The Session Initiation Protocol (SIP) "Join" Header
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 (2004).
Abstract
This document defines a new header for use with SIP multi-party
applications and call control. The Join header is used to logically
join an existing SIP dialog with a new SIP dialog. This primitive
can be used to enable a variety of features, for example: "Barge-In",
answering-machine-style "Message Screening" and "Call Center
Monitoring". Note that definition of these example features is non-
normative.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Applicability of RFC 2804 ("Raven"). . . . . . . . . . . . . 3
4. User Agent Server Behavior: Receiving a Join Header . . . . 4
5. User Agent Client Behavior: Sending a Join header . . . . . 6
6. Proxy behavior . . . . . . . . . . . . . . . . . . . . . . . 7
7. Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
7.1. The Join Header . . . . . . . . . . . . . . . . . . . 7
7.2. New option tag for Require and Supported headers . . . 8
8. Usage Examples . . . . . . . . . . . . . . . . . . . . . . . 8
8.1. Join accepted and transitioned to central conference . 9
8.2. Join rejected . . . . . . . . . . . . . . . . . . . . 12
9. Security Considerations . . . . . . . . . . . . . . . . . . 13
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . 14
10.1. Registration of "Join" SIP header. . . . . . . . . . . 14
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10.2. Registration of "join" SIP Option-tag. . . . . . . . . 14
11. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . 14
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 14
12.1. Normative References . . . . . . . . . . . . . . . . . 14
12.2. Informative References . . . . . . . . . . . . . . . . 15
13. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 16
14. Full Copyright Statement . . . . . . . . . . . . . . . . . . 17
1. Introduction
This document describes a SIP [1] extension header field as part of
the SIP multiparty applications architecture framework [12]. The
Join header is used to logically join an existing SIP dialog with a
new SIP dialog. This is especially useful in peer-to-peer call
control environments.
One use of the "Join" header is to insert a new participant into a
multimedia conversation (which may be a two-party call or a SIP
conference [15]). While this functionality is already available
using 3rd party call control [17], style call control, the 3pcc model
requires a central point of control which may not be desirable in
many environments. As such, a method of performing these same call
control primitives in a distributed, peer-to-peer fashion is very
desirable.
Use of an explicit Join header is needed in some cases instead of
addressing an INVITE to a conference URI for the following reasons:
o A conference may not yet exist--the new invitation may be trying
to join an ordinary two-party call.
o The party joining may not know if the dialog it wants to join is
part of a conference.
o The party joining may not know the conference URI.
The Join header enables services such as barge-in, real-time message
screening, and call center monitoring in a distributed peer-to-peer
way. This list of services is not exhaustive.
For example, the Boss has an established 2-party conversation with a
Customer, and using some out-of-band mechanism (e.g., voice,
gestures, or email) asks an Assistant to join the conversation. The
Assistant sends an INVITE with a Join header to the Boss with the
dialog information for the established dialog. The Assistant
obtained this information from some other mechanism, for example a
web-page, an instant message, or from the SIP session dialog package
[13].
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Assistant Boss Customer
| callid: [email protected] | callid: [email protected] |
| | |
| |<============>|
| | |
|INVITE------>| |
|Join: [email protected] | |
| |reINVITE----->|
|<----200-----|<----200------|
|-----ACK---->|<----ACK------|
| | |
| .. begins mixing .. |
| | |
|<===========>|<============>|
|<::::::::::::::::::::::::::>|
Note that this operation effectively creates a new conference. The
Boss needs to cause a new conference to start (and consequently
create or obtain a new conference URI). In our example, the Boss
mixes all media locally, so it needs to generate a new conference
URI, return the conference URI as the Contact to the Join INVITE
(with the "isfocus" Contact header field parameter as defined in [6],
and reINVITE or UPDATE [22] the Customer with the conference URI as
the new Contact. This scenario is also discussed in more detail in
[16].
2. Conventions
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 RFC 2119 [2].
This document refers frequently to the terms "confirmed dialog" and
"early dialog". These are defined in Section 12 of SIP [1].
3. Applicability of RFC 2804 ("Raven")
This primitive can be used to create services which are used for
monitoring purposes, however these services do not meet the
definition of a wiretap according to RFC 2804 [14]. The definition
from RFC 2804 is included here:
Wiretapping is what occurs when information passed across the
Internet from one party to one or more other parties is delivered
to a third party:
1. Without the sending party knowing about the third party
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2. Without any of the recipient parties knowing about the delivery
to the third party
3. When the normal expectation of the sender is that the
transmitted information will only be seen by the recipient
parties or parties obliged to keep the information in
confidence
4. When the third party acts deliberately to target the
transmission of the first party, either because he is of
interest, or because the second party's reception is of
interest.
Specifically, item 2 of this definition does not apply to this
extension, as one party is always aware of a Join request and can
even decline such requests. In addition, in many applications of
this primitive, some or all of the other items may not apply. For
example, in many call centers which handle financial transactions,
all conversations are recorded with the full knowledge and
expectation of all parties involved.
4. User Agent Server Behavior: Receiving a Join Header
The Join header contains information used to match an existing SIP
dialog (call-id, to-tag, and from-tag). Upon receiving an INVITE
with a Join header, the UA attempts to match this information with a
confirmed or early dialog. The to-tag and from-tag parameters are
matched as if they were tags present in an incoming request. In
other words the to-tag parameter is compared to the local tag, and
the from-tag parameter is compared to the remote tag.
If more than one Join header field is present in an INVITE, or if a
Join header field is present in a request other than INVITE, the UAS
MUST reject the request with a 400 Bad Request response.
The Join header has specific call control semantics. If both a Join
header field and another header field with contradictory semantics
(for example a Replaces [8] header field) are present in a request,
the request MUST be rejected with a 400 "Bad Request" response.
If the Join header field matches more than one dialog, the UA MUST
act as if no match is found.
If no match is found, but the Request-URI in the INVITE corresponds
to a conference URI, the UAS MUST ignore the Join header and continue
processing the INVITE as if the Join header did not exist. This
allows User Agents which receive an INVITE with Join to redirect the
request directly to a conference URI.
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Otherwise if no match is found, the UAS rejects the INVITE and
returns a 481 Call/Transaction Does Not Exist response. Likewise, if
the Join header field matches a dialog which was not created with an
INVITE, the UAS MUST reject the request with a 481 response.
If the Join header field matches a dialog which has already
terminated, the UA SHOULD decline the request with a 603 Declined
response.
If the Join header field matches an active dialog (n.b. unlike the
Replaces header, the Join header has no limitation on its use with
early dialogs), the UA MUST verify that the initiator of the new
INVITE is authorized to join the matched dialog. If the initiator of
the new INVITE has authenticated successfully as equivalent to the
user who is being joined, then the join is authorized. For example,
if the user being joined and the initiator of the joining dialog
share the same credentials for Digest authentication [4], or they
sign the join request with S/MIME [5] with the same private key and
present the (same) corresponding certificate used in the original
dialog, then the join is authorized.
Alternatively, the Referred-By mechanism [9] defines a mechanism that
the UAS can use to verify that a join request was sent on behalf of
the other participant in the matched dialog (in this case, triggered
by a REFER request). If the join request contains a Referred-By
header which corresponds to the user being joined, the UA SHOULD
treat the join as if it was authorized by the joined party. The
Referred-By header MUST reference a corresponding, valid Refererred-
By Authenticated Identity Body [10]. The UA MAY apply other local
policy to authorize the remainder of the request. In other words,
the UAS may apply different policy to the joined dialog than was
applied to the target dialog.
The UA MAY also maintain a list of authorized entities who are
allowed to join any dialog with certain characteristics (for example,
all dialogs placed in the call center context of the UA). In
addition, the UA MAY use other authorization mechanisms defined for
this purpose in standards track extensions. For example, an
extension could define a mechanism for transitively asserting
authorization of a join.
If authorization is successful, the UA attempts to accept the new
INVITE, and assign any mixing or conferencing resources necessary to
complete the join. If the UA cannot accept the new INVITE (for
example: it cannot establish required QoS or keying, or it has
incompatible media), the UA MUST return an appropriate error response
and MUST leave the matched dialog unchanged.
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A User Agent that accepts a Join header needs to setup dialogs or
conferences such that the requesting UAC is logically added to the
conversation space associated with the matched dialog. Any dialogs
which are already logically associated with the matched dialog in the
same conversation space are included as well. For a detailed
description of various conferencing mechanisms that could be used to
handle a Join, please consult the SIP conferencing framework [15].
If the UAS has sufficient resources to locally handle the Join
request, the UAS SHOULD accept the Join request and perform the
appropriate media mixing or combining. The UAS MAY rearrange
appropriate dialogs instead as described below, based on some local
policy.
If the UAS does not have sufficient resources locally to handle the
request, or does not wish to use these local resources, but is aware
of other resources which could be used to satisfy the request (e.g.,
a centralized conference server), the UA SHOULD create a conference
using this resource (e.g., INVITE the conference server to obtain a
conference URI), redirect the requestor to this resource, and request
other participants in the same conversation space to use this
resource. The UA MAY use any appropriate mechanism to transition
participants to the new resource (e.g., 3xx response, 3rd-party call
control reinvitiations, REFER requests, or reinvitations to a
multicast group). The UA SHOULD only use mechanisms which are
expected to be acceptable to the other participants. For example,
the UA SHOULD NOT attempt to transition the participants to a
multicast group unless the UA can reasonably expect that all the
participants can support multicast.
If the UAS is incapable of satisfying the Join request, it MUST
return a 488 "Not Acceptable Here" response.
5. User Agent Client Behavior: Sending a Join header
A User Agent that wishes to add a new dialog of its own to a single
existing early or confirmed dialog and any associated dialogs or
conferences, MAY send the target User Agent an INVITE request
containing a Join header field. The UAC places the Call-ID, to-tag,
and from-tag information for the target dialog in a single Join
header field and sends the new INVITE to the target.
If the User Agent receives a 300-class response, and acts on this
response by sending an INVITE to a Contact in the response, this
redirected INVITE MUST contain the same Join header which was present
in the original request. Although this is unusual, this allows
INVITE requests with a Join header to be redirected before reaching
the target UAS.
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Note that use of the Join mechanism does not provide a way to match
multiple dialogs, nor does it provide a way to match an entire call,
an entire transaction, or to follow a chain of proxy forking logic.
6. Proxy behavior
Proxy Servers do not require any new behavior to support this
extension. They simply pass the Join header field transparently as
described in the SIP specification.
Note that it is possible for a proxy (especially when forking based
on some application layer logic, such as caller screening or time-
of-day routing) to forward an INVITE request containing a Join header
field to a completely orthogonal set of Contacts than the original
request it was intended to replace. In this case, the INVITE request
with the Join header field will fail.
7. Syntax
7.1. The Join Header
The Join header field indicates that a new dialog (created by the
INVITE in which the Join header field in contained) should be joined
with a dialog identified by the header field, and any associated
dialogs or conferences. It is a request header only, and defined
only for INVITE requests. The Join header field MAY be encrypted as
part of end-to-end encryption. Only a single Join header field value
may be present in a SIP request
This document adds the following entry to Table 3 of [1]. Additions
to this table are also provided for extension methods defined at the
time of publication of this document. This is provided as a courtesy
to the reader and is not normative in any way. MESSAGE, SUBSCRIBE
and NOTIFY, REFER, INFO, UPDATE, PRACK, and PUBLISH are defined
respectively in [19], [20], [7], [21], [22], [23], and [24].
Header field where proxy ACK BYE CAN INV OPT REG MSG
------------ ----- ----- --- --- --- --- --- --- ---
Join R - - - o - - -
SUB NOT REF INF UPD PRA PUB
--- --- --- --- --- --- ---
Join R - - - - - - -
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The following syntax specification uses the augmented Backus-Naur
Form (BNF) as described in RFC 2234 [3].
Join = "Join" HCOLON callid *(SEMI join-param)
join-param = to-tag / from-tag / generic-param
to-tag = "to-tag" EQUAL token
from-tag = "from-tag" EQUAL token
A Join header MUST contain exactly one to-tag and exactly one from-
tag, as they are required for unique dialog matching. For
compatibility with dialogs initiated by RFC 2543 [11] compliant UAs,
a to-tag of zero matches both a to-tag value of zero and a null to-
tag. Likewise, a from-tag of zero matches both a to-tag value of
zero and a null from-tag.
Examples:
Join: [email protected]
;from-tag=r33th4x0r
;to-tag=ff87ff
Join: 12adf2f34456gs5;to-tag=12345;from-tag=54321
Join: [email protected];to-tag=24796;from-tag=0
7.2. New option tag for Require and Supported headers
This specification defines a new Require/Supported header option tag
"join". UAs which support the Join header MUST include the "join"
option tag in a Supported header field. UAs that want explicit
failure notification if Join is not supported MAY include the "join"
option in a Require header field.
Example:
Require: join, 100rel
8. Usage Examples
The following non-normative examples are not intended to enumerate
all the possibilities for the usage of this extension, but rather to
provide examples or ideas only. For more examples, please see
service-examples [18].
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8.1. Join accepted and transitioned to central conference
A B C conf
| | callid: [email protected] | |
| | | |
| |<-INVITE------| | *1
| |-----200----->| | *2
| |<----ACK------| | *3
| |<============>| |
| | | |
|INVITE------>| | | *4
|Join: [email protected] |--INVITE-------------------->| *5
| |<----200---------------------| *6
| |-----ACK-------------------->|
|<----302-----| | | *7
|-----ACK---->| | |
|INVITE------------------------------------>| *8
|<--200-------------------------------------| *9
|---ACK------------------------------------>|
| |--REFER------>| | *10
| |<---202-------| |
| |<--NOTIFY-----|--INVITE-*11->|
| |------200---->|<----200-*12--|
| |<--NOTIFY-----|-----ACK----->|
| |------200---->| |
| |---BYE------->| |
| |<--200--------| |
| | | |
|<=========================================>| mixes the
| |<===========================>| three sessions
| | |<============>| together
The conversation now appears identical to the locally mixed one from
the example in the Introduction. Details of how the Join are
implemented are transparent to A. B could have used 3rd party call
control instead to move the necessary sessions.
Message *1: C -> B
INVITE sip:[email protected] SIP/2.0
To: <[email protected]>
From: <[email protected]>;tag=xyz
Call-Id: [email protected]
CSeq 1 INVITE
Contact: <sip:[email protected]>
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Message *2: B -> C
SIP/2.0 200 OK
To: <[email protected]>;tag=pdq
From: <[email protected]>;tag=xyz
Call-Id: [email protected]
CSeq 1 INVITE
Contact: <sip:[email protected]>
Message *3: C -> B
ACK sip:[email protected] SIP/2.0
To: <[email protected]>;tag=pdq
From: <[email protected]>;tag=xyz
Call-Id: [email protected]
CSeq 1 INVITE
Message *4: A -> B
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Join: [email protected];to-tag=xyz;from-tag=pdq
Message *5: B -> conf
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=abc
Call-Id: [email protected]
CSeq: 1INVITE
Contact: <sip:[email protected]>
Message *6: conf -> B
SIP/2.0 200 OK
To: <sip:[email protected]>;tag=def
From: <sip:[email protected]>;tag=abc
Call-Id: [email protected]
CSeq: 1INVITE
Contact: <sip:[email protected]>;isfocus
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Message *7: B -> A
SIP/2.0 302 Moved Temporarily
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>;isfocus
Message *8: A -> conf
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>
Join: [email protected];to-tag=xyz;from-tag=pdq
Message *9: conf ->A
SIP/2.0 200 OK
To: <sip:[email protected]>;tag=jjj
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]>;isfocus
Message *10: B -> C
REFER sip:[email protected] SIP/2.0
To: <[email protected]>;tag=xyz
From: <[email protected]>;tag=pdq
Call-Id: [email protected]
CSeq: 1 REFER
Contact: <sip:[email protected]>
Refer-To: <sip:[email protected]>
Referred-By: <sip:[email protected]>
Message *11: C -> conf
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: <[email protected]>;tag=mmm
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Call-Id: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Referred-By: <sip:[email protected]>
Message *12: C -> conf
SIP/2.0 200 OK
To: <sip:[email protected]>
From: <[email protected]>;tag=mmm
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>;isfocus
Referred-By: <sip:[email protected]>
8.2. Join rejected
A B C
| | callid: [email protected] |
| | |
| |<============>|
| | |
|INVITE------>| *1 |
|Join: [email protected] | |
| | |
|<----486-----| *2 |
|-----ACK---->| |
| | |
In this example B is Busy (does not want to be disturbed), and
therefore does not wish to add A. B could also decline the request
with a 603 response.
Message *1: A -> B
INVITE sip:[email protected] SIP/2.0
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]>
Join: [email protected];to-tag=xyz;from-tag=pdq
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Message *2: B -> A
SIP/2.0 486 Busy
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=iii
Call-Id: [email protected]
CSeq: 1 INVITE
9. Security Considerations
The extension specified in this document significantly changes the
relative security of SIP devices. Currently in SIP, even if an
eavesdropper learns the Call-ID, To, and From headers of a dialog,
they cannot easily modify or destroy that dialog if Digest
authentication or end-to-end message integrity are used.
This extension can be used to insert or monitor potentially sensitive
content in a multimedia conversation. As such, invitations with the
Join header MUST only be accepted if the peer requesting replacement
has been properly authenticated using a standard SIP mechanism
(Digest or S/MIME), and authorized to be joined with the target
dialog. (All SIP implementations are already required to support
Digest Authentication.) Generally authorization for joins are
configured as a matter of local policy as long-duration persistent
relationships.
For example, the UAs used by call center agents might be configured
with a list of identities who could join their calls (supervisors and
any call center monitoring User Agents). Alternatively the call
center agents might rely on transitive authorization assertions from
a (shorter) list of authorized hosts (e.g., a certificate authority).
For answering-machine-style message screening this is even easier.
Presumably the user screening their messages already has some
credentials with their messaging server.
Some mechanisms for obtaining the dialog information needed by the
Join header (Call-ID, to-tag, and from-tag) include URIs on a web
page, subscriptions to an appropriate event package, and
notifications after a REFER request. Use of end-to-end security
mechanisms to integrity protect and encrypt this information is also
RECOMMENDED.
This extension was designed to take advantage of future signature or
authorization schemes defined by standards track extensions. In
general, call control features would benefit considerably from such
work.
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RFC 3911 SIP Join October 2004
Section 4 describes specific mechanisms for authorization using
Digest Authentication and S/MIME (RFC 3261) and Referred-by [9], the
currently available capabilities in SIP.
10. IANA Considerations
10.1. Registration of "Join" SIP header
Name of Header: Join
Short form: none
Normative description: section 7.1 of this document
10.2. Registration of "join" SIP Option-tag
Name of option: join
Description: Support for the SIP Join header
SIP headers defined: Join
Normative description: This document
11. Acknowledgments
Thanks to Robert Sparks, Alan Johnston, and Ben Campbell and many
other members of the SIP WG for their continued support of the cause
of distributed call control in SIP.
12. References
12.1. Normative References
[1] 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.
[2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[3] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
[4] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP Authentication:
Basic and Digest Access Authentication", RFC 2617, June 1999.
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RFC 3911 SIP Join October 2004
[5] Ramsdell, B., "Secure/Multipurpose Internet Mail Extensions
(S/MIME) Version 3.1 Message Specification", RFC 3851, July
2004.
[6] Rosenberg, J., "Indicating User Agent Capabilities in the
Session Initiation Protocol (SIP)", RFC 3840, August 2004.
12.2. Informative References
[7] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[8] Dean, R., Biggs, B., and R. Mahy, "The Session Initiation
Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.
[9] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By
Mechanism", RFC 3892, September 2004.
[10] Peterson, J., "Session Initiation Protocol (SIP) Authenticated
Identity Body (AIB) Format", RFC 3893, September 2004.
[11] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
[12] Mahy, R., "A Call Control and Multi-party usage framework for
the Session Initiation Protocol (SIP)", Work in Progress,
March 2003.
[13] Rosenberg, J. and H. Schulzrinne, "An INVITE Initiated Dialog
Event Package for the Session Initiation Protocol (SIP)", Work
in Progress, March 2003.
[14] IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, May 2000.
[15] Rosenberg, J., "A Framework for Conferencing with the Session
Initiation Protocol", Work in Progress, May 2003.
[16] Johnston, A. and O. Levin, "Session Initiation Protocol Call
Control - Conferencing for User Agents", Work in Progress,
April 2003.
[17] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo,
"Best Current Practices for Third Party Call Control (3pcc) in
the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April
2004.
[18] Johnston, A. and S. Donovan, "Session Initiation Protocol
Service Examples", Work in Progress, March 2003.
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[19] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and
D. Gurle, "Session Initiation Protocol (SIP) Extension for
Instant Messaging", RFC 3428, December 2002.
[20] Roach, A., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[21] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.
[22] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[23] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
Responses in Session Initiation Protocol (SIP)", RFC 3262, June
2002.
[24] Campbell, B., "SIMPLE Presence Publication Mechanism", Work in
Progress, February 2003.
13. Authors' Addresses
Rohan Mahy
Airespace
110 Nortech Parkway
San Jose, CA 95134
USA
EMail: [email protected]
Dan Petrie
Pingtel
400 West Cummings Park, Suite 2200
Woburn, MA 01801
USA
EMail: [email protected]
Mahy & Petrie Standards Track [Page 16]
RFC 3911 SIP Join October 2004
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