Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Expires: January 18, 2007                                  July 17, 2006

        RTP and the Datagram Congestion Control Protocol (DCCP)

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Copyright Notice

   Copyright (C) The Internet Society (2006).


   The Real-time Transport Protocol (RTP) is a widely used transport for
   real-time media on IP networks.  The Datagram Congestion Control
   Protocol (DCCP) is a newly defined transport protocol that provides
   desirable services for real-time applications.  This memo specifies a
   mapping of RTP onto DCCP, along with associated signalling, such that
   real-time applications can make use of the services provided by DCCP.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Rationale  . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Conventions Used in this Memo  . . . . . . . . . . . . . . . .  4
   4.  RTP over DCCP: Framing . . . . . . . . . . . . . . . . . . . .  4
     4.1.  RTP Data Packets . . . . . . . . . . . . . . . . . . . . .  4
     4.2.  RTP Control Packets  . . . . . . . . . . . . . . . . . . .  5
     4.3.  Multiplexing Data and Control  . . . . . . . . . . . . . .  6
     4.4.  RTP Sessions and DCCP Connections  . . . . . . . . . . . .  7
     4.5.  RTP Profiles . . . . . . . . . . . . . . . . . . . . . . .  8
   5.  RTP over DCCP: Signalling using SDP  . . . . . . . . . . . . .  8
     5.1.  Protocol Identification  . . . . . . . . . . . . . . . . .  8
     5.2.  Service Codes  . . . . . . . . . . . . . . . . . . . . . .  9
     5.3.  Connection Management  . . . . . . . . . . . . . . . . . . 10
     5.4.  Example  . . . . . . . . . . . . . . . . . . . . . . . . . 10
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 11
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 12
   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 12
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 12
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 13
   Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 15
   Intellectual Property and Copyright Statements . . . . . . . . . . 16

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1.  Introduction

   The Real-time Transport Protocol (RTP) [1] is widely used in video
   streaming, telephony, and other real-time networked applications.
   RTP can run over a range of lower-layer transport protocols, and the
   performance of an application using RTP is heavily influenced by the
   choice of lower-layer transport.  The Datagram Congestion Control
   Protocol (DCCP) [2] is a newly specified transport protocol that
   provides desirable properties for real-time applications running on
   unmanaged best-effort IP networks.  This memo describes how RTP can
   be framed for transport using DCCP, and discusses some of the
   implications of such a framing.  It also describes how the Session
   Description Protocol (SDP) [3] can be used to signal such sessions.

   The remainder of this memo is structured as follows: we begin with a
   rationale for the work in Section 2, describing why a mapping of RTP
   onto DCCP is needed.  Following a description of the conventions used
   in this memo in Section 3, the specification begins in Section 4 with
   the definition of how RTP packets are framed within DCCP; associated
   signalling is described in Section 5.  We conclude with a discussion
   of security considerations in Section 6, and IANA considerations in
   Section 7.

2.  Rationale

   With the widespread adoption of RTP have come concerns that many real
   time applications do not implement congestion control, leading to the
   potential for congestion collapse of the network [14].  The designers
   of RTP recognised this issue, stating that [4]:

      If best-effort service is being used, RTP receivers SHOULD monitor
      packet loss to ensure that the packet loss rate is within
      acceptable parameters.  Packet loss is considered acceptable if a
      TCP flow across the same network path and experiencing the same
      network conditions would achieve an average throughput, measured
      on a reasonable time-scale, that is not less than the RTP flow is
      achieving.  This condition can be satisfied by implementing
      congestion control mechanisms to adapt the transmission rate (or
      the number of layers subscribed for a layered multicast session),
      or by arranging for a receiver to leave the session if the loss
      rate is unacceptably high.

   While the goals are clear, the development of TCP friendly congestion
   control that can be used with RTP and real-time media applications is
   an open research question with many proposals for new algorithms, but
   little deployment experience.

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   Two approaches have been used to provide congestion control for RTP:
   1) develop new RTP profiles that incorporate congestion control; and
   2) provide mechanisms for running RTP over congestion controlled
   transport protocols.  The RTP Profile for TCP Friendly Rate Control
   [15] is an example of the first approach, extending the RTP packet
   formats to incorporate feedback information such that TFRC congestion
   control [16] can be implemented at the application level.  This
   approach has the advantage that congestion control can be added to
   existing applications, without needing operating system or network
   support, and offers flexibility to experiment with new congestion
   control algorithms as they are developed.  Unfortunately, there is
   also the consequent disadvantage that the complexity of implementing
   congestion control is passed onto the application author, a burden
   which many would prefer to avoid.

   The other approach is to run RTP on a lower-layer transport protocol
   that provides congestion control.  One possibility is to run RTP over
   TCP, as defined in [5], but the reliable nature of TCP and the
   dynamics of its congestion control algorithm make this inappropriate
   for most interactive real time applications (SCTP is inappropriate
   for similar reasons).  A better fit for such applications may be to
   run RTP over DCCP, since DCCP offers unreliable packet delivery and a
   choice of congestion control.  This gives applications the ability to
   tailor the transport to their needs, taking advantage of better
   congestion control algorithms as they come available, while passing
   complexity of implementation to the operating system.  If DCCP should
   come to be widely available, it is believed these will be compelling
   advantages.  Accordingly, this memo defines a mapping of RTP onto

3.  Conventions Used in this Memo

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [6].

4.  RTP over DCCP: Framing

   The following section defines how RTP and RTCP packets can be framed
   for transport using DCCP.  It also describes the differences between
   RTP sessions and DCCP connections, and the impact these have on the
   design of applications.

4.1.  RTP Data Packets

   Each RTP data packet MUST be conveyed in a single DCCP datagram.

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   Fields in the RTP header MUST be interpreted according to the RTP
   specification, and any applicable RTP Profile and Payload Format.
   Header processing is not affected by DCCP framing (in particular,
   note that the semantics of the RTP sequence number and the DCCP
   sequence number are not compatible, and the value of one cannot be
   inferred from the other).

   A DCCP connection is opened when an end system joins an RTP session,
   and it remains open for the duration of the session.  To ensure NAT
   bindings are kept open, an end system SHOULD send a zero length DCCP-
   Data packet once every 15 seconds during periods when it has no RTP
   data to send.  This removes the need for RTP no-op packets [17], and
   similar application level keep-alives, when using RTP over DCCP.

   RTP data packets MUST obey the dictates of DCCP congestion control.
   In some cases, the congestion control will require a sender to send
   at a rate below that which the payload format would otherwise use.
   To support this, an application should use either a rate adaptive
   payload format, or a range of payload formats (allowing it to switch
   to a lower rate format if necessary).  Details of the rate adaptation
   policy for particular payload formats are outside the scope of this

   TODO: provide more guidance on implementation of congestion control
   within an RTP application.

   DCCP allows an application to choose the checksum coverage, using a
   partial checksum to allow an application to receive packets with
   corrupt payloads.  Some RTP Payload Formats (e.g. [18]) can make use
   of this feature in conjunction with payload-specific mechanisms to
   improve performance when operating in environments with frequent non-
   congestive packet corruption.  If such a payload format is used, an
   RTP end system MAY enable partial checksums at the DCCP layer, in
   which case the checksum MUST cover at least the DCCP and RTP headers
   to ensure packets are correctly delivered.  Partial checksums MUST
   NOT be used unless supported by mechanisms in the RTP payload format.

4.2.  RTP Control Packets

   The RTP Control Protocol (RTCP) is used in the standard manner with
   DCCP.  RTCP packets are grouped into compound packets, as described
   in Section 6.1 of [1], and each compound RTCP packet is transported
   in a single DCCP datagram.

   The usual RTCP timing rules apply, with the additional constraint
   that RTCP packets MUST obey the DCCP congestion control algorithm
   negotiated for the connection.  This can prevent a participant from
   sending an RTCP packet at the expiration of the RTCP transmission

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   timer if there is insufficient network capacity available.  In such
   cases the RTCP packet is delayed and sent at the earliest possible
   instant when capacity becomes available.  The actual time the RTCP
   packet was sent is then used as the basis for calculating the next
   RTCP transmission time.

   RTCP packets comprise only a small fraction of the total traffic in
   an RTP session.  Accordingly, it is expected that delays in their
   transmission due to congestion control will not be common, provided
   the configured nominal "session bandwidth" (see Section 6.2 of [1])
   is in line with the bandwidth achievable on the DCCP connection.  If,
   however, the capacity of the DCCP connection is significantly below
   the nominal session bandwidth, RTCP packets may be delayed enough for
   participants to time out due to apparant inactivity.  In such cases,
   the session parameters SHOULD be re-negotiated to more closely match
   the available capacity, for example by performing a SIP re-invite
   with an updated "b=" line.

      Since the nominal session bandwidth is chosen based on media codec
      capabilities, a session where the nominal bandwidth is much larger
      than the available bandwidth will likely become unusable due to
      constraints on the media channel, and so require negotiation of a
      lower bandwidth codec, before it becomes unusable due to
      constraints on the RTCP channel.

   As noted in Section 17.1 of [2], there is the potential for overlap
   between information conveyed in RTCP packets and that conveyed in
   DCCP acknowledgement options.  In general this is not an issue since
   RTCP packets contain media-specific data that is not present in DCCP
   acknowledgement options, and DCCP options contain network-level data
   that is not present in RTCP.  Indeed, there is no overlap between the
   five RTCP packet types defined in the RTP specification [1] and the
   standard DCCP options [2].  There are, however, cases where overlap
   does occur: most clearly between the optional RTCP Extended Reports
   Loss RLE Blocks [19] and the DCCP Ack Vector option.  If there is
   overlap between RTCP report packets and DCCP acknowledgements, an
   application should use either RTCP feedback or DCCP acknowledgements,
   but not both (use of both types of feedback will waste available
   network capacity, but is not otherwise harmful).

4.3.  Multiplexing Data and Control

   The obvious mapping of RTP onto DCCP creates two DCCP connections for
   each RTP flow: one for RTP data packets, one for RTP control packets.
   A frequent criticism of RTP relates to the number of ports it uses,
   since large telephony gateways can support more than 32768 RTP flows
   between pairs of gateways, and so run out of UDP ports.  In addition,
   use of multiple ports complicates NAT traversal.  For these reasons,

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   it is RECOMMENDED that RTP and RTCP flows be multiplexed onto a
   single DCCP connection.

   RTP and RTCP packets multiplexed onto a single connection can be
   distinguished provided care is taken in assigning RTP payload types.
   The RTP payload type and marker bit(s) occupy the same space in the
   packet as does the RTCP packet type field.  Provided the RTP payload
   type is chosen such that the payload type, or the payload type plus
   128 (when the marker bit is set), does not clash with any of the used
   RTCP packet types, the two can be demultiplexed.  With the RTCP
   packet types registered at the time of this writing, this implies
   that RTP payload types 64-65 and 72-79 must be avoided.  None of the
   registered static payload type assignments are in this range, and
   typical practice is to make dynamic assignments in the range 96-127,
   so this restriction is not typically problematic.  This multiplexing
   does not otherwise impact the operation of RTP or RTCP.

   There may be circumstances where multiplexing RTP and RTCP is not
   desired, for example when translating from an RTP stream over non-
   DCCP transport that uses conflicting RTP payload types and RTCP
   packet types.  As specified in Section 5.1, the "a=rtcp:" SDP
   attribute MAY be used to signal use of non-multiplexed RTCP.

4.4.  RTP Sessions and DCCP Connections

   An end system should not assume that it will observe only a single
   RTP synchronisation source (SSRC) because it is using DCCP framing.
   An RTP session can span any number of transport connections, and can
   include RTP mixers or translators bringing other participants into
   the session.  The use of a unicast DCCP connection does not imply
   that the RTP session will have only two participants, and RTP end
   systems must assume that multiple synchronisation sources may be
   observed when using RTP over DCCP.

   An RTP translator bridging multiple DCCP connections to form a single
   RTP session needs to be aware of the congestion state of each DCCP
   connection, and must adapt the media to the available capacity of
   each.  In general, transcoding is required to perform adaptation:
   this is computationally expensive, induces delay, and generally gives
   poor quality results.  Depending on the payload, it might be possible
   to use some form of scalable coding.  Scalable media coding formats
   are an active research area, and are not in widespread use at the
   time of this writing.

   A single RTP session may also span a DCCP connection and some other
   type of transport connection.  An example might be an RTP over DCCP
   connection from an RTP end system to an RTP translator, with an RTP
   over UDP/IP multicast group on the other side of the translator.  A

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   second example might be an RTP over DCCP connection that links PSTN
   gateways.  The issues for such an RTP translator are similar to those
   when linking two DCCP connections, except that the congestion control
   algorithms on either side of the translator may not be compatible.
   Implementation of effective translators for such an environment is

4.5.  RTP Profiles

   In general, there is no conflict between new RTP Profiles and DCCP
   framing, and most RTP profiles can be negotiated for use over DCCP.
   The only potential for conflict occurs if an RTP profile changes the
   RTCP reporting interval or the RTP packet transmission rules, since
   this may conflict with DCCP congestion control.  If an RTP profile
   conflicts with DCCP congestion control, that profile MUST NOT be used
   with DCCP.

   Of the profiles currently defined, the RTP Profile for Audio and
   Video Conferences with Minimal Control [4], the Secure Real-time
   Transport Protocol [7], the Extended RTP Profile for RTCP-based
   Feedback [8], and the Extended Secure RTP Profile for RTCP-based
   Feedback [9] MAY be used with DCCP.  The RTP Profile for TFRC [15]
   MUST NOT be used with DCCP, since it conflicts with DCCP congestion
   control by providing alternative congestion control semantics.

5.  RTP over DCCP: Signalling using SDP

   The Session Description Protocol (SDP) [3] and the offer/answer model
   [10] are widely used to negotiate RTP sessions (for example, using
   the Session Initiation Protocol [20]).  This section describes how
   SDP is used to signal RTP sessions running over DCCP.

5.1.  Protocol Identification

   SDP uses a media ("m=") line to convey details of the media format
   and transport protocol used.  The ABNF syntax of a media line is as
   follows (from [3]):

       media-field = %x6d "=" media SP port ["/" integer] SP proto
                     1*(SP fmt) CRLF

   The proto field denotes the transport protocol used for the media,
   while the port indicates the transport port to which the media is
   sent.  Following [5] and [11] this memo defines the following five
   values of the proto field to indicate media transported using DCCP:

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   The "DCCP" protocol identifier is similar to the "UDP" and "TCP"
   protocol identifiers and describes the transport protocol, but not
   the upper-layer protocol.  An SDP "m=" line that specifies the "DCCP"
   protocol MUST further qualify the application layer protocol using a
   fmt identifier.  A single DCCP port is used, as denoted by the port
   field in the media line.  The "DCCP" protocol identifier MUST NOT be
   used to signal RTP sessions running over DCCP.

   The "DCCP/RTP/AVP" protocol identifier refers to RTP using the RTP
   Profile for Audio and Video Conferences with Minimal Control [4]
   running over DCCP.

   The "DCCP/RTP/SAVP" protocol identifier refers to RTP using the
   Secure Real-time Transport Protocol [7] running over DCCP.

   The "DCCP/RTP/AVPF" protocol identifier refers to RTP using the
   Extended RTP Profile for RTCP-based Feedback [8] running over DCCP.

   The "DCCP/RTP/SAVPF" protocol identifier refers to RTP using the
   Extended Secure RTP Profile for RTCP-based Feedback [9] running over

   By default, a single DCCP connection on the specified port is used
   for both RTP and RTCP packets.  The "a=rtcp:" attribute [12] MAY be
   used to specify an alternate DCCP port for RTCP, in which case a
   separate DCCP connection is opened to transport the RTCP data.

   Port 5004 is registered for use by RTP and SHOULD be the default port
   chosen by applications.  If more than one RTP session is active from
   a host, ports in the dynamic range SHOULD be used for the other
   sessions.  If RTCP is to be sent on a separate DCCP connection to
   RTP, the RTCP connection SHOULD use port 5005 by default.

5.2.  Service Codes

   In addition to the port number, specified on the SDP "m=" line, a
   DCCP connection has an associated service code.  A single new SDP
   attribute is defined to signal the service code:

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       dccp-service-attr = %x61 "=dccp-service-code:" service-code

       service-code      = hex-sc / decimal-sc / ascii-sc

       hex-sc            = "SC=x" *HEXDIG

       decimal-sc        = "SC="  *DIGIT

       ascii-sc          = "SC:"  *sc-char

       sc-char           = %d42-43, %d45-47, %d63-90, %d95, %d97-122

   where DIGIT and HEXDIG are as defined in [13].  The service code
   should be interpreted as defined in Section 8.1.2 of [2].  The
   following DCCP service codes are registered for use with RTP:

   o  SC:RTPA an RTP session conveying audio data

   o  SC:RTPV an RTP session conveying video data

   o  SC:RTPT an RTP session conveying textual data

   o  SC:RTPO an RTP session conveying other types of media

   To ease the job of middleboxes, applications SHOULD use these service
   codes to identify RTP sessions running within DCCP.

   The "a=dccp-service-code:" attribute is a media level attribute which
   is not subject to the charset attribute.

5.3.  Connection Management

   The "a=setup:" attribute indicates which of the end points should
   initiate the DCCP connection establishment (i.e. send the initial
   DCCP-Request packet).  The "a=setup:" attribute MUST be used in a
   manner comparable with [11], except that DCCP connections are being
   initiated rather than TCP connections.

   After the initial offer/answer exchange, the end points may decide to
   re-negotiate various parameters.  The "a=connection:" attribute MUST
   be used in a manner compatible with [11] to decide whether a new DCCP
   connection needs to be established as a result of subsequent offer/
   answer exchanges, or if the existing connection should still be used.

5.4.  Example

   An offerer at signals its availability for an H.261 video

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   session, using RTP/AVP over DCCP with service code "RTPV":

       o=alice 1129377363 1 IN IP4
       c=IN IP4
       t=0 0
       m=video 5004 DCCP/RTP/AVP 99
       a=rtpmap:99 h261/90000

   An answerer at receives this offer and responds with the
   following answer:

       o=bob 1129377364 1 IN IP4
       c=IN IP4
       t=0 0
       m=video 9 DCCP/RTP/AVP 99
       a=rtpmap:99 h261/90000

   The end point at then initiates a DCCP connection to port
   5004 at  Note that DCCP port 5004 is used for both the
   RTP and RTCP data, and port 5005 is unused.

6.  Security Considerations

   The security considerations in the RTP specification [1] and any
   applicable RTP profile (e.g. [4], [7], [8], or [9]) or payload format
   apply when transporting RTP over DCCP.

   The security considerations in the DCCP specification [2] apply.

   The SDP signalling described in Section 5 is subject to the security
   considerations of [3], [10], [11] and [5].

   It is not believed that any additional security considerations are
   introduced as a result of combining these protocols.  Indeed, the
   provision of effective congestion control for RTP will alleviate the
   potential for denial-of-service present when RTP flows ignore the
   advice in [1] to monitor packet loss and reduce their sending rate in

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   the face of persistent congestion.

7.  IANA Considerations

   The following SDP "proto" field identifiers are registered: "DCCP",
   (see Section 5.1 of this memo).

   One new SDP attribute ("a=dccp-service-code:") is registered (see
   Section 5.2 of this memo).

   The following DCCP service codes are registered: SC:RTPA, SC:RTPV,
   SC:RTPT, and SC:RTPO (see Section 5.2 of this memo).

   DCCP ports 5004 ("DCCP RTP") and 5005 ("DCCP RTCP") are registered
   for use as default RTP/RTCP ports (see Section 5.1 of this memo).
   The four services codes listed above are to be associated with these

8.  Acknowledgements

   This work was supported in part by the UK Engineering and Physical
   Sciences Research Council.  Thanks are due to to Philippe Gentric,
   Magnus Westerlund and the other members of the DCCP working group for
   their comments.

9.  References

9.1.  Normative References

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

   [2]   Kohler, E., Handley, M., and S. Floyd, "Datagram Congestion
         Control Protocol (DCCP)", RFC 4340, March 2006.

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

   [4]   Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
         Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

   [5]   Lazzaro, J., "Framing RTP and RTCP Packets over Connection-
         Oriented Transport", RFC 4571, June 2006.

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

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

   [8]   Ott, J., Wenger, S., Sato, N., and C. Burmeister, "Extended RTP
         Profile for RTCP-based Feedback(RTP/AVPF)", RFC 4585,
         June 2006.

   [9]   Ott, J. and E. Carrara, "Extended Secure RTP Profile for RTCP-
         based Feedback (RTP/SAVPF)", draft-ietf-avt-profile-savpf-02
         (work in progress), July 2005.

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

   [11]  Yon, D. and G. Camarillo, "TCP-Based Media Transport in the
         Session Description Protocol (SDP)", RFC 4145, September 2005.

   [12]  Huitema, C., "Real Time Control Protocol (RTCP) attribute in
         Session Description Protocol (SDP)", RFC 3605, October 2003.

   [13]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
         Specifications: ABNF", RFC 2234, November 1997.

9.2.  Informative References

   [14]  Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
         Control for Voice Traffic in the Internet", RFC 3714,
         March 2004.

   [15]  Gharai, L., "Use of the Extended RTP Profile for RTCP-based
         Feedback (RTP/AVPF) to Support TCP-Friendly Rate Control",
         draft-ietf-avt-tfrc-profile-06 (work in progress), June 2006.

   [16]  Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
         Friendly Rate Control (TFRC): Protocol Specification",
         RFC 3448, January 2003.

   [17]  Andreasen, F., "A No-Op Payload Format for RTP",
         draft-wing-avt-rtp-noop-03 (work in progress), May 2005.

   [18]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-
         Time Transport Protocol (RTP) Payload Format and File Storage
         Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-
         Rate Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.

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   [19]  Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
         Extended Reports (RTCP XR)", RFC 3611, November 2003.

   [20]  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.

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Author's Address

   Colin Perkins
   University of Glasgow
   Department of Computing Science
   17 Lilybank Gardens
   Glasgow  G12 8QQ


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