SIP Working Group                                              B. Biggs
Internet Draft                                                  R. Dean
Document: draft-ietf-sip-replaces-00.txt                        R. Mahy
                                                               (editor)
Expires: July 2002                                         January 2002


                        The SIP Replaces Header


Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
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   http://www.ietf.org/ietf/1id-abstracts.txt
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1. Abstract

   This document proposes a new header for use with the SIP call
   control architecture.  The Replaces header is used in peer-to-peer
   call control to logically replace an existing SIP dialog with a new
   SIP dialog.  This primitive can be used to enable a variety of
   features, for example: "Attended Transfer" and "Retrieve from Call
   Park".  Note that definition of these example features is non-
   normative.

2. Conventions used in this document

   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
   [RFC2119].

   Throughout this document, an "established dialog" means an active
   SIP dialog in which the request that created the dialog has received
   a successful final response (200 OK).  An "early dialog" means
   transaction and dialog state that exists after a request is sent
   which would create a new dialog, but before a final response is
   received for the initial request.



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                         SIP Replaces Header


3. Overview

   This document describes a [SIP] extension for distributed call
   control as part of the SIP call control architecture [framework].
   The Replaces header is used in peer-to-peer call control to
   logically replace an existing SIP dialog with a new SIP dialog.

   INVITEs are requests which can be accepted, rejected or declined.  A
   User Agent that accepts a request with call-control semantics agrees
   to take responsibility for setting up the appropriate requested
   media relationships.

   In the parlance of the SIP Call Control Model document [cc-models],
   the "Replaces" header is used to replace one participant with
   another in a conversation space. This functionality is already
   available using 3rd party call control [3pcc] 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 a new INVITE with a new header for dialog matching was chosen
   over making implicit associations in an incoming INVITE based on
   call-id or other fields for the following reasons:

    - An INVITE already has the correct semantics for a new call

    - Using an explicit Replaces header in a new request makes the
      intent of the request obvious.

    - A unique call-id may be given to the replacement call.  This
      avoids call-leg matching problems in any of the clients.

    - There are no adverse effects if the header is unsupported.

   The Replaces header enables services such as attended call transfer,
   retrieve from park, and transition from locally mixed conferences to
   two party calls in a distributed peer-to-peer way.  This list of
   services is not exhaustive.  Although the Replaces header is
   frequently used in combination with the [REFER] method as used in
   [cc-transfer], they may be used independently.

4. Syntax

4.1. The Replaces Header

   The Replaces header indicates that the dialog identified by the
   header is to be shut down and logically replaced by the incoming
   INVITE in which it is contained.  It is a request header only, and
   defined here only for INVITE requests.  The Replaces header MAY be
   encrypted as part of end-to-end encryption.

   This document adds the following entry to Table 3 of [bis]:

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                         SIP Replaces Header



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


   Note that the Replaces header has specific call control semantics.
   If both a Replaces header and another header with contradictory
   semantics are present in a request, the request MUST be rejected
   with a 400 "Bad Request" response.

4.1.1. Formal Syntax

   The following syntax specification uses the augmented Backus-Naur
   Form (BNF) as described in RFC-2234 [BNF].


      Replaces         = "Replaces" ":" 1#replaces-values
      replaces-values  =  callid *( ";" replaces-param )
      callid           = token [ "@" token ]
      replaces-param   = to-tag | from-tag | extension-param
      to-tag           = "to-tag=" UUID
      from-tag         = "from-tag=" UUID
      extension-param   = token [ "=" ( token | quoted-string ) ]

   A Replaces header MUST contain exactly one to-tag and exactly one
   from-tag, as they are required for unique dialog matching.  Since we
   rely on the tags for matching purposes, implementations which
   support Replaces MUST support the SIP [bis] specification, which
   requires tags.  For compatibility with early dialogs and dialogs
   initiated by RFC2543 compliant UAs, a tag of zero must match both
   tags of zero and null tags.

4.1.2. Examples

      Replaces: 98732@sip.billybiggs.com
                ;from-tag=r33th4x0r
                ;to-tag=ff87ff

      Replaces: 12345@149.112.118.3;to-tag=12345;from-tag=54321

      Replaces: 87134@171.161.34.23;to-tag=24796;from-tag=0


4.2 New option tag for Require and Supported headers

   This specification defines a new Require/Supported header option tag
   "replaces".  UAs which support the Replaces header MUST include the
   "replaces" option in the Supported header.  UAs that want explicit
   failure notification if Replaces is not supported MAY send include
   the "replaces" option in the Require header.

   Example:

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                         SIP Replaces Header



      Require: replaces, 100rel

4.3 687 Response Code: "Dialog Terminated"

   This specification defines a new SIP response code. The 687 "Dialog
   Terminated" response code indicates that an early dialog has been
   completely replaced by a new dialog.  A new response code was chosen
   from the 6xx class to prevent intervening proxies from attempting to
   fork additional branches of the replaced dialog.

5. User Agent Behavior: Receiving a Replaces Header

5.1 Matching Dialogs

   The Replaces header contains information used to match an existing
   SIP dialog (call-id, to-tag, and from-tag).  Upon receiving an
   INVITE with the Replaces header, the UA MUST attempt to match this
   information with an established 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 the Replaces header matches more than one dialog, the UA MAY use
   other headers if present (ex: the Referred-By header) to attempt to
   match a single dialog.  If a single matching dialog is not found,
   the UA MUST act as if no match is found.

   If no match is found, the UAS MUST ignore the header and process the
   INVITE normally.

        OPEN ISSUE: If no match is found, should the UAS ignore the
        header and process normally, or return a 481?

   If the Replaces header matches a dialog which was not created with
   an INVITE, the UAS MUST reject the request with an appropriate
   response.

   If the Replaces header matches a dialog which has already
   terminated, the UA SHOULD decline the request with a 603 Declined
   response.  This prevents phantom ringing in cases like example 6.4.

   Once a matching call-leg is found, the UAS MAY authenticate the
   INVITE request.  If the request is successfully authenticated or
   already preauthorized, the UAS SHOULD proceed with processing.  The
   UAS MAY prompt the user to accept or reject unauthenticated
   requests.  The UAS MAY reject the request with any appropriate
   response (for example: 603 "Decline", 403 "Forbidden", or 488 "Not
   Acceptable Here")

5.2. Replaces Semantics


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                         SIP Replaces Header


   If the Replaces header matches an established active dialog, the UA
   SHOULD attempt to accept the new INVITE, reassign the user interface
   and other resources of the matched dialog to the new INVITE, and
   shut down the replaced dialog by sending a BYE.  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 response and leave the matched dialog unchanged.

   If the Replaces header matches an early dialog that was initiated by
   the UA, the UA SHOULD attempt to accept the new INVITE.  If the UA
   cannot accept the new INVITE, the UA MUST return an appropriate
   response and leave the matched dialog unchanged.  If the UA
   successfully accepts the new INVITE, the UA MUST reassign the
   resources of the early dialog to the new INVITE, and CANCEL the
   replaced early dialog.

   If the Replaces header matches an early dialog that was not
   initiated by the UA, the UA SHOULD attempt to provisionally accept
   the new INVITE.  In other words, the UA should attempt whatever
   steps are necessary to return a provisional or final response
   suitable for the state of the resources used by the matched dialog.

   If this is successful, the UA MUST reassign the resources of the
   early dialog to the new INVITE, and respond to the replaced early
   dialog with a 687 "Transaction Terminated" response (defined later
   in this document).

6. Usage Examples

   The following non-normative examples are not intended to enumerate
   all the possibilities for the usage of these extensions, but rather
   to provide examples or ideas only.  For more examples, please see
   [service-examples].


6.1. Replacing an Active Dialog

   In this example, Alice is talking to Bob from phone1.  She transfers
   Bob to a Parking Place while she goes to the lab.  When she gets
   there she retrieves the "parked" call from phone2 by sending an
   INVITE with Replaces to Bob with the dialog information Bob shared
   with the Parking Place.  How did Alice get this information?  Maybe
   she subscribed to this information from the Parking Place, or went
   to a website and clicked on a URL.

        Alice          Alice                             Parking
        phone1         phone2            Bob               Place
        |               |                 |                   |
        |<===============================>|                   |
        |               |                 |                   |
        |        Alice transfers Bob to Parking Place         |
        |               |                 |                   |
        |------------REFER/200----------->|    *1    *2       |

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                         SIP Replaces Header


        |               |                 |--INVITE/200/ACK-->|
        |<-----------NOTIFY/200-----------|<=================>|
        |------------BYE/200------------->|                   |
        |               |                 |                   |
        |               |                 |                   |
        |  Alice later retrieves call from another phone      |
        |               |                 |                   |
        |            *3 |-INV w/Replaces->|                   |
        |            *4 |<--200-----------|    *5             |
        |               |---ACK---------->|----BYE/200------->|
        |               |<===============>|                   |
        |               |                 |                   |



   Message *1: Bob-> Parking Place

   INVITE sip:parkingplace@sip.org SIP/2.0
   To: <sip:parkingplace@sip.org>
   From: <sip:bob@sip.org>;tag=7743
   Call-ID: 425928@bobster.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@bobster.sip.org>
   Referred-By: <sip:alice@phone1.sip.org>

   Message *2: Parking Place -> Bob

   SIP/2.0 200 OK
   To: <sip:parkingplace@sip.org>;tag=6472
   From: <sip:bob@sip.org>;tag=7743
   Call-ID: 425928@bobster.sip.org
   CSeq: 1 INVITE
   Contact: <sip:parkplace@monopoly.sip.org>

   Message *3: Alice@phone2 -> Bob

   INVITE sip:bob@bobster.sip.org
   To: <sip:bob@sip.org>
   From: <sip:alice@phone2.sip.org>;tag=8983
   Call-ID: 09870@phone2.sip.org
   CSeq: 1 INVITE
   Contact: <sip:alice@phone2.sip.org>
   Replaces: 425928@bobster.sip.org;to-tag=7743;from-tag=6472

   Message *4: Bob -> Alice@phone2

   SIP/2.0 200 OK
   To: <sip:bob@sip.org>;tag=9343
   From: <sip:alice@phone2.sip.org>;tag=8983
   Call-ID: 09870@phone2.sip.org
   CSeq: 1 INVITE
   Contact: <sip:alice@phone2.sip.org>


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                         SIP Replaces Header


   Message *5: Bob -> Parking Place

   BYE sip:parkingplace@sip.org SIP/2.0
   To: <sip:parkingplace@sip.org>;tag=6472
   From: <sip:bob@sip.org>;tag=7743
   Call-ID: 425928@bobster.sip.org
   CSeq: 2 BYE
   Contact: <sip:bob@bobster.sip.org>

6.2 Replacing an Early Dialog initiated by someone else

   In this example, a Customer tries calling a call center and for some
   reason cannot get through properly.  The customer calls an Operator
   and asks for help.  The operator calls the contact center, and upon
   receiving a provisional response, assumes that everything is OK and
   transfers the Customer to the Call Center, replacing the operator's
   place in the queue.

                                                       Call
      Operator               Customer                  Center
        |                       |                        |
        |<--INVITE/180/200/ACK--|                        |
        |<=====================>|  "Hello, I'm having    |
        |                       |   trouble calling ..." |
        |"OK, I'll try it and   |                        |
        | transfer you if it    |                        |
        | works for me"         |                        |
        |                       |                        |
     *1 |-----INVITE ----------------------------------->|
     *2 |<----182: You are caller number 7---------------|
        |                       |                        |
        |  completes transfer   |                        |
        |                       |                        |
        |---REFER/200---------->|                        |
        |                       |--INVITE with Replaces->| *3
        |                       |<----182: caller #7-----| *4
        |<----687 Dialog Terminated----------------------| *5
        |-----ACK--------------------------------------->|
        |<--NOTIFY/200----------|                        |
        |---BYE/200------------>|                        |
        |                       |  ...time passes..      |
        |                       |                        |
        |                       |                        |
        |                       |                        |
        |                       |<---200 OK--------------|
        |<--NOTIFY/200----------|----ACK---------------->|
        |                       |                        |
        |                       |                        |

   Message *1: Operator -> Call Center

   INVITE sip:helpdesk@clueless.org SIP/2.0
   To: <sip:helpdesk@clueless.org>

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                         SIP Replaces Header


   From: <sip:operator@acme.com>;tag=7743
   Call-ID: 425928@dhcp23311.acme.com
   CSeq: 1 INVITE
   Contact: <sip:jdoe@dhcp2311.acme.com>
   Accept-Language: en

   Message *2: Call Center -> Operator

   SIP/2.0 182 You are 7th in Queue
   To: <sip:helpdesk@clueless.org>;tag=6472
   From: <sip:operator@acme.com>;tag=7743
   Call-ID: 425928@dhcp23311.acme.com
   CSeq: 1 INVITE
   Contact: <sip:helpdesk@frontline.clueless.org>


   Message *3: Customer -> Call Center

   INVITE sip:helpdesk@frontline.clueless.org
   To: <sip:helpdesk@clueless.org>
   From: <sip:customer@acme.com>;tag=8983
   Call-ID: 09870@lobby12.acme.com
   CSeq: 1 INVITE
   Contact: <sip:customer@lobby12.acme.com>
   Replaces: 425928@dhcp23311.acme.com;to-tag=7743;from-tag=6472
   Accept-Language: en
   Referred-By: <sip:jdoe@dhcp2311.acme.com>

   Message *4: Call Center -> Customer

   SIP/2.0 182 You are 7th in Queue
   To: <sip:helpdesk@clueless.org>
   From: <sip:customer@acme.com>;tag=8983
   Call-ID: 09870@lobby12.acme.com
   CSeq: 1 INVITE
   Contact: <sip:helpdesk@frontline.clueless.org>


   Message *5: Call Center -> Operator

   SIP/2.0 687 Dialog Terminated
   To: <sip:helpdesk@clueless.org>;tag=6472
   From: <sip:operator@acme.com>;tag=7743
   Call-ID: 425928@dhcp23311.acme.com
   CSeq: 1 INVITE
   Contact: <sip:helpdesk@frontline.clueless.org>


6.3. Replacing an Early Dialog you initiated

   In this example, Bob just arrived in the lab and hasn't registered
   there yet.  He hears his desk phone ring.  He quickly logs into a
   software UA on a nearby computer.  Among other things, the software

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                         SIP Replaces Header


   UA subscribes to the call-state of his desk phone.  When it notices
   that his phone is ringing it offers him the choice to take the call
   there.  The software UA sends an INVITE with Replaces to Alice.
   When Alice's UA receives this new INVITE, it CANCELs her original
   INVITE and connects Alice to Bob.

                              Bob                      Bob
       Alice                  desk                     lab
        |                       |                        |
    *1  |-----INVITE----------->|                        |
    *2  |<----180---------------|  Bob hears desk phone  |
        |                       |  ringing from lab but  |
        |                       |  isn't REGISTERed yet  |
        |                       |                        |
        |                       |<--SUB callpackage/200--|
        |                       |---NOTIFY/200---------->|
   *3/4 |<-----INVITE with Replaces/200/ACK--------------|
   *5/6 |------CANCEL/200------>|                        |
   *7   |<-----487--------------|                        |
        |------ACK------------->|                        |
        |                       |                        |
        |                       |                        |

   Message *1: Alice -> Bob's desk phone

   INVITE sip:bob@sip.org SIP/2.0
   To: <sip:bob@sip.org>
   From: <sip:alice@sip.org>;tag=7743
   Call-ID: 425928@phone.sip.org
   CSeq: 1 INVITE
   Contact: <sip:alice@phone.sip.org>

   Message *2: Bob's desk phone -> Alice

   SIP/2.0 180 Ringing
   To: <sip:bob@sip.org>;tag=6472
   From: <sip:alice@sip.org>;tag=7743
   Call-ID: 425928@phone.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@bobster.sip.org>

   Message *3: Bob in lab -> Alice

   INVITE sip:alice@phone.sip.org
   To: <sip:alice@sip.org>
   From: <sip:bob@sip.org>;tag=8983
   Call-ID: 09870@labpc.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@labpc.sip.org>
   Replaces: 425928@phone.sip.org;to-tag=7743;from-tag=6472

   Message *4: Alice -> Bob in lab


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                         SIP Replaces Header


   SIP/2.0 200 OK
   To: <sip:alice@sip.org>;tag=9232
   From: <sip:bob@sip.org>;tag=8983
   Call-ID: 09870@labpc.sip.org
   CSeq: 1 INVITE
   Contact: <sip:alice@phone.sip.org>

   Message *5: Alice -> Bob's desk

   CANCEL sip:bob@sip.org SIP/2.0
   To: <sip:bob@sip.org>
   From: <sip:alice@sip.org>;tag=7743
   Call-ID: 425928@phone.sip.org
   CSeq: 1 CANCEL
   Contact: <sip:alice@phone.sip.org>

   Message *6: Bob's desk -> Alice

   SIP/2.0 200 OK
   To: <sip:bob@sip.org>
   From: <sip:alice@sip.org>;tag=7743
   Call-ID: 425928@phone.sip.org
   CSeq: 1 CANCEL
   Contact: <sip:bob@bobster.sip.org>

   Message *7: Bob's desk -> Alice

   SIP/2.0 487 Request Terminated
   To: <sip:bob@sip.org>;tag=6472
   From: <sip:alice@sip.org>;tag=7743
   Call-ID: 425928@phone.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@bobster.sip.org>


6.4. Handling Replaces for a Terminated Dialog

   In this example, Alice, Bob, and Cathy participate in a 3-way call
   mixed locally by Bob's UA.  Bob's UA is programmed to revert to a
   simple 2-party call when any party hangs up (including Bob).
   Ordinarily this would be a very polite feature--Cathy and Alice
   could continue to talk after Bob hangsup.  If all three hang up at
   about the same time, but Bob hangs up first (this will happen about
   one-third of the time), an INVITE with Replaces header can arrive at
   Cathy's UA shortly after she has hung up.  Because Cathy's UA needs
   to keep transaction state around for a while anyway (typically 32
   seconds), the dialog information in the Replaces header should match
   a terminated dialog.  Cathy declines the INVITE, and cleanup
   proceeds normally.

    Alice                   Bob                     Cathy
     |                                                |
     |   Alice, Bob, and Cathy are participants in    |

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                         SIP Replaces Header


     |   a 3-way call mixed by Bob                    |
     |                                                |
     |<=====================>#<======================>|
     |                       |                        |
     | All three hang up at  |                        |
     | about the same time   |                        |
     |                       |                        |
     | Bob's UA tries to     |                        |
     | setup a 2-way call    |                        |
     | btwn Alice and Cathy  |                        |
     |                       |                        |
     |<---REFER--------------|                        |
     |----INVITE with Replaces--->XX (lost or late)   |
     |                       |                        |
     |                       |<-----BYE/200-----------| the dialog is
     |                       |                        | already dead
     |----INVITE with Replaces----------------------->|    so
     |<---603 Declined--------------------------------| Cathy Declines
     |----ACK---------------------------------------->|
     |----NOTIFY/200-------->|                        |
     |                       |                        |
     |<-----BYE/200--------->|                        |
     |  (either side sends)  |                        |
     |                       |                        |
     |                       |                        |

6.5. An Error Case

   The following example illustrates one reason an INVITE with Replaces
   may fail.  In this example, both Bob and Cathy have a common audio
   codec with Alice, but Bob and Cathy do not share a common codec.
   When Cathy receives an INVITE from Bob with the Replaces header,
   Cathy determines she cannot communicate, sends a 488 response to
   Bob, and maintains her session with Alice.


       Alice                   Bob                        Cathy
        |                       |                           |
        |--INVITE/200/ACK------>|                           |
        |                       |                           |
        |<=audio w/GSM codec===>|                           |
        |                       |                           |
        |----INVITE/200/ACK-------------------------------->|
        |                       |                           |
        |<===audio with G.729 codec========================>|
        |                       |                           |
        |                       |                           |
        |--REFER/200----------->|                           |
        |                       |--INVITE w/Replaces------->|
        |                       |                           |
        |                       |  no codec in common!      |
        |                       |                           |
        |                       |<-488 Not Acceptable Here--|

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                         SIP Replaces Header


        |<--NOTIFY/200----------|--ACK--------------------->|
        |                       |                           |
        |<=====================>|                           |
        |<=================================================>|
        |                       |                           |

6.6. Backwards compatibility with RFC2543 User Agents

   In this example, both Alice and Bob use tags, but Alice wishes to
   replace a dialog at Bob that was initiated by a User Agent that does
   not support tags.


                                                         RFC2543
                                                         User Agent
       Alice                   Bob                       (no tags)
        |                       |                           |
        |                       |<---------INVITE-----------| *1
        |                       |----------200--------------| *2
        |                       |<---------ACK--------------|
        |                       |                           |
        |                       |<=========================>|
        |                       |                           |
        |                       |                           |
     *3 |--INVITE w/Replaces--->|                           |
     *4 |<----200 OK------------|----------BYE------------->| *5
        |-----ACK-------------->|<---------200--------------|
        |                       |                           |
        |<=====================>|                           |
        |                       |                           |

   Message *1: Oldtimer (RFC 2543 User Agent)-> Bob

   INVITE sip:bob@sip.org SIP/2.0
   To: <sip:bob@sip.org>
   From: <sip:oldtimer@sip.org>
   Call-ID: 425928@test-ua.sip.org
   CSeq: 1 INVITE
   Contact: <sip:oldtimer@test-ua.sip.org>

   Message *2: Bob -> Oldtimer

   SIP/2.0 200 OK
   To: <sip:bob@sip.org>;tag=3245
   From: <sip:oldtimer@sip.org>
   Call-ID: 425928@test-ua.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@bobster.sip.org>

   Message *3: Alice -> Bob

   INVITE sip:bob@bobster.sip.org
   To: <sip:bob@sip.org>

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                         SIP Replaces Header


   From: <sip:alice@phone2.sip.org>;tag=8983
   Call-ID: 09870@phone2.sip.org
   CSeq: 1 INVITE
   Contact: <sip:alice@phone2.sip.org>
   Replaces: 425928@test-ua.sip.org;to-tag=3245;from-tag=0

   Message *4: Bob -> Alice

   SIP/2.0 200 OK
   To: <sip:bob@sip.org>;tag=9343
   From: <sip:alice@phone2.sip.org>;tag=8983
   Call-ID: 09870@phone2.sip.org
   CSeq: 1 INVITE
   Contact: <sip:bob@bobster.sip.org>

   Message *5: Bob -> Oldtimer

   BYE sip:oldtimer@test-ua.sip.org SIP/2.0
   To: <sip:oldtimer@sip.org>
   From: <sip:bob@sip.org>;tag=3245
   Call-ID: 425928@test-ua.sip.org
   CSeq: 2 BYE
   Contact: <sip:bob@bobster.sip.org>



7. Security Considerations

   This extension can be used to disconnect or replace participants of
   a multimedia conversation with an attacker.  As such, invitations
   with the Replaces header SHOULD only be accepted in a dialog in
   which the peer has been properly authenticated using a standard SIP
   mechanism, and for which message integrity is checked so that the
   header cannot be added or modified in transit.

   The extensions proposed in this document do not significantly change
   the relative security of SIP devices.  Currently in SIP, an
   eavesdropper who learns the Call-ID, To, and From headers can easily
   modify or destroy a dialog using a reINVITE.  In practice, dialog
   information (Call-ID, to-tag, and from-tag) for most uses of
   Replaces is obtained via subscription to a "call-package" event
   package or via transitivity using the REFER method.  Encryption of
   SIP signaling to insure confidentiality of this information is
   RECOMMENDED.

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

8. IANA Considerations

8.1 Registration of "Replaces" SIP header

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                         SIP Replaces Header



   Name of Header: Replaces

   Short form: none

   Registrant:  Rohan Mahy
                rohan@cisco.com

   Normative description: section 4.1 of this document


8.2 Registration of "replaces" SIP Option-tag

   Name of option: replaces

   Description: Support for the SIP Replaces header

   SIP headers defined: Replaces

   Normative description: This document

   Registrant:  Rohan Mahy
                rohan@cisco.com


8.3 Registration of "687" SIP Response code

   Number of response code: 687

   Default reason phrase: Dialog Terminated

   Registrant:  Rohan Mahy
                rohan@cisco.com

   Normative description: section 4.3 of this document


9. To Do and Open Issues

   Open Issues:

   - MAJOR OPEN ISSUE: When no matching dialog is found should we
   ignore the Replaces header and accept the INVITE, or reject the
   INVITE?

   - Are the proposals for early dialog and terminated dialog matching
   acceptable?

   - Is the proposed tag matching scheme for pre-bis UAs acceptable?

   To Do:

   - Update references

 Biggs, Dean, Mahy        Expires: July 2002                        14

10. References


   [SIP] M. Handley, E. Schooler, and H. Schulzrinne, "SIP: Session
   Initiation Protocol", RFC2543, Internet Engineering Task Force,
   Nov 1998.

   [bis] Rosenberg, Schulzrinne, Camarillo, Johnston, Peterson, Sparks,
   Handley, and Schooler, "SIP: Session Initiation Protocol", Internet-
   Draft <draft-ietf-sip-2543bis-05.txt>, IETF, Oct 2001.

   [cc-framework] B. Campbell, "SIP Call Control - Framework ",
   Internet Draft <draft-campbell-sip-cc-framework-02.txt>, IETF,
   Mar 2001.  Work in progress.

   [cc-models] R. Mahy, "A Call Control Model for SIP", Internet Draft
   <draft-mahy-sip-cc-models-00.txt>, IETF; July 2001. Work in
   progress.

   [REFER] R. Sparks, "The REFER Method", Internet Draft <draft-ietf-
   sip-refer-02.txt>, IETF; Oct 2001. Work in progress.

   [cc-transfer] R. Sparks, "SIP Call Control - Transfer", Internet
   Draft <draft-ietf-sip-cc-transfer-05.txt>, IETF; July 2001. Work in
   progress.

   [3pcc] J. Rosenberg, J. Peterson, H. Schulzrinne, "Third Party Call
   Control in SIP", Internet Draft <draft-rosenberg-sip-3pcc-01.txt>,
   IETF;  Nov. 2000.  Work in progress

   [service-examples] A. Johnston, et al., "SIP Service Examples",
   Internet Draft <draft-ietf-sipping-service-examples-03.txt>, IETF;
   Nov 2001.  Work in progress.

   [RFC2026] S Bradner, "The Internet Standards Process -- Revision 3",
   RFC2026 (BCP), IETF, October 1996.

   [RFC2119] S. Bradner, "Key words for use in RFCs to indicate
   requirement levels," Request for Comments (Best Current Practice)
   2119, Internet     Engineering Task Force, Mar. 1997.

   [BNF] D Crocker, P Overell, "Augmented BNF for Syntax
   Specifications: ABNF", RFC2234, IETF, Nov 1997.


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. Author's Addresses

 Biggs, Dean, Mahy        Expires: July 2002                        15
                         SIP Replaces Header



   Billy Biggs
   bbiggs@dumbterm.net

   Rick Dean
   rfc@fdd.com

   Rohan Mahy
   Cisco Systems
   170 West Tasman Dr, MS: SJC-21/3/3
   Phone: +1 408 526 8570
   Email: rohan@cisco.com



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