BEHAVE Working Group X. Marjou, Ed.
Internet-Draft France Telecom
Intended status: Informational February 2, 2007
Expires: August 6, 2007
Application Mechanism for maintaining alive the Network Address
Translator (NAT) mappings associated to RTP flows.
draft-marjou-behave-app-rtp-keepalive-01
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Copyright (C) The IETF Trust (2007).
Abstract
This document defines a mechanism that enables applications using
Real Time Protocol (RTP) to maintain their RTP Network Address
Translator (NAT) mappings alive.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. List of Alternatives for Performing RTP Keepalive . . . . . . . 4
4.1. UDP Packet of 0-byte . . . . . . . . . . . . . . . . . . . 5
4.2. RTCP Packets Multiplexed with RTP Packets . . . . . . . . . 5
4.3. STUN Packet . . . . . . . . . . . . . . . . . . . . . . . . 5
4.4. RTP Packet with Comfort Noise Payload . . . . . . . . . . . 5
4.5. RTP Packet with No-Op Payload . . . . . . . . . . . . . . . 6
4.6. RTP Packet with Incorrect Version Number . . . . . . . . . 6
4.7. RTP Packet with Unknown Payload Type . . . . . . . . . . . 6
5. Recommended Solution . . . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . . 7
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 7
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 8
Intellectual Property and Copyright Statements . . . . . . . . . . 9
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1. Introduction
[Note: The content of this draft is basically a copy and paste of the
current 7.12 section of ICE [5] concerning binding keepalives
requirements that apply to a non ICE agent, or that apply to an ICE
agent that communicates with a non-ICE agent. It thus makes sense to
extract it in a separate document so that non-ICE agents can refer to
non-ICE specification.]
Documents [2] and [3] describe NAT behaviors and point-out that two
key aspects of NAT are mappings (a.k.a. bindings) and their
refreshment. 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 maintain the NAT
mappings alive.
When applied to applications using RTP [4], the RTP media stream
packets themselves normally fulfill this requirement. However, as
described in ICE [5], there exist some cases where RTP do not
generate a minimum flow activity.
The examples are:
o Firstly, in some RTP usages, such as SIP, the media streams can be
"put on hold". This is accomplished by using the SDP "sendonly"
or "inactive" attributes, as defined in RFC 3264 [6]. RFC 3264
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 Secondly, some RTP payload formats, such as the payload format for
text conversation [7], may send packets so infrequently that the
interval exceeds the NAT binding timeouts.
o Thirdly, if silence suppression is in use, long periods of silence
may cause media transmission to cease sufficiently long for NAT
bindings to time out.
This document first states the requirements that must be supported to
perform RTP keepalives (Section 3). In a second step, several
alternatives are laid-out to overcome this problem (Section 4).
Finally a single solution is recommended, in order to achieve
interoperability (Section 5).
The scope of the draft is 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
(SIP).
o RTCP flows.
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* Recall that [4] recommends a minimum interval of 5 seconds and
that "on hold" procedures of [6] do not impact RTCP
transmissions. Therefore, when in use, there is always some
RTCP flow activity.
o Other types of flows, such as the Binary Floor Control Protocol
(BFCP)
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 [1]
3. Requirements
This section outlines the key requirements that the solution need to
satisfy in order to provide RTP media keepalive.
REQ 1. The recommended mechanism MUST generate activity within the
RTP media stream
REQ 2. The activity is generated periodically for the whole duration
of the RTP media stream.
REQ 3. Any type of transport (e.g. UDP, TCP) MUST be supported.
REQ 4. Any type media of stream (e.g. audio, video, text) MUST be
supported.
REQ 5. Any type of payload format (e.g. G.711, H.263) MUST be
supported.
REQ 6: Session signaling protocols SHOULD not be impacted.
REQ 7: Session description protocols SHOULD not be impacted.
REQ 8: Impacts on existing software SHOULD be minimized.
REQ 9: Remote peer SHOULD not be impacted.
REQ 10: One single mechanism MUST be recommended.
4. List of Alternatives for Performing RTP Keepalive
This section lists some alternatives that could be used in order to
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perform a keepalive message within RTP media streams.
A common drawback of most of these alternatives is that they require
media packets be sent by the application during "on hold" procedures,
which violates the behavior of the inactive and recvonly attributes
specified in SDP-NEW [10] and in RFC3264 [6]. Although there can
exist some debate whether STUN is a media flow or not, STUN also
requires the application to send some packets within the media stream
during on-hold procedures.
4.1. UDP Packet of 0-byte
The application sends an empty UDP packet.
Cons:
o This alternative is specific to UDP.
o There may be some implementations that will not ignore these
packets.
4.2. RTCP Packets Multiplexed with RTP Packets
The application sends RTCP packets in the RTP media stream itself
(i.e. same tuples for both RTP and RTCP packets) [8]. RTCP packets
therefore maintain the NAT mappings open.
Cons:
o Multiplexing RTP and RTCP must be supported by the remote peer.
o Multiplexing RTP and RTCP must be signalled in SDP offer/answer.
o This alternative may significantly impact existing software and
specifications.
4.3. STUN Packet
The application sends a STUN Binding Request packet and receives a
STUN Binding Response [9]
Cons:
o This alternative requires that the remote endpoint support STUN.
4.4. RTP Packet with Comfort Noise Payload
The application sends a RTP packet with a comfort-noise payload [11].
Cons:
o This alternative is limited to voice payload formats only.
o For each payload type, the content of the payload needs to be
specified.
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4.5. RTP Packet with No-Op Payload
The application sends a RTP No-OP payload [12] .
Cons:
o This payload type needs to be supported by the remote peer.
o This payload type needs to be signalled in SDP offer/answer.
4.6. RTP Packet with Incorrect Version Number
The application sends a RTP with an incorrect version number.
Based on RTP specification [4], the peer should perform a header
validity check, and therefore ignore these types of packet.
Cons:
o Only four version numbers are possible. Using one of them for RTP
keepalive would be wasteful.
4.7. RTP Packet with Unknown Payload Type
The application sends a RTP packet with an unknown payload type.
Normally the peer will ignore it, as RTP [4] states that "a receiver
MUST ignore packets with payload types that it does not understand".
For example, the keepalive RTP packets contain a dynamic payload type
that has not been negotiated for the session.
[Note: more details on the selection of the payload type are needed
here.]
Cons:
o None
5. Recommended Solution
An application supporting this specification MUST send keepalive
packets under the form of ... [Note: The recommended solution needs
to be discussed. However recommending a single method among the
alternatives of the previous section is the best in term of
interoperability. Proposal is the alternative of Section 4.7]
Keepalives packets MUST be sent for each RTP stream regardless of
whether the media stream is currently inactive, sendonly, recvonly or
sendrecv.
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Keepalives packets within a particular RTP media stream MUST use the
tuple (source IP address, source TCP/UDP ports, target IP address,
target TCP/UDP Port) of the regular RTP packets."
Keepalive packets MUST be sent every Tr seconds. Tr SHOULD be
configurable, and otherwise MUST default to 15 seconds. [Note: same
value as in [5].]
An application starts sending keepalives packet as soon as the first
regular RTP packet of the media stream has been sent. It ceases
sending these keepalives packet when the media stream is disabled, or
when the communication terminates.
6. Security Considerations
T.B.D.
7. Acknowledgements
Jonathan Rosenberg, via the ICE specification, provided the major
inputs for this draft. In addition, thanks to the following folks
for useful inputs and comments: Dan Wing, and Aurelien Sollaud.
8. References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", RFC 2119, March 1997.
[2] Audet, F. and C. Jennings, "Network Address Translation (NAT)
Behavioral Requirements for Unicast UDP", RFC 4787,
January 2007.
[3] Guha, S., Biswas, K., Ford, B., Francis, P., Sivarkumar, S.,
and P. Srisuresh, "NAT Behavioral Requirements for TCP",
draft-ietf-behave-tcp-04 (work in progress), January 2007.
[4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications",
RFC 3550, July 2003.
[5] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols", draft-ietf-mmusic-ice-13 (work in
progress), January 2007.
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[6] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
the Session Description Protocol (SDP)", RFC 3264, June 2002.
[7] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, June 2005.
[8] Perkins, C. and M. Magnus, "Multiplexing RTP Data and Control
Packets on a Single Port", draft-ietf-avt-rtp-and-rtcp-mux-03
(work in progress), December 2006.
[9] Rosenberg, J., Huitema, C., Mahy, R., and D. Wing, "Simple
Traversal Underneath Network Address Translators (NAT) (STUN)",
draft-ietf-behave-rfc3489bis-05 (work in progress),
October 2006.
[10] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[11] Robert, R., "Real-time Transport Protocol (RTP) Payload for
Comfort Noise (CN)", RFC 3389, September 2002.
[12] Andreason, F., Oran, D., and D. Wing, "A No-Op Payload Format
for RTP", draft-ietf-avt-rtp-no-op-00 (work in progress),
May 2005.
Author's Address
Xavier Marjou (editor)
France Telecom
2, hent Pierre Marzin
Lannion, Brittany 22307
France
Email: xavier.marjou@orange-ftgroup.com
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