Re: [Sipping] Progress on draft-jesske-sipping-etsi-ngn-reason-03

["DRAGE, Keith \(Keith\)" <drage at alcatel-lucent.com>]

For REQ-1:

I think we need to capture in this requirement (or in association with
this requirement) somehow:

1)	that this is between two "PSTN gateways".

2)	that scenarios exist between PSTN gateways where SIP-I is not a
solution (and you may need to describe one of these in a use cases part
of the document).

For REQ-2: 

Is this the only use for the requirement. There are two ways of building
this requirement:

-	the gateway passes on the cause and some announcement server
translates the cause into the appropriate announcement.
-	the gateway translates the cause into an identifier of a
particular announcement, and passes that information to the announcement
server.

Both fulfil this requirement. Is there some other requirement that
causes these to be distinguished.

I would also note that neither of these requirements justify the
following "display" UA procedure in the document:

   A UA that supports the Reason header field can process the Q.850 
   Cause Value and display it or an equivalent text. The inclusion of a 
   Reason header field by UA is only for 2B2 UA interworking with the 
   PSTN/ISDN or providing services foreseen.

Are we missing a requirement or is the procedure text subject to
revision.

Regards

Keith 

> -----Original Message-----
> From: sipping-bounces at ietf.org 
> [mailto:sipping-bounces at ietf.org] On Behalf Of Dale.Worley at comcast.net
> Sent: Tuesday, July 01, 2008 7:08 PM
> To: sipping at ietf.org
> Subject: Re: [Sipping] Progress on 
> draft-jesske-sipping-etsi-ngn-reason-03
> 
>    From: "Jesske, R" <R.Jesske at telekom.de>
> 
>    I'll try to reword the requirements:
> 
>    REQ-1:
>    It should be possible to support PSTN-SIP-PSTN scenarios where the
>    reason of a call release can be transferred though the SIP domain
>    without any loss of information and no change of reason. 
> 
>    REQ-2: 
>    It should be possible to provide correct announcements to a SIP
>    user based on the reason for call clearing within the PSTN network
>    or the PSTN user.
> 
>    Is that better?
> 
> Yes, that is much clearer!
> 
> I assume that the ISUP reasons are two-digit codes, and there 
> is no intrinsic restriction on what two-digit combinations 
> they can be.
> That is, for REQ-1, codes 00 through 99 must be supported, 
> even if they have no assigned meaning now.
> 
> Dale
> _______________________________________________
> Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP 
> Use sip-implementors at cs.columbia.edu for questions on current 
> sip Use sip at ietf.org for new developments of core SIP
> 
_______________________________________________
Sipping mailing list  https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sip at ietf.org for new developments of core SIP