[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
RE: [Sip] Update of display name during a call
That's exactly what we proposed in
http://www.ietf.org/internet-drafts/draft-venkatar-sipping-called-name-00.tx
t.
regards,
Venkatesh
-----Original Message-----
From: Henry Chen [mailto:hjlechen@cisco.com]
Sent: Monday, September 01, 2003 8:39 PM
To: Cullen Jennings; Elwell, John; sip@ietf.org
Subject: RE: [Sip] Update of display name during a call
Not really since a UA only expect to receive PAI, called party receives
it in a request and calling party receives it in the response.
-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: Sunday, August 31, 2003 10:56 AM
To: Henry Chen; Elwell, John; sip@ietf.org
Subject: Re: [Sip] Update of display name during a call
Well the PAI is a network asserted From. You are talking about a network
asserted To. I suspect putting both in the same field with lead to
problems.
On 8/29/03 10:49, "Henry Chen" <hjlechen@cisco.com> wrote:
>
> The caller's name can be authenticated and put in a PAI header by a
> proxy in the initial Invite according to RFC3325.
>
> The problem is how to pass called party name to the call originator
> when the call is forwarded by the proxy.
>
> Is there any problem to put called party id or display name in a PAI
> header in the response to Invite by the proxy in the trusted domain
> serving the call?
>
> Thanks,
>
> Henry
>
>
> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Friday, August 29, 2003 8:32 AM
> To: 'Cullen Jennings'; sip@ietf.org
> Subject: RE: [Sip] Update of display name during a call
>
>
> Cullen,
>
> In trusted situations where P-Asserted-ID is in use, I think it would
> be reasonable to use P-Asserted-ID rather than AIB in the re-INVITE or
> UPDATE.
>
> John (john.elwell@siemens.com)
>
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: 28 August 2003 21:11
> To: sip@ietf.org
> Cc: Cullen Jennings
> Subject: [Sip] Update of display name during a call
>
>
>
> There has been some discussion on the issues of changing the display
> name associated with a call during the call. This has come up in QSIG
> cases, it has come up in B2BUA cases, and in some other PBX and PSTN
> interoperability issues. I think it is worth solving.
>
> How about this as a proposal. The UA can send an re-INVITE or UPDATE
> (as
> appropriate) with an AIB. The AIB can be authenticated in the normal
> way. The From and To inside the AIB can have a modified display name.
> The tag will not be changed. If a 2543 device receives one of these
> message, it may end up ignoring it but it will not cause harm. If a
> device that receives one of these and understands it, it SHOULD update
> the display name information. This can happen both from callee to
caller
> and caller to callee. The AIB might have a PAI in it too.
>
> Section 12.2.1.1 of 3261 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.
>
> Clearly this is in line with that.
>
> AIBs are about identity and solve some of the complex issues of third
> parties asserting identity on behalf of others. I think this is a good
> place to deal with issues that is about the displayed identity.
>
> Thoughts? Issues with this? Better Ideas...
>
> Thanks, Cullen
>
>
>
>
>
>
> _______________________________________________
> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip