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MMUSIC WG                                                        R. Even
Internet-Draft                                       Huawei Technologies
Intended status: Informational                                 J. Lennox
Expires: October 13, 2014                                          Vidyo
                                                                   Q. Wu
                                                     Huawei Technologies
                                                          April 11, 2014


   The Session Description Protocol (SDP) Application Token Attribute
               draft-even-mmusic-application-token-03.txt

Abstract

   The RTP fixed header includes the payload type number and the SSRC
   values of the RTP stream.  RTP defines how to de-multiplex streams
   within an RTP session, but in some use cases applications need
   further identifiers in order to identify the signaling descriptions
   associated with particular RTP media streams.

   This document defines a mechanism to provide the mapping between the
   SSRCs of RTP streams and the SDP m-line description by defining
   extensions to RTP and RTCP messages.

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 October 13, 2014.

Copyright Notice

   Copyright (c) 2014 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



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   (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.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Proposal for Application ID  . . . . . . . . . . . . . . . . .  4
     3.1.  appId token  . . . . . . . . . . . . . . . . . . . . . . .  5
       3.1.1.  RTCP SDES message  . . . . . . . . . . . . . . . . . .  7
       3.1.2.  RTP Header Extension . . . . . . . . . . . . . . . . .  8
       3.1.3.  recv-appId . . . . . . . . . . . . . . . . . . . . . .  8
   4.  Using Application ID token in Offer / Answer . . . . . . . . .  9
   5.  Other Considerations . . . . . . . . . . . . . . . . . . . . . 10
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 12
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 12
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 12
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 13
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 14
























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1.  Introduction

   The RTP [RFC3550] header includes the payload type number and the
   SSRC values of the RTP stream.  RTP defines how to de-multiplex
   streams within an RTP session, but in some use cases, applications
   need further identifiers in order to map an RTP media stream to its
   SDP m-line description.

   SDP [RFC4566] can be used to describe multiple RTP media streams in
   one or more m-lines that define a single SSRC multiplexed RTP session
   (as specified in [RFC3550]).  This addresses the WebRTC architecture
   [I-D.ietf-rtcweb-overview].

   A Unified Plan for Using SDP with Large Numbers of Media Flows
   [I-D.roach-mmusic-unified-plan] proposes that each m-line will
   represent a media source [I-D.ietf-avtext-rtp-grouping-taxonomy].  In
   the simple case a media source will be one video or audio RTP stream.
   Media source description becomes more complicated when, for robust
   applications, techniques like retransmission (RTX) and Forward Error
   Correction (FEC) are used to protect media, or simulcast or layered
   coding can be used to provide support to heterogeneous receivers.  In
   these cases a media source may send more than one RTP stream: for
   example, a scalable video stream base layer, an enhancement layer and
   a FEC stream.

   Multiple SDP m-lines can be multiplexed to a single RTP session using
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  The same payload type
   number can be used in multiple bundled m-lines.  An [RFC3264] offerer
   may receive an RTP media stream before the SDP answer, and if the
   same payload type number is used in multiple bundled m-lines, the
   offerer will not be able to associate incoming media using that pt
   number to a specific m-line.

   Some applications may require more information about the usage of the
   RTP streams: for example, RTP streams from different cameras that
   need to be identified by the application in order to render them
   correctly, or a source that can send multiple versions of the same
   stream in different resolutions (i.e. simulcast
   [I-D.westerlund-avtcore-rtp-simulcast]).

   A selective forwarding middlebox as described in RTP topologies
   section 3.7 [topologies] may project the RTP stream from the source
   to the destination and forward new SSRCs without any signaling.

   A three camera telepresence system may send two video media stream of
   the two recent active speakers to a system with two monitors.  In
   this case it may send first the video from the left and center camera
   (this will cause the video from the center camera to be displayed on



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   the right) and later the video from the center and right camera (this
   will cause the video from the center camera to be displayed on the
   left).  The SSRC of the video stream from the center camera will
   remain the same but the mapping to the stream description will
   change.

   As discussed in [I-D.roach-mmusic-unified-plan] during call
   establishment, circumstances may arise under which an endpoint can
   send an offer to receive a new stream, and begin receiving media for
   that media stream prior to receiving the SDP that correlates its SSRC
   to the m-line.  For such cases, the endpoint will not know how to
   handle the media, and will most probably be forced to discard it.


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.  Proposal for Application ID

   As we saw in the introduction, the SSRC of the RTP stream which
   should be created by the RTP layer is used by the SDP in the offer to
   map an RTP stream description to the SDP description.  There are
   cases where the SSRCs of the RTP streams may not be available or do
   not provide all the required information.

   There is a need to have a token that will allow the mapping between a
   single RTP stream configuration in an m-line to the actual RTP media
   stream.  For example, stream1 is the RTP stream from the left camera
   and stream2 is the RTP stream from the right camera and stream3 is
   the FEC stream that protects both streams.

   There is also a need for a token that will allow the offerer to
   correlate a received RTP stream to an SDP m-line before receiving the
   answer from the remote side.

   In order to address these requirements this document defines an SDP
   token attribute appId that provides a level of indirection for the
   binding.  The actual binding is done in RTP by associating the appId
   with an SSRC using a new RTCP SDES message and a new RTP header
   extension that define the mapping from appId to a specific SSRC.
   Having the binding in RTP/RTCP allows the RTP layer to change the
   SSRC of a media stream by sending a new binding message (SDES an RTP
   header extension) without a need to have an SDP level offer/answer



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   exchange.

   For the case when the offerer receives an RTP stream before the SDP
   answer, we define a new optional attribute recv-appId to be used for
   correlating this received RTP stream.

   Note: the name appId does not represent the token functionality very
   well.  We are looking for a better name (SSID source stream ID was
   proposed on the mailing list but is also well know for wifi networks)

3.1.  appId token

   AppId is a general-purpose token associated with an RTP stream,
   allowing the binding of the SDP representation to an SSRC.

   The token is chosen by the sender, and represents the RTP stream that
   will be sent to the receiver.

   The proposed token can be sent using SDP, RTCP SDES messages
   [RFC3550], or an RTP header extension [RFC5285]

   The SSRC mapping may be available to the receiver when receiving the
   RTP stream through the RTP header extension, but may also be
   available ahead of time via an RTCP SDES message conveyed before the
   source starts sending, even if the receiver has not seen any RTP
   packets from this source (as in a multipoint conference).

   The receiver can receive new sources that may be of two kinds.

   o  A new RTP stream replacing an existing RTP stream, in which case
      the AppId of the replaced RTP stream will be assigned to the new
      SSRC.
   o  A new RTP stream requiring a different AppId, for example, when
      adding a presentation stream to an existing call with two video
      cameras from a room.

   The solution supports an RTP session as described using SDP.  The RTP
   session may use Bundle [I-D.ietf-mmusic-sdp-bundle-negotiation] with
   more than one m-line.  In this case, if the SSRCs of all RTP streams
   are not known in advance, the AppIds associated with each m-line need
   to be available to the media receiver in order to map each SSRC to a
   specific m-line configuration.  The appIds MUST be unique in an SDP
   session.

   Editor Note (is this required?): It is preferable that they will be
   unique in an RTP session which is not a problem in a point to point
   call or in a multipoint conference with a central signaling point.




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   The document defines a new SDP media level attribute a=appId that can
   be used to list all the appIds that an application may use.

   The appId syntax provides a token identifier.  Each value of the
   AppId maps to one SSRC at a time.  When a new SSRC is mapped to an
   existing AppId using an RTP header extension or SDES message, it
   replaces the previous RTP stream for this application usage.

   The definition is

   a=appId:token

   a=appId:token <attribute>

   a=appId:token <attribute>:<value>

   The formal representation of the appId token is:
      appId-attribute = "appId:" token *[WSP attribute]
      attribute =/ appId-attr
      ; The base definition of "attribute" is in [RFC4566].
      ; (It is the content of "a=" lines.)
      ; WSP and DIGIT defined in [RFC5234]
      ; token defined in [RFC4566]

   AppId attributes are not defined in this specification but are
   provided for future extensibility.  (TODO: define an IANA registry
   for them.)

   Examples:

   The SDP offer specifies an appId that will be used for mapping to
   specific SSRCs.  The example shows two RTP streams having different
   content [RFC4796] with the same payload type number.  The appId can
   be used to identify the content of the RTP stream.

      a=group:BUNDLE m1 m2
      m=video 49200 RTP/AVP 99
      a=rtpmap:99 H264/90000
      a=mid:m1
      a=content:main
      a=appId:2
      m=video 49200 RTP/AVP 99
      a=rtpmap:99 H264/90000
      a=mid:m2
      a=content:alt
      a=appId:3

   The second example is when the application usage of the RTP stream is



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   specified using SDP to specify different content [RFC4796] , and each
   RTP stream has a Retransmission stream.  The media receiver can map
   the received SSRC of the RTP stream or the retransmission to the
   specific content based on the appId.

      a=group:BUNDLE m1 m2
      m=video 49200 RTP/AVP 97,98
      a=rtpmap:98 H264/90000
      a=mid:m1
      a=content:main
      a=rtpmap:97 rtx/90000
      a=fmtp:97 apt=98;rtx-time=3000
      a=appId:2
      a=appId:3
      m=video 49200 RTP/AVP 97,98
      a=rtpmap:98 H264/90000
      a=mid:m2
      a=content:alt
      a=rtpmap:97 rtx/90000
      a=fmtp:97 apt=98;rtx-time=3000
      a=appId:4
      a=appId:5

   In this example, the SDP signaling does not provide enough
   information to establish which appId value will be used for the H.264
   encoded stream, and which will be used for the RTX retransmission
   stream.  However, it does establish the relationships among the
   streams identified by appId values, allowing the receiver to properly
   associate the related streams once it receives them.

3.1.1.  RTCP SDES message

   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
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |    AppId = XXX    |     length    |AppId token
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |   ....

   This AppId is the same token as defined in the new SDP attribute 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.

   Editor Note: Need guidance on how often the SDES message should be



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   sent.

3.1.2.  RTP Header Extension

   The Application ID 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:AppId

   Packets tagged by the sender with the AppId 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=1            |   Len-1              |    AppId
        +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
        |  AppID   ..............       |
        +-+-+-+-+-+-+-+-+

   To add or modify the AppId by an intermediary can be an expensive
   operation, particularly if SRTP is used to authenticate the packet.
   Modification to the contents of the RTP header requires a re-
   authentication of the complete packet, and this could prove to be a
   limiting factor in the throughput of a multipoint device.

   There is no need to send the AppId header extension with all RTP
   packets.  Senders MAY choose to send it only when a new SSRC is sent,
   or when an SSRC changes its association to an AppId.  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.  For
   codecs with decoder refresh points (such as I-Frames in video
   codecs), senders also SHOULD send the AppId header extension along
   with the packets carrying the decoder refresh.

3.1.3.  recv-appId

   An offer may include a recv-appId attribute allowing the offerer to
   request from the answerer to use this token for the RTP stream sent
   from the answerer for a sendrecv or recvonly RTP stream.  This is
   important in order to support early media from the answerer that may
   be received by the offerer before the answer SDP arrives.

   The answerer should use the recv-appId as the token in the RTCP SDES
   and RTP header extension for the RTP stream sent to the offerer.




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   The formal representation of the appId token is:
      appId-attribute = "recv-appId:" token
      ; The base definition of "attribute" is in [RFC4566].
      ; (It is the content of "a=" lines.)

   An example of an offer using bundle with two video streams using the
   same payload type number:

      a=group:BUNDLE m1 m2
      m=video 49200 RTP/AVP 99
      a=rtpmap:99 H264/90000
      a=mid:m1
      a=content:main
      a=appId:2
      a=recv-appId:4
      m=video 49200 RTP/AVP 99
      a=rtpmap:99 H264/90000
      a=mid:m2
      a=content:alt
      a=appId:3
      a=recv-appId:5


4.  Using Application ID token in Offer / Answer

   The appId may be used in offer answer.  Some use cases are provided.
   They only show part of the SDP that can demonstrate the usage.

   A simple case is when each SDP m-line describes one RTP stream and
   the m-lines are bundled .  The recv-appId is offered so when the
   offerer sees an RTP stream with appId token value 10 it knows it is
   the main video.

   The offer is:
      a=group:BUNDLE m1 m2
      m=video 49200 RTP/AVP 98
      a=rtpmap:98 H264/90000
      a=mid:m1
      a=content:main
      a=appId:2
      a=recv-appId:10
      m=video 49200 RTP/AVP 99
      a=rtpmap:99 H264/90000
      a=mid:m2
      a=content:alt
      a=appId:3





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      a=recv-appId:20

   An offerer sending a BUNDLE group MUST indicate at least one recv-
   appId for every RTP m=line in the group on which it could receive
   media (i.e., for every m-line which is marked a=sendrecv (possibly
   implicitly) or a=recvonly.  On m-lines whose configuration is such
   that multiple packet streams are expected to be sent simultaneously
   (e.g., one with rtx or fec payload types configured), the offerer
   MUST specify as many recv-appId values as the number of simultaneous
   packet streams.

   Additionally, an offerer sending a BUNDLE group MUST indicate at
   least one appId for every m= line on which it expects to send media
   (i.e., every a=sendrecv or a=sendonly m= line), and MUST send
   multiple appIds for m= lines on which it expects to send multiple
   packet streams simultaneously.

   An answerer, receiving an offer containing appId or recv-appId
   attributes, MUST respond with mirrored recv-appId and appId values
   for the subset of m= lines and packet streams indicated in the
   answer, maintaining the same recv-appId and appId values.

   In subsequent offers, appId and recv-appId values SHOULD be
   maintained per m=line unless the offer is recycling the m=line for a
   fundamentally new purpose, in which case new appId and recv-appId
   values SHOULD be used.  AppId and recv-appId values MUST NOT be
   reused, in the same session, for a different m= line than the one to
   which they were originally assigned, unless at least two times the
   maximum segment lifetime (MSL) has passed since the previous offer/
   answer exchange in which the values were used.


5.  Other Considerations

   During the work on the document we identified that there are two
   different problems that we were trying to address.  One has to do
   with mapping the RTP stream to an m-line addressed in the current
   version of the document.  The other problem is to provide semantics
   to an RTP stream description in the SDP description for the cases
   where the SSRC of the RTP stream is not known to the SDP signaling
   layer and we also identified issues not addressed by the SSRC
   attribute [RFC5576].

   A single RTP media stream can be identified in SDP by using the SSRC
   attribute [RFC5576].  Relations between RTP streams in the same
   session can be specified using the ssrc-group attribute [RFC5576].
   Using the SSRC to identify RTP streams in an SDP session assumes that
   this information is available to the SDP signaling layer.  The SSRC



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   is RTP layer information and may change in the RTP layer during a
   session.

   Support of FEC, SVC and simulcast brings more requirements as
   explained using the following examples.

   The following example is of a unified plan
   [I-D.roach-mmusic-unified-plan] offer of one audio source and one
   video source.  The video source includes two SVC RTP streams a base
   layer and an enhancement layer.  There are also two FEC options:

   Base layer S1 is protected by FEC repair stream R1

   Base Layer S1 and Enhancement layer S2 are protected by FEC repair
   stream R2.

   This enables the answer to select the base layer with R1 or the Base
   + enhancement layers both protected by R2.

   This example uses the SSRC and SSRC-GROUP attributes which requires
   the pre-knowledge of the SSRCs that are RTP layer values.

   SDP Offer:
      v=0
      o=- 20518 0 IN IP4 198.51.100.1
      s=FEC Grouping Semantics for SSRC Multiplexing
      t=0 0
      c=IN IP4 203.0.113.1
      a=group:BUNDLE m1 m2
      m=audio 56600 RTP/SAVPF 0 109
      a=msid:ma ta
      a=mid:m1
      a=ssrc:53280
      a=rtpmap:0 PCMU/8000
      a=rtpmap:109 opus/48000
      m=video 56602 RTP/AVPF 100 101 110 111 - Main camera
      a=msid:ma tb
      a=mid:m2
      a=rtpmap:100 H264/90000 - Base layer
      a=rtpmap:101 H264-SVC/90000 - Enhancement layer.
      a=depend:101 lay L1:100 - dependencies
      a=rtpmap:110 1d-interleaved-parityfec/90000
      a=fmtp:110 L=5; D=10; repair-window=200000
      a=rtpmap:111 1d-interleaved-parityfec/90000
      a=fmtp:111 L=10; D=10; repair-window=400000
      a=ssrc:1000 cname:MSTFEC@example.com





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      a=ssrc:1010 cname:MSTFEC@example.com
      a=ssrc:2110 cname:MSTFEC@example.com
      a=ssrc:2120 cname:MSTFEC@example.com
      a=ssrc-group:FEC-FR 1000 2110
      a=ssrc-group:FEC-FR 1000 1010 2120
      a=ssrc-group:DDP 1000 1010

   In this case all video streams are from the same source and can be
   described using a single m-line.  The grouping relations are
   specified using the SSRCs values that need to be available in the
   offer.  It is also not clear based on the offer which rtpmap line
   corresponds to each of the a=ssrc lines, e.g. which rtpmap line will
   be sent using ssrc = 2110.  The answerer can deduce the information
   based on analyzing the ssrc-group information but there can be case
   that it will not be possible..

   Note: These issues will be addressed in a separate document based on
   the feedback from the working group that even though these are real
   issues they should have a separate solution in order to differentiate
   between the token and the semantics needed.


6.  Acknowledgements

   Place Holder


7.  IANA Considerations

   TBD


8.  Security Considerations

   TBD.


9.  References

9.1.  Normative References

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

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.




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   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

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

9.2.  Informative References

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
              "A Taxonomy of Grouping Semantics and Mechanisms for Real-
              Time Transport Protocol (RTP) Sources",
              draft-ietf-avtext-rtp-grouping-taxonomy-01 (work in
              progress), February 2014.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Multiplexing Negotiation Using Session Description
              Protocol (SDP) Port Numbers",
              draft-ietf-mmusic-sdp-bundle-negotiation-06 (work in
              progress), April 2014.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-09 (work
              in progress), February 2014.

   [I-D.roach-mmusic-unified-plan]
              Roach, A., Uberti, J., and M. Thomson, "A Unified Plan for
              Using SDP with Large Numbers of Media Flows",
              draft-roach-mmusic-unified-plan-00 (work in progress),
              July 2013.

   [I-D.westerlund-avtcore-rtp-simulcast]
              Westerlund, M., Lindqvist, M., and F. Jansson, "Using
              Simulcast in RTP Sessions",
              draft-westerlund-avtcore-rtp-simulcast-03 (work in
              progress), October 2013.

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

   [RFC3388]  Camarillo, G., Eriksson, G., Holler, J., and H.
              Schulzrinne, "Grouping of Media Lines in the Session
              Description Protocol (SDP)", RFC 3388, December 2002.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session



Even, et al.            Expires October 13, 2014               [Page 13]


Internet-Draft            SDP application token               April 2014


              Description Protocol", RFC 4566, July 2006.

   [RFC4756]  Li, A., "Forward Error Correction Grouping Semantics in
              Session Description Protocol", RFC 4756, November 2006.

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

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.


Authors' Addresses

   Roni Even
   Huawei Technologies
   Tel Aviv,
   Israel

   Email: roni.even@mail01.huawei.com


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

   Email: jonathan@vidyo.com


   Qin Wu
   Huawei Technologies

   Email: bill.wu@huawei.com













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