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Internet Engineering Task Force Robert Sparks
Internet Draft dynamicsoft
draft-ietf-sip-cc-transfer-02.txt
November 2000
Expires May 2001
SIP Call Control
Transfer
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with all
provisions of Section 10 of RFC2026 [1].
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-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts 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
Abstract
This document defines a SIP extension within the Call Control
Framework to provide Call Transfer capabilities.
Robert Sparks [Page 1]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
1 Overview...........................................................3
2 Changes from draft-sparks-sip-cc-transfer-01.......................4
3 The REFER Method...................................................5
3.1 The Refer-To Header............................................5
3.1.1 Examples.....................................................5
3.1.2 A PGP based signature-scheme.................................6
3.1.3 Examples.....................................................7
3.2 Header Field Support for the REFER Method......................7
3.3 Message Body Inclusion.........................................8
3.4 Responses to the REFER Method..................................8
3.5 Behavior of SIP User Agents....................................8
3.6 Behavior of SIP Registrars/Redirect Servers....................9
3.7 Behavior of SIP Proxies........................................9
3.8 Security Considerations........................................9
4 Call Transfer.....................................................10
4.1 Actors and Roles..............................................10
4.2 Requirements..................................................11
4.3 Using REFER to achieve Call Transfer..........................12
4.4 Unattended Transfer...........................................12
4.4.1 Successful Unattended Transfer..............................13
4.4.2 Failed Unattended Transfer..................................14
4.5 Unattended Transfer with Consultation Hold....................14
4.5.1 Variation 1 : Exposes transfer target.......................15
4.5.2 Variation 2 : Protects transfer target......................15
4.5.3 Consultation Hold in the presence of forking proxies........16
4.6 Attended Transfer.............................................17
4.7 Transfer with multiple parties................................17
5 Editor's Address..................................................18
6 Acknowledgments...................................................18
7 References........................................................18
Robert Sparks [Page 2]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
1 Overview
This document defines a SIP [2] extension and details its use to
provide Call Transfer capabilities. This is part of a family of Call
Control extensions described in the Call Control Framework document
[3].
The mechanisms discussed here are most closely related to traditional
unattended and consultation hold transfers. Discussion of attended
transfer (where all parties are briefly in a conference) is deferred
until the conferencing features in this framework are addressed.
This work has roots in draft-ietf-sip-cc-01 [4] but some basic
semantics are different. In particular, transfers are achieved through
a new method that does not terminate the original signaling
relationship. By disassociating transfers from the processing of BYE,
these changes facilitate recovery of failed transfers and clarify
state management in the participating entities.
Implementers that started with the sip-cc-01 BYE-ALSO technique for
blind-transfer should find it straightforward to migrate to the
mechanisms set forth here.
Robert Sparks [Page 3]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
2 Changes from draft-sparks-sip-cc-transfer-01
. Allowed the ref= parameter in the Referred-By header to occur
either before or after the optional signature.
. Allowed name-addr|addr-spec for the referrer-url in a Referred-By
header
. Quoted the referenced-url with <> and required escaping <> if
they occur within the referenced-url
. Captured the results of the list discussion on having the REFERed
invite reach a particular endpoint in the presence of forking
proxies.
. Added example Refer-To: and Referred-By: headers.
Robert Sparks [Page 4]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
3 The REFER Method
REFER is a SIP method as defined by [2]. The REFER method indicates
that the recipient should contact a third party using the contact
information provided in the method. A success response indicates that
the recipient was able to contact the third party. The REFER method
follows the session's current signaling path. In particular, the
Request-URI of the REFER method identifies the recipient.
Unless stated otherwise, the protocol for emitting and responding to a
REFER request are identical to those for a BYE request in [2]. The
behavior of SIP entities not implementing the REFER (or any other
unknown) method is explicitly defined in [2] and is not discussed
further here.
3.1 The Refer-To Header
Refer-To is a request-header as defined by [2]. It may only appear in
a REFER request.
Refer-To = ("Refer-To" | "r") ":" URL
A REFER method MUST contain exactly one Refer-To header.
The Refer-To header MAY be encrypted as part of end-end encryption.
The Contact header is an important part of the
Route/Record-Route mechanism and is not available
for this task.
3.1.1 Examples
Refer-To: sip:alice@atlanta.com
Refer-To: sip:bob@biloxi.com?Accept-Contact=sip:bobsdesk.biloxi.com?Ca
ll-ID=55432@alicepc.atlanta.com
Refer-To: sip:carol@cleveland.com;method=SUBSCRIBE
Refer-To: http://www.ietf.org
Robert Sparks [Page 5]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
The Referred-By Header
Referred-By is a request-header as defined by [2]. It can appear in
any request. It conveys the identity of the original REFERrer to the
referred-to party, optionally proving the identity and that the
REFERrer actually issued this reference.
Referred-By = ("Referred-By" | "b") ":" referrer-url ";"
( referenced-url
| ( referenced-url ";" ref-signature )
| ( ref-signature ";" referenced-url )
)
referrer-url = ( name-addr | addr-spec )
referenced-url = "ref" "=" "<" URL ">"
ref-signature = signature-scheme *( ";" sig-scheme-params )
signature-scheme = "scheme" "=" token
sig-scheme-parms = token "=" ( token | quoted-string )
The referrer-url contains the SIP URL of the party sending the REFER
request. The referenced-url contains a copy of the URL placed in the
Refer-To: header. Any occurrences of < or > in the referenced-url MUST
be escaped. The ref-signature contains a signature over the
concatenation of referrer-url and referenced-url. An example
signature scheme is given in section 3.1.2.
A REFER request MUST contain exactly one Referred-By header.
The Referred-By header SHOULD be signed to help detection of REFERs
from unauthorized third parties. A signed Referred-By header SHOULD
include a Date header in the referrer-url to facilitate detection of
replay attacks.
A UA MAY reject a request containing an unsigned Referred-By header. A
UA SHOULD verify the signature on any Referred-By header it receives.
The Referred-By header MAY be encrypted as part of end-end encryption.
3.1.2 A PGP based signature-scheme
One signature-scheme for Referred-By headers uses PGP as follows:
signature-scheme = "scheme" "=" "pgp"
sig-scheme-parms = pgp-version | signed-by | pgp-signature
pgp-version, signed-by and pgp-signature are defined in section 15.1
of RFC2543, with the modification that the signature is computed
across the concatenation of the referrer-url and the referenced-url.
Robert Sparks [Page 6]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
3.1.3 Examples
Referred-By: sip:alice@atlanta.com;ref=<http:www.ietf.org>
Referred-By: "Bob" <sip:bob@biloxi.com>;ref=<sip:alice@atlanta.com>;
scheme=pgp;pgp-version="5.0";signature="the signature"
(Note that in the last example, the signature would be over the
string "sip:bob@biloxi.comsip:alice@atlanta.com")
3.2 Header Field Support for the REFER Method
This table adds a column to tables 4 and 5 in [2], describing header
presence in a REFER method. See [2] for a key for the symbols used. A
row for the Refer-To: and Referred-By request-header should be
inferred, each mandatory for REFER. Refer-To is not applicable for all
other methods. Referred-By is a general Request header. The enc and e-
e columns in [2] apply to the REFER method unmodified.
Header Where REFER
Accept R -
Accept-Encoding R -
Accept-Language R o
Allow R -
Allow 405 m
Authorization R o
Call-ID gc m
Contact R o
Contact 1xx -
Contact 2-6xx o
Content-Encoding e -
Content-Length e o
Content-Type e -
CSeq gc m
Date g o
Encryption g o
Expires R o
From gc m
Hide R o
Max-Forwards R o
Organization g o
Priority R -
Proxy-Authenticate 407 o
Proxy-Authorization R o
Robert Sparks [Page 7]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
Proxy-Require R o
Require R o
Retry-After R -
Retry-After 404,480,486 o
Retry-After 503 o
Retry-After 600,603 o
Response-Key R o
Record-Route R o
Record-Route 2xx o
Route R o
Server r o
Subject R -
Timestamp g o
To gc(1) m
Unsupported 420 o
User-Agent g o
Via gc(2) m
Warning r o
WWW-Authenticate 401 o
3.3 Message Body Inclusion
A REFER method may contain a body which SHOULD be processed according
to its Content-Type.
3.4 Responses to the REFER Method
An agent responding to a REFER Method MUST return a 400 Bad Request if
the request contained zero or more than one Refer-To headers. An agent
responding to a REFER Method MUST return a 400 Bad Request if the
request contained zero or more than one Referred-By headers. An agent
(including proxies generating local responses) MAY return a 100 Trying
or any appropriate 400-600 class response as prescribed by [2]. If the
recipient's agent decides to contact the resource in the Refer-To
header, a 200 OK response MUST be returned if it the contact was
successful, otherwise a 503 Service Unavailable MUST be returned. The
503 response MAY contain a Retry-After: header indicating when the
REFER may be attempted again.
3.5 Behavior of SIP User Agents
A UA receiving a well-formed REFER request SHOULD request approval
from the user to proceed (this request could be interactive or through
configuration). Upon receiving approval from the user, the UA MUST
contact the resource identified by the URL in the Refer-To: header.
Robert Sparks [Page 8]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
Note that if the URL is a SIP URL, it could contain header fields such
as Call-Id that will be used to form the resulting request. If the URL
is a SIP URL, the Referred-By header in the REFER request should be
copied into the request sent to the referred-to resource. In
accordance with [2], the UA SHOULD issue a provisional response to the
REFER method if it cannot issue a final response within 200ms of its
receipt. The appropriate response to issue to the REFER on receipt of
a final response from the referred-to resource is discussed in
"Responses to the REFER Method".
3.6 Behavior of SIP Registrars/Redirect Servers
Registrars and Redirect Servers SHOULD return a 603 to a REFER
request, unless they are also playing some other SIP role.
3.7 Behavior of SIP Proxies
SIP Proxies do not require modification to support the REFER method.
Specifically, as required by [2], a proxy should process a REFER
request the same way it processes an OPTIONS request.
3.8 Security Considerations
The security requirements of [2] apply to the REFER method.
This mechanism relies on providing contact information for the
referred-to resource to the party being referred. Care should be taken
to provide a suitably restricted URI if the referred to resource
should be protected.
Care should be taken when implementing the logic that determines
whether or not to accept the REFER request. A UA not capable of
accessing non-SIP URLs SHOULD NOT accept REFER requests to them.
Robert Sparks [Page 9]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4 Call Transfer
4.1 Actors and Roles
There are three actors in a given transfer event, each playing one of
the following roles:
Transferee - the party being transferred to the Transfer
Target.
Transferor - the party initiating the transfer
Transfer Target - the new party being introduced into a call with
the Transferee.
The following roles are used to describe transfer requirements and
scenarios:
Originator - wishes to place a call to the Recipient. This actor
is the source of the first INVITE in a session, to
either a Facilitator or a Screener.
Facilitator - receives a call or out-of-band request from the
Originator, establishes a call to the Recipient
through the Screener, and connects the Originator to
the Recipient.
Screener - receives a call ultimately intended for the Recipient
and transfers the calling party to the Recipient if
appropriate.
Recipient - the party the Originator is ultimately connected to.
Robert Sparks [Page 10]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.2 Requirements
1. Any party in a SIP session MUST be able to transfer any other
party in that session at any point in that session.
2. The Transferor and the Transferee MUST NOT be removed from a
session as part of a transfer transaction.
At first glance, requirement 2 may seem to indicate
that the user experience in a transfer must be
significantly different from what a current PBX or
Centrex user expects. As the call-flows in this
document show, this is not the case. A client MAY
preserve the current experience. In fact, without
this requirement, some forms of the current
experience (ringback on unattended transfer failure
for instance) will be lost.
3. The Transferor MUST know whether or not the transfer was
successful (this is significantly different from the requirements
of draft-ietf-sip-cc-01).
Robert Sparks [Page 11]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.3 Using REFER to achieve Call Transfer
A REFER can be issued by the Transferor to cause the Transferee to
issue an INVITE to the Transfer-Target. Note that a successful REFER
transaction does not terminate the session between the Transferor and
the Transferee. If those parties wish to terminate their session, they
must do so with a subsequent BYE request. The media negotiated between
the transferee and the transfer target is not affected by the media
that had been negotiated between the transferor and the transferee. In
particular, the INVITE issued by the Transferee will have the same SDP
body it would have if he Transferee had initiated that INVITE on its
own. Further, the disposition of the media streams between the
Transferor and the Transferee is not altered by the REFER method.
Agents may alter a session's media through additional signaling. For
example, they may make use of the SIP hold re-INVITE [2] or the
conferencing extensions provided by this framework.
4.4 Unattended Transfer
Unattended Transfer consists of the Transferor providing the Transfer
Target's contact to the Transferee. The Transferee attempts to
establish a session using that contact and reports the results of that
attempt to the Transferor. The signaling relationship between the
Transferor and Transferee is not terminated, so the call is
recoverable if the Transfer Target cannot be reached. Note that the
Transfer Target's contact information has been exposed to the
Transferee. The provided contact can be used to make new calls in the
future.
The diagrams below show indicate the first line of each message. All
messages in a particular diagram share the same Call-ID. In these
diagrams, media is managed through reINVITE holds, but other
mechanisms (mixing multiple media streams at the UA or using the
conferencing extensions for example) are valid.
Robert Sparks [Page 12]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.4.1 Successful Unattended Transfer
Transferor Transferee Transfer
| | Target
| INVITE | |
|<-------------------| |
| 200 OK | |
|------------------->| |
| ACK | |
|<-------------------| |
| INVITE (hold) | |
|------------------->| |
| 200 OK | |
|<-------------------| |
| ACK | |
|------------------->| |
| REFER | |
|------------------->| |
| 100 Trying | |
|<-------------------| |
| | INVITE |
| |------------------->|
| | 200 OK |
| |<-------------------|
| | ACK |
| |------------------->|
| 200 OK | |
|<-------------------| |
| BYE | |
|------------------->| |
| 200 OK | |
|<-------------------| |
| | BYE |
| |<-------------------|
| | 200 OK |
| |------------------->|
Robert Sparks [Page 13]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.4.2 Failed Unattended Transfer
Transferor Transferee Transfer
| | Target
| | |
| INVITE | |
|<-------------------| |
| 200 OK | |
|------------------->| |
| ACK | |
|<-------------------| |
| INVITE (hold) | |
|------------------->| |
| 200 OK | |
|<-------------------| |
| ACK | |
|------------------->| |
| REFER | |
|------------------->| |
| 100 Trying | |
|<-------------------| |
| | INVITE |
| |------------------->|
| | 486 Busy Here |
| |<-------------------|
| | ACK |
| |------------------->|
| 503 Service Unavailable |
|<-------------------| |
| INVITE (unhold) | |
|------------------->| |
| 200 OK | |
|<-------------------| |
| ACK | |
|------------------->| |
| BYE | |
|------------------->| |
| 200 OK | |
|<-------------------| |
4.5 Unattended Transfer with Consultation Hold
Transfer with Consultation Hold involves a session between the
transferor and the transfer target before the transfer actually takes
place. This is implemented with SIP Hold and Unattended Transfer as
described above.
Robert Sparks [Page 14]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.5.1 Variation 1 : Exposes transfer target
The transferor places the transferee on hold, establishes a call with
the transfer target to alert them to the impending transfer,
terminates the connection with the transfer target, then proceeds with
unattended transfer as above. This variation can be used to provide an
experience similar to that expected by current PBX and Centrex users.
To (hopefully) improve clarity, non-REFER transactions have been
collapsed into one indicator with the arrow showing the direction of
the request.
Transferor Transferee Transfer
| | Target
| | |
Call-ID:1 | INVITE/200 OK/ACK | |
|<-------------------| |
Call-ID:1 | INVITE (hold)/200 OK/ACK |
|------------------->| |
Call-ID:2 | INVITE/200 OK/ACK | |
|---------------------------------------->|
Call-ID:2 | BYE/200 OK | |
|---------------------------------------->|
Call-ID:1 | REFER | |
|------------------->| |
| 100 Trying | |
|<-------------------| |
Call-ID:1 | | INVITE/200 OK/ACK |
| |------------------->|
| 200 OK | |
|<-------------------| |
Call-ID:1 | BYE/200 OK | |
|------------------->| |
Call-ID:1 | | BYE/200 OK |
| |<-------------------|
4.5.2 Variation 2 : Protects transfer target
The transferor places the transferee on hold, establishes a call with
the transfer target and then reverses their roles, transferring the
original transfer target to the original transferee. This has the
advantage of hiding information about the original transfer target
from the original transferee. On the other hand, the Transferee's
experience is different that in current systems. The Transferee is
effectively "called back" by the Transfer Target.
Robert Sparks [Page 15]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
Transferor Transferee Transfer
| | Target
| | |
Call-ID:1 | INVITE/200 OK/ACK | |
|<-------------------| |
Call-ID:1 | INVITE (hold)/200 OK/ACK |
|------------------->| |
Call-ID:2 | INVITE/200 OK/ACK | |
|---------------------------------------->|
Call-ID:2 | INVITE (hold)/200 OK/ACK |
|---------------------------------------->|
Call-ID:2 | REFER | |
|---------------------------------------->|
| 100 Trying | |
|<----------------------------------------|
Call-ID:2 | | INVITE/200 OK/ACK |
| |<-------------------|
| 200 OK | |
|<----------------------------------------|
Call-ID:1 | BYE/200 OK | |
|------------------->| |
Call-ID:2 | BYE/200 OK | |
|---------------------------------------->|
Call-ID:2 | | BYE/200 OK |
| |------------------->|
4.5.3 Consultation Hold in the presence of forking proxies
It is worth noting that the examples given above abstract away any
proxies that might be between the three parties. In 4.5.1 for example,
the URL used to reach the Transfer Target may go through a forking
proxy. There is no guarantee that the Transferee's and Transferor's
invitations to the Transfer Target will reach the same endpoint. If
the proxy forked in parallel, both invitations could cause multiple
endpoints to ring. To increase the probability of the desired behavior
of having the referred invite reach and ring only the same endpoint as
the consultation invite, the Transferor SHOULD issue the REFER request
with the Refer-To: header containing the Contact the Transfer Target
provided in its 200 OK to the Transferor's INVITE. If that REFER
fails, the Transferor SHOULD issue another REFER with the Refer-To:
header containing the URL it used to reach the Transfer Target,
augmented with an Accept-Contact header containing the Contact the
Transfer Target provided.
Robert Sparks [Page 16]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
4.6 Attended Transfer
In an attended transfer, the three actors participate in an ad-hoc
conference as part of the event. Discussion of the implementation of
attended transfer is thus deferred until the conferencing portion of
the Call Control framework has been addressed.
4.7 Transfer with multiple parties
In this example the Originator places call to the Facilitator who
reaches the Recipient through the Screener. The Recipient's contact
information is exposed to the Facilitator and the Originator. This
example is provided for clarification of the semantics of the REFER
method only and should not be used as the design of an
implementation.
Originator Facilitator Screener Recipient
Call-ID | | | |
1 |INVITE/200 OK/ACK | |"Get Fred for me!"
|----------->| | | "Right away!"
1 |INVITE (hold)/200 OK/ACK | |
|<-----------| | |
2 | |INVITE/200 OK/ACK |"I have a call
| |----------->| |from Mary for Fred"
2 | |INVITE (hold)/200 OK/ACK "Hold please"
| |<-----------| |
3 | | |INVITE/200 OK/ACK
| | |--------->|"You have a call
| | | |from Mary"
| | | | "Put her through"
3 | | |INVITE (hold)/200 OK/ACK
| | |--------->|
2 | |REFER | |
| |<-----------| |
| |100 Trying | |
| |----------->| |
2 | |INVITE/200 OK/ACK |
| |---------------------->|"This is Fred"
| |200 OK | | "Please hold for
| |----------->| | Mary"
2 | |BYE/200 OK | |
| |<-----------| |
Robert Sparks [Page 17]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
3 | | |BYE/200 OK|
| | |--------->|
2 | |INVITE (hold)/200 OK/ACK
| |---------------------->|
1 |REFER | | |
|<-----------| | |
|100 Trying | | |
|----------->| | |
1 |INVITE/200 OK/ACK | |
|----------------------------------->| "Hey Fred"
|200 OK | | | "Hello Mary"
|----------->| | |
1 |BYE/200 OK | | |
|<-----------| | |
2 | |BYE/200 OK | |
| |---------------------->|
1 |BYE/200 OK | | |
|<-----------------------------------| "See you later"
5 Editor's Address
Robert Sparks
dynamicsoft
200 Executive Drive
Suite 120
West Orange, NJ 07052
email: rsparks@dynamicsoft.com
6 Acknowledgments
This draft is a collaborative product of the SIP working group. The
editor thanks the following for their early contributions to this
work: Ben Campbell, Chris Cunningham, Steve Donovan, Alan Johnston,
Kevin Summers and Dean Willis.
7 References
[1] S. Bradner, "The Internet Standards Process -- Revision 3",
BCP9, RFC2026, October 1996.
[2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg,
"SIP:Session Initiation Protocol", RFC 2543, March 1999.
Robert Sparks [Page 18]
Internet Draft draft-ietf-sip-cc-transfer-02.txt November 2000
[3] B. Campbell, "Framework for SIP Call Control Extensions",
Internet Draft draft-ietf-sip-cc-framework-00, Internet
Engineering Task Force, March 2000. Work in Progress.
[4] H. Schulzrinne, J. Rosenberg, "SIP Call Control Services",
Internet Draft draft-ietf-sip-cc-01, Internet Engineering Task
Force, June 17, 1999 Work in Progress (expired).
Robert Sparks [Page 19]
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