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Versions: 00 01 02

Network Working Group                                      M. Westerlund
Internet-Draft                                                 B. Burman
Intended status: Standards Track                              F. Jansson
Expires: January 17, 2013                                       Ericsson
                                                           July 16, 2012


    Multiple Synchronization sources (SSRC) in RTP Session Signaling
                  draft-westerlund-avtcore-max-ssrc-02

Abstract

   RTP has always been a protocol that supports multiple participants
   each sending their own media streams in an RTP session.
   Unfortunately, many implementations are designed only for point to
   point voice over IP with a single source in each end-point.  Even
   client implementations aimed at video conferences have often been
   built with the assumption around central mixers that only deliver a
   single media stream per media type.  Thus any application that wants
   to allow for more advanced usage, where multiple media streams are
   sent and received by an end-point, has an issue with legacy
   implementations.  This document describes the problem and proposes a
   signalling solution for how to use multiple SSRCs within one RTP
   session and at the same time handle the legacy issues.

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 January 17, 2013.

Copyright Notice

   Copyright (c) 2012 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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   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.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  Background . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Requirements Language  . . . . . . . . . . . . . . . . . .  4
     2.2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Multiple Streams Issues  . . . . . . . . . . . . . . . . . . .  5
     3.1.  Legacy Behaviors . . . . . . . . . . . . . . . . . . . . .  5
     3.2.  Receiver Limitations . . . . . . . . . . . . . . . . . . .  7
     3.3.  Transmission Declarations  . . . . . . . . . . . . . . . .  7
   4.  Multiple Streams SDP Extension . . . . . . . . . . . . . . . .  8
     4.1.  Signaling Support for Multiple Streams . . . . . . . . . .  8
     4.2.  Declarative Use  . . . . . . . . . . . . . . . . . . . . .  9
     4.3.  Use in Offer/Answer  . . . . . . . . . . . . . . . . . . . 10
     4.4.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . 10
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 11
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 11
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 12
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13



















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1.  Introduction

   This document discusses the issues of non basic usage of RTP
   [RFC3550] where there is multiple media sources sent over an RTP
   session using the SSRC source identifier to distinguish between the
   sources.  This include multiple sources from the same end-point,
   multiple end-points each having a source, or an application that
   sends or receive multiple encodings of a particular media source
   using multiple SSRCs.

1.1.  Background

   RTP sessions are a concept which most fundamental part is an SSRC
   space.  This space can encompass a number of network nodes and
   interconnected transport flows between these nodes.  Each node may
   have zero, one or more source identifiers (SSRCs) used to either
   identify a real media source such as a camera or a microphone, a
   conceptual source (like the most active speaker selected by an RTP
   mixer that switches between incoming media streams based on the media
   stream or additional information), or simply as an identifier for a
   receiver that provides feedback and reports on reception.  There are
   also RTP nodes, like translators that are manipulating data,
   transport or session state without making their presence aware to the
   other session participants.

   RTP was designed to support multiple participants in a session from
   the beginning.  This was not restricted to multicast as many believe
   but also unicast using either multiple transport flows below RTP or a
   network node that redistributes the RTP packets, either unchanged in
   the form of a transport translator (relay) or modified in an RTP
   mixer.  There is also the case where a single end-point have multiple
   media sources, like multiple cameras or microphones.

   However, the most common use cases for RTP have been point to point
   Voice over IP (VoIP) or streaming applications where there have
   commonly not been more than one media source per end-point.  Even in
   conferencing applications, especially voice only, the conference
   focus or bridge have provided a single stream being a mix of the
   other participants to each participant.  Thus there has been little
   need for handling multiple SSRCs in implementations.  This has
   resulted in an installed legacy base that is not fully RTP
   specification compliant and will have different issues if they
   receive multiple SSRCs of media, either simultaneously or in
   sequence.  These issues will manifest themselves in various ways,
   either by software crashes, or simply in limited functionality, like
   only decoding and playing back the first or latest SSRC received and
   discarding any other SSRCs.




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   The signaling solutions around RTP, especially the SDP [RFC4566]
   based, have not considered the fundamental issues around an RTP
   session's theoretical support of up to 4 billion plus sources all
   sending media.  No end-point has infinite processing resources to
   decode and mix any number of media sources.  In addition the memory
   for storing related state, especially decoder state is limited, and
   the network bandwidth to receive multiple streams is also limited.
   Today, the most likely limitations are processing and network
   bandwidth although for some use cases memory or other limitations may
   also exist.  The issue is that a given end-point will have some
   limitations in the number of streams it simultaneously can receive,
   decode and playback.  These limitations need to be possible to expose
   and enabling the session participants to take them into account.

   In similar ways there is a need for an end-point to express if it
   intends to produce one or more media streams in an RTP session.
   Todays SDP signaling support for this is basically the directionality
   attribute which indicates an end-point intent to send media or not.
   There is however no way to indicate how many media streams will be
   sent.

   Taking these things together there exist a clear need to enable the
   usage of multiple simultaneous media streams within an RTP session in
   a way that allows a system to take legacy implementations into
   account in addition to negotiate the actual capabilities around the
   multiple streams in an RTP session.


2.  Definitions

2.1.  Requirements Language

   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].

2.2.  Terminology

   The following terms and abbreviations are used in this document:

   Encoding:  A particular encoding is the choice of the media encoder
      (codec) that has been used to compress the media, the fidelity of
      that encoding through the choice of sampling, bit-rate and other
      configuration parameters.







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   Different encodings:  An encoding is different when some parameter
      that characterize the encoding of a particular media source has
      been changed.  Such changes can be one or more of the following
      parameters; codec, codec configuration, bit-rate, sampling.


3.  Multiple Streams Issues

   This section attempts to go a bit more in depth around the different
   issues when using multiple media streams in an RTP session to make it
   clear that although in theory multi-stream applications should
   already be possible to use, there are good reasons to create
   extensions for signaling.  In addition, the RTP specification could
   benefit from clarifications on how certain mechanisms should be
   working when an RTP session contains more than two SSRCs.

3.1.  Legacy Behaviors

   It is a common assumption among many applications using RTP that they
   do not have a need to support more than one incoming and one outgoing
   media stream per RTP session.  For a number of applications this
   assumption has been correct.  For VoIP and Streaming applications it
   has been easiest to ensure that a given end-point only receives
   and/or sends a single stream.  However, all end-points should support
   a source changing SSRC value during a session, e.g due to SSRC value
   collision between participants in a conference and the requirement to
   always use unique SSRC values.

   Some RTP extension mechanisms require the RTP stacks to handle
   additional SSRCs, like SSRC multiplexed RTP retransmission described
   in [RFC4588].  However, that still has only required handling a
   single media decoding chain.

   There are however applications that clearly can benefit from
   receiving and using multiple media streams simultaneously.  A very
   basic case would be T.140 conversational text, where the text
   characters are transmitted as a real-time media stream as you type.
   When used in a multi-party chat scenario, an end-point can receive
   input from multiple sending end-points where the T.140 RTP Payload
   Format [RFC4103] text media is both low bandwidth and where there is
   no obvious method to algorithmically distinguish between multiple
   sources of text, making simple multiplex and identification of
   separate sources through an identifier (SSRC) a good choice.

   An RTP session that contains an end-point with more than two SSRCs
   actively sending media streams puts some requirements on the
   receiving client, which is not necessarily fulfilled by a legacy
   client:



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   1.  The receiving client needs to handle receiving more than one
       stream simultaneously rather than replacing the already existing
       stream with the new one.

   2.  Be capable of decoding multiple streams simultaneously.

   3.  Be capable of rendering multiple streams simultaneously.

   An application using multiple streams may be very similar to existing
   one media stream applications at signaling level.  To avoid
   connecting two different implementations, one that is built to
   support multiple streams and one that is not, it is important that
   the capabilities are signaled.  This also enables a particular
   application to separate legacy clients that do not signal it
   explicitly from the ones that do, and take appropriate measures for
   the legacy application.  Due to that there exist both legacy
   applications that only handle a single stream and applications that
   handle multiple streams, a single authoritative interpretation cannot
   be defined by this specification.

   There exist many models for legacy behaviors when it comes to
   handling multiple SSRCs.  This briefly describes some of the more
   common ones:

   Single SSRC per Direction:  As previously discussed there exist a
      large number of applications where each end-point has only a
      single SSRC and the number of end-points in the RTP session are
      only two.  This class of applications needs to have a legacy
      fallback behavior that provide only a single SSRC, or issues will
      likely occur.

   Multiple SSRCs per Direction Single Media Stream:  There exist some
      applications that uses multiple concurrent SSRCs but in the end
      only consumes a single media stream.  This include the streaming
      applications using RTP retransmission with SSRC multiplexing, but
      also applications switching between sources although only
      consuming one at a time.  When an offer or answer without the
      signalling extension is received, the end-point can potentially
      determine this by inspecting payload types in use and correctly
      determine the legacy behavior.  In some cases it must be
      determined through application context or SDP external signaling
      indications.

   Multi-Party Each Using Multiple SSRCs:  Multicast applications
      commonly have multiple SSRCs, if for no other reason than the
      existence of multiple end-points in the same RTP session.  Similar
      considerations exist in multi-party applications that uses central
      nodes.  There may be no indications in the SDP regarding how the



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      application will handle multiple concurrent SSRCs and the
      expectancy to decode these.  Thus also this legacy behavior must
      commonly be determined from the application context and external
      signaling.

3.2.  Receiver Limitations

   An RTP end-point that intends to process the media in an RTP session
   needs to have sufficient resources to receive and process all the
   incoming streams.  It is extremely likely that no receiver is capable
   to handle the theoretical upper limit of more than 4 billion media
   sources in an RTP session.  Instead, one or more properties will
   limit the end-points' capabilities to handle simultaneous media
   streams.  These properties are for example memory, processing,
   network bandwidth, memory bandwidth, or rendering estate to mention a
   few possible limitations.  Those limits in number of simultaneous
   streams may also differ depending on what codec (payload type) that
   is used.

   We have also considered the issue of how many simultaneous non-active
   sources an end-point can handle.  We cannot see that inactive media
   sending SSRCs result in significant resource consumption and there
   should thus be no need to limit them.

   A potential issue that needs to be acknowledged is where a limited
   set of simultaneously active sources varies within a larger set of
   session members.  As each media decoding chain may contain state, it
   is important that a receiver can flush a decoding state for an
   inactive source and if that source becomes active again it does not
   assume that this previous state exists.  Thus, we see need for a
   signaling solution that allows a receiver to indicate its upper limit
   in terms of capability to handle simultaneous media streams.  We see
   little need for an upper limitation of RTP session members.
   Applications will need to account for its own capability to use
   different codecs simultaneously when choosing general and payload
   specific limits.

3.3.  Transmission Declarations

   In an RTP based system where an end-point may either be legacy or has
   an explicit upper limit in the number of simultaneous streams, one
   will encounter situations where the end-point can not receive and
   process all simultaneous active streams in the session.  Instead, the
   sending end-points or central nodes, like RTP mixers, will have to
   provide the end-point with a selected set of streams based on various
   metrics, such as most active, most interesting, or user selected.  In
   addition, the central node may combine multiple media streams using
   mixing or composition into a new media stream to enable an end-point



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   to get sufficient source coverage in the session, despite existing
   limitations.

   For such a system to be able to correctly determine the need for
   central processing, the capabilities needed for such a central
   processing node, and the potential need for an end-point to do sender
   side limitations, it is necessary for an end-point to declare how
   many simultaneous streams it may send.  Thus, enabling negotiation of
   the number of streams an end-point sends.  The limit in number of
   simultaneous streams may also differ depending on what codec (payload
   type) that is used.


4.  Multiple Streams SDP Extension

   This section describes an extension of the media-level SDP attributes
   to support signaling of the end point's multiple stream capabilities.

4.1.  Signaling Support for Multiple Streams

   A solution to the issues described in the previous section needs to:

   o  Enable signaling between the RTP sender and receiver how many
      simultaneous RTP streams that can be handled, including a
      possibility to specify a different number of streams depending on
      what codec (payload type) that is used.

   o  Be able to handle the case where the number of RTP streams that
      can be sent from a client do not match the number of streams that
      can be received by the same client.

   It is also a requirement that a multiple streams capable RTP sender
   MUST be able to adapt the number of sent streams to the RTP receiver
   capability.

   For this purpose and for use in SDP, two new media-level SDP
   attributes are defined, max-send-ssrc and max-recv-ssrc, which can be
   used independently to establish a limit to the number of
   simultaneously active SSRCs for the send and receive directions,
   respectively.  Active SSRCs are the ones counted as senders according
   to [RFC3550], i.e. they have sent RTP packets during the last two
   regular RTCP reporting intervals.  Alternatively, if the end-points
   supports PAUSE and RESUME signaling
   [I-D.westerlund-avtext-rtp-stream-pause], any stream that the media
   sender sent a PAUSED indication for can be regarded as non-Active and
   can immediately be replaced by another SSRC, considering the limits
   established by this specification, including possible changes in the
   applicable limit required by a change of codec.



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   The syntax for the attributes is in ABNF [RFC5234]:

   max-ssrc    = "a="( "max-send-ssrc:" / "max-recv-ssrc:" ) alt-list
   alt-list    = alt-set *(WSP alt-set)
   alt-set     = "{" alt *("&" alt)) "}"
   alt         = pt ":" limit
   pt          = ( pt-list / pt-wildcard )
   pt-list     = ( pt-value / pt-range ) *(","( pt-value / pt-range ))
   pt-value    = 1*3DIGIT
   pt-range    = pt-value "-" pt-value
   pt-wildcard = "*"
   limit       = 1*8DIGIT
   ; WSP and DIGIT defined in [RFC5234]


   A Payload Type (PT)-agnostic upper limit to the total number of
   simultaneous SSRCs that can be sent or received in this RTP session
   is signaled with a "*" instead of the PT number.  A value of 0 MAY be
   used as maximum number of SSRC, but it is then RECOMMENDED that this
   is also reflected using the sendonly or recvonly attribute.

   A PT-specific upper limit to the total number of simultaneous SSRCs
   in the RTP session with that specific PT is signaled with a defined
   PT (static, or dynamic through rtpmap).  Multiple, alternative sets
   of PT and limit MAY be specified on the same line, where each set
   indicates the codec-dependent limit when a certain PT is used.  A
   combination of a comma-separated list of PT and a range of PT sharing
   a single limit MAY be used.  Within the alternative set, there MAY be
   a specification of limits valid when different PT are used
   simultaneously.  Any PT-agnostic specification on a line MUST be
   interpreted as valid for any PT that was not included within an
   explicit limit within that alternative set.  Multiple lines with max-
   send-ssrc or max-recv-ssrc attributes specifying a single PT MAY be
   used, but MUST NOT contain conflicting limits.  PT values that are
   not defined in the media block MUST be ignored.

   When max-send-ssrc or max-recv-ssrc are not included in the SDP, it
   is to be interpreted in the application's context.  The default
   number of allowed SSRCs can vary depending on the type of
   application.

4.2.  Declarative Use

   When used as a declarative media description, the specified limit in
   max-send-ssrc indicates the maximum number of simultaneous streams of
   the specified payload types that the configured end-point may send at
   any single point in time.  Similarly, max-recv-ssrc indicates the
   maximum number of simultaneous streams of the specified payload types



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   that may be sent to the configured end-point.  Payload-agnostic
   limits MAY be used with or without additional payload-specific
   limits.

4.3.  Use in Offer/Answer

   When used in an offer [RFC3264], the specified limits indicate the
   agent's intent of sending and/or capability of receiving that number
   of simultaneous SSRCs.  An offerer supporting this specification MUST
   include both attributes for sendrecv media streams, even if one or
   both has a value of 1.  For sendonly or recvonly m= blocks, the one
   matching the Offer/Answer agent's role MUST be included when using
   this extension, and the other directionality MAY be included for
   informational purpose, if bi-directionality can potentially be used
   in the future for this m= block.  The answerer MUST reverse the
   directionality of recognized attributes such that max-send-ssrc
   becomes max-recv-ssrc and vice versa.  The answerer SHOULD modify the
   offered limits in the answer to suit the answering client's
   capability and intentions.  A sender MUST NOT send more simultaneous
   streams of the specified payload type(s) than the receiver has
   indicated ability to receive, taking into account also any payload
   type-agnostic limit.

   In case an answer fails to include any of the limitation attributes,
   the agent is RECOMMENDED to have defined a suitable interpretation
   for the application context.  This may be the choice of being capable
   of supporting only a single stream (SSRC) in the direction for which
   attributes are missing.  It may also indicate multiple SSRC support
   depending on the application.  In case the offer lacks both max-send-
   ssrc and max-recv-ssrc, they MUST NOT be included in the answer.

4.4.  Examples

   The SDP examples below are not complete.  Only relevant parts have
   been included, for brevity and readability.

     m=video 49200 RTP/AVP 99
     a=rtpmap:99 H264/90000
     a=max-send-ssrc:{*:2}
     a=max-recv-ssrc:{*:4}

   An offer with a stated intention of sending 2 simultaneous SSRCs and
   a capability to receive 4 simultaneous SSRCs.

     m=video 50324 RTP/AVP 96 97
     a=rtpmap:96 H264/90000
     a=rtpmap:97 H263-2000/90000
     a=max-recv-ssrc:{96:2&97:3} {96:1&97:4} {97:5}



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     a=max-send-ssrc:{* 1}

   An offer to receive at most 5 SSRCs, at most 2 of which using payload
   type 96 and the rest using payload type 97.  The "max- send-ssrc" is
   used to explicitly indicate the value of 1.

     m=video 50324 RTP/AVP 96 97 98
     a=rtpmap:96 H264/90000
     a=rtpmap:97 H263-2000/90000
     a=rtpmap:98 H263/90000
     a=max-recv-ssrc:{96:2&97:3} {96-97:2&98:1} {96,98:2&97:1}
     a=max-recv-ssrc:{96,98:1&97:3} {96:1&97-98:2} {96-97:1&98:3}
     a=max-recv-ssrc:{96:1&98:4} {97:3&98:2} {97:2&98:3} {97:1&98:4}
     a=max-recv-ssrc:{98:5}
     a=max-send-ssrc:{* 1}

   An offer to receive at most 5 SSRCs, at most 2 of which using payload
   type 96, at most 3 of which using payload type 97, and at most 5
   using payload type 98.  Since there are many combinations, more than
   one "max-recv-ssrc" line is used, which is OK since no specifications
   on those lines are in conflict.


5.  IANA Considerations

   This document registers two media level SDP attributes.


6.  Security Considerations

   The SDP attributes defined in this document "a=max-recv-ssrc" and
   "a=max-send-ssrc" signals capabilities of the end-point.  Thus they
   are vulnerable to attacks.  The primary security concerns would be
   with third parties that modifies the values of the attributes or
   inserts the attributes in a signalling context.  Thus changing the
   peers view of the others peers capabilities and proposals.  A
   modification reducing either of send or receive values will degrade
   the service, potentially preventing the service all together.
   Increasing the value or inserting the attribute with a value
   different from 1 have the potential of being even more effective.  It
   can result in that an end-point that only supports a single stream
   will be sent multiple streams.  First of all, this potentially
   exposes software flaws regarding handling of multiple streams, thus
   causing crashes, less severe it can cause media degradation as the
   receiving entity flaps between media streams, or plays only a single
   one, where the other side assumes both will be played.  In addition,
   negotiation of several streams has transport impact, potentially
   increasing the bit-rate consumed towards the end-point, and in



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   addition forcing an adaptation response over a limited path, thus
   degrading the media stream the end-point may play out.

   To prevent third party manipulation of the SDP, it should be source
   authenticated and integrity protected.  The solution suitable for
   this depends on the signalling protocol being used.  For SIP S/MIME
   [RFC3261] is the ideal, and hop by hop TLS provides at least some
   protection, although not perfect.  For SDPs retrieved using RTSP
   DESCRIBE [RFC2326], TLS would be the RECOMMENDED solution.


7.  References

7.1.  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.

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

7.2.  Informative References

   [I-D.westerlund-avtext-rtp-stream-pause]
              Akram, A., Burman, B., Grondal, D., and M. Westerlund,
              "RTP Media Stream Pause and Resume",
              draft-westerlund-avtext-rtp-stream-pause-02 (work in
              progress), July 2012.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

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

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, June 2005.




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

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.


Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


   Bo  Burman
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 13 11
   Email: bo.burman@ericsson.com


   Fredrik Jansson
   Ericsson
   Farogatan 6
   Kista,   SE-164 80
   Sweden

   Phone: +46 10 719 00 00
   Fax:
   Email: fredrik.k.jansson@ericsson.com
   URI:











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