Network Working Group                                          X. Marjou
Internet-Draft                                                A. Sollaud
Intended status: Standards Track                   France Telecom Orange
Expires: September 5, 2011                                 March 28, 4, 2011                               September 24, 2010

 Application Mechanism for keeping alive the Network Address Translator
              (NAT) mappings associated to RTP RTP/RTCP flows.


   This document lists the different mechanisms that enable applications
   using Real-time Transport Protocol (RTP) and RTP control protocol
   (RTCP) to maintain their RTP Network Address Translator (NAT)
   mappings alive.  It also makes a recommendation for a preferred
   mechanism.  This document is not applicable to Interactive
   Connectivity Establishment (ICE) agents.

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on March 28, September 5, 2011.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  List of Alternatives for Performing RTP Keepalive  . . . . . .  5
     4.1.  Transport Packet of 0-byte . . . . . . . . . . . . . . . .  5
     4.2.  RTP Packet with Comfort Noise Payload  . . . . . . . . . .  5
     4.3.  RTCP Packets Multiplexed with RTP Packets  . . . . . . . .  6  5
     4.4.  STUN Indication Packet . . . . . . . . . . . . . . . . . .  6
     4.5.  RTP Packet with Incorrect Version Number . . . . . . . . .  6
     4.6.  RTP Packet with Unknown Payload Type . . . . . . . . . . .  6
   5.  Recommended Solution for Keepalive Mechanism . . . . . . . . .  7
   6.  Media Format Exceptions  . . . . . . . . . . . . . . . . . . .  7
   7.  Timing and Transport Considerations  . . . . . . . . . . . . .  7
   8.  Security Considerations  RTCP Flow Keepalive  . . . . . . . . . . . . . . . . . . . . .  8
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . .  9
   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  8
   10.  9
   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . .  8
   11.  9
   12. References . . . . . . . . . . . . . . . . . . . . . . . . . .  8
     11.1. 10
     12.1. Normative references . . . . . . . . . . . . . . . . . . .  8
     11.2. 10
     12.2. Informative references . . . . . . . . . . . . . . . . . .  9 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 10 11

1.  Introduction

   Documents [RFC4787] and [RFC5382] describe Network Address Translator
   (NAT) behaviors and point out that two key aspects of NAT are
   mappings (a.k.a. bindings) and keeping them refreshed.  This
   introduces a derived requirement for applications engaged in a
   multimedia session involving NAT traversal: they need to generate a
   minimum of flow activity in order to create NAT mappings and maintain

   When applied to applications using the real-time transport protocol
   (RTP) [RFC3550], the RTP media stream packets themselves normally
   fulfill this requirement.  However there exist some cases where RTP
   does not generate the minimum required flow activity.

   The examples are:

   o  In some RTP usages, such as the Session Inititation Protocol (SIP)
      [RFC3550], agents can negotiate a unidirectional media stream by
      using the Session Description Protocol (SDP) [RFC4566] "recvonly"
      attribute on one agent and "sendonly" on the peer, as defined in
      [RFC3264].  [RFC3264] directs implementations not to transmit
      media on the receiving agent.  In case the agent receiving the
      media is located in the private side of a NAT, it will never
      receive RTP packets from the public peer if the NAT mapping has
      not been created.

   o  Similarly, a bidirectional media stream can be "put on hold".
      This is accomplished by using the SDP "sendonly" or "inactive"
      attributes.  Again [RFC3264] directs implementations to cease
      transmission of media in these cases.  However, doing so may cause
      NAT bindings to timeout, and media won't be able to come off hold.

   o  Some RTP payload formats, such as the payload format for text
      conversation [RFC4103], may send packets so infrequently that the
      interval exceeds the NAT binding timeouts.

   To solve these problems, an agent therefore needs to periodically
   send keepalive data within the outgoing RTP session of an RTP media
   stream regardless of whether the media stream is currently inactive,
   sendonly, recvonly or sendrecv, and regardless of the presence or
   value of the bandwidth attribute.

   It is important to note that the above examples NAT traversals constraints also usually
   require the agents to use symmetric Symmetric RTP / RTP Control Protocol (RTCP)
   [RFC4961] in addition to RTP keepalive.

   This document first states the requirements that must be supported to
   perform RTP keepalives (Section 3).  In a second step, the document
   reports the different mechanisms to overcome this problem
   (Section 4).  Section 5 finally states the recommended solution for
   RTP keepalive.  Section 6 discusses some media format exceptions.
   Section 7 adds details about timing and transport considerations.
   Section 8 documents how to maintain NAT bindings for RTCP.

   This document is not applicable to Interactive Connectivity
   Establishment (ICE) [RFC5245] agents.  Indeed, the ICE protocol
   together with Session Traversal Utilities for NAT (STUN) [RFC5389]
   and Traversal Using Relays around NAT (TURN) [RFC5766] solve the
   overall Network Address Translator (NAT) traversal mechanism of media
   streams.  In the context of RTP media streams, some agents may not
   require all ICE functionalities and may only need a keepalive
   mechanism.  This document thus applies to such agents, and does not
   apply to agents implementing ICE.

   The scope of the draft is also limited to RTP flows.  In particular,
   this document does not address keepalive activity related to:

   o  Session signaling flows, such as the Session Initiation Protocol

   o  RTP Control Protocol (RTCP) flows.
         Recall that [RFC3550] recommends a minimum interval of 5
         seconds and that "on hold" procedures of [RFC3264] do not
         impact RTCP transmissions.  Therefore, when in use, there is
         always some RTCP flow activity.

   Note that if a given media uses a codec that already integrates a
   keepalive mechanism, no additional keepalive mechanism is required at
   the RTP level.

   As mentionned mentioned in Section 3.5 of [RFC5405] "It is important to note
   that keep-alive keepalive messages are NOT RECOMMENDED for general use -- they
   are unnecessary for many applications and can consume significant
   amounts of system and network resources."

2.  Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   and "OPTIONAL" are to be interpreted as described in RFC 2119

3.  Requirements

   This section outlines the key requirements that need to be satisfied
   in order to provide RTP media keepalive.

   REQ-1  Some data is sent periodically within the outgoing RTP session
          for the whole duration of the RTP media stream.

   REQ-2  Any type of transport (e.g.  UDP, TCP) MUST be supported.

   REQ-3  Any media type (e.g. audio, video, text) MUST be supported.

   REQ-4  Any media format (e.g.  G.711, H.263) MUST be supported.

   REQ-5  Session signaling protocols SHOULD NOT be impacted.

   REQ-6  Impacts on existing software SHOULD be minimized.

   REQ-7  Remote peer SHOULD NOT be impacted.

   REQ-8  The support for RTP keepalive SHOULD be described in the SDP.

   REQ-9  The solution SHOULD cover the integration with RTCP.

4.  List of Alternatives for Performing RTP Keepalive

   This section lists, in no particular order, some alternatives that
   can be used to perform a keepalive message within RTP media streams.

4.1.  Transport Packet of 0-byte

   The application sends an empty transport packet (e.g.  UDP packet,
   DCCP packet).

   o  This alternative is specific to each transport protocol.

4.2.  RTP Packet with Comfort Noise Payload

   The application sends an RTP packet with a comfort-noise payload

   o  This alternative is limited to audio formats only.
   o  Comfort Noise needs to be supported by the remote peer.
   o  Comfort Noise needs to be signalled in SDP offer/answer.
   o  The peer is likely to render comfort noise at the other side, so
      the content of the payload (the noise level) needs to be carefully

4.3.  RTCP Packets Multiplexed with RTP Packets

   The application sends RTCP packets in the RTP media path itself (i.e.
   same tuples for both RTP and RTCP packets) [RFC5761].  RTCP packets
   therefore maintain the NAT mappings open. open as long as the requirements
   on parameter selection are fulfilled as discussed in Section 8.

   Note: "on hold" procedures of [RFC3264] do not impact RTCP

   o  Multiplexing RTP and RTCP must be supported by the remote peer.
   o  Some RTCP monitoring tools expect that RTCP packets are not
   o  RTCP must be configured so that Tmin value [RFC3550] is lower or
      equal to the Tr interval.

4.4.  STUN Indication Packet

   The application sends a STUN [RFC5389] Binding Indication packet as
   specified in ICE [RFC5245].

   Thanks to the RTP validity check, STUN packets will be ignored by the
   RTP stack.

   o  The sending agent needs to support STUN.

4.5.  RTP Packet with Incorrect Version Number

   The application sends an RTP packet with an incorrect version number,
   which value is zero.

   Based on RTP specification [RFC3550], the peer should perform a
   header validity check, and therefore ignore these types of packet.

   o  Only four version numbers are possible.  Using one of them for RTP
      keepalive would be wasteful.
   o  [RFC4566] and [RFC3264] mandate not to send media with inactive
      and recvonly attributes, however this is mitigated as no real
      media is sent with this mechanism.

4.6.  RTP Packet with Unknown Payload Type

   The application sends an RTP packet of 0 length with a dynamic
   payload type that has not been negotiated by the peers (e.g. not
   negotiated within the SDP offer/answer, and thus not mapped to any
   media format).

   The sequence number is incremented by one for each packet, as it is
   sent within the same RTP session as the actual media.  The timestamp
   contains the same value a media packet would have at this time.  The
   marker bit is not significant for the keepalive packets and is thus
   set to zero.

   The SSRC is the same as for the media for which keepalive is sent.

   Normally the peer will ignore this packet, as RTP [RFC3550] states
   that "a receiver MUST ignore packets with payload types that it does
   not understand".

   o  [RFC4566] and [RFC3264] mandate not to send media with inactive
      and recvonly attributes, however this is mitigated as no real
      media is sent with this mechanism.
   o  [RFC3550] does not preclude examination of received packets by the
      peer in an attempt to determine if it is under attack.
   o  The statement "RTP Packet with Unknown Payload Type" of RFC3550 is
      not always observed in real life.
   o  There is no RTCP reporting for the keepalive packets as RFC3550
      mandates to ignore "RTP Packet with Unknown Payload Type".
   o  Some RTP payload formats do not handle gaps in RTP sequence number

5.  Recommended Solution for Keepalive Mechanism

   The RECOMMENDED mechanism is the "RTCP packets multiplexed with RTP
   packets" (Section 4.3).  This mechanism is desirable because it
   reduces the number of ports when RTP and RTCP are used.  It also has
   the advantage of taking into account RTCP aspects, which is not the
   case of other mechanisms.

   Other mechanisms (Section 4.1, Section 4.2, Section 4.4, Section 4.5,
   Section 4.6) are NOT RECOMMENDED.

6.  Media Format Exceptions

   When a given media format does not allow the keepalive solution
   recommended in Section 5, an alternative mechanism SHOULD be defined
   in the payload format specification for this media format.

7.  Timing and Transport Considerations

   An application supporting this specification MUST transmit either
   keepalive packets or media packets at least once every Tr seconds
   during the whole duration of the media session.

   Tr has different value according to the transport protocol

   For UDP, the minimum RECOMMENDED Tr value is 15 seconds, and Tr
   SHOULD be configurable to larger values.

   For TCP, the recommended Tr value is 7200 seconds.

   When using the "RTCP packets multiplexed with RTP packets" solution
   for keepalive, Tr MUST comply with the RTCP timing rules of

   Keepalive packets within a particular RTP session MUST use the tuple
   (source IP address, source TCP/UDP ports, target IP address, target
   TCP/UDP Port) of the regular RTP packets.

   The agent SHOULD only send RTP keepalive when it does not send
   regular RTP packets.

8.  RTCP Flow Keepalive

   RTCP packets are sent periodically and can thus normally maintain the
   NAT mappings open as long as they are sent frequently enough.  There
   are two conditions for that.  First RTCP needs to be used bi-
   directionally and in a symmetric fashion, as described in [RFC4961].
   Secondly, RTCP needs to be sent frequently enough.  However, there
   are certain configurations that can break this latter assumption.

   There are two factors that need to be considered to ensure that RTCP
   is sent frequently enough.  First the RTCP bandwidth needs to be
   sufficiently large so that transmission will occur more frequently
   than the longest acceptable packet transmission interval (Tr).  The
   worst case RTCP interval (Twc) can be calculated using this formula
   by inserting the max value of the following parameters:
   o  Maximum RTCP packet size (avg_rtcp_size_max)
   o  Maximum number of participants (members_max)
   o  RTCP receiver bandwidth (rtcp_bw)

   The RTCP bandwidth value to use here is for a worst case, which will
   be the receiver proportion when all members are not senders except
   one.  This can be approximated to be all members.  Thus for sessions
   where RR and RS values are used, then rtcp_bw shall be set to RR.
   For sessions where the [RFC3550] defines proportions of 1/4 for
   sender and 3/4 for receivers are used, then rtcp_bw will be 5% of 3/4
   of the AS value in bits per second.

   Twc = 1.5 / 1.21828 * members_max * rtcp_bw / avg_rtcp_size_max * 8

   The second factor is the minimum RTCP interval Tmin defined in
   [RFC3550].  Its base value is 5 seconds, but it might also be scaled
   to 360 divided by the session bandwidth in kbps.  The Extended RTP
   Profile for Real-time Transport Control Protocol (RTCP)-Based
   Feedback (RTP/AVPF) [RFC4585] also allows for the setting of a trr-
   int parameter which is a minimal RTCP interval for regular RTCP
   packets.  It is also used as the Tmin value in the regular Td
   calculation.  An analysis of the algorithm gives that the longest
   possible regular RTCP interval possible are:

   RTCP_int_max = trr-int * 1.5 + Td * 1.5 / 1.21828

   And as long as the there is sufficient bandwidth according to
   criteria 1, then this can be simplified by setting Td = trr-int

   RTCP_int_max = trr-int * (1.5 + 1.5 / 1.21828) = 2.73123 * trr-int

   Thus the requirements on the RTCP parameters are the following for
   functioning keepalive:
   1.  Ensure that sufficient RTCP bandwidth is provided by calculating
       Twc and ensure that this is less than or equal to Tr.
   2.  If AVP or SAVP is used the Tmin value can't be greater that Tr
       divided by 1.5 / (e-3/2).
   3.  If AVPF or SAVPF is to be used trr-min must not be set to a
       greater value than Tr / 3.

9.  Security Considerations

   The RTP keepalive packets are sent on the same path as regular RTP
   media packets and may be perceived as an attack by a peer.  However,
   [RFC3550] mandates a peer to "ignore packets with payload types that
   it does not understand".  A peer that does not understand the
   keepalive message will thus appropriately drop the received packets.


10.  IANA Considerations



11.  Acknowledgements

   Jonathan Rosenberg provided the major inputs for this draft via the
   ICE specification.  Magnus Westerlund provided the text for the RTCP
   flow keepalive section.  In addition, thanks to Alfred E. Heggestad,
   Colin Perkins, Dan Wing, Gunnar Hellstrom, Hadriel Kaplan, Magnus
   Westerlund, Randell
   Jesup, Remi Denis-Courmont, Robert Sparks, and Steve Casner for their
   useful inputs and comments.


12.  References


12.1.  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.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405,
              November 2008.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.


12.2.  Informative references

   [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.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.

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

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

   [RFC4585]  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.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5382]  Guha, S., Biswas, K., Ford, B., Sivakumar, S., and P.
              Srisuresh, "NAT Behavioral Requirements for TCP", BCP 142,
              RFC 5382, October 2008.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

Authors' Addresses

   Xavier Marjou
   France Telecom Orange
   2, avenue Pierre Marzin
   Lannion  22307


   Aurelien Sollaud
   France Telecom Orange
   2, avenue Pierre Marzin
   Lannion  22307