SIP WG R. Mahy
Internet-Draft Cisco Systems, Inc.
Expires: March 31, 2004 O. Levin
Microsoft Corporation
Oct 2003
Remote Call Control in SIP using the REFER method and the
session-oriented dialog package
draft-mahy-sip-remote-cc-00.txt
Status of this Memo
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
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This Internet-Draft will expire on March 31, 2004.
Copyright Notice
Copyright (C) The Internet Society (2003). 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. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Examples of Remote Call Control Operations . . . . . . . . . . 4
4. User Agent Behavior . . . . . . . . . . . . . . . . . . . . . 8
4.1 Organizing requests within dialogs . . . . . . . . . . . . . . 8
4.2 Addressing the relevant parties . . . . . . . . . . . . . . . 10
4.3 Selecting an existing dialog context for the triggered
request . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
4.4 Accessing Local Services Remotely . . . . . . . . . . . . . . 11
4.5 Authorizing remote call control requests . . . . . . . . . . . 12
5. More complex examples . . . . . . . . . . . . . . . . . . . . 13
6. Handling DTMF . . . . . . . . . . . . . . . . . . . . . . . . 15
7. Formal Syntax . . . . . . . . . . . . . . . . . . . . . . . . 15
8. Security Considerations . . . . . . . . . . . . . . . . . . . 16
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 16
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 16
Normative References . . . . . . . . . . . . . . . . . . . . . 16
Informational References . . . . . . . . . . . . . . . . . . . 17
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 18
Intellectual Property and Copyright Statements . . . . . . . . 19
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1. Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC-2119 [2].
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 [11] 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 [13] 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 [12], 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.)
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
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
[16] or Megaco [17]. The authors believe that remote call control
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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.
3. Examples of Remote Call Control Operations
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.
In the requests labeled with 1 and 2, Alice's PC or PDA sets up a
subscription to the dialog package from Alice's phone (messages shown
in a later section). All of the subsequent NOTIFY messages are
notifications about changes in the dialog state at Alice's phone. In
message 3, Alice's PC or PDA asks her phone to "call Bob" (message
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4), which eventually results in an early dialog (5) with one of Bob's
Contacts. [Note Well: Parts of the flow marked in parentheses
(including messages 6, 7, 9, and 10) show alternative outcomes in the
call flow.]
Alice's Alice's Bob Cathy
PC or PDA Phone
| Call-ID: 123 | Call-ID: 456 | Call-ID: 789 |
| | | |
|---SUBSCRIBE/200--->| 1 | |
|<--NOTIFY/200-------| 2 | |
| | | |
| | | |
|---REFER/202------->| 3 | |
|<--NOTIFY/200-------|---INVITE--------->| 4 |
| |<-----180----------| 5 |
|<--NOTIFY/200-------| | |
| | | |
| | | |
( |---REFER/202------->| 6 | ) |
( | |---CANCEL/200----->| ) |
( |<--NOTIFY/200-------|<-----487/ACK------| ) |
| | | |
| | | |
| |<-----200/ACK------| |
|<--NOTIFY/200-------| | |
| | | |
( |---REFER/202------->| 7 | ) |
( | |---BYE/200-------->| ) |
( |<--NOTIFY/200-------| | ) |
| | | |
| | | |
| |<----------------------INVITE/180-----| 8
|<--NOTIFY/200-------| | |
| | | |
| | | |
( |---REFER/202------->| 9 | | )
( |<--NOTIFY/200-------|--------------------------302/ACK---->| )
| | | |
| | | |
( |---REFER/202------->| 10 | | )
( |<--NOTIFY/200-------|--------------------------486/ACK---->| )
| | | |
| | | |
|---REFER/202------->| 11 | |
|<--NOTIFY/200-------|--------------------------200/ACK---->|
| | | |
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| | | |
Messages 3, 4, and 5 follow. The norefersub option-tag on each REFER
suppresses the implicit subscription which would normally follow the
REFER (the notifications in the call flow diagram are for the dialog
package subscription in messages 1 and 2).
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. Finally, some of the URIs shown here are not escaped
properly to aid in readability. In message 9 the @ in the Refer-To
URI should be escaped.
Message 3:
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: remotecc
Supported: norefersub
Refer-To: "Bob" <sip:bob@example.net>
Message 4:
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 5:
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>
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Content-Type: application/sdp
Content-Length: xxx
The rest of the REFER messages in this example are identical to the
REFER in Message 3 except for the Refer-To header , CSeq header. and
Via branch id (not shown). Message fragment 6 and 7 show the Refer-To
headers which Alice's PC or PDA would send to cause Alice's phone to
terminate the session which Message 4 attempted to originate. The
extra parameters in the Refer-To header are used to explicitly match
a specific dialog. They are more fully described in a later section.
Header for Message 6:
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5;method=CANCEL>
;call-id=456;remote-tag=;local-tag=xyz
Header for Message 7:
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5;method=BYE>
;call-id=456;remote-tag=uvw;local-tag=xyz
Message 8 is an invitation received by Alice's phone from Cathy.
Refer-To headers for Messages 9, 10, and 11 are shown below which
would cause Alice's phone to redirect, reject, or accept the
invitation. They use the response URI parameter defined in [6]. Note
that some specialized User Agents might not be capable of accepting
an invitation autonomously. For example, a SIP user agent which
connect to an analog telephone cannot physically force the phone to
go offhook.
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Message 8:
INVITE sip:reg2@10.1.1.3 SIP/2.0
To: "Alice" <sip:alice@example.com>
From: "Cathy" <sip:cathy@example.net>;tag=ijk
Call-ID: 789
CSeq: 1 INVITE
Contact: "Cathy's Contact" <sip:cathy-pc.example.net>
Content-Type: application/sdp
Content-Length: xxx
Header for 9:
Refer-To: <sip:cathy-pc.example.net;method=INVITE;response=302 \
?Contact=sip:doug@example.com>
;call-id=789;remote-tag=ijk;local-tag=lmn
Header for 10:
Refer-To: <sip:cathy-pc.example.net;method=INVITE;response=486>
;call-id=789;remote-tag=ijk;local-tag=lmn
Header for 11:
Refer-To: <sip:cathy-pc.example.net;method=INVITE;response=200>
;call-id=789;remote-tag=ijk;local-tag=lmn
4. User Agent Behavior
4.1 Organizing requests within dialogs
REFER messages used for call transfer usually 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 can be addressed by including 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:
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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 should
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
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 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 intermittent 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].
OPEN ISSUE: Should we restrict usage to one of these three models?
OPEN ISSUE: Are there other models possible?
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
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4.2 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
defines the semantics of each field.
In most cases, remote call control seeks to manipulate dialogs or
sessions on a specific UA. For this reason, the Request URI of the
REFER request SHOULD be a valid GRUU for a singel UA (a Contact URI).
Contact URIs for a UA can be discovered by subscribing to the
registration package for the relevant AORs.
In the rare exceptions when the controller does not care which
specifc UA it manipulates, an AOR MAY be used instead. When an AOR is
used, the REFER request can include appropriate caller-preferences to
encourage selection of an appropriate Contact. The norefersub
option-tag MUST NOT be used when the REFER Request-URI is an AOR, as
the REFER Request could fork and cause incorrect behavior. While,
the controller can discourage a proxy from forking remote call
control request by using the Request-Disposition: no-fork header
field, insuring that no proxy forks requires the use of the
callerpref option-tag in a Proxy-Require header field value. Because
any proxy in the chain of this request which did not support caller
preferences would cause the request to fail, use of Proxy-Require is
NOT RECOMMENDED.
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 remotecc
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 be
inserted by a user of the Refer-Receiver (if operated autonomously).
A REFER triggering a standalone request or dialog starting request,
could send to either an AOR or a Contact address, but typically to an
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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.
4.3 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, which was not necessarily initiated by the controller. 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
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.
4.4 Accessing Local Services Remotely
It may be desirable to have a URI convention for contacts on some UAs
which gives autoanswer behavior. This allows for the development of
several services if properly authorized (for example, an intercom
service). In the context of remote call control, this URI could be
placed in the Request-URI of a REFER requesting a new dialog (as in
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Message 3 in the examples) instead of Alice's regular Contact address
(at issue is if non-autoanswer behavior if desirable on devices which
are capable of autoanswer). There are several possible syntactic
choices for an autoanswer URI based on Alice's registration. (We
cannot restrict ourselves to only one autoanswer URI on Alice's
phone, since multiple registrations may exhibit different
authorization, alerting, and other behavior.) Three of these choices
are listed below.
(option1) sip:reg2+autoanswer@10.1.1.3
(option2) sip:reg2;autoanswer@10.1.1.3
(option3) sip:reg2@10.1.1.3;autoanswer
Hold is a very common local service on phones. Unfortunately hold has
two semantics. Hold is often used to describe a primitive operation
of setting all media lines in a session to inactive. (We will call
this Simple Hold). Hold also describes a server running on a phone
which can cause different behavior, for example, music on hold, tone
on hold, or simple hold. It is desirable to be able to access the
"Hold" service on Alice's phone via a URI. In the absense of any
other convention, we will use the following URI:
sip:service.hold@10.1.1.3 . Using this "service URI" on the phone,
Alice's PC or PDA can ask the phone to place a specific session on or
off hold as shown below.
REFER sip:reg2@10.1.1.3 SIP/2.0
Refer-To: <sip:service.hold@10.1.1.3>
;call-id=789;remote-tag=ijk;local-tag=lmn
REFER sip:reg2@10.1.1.3 SIP/2.0
Refer-To: <sip:service.unhold@10.1.1.3>
;call-id=789;remote-tag=ijk;local-tag=lmn
The SIP conferencing framework [14] and the cc-conferencing describe
the Conference Factory as a service which assigns new conference
URIs. The conference factory is itself accessible via a URI. In some
cases, this conference factory is colocated with a phone or other
single-user UA. In the absense of any other convention, we use the
following URI to reference the conference factory service on Alice's
phone: sip:service.conf-factory@10.1.1.3 .
OPEN ISSUE: How are service URIs registered?
OPEN ISSUE: Is this a complete set of services that are needed for
remote call control?
4.5 Authorizing remote call control requests
To be written.
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5. More complex examples
The following example shows how a controller can cause the
Refer-Recevier to complete a transfer between two existing calls.
Alice's Alice's Bob Cathy
PC or PDA Phone
| | Call-ID: 456 | Call-ID: 789 |
| |<=================>| |
| |<====================================>|
| | | |
| | | |
| | | |
A |---REFER/202------->| | |
| B |---REFER/202------>| |
| C |<--NOTIFY/200------|---INVITE-------->|
| | | w/Replaces |
| | |<--200/ACK--------|
| D |<--NOTIFY/200------| |
| |<--BYE/200----------------------------|
E |<--NOTIFY/200-------| | |
| |---BYE/200-------->| |
F |<--NOTIFY/200-------| |<================>|
| | | |
| | | |
Message A: (CHECK TAGS!) (character escaping)
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5;method=REFER \
?Refer-To=<sip::cathy-pc.example.net?Replaces=789;to-tag=ijk;from-tag=lmn>>
;call-id=456;remote-tag=uvw;local-tag=xyz
Messages C and D are notifications to the refer package.
Messages E and F are notifications to the dialog package.
The following example shows how a controller can cause the
Refer-Receiver to join two existing sessions using a SIP conference
server (labeled "Focus" in the call flow diagram). Note that < > ;
and = characters in these URIs need to be escaped
Alice's Focus Alice's Bob Cathy
PC or PDA Phone
| | | Call-ID: 456 | Call-ID: 789 |
| | |<============>| |
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| | |<===========================>|
| | | | |
| | | | |
G |---REFER/202------------------>| | |
| |<--INVITE-----| | |
|<--NOTIFY/200--(100)-----------| | |
H | |---200/ACK--->| | |
|<--NOTIFY/200----(200)---------| | |
| |<============>| | |
| | | | |
I |---REFER/202--->| | | |
| |---INVITE-w/ Replaces---------------------->|
| |<--200/ACK----------------------------------|
| | |<---BYE/200------------------|
|<--NOTIFY/200------------------| | |
| |<==========================================>|
| | | | |
J |---REFER/202--->| | | |
| |---INVITE-w/ Replaces------->| |
| |<--200/ACK-------------------| |
| | |<---BYE/200---| |
|<--NOTIFY/200------------------| | |
| |<===========================>| |
| | | | |
| | | | |
Message G:
Refer-To: <sip:service.conf-factory@10.1.1.3>
;call-id=456;remote-tag=uvw;local-tag=xyz
Message H:
SIP/2.0 200 OK
To: "Conference Factory" <sip:service.conf-factory@10.1.1.3>;tag=ccc
From: "Alice" <sip:alice@example.com>;tag=bbb
Call-ID: aaa
CSeq 1 INVITE
Contact: "Conference #3" <sip:conf3@10.1.1.3>;isFocus
Content-Type: application/sdp
Content-Length: xxx
Message I:
REFER sip:conf3@10.1.1.3 SIP/2.0
Refer-To: <sip:cathy-pc.example.net \
?Replaces=789;to-tag=ijk;from-tag=lmn>
Referred-By: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;cid="333@444"
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Message J:
REFER sip:conf3@10.1.1.3 SIP/2.0
Refer-To: "Bob's Contact" <sip:line1@192.168.0.5 \
?Replaces=456;to-tag=uvw;from-tag=xyz>
Referred-By: "Alice's PC or PDA" <sip:alice1@10.1.1.2>;cid="111@222"
6. Handling DTMF
Occasionally it is useful for one UA to collect digits on behalf of a
User Agent which can actually send them using RFC2833 [19]. One of
the options is that the collecting UA send KPML [20] responses to the
UA capable of turning these keypad events into DTMF media. The method
used for sending markup responses is under discussion currently, but
one proposal is a new method called FEEDBACK as part of [21]. For
example, it is possible that the KPML response tag include new
parameters as shown below to identify the dialog for which the keypad
markup is intended.
FEEDBACK 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: 27 FEEDBACK
Content-Type: application/kpml+xml
Content-Length: xxx
<?xml version="1.0">
<kpml version="1.0">
<response digits="9999#" call-id="456"
remote-tag="uvw" local-tag="xyz"/>
</kpml>
OPEN ISSUE: Need further investigation of mechanism to carry DTMF.
Can this be generalized?
7. Formal Syntax
The following syntax specification extends the Refer-To header
described in RFC3515 using the augmented Backus-Naur Form (BNF) as
described in RFC-2234 [3].
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Refer-To = ("Refer-To" / "r") HCOLON ( name-addr / addr-spec ) *
(SEMI referto-params)
referto-params = callid-param / rtag-param / ltag-param /
event-param / eventid-param / generic-param
callid-param = "call-id" EQUAL ( token / LDQOT callid RDQUOT )
rtag=param = "remote-tag" EQUAL token
ltag=param = "local-tag" EQUAL token
event-param = "event" EQUAL event-type
eventid-param = "event-id" EQUAL token
8. 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 [10] via a SIPS URI, or individually signed
usingSMIME [9]. 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
verification, and SHOULD implement separate signing of Referred-By
headers in an appropriate authenticated identity body.
9. IANA Considerations
Need to register the remotecc option-tag, the Refer-To header
parameters, and the SIP URI parameters
10. 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.
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[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-02 (work in progress), July
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-03 (work in progress), August 2003.
[8] Peterson, J., "SIP Authenticated Identity Body (AIB) Format",
draft-ietf-sip-authid-body-02 (work in progress), July 2003.
[9] Ramsdell, B., "S/MIME Version 3 Message Specification", RFC
2633, June 1999.
[10] Dierks, T., Allen, C., Treese, W., Karlton, P., Freier, A. and
P. Kocher, "The TLS Protocol Version 1.0", RFC 2246, January
1999.
Informational References
[11] 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.
[12] Sparks, R. and A. Johnston, "Session Initiation Protocol Call
Control - Transfer", draft-ietf-sipping-cc-transfer-01 (work in
progress), February 2003.
[13] Rosenberg, J., Peterson, J., Schulzrinne, H. and G. Camarillo,
"Best Current Practices for Third Party Call Control in the
Session Initiation Protocol", draft-ietf-sipping-3pcc-04 (work
in progress), July 2003.
[14] Rosenberg, J., "A Framework for Conferencing with the Session
Initiation Protocol",
draft-ietf-sipping-conferencing-framework-00 (work in
progress), May 2003.
[15] Rosenberg, J., Schulzrinne, H. and P. Kyzivat, "Caller
Preferences for the Session Initiation Protocol (SIP)",
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draft-ietf-sip-callerprefs-09 (work in progress), July 2003.
[16] Andreasen, F. and B. Foster, "Media Gateway Control Protocol
(MGCP) Version 1.0", RFC 3435, January 2003.
[17] Groves, C., Pantaleo, M., Anderson, T. and T. Taylor, "Gateway
Control Protocol Version 1", RFC 3525, June 2003.
[18] Burger, E., "Basic Network Media Services with SIP",
draft-burger-sipping-netann-07 (work in progress), September
2003.
[19] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals", RFC 2833, May 2000.
[20] Burger, E., "Keypad Stimulus Protocol (KPML)",
draft-ietf-sipping-kpml-00 (work in progress), September 2003.
[21] Jennings, C., "SIP Support for Application Initiation",
draft-jennings-sip-app-info-01 (work in progress), July 2003.
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
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