One document matched: draft-jennings-sipping-connected-00.txt
SIPPING WG C. Jennings
Internet-Draft Cisco Systems, Inc.
Expires: August 7, 2004 February 7, 2004
Updating the Connected Party in SIP
draft-jennings-sipping-connected-00.txt
Status of this Memo
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Copyright Notice
Copyright (C) The Internet Society (2004). All Rights Reserved.
Abstract
Communication systems often have situations in which the party at the
far end of the call changes, making it necessary to have a way to
notify the near end of the new identity at the far end. This draft
discusses ways to update this "connected party" information in SIP.
This work is being discussed on the sipping@ietf.org mailing list.
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Table of Contents
1. Conventions and Definitions . . . . . . . . . . . . . . . . . 3
2. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3.1 Connected UAS or UAC Updated . . . . . . . . . . . . . . . . . 3
3.2 Early UAC Updated . . . . . . . . . . . . . . . . . . . . . . 3
3.3 Early UAS Updated . . . . . . . . . . . . . . . . . . . . . . 4
4. Solutions Using Transfer Mechanisms . . . . . . . . . . . . . 4
4.1 Connected UAS or UAC Updated . . . . . . . . . . . . . . . . . 4
4.2 Early UAC Updated . . . . . . . . . . . . . . . . . . . . . . 4
4.3 Early UAS Updated . . . . . . . . . . . . . . . . . . . . . . 5
5. Solutions Based on UPDATE . . . . . . . . . . . . . . . . . . 5
6. Why PAI is Useless . . . . . . . . . . . . . . . . . . . . . . 6
7. Recommendations . . . . . . . . . . . . . . . . . . . . . . . 7
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 7
Normative References . . . . . . . . . . . . . . . . . . . . . 7
Informative References . . . . . . . . . . . . . . . . . . . . 7
Author's Address . . . . . . . . . . . . . . . . . . . . . . . 8
Intellectual Property and Copyright Statements . . . . . . . . 9
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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 [6].
2. Introduction
SIP[5] initiates sessions but it also provides information on the
identities of the parties at both ends of a session. This is
necessary as users find it very desirable for knowing how to deal
with the communications that SIP is initiating. 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. The
use cases here are split into two categories: where the identity
change happens in an early dialog or a connected call, or where the
callee's or the caller's identity changes. How the identity of the
called party is provided to the caller is not well defined in SIP.
This is early work discussing the problems and looking at possible
solutions to this problem. It needs considerable work and
consideration for how the approach would interact with existing
systems. It is being discussed on the sipping@ietf.org mailing list.
3. Use Cases
Identity information often contains both a human readable name and a
network address such as a phone number or SIP URL.
3.1 Connected UAS or UAC Updated
The classic case is when Alice calls Bob who forwarded to Charlie who
answers the phone. Alice would like to know she is talking to Charlie
not Bob.
These cases do not involve early dialogs. Since the call has been
established, the use cases are symmetrical. Either the UAS or UAC
could need to be updated in any of these cases.
The classic case is when a call is transferred.
The UAS has originally answered the call with the identity Help Desk
but wishes to change it, mid-call, to Alice.
3.2 Early UAC Updated
In a typical example, phone A rings but user B remotely picks up the
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ringing phone from a different one.
Call queuing in an ACD is another example. The call may initially be
queued for the "Help Desk" but later be moved to ring Bob's phone.
It should be noted that the history requirements[9] cover this case.
3.3 Early UAS Updated
First, Alan dials Charlie on behalf of another user, Bob in this
case. While Charlie's phone is ringing, Alan steps out of the call
and lets Bob talk to Charlie.
4. Solutions Using Transfer Mechanisms
Many of these use cases seem to come up because there is a B2BUA
involved. This B2BUA may be acting as a call control system or it may
be a PSTN inter-working GW which is effectively a B2BUA with the PSTN
connecting the two UAs that form the B2BUA. It's fine to have a B2BUA
connected to a P2P system; however, the whole design of the B2BUA
concept in SIP is to make B2BUA just look like a valid peer to the
P2P system.
These solutions are derived by looking at what would happen if
approximately the same functionality were done in a P2P system, and
looking at signaling from the point of view of the UA that is
receiving the updated identity.
4.1 Connected UAS or UAC Updated
When A is in a connected call with B and B wishes to update the
identity that A sees to C, this is the same from A's point of view as
B transferring the call to C. In this case the identity can be
updated by sending a new INVITE to A with the From set to C and
including a replaces header to indicate it replaces the original
call.
This approach may have bad interaction with who gets billed.
4.2 Early UAC Updated
Here the UAS wishes to change its identity and update the UAC. The
UAS sends back a new early dialog with the new identity. From the UAC
point of view, this looks like the call to a UAS that forked to form
a new early dialog with a different UAS. This might have bad
interactions with early media.
Trying to solve this the same way as the when the UAC is connected
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does not work because the new INVITE with the replaces results in a
completely wrong view about which UA is in the ringing state and
which is already off hook.
4.3 Early UAS Updated
Here the UAC wishes to change its identity and update it on all the
UASs that it has forked to while it had an outstanding INVITE. The
transfer issues draft [3] clearly points out that the UAS is a moving
target due to forking with the "Consultative Turned Blind" issues.
Changing the identity of UAC complicates this even further because
the routing of the invite may have been dependent on the identity of
the UAS. For example, calls from Private may go straight to voice
mail while calls from Boss get forked to office phone and cell phone.
It is probable that whatever solution is chosen to deal with the
"Consultative Turned Blind" problem in [3] can also be used to update
identity in this scenario. The recommendation here is to hold off on
defining something special for identity updates until the solution
for the transfer issues problem is resolved.
The idea of sending a new INVITE with replaces, waiting a little
while, and then sending a CANCEL to the original INVITE has been
suggested. This fails in the case where the call is to the PSTN and
forks to a different gateway.
5. Solutions Based on UPDATE
User agents could send UPDATE requests that do not contain SDP, but
do change the To and From headers. The tags in the To or From headers
would not be changed. Any UPDATES received by an RFC 2543 [10]
system would fail to match the transaction due to the changed To or
From. This would not result in the call failing but would only result
in the identity failing to be displayed correctly.
Changing the To and From headers was contemplated in Section 12.2.1.1
of RFC 3261 which says "Usage of 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."
As an alternative to changing the To and From headers, a new Name
header could be created that represents the identity of the sender of
the message.
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The biggest problem with these approaches is that they make it very
hard to apply new policy to the call if the user's identity changes.
Consider the example above. The identity of the caller determines
where the call is forked; with this approach, the system would not
change where the call was being forked if the identity of the caller
changed.
Another issue is how this interacts with outstanding UPDATES that are
being used for offer/answer negotiation.
6. Why PAI is Useless
PAI, described in RFC 3325 [4], was designed to work in very limited
trust domains. It is primarily useful when a UAC wishes its identity
to be anonymous to the UAS but the intermediate trust domain needs to
know the UAC's identity for legal call trace reasons. There are no
legal call trace requirements of this type from the UAS to the UAC.
If the UAS wishes to change its identity to Anonymous, there is no
need to carry the real identity along. It is important to understand
this: the requirement solved by PAI is asymmetrical and does not
apply in the backwards direction.
The requirements driving the work in this document are about
informing the party at the far end of the call of a changed identity.
PAI will never work for this. The fundamental requirement for PAI is
that as the PAI crosses from one trust domain A to the next trust
domain B, the next trust domain B must discard it unless it can
independently verify it. This is so because if B passes it to any
other element even inside its trust domain, those elements will
assume the PAI is true even though B does not know this to be true.
The end user device that is capable of displaying the PAI information
is never in the same trust domain as the network elements, so it must
discard this information as it comes in. RFC 3325 specifically says
"However, if a User Agent Server receives a message from a previous
element that it does not trust, it MUST NOT use the
P-Asserted-Identity header field in any way." A UAS that sits in the
user's hand is not going to be trusted by the "network" trust domain.
Trust domains are symmetrical - you cannot have a trust domain in
which the phone trusts the network but the network does not trust the
phone.
To summarize, given two trust domains A and B, even if you passed the
PAI around in responses in A, as soon as it got passed to B, it would
be discarded or unusable. If it got passed in a response to a proxy
in B, 3325 says that "the proxy MUST authenticate the originator of
the message" which of course the proxy cannot do for a response, so
it would have to drop the PAI. There is no use case in which the UA
producing the PAI and the UA that could display a PAI to an end user
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are in the same trust domain. A new header, like Name, could be
defined as described in Section 5 but PAI is not useful for this.
7. Recommendations
Deprecating forking and early media do not seem feasible. The early
unattended transfer problem has been floating around for a long time
with no good solution. Using UPDATE or some new method to update
connected party information after a dialog is formed seems very
appealing. This name could be in a changed To or From header or in a
new header. This same approach could be used when the caller identity
changed in an early dialog. History seem like a good approach for
coping with the ever changing identity of various UASs during early
dialogs.
8. Acknowledgments
Thanks to Randy Baird and Rohan Mahy.
Normative References
[1] Sparks, R., "The SIP Referred-By Mechanism",
draft-ietf-sip-referredby-03 (work in progress), August 2003.
[2] Biggs, B., Dean, R. and R. Mahy, "The Session Inititation
Protocol (SIP) 'Replaces' Header", draft-ietf-sip-replaces-04
(work in progress), August 2003.
[3] Petrie, D., "SIP Transfer Issues",
draft-petrie-sipping-xfer-issues-00 (work in progress), October
2003.
[4] Jennings, C., Peterson, J. and M. Watson, "Private Extensions to
the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks", RFC 3325, November 2002.
[5] 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.
[6] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
Informative References
[7] Elwell, J., "Interworking between SIP and QSIG",
draft-ietf-sipping-qsig2sip-02 (work in progress), August 2003.
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[8] Venkataramanan, V., "Enhancements to Asserted Identity to
Enable Called Party Name Delivery using SIP",
draft-venkatar-sipping-called-name-00 (work in progress), June
2003.
[9] Barnes, M., "SIP Generic Request History Capability Ãû
Requirements", draft-ietf-sipping-req-history-04 (work in
progress), June 2003.
[10] Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
Author's Address
Cullen Jennings
Cisco Systems, Inc.
170 West Tasman Dr.
MS: SJC-21/2
San Jose, CA 95134
USA
Phone: +1 408 902 3341
EMail: fluffy@cisco.com
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