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
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
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as reference material or to cite them other than as "work in
progress."
The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
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.
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
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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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