-rtp-topologies-update].  In
   the second case, the mixer can use a conference-wide RTP session
   similar to [I-D.ietf-avtcore-rtp-topologies-update] Topo-mixer or
   Topo-Video-switching.  The major difference is that for the second
   case, the mixer uses conference-wide RTP sessions, and distributes
   the RTCP reports to all the RTP session participants, enabling them
   to learn all the CNAMEs and SSRCs of the participants and know the
   contributing source or sources (CSRCs) of the original streams from
   the RTP header.  In the first case, the Mixer terminates the RTCP and
   the participants cannot know all the available sources based on the
   RTCP information.  The conference roster information including
   conference participants, endpoints, media and media-id (SSRC) can be
   available using the conference event package [RFC4575] element.

   In the Media-Switching Mixer topology
   [I-D.ietf-avtcore-rtp-topologies-update], 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.

   In the Selective Forwarding middlebox topology, the peer to mixer
   communication is unicast with RTCP mixer feedback.  Every potential
   sender in the conference has a source which may be "projected" by the
   mixer into every other RTP session in the conference; thus, every
   original source is maintained with an independent RTP identity to
   every receiver, maintaining separate decoding state and its original
   RTCP SDES information.  However, RTCP is terminated at the mixer,
   which might also perform reliability, repair, rate adaptation, or
   transcoding on the stream.  Senders' SSRCs may be renumbered by the
   mixer.  The sender may turn the projected sources on and off at any
   time, depending on which sources it thinks are most relevant for the
   receiver; this is the primary reason why this topology must act as an
   RTP mixer rather than as a translator, as otherwise these disabled
   sources would appear to have enormous packet loss.  Source switching
   is accomplished through this process of enabling and disabling
   projected sources, with the higher-level semantic assignment of
   reason for the RTP streams assigned externally.



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   The above topologies demonstrate two major RTP/RTCP behaviors:

   1.  The mixer may either use the source SSRC when forwarding RTP
       packets, or use its own created SSRC.  Still the mixer will
       distribute all RTCP information to all participants creating
       conference-wide RTP session/s.  This allows the participants to
       learn the available RTP sources in each RTP session.  The
       original source information will be the SSRC or in the CSRC
       depending on the topology.  The point to point case behaves like
       this.

   2.  The mixer terminates the RTCP from the source, creating separate
       RTP sessions with the peers.  In this case the participants will
       not receive the source SSRC in the CSRC.  Since this is usually a
       mixer topology, the source information is available from the SIP
       conference event package [RFC4575].  Subscribing to the
       conference event package allows each participant to know the
       SSRCs of all sources in the conference.

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 separate 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 works 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.  (The
   multiplexing is not at the SSRC level).

   Even though Session multiplexing is supported by CLUE, for scaling
   reasons, CLUE recommends using SSRC multiplexing in a single or
   multiple sessions using [I-D.ietf-mmusic-sdp-bundle-negotiation].  So
   we need to look at how to map RTP streams to Captures Encodings when
   SSRC multiplexing is used.

   When looking at SSRC multiplexing we can see that in various
   topologies, the SSRC behavior may be different:



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   1.  The SSRCs are static (assigned by the MCU/Mixer), and there is an
       SSRC for each media capture encoding defined in the CLUE
       protocol.  Source information may be conveyed using CSRC, or, in
       the case of topo-RTCP-Terminating MCU, is not conveyed.

   2.  The SSRCs are dynamic, representing the original source and are
       relayed by the Mixer/MCU to the participants.

   In the above two cases the MCU/Mixer may create an advertisement,
   with a virtual room capture scene.

   Another case we can envision is that the MCU / Mixer relays all the
   capture scenes from all advertisements to all consumers.  This means
   that the advertisement will include multiple capture scenes, each
   representing a separate TelePresence room with its own coordinate
   system.

   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
   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.

   Editor's note: This section provides an overview of the RFCs and
   drafts that are 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 tools.

   When looking at the available tools based on current work in MMUSIC,
   AVTcore and AVText 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=").
   [bundle] specifies how to associate a received RTP stream with the



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   m-line describing it.  The assumption in the work 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] 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 but in order to offer
   multiple RTP streams with different resolutions it does it 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
   [I-D.ietf-mmusic-sdp-simulcast]

   In the next sections, the document will propose mechanisms to map the
   RTP streams to media captures addressing.

4.2.  Requirements of a solution

   This section lists, more briefly, the requirements a media
   architecture for Clue telepresence needs to achieve, summarizing the
   discussion of previous sections.  In this section, RFC 2119 [RFC2119]
   language refers to requirements on a solution, not an implementation;
   thus, requirements keywords are not written in capital letters.

   Media-1: It must not be necessary for a Clue session to use more than
   a single transport flow for transport of a given media type (video or
   audio).

   Media-2: It must, however, be possible for a Clue session to use
   multiple transport flows for a given media type where it is
   considered valuable (for example, for distributed media, or
   differential quality-of-service).

   Media-3: It must be possible for a Clue endpoint or MCU to
   simultaneously send sources corresponding to static captures and to
   both composited and switched multi-content captures in the same
   transport flow.  (Any given device might not necessarily be able send




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   all of these source types; but for those that can, it must be
   possible for them to be sent simultaneously.)

   Media-4: It must be possible for an original source to move among
   multi-content captures (i.e. at one time be sent for one MCC, and at
   a later time be sent for another one).

   Media-5: It must be possible for a source to be placed into a MCC
   even if the source is a "late joiner", 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.








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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.




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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.





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   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.







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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





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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.



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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
   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.



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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
   [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.





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   [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>.



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   [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



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

   Email: jonathan@vidyo.com











































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