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Versions: (draft-perkins-avt-rtp-and-rtcp-mux) 00 01 02 03 04 05 06 07 RFC 5761

Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Updates: 3550 (if approved)                                M. Westerlund
Intended status: Standards Track                                Ericsson
Expires: November 22, 2007                                  May 21, 2007


       Multiplexing RTP Data and Control Packets on a Single Port
                 draft-ietf-avt-rtp-and-rtcp-mux-05.txt

Status of this Memo

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   This Internet-Draft will expire on November 22, 2007.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   This memo discusses issues that arise when multiplexing RTP data
   packets and RTP control protocol (RTCP) packets on a single UDP port.
   It updates RFC 3550 to describe when such multiplexing is, and is
   not, appropriate, and explains how the Session Description Protocol
   (SDP) can be used to signal multiplexed sessions.





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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Background . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Distinguishable RTP and RTCP Packets . . . . . . . . . . . . .  4
   5.  Multiplexing RTP and RTCP on a Single Port . . . . . . . . . .  5
     5.1.  Unicast Sessions . . . . . . . . . . . . . . . . . . . . .  6
       5.1.1.  SDP Signalling . . . . . . . . . . . . . . . . . . . .  6
       5.1.2.  Interactions with SIP forking  . . . . . . . . . . . .  7
       5.1.3.  Interactions with ICE  . . . . . . . . . . . . . . . .  7
       5.1.4.  Interactions with Header Compression . . . . . . . . .  8
     5.2.  Any Source Multicast Sessions  . . . . . . . . . . . . . .  8
     5.3.  Source Specific Multicast Sessions . . . . . . . . . . . .  9
   6.  Multiplexing, Bandwidth, and Quality of Service  . . . . . . .  9
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 11
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 11
     10.2. Informative References . . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13
   Intellectual Property and Copyright Statements . . . . . . . . . . 14




























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1.  Introduction

   The Real-time Transport Protocol (RTP) [1] comprises two components:
   a data transfer protocol, and an associated control protocol (RTCP).
   Historically, RTP and RTCP have been run on separate UDP ports.  With
   increased use of Network Address Translation (NAT) this has become
   problematic, since opening multiple NAT pinholes can be costly.  This
   memo discusses how the RTP and RTCP flows for a single media type can
   be run on a single port, to ease NAT traversal, and considers when
   such multiplexing is appropriate.  The multiplexing of several types
   of media (e.g. audio and video) onto a single port is not considered
   here (but see Section 5.2 of [1]).

   This memo is structured as follows: in Section 2 we discuss the
   design choices which led to the use of separate ports, and comment on
   the applicability of those choices to current network environments.
   We discuss terminology in Section 3, how to distinguish multiplexed
   packets in Section 4, and then specify when and how RTP and RTCP
   should be multiplexed, and how to signal multiplexed sessions, in
   Section 5.  Quality of service and bandwidth issues are discussion in
   Section 6.  We conclude with security considerations in Section 7.

   This memo updates Section 11 of [1].


2.  Background

   An RTP session comprises data packets and periodic control (RTCP)
   packets.  RTCP packets are assumed to use "the same distribution
   mechanism as the data packets" and the "underlying protocol MUST
   provide multiplexing of the data and control packets, for example
   using separate port numbers with UDP" [1].  Multiplexing was deferred
   to the underlying transport protocol, rather than being provided
   within RTP, for the following reasons:

   1.  Simplicity: an RTP implementation is simplified by moving the RTP
       and RTCP demultiplexing to the transport layer, since it need not
       concern itself with the separation of data and control packets.
       This allows the implementation to be structured in a very natural
       fashion, with a clean separation of data and control planes.

   2.  Efficiency: following the principle of integrated layer
       processing [14] an implementation will be more efficient when
       demultiplexing happens in a single place (e.g. according to UDP
       port) than when split across multiple layers of the stack (e.g.
       according to UDP port then according to packet type).





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   3.  To enable third party monitors: while unicast voice-over-IP has
       always been considered, RTP was also designed to support loosely
       coupled multicast conferences [15] and very large-scale multicast
       streaming media applications (such as the so-called "triple-play"
       IPTV service).  Accordingly, the design of RTP allows the RTCP
       packets to be multicast using a separate IP multicast group and
       UDP port to the data packets.  This not only allows participants
       in a session to get reception quality feedback, but also enables
       deployment of third party monitors which listen to reception
       quality without access to the data packets.  This was intended to
       provide manageability of multicast sessions, without compromising
       privacy.

   While these design choices are appropriate for many uses of RTP, they
   are problematic in some cases.  There are many RTP deployments which
   don't use IP multicast, and with the increased use of Network Address
   Translation (NAT) the simplicity of multiplexing at the transport
   layer has become a liability, since it requires complex signalling to
   open multiple NAT pinholes.  In environments such as these, it is
   desirable to provide an alternative to demultiplexing RTP and RTCP
   using separate UDP ports, instead using only a single UDP port and
   demultiplexing within the application.

   This memo provides such an alternative by multiplexing RTP and RTCP
   packets on a single UDP port, distinguished by the RTP payload type
   and RTCP packet type values.  This pushes some additional work onto
   the RTP implementation, in exchange for simplified NAT traversal.


3.  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 RFC 2119 [2].


4.  Distinguishable RTP and RTCP Packets

   When RTP and RTCP packets are multiplexed onto a single port, they
   can be distinguished provided: 1) the RTP payload type (PT) values
   used are distinct from the RTCP packet types used; and 2) for each
   RTP payload type, PT+128 is distinct from the RTCP packet types used.
   The first constraint precludes a direct conflict between RTP payload
   type and RTCP packet type, the second constraint precludes a conflict
   between an RTP data packet with marker bit set and an RTCP packet.
   This demultiplexing method works because the RTP payload type and
   RTCP packet type occupy the same position within the packet.




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   The following conflicts between RTP and RTCP packet types are known:

   o  RTP payload types 64-65 conflict with the (obsolete) RTCP FIR and
      NACK packets defined in the original RTP Payload Format for H.261
      [3].

   o  RTP payload types 72-76 conflict with the RTCP SR, RR, SDES, BYE
      and APP packets defined in the RTP specification [1].

   o  RTP payload types 77-78 conflict with the RTCP RTPFB and PSFB
      packets defined in the RTP/AVPF profile [4].

   o  RTP payload type 79 conflicts with RTCP Extended Report (XR) [5]
      packets.

   o  RTP payload type 80 conflicts with Receiver Summary Information
      (RSI) packets defined in the RTCP Extensions for Single-Source
      Multicast Sessions with Unicast Feedback [6].

   New RTCP packet types may be registered in future, and will further
   reduce the RTP payload types that are available when multiplexing RTP
   and RTCP onto a single port.  To allow this multiplexing, future RTCP
   packet type assignments SHOULD be made after the current assignments
   in the range 209-223, then in the range 194-199, so that only the RTP
   payload types in the range 64-95 are blocked.

   Given these constraints, it is RECOMMENDED to follow the guidelines
   in the RTP/AVP profile [7] for the choice of RTP payload type values,
   with the additional restriction that payload type values in the range
   64-95 MUST NOT be used.  Specifically, dynamic RTP payload types
   SHOULD be chosen in the range 96-127 where possible.  Values below 64
   MAY be used if that is insufficient, in which case it is RECOMMENDED
   that payload type numbers that are not statically assigned by [7] be
   used first.

      Note: since all RTCP packets MUST be sent as compound packets
      beginning with an SR or an RR packet ([1] Section 6.1), one might
      wonder why RTP payload types other than 72 and 73 are prohibited
      when multiplexing RTP and RTCP.  This is done to ensure robustness
      against broken nodes which send non-compliant RTCP packets, which
      might otherwise be confused with multiplexed RTP packets.


5.  Multiplexing RTP and RTCP on a Single Port

   The procedures for multiplexing RTP and RTCP on a single port depend
   on whether the session is a unicast session or a multicast session.
   For multicast sessions, the procedures also depend on whether Any



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   Source Multicast (ASM) or Source Specific Multicast (SSM) multicast
   is to be used.

5.1.  Unicast Sessions

   It is acceptable to multiplex RTP and RTCP packets on a single UDP
   port to ease NAT traversal for unicast sessions, provided the RTP
   payload types used in the session are chosen according to the rules
   in Section 4, and provided that multiplexing is signalled in advance.
   The following sections describe how such multiplexed sessions can be
   signalled using the Session Initiation Protocol (SIP) with the offer/
   answer model.

5.1.1.  SDP Signalling

   When the Session Description Protocol (SDP) [8] is used to negotiate
   RTP sessions following the offer/answer model [9], the "a=rtcp-mux"
   attribute (see Section 8) indicates the desire to multiplex RTP and
   RTCP onto a single port.  The initial SDP offer MUST include this
   attribute to request multiplexing of RTP and RTCP on a single port.
   For example:

       v=0
       o=csp 1153134164 1153134164 IN IP6 2001:DB8::211:24ff:fea3:7a2e
       s=-
       c=IN IP6 2001:DB8::211:24ff:fea3:7a2e
       t=1153134164 1153137764
       m=audio 49170 RTP/AVP 97
       a=rtpmap:97 iLBC/8000
       a=rtcp-mux

   This offer denotes a unicast voice-over-IP session using the RTP/AVP
   profile with iLBC coding.  The answerer is requested to send both RTP
   and RTCP to port 49170 on IPv6 address 2001:DB8::211:24ff:fea3:7a2e.

   If the answerer wishes to multiplex RTP and RTCP onto a single port
   it MUST include an "a=rtcp-mux" attribute in the answer.  The RTP
   payload types used in the answer MUST conform to the rules in
   Section 4.

   If the answer does not contain an "a=rtcp-mux" attribute, the offerer
   MUST NOT multiplex RTP and RTCP packets on a single port.  Instead,
   it should send and receive RTCP on a port allocated according to the
   usual port selection rules (either the port pair, or a signalled port
   if the "a=rtcp:" attribute [10] is also included).  This will occur
   when talking to a peer that does not understand the "a=rtcp-mux"
   attribute.




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   When SDP is used in a declarative manner, the presence of an "a=rtcp-
   mux" attribute signals that the sender will multiplex RTP and RTCP on
   the same port.  The receiver MUST be prepared to receive RTCP packets
   on the RTP port, and any resource reservation needs to be made
   including the RTCP bandwidth.

5.1.2.  Interactions with SIP forking

   When using SIP with a forking proxy, it is possible that an INVITE
   request may result in multiple 200 (OK) responses.  If RTP and RTCP
   multiplexing is offered in that INVITE, it is important to be aware
   that some answerers may support multiplexed RTP and RTCP, some not.
   This will require the offerer to listen for RTCP on both the RTP port
   and the usual RTCP port, and to send RTCP on both ports, unless
   branches of the call that support multiplexing are re-negotiated to
   use separate RTP and RTCP ports.

5.1.3.  Interactions with ICE

   It is common to use the Interactive Connectivity Establishment (ICE)
   [16] methodology to establish RTP sessions in the presence of Network
   Address Translation (NAT) devices or other middleboxes.  If RTP and
   RTCP are sent on separate ports, the RTP media stream comprises two
   components in ICE (one for RTP and one for RTCP), with connectivity
   checks being performed for each component.  If RTP and RTCP are to be
   multiplexed on the same port some of these connectivity checks can be
   avoided, reducing the overhead of ICE.

   If it is desired to use both ICE and multiplexed RTP and RTCP, the
   initial offer MUST contain an "a=rtcp-mux" attribute to indicate that
   RTP and RTCP multiplexing is desired, and MUST contain "a=candidate:"
   lines for both RTP and RTCP along with an "a=rtcp:" line indicating a
   fallback port for RTCP in the case that the answerer does not support
   RTP and RTCP multiplexing.  This MUST be done for each media where
   RTP and RTCP multiplexing is desired.

   If the answerer wishes to multiplex RTP and RTCP on a single port, it
   MUST generate an answer containing an "a=rtcp-mux" attribute, and a
   single "a=candidate:" line corresponding to the RTP port (i.e. there
   is no candidate for RTCP), for each media where it is desired to use
   RTP and RTCP multiplexing.  The answerer then performs connectivity
   checks for that media as if the offer had contained only a single
   candidate for RTP.  If the answerer does not want to multiplex RTP
   and RTCP on a single port, it MUST NOT include the "a=rtcp-mux"
   attribute in its answer, and MUST perform connectivity checks for all
   offered candidates in the usual manner.

   On receipt of the answer, the offerer looks for the presence of the



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   "a=rtcp-mux" line for each media where multiplexing was offered.  If
   this is present, then connectivity checks proceed as if only a single
   candidate (for RTP) were offered, and multiplexing is used once the
   session is established.  If the "a=rtcp-mux" line is not present, the
   session proceeds with connectivity checks using both RTP and RTCP
   candidates, eventually leading to a session being established with
   RTP and RTCP on separate ports (as signalled by the "a=rtcp:"
   attribute).

5.1.4.  Interactions with Header Compression

   Multiplexing RTP and RTCP packets onto a single port may negatively
   impact header compression schemes, for example Compressed RTP (CRTP)
   [17] and RObust Header Compression (ROHC) [18].  Header compression
   exploits patterns of change in the RTP headers of consecutive packets
   to send an indication that the packet changed in the expected way,
   rather than sending the complete header each time.  This can lead to
   significant bandwidth savings if flows have uniform behaviour.

   The presence of RTCP packets multiplexed with RTP data packets can
   disrupt the patterns of change between headers, and has the potential
   to significantly reduce header compression efficiency.  The extent of
   this disruption depends on the header compression algorithm used, and
   on the way flows are classified.  A well designed classifier should
   be able to separate RTP and RTCP multiplexed on the same port into
   different compression contexts, using the payload type field, such
   that the effect on the compression ratio is small.  A classifier that
   assigns compression contexts based only on the IP addresses and UDP
   ports will not perform well.  It is expected that implementations of
   header compression will need to be updated to efficiently support RTP
   and RTCP multiplexed on the same port.

   This effect of multiplexing RTP and RTCP on header compression may be
   especially significant in those environments, such as some wireless
   telephony systems, which rely on the efficiency of header compression
   to match the media to a limited capacity channel.  The implications
   of multiplexing RTP and RTCP should be carefully considered before
   use in such environments.

5.2.  Any Source Multicast Sessions

   The problem of NAT traversal is less severe for any source multicast
   (ASM) RTP sessions than for unicast RTP sessions, and the benefit of
   using separate ports for RTP and RTCP is greater, due to the ability
   to support third party RTCP only monitors.  Accordingly, RTP and RTCP
   packets SHOULD NOT be multiplexed onto a single port when using ASM
   multicast RTP sessions, and SHOULD instead use separate ports and
   multicast groups.



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5.3.  Source Specific Multicast Sessions

   RTP sessions running over Source Specific Multicast (SSM) send RTCP
   packets from the source to receivers via the multicast channel, but
   use a separate unicast feedback mechanism [6] to send RTCP from the
   receivers back to the source, with the source either reflecting the
   RTCP packets to the group, or sending aggregate summary reports.

   Following the terminology of [6], we identify three RTP/RTCP flows in
   an SSM session:

   1.  RTP and RTCP flows between media sender and distribution source.
       In many scenarios, the media sender and distribution source are
       co-located, so multiplexing is not a concern.  If the media
       sender and distribution source are connected by a unicast
       connection, the rules in Section 5.1 of this memo apply to that
       connection.  If the media sender and distribution source are
       connected by an Any Source Multicast connection, the rules in
       Section 5.2 apply to that connection.  If the media sender and
       distribution source are connected by a Source Specific Multicast
       connection, the RTP and RTCP packets MAY be multiplexed on a
       single port, provided this is signalled (using "a=rtcp-mux" if
       using SDP).

   2.  RTP and RTCP sent from the distribution source to the receivers.
       The distribution source MAY multiplex RTP and RTCP onto a single
       port to ease NAT traversal issues on the forward SSM path,
       although doing so may hinder third party monitoring devices if
       the session uses the simple feedback model.  When using SDP, the
       multiplexing SHOULD be signalled using the "a=rtcp-mux"
       attribute.

   3.  RTCP sent from receivers to distribution source.  This is an RTCP
       only path, so multiplexing is not a concern.

   Multiplexing RTP and RTCP packets on a single port in an SSM session
   has the potential for interactions with header compression described
   in Section 5.1.4.


6.  Multiplexing, Bandwidth, and Quality of Service

   Multiplexing RTP and RTCP has implications on the use of Quality of
   Service (QoS) mechanism that handles flow that are determined by a
   three or five tuple (protocol, port and address for source and/or
   destination).  In these cases the RTCP flow will be merged with the
   RTP flow when multiplexing them together.  Thus the RTCP bandwidth
   requirement needs to be considered when doing QoS reservations for



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   the combined RTP and RTCP flow.  However from an RTCP perspective it
   is beneficial to receive the same treatment of RTCP packets as for
   RTP as it provides more accurate statistics for the measurements
   performed by RTCP.

   The bandwidth required for a multiplexed stream comprises the session
   bandwidth of the RTP stream, plus the bandwidth used by RTCP.  In the
   usual case, the RTP session bandwidth is signalled in the SDP "b=AS:"
   (or "b=TIAS:" [11]) line, and the RTCP traffic is limited to 5% of
   this value.  Any QoS reservation SHOULD therefore be made for 105% of
   the "b=AS:" value.  If a non-standard RTCP bandwidth fraction is
   used, signalled by the SDP "b=RR:" and/or "b=RS:" lines [12], then
   any QoS reservation SHOULD be made for bandwidth equal to (AS + RS +
   RR), taking the RS and RR values from the SDP answer.


7.  Security Considerations

   The usage of multiplexing RTP and RTCP is not believed to introduce
   any new security considerations.  Known major issues are, integrity
   and authentication of the signalling used to setup the multiplexing,
   the integrity, authentication and confidentiality of the actual RTP
   and RTCP traffic.  The security considerations in the RTP
   specification [1] and any applicable RTP profile (e.g. [7]) and
   payload format(s) apply.

   If the Secure Real-time Transport Protocol (SRTP) [13] is to be used
   in conjunction with multiplexed RTP and RTCP, then multiplexing MUST
   be done below the SRTP layer.  The sender generates SRTP and SRTCP
   packets in the usual manner, based on their separate cryptographic
   contexts, and multiplexes them onto a single port immediately before
   transmission.  At the receiver, the cryptographic context is derived
   from the SSRC, destination network address and destination transport
   port number in the usual manner, augmented using the RTP payload type
   and RTCP packet type to demultiplex SRTP and SRTCP according to the
   rules in Section 4 of this memo.  After the SRTP and SRTCP packets
   have been demultiplexed, cryptographic processing happens in the
   usual manner.


8.  IANA Considerations

   Following the guidelines in [8], the IANA is requested to register
   one new SDP attribute:

   o  Contact name/email: authors of RFC XXXX





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   o  Attribute name: rtcp-mux

   o  Long-form attribute name: RTP and RTCP multiplexed on one port

   o  Type of attribute: media level

   o  Subject to charset: no

   This attribute is used to signal that RTP and RTCP traffic should be
   multiplexed on a single port (see Section 5 of this memo).  It is a
   property attribute, which does not take a value.

   Note to RFC Editor: please replace "RFC XXXX" above with the RFC
   number of this memo, and remove this note.


9.  Acknowledgements

   We wish to thank Steve Casner, Joerg Ott, Christer Holmberg, Gunnar
   Hellstrom, Randell Jesup, Hadriel Kaplan, Harikishan Desineni,
   Stephan Wenger, Jonathan Rosenberg, Roni Even, Ingemar Johansson,
   Dave Singer, and Kevin Johns for their comments on this memo.  This
   work was supported in part by the UK Engineering and Physical
   Sciences Research Council.


10.  References

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

   [3]   Turletti, T., "RTP Payload Format for H.261 Video Streams",
         RFC 2032, October 1996.

   [4]   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, July 2006.

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

   [6]   Chesterfield, J., Ott, J., and E. Schooler, "RTCP Extensions



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         for Single-Source Multicast Sessions with Unicast Feedback",
         draft-ietf-avt-rtcpssm-13 (work in progress), March 2007.

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

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

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

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

   [11]  Westerlund, M., "A Transport Independent Bandwidth Modifier for
         the Session Description Protocol (SDP)", RFC 3890,
         September 2004.

   [12]  Casner, S., "Session Description Protocol (SDP) Bandwidth
         Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
         July 2003.

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

10.2.  Informative References

   [14]  Clark, D. and D. Tennenhouse, "Architectural Considerations for
         a New Generation of Protocols", Proceedings of ACM
         SIGCOMM 1990, September 1990.

   [15]  Casner, S. and S. Deering, "First IETF Internet Audiocast", ACM
         SIGCOMM Computer Communication Review, Volume 22, Number 3,
         July 1992.

   [16]  Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Methodology for Network  Address Translator (NAT) Traversal for
         Offer/Answer Protocols", draft-ietf-mmusic-ice-15 (work in
         progress), March 2007.

   [17]  Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for
         Low-Speed Serial Links", RFC 2508, February 1999.

   [18]  Bormann, C., Burmeister, C., Degermark, M., Fukushima, H.,
         Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K.,
         Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke, T.,



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         Yoshimura, T., and H. Zheng, "RObust Header Compression (ROHC):
         Framework and four profiles: RTP, UDP, ESP, and uncompressed",
         RFC 3095, July 2001.


Authors' Addresses

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

   Email: csp@csperkins.org


   Magnus Westerlund
   Ericsson
   Torshamgatan 23
   Stockholm  SE-164 80
   Sweden

   Email: magnus.westerlund@ericsson.com



























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