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Versions: (draft-mahy-sip-join-and-fork) 00 01 02 03 RFC 3911

SIP WG                                                           R. Mahy
Internet-Draft                                       Cisco Systems, Inc.
Expires: April 1, 2003                                         D. Petrie
                                                                 Pingtel
                                                                Oct 2002


          The Session Inititation Protocol (SIP) "Join" Header
                       draft-ietf-sip-join-00.txt

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   The list of current Internet-Drafts can be accessed at http://
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   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on April 1, 2003.

Copyright Notice

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

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.






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Table of Contents

   1.   Conventions  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.   Overview . . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.   Applicability of RFC2804 ("Raven") . . . . . . . . . . . . .   4
   4.   User Agent Server Behavior: Receiving a Join Header  . . . .   5
   5.   User Agent Client Behavior: Sending a Join header  . . . . .   7
   6.   Proxy behavior . . . . . . . . . . . . . . . . . . . . . . .   7
   7.   Syntax . . . . . . . . . . . . . . . . . . . . . . . . . . .   7
   7.1  The Join Header  . . . . . . . . . . . . . . . . . . . . . .   8
   7.2  New option tag for Require and Supported headers . . . . . .   9
   8.   Usage Examples . . . . . . . . . . . . . . . . . . . . . . .   9
   8.1  Join accepted and transitioned to central mixer  . . . . . .  10
   8.2  Join rejected  . . . . . . . . . . . . . . . . . . . . . . .  11
   9.   Security Considerations  . . . . . . . . . . . . . . . . . .  11
   10.  IANA Considerations  . . . . . . . . . . . . . . . . . . . .  11
   10.1 Registration of "Join" SIP header  . . . . . . . . . . . . .  12
   10.2 Registration of "join" SIP Option-tag  . . . . . . . . . . .  12
   11.  Changes  . . . . . . . . . . . . . . . . . . . . . . . . . .  12
   11.1 Changes Since draft-mahy-join-and-fork-01  . . . . . . . . .  12
   11.2 Changes Since -00  . . . . . . . . . . . . . . . . . . . . .  12
   12.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . .  13
        Normative References . . . . . . . . . . . . . . . . . . . .  13
        Informational References . . . . . . . . . . . . . . . . . .  13
        Authors' Addresses . . . . . . . . . . . . . . . . . . . . .  14
        Full Copyright Statement . . . . . . . . . . . . . . . . . .  15

























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1. 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].

2. Overview

   This document describes a SIP [1] extension header field as part of
   the SIP multiparty applications architecture framework [7].  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 a two-party call or a conference).
   While this functionality is already available using 3rd party call
   control [12] 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 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 (ex: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 [8].




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   Assitant     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, and
   reINVITE the Customer with the conference URI as the new Contact.

3. Applicability of RFC2804 ("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 RFC2804 [9].  The definition
   from RFC2804 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

      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.



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   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 are matched as if
   they were present in an incoming request.  In other words the to-tag
   is compared to the local tag, and the from-tag 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 [5] 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 to a conference.

   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 an appropriate response
   (ex: 400, 481, or 501).

   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 a active dialog, the UA SHOULD
   verify that the initiator of the new INVITE is authorized to join the



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   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.  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.

   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.  All the participants
   in a conversation space should have access to all the media/content
   sent in the context of that conversation space.  That a participant
   does not negotiate a specific type of media does not mean that it is
   not otherwise a full participant.  For a detailed description of
   various conferencing mechanisms that could be used to handle a Join,
   please consult the SIP conferencing framework [10].

   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 (ex: a
   centralized mixer), the UA SHOULD create a conference using this
   resource (ex: INVITE the centralized mixer 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 (ex: 3xx repsonse, 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



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   the UA can reasonably expect that all the particpants 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.

   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.
   For example, if Alice replaces Cathy in an early dialog with Bob, but
   he does not answer, Alice's replacement request will not match other
   dialogs to which Bob's UA redirects, nor other branches to which his
   proxy forwards.

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







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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.  SUBSCRIBE and NOTIFY,
   REFER, INFO, UPDATE, and PRACK are defined respectively in [14], [4],
   [15], [16], and [17].

        Header field    where   proxy   ACK  BYE  CAN  INV  OPT  REG
        ------------    -----   -----   ---  ---  ---  ---  ---  ---
        Join              R              -    -    -    o    -    -


                                        SUB  NOT  REF  INF  UPD  PRA
                                        ---  ---  ---  ---  ---  ---
        Join              R              -    -    -    -    -    -

   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 RFC2543 [6] compliant UAs, a
   tag of zero matches both tags of zero and null tags.












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   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 [13].




















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8.1 Join accepted and transitioned to central mixer

   A             B              C            mixer
   | callid: 4@A |  callid: 7@c |              |
   |             |              |              |
   |             |<============>|              |
   |             |              |              |
   |INVITE------>|              |              |
   |Join: 7@c    |--INVITE-------------------->|
   |             |<----200---------------------|
   |             |-----ACK-------------------->|
   |<----300-----|                             |
   |INVITE------------------------------------>|
   |<--200-------------------------------------|
   |---ACK------------------------------------>|
   |             |--REFER------>|              |
   |             |<---200-------|--INVITE----->|
   |             |              |<----200------|
   |             |<--NOTIFY-----|-----ACK----->|
   |             |------200---->|              |
   |             |---BYE------->|              |
   |             |<--200--------|              |
   |             |              |              |
   |<=========================================>| mixes the
   |             |<===========================>| three sessions
   |             |              |<============>| together

   The conversation space now looks identical to the locally mixed
   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.

   [ B , C ]  -->  [ A , B , C ]


















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8.2 Join rejected

   A             B              C
   | callid: 4@A |  callid: 7@c |
   |             |              |
   |             |<============>|
   |             |              |
   |INVITE------>|              |
   |Join: 7@c    |              |
   |             |              |
   |<----486-----|              |
   |-----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.

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 SHOULD only be accepted if the peer requesting
   replacement has been properly authenticated using a standard SIP
   mechanism, and authorized to by joined with the target dialog.

   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 notifcations
   after a REFER request.  Use of end-to-end security mechanisms to
   encrypt this information is also RECOMMENDED.

   This extension was designed to take advantage of future signature or
   authorization schemes defined by the SIP Working Group.  In general,
   call control features would benefit considerably from such work.

10. IANA Considerations








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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. Changes

11.1 Changes Since draft-mahy-join-and-fork-01

   o  Added discussion about handling of 300-class responses to an
      INVITE with Join

   o  Fixed several typos

   o  Updated references

   o  Resubmitted as a Working Group item


11.2 Changes Since -00

   o  Realigned the text to mirror the outline of Replaces

   o  Removed the fork header

   o  Added a section to explain how this is not a "Raven" wiretap
      mechanism

   o  Reorganized motivational overview material

   o  Added authorization language in UAS behavior section

   o  Updated and Added references



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12. 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.

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.

Informational References

   [4]   Sparks, R., "The SIP Refer Method", draft-ietf-sip-refer-06
         (work in progress), July 2002.

   [5]   Dean, R., Biggs, B. and R. Mahy, "The Session Inititation
         Protocol (SIP) 'Replaces' Header", draft-ietf-sip-replaces-02
         (work in progress), May 2002.

   [6]   Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
         "SIP: Session Initiation Protocol", RFC 2543, March 1999.

   [7]   Mahy, R., "A Multi-party Application Framework for SIP", draft-
         ietf-sipping-cc-framework-01 (work in progress), July 2002.

   [8]   Rosenberg, J. and H. Schulzrinne, "A Session Initiation
         Protocol (SIP) Event Package for Dialog State", draft-ietf-
         sipping-dialog-package-00 (work in progress), June 2002.

   [9]   IAB and IESG, "IETF Policy on Wiretapping", RFC 2804, May 2000.

   [10]  Rosenberg, J., "SIP Conferencing Framework", draft-rosenberg-
         sipping-conferencing-framework-00.txt (work in progress), Oct
         2002.

   [11]  Sparks, R. and A. Johnston, "SIP Call Control - Transfer",
         draft-ietf-sipping-cc-transfer-00.txt (work in progress), Oct
         2002.

   [12]  Rosenberg, J., Schulzrinne, H., Camarillo, G. and J. Peterson,



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         "Best Current Practices for Third Party Call Control in the
         Session  Initiation Protocol", draft-ietf-sipping-3pcc-02 (work
         in progress), June 2002.

   [13]  Johnston, A., "SIP Service Examples", draft-ietf-sipping-
         service-examples-02 (work in progress), July 2002.

   [14]  Roach, A., "Session Initiation Protocol (SIP)-Specific Event
         Notification", RFC 3265, June 2002.

   [15]  Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.

   [16]  Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
         Method", RFC 3311, October 2002.

   [17]  jdrosen@dynamicsoft.com and schulzrinne@cs.columbia.edu,
         "Reliability of Provisional Responses in Session Initiation
         Protocol (SIP)", RFC 3262, June 2002.


Authors' Addresses

   Rohan Mahy
   Cisco Systems, Inc.
   101 Cooper Street
   Santa Cruz, CA  95060
   USA

   EMail: rohan@cisco.com


   Dan Petrie
   Pingtel
   400 West Cummings Park, Suite 2200
   Woburn, MA  01801
   USA

   EMail: dpetrie@pingtel.com













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Full Copyright Statement

   Copyright (C) The Internet Society (2002).  All Rights Reserved.

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   The limited permissions granted above are perpetual and will not be
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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.



















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