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PROPOSED STANDARD

Internet Engineering Task Force (IETF)                         X. Marjou
Request for Comments: 6263                                    A. Sollaud
Category: Standards Track                          France Telecom Orange
ISSN: 2070-1721                                                June 2011


        Application Mechanism for Keeping Alive the NAT Mappings
        Associated with RTP / RTP Control Protocol (RTCP) Flows

Abstract

   This document lists the different mechanisms that enable applications
   using the Real-time Transport Protocol (RTP) and the RTP Control
   Protocol (RTCP) to keep 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 is an Internet Standards Track document.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Further information on
   Internet Standards is available in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc6263.

Copyright Notice

   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.





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Table of Contents

   1. Introduction ....................................................2
   2. Terminology .....................................................4
   3. Requirements ....................................................4
   4. List of Alternatives for Performing RTP Keepalive ...............4
      4.1. Empty (0-Byte) Transport Packet ............................4
      4.2. RTP Packet with Comfort Noise Payload ......................5
      4.3. RTCP Packets Multiplexed with RTP Packets ..................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. RTCP Flow Keepalive .............................................8
   9. Security Considerations .........................................9
   10. Acknowledgements ...............................................9
   11. References ....................................................10
      11.1. Normative References .....................................10
      11.2. Informative References ...................................10

1.  Introduction

   [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 them.

   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 Initiation Protocol (SIP)
      [RFC3261], 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.  If the agent receiving the media is
      located on the private side of a NAT, it will never receive RTP
      packets from the public peer if the NAT mapping has not been
      created.




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   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 time out, 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 NAT traversal constraints also usually
   require that the agents use 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], solves 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.

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




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2.  Terminology

   In this document, the key words "MUST", "MUST NOT", "REQUIRED",
   "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
   and "OPTIONAL" are to be interpreted as described in RFC 2119
   [RFC2119].

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  The 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.  Empty (0-Byte) Transport Packet

   The application sends an empty transport packet (e.g., UDP packet,
   Datagram Congestion Control Protocol (DCCP) packet).

   Con:

   o  This alternative is specific to each transport protocol.







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4.2.  RTP Packet with Comfort Noise Payload

   The application sends an RTP packet with a comfort noise payload
   [RFC3389].

   Cons:

   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 signaled 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
      chosen.

4.3.  RTCP Packets Multiplexed with RTP Packets

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

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

   Cons:

   o  Multiplexing RTP and RTCP must be supported by the remote peer.

   o  Some RTCP monitoring tools expect that RTCP packets are not
      multiplexed.

   o  RTCP must be configured so that the Tmin value [RFC3550] is less
      than or equal to the Tr interval.














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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.

   Con:

   o  The sending agent needs to support STUN.

4.5.  RTP Packet with Incorrect Version Number

   The application sends an RTP packet with a version number set to zero
   (i.e., an incorrect version number).

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

   Cons:

   o  Only four version numbers are possible.  Using one of them for RTP
      keepalive would be wasteful.

   o  [RFC4566] and [RFC3264] mandate that media with inactive and
      recvonly attributes not be sent; 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 that 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 synchronization source (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".



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   Cons:

   o  [RFC4566] and [RFC3264] mandate that media with inactive and
      recvonly attributes not be sent; 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 "a receiver MUST ignore packets with payload types
      that it does not understand" of [RFC3550] is not always observed
      in real life.

   o  There is no RTCP reporting for the keepalive packets, as [RFC3550]
      mandates that RTP packets with payload types that the receiver
      does not understand be ignored.

   o  Some RTP payload formats do not handle gaps in RTP sequence number
      well.

5.  Recommended Solution for Keepalive Mechanism

   The RECOMMENDED mechanism is that discussed in "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 with other mechanisms.

   Other mechanisms (Sections 4.1, 4.2, 4.4, 4.5, and 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.




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   For TCP, the recommended Tr value is 7200 seconds.

   When using the "RTCP packets multiplexed with RTP packets" solution
   (Section 4.3) for keepalive, Tr MUST comply with the RTCP timing
   rules of [RFC3550].

   Keepalive packets within a particular RTP session MUST use the tuple
   (source IP address, source TCP/UDP port, 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 keep the NAT
   mappings open as long as they are sent frequently enough.  There are
   two conditions for that.  First, RTCP needs to be used
   bidirectionally 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 except one are not
   senders.  This can be approximated to be all members.  Thus, for
   sessions where RR and RS values [RFC3556] are used, then rtcp_bw
   shall be set to RR.  For sessions where the [RFC3550]-defined
   proportions of RTCP bandwidth are used (i.e., 1/4 of the bandwidth
   for senders and 3/4 of the bandwidth for receivers), then rtcp_bw
   will be 5% of 3/4 of the AS value [RFC4566] in bits per second.

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





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   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 shows that the longest
   possible regular RTCP interval is:

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

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

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

   Thus, the requirements for the RTCP parameters are as follows for
   functioning keepalive:

   1.  Ensure that sufficient RTCP bandwidth is provided by calculating
       Twc, and ensure that the resulting value is less than or equal
       to Tr.

   2.  If AVP or SAVP [RFC3711] is used, the Tmin value can't be greater
       than Tr divided by 1.5 / (e-3/2).

   3.  If AVPF or SAVPF [RFC5124] is to be used, trr-min must not be set
       to a value greater 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 that a peer "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.  Acknowledgements

   Jonathan Rosenberg provided the major inputs for this document 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,
   Randell Jesup, Remi Denis-Courmont, Robert Sparks, and Steve Casner
   for their useful inputs and comments.




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11.  References

11.1.  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.

11.2.  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.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 3556, July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [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.



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

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.

   [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., Ed., 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.






















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Authors' Addresses

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

   EMail: xavier.marjou@orange-ftgroup.com


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

   EMail: aurelien.sollaud@orange-ftgroup.com

































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