One document matched: draft-andreasen-mmusic-connectivityprecondition-02.txt
Differences from draft-andreasen-mmusic-connectivityprecondition-01.txt
Internet Engineering Task Force Flemming Andreasen
MMUSIC Working Group Dave Oran
INTERNET-DRAFT Dan Wing
EXPIRES: August 2005 Cisco Systems
February, 2005
Connectivity Preconditions for
Session Description Protocol Media Streams
<draft-andreasen-mmusic-connectivityprecondition-02.txt>
Status of this memo
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Copyright Notice
Copyright (C) The Internet Society (2005). All Rights Reserved.
Abstract
This document defines a new connectivity precondition for the
Session Description Protocol precondition framework described in RFC
3312. A connectivity precondition can be used to delay session
establishment or modification until media stream connectivity has
been verified successfully.
INTERNET-DRAFT Connectivity Preconditions February, 2005
1 Notational Conventions............................................2
2 Introduction......................................................2
3 Connectivity Precondition Definition..............................2
3.1 Verifying Connectivity........................................4
4 Examples..........................................................5
5 Security Considerations...........................................8
6 IANA Considerations...............................................9
7 Acknowledgements..................................................9
8 Authors' Addresses................................................9
9 Normative References..............................................9
10 Informative References..........................................9
11 Intellectual Property Statement................................10
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
The concept of a Session Description Protocol (SDP) [SDP]
precondition in the Session Initiation Protocol (SIP) [SIP] is
defined in [RFC3312] and [RFC3312upd]. A precondition 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 one or more Network Address Translators (NATs) or
firewalls in the media path, the media stream may not be received by
the far end. The connectivity precondition defined in this document
ensures, that session progress is delayed until media stream
connectivity has been verified, or the session itself is abandoned.
3 Connectivity Precondition Definition
The connectivity precondition type is defined by the string "cntv"
and hence we modify the grammar found in RFC 3312 as follows:
precondition-type = "cntv" | "qos" | token
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RFC 3312 defines support for two kinds of status types, namely
segmented and end-to-end. The connectivity precondition-type
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 mandatory 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 delay of session establishment defined here implies that
alerting of the called party MUST NOT occur until the precondition
has been satisfied. Packets may be both sent and received on the
media streams in question, however such packets SHOULD be limited to
packets that are necessary to verify connectivity between the two
endpoints involved on the media stream, i.e. the underlying media
stream SHOULD NOT be cut through. For example, STUN packets [STUN],
RTP No-Op packets and corresponding RTCP reports, as well as TCP SYN
and ACK packets can be exchanged on media streams that support them
as a way of verifying connectivity.
The direction attributes defined in RFC 3312 are interpreted as
follows:
* send: This party is sending packets on the media stream to the
other party, and the other party has received at least one of
those packets, i.e., there is connectivity in the forward
(sending) direction.
* recv: The other party is sending packets on the media stream to
this party, and this party has received at least one of those
packets, i.e., there is connectivity in the backwards (receiving)
direction.
When the media stream consists of multiple destination addresses,
connectivity to all of them MUST be verified in order for the
precondition to be met. In the case of RTP-based media streams,
RTCP connectivity however is not a requirement.
Note that a "send" connectivity precondition from the offerer's
point of view corresponds to a "recv" connectivity precondition from
the answerer's point of view, and vice versa. If media stream
connectivity in both directions is required before session
establishment or modification continues, the desired status MUST be
set to "sendrecv".
Connectivity preconditions may have a strength-tag of either
"mandatory" or "optional". When a mandatory connectivity
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precondition is offered, and the answerer cannot satisfy the
connectivity precondition, e.g., because the offer does not include
parameters that enable connectivity to be verified without media cut
through, the offer MUST be rejected as described in RFC 3312. When
an optional connectivity precondition is offered, the answerer MUST
generate its answer SDP as soon as possible; since session progress
is not delayed in this case, it is not known whether the associated
media streams will have connectivity. If the answerer wants to
delay session progress until connectivity has been verified, the
answerer MUST increase the strength of the connectivity precondition
by using a strength-tag of "mandatory" in the answer.
Note that use of a "mandatory" precondition requires the presence
of a SIP "Require" header with the option tag "precondition": Any
SIP UA that does not support a mandatory precondition will reject
such requests. To get around this issue, an optional connectivity
precondition and the SIP "Supported" header with the option tag
"precondition" can be used instead.
Offers with connectivity preconditions in re-INVITEs or UPDATEs
follow the rules given in Section 6 of RFC 3312, i.e.:
"Both user agents SHOULD continue using the old session parameters
until all the mandatory preconditions are met. At that moment,
the user agents can begin using the new session parameters."
It should be noted, that connectivity may not exist between two
entities initially, e.g., when one or both entities are behind a
symmetric NAT. Subsequent packet exchanges however may create the
necessary address bindings in the NAT(s) thereby creating
connectivity. The ICE methodology [ICE] for example ensures that
such bindings are created following an offer/answer exchange.
3.1 Verifying Connectivity
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.
Editor's Note: The above obviously only applies to RTP-based media
streams.
The above definitions of send and receive connectivity preconditions
beg two questions: How does the sender of a packet know the other
party received it, and how does the receiver of a packet know who
sent it (in particular, the correlation between an incoming media
packet and a particular SIP dialog may not be obvious). The
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determination depends on the exact method being used to verify
connectivity, however the following provides procedures for three
specific approaches:
* RTP No-Op [no-op]: The sender of an RTP No-Op payload can verify
send connectivity by examining the RTCP report being returned. In
particular, the source SSRC in the RTCP report block is used for
correlation. The RTCP report block also contains the SSRC of the
sender of the report and the SSRC of incoming RTP No-Op packets
identifies the sender of the RTP packet. Thus, once send
connectivity has been ascertained, receipt of an RTP No-Op packet
from the same SSRC provides the necessary correlation to determine
receive connectivity. Alternatively, the duality of send and
receive preconditions can be exploited, with one side confirming
when his send precondition is satisfied, which in turn implies the
other sides recv precondition is satisfied.
* ICE [ICE]: The STUN binding request message sent to check
connectivity contains a transaction ID which is returned in the
STUN binding response, thus send connectivity is verified easily.
STUN binding requests also contain a username and a password which
ICE communicates via SIP. When an incoming STUN message is
received, it is therefore easy to determine the source of that
message and hence receive connectivity can be determined that way.
ICE presents the peer with a number of alternative candidate
addresses for a particular media stream. Once connectivity has
been verified for one of those candidate addresses, connectivity
has been verified, regardless of whether this candidate address is
the one that ends up being used. If a media stream consists of
multiple destination addresses, verification of a candidate
address for each must occur in order for the precondition to be
satisfied.
* TCP [TCP]: TCP connections are bidirectional and hence there
is no difference between send and recv connectivity preconditions.
Once the TCP three-way hand shake has completed (SYN, SYN-ACK,
ACK), the TCP connection is established and data can be sent and
received by either party, i.e. both a send and a receive
connectivity precondition has been satisfied.
4 Examples
The call flow of Figure 1 shows a basic session establishment with
the Session Initiation Protocol using SDP connectivity preconditions
and RTP No-Op. Note that not all SDP details are provided in the
following.
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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
SDP1: A includes a mandatory 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". A's
local status table (see RFC 3312) for the connectivity precondition
is as follows:
Direction | Current | Desired Strength | Confirm
-----------+----------+------------------+----------
send | no | mandatory | no
recv | no | mandatory | no
and the resulting offer SDP is:
m=audio 20000 RTP/AVP 0 96
c=IN IP4 192.0.2.1
a=rtpmap:96 no-op/8000
a=curr:cntv e2e none
a=des:cntv mandatory e2e sendrecv
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SDP2: When B receives the offer, B sees the mandatory sendrecv
connectivity precondition. B can ascertain connectivity to A
("send" from B's point of view) by use of the RTP No-Op, however B
wants A to inform it about connectivity in the other direction
("recv" from B's point of view). B's local status table therefore
looks as follows:
Direction | Current | Desired Strength | Confirm
-----------+----------+------------------+----------
send | no | mandatory | no
recv | no | mandatory | no
Since B wants to ask A for confirmation about the "recv" (from B's
point of view) connectivity precondition, the resulting answer SDP
becomes:
m=audio 30000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.4
a=curr:cntv e2e none
a=des:cntv mandatory e2e sendrecv
a=conf:cntv e2e recv
Meanwhile, B performs a connectivity check to A, which succeeds and
hence B's local status table is updated as follows:
Direction | Current | Desired Strength | Confirm
-----------+----------+------------------+----------
send | yes | mandatory | no
recv | no | mandatory | no
Since the "recv" connectivity precondition (from B's point of view)
is still not satisfied, session establishment remains suspended.
SDP3: When A receives the answer SDP, A notes that confirmation was
requested for B's "recv" connectivity precondition, which is the
"send" precondition from A's point of view. A performs a
connectivity check to B, which succeeds, and A's local status table
becomes:
Direction | Current | Desired Strength | Confirm
-----------+----------+------------------+----------
send | yes | mandatory | yes
recv | no | mandatory | no
Since B asked for confirmation about the "send" connectivity (from
A's point of view), A now sends an UPDATE (5) to B to confirm the
connectivity from A to B:
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m=audio 20000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.1
a=curr:cntv e2e send
a=des:cntv mandatory e2e sendrecv
SDP4: Upon receiving the updated offer, B now knows that there is
connectivity from A to B and updates the local status table as
follows ("send" from A corresponds to "recv" from B's point of
view):
Direction | Current | Desired Strength | Confirm
-----------+----------+------------------+----------
send | yes | mandatory | no
recv | yes | mandatory | no
B 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:cntv e2e sendrecv
a=des:cntv mandatory e2e sendrecv
At this point in time, session establishment resumes and B returns a
180 (Ringing) response (7).
5 Security Considerations
In addition to the general security considerations for preconditions
provided in RFC 3312, the following security issues, which are
specific to connectivity preconditions, should be considered.
Connectivity preconditions rely on mechanisms beyond SDP, e.g. RTP
No-Op [no-op] or STUN [stun], to establish and verify connectivity
between an offerer and an answerer. An attacker that prevents those
mechanism from succeeding can prevent media sessions from being
established and hence it is RECOMMENDED that such mechanisms are
adequately secured by message authentication and integrity
protection. Also, the mechanisms SHOULD consider how to prevent
denial of service attacks. Similarly, an attacker that can forge
packets for these mechanisms can enable sessions to be established
when there in fact is no media connectivity, which may lead to a
poor user experience. Authentication and integrity protection of
such mechanisms can prevent this type of attacks and hence use of it
is RECOMMENDED.
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6 IANA Considerations
IANA is hereby requested to register a RFC 3312 precondition type
called "cntv" 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.
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.
[SIP] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.
Peterson, R. Sparks, M. Handley, E. Schooler, "SIP: Session
Initiation Protocol", RFC 3261, June 2002.
10 Informative References
[RFC3551] H. Schulzrinne, and S. Casner "RTP Profile for Audio and
Video Conferences with Minimal Control", RFC 3550, July 2003.
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[no-op] F. Andreasen, D. Oran, and D. Wing, "RTP No-Op Payload
Format", Work in Progress
[stun] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy, "STUN -
Simple Traversal of User Datagram Protocol (UDP) Through Network
Address Translators (NATs)", RFC 3489, March 2003.
[RFC3312upd] G. Camarillo and P. Kyzivat, "Update to the Session
Initiation Protocol (SIP) Preconditions Framework", IETF, work in
progress.
[ICE] J. Rosenberg, "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Multimedia Session Establishment Protocols", IETF, work in progress.
[TCP] J. Postel, "Transmission Control Protocol", RFC 793,
September 1981.
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