SIP WG R. Mahy
Internet-Draft Cisco Systems, Inc.
Expires: August 1, 2004 O. Levin
Microsoft Corporation
F. Audet
Nortel Networks
Feb 2004
Remote Call Control in SIP using the REFER method and the
session-oriented dialog package
draft-mahy-sip-remote-cc-01.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
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This Internet-Draft will expire on August 1, 2004.
Copyright Notice
Copyright (C) The Internet Society (2004). All Rights Reserved.
Abstract
This document describes how to use the SIP REFER method and the
dialog package to manipulate conversations, dialogs, and sessions on
remote User Agents. This functionality is most useful for collections
of loosely coupled User Agents that wish to present a coordinated
user experience. It does not require a Third-Party Call Control
controller to be involved in any of the manipulated dialogs.
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Table of Contents
1. Terminology . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Remote control operations . . . . . . . . . . . . . . . . . 4
4. Implementing these operations . . . . . . . . . . . . . . . 6
5. Examples of Remote Call Control Operations SIP Call Flows . 8
5.1 Make Call Operation . . . . . . . . . . . . . . . . . . . . 8
5.2 Answer Call Operation . . . . . . . . . . . . . . . . . . . 10
5.3 Clear Connection . . . . . . . . . . . . . . . . . . . . . . 12
5.4 Deflect Call . . . . . . . . . . . . . . . . . . . . . . . . 14
5.5 Single Step Transfer Call . . . . . . . . . . . . . . . . . 16
5.6 Complete Transfer Between Sessions . . . . . . . . . . . . . 18
5.7 Hold Call . . . . . . . . . . . . . . . . . . . . . . . . . 18
5.8 Retrieve Call . . . . . . . . . . . . . . . . . . . . . . . 20
5.9 Conference Call . . . . . . . . . . . . . . . . . . . . . . 21
5.10 Single Step Conference Call . . . . . . . . . . . . . . . . 23
5.11 Set Do Not Disturb . . . . . . . . . . . . . . . . . . . . . 25
5.12 Set Forwarding . . . . . . . . . . . . . . . . . . . . . . . 26
5.13 Alternate Call . . . . . . . . . . . . . . . . . . . . . . . 28
5.14 Consultation Call . . . . . . . . . . . . . . . . . . . . . 29
6. Examples of implementing remote call control operations
with Refer-To URI . . . . . . . . . . . . . . . . . . . . . 30
6.1 Make Call Operation . . . . . . . . . . . . . . . . . . . . 30
6.2 Answer Call Operation . . . . . . . . . . . . . . . . . . . 30
6.3 Clear Connection . . . . . . . . . . . . . . . . . . . . . . 30
6.4 Deflect Call . . . . . . . . . . . . . . . . . . . . . . . . 31
6.5 Complete Transfer Between Calls . . . . . . . . . . . . . . 31
7. User Agent Behavior . . . . . . . . . . . . . . . . . . . . 31
7.1 Organizing requests within dialogs . . . . . . . . . . . . . 31
7.2 Addressing the relevant parties . . . . . . . . . . . . . . 32
7.3 Selecting an existing dialog context for the triggered
request . . . . . . . . . . . . . . . . . . . . . . . . . . 33
8. Authorizing remote call control requests . . . . . . . . . . 34
9. Security Considerations . . . . . . . . . . . . . . . . . . 34
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . 35
11. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . 35
Normative References . . . . . . . . . . . . . . . . . . . . 35
Informational References . . . . . . . . . . . . . . . . . . 36
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 37
Intellectual Property and Copyright Statements . . . . . . . 38
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1. Terminology
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].
To simplify discussions related to the REFER method and its
extensions, three new terms will be used:
REFER-Issuer: the UA issuing the REFER request. Sometimes this
document will also use the term "controller".
REFER-Recipient: the UA receiving the REFER request
REFER-Target: the UA designated in the Refer-To URI
2. Introduction
The SIP [1] core protocol describes how User Agents originate and
terminate sessions. The SIP call control framework [13] also
describes how User Agents involved in these sessions can manipulate
conversations based on the sessions to provide functionality such as
transfer, pickup, and barge-in. Third-Party Call Control [15] goes on
to describe how a controller can setup dialogs with a number of
participants in order to manipulate sessions among the participants.
Remote call control is the manipulation of conversations and
session-oriented dialogs by a UA that is not directly involved in any
of the relevant conversations, dialogs, or sessions. This
manipulation generally involves sending REFER [4] requests to a UA
which is directly involved, using information obtained via the dialog
package [5]. (Although many are familiar with REFER only as used to
implement call transfer [14], the authors of the REFER method never
intended this limitation. In fact the REFER method was created when
the SIP working group realized that a generic request to ask another
UA to do something on your behalf was much more powerful than just
doing transfers.) The Extensions to the REFER mechanism [6] describes
the use of REFER for that purpose.
Unlike the Third-Party Call Control (3pcc) model which requires its
controller to act as a B2BUA and maintain dialog state for all
relevant dialogs, all the SIP entities involved in remote call
control using REFER are just regular SIP User Agents. For convenience
we can still describe the SIP entity that sends requests to
manipulate remote sessions "the controller", but this is just a
logical role. A UA that acts as a controller for one request can
terminate and originate its own sessions, and even receive remote
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call control requests as other requests.
Some readers may question if remote call control is an appropriate
use of SIP, instead possibly something more appropriate for MGCP
[19] or Megaco [20]. The authors believe that remote call control
is an appropriate and natural extension of SIP. Manipulating
sessions and dialogs is certainly consistent with core
functionality of SIP. This usage of SIP is much different from an
MGCP or Megaco master/slave approach. For example, multiple UAs
can send remote call control requests. All remote call control
requests can be refused based on local authorization policy or if
the request doesn't make sense. Finally, each UA is still fully
responsible and authoritative for their own dialog and session
state. In other words, each UA still has the last word on its
sessions and dialogs, even if asked to perform manipulations on
that state by another entity. This seems completely appropriate
with the design of SIP. In fact these requirements and goals are
well documented in the SIP Call Control Framework.
Remote call control is especially useful for collections of loosely
coupled User Agents which would like to present a coordinated user
experience. Among other things, this allows User Agents which handle
orthogonal media types but which would like to be present in a single
conversation to add and remove each other from the conversation as
needed. This is especially appropriate when coordinating
conversations among organizers, general purpose computers, and
special purpose communications appliances like telephones, Internet
televisions, in-room video systems, electronic whiteboards, and
gaming devices.
For example using remote call control, an Instant Messaging client
could initiate a multiplayer gaming session and an audio session to a
chat conversation. Likewise a telephone could add an electronic
whiteboard session to a voice conversation. Finally, a computer or
organizer could cause a nearby phone to dial from numbers or URIs in
a document, email, or address book; allow users to answer or deflect
incoming calls without removing hands from the computer keyboard;
place calls on hold; and join other sessions on the phone or
otherwise.
Remote call control can also be used in two directions. A computer
could remote control a nearby phone and make it dial a SIP URI, but
the SIP phone could then also remote control the computer into
terminating the session upon the user hanging up the phone.
3. Remote control operations
Remote call control can be used to request a variety of operations.
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Commonly used operations include the following:
Make Session - Initiate a new session.
Clear Session - Terminate a session.
Answer - Succesfully respond to a session invitation.
Deflect Session - Redirect a session invitation.
Reject Session - Reject a session invitation.
Single Step Transfer Session - Transfer a session to another UA in
a single step. The transferring device is no longer involved with
the session after single step transfer is completed. This is
described as a "Blind Transfer" in [14]
Complete Transfer Between Sessions - Transfer the remote UA of one
existing session to communicate directly with the remote UA of
another existing session. Once the transfer completes, the remote
controlled UA is no longer involved with either session.
Hold Session - Holds a call at the holding UA. Note that this
operation would cause whatever call control would occur locally
when this operation is selected (for example a simple hold which
makes the call inactive, or a service such as music on hold using
a remote stream.
Retrieve Session - Retrieves a held call at the retrieving device.
Merge Sessions - Conferences together two existing sessions at a
UA.
Single Step Conference Call - Initiate another session and merge
it to an existing session into a new conference.
Alternate Sessions - Place an existing session on hold, and
retrieves a previously held session. This operation is a
combination of the Hold Call and Retrieve Call operations.
Consultation Session - Places an existing session on hold at the
UA and initiates a new session from the UA. This operation is a
combination of the Hold Call and Make Call operations.
Set Do Not Disturb - Will cause the remote controlled UA to reject
further session invitations with a proper response indicating that
it is not availble. This operation does not require the
participation of the controller for subsequent session
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invitations. The target may cause this operation via local
processing or for example by updating presence [17] status which
is consumed by systems performing call routing.
Set Forwarding - Will cause the remote controlled UA to redirect
further session invitations to another URI. This operation does
not require the participation of the controller for subsequent
session invitations. The target may cause this operation via local
processing or for example by manipulating SIP registrations.
4. Implementing these operations
In order to convey requests for remote call control operations, there
are several syntactic approaches possible. The most obvious is to
use the existing Refer-To URI syntax. However, escaping long URIs is
error-prone and obfuscates the intent of a request. Another option
mentioned as a REFER extension is carrying the Refer-To target as a
message/sipfrag [12] body. However, encoding remote call control
operations which deal with with more that one session in a single URI
are still cumbersome. Also, both these approaches rely on implicit
behavior or undefined URI conventions. This document uses this
approach for operations which only require a straightforward
encoding.
Alternatively, the Refer-To URI could be a Universal Resource Name
(URN) [21] which could describe a particular operation such as Hold
or Retrieve. Combined with the dialog-identifiers of an existing
session conveyed as parameters of the Refer-To header, this would
permit explicit operations which do not need additional parameters or
handle more than a single session. For example, the following could
represent a Hold operation of a session with the Call-ID "123":
Refer-To: <urn:ietf:params:sip:remotecc:hold>
;call-id=123;remote-tag=aaa;local-tag=bbb
Note however that the most interesting remote call control operations
(such as Complete Transfer Between Sessions and Merge) operate on
more than one session and may require additional parameters. These
are still abstract operations, but they operate on more than one
target. Using an explicit description of these parameters in a new
MIME body is an ideal way to provide this additional functionality,
and the only approach which works with all the sample remote call
control operations in this document.
An additional benefit of a remote call control body is that certain
details of these operations can be abstracted. For example, a Clear
Session operation can cause either a CANCEL, BYE or appropriate
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response to be sent depending on context. A Hold operation can
result in whatever user-visible functionality occurs when a Hold is
selected locally (for example a simple hold, tone-on-hold,
music-on-hold, animated cartoon characters, etc.). A Merge Sessions
operation can use whatever conference resource would be used by the
UA itself (a local conferencing focus, a discovered focus, or an
administratively configured focus).
This document therefore describes a MIME body for remote call control
operations conveyed in the body of a REFER request. Remote call
control operations using a remote call control MIME type body are
operations that are typically more abstract or complex information
than can be practically be achieved with a message/sipfrag body or a
Refer-to URI.
This document makes frequent use of the REFER extensions defined in
[6] to carry out these operation. In particular, we frequently
reference bodies in the Refer-To header using a Content-ID URI
(cid:).
While a remote call control MIME body is not defined in this
document, we use the MIME type application/remotecc in our examples.
The following is an example of a REFER with a Remote Call Control
operation with such a body:
REFER sip:reg2@10.1.1.3 SIP/2.0
Via: SIP/2.0/TCP issuer.example.com.com;branch=z9hG4bK-a-1
To: "Alice's phone" <sip:reg2@10.1.1.3>
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123@issuer.example.com
CSeq: 2 REFER
Max-Forwards: 70
Contact: sip:alice1@10.1.1.2
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: <cid:1239103912039@issuer.example.com>
;call-id=
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
----------------------------
| Remote Call Control Body |
----------------------------
The application/dialog-info+xml package can be used to provide
information about the status of dialogs. The examples in this
specification assume that the dialog event package is sufficient to
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provide the necessary feedback for remote call control operations.
5. Examples of Remote Call Control Operations SIP Call Flows
This entire section provide non-normative examples of functionality
where a computer or PDA manipulates a telephone. The behavior for
remote call control with other types of devices is similar, but
describing similar manipulations for other media or device types
would naturally use a different set of vocabulary.
The following sub-sections provide an example for every operation
described in the previous section.
The following notes are applicable to all the call flows in the
subsections below:
It is assumed that Alice's PC or PDA has subscribed to Alice's
Phone dialog package. All of the NOTIFY messages are notifications
about changes in the dialog state at Alice's phone. No additional
remote call control event packages are shown, but it is not
precluded that one be defined later.
As specified in [6], there is no no implicit subscrition on all
REFER messages between Alice's PDA or PC and Alice's Phone with
the extended REFER mechanism.
Via and Max-Forward headers and session descriptions are omitted
for brevity and clarity. In some cases, display names are added
for simplify the task of the reader following the examples. Note
that URIs in SIP cannot wrap lines. Due to RFC formatting
conventions, this draft splits URIs across lines where the URI
would exceed 72 characters. A backslash character marks where this
line folding has taken place.
5.1 Make Call Operation
In message 1, Alice's PC or PDA asks her phone to "call Bob" (message
2), which eventually results in an early dialog (3) with one of Bob's
Contacts. Bob sends a ringing indication (4) which triggers Alice's
phone to send a notification (5) of "early" to Alice's PC or PDA.
Then Bob answers the phone (6) wich triggers Alice's phone to send a
notification (7) of "confirmed" to Alice's PC or PDA.
Alice's Alice's Bob
PC or PDA Phone
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| Call-ID: 123 | Call-ID: 456 |
| | |
1 |---REFER/202------->| |
| 2 |---INVITE--------->|
|<--NOTIFY/200-------| 3 |
| |<----180-----------| 4
|<--NOTIFY/200-------| 5 |
| |<----200/ACK-------| 6
|<--NOTIFY/200-------| 7 |
| | |
In this first example, in Message 1a, traditional Refer-To encoding
is used. Message 1b shows how to request this same operation with an
embedded remote call control MIME body.
Message 1a:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Refer-To: "Bob" <sip:bob@example.net;method=INVITE>
Message 1b:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| MakeCall |
| From: sip:reg2@10.1.1.3 |
| To: bob@example.net |
| other parameters |
--------------------------------
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Message 2:
INVITE sip:bob@example.net SIP/2.0
To: "Bob" <sip:bob@example.net>
From: "Alice" <sip:alice@example.com>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Content-Type: application/sdp
Content-Length: xxx
Message 3:
NOTIFY indicates "trying".
Message 4:
SIP/2.0 180 Ringing
To: "Bob" <sip:bob@example.net>;tag=uvw
From: "Alice's phone" <sip:reg2@10.1.1.3>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Message 5:
NOTIFY will indicates "early".
Message 6:
SIP/2.0 200 OK
To: "Bob" <sip:bob@example.net>;tag=uvw
From: "Alice's phone" <sip:reg2@10.1.1.3>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 7:
NOTIFY indicates "confirmed".
5.2 Answer Call Operation
In message 1, Bob makes a call to Alice's Phone. A notification (2)
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of "trying" is sent to Alice. Alice's phone automatically sends a
"ringing" (3) to Bob. Another notification (4) of "early" is then
sent to Alice's PC. Alice then instructs (5) her PDA to tell the
phone to answer the call (6). Alice's phone sends a notification (7)
of "confirmed" to Alice's PDA.
Alice's Alice's Bob
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 |
| | |
| |<--INVITE----------| 1
|<--NOTIFY/200-------| 2 |
| 3 |------180--------->|
|<--NOTIFY/200-------| 4 |
5 |---REFER/202------->| |
| 6 |------200/ACK----->|
|<--NOTIFY/200-------| 7 |
| | |
Message 1:
INVITE sip:alice@example.com SIP/2.0
To: "Alice" <sip:alice@example.com>
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 2:
NOTIFY indicates "trying".
Message 3:
SIP/2.0 180 Ringing
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Message 4:
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NOTIFY indicates "early".
Message 5:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| answercall |
| Call= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| other parameters |
--------------------------------
Meassage 6:
SIP/2.0 200 OK
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Content-Type: application/sdp
Content-Length: xxx
Message 7:
NOTIFY indicates "confirmed".
5.3 Clear Connection
Alice's Phone and Bob's contact are currently in an established
dialog. In message 1, Alice's PC or PDA asks her phone to "clear the
connection" with Bob's phone. (message 2).
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Alice's Alice's Bob
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 |
| | |
| |<==Estab. dialog==>|
1 |---REFER/202------->| |
| 2 |------BYE/200----->|
|<--NOTIFY/200-------| 3
| | |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| clearconnection |
| Call= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| other parameters |
--------------------------------
Message 2:
BYE is sent to Bob's contact.
Message 3:
NOTIFY indicates "trying".
Message 3:
NOTIFY will indicates "terminated".
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5.4 Deflect Call
In message 1, Bob makes a call to Alice's Phone. A notification (2)
of "trying" is sent to Alice. Alice's phone automatically sends a
"ringing" (3) to Bob. Another notification (4) of "early" is then
sent to Alice's PC. Alice then instructs (5) her PDA to tell the
phone to deflect the call (6) to Cathy. Alice's phone sends a
notification (7) of "terminated" to Alice's PDA. Bob's will attempt
the call to Cathy (8).
Alice's Alice's Bob Cathy
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | |
| | | |
| |<--INVITE----------| 1 |
|<--NOTIFY/200-------| 2 | |
| 3 |------180--------->| |
|<--NOTIFY/200-------| 4 | |
5 |---REFER/202------->| | |
| 6 |------302/ACK----->| |
|<--NOTIFY/200-------| 7 | |
| 2 | 8 |----INVITE-------->|
Message 1:
INVITE sip:alice@example.com SIP/2.0
To: "Alice" <sip:alice@example.com>
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 2:
NOTIFY indicates "trying".
Message 3:
SIP/2.0 180 Ringing
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
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CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Message 4:
NOTIFY indicates "early".
Message 5:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| deflectcall |
| Call= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| other parameters |
--------------------------------
Message 6:
SIP/2.0 302 Moved Temporarily
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Cathy" <sip:cathy@example.net>
Message 7:
NOTIFY indicates "rejected".
Mesage 8:
INVITE sip:cathy@example.net SIP/2.0
To: "Cathy" <sip:cathy@example.net>
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From: "Bob" <sip:bob@example.net>;tag=pqr
Call-ID: 789
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
5.5 Single Step Transfer Call
Alice's Phone and Bob's contact are currently in an established
dialog. In message 1, Alice's PC or PDA requests that a request be
made to transfer the call to Cathy. Alice's phone sends a request (2)
to Bob's contact to transfer the call to Cathy (3). Call from Bob's
contact to Cathy rings (4), is answered (5). Bob's contact sends a
notification (6) to Alice's phone because of the REFER implicit
subsription. Alice's phone then terminates the session with Bob's
contact (7) and sends a notification of "terminated" to Alice's PC or
PDA.
Alice's Alice's Bob Cathy
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | |
| | | |
| |<==Estab. dialog==>| |
1 |---REFER/202------->| | |
| 2 |----REFER/202----->| |
| | 3 |----INVITE-------->|
| | |<----180-----------| 4
| | |<----200/ACK-------| 5
| |<--NOTIFY/200------| 6 |
| 7 |---BYE/200-------->| |
|<--NOTIFY/200-------| 8 |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
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Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| transfer |
| FirstCall= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| SecondCall= |
| sip:cathy@example.net |
| other parameters |
--------------------------------
Message 2:
REFER sip:bob@example.net SIP/2.0
To: "Bob" <sip:bob@example.net>
From: "Alice" <sip:alice@example.com>;tag=xyz
Call-ID: 456
CSeq: 1 REFER
Refer-To: "Cathy" <sip:cathy@example.net;method=INVITE>
Mesage 3:
INVITE sip:cathy@example.net SIP/2.0
To: "Cathy" <sip:cathy@example.net>
From: "Bob" <sip:bob@example.net>;tag=pqr
Call-ID: 789
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Messages 4 & 5:
180, 200, ACK when call is set up with Cathy.
Message 6:
NOTIFY will include the sigfrag as per the REFER implicit subsription.
Message 7:
Bob's contact clears the call.
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Message 8:
NOTIFY indicates "confirmed".
5.6 Complete Transfer Between Sessions
TBD
5.7 Hold Call
In message 1, Alice's PC or PDA asks her phone to put on hold the
already established dialog with Bob. Alice's phone sends a re-INVVITE
to Bob's contact to put the media stream on hold. Note that a call
hold is different concept than held media. In fact, a user can be
placed on hold, and be provided with music on hold. A held call is a
logical state which could be useful for a number of things such as
monitoring the amount of time a user stays in a queue. This diagram
does not illustrate any event package to illustrate that a can can be
held.
Alice's Alice's Bob
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 |
| | |
| |<==Estab. dialog==>|
1 |---REFER/202------->| |
| 2 |---INVITE--------->|
|<--NOTIFY/200-------| 3 |
| |<----200/ACK-------| 4
|<--NOTIFY/200-------| 5 |
| | |
Message 1:
Accept: application/dialog-info+xml
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
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Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| hold |
| Call= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| other parameters |
--------------------------------
Message 2:
INVITE sip:bob@example.net SIP/2.0
To: "Bob" <sip:bob@example.net>
From: "Alice" <sip:alice@example.com>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Content-Type: application/sdp
Content-Length: xxx
SDP to indicate held media for example.
Message 3:
NOTIFY indicates "trying".
Message 4:
SIP/2.0 200 OK
To: "Bob" <sip:bob@example.net>;tag=uvw
From: "Alice's phone" <sip:reg2@10.1.1.3>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 5:
NOTIFY indicates "confirmed".
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5.8 Retrieve Call
In message 1, Alice's PC or PDA asks her phone to retreive an held
call with Bob. Alice's phone sends a re-INVVITE to Bob's contact to
resume the media stream which was already on hold. Note that a call
hold is different concept than held media. In fact, a user can be
placed on hold, and be provided with music on hold. A held call is a
logical state which could be useful for a number of things such as
monitoring the amount of time a user stays in a queue. This diagram
does not illustrate any event package to illustrate that a can can be
held.
Alice's Alice's Bob
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 |
| | |
| |<==Estab. dialog==>|
1 |---REFER/202------->| |
| 2 |---INVITE--------->|
|<--NOTIFY/200-------| 3 |
| |<----200/ACK-------| 4
|<--NOTIFY/200-------| 5 |
| | |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| retreive |
| Call= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
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| other parameters |
--------------------------------
Message 2:
INVITE sip:bob@example.net SIP/2.0
To: "Bob" <sip:bob@example.net>
From: "Alice" <sip:alice@example.com>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Content-Type: application/sdp
Content-Length: xxx
SDP to indicate re-established media.
Message 3:
NOTIFY indicates "trying".
Message 4:
SIP/2.0 200 OK
To: "Bob" <sip:bob@example.net>;tag=uvw
From: "Alice's phone" <sip:reg2@10.1.1.3>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 5:
NOTIFY indicates "confirmed".
5.9 Conference Call
Alice's Phone and Bob's contact are currently in an established
dialog. Alice's Phone and Cathy's contact are also currently in an
established dialog. In message 1, Alice's PC or PDA requests that a
conference be established between the two calls (i.e., a conference
between Alice's Phone, Bob's contact and Cathy's contact. Alice's
phone establish a call with a conference bridge (2-5). Alice's phone
sends a request (6) to Bob's contact to transfer the call to the same
conference bridge (7). Alice's phone is notified (implicit REFER
subscription) of the successful transfer to the conference bridge (8)
and clears the call with Bob (9). Alice's phone sends a request (10)
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to Cathy's contact to transfer the call to the same conference bridge
(11). Alice's phone is notified (implicit REFER subscription) of the
successful transfer to the conference bridge (12) and clears the call
with Cathy (13). The call flow does not show an event package for the
successful remote conference invocation.
Alice's Alice's Bob Cathy Conf. Bridge
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | Call-ID: 789 | Call-ID: ABC |
| | | | |
| |<=Est.dialog=>| | |
| |<===Established dialog======>| |
1 |---REFER/202-->| | | |
| 2 |-------------------INVITE------------------>|
|<--NOTIFY/200--| 3 | | |
| |<----------------200/ACK--------------------| 4
|<--NOTIFY/200--| 5 | | |
| 6 |--REFER/202-->| | |
| | 7 |--------INVITE/200/ACK------>|
| |<-NOTIFY/200--| 8 | |
| 9 |---BYE/200--->| | |
| 10 |-----------REFER/202-------->| |
| | 8 | 11 |--INVITE----->|
| | | 12 |<--200/ACK----|
| |<-----------NOTIFY/200-------| 13 |
| 14 |--------------BYE/200------->| |
| | | | |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
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| conference |
| FirstCall= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| SecondCall= |
| sip:cathy-pc@192.168.8.8 |
| call-id:789 |
| remote-tag=abc |
| local-tag=def |
| other parameters |
--------------------------------
Message 3:
NOTIFY indicates "trying".
Mesage 5:
NOTIFY indicates "confirmed".
5.10 Single Step Conference Call
A single step conference call is the same operation as a conference
call, except that one of the legs is a SIP URI instead of an
established dialog. Alice's Phone and Bob's contact are currently in
an established dialog. In message 1, Alice's PC or PDA requests that
a conference be established between the the existing call with Bob's
contact and with Cathy (i.e., a conference between Alice's Phone,
Bob's contact and Cathy. Alice's phone establish a call with a
conference bridge (2-5). Alice's phone sends a request (6) to Bob's
contact to transfer the call to the same conference bridge (7).
Alice's phone is notified (implicit REFER subscription) of the
successful transfer to the conference bridge (8) and clears the call
with Bob (9). Alice's phone sends a request (10) to Cathy's contact
to transfer the call to the same conference bridge (11). Alice's
phone is notified (implicit REFER subscription) of the successful
transfer to the conference bridge (12). The call flow does not show
an event package for the successful remote single step conference
invocation.
Alice's Alice's Bob Cathy Conf. Bridge
PC or PDA Phone
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| Call-ID: 123 | Call-ID: 456 | Call-ID: 789 | Call-ID: ABC |
| | | | |
| |<=Est.dialog=>| | |
1 |---REFER/202-->| | | |
| 2 |-------------------INVITE------------------>|
|<--NOTIFY/200--| 3 | | |
| |<----------------200/ACK--------------------| 4
|<--NOTIFY/200--| 5 | | |
| 6 |--REFER/202-->| | |
| | 7 |--------INVITE/200/ACK------>|
| |<-NOTIFY/200--| 8 | |
| 9 |---BYE/200--->| | |
| 10 |-----------REFER/202-------->| |
| | 8 | 11 |--INVITE----->|
| | | 12 |<--200/ACK----|
| |<-----------NOTIFY/200-------| 13 |
| | | | |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| conference |
| FirstCall= |
| sip:line1@192.168.0.5 |
| call-id:456 |
| remote-tag=uvw |
| local-tag=xyz |
| SecondCall= |
| sip:cathy@example.net |
| other parameters |
--------------------------------
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Message 3:
NOTIFY indicates "trying".
Mesage 5:
NOTIFY indicates "confirmed".
5.11 Set Do Not Disturb
In message 1, Alice sends a request so that her phone will be in "Do
not disturb" or "Make set busy" mode. Any subsequent invitation (2)
send to Alice's phone will result in the session being rejected with
response 480 "Temporarily not available" (or 486 "Busy Here", or any
other appropriate code) without any interaction from Alice's PC or
PDA.
Alice's Alice's Bob
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 |
| | |
1 |---REFER/202------->| |
| |<--INVITE----------| 2
|<--NOTIFY/200-------| 3 |
| 4 |------480/ACK----->|
|<--NOTIFY/200-------| 5 |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
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--------------------------------
| Remote Call Control Body |
| donotdisturb |
| reason=480 |
| other parameters |
--------------------------------
Message 2:
INVITE sip:alice@example.com SIP/2.0
To: "Alice" <sip:alice@example.com>
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 3:
NOTIFY indicates "trying".
Message 4:
SIP/2.0 480 Temporarily unavailable
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Alice's Phone" <sip:reg2@10.1.1.3>
Message 5:
NOTIFY indicates "rejected".
5.12 Set Forwarding
In message 1, Alice sends a request so that her phone will be "call
forwarded" to Cathy. Any subsequent invitation (2) send to Alice's
phone will result in the session being forewared with response 302
"Move temporarily" without any interaction from Alice's PC or PDA.
Alice's Alice's Bob Cathy
PC or PDA Phone
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| Call-ID: 123 | Call-ID: 456 | |
| | | |
1 |---REFER/202------->| | |
| |<--INVITE----------| 2 |
|<--NOTIFY/200-------| 3 | |
| 4 |------302/ACK----->| |
|<--NOTIFY/200-------| 5 6 |----INVITE-------->|
| | | |
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: extended-refer
Refer-To: cid:1239103912039@issuer.example.com
Content-Type: application/remotecc
Content-Id: <1239103912039@issuer.example.com>
Content-Length: ...
--------------------------------
| Remote Call Control Body |
| setforwarding |
| destination |
| sip:cathy@example.net |
| forwardingtype=always |
| other parameters |
--------------------------------
Message 2:
INVITE sip:alice@example.com SIP/2.0
To: "Alice" <sip:alice@example.com>
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
Message 3:
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NOTIFY indicates "trying".
Message 4:
SIP/2.0 302 Moved temporarily
To: "Alice" <sip:alice@example.com>;tag=uvw
From: "Bob" <sip:bob@example.net>;tag=xyz
Call-ID: 456
CSeq: 1 INVITE
Contact: "Cathy" <sip:cathy@example.net>
Message 5:
NOTIFY indicates "rejected".
Message 6:
INVITE sip:cathy@example.net SIP/2.0
To: "Cathy" <sip:cathy@example.net>
From: "Bob" <sip:bob@example.net>;tag=pqr
Call-ID: 789
CSeq: 1 INVITE
Contact: "Bob's Contact" <sip:line1@192.168.0.5>
Content-Type: application/sdp
Content-Length: xxx
5.13 Alternate Call
Alternate call is not really an operation by itself. It is a a hold
operation followed by a retrieve operation. This section is included
only to illustrate how those two operations can be combined to
provide an Alternate Call service. In message 1, Alice's PC or PDA
asks her phone to put on hold the already established dialog with
Bob. Alice's phone sends a re-INVITE to Bob's contact (2) to put the
media stream on hold. Alice's PC or PDA then asks her phone (6) to
retrieve her previously held call with Cathy (7).
Alice's Alice's Bob Cathy
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | Call-ID- 789 |
| | | |
| |<==Estab. dialog==>| |
| |<========Call on hold=================>|
1 |---REFER/202------->| | |
| 2 |---INVITE--------->| |
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|<--NOTIFY/200-------| 3 | |
| |<----200/ACK-------| 4 |
|<--NOTIFY/200-------| 5 | |
| | | |
6 |---REFER/202------->| | |
| 7 |-----------------INVITE--------------->|
|<--NOTIFY/200-------| 8 | |
| |<----200/ACK---------------------------| 9
|<--NOTIFY/200-------| 10 | |
| | | |
5.14 Consultation Call
Consultation call is not really an operation by itself. It is a hold
opeeraiton followed by a make call operation. This section is
included only to illustrate how those two operation can be combined
to provide a Consultation Call service. In message 1, Alice's PC or
PDA asks her phone to put on hold the already established dialog with
Bob. Alice's phone. Alice's phone sends a re-INVITE to Bob's contact
(2) to put the media stream on hold. Alice's PC or PDA then asks her
phone (6) to make a call to Cathy (7).
Alice's Alice's Bob Cathy
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | Call-ID- 789 |
| | | |
| |<==Estab. dialog==>| |
1 |---REFER/202------->| | |
| 2 |---INVITE--------->| |
|<--NOTIFY/200-------| 3 | |
| |<----200/ACK-------| 4 |
|<--NOTIFY/200-------| 5 | |
| | | |
6 |---REFER/202------->| | |
| 7 |-----------------INVITE--------------->|
|<--NOTIFY/200-------| 8 | |
| |<-----------------180------------------| 9
|<--NOTIFY/200-------| 10 | |
| |<---------------200/ACK----------------| 11
|<--NOTIFY/200-------| 12 | |
| | | |
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6. Examples of implementing remote call control operations with Refer-To
URI
This section provided examples of how to implement some of the simple
operations of the previous sections without using a REFER MIME body
and relying instead of the Refer-To URI. All the call flows are
assumed to be the same as per the previous section. Only the changed
REFER message is shown.
6.1 Make Call Operation
This example is already discussion in a previous section.
6.2 Answer Call Operation
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: response-refer
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5;method=INVITE;response=200?
Call-ID=456&To=alice%40example.com;tag=uvw&From=bob%40example.net;tag=xyz>
6.3 Clear Connection
Message 1:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: response-refer
Refer-To: "Bob's contact" <sip:line1@192.168.0.5;method=BYE?
Call-ID=456&To=alice%40example.com;tag=uvw&From=bob%40example.net;tag=xyz>
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6.4 Deflect Call
Message 5:
REFER sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 2 REFER
Accept: application/dialog-info+xml
Require: response-refer
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5;method=INVITE;response=302?
Call-ID=456&To=alice%40example.com;tag=uvw&From=bob%40example.net;tag=xyz>
6.5 Complete Transfer Between Calls
7. User Agent Behavior
7.1 Organizing requests within dialogs
REFER messages used for call transfer always arrive within an
existing dialog which was created with the INVITE method. In general,
REFER messages can be sent within an existing dialog, or they can
start a new dialog (the dialog used by the implicit subscription they
create). In many use cases of remote call control, receiving
notifications about the status of a REFER request are superfluous, as
the Refer-Issuer typically maintains a long duration subscription to
the dialog package. This situation is complicated by the possible
presence of the norefersub option-tag, defined in section 7 of [6].
When the norefersub option tag is present, a REFER request which
would have created a new subscription and dialog becomes a standalone
transaction instead. Each such standalone REFER transaction MUST use
a new (unique) Call-Id header field value. The following three use
cases are suggested:
1. In the most common usage, the controller maintains a long duration
subscription to the dialog package, and sends REFER requests within
that dialog. Each REFER is sent within the context of the dialog
created for the subscription to the dialog package, and could
include the norefersub option-tag in a Supported header field value.
2. Occasionally the dialog package is only supported via a dialog
state agent separate from the Refer-Receiver, in which case the
controller maintains a long duration subscription to the dialog
package to a dialog state agent, and the controller sends these
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individual REFER requests as standalone requests each with a
different (unique) Call-ID header field value, which could also
include the norefersub option-tag in a Supported header field value.
3. In some cases, the controller does not typically maintain a dialog
package subscription for the Refer-Receiver. This might be the case
for a "webdialer" or other application which associates with other
UAs on an adhoc and intermitent basis. An initial REFER request is
sent to start a new dialog, which is followed by notifications for
the refer event type (the norefersub option-tag SHOULD NOT be used in
this case). These notifications could contain message/sipfrag or
application/dialog-info+xml notification bodies as described in
Section 4 of [6].
Message 1:
SUBSCRIBE sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's phone" <sip:reg2@10.1.1.3>
From: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
Call-ID: 123
CSeq: 1 SUBSCRIBE
Event: dialog
Contact: <sip:alice1@10.1.1.2>
Message 2:
NOTIFY sip:reg2@10.1.1.3 SIP/2.0
To: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;tag=abc
From: "Alice's phone" <sip:reg2@10.1.1.3>;tag=def
Call-ID: 123
CSeq: 1 NOTIFY
Event: dialog
Contact: <sip:reg2@10.1.1.3>
Subscription-State: active;expires=3600
Content-Type: application/dialog-info+xml
Content-Length: xxx
7.2 Addressing the relevant parties
REFER requests contain a number of URIs which need to address the
appropriate parties. A list of the relevant fields include the
Request-URI, To header URI, From header URI, Contact header URI,
Refer-To header URI, and the Referred-By header URI. This section
attempts to clarify what needs to be placed in each field.
In most cases, remote call control seeks to manipulate dialogs or
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sessions on a specific UA. For this reason, the Request URI of the
REFER request MUST be a valid Globally Routable Unique URI (GRUU)
[9] for a single UA (a Contact URI). Contact URIs for a UA can be
discovered by subscribing to the Registration Package [22] for the
relevant AORs.
For remote call control requests to operate as expected, the
Refer-Issuer needs to be confident that the Refer-Receiver supports
the extensions and conventions described here. Otherwise, the
triggered request might have completely different semantics from the
request which was indicated in the Refer-To header. (Most
implementations ignore unknown URI and header parameters). For
example a REFER intended to cause the Refer-Receiver to send a 486
Busy Here response for an existing dialog, might instead trigger a
new INVITE to the sender of the original INVITE. Implementations
which send remote call control requests MUST include the
refer-response option-tag in a Require header field value in each
REFER request. (Note that support for this option-tag also implies
support for the response URI parameter in a Refer-To header.)
The To header field in the REFER request should contain the same URI
as in the Request-URI, and the From identifies the AOR of the
controller. The Refer-To is set to whatever URI would normally
appear in the triggered request if the request were initiated
autonomously by the Refer-Receiver. A REFER triggering a standalone
request or dialog starting request, could send to either an AOR or a
Contact address, but typically to an AOR. A REFER request triggering
a request which is in a dialog MUST always place a Contact URI in
the Refer-To header.
When set, the Referred-By [7] header field SHOULD be the same URI as
the URI in the Contact address of the REFER. If included by the
Refer-Issuer, it SHOULD be protected with a signed authenticated
identity body [8] as recommended in the Referred-By specification.
7.3 Selecting an existing dialog context for the triggered request
Many uses of remote call control require that the Refer-Receiver
generate a new request or response in the context of an existing
dialog. For example, the controller might want the Refer-Receiver to
send a BYE, CANCEL, or response to an INVITE in the context of a
dialog created with INVITE. For subscriptions, the controller might
want the Refer-Receiver to unsubscribe (send a SUBSCRIBE with an
Expires header field of 0).
To select the appropriate dialog from which to source the request,
this document proposes a few new (header) parameters to the Refer-To
header (the call-id, remote-tag, and local-tag parameters). Explicit
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header parameters were selected because they can apply to non SIP
URIs. For example, the following URI, loads a "How To" website in
the context of an existing dialog (presumably one created with an
INVITE). When the associated dialog completes, the content may be
hidden or dismissed with the context with which it was associated
Refer-To: <http://support.example.com/howto.html>
;call-id=xyz;remote-tag=123;local-tag=456
When describing the context of a subscription, the event and
event-id parameters are also used. These correspond to the event type
and the event-id parameter in the Event header (if present).
Explicit matching of target dialogs and subscriptions was
intentonally selected instead of including the appropriate values in
embedded Call-ID, To, From, and Event headers. Among other benefits,
this reduces the length of the URI portion of the Refer-To header and
simplifies URI encoding requirements dramatically.
OPEN ISSUE: These parameter extensions should be incorporated in
the REFER extensions draft.
8. Authorizing remote call control requests
User Agents MUST authorize all remote call control requests.
Requests from user agents which can authenticate themselves (using
Digest authentication, mutual TLS authentication, or S/MIME) as
representing the AOR on the target UA SHOULD be authorized unless
local policy directs otherwise. In addition, some user agents may
need introduction using one-time credentials which have additional
authorization restrcitions. For example, an electronic whiteboard in
a conference room could authorize participants only if they had
scheduled a meeting in the corresponding conference room for the
current time. [More explanation needed.]
9. Security Considerations
The functionality described in this document allows an authorized
party to manipulate SIP sessions and dialogs in arbitrary ways.
Implementations need to take reasonable precautions to insure
authenticity of remote call control request, which MUST be sent using
either hop-by-hop TLS [11] via a SIPS URI, or individually signed
using SMIME [10]. Signing remote call control requests with SMIME is
RECOMMENDED. In addition, UAs which support remote call control
SHOULD sign Referred-By headers in remote call control requests in an
appropriate authenticated identity body. UAs which support remote
call control MUST implement SIPS, SHOULD implement SMIME signing and
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verification, and SHOULD implement separate signing of Referred-By
headers in an appropriate authenticated identity body.
10. IANA Considerations
No action by IANA is required.
11. Acknowledgments
Many thanks to Sean Olson, Robert Sparks, and Alan Johnston.
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] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[5] Rosenberg, J. and H. Schulzrinne, "An INVITE Inititiated Dialog
Event Package for the Session Initiation Protocol (SIP",
draft-ietf-sipping-dialog-package-01 (work in progress), March
2003.
[6] Olson, S., "Extensions to the REFER mechanism for Third Party
Call Control", draft-olson-sipping-refer-extensions-00 (work in
progress), June 2003.
[7] Sparks, R., "The SIP Referred-By Mechanism",
draft-ietf-sip-referredby-01 (work in progress), February 2003.
[8] Peterson, J., "SIP Authenticated Identity Body (AIB) Format",
draft-ietf-sip-authid-body-01 (work in progress), March 2003.
[9] Rosenberg, J., "Obtaining and Using Globally Routable User
Agent (UA) URIs (GRUU) in the Session Initiation Protocol
(SIP)", draft-ietf-sip-gruu-00 (work in progress), January
2004.
[10] Ramsdell, B., "S/MIME Version 3 Message Specification", RFC
2633, June 1999.
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[11] Dierks, T., Allen, C., Treese, W., Karlton, P., Freier, A. and
P. Kocher, "The TLS Protocol Version 1.0", RFC 2246, January
1999.
[12] Sparks, R., "Internet Media Type message/sipfrag", RFC 3420,
November 2002.
Informational References
[13] Mahy, R., "A Call Control and Multi-party usage framework for
the Session Initiation Protocol (SIP)",
draft-ietf-sipping-cc-framework-02 (work in progress), March
2003.
[14] Sparks, R. and A. Johnston, "Session Initiation Protocol Call
Control - Transfer", draft-ietf-sipping-cc-transfer-01 (work in
progress), February 2003.
[15] Rosenberg, J., Schulzrinne, H., Camarillo, G. and J. Peterson,
"Best Current Practices for Third Party Call Control in the
Session Initiation Protocol", draft-ietf-sipping-3pcc-03 (work
in progress), March 2003.
[16] Rosenberg, J., "A Framework for Conferencing with the Session
Initiation Protocol",
draft-ietf-sipping-conferencing-framework-00 (work in
progress), May 2003.
[17] Rosenberg, J., "A Presence Event Package for the Session
Initiation Protocol (SIP)", draft-ietf-simple-presence-10 (work
in progress), January 2003.
[18] Rosenberg, J., Schulzrinne, H. and P. Kyzivat, "Caller
Preferences and Callee Capabilities for the Session Initiation
Protocol (SIP)", draft-ietf-sip-callerprefs-08 (work in
progress), March 2003.
[19] Andreasen, F. and B. Foster, "Media Gateway Control Protocol
(MGCP) Version 1.0", RFC 3435, January 2003.
[20] Groves, C., Pantaleo, M., Anderson, T. and T. Taylor, "Gateway
Control Protocol Version 1", RFC 3525, June 2003.
[21] Moats, R., "URN Syntax", RFC 2141, May 1997.
[22] Rosenberg, J., "A Session Initiation Protocol (SIP) Event
Package for Registrations", draft-ietf-sipping-reg-event-00
(work in progress), October 2002.
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Authors' Addresses
Rohan Mahy
Cisco Systems, Inc.
5617 Scotts Valley Drive, Suite 200
Scotts Valley, CA 95066
USA
EMail: rohan@cisco.com
Orit Levin
Microsoft Corporation
One Microsoft Way
Redmond, WA 98052
USA
EMail: oritl@microsoft.com
Francois Audet
Nortel Networks
4655 Great America Parkway
Santa Clara, CA 95054
USA
EMail: audet@nortelnetworks.com
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