Internet DRAFT - draft-perkins-avt-rtp-and-rtcp-mux


Network Working Group                                         C. Perkins
Internet-Draft                                     University of Glasgow
Updates: 3550 (if approved)                                M. Westerlund
Expires: March 24, 2007                                         Ericsson
                                                      September 20, 2006

       Multiplexing RTP Data and Control Packets on a Single Port

Status of this Memo

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Copyright Notice

   Copyright (C) The Internet Society (2006).


   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
     5.2.  Any Source Multicast Sessions  . . . . . . . . . . . . . .  7
     5.3.  Source Specific Multicast Sessions . . . . . . . . . . . .  7
   6.  Multiplexing, Bandwidth, and Quality of Service  . . . . . . .  8
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . .  9
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  9
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . .  9
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . .  9
     10.1. Normative References . . . . . . . . . . . . . . . . . . .  9
     10.2. Informative References . . . . . . . . . . . . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11
   Intellectual Property and Copyright Statements . . . . . . . . . . 12

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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 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 [13] 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 [14] 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

   While these design choices are appropriate for many use 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",
   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 RTCP FIR and NACK
      packets defined in the 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]

   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 a multicast sessions, also depends whether ASM or SSM multicast
   is to be used.

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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.

   When the Session Description Protocol (SDP) [8] is used to negotiate
   RTP sessions according to the offer/answer model [9], the "a=rtcp:"
   attribute [10] is used to indicate the port chosen for RTCP traffic,
   if the default of using an odd/even port pair is not desirable.  If
   RTP and RTCP are to be multiplexed on a single port, this attribute
   MUST be included in the initial SDP offer, and MUST indicate the the
   same port as included in the "m=" line.  For example:

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

   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 supports multiplexing of RTP and RTCP onto a single
   port it MUST include an "a=rtcp:" attribute in the answer.  The port
   specified in that attribute MUST be the same as that used for RTP in
   the "m=" line of the answer.  The RTP payload types used MUST conform
   to the rules in Section 4, and the answer MUST be rejected if there
   is a conflict between the chosen RTP payload types and the expected
   RTCP packet types.

   If the answer does not contain an "a=rtcp:" attribute, the offerer
   MUST NOT multiplex RTP and RTCP packets on a single port.  Instead,
   it must send and receive RTCP on a port allocated according to the
   usual port pair rules.  This will occur when talking to a peer that
   does not understand the "a=rtcp:" attribute or this specification.

   If the answer contains an "a=rtcp:" attribute, but that attribute
   specifies a different port for RTCP than for RTP, then the answer
   MUST be rejected, and the session re-negotiated using separate RTP
   and RTCP ports.

   Answerers which support the "a=rtcp:" attribute but do not understand

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   this memo should reject sessions where the RTP and RTCP ports are the
   same (Section 11 of [1] prohibits such sessions unless the mechanisms
   described in this memo are used).  It is likely that this is a poorly
   tested feature of older implementations, however, and implementations
   should be robust to unexpected behaviour.  If the offerer suspects a
   session was rejected due to the presence of multiplexed RTP and RTCP,
   it MAY retry the offer using separate ports for RTP and RTCP.

   When using SIP with a forking proxy, it is possible that multiple 200
   (OK) answers will be received, some supporting multiplexed RTP and
   RTCP, some not.  This is not an issue if a separate RTP session is
   established with each answerer, since multiplexing occurs on a per
   session basis, but does prevent a single RTP session being opened
   between the offerer and all answerers.

   TODO: expand this discussion.  Does SIP define if this should be a
   single RTP session or multiple sessions?

   TODO: discuss interactions between multiplexed RTP and RTCP, and
   Interactive Connectivity Establishment (ICE) [15].

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.

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
       collocated, 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

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       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 (for example, using
       "a=rtcp:" with the same port number as specified for RTP on the
       "m=" line, 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, since
       this does not hinder third party monitoring.  When using SDP, the
       multiplexing MUST be signalled using the "a=rtcp:" attribute [10]
       with the same port number as specified for RTP on the "m=" line.

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

   Multiplexing RTP and RTCP onto a single port is more acceptable for
   an SSM session than for an ASM session, since SSM sessions cannot
   readily make use of third party reception quality monitoring devices
   that listen to the multicast RTCP traffic but not the data traffic
   (since the RTCP traffic is unicast to the distribution source, rather
   than multicast, and since one cannot subscribe to only the RTCP
   packets on the SSM channel, even if sent on a separate port).

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
   the combinded 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:"
   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 [11], then any QoS reservation
   SHOULD be made for bandwidth equal to (AS + RS + RR), taking the RS

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   and RR values from the SDP answer.

7.  Security Considerations

   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) [12] 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

   No IANA actions are required.

9.  Acknowledgements

   We wish to thank Steve Casner, Joerg Ott, Christer Holmberg, Gunnar
   Hellstrom, Randell Jesup, Hadriel Kaplan and Harikishan Desineni for
   their comments on this memo.  This work is 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.

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   [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., "RTCP Extensions for Single-Source Multicast
         Sessions with Unicast Feedback", draft-ietf-avt-rtcpssm-11
         (work in progress), March 2006.

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

   [12]  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

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

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

   [15]  Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Methodology for Network  Address Translator (NAT) Traversal for
         Offer/Answer Protocols", draft-ietf-mmusic-ice-10 (work in
         progress), August 2006.

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Authors' Addresses

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


   Magnus Westerlund
   Torshamgatan 23
   Stockholm  SE-164 80


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