One document matched: draft-andreasen-mmusic-connectivityprecondition-00.txt
Internet Engineering Task Force Flemming Andreasen
MMUSIC Working Group Dave Oran
INTERNET-DRAFT Dan Wing
EXPIRES: August 2004 Cisco Systems
February, 2004
Connectivity Preconditions for
Session Description Protocol Media Streams
<draft-andreasen-mmusic-connectivityprecondition-00.txt>
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Copyright Notice
Copyright (C) The Internet Society (2004). All Rights Reserved.
Abstract
This document defines a new connectivity precondition for the
Session Description Protocol precondition framework described in RFC
3312. A connecitivity precondition can be used to delay session
establishment or modification until media stream connectivity has
been verified successfully.
INTERNET-DRAFT Connectivity Preconditions February, 2004
1. Notational Conventions..........................................2
2. Introduction....................................................2
3. Connectivity Precondition Definition............................2
4. Examples........................................................3
5. Security Considerations.........................................5
6. IANA Considerations.............................................5
7. Acknowledgements................................................5
8. Authors' Addresses..............................................6
9. Normative References............................................6
10. Informative References..........................................6
Intellectual Property Statement......................................7
Acknowledgement......................................................8
1. Notational Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "MUST", "MUST NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
2. Introduction
RFC 3312 defines the concept of a Session Description Protocol (SDP)
[SDP] precondition, which is a condition that has to be satisfied
for a given media stream in order for session establishment or
modification to proceed. When the precondition is not met, session
progress is delayed until the precondition is satisfied, or the
session establishment fails. For example, RFC 3312 defines the
Quality of Service precondition, which is used to ensure
availability of network resources prior to establishing (i.e.
alerting) a call.
SIP sessions are typically established in order to setup one or more
media streams. Even though a media stream may be negotiated
successfully, the actual media stream itself may fail. For example,
when there is a NAT or firewall in the media path, the media stream
may not be received. The connectivity precondition defined in this
document ensures, that session progress is delayed until media
stream connectivity is established, or the session itself is
abandoned.
3. Connectivity Precondition Definition
The connectivity precondition type is defined by the string "con"
and hence we modify the grammar found in RFC 3312 as follows:
precondition-type = "con" | "qos" | token
RFC 3312 defines support for two kinds of status types, namely
segmented and end-to-end. The connectivity precondition-type
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defined here MUST be used with the end-to-end status type; use of
the segmented status type is undefined.
An entity that wishes to delay session establishment or modification
until media stream connectivity has been established uses this
precondition-type in an offer. When a connectivity precondition is
received in an offer, session establishment or modification MUST be
delayed until the connectivity precondition has been met, i.e. media
stream connectivity has been established in the desired
direction(s).
The direction attributes are interpreted as follows:
* send: The offerer/answerer is sending media stream packets to the
other party, and the offerer/answer knows the other party is
receiving those media stream packets.
* recv: The offerer/answerer knows that the other party has
ascertained media stream connectivity to it.
If media stream connectivity in both directions is required, the
desired status should be set to "sendrecv".
Media stream connectivity can be ascertained in different ways and
this document does not mandate any particular mechanism for doing
so. It is however RECOMMENDED that the No-Op RTP payload format
defined in [no-op] is supported by entities that support
connectivity preconditions. This will ensure that all entities that
support the connectivity preconditions have at least one common way
of ascertaining connectivity.
4. Examples
The call flow of Figure 1 shows a basic session establishment using
connectivity preconditions and using RTP no-op. Note that not all
SDP details are provided in the following.
SDP1: A includes the end-to-end connectivity precondition with a
desired status of "sendrecv"; this will ensure media stream
connectivity in both directions before continuing with the session
setup. Since media stream connectivity in either direction is
unknown at this point, the current status is set to "none":
m=audio 20000 RTP/AVP 0 96
c=IN IP4 192.0.2.1
a=rtpmap:96 no-op/8000
a=curr:con e2e none
a=des:con mandatory e2e sendrecv
SDP2: When B receives the offer, B sees the bidirectional
connectivity preconditions. B can ascertain connectivity to A by
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use of the RTP no-op, however B needs A to inform it about
connectivity in the other direction. Consequently, B includes a
request for confirmation in the "receive" direction:
m=audio 30000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.4
a=curr:con e2e none
a=des:con mandatory e2e sendrecv
a=conf:con e2e recv
Meanwhile, B performs a connectivity check to A, which succeeds.
Also, A performs a connectivity check to B, which also succeeds.
SDP3: A now sends an UPDATE (5) to B to confirm the connectivity
from A to B, as requested by B:
m=audio 20000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.1
a=curr:con e2e send
a=des:qos mandatory e2e sendrecv
SDP4: Upon receiving the updated offer, B now knows that there is
connectivity from A to B; connectivity from B to A was verified
earlier. B therefore responds with an answer (6) which contains the
current status of the connectivity precondition (i.e., sendrecv)
from B's point of view:
m=audio 30000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.4
a=curr:con e2e sendrecv
a=des:con mandatory e2e sendrecv
At this point in time, session establishment resumes and B returns a
180 (Ringing) response (7).
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A B
| |
|-------------(1) INVITE SDP1--------------->|
| |
|<------(2) 183 Session Progress SDP2--------|
| |
|<~~~~~ Connectivity check to A ~~~~~~~~~~~~~|
| |
|----------------(3) PRACK------------------>|
| |
|~~~~~ Connectivity to A OK ~~~~~~~~~~~~~~~~>|
| |
|<-----------(4) 200 OK (PRACK)--------------|
| |
|~~~~~ Connectivity check to B ~~~~~~~~~~~~~>|
|<~~~~ Connectivity to B OK ~~~~~~~~~~~~~~~~~|
| |
|-------------(5) UPDATE SDP3--------------->|
| |
|<--------(6) 200 OK (UPDATE) SDP4-----------|
| |
|<-------------(7) 180 Ringing---------------|
| |
| |
| |
Figure 1: Example using the connectivity precondition
5. Security Considerations
TBD
6. IANA Considerations
IANA is hereby requested to register a RFC 3312 precondition type
called "con" with the name "Connectivity precondition". The
reference for this precondition type is the current document.
7. Acknowledgements
The concept of a "connectivity precondition" is the result of
discussions with numerous people over a long period of time; the
authors greatly appreciate these contributions.
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8. Authors' Addresses
Flemming Andreasen
Cisco Systems, Inc.
499 Thornall Street, 8th Floor
Edison, New Jersey 08837 USA
EMail: fandreas@cisco.com
David Oran
Cisco Systems, Inc.
7 Ladyslipper Lane
Acton, MA 01720 USA
EMail: oran@cisco.com
Dan Wing
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134 USA
EMail: dwing@cisco.com
9. Normative References
[RFC3312] G. Camarillo, W. Marshall, J. Rosenberg, "Integration of
Resource Management and Session Initiation Protocol (SIP)", RFC
3312, October 2002.
[RFC2327] M. Handley and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
10. Informative References
[RFC3551] H. Schulzrinne, and S. Casner "RTP Profile for Audio and
Video Conferences with Minimal Control", RFC 3550, July 2003.
[NO-OP] F. Andreasen, D. Oran, and D. Wing, "RTP No-Op Payload
Format", February 2004, http://search.ietf.org/internet-drafts/
draft-wing-avt-rtp-noop-00.txt, Work in Progress
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Acknowledgement
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