SIP WG J. Elwell
Internet-Draft Siemens plc
Expires: August 23, 2006 February 19, 2006
Connected Identity in the Session Initiation Protocol (SIP)
draft-elwell-sip-connected-identity-01.txt
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Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
Because of retargeting of a dialog-forming request, the UAS can have
a different identity from that in the To header. This document
provides a means for that UA to supply its identity to the peer UA by
means of a request in the reverse direction and for that identity to
be signed by an authentication service. The same mechanism can be
used to indicate a change of identity during a dialog, e.g., because
of some action in a PSTN behind a gateway.
This work is being discussed on the sip@ietf.org mailing list.
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Table of Contents
1. Conventions and Definitions . . . . . . . . . . . . . . . . . 3
2. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Existing mechanisms for conveying identity in the context
of a call . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Existing methods for providing authenticated identity
information . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Overview of solution . . . . . . . . . . . . . . . . . . . . . 5
6. Behaviour . . . . . . . . . . . . . . . . . . . . . . . . . . 6
6.1. Behaviour of a UA that issues an INVITE request . . . . . 6
6.2. Behaviour of a UA that receives an INVITE request . . . . 7
6.3. Behaviour of a UA during an established
INVITE-initiated dialog . . . . . . . . . . . . . . . . . 8
7. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
7.1. Sending connected identity after answering a call . . . . 8
7.2. Sending revised connected identity during a call . . . . . 11
8. IANA considerations . . . . . . . . . . . . . . . . . . . . . 16
9. Security considerations . . . . . . . . . . . . . . . . . . . 16
10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 17
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 17
11.1. Normative References . . . . . . . . . . . . . . . . . . . 17
11.2. Informative References . . . . . . . . . . . . . . . . . . 17
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 18
Intellectual Property and Copyright Statements . . . . . . . . . . 19
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1. Conventions and Definitions
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].
2. Introduction
SIP[1]initiates sessions but it also provides information on the
identities of the parties at both ends of a session. Users need this
information to help determine how to deal with communications
initiated by SIP. As a call proceeds, these identities may change.
This can happen for many reasons: calls are forwarded, calls are
parked and picked up, calls are transferred, calls are queued to be
picked up by a pool of agents, and so on. This can have impact on
the identity of the party that answers a call. It can also cause the
identity of a party to change during an established call.
This document extends the use of the From header field to allow it to
convey "connected identity" information in either direction within
the context of an existing INVITE-initiated dialog.
The provision of "response identity" for requests outside the context
of an INVITE-initiated dialog is outside the scope of this document.
3. Existing mechanisms for conveying identity in the context of a call
When establishing a call and its session, the SIP From header field
in the INVITE request provides a means for conveying the identity of
the caller from the User Agent Client (UAC) to the User Agent Server
(UAS), thereby allowing the caller's identity to be presented to the
callee. There is no corresponding mechanism specified for conveying
the identity of the callee from the UAS to the UAC, to allow the
callee's identity to be presented to the caller. The identity of the
callee is normally expected to be the identity placed in the To
header field of the INVITE request, but often this expectation is not
met because a different party answers the call, e.g., because of call
forwarding.
History information [5] gathered during the routing of a request
and returned in the response can provide additional information to
the UAC. However, this does not necessarily clearly indicate the
AoR of the UAS. Also the methods described in Section 4 for
authentication do not apply to history information, which relies
instead on hop-by-hop security and transitive trust.
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The Reply-To header field has its own meaning and cannot be relied
on in all circumstances.
The Contact header field provides a contact URI, which may not
reveal the identity (Address of Record) of the user on whose
behalf the response is sent.
Parties involved in a call can change owing to actions such as call
transfer. If such actions are achieved by issuing a new INVITE
request (with a Replaces header field) between the two UAs that are
to be involved in the re-arranged call, the SIP From header field in
the INVITE request can provide identity information in one direction,
but again there is no mechanism for conveying identity information in
the reverse direction.
However, call re-arrangements are not always carried out using a new
INVITE request. Sometimes a B2BUA performs call re-arrangements
using third party call control (3PCC) techniques. With such
techniques the UA involved in the original call and still involved in
the re-arranged call receives only a re-INVITE or UPDATE request in
the context of the original dialog between that UA and the B2BUA.
This forces the UA to re-negotiate the session with the new remote
party, but introduces a need to convey the identity of the new remote
party to the UA. Because there is no new INVITE request (outside the
context of the existing dialog), techniques applicable to new calls
do not apply.
Another case where call re-arrangements are not carried out using a
new INVITE request is where one of the UAs is a gateway to a PSTN and
a call re-arrangement such as call transfer has occurred within the
PSTN. The gateway then has a need to convey the identity of the new
party within the PSTN to the remote UA. This needs to be done within
the context of the existing dialog between the gateway and the remote
UA. In this case there is probably not even any need to re-negotiate
the session - the only requirement is to update the identity
information.
4. Existing methods for providing authenticated identity information
Because the From header field in a request is generated by the UAC
itself it can be subject to falsification. SIP has several means of
providing cryptographic authentication of a request's source
identity.
One such means for requests is HTTP-based digest authentication, as
specified in [1]. Although a UAS can require digest authentication,
it is not usually feasible between an arbitrary pair of UAs because
of reliance on a shared secret. To achieve scalability, methods
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based on public key cryptography are essential.
Another method is specified in [6], and is applicable to responses as
well as requests. This requires a UA to have a private key and
associated certificate in order to sign an Authenticated Identity
Body (AIB) in the request or response. However, this has seen little
deployment, since the public key infrastructures needed to support
private keys and certificates in every UA are not generally
available.
A third method is specified in [3]. For signature this uses a
private key and certificate associated with the domain indicated in
the From header URI. An authentication service, typically located at
the outbound proxy, authenticates the UAC by some means, using digest
authentication for example, and then inserts an Identity header and
an Identity-Info header in the forwarded request. The Identity
header contains a signature using the domain's private key and the
Identity-Info header references the corresponding certificate.
5. Overview of solution
A mid-dialog request is used to provide connected identity. The UAC
for that request inserts its identity in the From header field of the
request and the Identity header can be used to provide
authentication.
A request in the opposite direction to the INVITE request prior to or
at the time the call is answered can indicate the identity of the
alerting or answering party. A request in the same direction as the
INVITE request prior to answer can indicate a change of calling
party. A request in either direction after answer can indicate a
change of party. In all cases a dialog (early or confirmed) must be
established before such a request can be sent.
Note that it might also be possible to provide a means of
indicating the identity of the alerting or answering party in the
response to the INVITE request. However, at present the problem
of authenticating a response is still subject to study. In the
absence of a solution to the response identity problem, the simple
solution of using a request in the opposite direction to the
INVITE request is sufficient.
This solution involves changing the URI (not the tags) in the To and
From header fields of mid-dialog requests and their responses,
compared with the corresponding values in the dialog forming request
and response. Changing the To and From header field URIs was
contemplated in Section 12.2.1.1 of RFC 3261, which says "Usage of
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the URI from the To and From fields in the original request within
subsequent requests is done for backwards compatibility with RFC
2543, which used the URI for dialog identification. In this
specification, only the tags are used for dialog identification. It
is expected that mandatory reflection of the original To and From URI
in mid-dialog requests will be deprecated in a subsequent revision of
this specification."
This document therefore deprecates mandatory reflection of the
original To and From URIs in mid-dialog requests and their responses.
It is assumed that deployed proxies will already be able to tolerate
a change of URI, since this has been expected for a considerable
time. To cater for any UAs that are not able to tolerate a change of
URI, a new option tag "dialogUriChange" is introduced for providing a
positive indication of support in the Supported header field.
OPEN ISSUE. Should this be extended to allow a URI in the To
header field of a response to change compared with the To header
field in a request? This could convey a connected identity in a
response to an INVITE request, but it would not be authenticated.
Authentication would have to rely on transitive trust, which might
be feasible in a closed environment where the sips URI scheme is
used.
6. Behaviour
6.1. Behaviour of a UA that issues an INVITE request
When issuing an INVITE request, a UA that supports changes of URI in
the From and To headers during a dialog SHOULD include the
dialogUriChange option tag in the Supported header field.
After a dialog has been formed (after receipt of a reliable response
to the INVITE request), if the dialogUriChange option tag has been
received in a Supported header field and if the identity associated
with the UA changes, the UA SHOULD issue a request on the same dialog
containing the new identity in the URI of the From header field.
Unless there is a need to invoke any other method, the UPDATE method
[4] SHOULD be used if supported by the peer UA. If the UPDATE method
is not supported by the peer UA, the re-INVITE method SHOULD be used.
but this necessitates waiting until the dialog is confirmed.
OPEN ISSUE. RFC 3311 talks about circumstances in which an UPDATE
request cannot contain an SDP offer, yet does not explicitly talk
about the use of UPDATE requests without SDP offers. It needs to
be resolved whether an UPDATE request can be used in order to
convey just a revised URI in the From header field even though no
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SDP offer needs to be sent at the time. If the outcome is that an
UPDATE request must contain an SDP offer, then SDP offer will need
to be included in the UPDATE request.
After sending a request with a revised URI in the From header field,
the UA SHALL send the same URI in the From header field of any future
requests on the same dialog, unless the identity changes again.
If the UA has received a request from the peer UA in which the From
header field URI differs from the To header field in the last request
the UA sent on that dialog, the UA MAY indicate the revised identity
to the user. In addition the UA SHOULD use this revised URI in the
To header field of any future requests it sends on the same dialog.
OPEN ISSUE. Is this a good idea to change the content of the To
header field URI on subsequent requests? How should a UAS for a
subsequent request react it if receives the wrong value in the To
header field URI, e.g., it receives the old URI when it has
already sent an UPDATE request containing a new From header field
URI?
6.2. Behaviour of a UA that receives an INVITE request
After receiving an INVITE request, a UA that supports changes of URI
in the From and To headers during a dialog SHOULD include the
dialogUriChange option tag in the Supported header field of any
dialog-forming response.
After a dialog has been formed (after sending a reliable response to
the INVITE request, i.e., a 2xx response or a reliable 1xx response),
if the dialogUriChange option tag has been received in a Supported
header field and if the identity associated with the UA differs from
that received in the To header field of the INVITE request, the UA
SHOULD issue a request on the same dialog containing the new identity
in the URI of the From header field. Unless there is a need to
invoke any other method, the UPDATE method SHOULD be used.
After sending a request with a revised URI in the From header field,
the UA SHALL send the same URI in the From header field of any future
requests on the same dialog, unless the identity changes again.
If the UA has received a request from the peer UA in which the From
header field URI differs from that received in the previous request
on that dialog, the UA MAY indicate the revised identity to the user.
In addition the UA SHOULD use this revised URI in the To header field
of any future requests it sends on the same dialog.
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6.3. Behaviour of a UA during an established INVITE-initiated dialog
If the dialogUriChange option tag has been received in a Supported
header field and if the identity associated with the UA differs from
that received in the To header field of the INVITE request, the UA
SHOULD issue a request on the same dialog containing the new identity
in the URI of the From header field. Unless there is a need to
invoke any other method, the UPDATE method SHOULD be used.
After sending a request with a revised URI in the From header field,
the UA SHALL send the same URI in the From header field of any future
requests on the same dialog, unless the identity changes again.
If the UA has received a request from the peer UA in which the From
header field URI differs from that received in the previous request
on that dialog, the UA MAY indicate the revised identity to the user.
In addition the UA SHOULD use this revised URI in the To header field
of any future requests it sends on the same dialog.
7. Examples
7.1. Sending connected identity after answering a call
In this example Carol's UA has been reached by retargeting at the
proxy and thus her identity (AoR) is not equal to that in the To
header field of the received INVITE request (Bob). Carol's UA
therefore conveys it identity in the From header field of an UPDATE
request. The proxy also provides an authentication service and
therefore adds Identity and Identity-Info header field to the UPDATE
request.
Alice's UA PROXY Carol's UA
INVITE(1) INVITE(2)
----------------> ---------------->
200(3) 200(4)
<---------------- <----------------
ACK(5) ACK(6)
----------------> ---------------->
UPDATE(8) UPDATE(7)
<---------------- <----------------
200(9) 200(10)
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----------------> ---------------->
INVITE (1):
INVITE sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Alice@ua1.example.com>
etc.
INVITE(2):
INVITE sip:Carol@ua2.example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Alice@ua1.example.com>
Identity: "dKJ97..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
200(3):
SIP/2.0 200 OK
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>;tag=2
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Carol@ua2.example.com>
etc.
200(4):
SIP/2.0 200 OK
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>;tag=2
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Carol@ua2.example.com>
etc.
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ACK (5):
ACK sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 ACK
etc.
ACK (6):
ACK sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 ACK
etc.
UPDATE (7):
UPDATE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Carol@ua2.example.com>
etc.
Note that the URI in the From header differs from that in the To
header in the INVITE request/response. However, the tag is the
same as that in the INVITE response.
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UPDATE (8):
UPDATE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Carol@ua2.example.com>
Identity: "cdKJH43..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
200(9):
SIP/2.0 200 OK
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Alice@ua1.example.com>
etc.
200(10):
SIP/2.0 200 OK
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Alice@ua1.example.com>
etc.
7.2. Sending revised connected identity during a call
In this example a call is established between Alice and Bob, where
Bob lies behind a B2BUA or gateway to a PSTN. Then call transfer
occurs in the B2BUA or PSTN, such that Alice becomes connected to
Carol, and a re-INVITE request is issued allowing the session to be
renegotiated. The B2BUA (or an entity behind it) or the gateway
provides the authentication service and thus generates the Identity
header in the re-INVITE request to provide authentication of Carol's
identity.
Alice's UA PROXY B2BUA or gateway
INVITE(1) INVITE(2)
----------------> ---------------->
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200(3) 200(3)
<---------------- <----------------
ACK(5) ACK(6)
----------------> ---------------->
UPDATE(8) UPDATE(7)
<---------------- <----------------
200(9) 200(10)
----------------> ---------------->
re-INVITE(11) re-INVITE(12)
<---------------- <----------------
200(13) 200(14)
----------------> ---------------->
ACK(15) ACK(16)
<--------------- <----------------
INVITE (1):
INVITE sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Alice@ua1.example.com>
etc.
INVITE(2):
INVITE sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Alice@ua1.example.com>
Identity: "dKJ97..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
200(3):
SIP/2.0 200 OK
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From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>;tag=2
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Bob@ua2.example.com>
etc.
200(4):
SIP/2.0 200 OK
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>;tag=2
Call-ID: 12345600@example.com
CSeq: 1 INVITE
Supported: dialogUriChange
Contact: <sip:Bob@ua2.example.com>
etc.
ACK (5):
ACK sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 ACK
etc.
ACK (6):
ACK sip:Bob@example.com SIP/2.0
From: <sip:Alice@example.com>;tag=1
To: <sip:Bob@example.com>
Call-ID: 12345600@example.com
CSeq: 1 ACK
etc.
UPDATE (7):
UPDATE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Bob@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Bob@ua2.example.com>
Identity: "cdKJH43..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
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UPDATE (8):
UPDATE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Bob@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Bob@ua2.example.com>
Identity: "cdKJH43..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
200(9):
SIP/2.0 200 OK
From: <sip:Bob@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Alice@ua1.example.com>
etc.
200(10):
SIP/2.0 200 OK
From: <sip:Bob@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 2 UPDATE
Contact: <sip:Alice@ua1.example.com>
etc.
re-INVITE (11):
INVITE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 INVITE
Contact: <sip:Carol@ua3.example.com>
Identity: "ecdFG24..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
re-INVITE (12):
INVITE sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
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To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 INVITE
Contact: <sip:Carol@ua3.example.com>
Identity: "ecdFG24..."
Identity-Info: <https://example.com/cert>;alg=rsa-sha1
etc.
200(13):
SIP/2.0 200 OK
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 INVITE
Contact: <sip:Alice@ua1.example.com>
etc.
200(14):
SIP/2.0 200 OK
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 INVITE
Contact: <sip:Alice@ua1.example.com>
etc.
ACK (15):
ACK sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 ACK
etc.
ACK (16):
ACK sip:Alice@ua1.example.com SIP/2.0
From: <sip:Carol@example.com>;tag=2
To: <sip:Alice@example.com>;tag=1
Call-ID: 12345600@example.com
CSeq: 3 ACK
etc.
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8. IANA considerations
This specification registers a new SIP option tag, as per the
guidelines in Section 27.1 of RFC 3261.
Name: dialogUriChange
Description: This option tag is used to indicate that a UA supports
changes to URIs in From and To header fields during a dialog.
9. Security considerations
[3] discusses security considerations relating to the Identity header
in some detail. Those same considerations apply when using the
Identity header to authenticate a connected identity in the From
header URI of a mid-dialog request.
A received From header field in a mid-dialog request that is not
accompanied by a valid Identity header field (or other means of
authentication) cannot be trusted (except in very closed
environments) and should be treated in a similar way to a From header
field in a dialog-initiating request that is not backed up by a valid
Identity header field.
A signed connected identity in a mid-dialog request (URI in the From
header field accompanied by a valid Identity header field) provides
information about the peer UA in a dialog. In the case of the UA
that was the UAS in the dialog-forming request, this identity is not
necessarily the same as that in the To header field of the dialog-
forming request. This is because of retargeting during the routing
of the dialog-forming request. A signed connected identity says
nothing about the legitimacy of such retargeting, but merely reflects
the result of that retargeting.
Likewise, when a signed connected identity indicates a change of
identity during a dialog, it conveys no information about the reason
for such change of identity or its legitimacy.
Use of the sips URI scheme can minimise the chances of attacks in
which inappropriate connected identity information is sent, either at
call establishment time or during a call.
Privacy may be required by the user of a connected UA. To achieve
privacy the UA MUST either decline to change the URI in the From
header field of a mid-dialog request or populate it in the way
described in [3] when anonymity is required.
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10. Acknowledgments
Thanks to Francois Audet, Frank Derks, Steffen Fries, Cullen Jennings
and Jon Peterson for providing valuable comments.
11. References
11.1. 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] Peterson, J. and C. Jennings, "Enhancements for Authenticated
Identity Management in the Session Initiation Protocol (SIP)",
draft-ietf-sip-identity-06 (work in progress), October 2005.
[4] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3893, September 2002.
[5] Barnes, M., "An Extension to the Session Initiation Protocol
(SIP) for Request History Information", RFC 3893, November 2005.
11.2. Informative References
[6] Peterson, J., "Session Initiation Protocol (SIP) Authenticated
Identity Body (AIB) Format", RFC 3893, September 2004.
[7] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
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Author's Address
John Elwell
Siemens plc
Technology Drive
Beeston, Nottingham NG9 1LA
UK
Phone: +44 115 943 4989
Email: john.elwell@siemens.com
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