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rfc:rfc3911

Network Working Group R. Mahy Request for Comments: 3911 Airespace Category: Standards Track D. Petrie

                                                               Pingtel
                                                          October 2004
        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

Mahy & Petrie Standards Track [Page 1] RFC 3911 SIP Join October 2004

      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].

Mahy & Petrie Standards Track [Page 2] RFC 3911 SIP Join October 2004

 Assistant     Boss        Customer
 | callid: 4@A |  callid: 7@c |
 |             |              |
 |             |<============>|
 |             |              |
 |INVITE------>|              |
 |Join: 7@c    |              |
 |             |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

Mahy & Petrie Standards Track [Page 3] RFC 3911 SIP Join October 2004

    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.

Mahy & Petrie Standards Track [Page 4] RFC 3911 SIP Join October 2004

 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.

Mahy & Petrie Standards Track [Page 5] RFC 3911 SIP Join October 2004

 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.

Mahy & Petrie Standards Track [Page 6] RFC 3911 SIP Join October 2004

 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              -    -    -    -    -    -    -

Mahy & Petrie Standards Track [Page 7] RFC 3911 SIP Join October 2004

 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: 98732@sip.example.com
           ;from-tag=r33th4x0r
           ;to-tag=ff87ff
    Join: 12adf2f34456gs5;to-tag=12345;from-tag=54321
    Join: 87134@192.0.2.23;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].

Mahy & Petrie Standards Track [Page 8] RFC 3911 SIP Join October 2004

8.1. Join accepted and transitioned to central conference

 A             B              C            conf
 |             |  callid: 7@c |              |
 |             |              |              |
 |             |<-INVITE------|              | *1
 |             |-----200----->|              | *2
 |             |<----ACK------|              | *3
 |             |<============>|              |
 |             |              |              |
 |INVITE------>|              |              | *4
 |Join: 7@c    |--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:bob@example.org SIP/2.0
 To: <bob@example.org>
 From: <carol@example.org>;tag=xyz
 Call-Id: 7@c.example.org
 CSeq 1 INVITE
 Contact: <sip:carol@c.example.org>

Mahy & Petrie Standards Track [Page 9] RFC 3911 SIP Join October 2004

 Message *2: B -> C
 SIP/2.0 200 OK
 To: <bob@example.org>;tag=pdq
 From: <carol@example.org>;tag=xyz
 Call-Id: 7@c.example.org
 CSeq 1 INVITE
 Contact: <sip:bob@b.example.org>
 Message *3: C -> B
 ACK sip:carol@c.example.org SIP/2.0
 To: <bob@example.org>;tag=pdq
 From: <carol@example.org>;tag=xyz
 Call-Id: 7@c.example.org
 CSeq 1 INVITE
 Message *4: A ->  B
 INVITE sip:bob@b.example.org SIP/2.0
 To: <sip:bob@example.org>
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 CSeq: 1 INVITE
 Contact: <sip:alice@a.example.org>
 Join: 7@c.example.org;to-tag=xyz;from-tag=pdq
 Message *5: B -> conf
 INVITE sip:conf-factory@example.org SIP/2.0
 To: <sip:conf-factory@example.org>
 From: <sip:bob@example.org>;tag=abc
 Call-Id: 999@b.example.org
 CSeq: 1INVITE
 Contact: <sip:bob@b.example.org>
 Message *6: conf -> B
 SIP/2.0 200 OK
 To: <sip:conf-factory@example.org>;tag=def
 From: <sip:bob@example.org>;tag=abc
 Call-Id: 999@b.example.org
 CSeq: 1INVITE
 Contact: <sip:conf456@conf-srv2.example.org>;isfocus

Mahy & Petrie Standards Track [Page 10] RFC 3911 SIP Join October 2004

 Message *7: B -> A
 SIP/2.0 302 Moved Temporarily
 To: <sip:bob@example.org>
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 CSeq: 1 INVITE
 Contact: <sip:conf456@conf-srv2.example.org>;isfocus
 Message *8: A -> conf
 INVITE sip:conf456@conf-srv2.example.org SIP/2.0
 To: <sip:bob@example.org>
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 CSeq: 2 INVITE
 Contact: <sip:alice@a.example.org>
 Join: 7@c.example.org;to-tag=xyz;from-tag=pdq
 Message *9: conf ->A
 SIP/2.0 200 OK
 To: <sip:bob@example.org>;tag=jjj
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 CSeq: 2 INVITE
 Contact: <sip:conf456@conf-srv2.example.org>;isfocus
 Message *10: B -> C
 REFER sip:carol@c.example.org SIP/2.0
 To: <carol@example.org>;tag=xyz
 From: <bob@example.org>;tag=pdq
 Call-Id: 7@c.example.org
 CSeq: 1 REFER
 Contact: <sip:bob@b.example.org>
 Refer-To: <sip:conf456@conf-srv2.example.org>
 Referred-By: <sip:bob@b.example.org>
 Message *11: C -> conf
 INVITE sip:conf456@conf-srv2.example.org SIP/2.0
 To: <sip:conf456@conf-srv2.example.org>
 From: <carol@example.org>;tag=mmm

Mahy & Petrie Standards Track [Page 11] RFC 3911 SIP Join October 2004

 Call-Id: 34343@c.example.com
 CSeq: 1 INVITE
 Contact: <sip:carol@c.example.com>
 Referred-By: <sip:bob@b.example.org>
 Message *12: C -> conf
 SIP/2.0 200 OK
 To: <sip:conf456@conf-srv2.example.org>
 From: <carol@example.org>;tag=mmm
 Call-Id: 34343@c.example.com
 CSeq: 1 INVITE
 Contact: <sip:conf456@conf-srv2.example.org>;isfocus
 Referred-By: <sip:bob@b.example.org>

8.2. Join rejected

 A             B              C
 |             |  callid: 7@c |
 |             |              |
 |             |<============>|
 |             |              |
 |INVITE------>|  *1          |
 |Join: 7@c    |              |
 |             |              |
 |<----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:bob@b.example.org SIP/2.0
 To: <sip:bob@example.org>
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 CSeq: 1 INVITE
 Contact: <sip:alice@a.example.org>
 Join: 7@c.example.org;to-tag=xyz;from-tag=pdq

Mahy & Petrie Standards Track [Page 12] RFC 3911 SIP Join October 2004

 Message *2: B -> A
 SIP/2.0 486 Busy
 To: <sip:bob@example.org>
 From: <sip:alice@example.org>;tag=iii
 Call-Id: 777@a.example.org
 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.

Mahy & Petrie Standards Track [Page 13] 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.

Mahy & Petrie Standards Track [Page 14] 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.

Mahy & Petrie Standards Track [Page 15] RFC 3911 SIP Join October 2004

 [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: rohan@airespace.com
 Dan Petrie
 Pingtel
 400 West Cummings Park, Suite 2200
 Woburn, MA  01801
 USA
 EMail: dpetrie@pingtel.com

Mahy & Petrie Standards Track [Page 16] RFC 3911 SIP Join October 2004

14. Full Copyright Statement

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 retain all their rights.
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Acknowledgement

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Mahy & Petrie Standards Track [Page 17]

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