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Network Working Group C. Huitema
Request for Comments: 3605 Microsoft
Category: Standards Track October 2003
Real Time Control Protocol (RTCP) attribute in
Session Description Protocol (SDP)
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
The Session Description Protocol (SDP) is used to describe the
parameters of media streams used in multimedia sessions. When a
session requires multiple ports, SDP assumes that these ports have
consecutive numbers. However, when the session crosses a network
address translation device that also uses port mapping, the ordering
of ports can be destroyed by the translation. To handle this, we
propose an extension attribute to SDP.
1. Introduction
The session invitation protocol (SIP, [RFC3261]) is often used to
establish multi-media sessions on the Internet. There are often
cases today in which one or both ends of the connection are hidden
behind a network address translation device [RFC2766]. In this case,
the SDP text must document the IP addresses and UDP ports as they
appear on the "public Internet" side of the NAT. In this memo, we
will suppose that the host located behind a NAT has a way to obtain
these numbers. A possible way to learn these numbers is briefly
outlined in section 3, however, just learning the numbers is not
enough.
The SIP messages use the encoding defined in SDP [RFC2327] to
describe the IP addresses and TCP or UDP ports used by the various
media. Audio and video are typically sent using RTP [RFC3550], which
requires two UDP ports, one for the media and one for the control
protocol (RTCP). SDP carries only one port number per media, and
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states that "other ports used by the media application (such as the
RTCP port) should be derived algorithmically from the base media
port." RTCP port numbers were necessarily derived from the base
media port in older versions of RTP (such as [RFC1889]), but now that
this restriction has been lifted, there is a need to specify RTCP
ports explicitly in SDP. Note, however, that implementations of RTP
adhering to the earlier [RFC1889] specification may not be able to
make use of the SDP attributes specified in this document.
When the NAT device performs port mapping, there is no guarantee that
the mappings of two separate ports reflects the sequencing and the
parity of the original port numbers; in fact, when the NAT manages a
pool of IP addresses, it is even possible that the RTP and the RTCP
ports may be mapped to different addresses. In order to successfully
establish connections despite the misordering of the port numbers and
the possible parity switches caused by the NAT, we propose to use a
specific SDP attribute to document the RTCP port and optionally the
RTCP address.
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 [RFC2119].
2. Description of the Solution
The main part of our solution is the declaration of an SDP attribute
for documenting the port used by RTCP.
2.1. The RTCP Attribute
The RTCP attribute is used to document the RTCP port used for media
stream, when that port is not the next higher (odd) port number
following the RTP port described in the media line. The RTCP
attribute is a "value" attribute, and follows the general syntax
specified page 18 of [RFC2327]: "a=<attribute>:<value>". For the
RTCP attribute:
* the name is the ascii string "rtcp" (lower case),
* the value is the RTCP port number and optional address.
The formal description of the attribute is defined by the following
ABNF [RFC2234] syntax:
rtcp-attribute = "a=rtcp:" port [nettype space addrtype space
connection-address] CRLF
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In this description, the "port", "nettype", "addrtype" and
"connection-address" tokens are defined as specified in "Appendix A:
SDP Grammar" of [RFC2327].
Example encodings could be:
m=audio 49170 RTP/AVP 0
a=rtcp:53020
m=audio 49170 RTP/AVP 0
a=rtcp:53020 IN IP4 126.16.64.4
m=audio 49170 RTP/AVP 0
a=rtcp:53020 IN IP6 2001:2345:6789:ABCD:EF01:2345:6789:ABCD
The RTCP attribute MAY be used as a media level attribute; it MUST
NOT be used as a session level attribute. Though the examples below
relate to a method that will return only unicast addresses, both
unicast and multicast values are valid.
3. Discussion of the Solution
The implementation of the solution is fairly straightforward. The
questions that have been most often asked regarding this solution are
whether this is useful, i.e., whether a host can actually discover
port numbers in an unmodified NAT, whether it is sufficient, i.e.,
whether or not there is a need to document more than one ancillary
port per media type, and whether why should not change the media
definition instead of adding a new attribute.
3.1. How do we Discover Port Numbers?
The proposed solution is only useful if the host can discover the
"translated port numbers", i.e., the value of the ports as they
appear on the "external side" of the NAT. One possibility is to ask
the cooperation of a well connected third party that will act as a
server according to STUN [RFC3489]. We thus obtain a four step
process:
1 - The host allocates two UDP ports numbers for an RTP/RTCP pair,
2 - The host sends a UDP message from each port to the STUN server,
3 - The STUN server reads the source address and port of the packet,
and copies them in the text of a reply,
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4 - The host parses the reply according to the STUN protocol and
learns the external address and port corresponding to each of the
two UDP ports.
This algorithm supposes that the NAT will use the same translation
for packets sent to the third party and to the "SDP peer" with which
the host wants to establish a connection. There is no guarantee that
all NAT boxes deployed on the Internet have this characteristic.
Implementers are referred to the STUN specification [RFC3489] for an
extensive discussion of the various types of NAT.
3.2. Do we need to Support Multiple Ports?
Most media streams are transmitted using a single pair of RTP and
RTCP ports. It is possible, however, to transmit a single media over
several RTP flows, for example using hierarchical encoding. In this
case, SDP will encode the port number used by RTP on the first flow,
and the number of flows, as in:
m=video 49170/2 RTP/AVP 31
In this example, the media is sent over 2 consecutive pairs of ports,
corresponding respectively to RTP for the first flow (even number,
49170), RTCP for the first flow (odd number, 49171), RTP for the
second flow (even number, 49172), and RTCP for the second flow (odd
number, 49173).
In theory, it would be possible to modify SDP and document the many
ports corresponding to the separate encoding layers. However,
layered encoding is not much used in practice, and when used is
mostly used in conjunction with multicast transmission. The
translation issues documented in this memo apply uniquely to unicast
transmission, and thus there is no short term need for the support of
multiple port descriptions. It is more convenient and more robust to
focus on the simple case in which a media is sent over exactly one
RTP/RTCP stream.
3.3. Why not Expand the Media Definition?
The RTP ports are documented in the media description line, and it
would seem convenient to document the RTCP port at the same place,
rather than create an RTCP attribute. We considered this design
alternative and rejected it for two reasons: adding an extra port
number and an option address in the media description would be
awkward, and more importantly it would create problems with existing
applications, which would have to reject the entire media description
if they did not understand the extension. On the contrary, adding an
attribute has a well defined failure mode: implementations that don't
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understand the "a=rtcp" attribute will simply ignore it; they will
fail to send RTCP packets to the specified address, but they will at
least be able to receive the media in the RTP packets.
4. UNSAF Considerations
The RTCP attribute in SDP is used to enable establishment of RTP/RTCP
flows through NAT. This mechanism can be used in conjunction with an
address discovery mechanism such as STUN [RFC3489]. STUN is a short
term fix to the NAT traversal problem, which requires thus
consideration of the general issues linked to "Unilateral self-
address fixing" [RFC3424].
The RTCP attribute addresses a very specific problem, the
documentation of port numbers as they appear after address
translation by a port-mapping NAT. The RTCP attribute SHOULD NOT be
used for other applications.
We expect that, with time, one of two exit strategies can be
developed. The IETF may develop an explicit "middlebox control"
protocol that will enable applications to obtain a pair of port
numbers appropriate for RTP and RTCP. Another possibility is the
deployment of IPv6, which will enable use of "end to end" addressing
and guarantee that the two hosts will be able to use appropriate
ports. In both cases, there will be no need for documenting a "non
standard" RTCP port with the RTCP attribute.
5. Security Considerations
This SDP extension is not believed to introduce any significant
security risk to multi-media applications. One could conceive that a
malevolent third party would use the extension to redirect the RTCP
fraction of an RTP exchange, but this requires intercepting and
rewriting the signaling packet carrying the SDP text; if an
interceptor can do that, many more attacks are available, including a
wholesale change of the addresses and port numbers at which the media
will be sent.
In order to avoid attacks of this sort, when SDP is used in a
signaling packet where it is of the form application/sdp, end-to-end
integrity using S/MIME [RFC3369] is the technical method to be
implemented and applied. This is compatible with SIP [RFC3261].
6. IANA Considerations
This document defines a new SDP parameter, the attribute field
"rtcp", which per [RFC2327] has been registered by IANA. This
attribute field is designed for use at media level only.
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7. Intellectual Property Statement
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use other technology described in
this document or the extent to which any license under such rights
might or might not be available; neither does it represent that it
has made any effort to identify any such rights. Information on the
IETF's procedures with respect to rights in standards-track and
standards-related documentation can be found in BCP-11. Copies of
claims of rights made available for publication and any assurances of
licenses to be made available, or the result of an attempt made to
obtain a general license or permission for the use of such
proprietary rights by implementers or users of this specification can
be obtained from the IETF Secretariat.
The IETF invites any interested party to bring to its attention any
copyrights, patents or patent applications, or other proprietary
rights which may cover technology that may be required to practice
this standard. Please address the information to the IETF Executive
Director.
8. Acknowledgements
The original idea for using the "rtcp" attribute was developed by Ann
Demirtjis. The document was reviewed by the MMUSIC and AVT working
groups of the IETF.
9. References
9.1. Normative References
[RFC1889] Schulzrinne, H., Casner, S., Frederick, R. and V.
Jacobson. "RTP: A Transport Protocol for Real-Time
Applications", RFC 1889, January 1996.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
[RFC2327] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
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[RFC3489] Rosenberg, J., Weinberger, J., Huitema, C. and R. Mahy.
"STUN - Simple Traversal of User Datagram Protocol (UDP)
Through Network Address Translators (NATs)", RFC 3489,
March 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R. and V.
Jacobson. "RTP: A Transport Protocol for Real-Time
Applications", RFC 3550, July 2003.
9.2. Informative References
[RFC2766] Tsirtsis, G. and P. Srisuresh. "Network Address
Translation - Protocol Translation (NAT-PT)", RFC 2766,
February 2000.
[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.
[RFC3369] Housley, R., "Cryptographic Message Syntax (CMS)", RFC
3369, August 2002.
[RFC3424] Daigle, L., "IAB considerations for UNilateral Self-
Address Fixing (UNSAF) across network address
translation", RFC 3424, November 2002.
10. Author's Address
Christian Huitema
Microsoft Corporation
One Microsoft Way
Redmond, WA 98052-6399
EMail: huitema@microsoft.com
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11. Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assignees.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
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