MMUSIC Working Group F. Andreasen
Internet-Draft Cisco System, Inc.
Expires: December 3, 2006 G. Camarillo
Ericsson
D. Oran
Cisco Systems, Inc
D. Wing
Cisco Systems, Inc.
June 1, 2006
Connectivity Preconditions for Session Description Protocol Media
Streams
draft-ietf-mmusic-connectivity-precon-02.txt
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Copyright (C) The Internet Society (2006).
Abstract
This document defines a new connectivity precondition for the Session
Description Protocol precondition framework described in RFC 3312
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(and its update, RFC4032). A connectivity precondition can be used
to delay session establishment or modification until media stream
connectivity has been verified successfully. The method of
verification may vary depending on the type of transport used for the
media. For reliable connection-oriented transports such as TCP
verification is achieved by successful connection establishment. For
unreliable datagram transports such as UDP, verification involves
probing the stream with data or control packets.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Connectivity Precondition Definition . . . . . . . . . . . . . 3
3.1. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.2. Operational semantics . . . . . . . . . . . . . . . . . . 4
3.3. Status type . . . . . . . . . . . . . . . . . . . . . . . 4
3.4. Direction tag . . . . . . . . . . . . . . . . . . . . . . 4
3.5. Precondition strength . . . . . . . . . . . . . . . . . . 5
4. Verifying connectivity . . . . . . . . . . . . . . . . . . . . 6
4.1. Procedures for connection-oriented transports . . . . . . 7
4.2. Procedures for datagram transports . . . . . . . . . . . . 8
5. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 14
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14
8. Change Log . . . . . . . . . . . . . . . . . . . . . . . . . . 15
8.1. Changes since -01 . . . . . . . . . . . . . . . . . . . . 15
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
9.1. Normative References . . . . . . . . . . . . . . . . . . . 15
9.2. Informative References . . . . . . . . . . . . . . . . . . 15
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17
Intellectual Property and Copyright Statements . . . . . . . . . . 18
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1. Introduction
The concept of a Session Description Protocol (SDP) [2] precondition
in the Session Initiation Protocol (SIP) [SIP] is defined in RFC3312
[4] (updated by RFC4032 [6]). 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, RFC3312
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, through an SDP offer-answer exchange, 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. In cases where
the media is carried over a connection-oriented transport such as TCP
[8], the connection-establishment procedures may fail. 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.
The connectivity precondition type defined in this document follows
the guidelines provided in RFC4032 [6] to extend the SIP
preconditions framework.
2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT
RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as
described in BCP 14, RFC 2119 [1] and indicate requirement levels for
compliant implementations.
3. Connectivity Precondition Definition
3.1. Syntax
The connectivity precondition type is defined by the string "conn"
and hence we modify the grammar found in RFC 3312 as follows:
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precondition-type = "conn" | "qos" | token
This precondition tag is registered with the IANA in Section 7.
3.2. Operational semantics
According to RFC4032 [6], documents defining new precondition types
need to describe the behavior of UAs from the moment session
establishment is suspended due to a set of preconditions until is
resumed when these preconditions are met. 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 is 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 does 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.
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.
3.3. Status type
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.
3.4. Direction tag
The direction attributes defined in RFC 3312 are interpreted as
follows:
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o send: The party who generated the session description (the offerer
in an offer-answer exchange) 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.
o recv: The other party (the answerer in an offer-answer exchange)
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.
o sendrecv: Both the send and recv conditions hold. In the case of
a connection-oriented transport such as TCP, once established the
connection would usually have an associated direction tag of
sendrecv because it can carry data in both directions.
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".
3.5. Precondition strength
Connectivity preconditions may have a strength-tag of either
"mandatory" or "optional".
When a mandatory connectivity 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.:
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"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 [7] methodology for example ensures that such
bindings are created following an offer/answer exchange.
4. Verifying connectivity
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) ?
Media stream connectivity can be ascertained in a variety of ways.
This document does not mandate any particular mechanism for doing so,
however the appropriate machinery is likely to vary depending on the
type of transport used for media carriage. In order to comply with
the intent of an endpoint requiring connectivity preconditions, the
following general principles apply:
o The 3-way handshake connection establishment procedures of a
reliable transport protocol such as TCP are usually adequate to
demonstrate bi-directional connectivity (and hence "sendrecv"
media capability). Probe packets sent over the connection are
generally not required to satisfy the precondition.
o A pure datagram transport such as UDP (whether carrying RTP or
some other protocol) by itself provides no useful feedback about
connectivity. Hence, some sort of probe traffic is necessary to
ascertain whether packets are being received successfully.
o Connectivity preconditions are used to verify connectivity based
on the address information exchanged in offers and answers. When
overlapping IP address spaces are used (e.g. because one or both
endpoints are behind a Network Address Translator), it is possible
to inadvertently verify connectivity with an unrelated entity. In
order to address this issue, a correlation mechanism is needed
between media stream packets on one side and offers and answers on
the other side. ICE [7] defines one such correlation mechanism,
however use of it is above and beyond the connection-oriented
connectivity preconditions defined here.
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o Some connection-oriented transport protocols may allow the data
transfer phase to operate in an unreliable mode (today there is no
standards-track IETF protocol which exhibits this characteristic).
In such cases the success of connection establishment may not
definitively demonstrate connectivity in the data phase, and hence
probe traffic MAY be necessary to ascertain if the precondition is
met.
o Hybrid protocols such as DCCP [14] provide their own feedback
channel and initialization procedures, which can serve to verify
connectivity without the use of explicit probe traffic.
The determination depends on the exact method being used to verify
connectivity.
4.1. Procedures for connection-oriented transports
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. Implementations SHOULD NOT require the receipt of
probe traffic in order to consider the precondition satisfied.
SCTP [9] connections have similar semantics as TCP and SHOULD be
treated the same as TCP.
When a connection-oriented transport is part of an offer, it may be
passive, active, or active/passive [12]. When it is passive, the
offerer expects the answerer to initiate the connection
establishment, and when it is active, the offerer wants to initiate
the connection establishment. When it is active/passive, the
answerer decides.
SIP and SDP do not provide any inherent capabilities for associating
an incoming media stream packet with a particular dialog. Thus, when
the offerer is passive and an incoming connection is being
established, the offerer cannot guarantee that the packet is
associated with a particular dialog. When SIP forking is being used,
this implies that the offerer cannot determine which of the early
dialogs now has its recv connectivity precondition satisfied - a
correlation mechanism is missing. This turns out not to be a problem
however, since the successful completion of the connection-
establishment procedure itself (e.g. receipt of SYN-ACK in the case
of TCP) informs the answerer that the precondition has been
satisfied, and hence there is no need for the offerer to explicitly
inform the answerer of this (by sending a SIP UPDATE message). In
the absence of a correlation mechanism (e.g. ICE), an answerer
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therefore MUST NOT require the offerer to confirm a connectivity
precondition on a connection-oriented transport.
4.2. Procedures for datagram transports
Verification of connectivity on datagram transports usually entails
the sending of probe traffic with some form of feedback to inform the
sender whether reception was successful. Techniques that can be used
to verify connectivity on datagram transports include:
o ICE [7]: ICE provides one or more candidate addresses in signaling
between the offerer and the answerer and then uses STUN Binding
Requests to determine which pairs of candidate addresses have
connectivity. Each STUN Binding Request contains a password which
is communicated in the SDP as well; this enables correlation
between STUN Binding Requests and candidate addresses for a
particular media stream. In ICE, connectivity is always checked
in both directions by following a state machine with a set of
states for the offerer and a set of states for the answerer: The
offerer ascertains "recv" connectivity for a particular transport
address pair by transitioning into the "validating" state, whereas
"send" connectivity is ascertained by transitioning into the
"valid" state. The answerer ascertains both "send" and "recv"
connectivity for a particular transport address pair by
transitioning into the "send-valid" state. As a consequence of
this, there is never a need for the answerer to request
confirmation of the connectivity precondition when using ICE: the
answerer can determine the status locally. When ICE is used to
verify connectivity preconditions, the precondition is satisfied
as soon as one of the candidates becomes valid, i.e. connectivity
has been verified for all the component transport addresses used
by the media stream. For example, with an RTP-based media stream
where RTCP is not suppressed, connectivity must be ascertained for
both RTP and RTCP; this is a tightening of the general operational
semantics provided in Section 3.2 imposed by ICE. Finally, it
should be noted, that though connectivity has been ascertained, a
new offer/answer exchange may be required before media can
actually flow (per ICE).
o RTP no-op [13]: The sender of an RTP No-Op payload can verify send
connectivity by examining the RTCP report(s) 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
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when his send precondition is satisfied, which in turn implies the
other sides recv precondition is satisfied.
The above are merely examples of techniques that can be used. Other
techniques which meet the requirements of Section 4 above can be used
as well. It is however RECOMMENDED that ICE be supported by entities
that support connectivity preconditions for datagram transports. Use
of ICE has the benefit of working for all datagram based media
streams (not just RTP) as well as facilitate NAT and firewall
traversal, which may otherwise interfere with connectivity.
Furthermore, the ICE recommendation provides a baseline to ensure
that all entities that require probe traffic to support the
connectivity preconditions have at least one common way of
ascertaining connectivity.
5. Examples
The first example uses the connectivity precondition with TCP in the
context of a session involving a wireless access medium. Both UAs
use a radio access network that does not allow them to send any data
(not even a TCP SYN) until a radio bearer has been setup for the
connection. Figure 1 shows the message flow of this example (the
PRACK transaction has been omitted for clarity):
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A B
| INVITE |
| a=curr:conn e2e none |
| a=des:conn mandatory e2e sendrecv |
| a=setup:holdconn |
|----------------------------------->|
| |
| 183 Session Progress |
| a=curr:conn e2e none |
| a=des:conn mandatory e2e sendrecv |
| a=setup:holdconn |
|<-----------------------------------|
| |
| UPDATE |
| a=curr:conn e2e none |
| a=des:conn mandatory e2e sendrecv |
A's radio | a=setup:actpass |
bearer is +----------------------------------->|
up | |
| 200 OK |
| a=curr:conn e2e none |
| a=des:conn mandatory e2e sendrecv |
| a=setup:active |
|<-----------------------------------|
| |
| |
| |
| | B's radio
|<---TCP Connection Establishment--->+ bearer is up
| | B sends TCP SYN
| |
| |
| 180 Ringing | TCP connection
|<-----------------------------------+ is up
| | B alerts the user
| |
Figure 1: Message flow with two types of preconditions
A sends an INVITE requesting connection-establishment preconditions.
The setup attribute in the offer is set to holdconn because A cannot
send or receive any data before setting up a radio bearer for the
connection.
B agrees to use the connectivity precondition by sending a 183
(Session Progress) response. The setup attribute in the answer is
also set to holdconn because B, like A, cannot send or receive any
data before setting up a radio bearer for the connection.
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When A's radio bearer is ready, A sends an UPDATE to B with a setup
attribute with a value of actpass. This attribute indicates that A
can perform an active or a passive TCP open. A is letting B choose
which endpoint will initiate the connection.
Since B's radio bearer is not ready yet, B chooses to be the one
initiating the connection and indicates so with a setup attribute
with a value of active. At a later point, when B's radio bearer is
ready, B initiates the TCP connection towards A.
Once the TCP connection is established successfully, B alerts the
callee and sends a 180 (Ringing) response.
The second example shows a basic SIP session establishment using SDP
connectivity preconditions and RTP No-Op. Note that not all SDP
details are provided in the following. below shows the message flow
for this scenario shown in Figure 2 below.
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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 2: Connectivity precondition with RTP no-op
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:
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m=audio 20000 RTP/AVP 0 96
c=IN IP4 192.0.2.1
a=rtpmap:96 no-op/8000
a=curr:conn e2e none
a=des:conn mandatory e2e sendrecv
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:conn e2e none
a=des:conn mandatory e2e sendrecv
a=conf:conn 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:
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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:
m=audio 20000 RTP/AVP 0 96
a=rtpmap:96 no-op/8000
c=IN IP4 192.0.2.1
a=curr:conn e2e send
a=des:conn mandatory e2e sendrecv
6. 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.
TCP[8] connection establishment, RTP No-Op [13] or STUN [10], 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.
It is also strongly RECOMMENDED that integrity protection be applied
to the SDP session descriptions. S/MIME [5] is the natural choice to
provide such end-to-end integrity protection, as described in RFC
3261 [3].
7. IANA Considerations
IANA is hereby requested to register a RFC 3312 precondition type
called "conn" with the name "Connectivity precondition". The
reference for this precondition type is the current document.
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8. Change Log
8.1. Changes since -01
There are no changes since the previous version of the document.
9. References
9.1. Normative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[2] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[3] 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.
[4] Camarillo, G., Marshall, W., and J. Rosenberg, "Integration of
Resource Management and Session Initiation Protocol (SIP)",
RFC 3312, October 2002.
[5] Peterson, J., "S/MIME Advanced Encryption Standard (AES)
Requirement for the Session Initiation Protocol (SIP)",
RFC 3853, July 2004.
[6] Camarillo, G. and P. Kyzivat, "Update to the Session Initiation
Protocol (SIP) Preconditions Framework", RFC 4032, March 2005.
[7] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
Methodology for Network Address Translator (NAT) Traversal for
Offer/Answer Protocols", draft-ietf-mmusic-ice-08 (work in
progress), March 2006.
9.2. Informative References
[8] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
[9] Stone, J., Stewart, R., and D. Otis, "Stream Control
Transmission Protocol (SCTP) Checksum Change", RFC 3309,
September 2002.
[10] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy, "STUN
- Simple Traversal of User Datagram Protocol (UDP) Through
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Network Address Translators (NATs)", RFC 3489, March 2003.
[11] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[12] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the
Session Description Protocol (SDP)", RFC 4145, September 2005.
[13] Andreasen, F., "A No-Op Payload Format for RTP",
draft-wing-avt-rtp-noop-03 (work in progress), May 2005.
[14] Kohler, E., "Datagram Congestion Control Protocol (DCCP)",
draft-ietf-dccp-spec-13 (work in progress), December 2005.
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Authors' Addresses
Flemming Andreasen
Cisco System, Inc.
499 Thornall Street, 8th Floor
Edison, NJ 08837
USA
Email: fandreas@cisco.com
Gonzalo Camarillo
Ericsson
Hirsalantie 11
Jorvas 02420
Finland
Email: Gonzalo.Camarillo@ericsson.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 94301
USA
Email: dwing@cisco.com
Andreasen, et al. Expires December 3, 2006 [Page 17]
Internet-Draft Connectivity Precondition June 2006
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Andreasen, et al. Expires December 3, 2006 [Page 18]
