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nagrawala | 3 Apr 2002 14:26

Re: BYE from server side


Naresh K Agrawala
04/03/2002 06:52 PM

To:   sipping <at> ietf.org
cc:

Subject:  BYE from server side

Hi,
 the scenario is like this.
Suppose an UAC sends an Cseq with 1 INVITE & call is established.
Now from another side  BYE is received with Cseq  1 BYE. what should be the
behaiour of UAC?

Another case is:
Gateway( MGC) receives an INVITE with Cseq as 5 INVITE and call is established.
Now when the callee releases the call, what should be the Cseq in the BYE sent
from the gateway?

Thanks & Regards,
 naresh

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SOUMENDU GHARA | 3 Apr 2002 17:55

Isup to sip mapping

Hi,
Do we have a doc which talks about how to map a ISUP APM message to SIP-T. How should be the handling of the segmented APM.

Regards,

Soumendu Ghara
Next Generation Networks
JSLEE Development Team
Office: 91+80+5538301 extn: 2210/2753
~ I am ready to meet my Maker. Whether my Maker is prepared for the great ordeal of meeting me is another matter - Winston Churchill

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Sanjoy Sen | 3 Apr 2002 20:51

SIP & T.120


Is there any ongoing work or planned work in the WG to integrate SIP and T.120/whiteboarding ?  Is this the right place for this work or should this be handled at MMUSIC?

The fundamental issues that come to mind :
1) T.120 family does everything from capability negotiation (T.121), session set-up (T.122), to conference control (T.124). Should we, for simplicity's sake, use SIP just to set-up transport for T.120 and leave everything else for T.120 to handle? This makes some sense given the wide deployment of complete T.120 stacks. However, there're lot of overlap between the Generic Conferencing Model (T.124) and SIP conferencing models and framework, and their interactions need to be sorted out (e.g., Join). What're some of the new SDP attributes that need to be defined?

2) The alternative is, of course, to recreate "whiteboarding" using SDP & SIP conferencing models. Is anybody aware of any ongoing work in other bodies (e.g., W3C)?

What is the general opinion of the WG on starting some work on this?

Thanks,

Sanjoy
sanjoy <at> nortelnetworks.com

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Tom-PT Taylor | 3 Apr 2002 21:02

RE: Isup to sip mapping


That will be covered in current work on SIP-BICC interworking in ITU-T SG
11.

-----Original Message-----
From: SOUMENDU GHARA [mailto:soumendu.ghara <at> wipro.com]
Sent: Wednesday, April 03, 2002 10:55 AM
To: Sipping <at> ietf.org
Subject: [Sipping] Isup to sip mapping

Hi, 
Do we have a doc which talks about how to map a ISUP APM message to SIP-T.
How should be the handling of the segmented APM.
Regards, 
Soumendu Ghara 
Next Generation Networks 
JSLEE Development Team
Office: 91+80+5538301 extn: 2210/2753 
~ I am ready to meet my Maker. Whether my Maker is prepared for the great
ordeal of meeting me is another matter - Winston Churchill

_______________________________________________
Sipping mailing list  https://www1.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
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Even, Roni | 4 Apr 2002 13:12
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RE: SIP & T.120

Sanjoy,
 
The issue of integrating SIP and T.120 was mentioned in draft-levin-sip-for -video. the work is continued after the Minneapolis meeting by a group that tries to define the requirements and framework for multimedia conferencing.
Using T.120 for control was discussed in the initial phases of H.323 and was not accepted due to the complexity of T.120 as well as the low availability of a stack. I would argue your point of wide deployment. It is true that Netmeeting is a T.120 client but if you want a stack for an embedded platform you have a problem. Databeam who used to be the major vendor of such stack does not do it this days.
As for data collaboration there is no other protocol and I think that the industry would welcome such a protocol. Today you can see web collaboration applications but they do not have full application sharing capabilities.
Roni Even
Polycom
-----Original Message-----
From: Sanjoy Sen [mailto:sanjoy <at> nortelnetworks.com]
Sent: Wednesday, April 03, 2002 9:52 PM
To: sipping <at> ietf.org
Subject: [Sipping] SIP & T.120


Is there any ongoing work or planned work in the WG to integrate SIP and T.120/whiteboarding ?  Is this the right place for this work or should this be handled at MMUSIC?

The fundamental issues that come to mind :
1) T.120 family does everything from capability negotiation (T.121), session set-up (T.122), to conference control (T.124). Should we, for simplicity's sake, use SIP just to set-up transport for T.120 and leave everything else for T.120 to handle? This makes some sense given the wide deployment of complete T.120 stacks. However, there're lot of overlap between the Generic Conferencing Model (T.124) and SIP conferencing models and framework, and their interactions need to be sorted out (e.g., Join). What're some of the new SDP attributes that need to be defined?

2) The alternative is, of course, to recreate "whiteboarding" using SDP & SIP conferencing models. Is anybody aware of any ongoing work in other bodies (e.g., W3C)?

What is the general opinion of the WG on starting some work on this?

Thanks,

Sanjoy
sanjoy <at> nortelnetworks.com

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Rohan Mahy | 4 Apr 2002 19:01
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Poll for interest in an interim SIP/SIPPING meeting

Hi,

I've heard from some folks that they would like to have another interim
SIP/SIPPING meeting like we did in Dallas between IETF 49 and 50.  I'd
like to here from folks *privately* please if they are interested in such
an interim meeting, probably during the month of May.  Also if you are
strongly opposed to such a meeting, I would like to hear from you
*privately* as well.

As for the possible locations, I am currently entertaining both Flagstaff,
Arizona, and Boston.  I can get a very good room rate for one, and the
other is easier for most folks to get to.  If you send me *private* mail
in support of an interim meeting, your location suggestion is also appreciated.

As always, any interim meeting has to be announced on ietf-announce at
least 30 days before the meeting, so please get me your comments early.
For your convenience, you can just click on the apropriate link.

Yes Reply:  mailto:rohan <at> cisco.com?Subject=[sip-interim-yes]

No  Reply:  mailto:rohan <at> cisco.com?Subject=[sip-interim-no]

thanks,
-rohan

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Michael Hammer | 4 Apr 2002 22:50
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draft-yu-tel-url-04.txt comment

James,

Section 6: npdi-ident = "yes" | "no"
Yet, section 7.3, example 1: "npdi=yes <at> sip.abc,com"

Would that be rejected or truncated?

Also, cic-ident = *phonedigit
Section 7.3, example 2: "cic=+1-6789 <at> sip.xyz.com"

Does phonedigit include all alphanumeric?

Mike

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Rohan Mahy | 5 Apr 2002 01:33
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possible agenda/motivation for interim meeting

Hi,

A couple folks have asked for my motivation for suggesting an interim.
Basically we've got a lot of stuff on our plate in both WGs, and much of
it is in that midway done phase were face time is most valuable.
An interim would give us time to resolve several sticky technical issues
and then get drafts out before the cutoff date for IETF54.  I think
we could have a much more productive meeting in Japan as a result.

Topics in serious need of face time:

- the longer term privacy solution
- a whole raft of security issues
- REFER, replaces, and cc-transfer
- call info and conf info packages
- multiparty and conferencing requirements/frameworks
- possibly more ISUP interworking issues
- possibly a working session to revise the AAA requirements draft

The last two would depend on critical mass of folks willing to work on
these.

thanks,
-rohan

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Brett Tate | 5 Apr 2002 02:55
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RE: draft-yu-tel-url-04.txt comment

> Section 6: npdi-ident = "yes" | "no"
> Yet, section 7.3, example 1: "npdi=yes <at> sip.abc,com"
> 
> Would that be rejected or truncated?

The ' <at> ' represents the end of the 
telephone-subscriber part of the sip-url.

Thus the npdi has a value of yes.

> Also, cic-ident = *phonedigit
> Section 7.3, example 2: "cic=+1-6789 <at> sip.xyz.com"
> 
> Does phonedigit include all alphanumeric?

The ' <at> ' represents the end of the 
telephone-subscriber part of the sip-url.

Thus the cic-ident has a value of +1-6789.

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Beck01, Wolfgang | 5 Apr 2002 15:04
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SIP billing service, new draft-beck-sipping-billing-scen-00.txt


During the Minneapolis meeting an idea came up to use SIP
to invoke billable services ("SIP = Service Invoice Protocol").
The draft
http://www.ietf.org/internet-drafts/draft-beck-sippping-billing-scen-00.txt
outlines some scenarios, security considerations and mechanisms.

What is a billable service?

-  a PSTN gateway?
-  a non-free support hotline?
-  accepting SIP calls from obtrusive salespeople?
-  accepting RTP traffic that will show up on the bill of my volume-based
IP access?

Extra header(s) are needed to convey pricing information and to protect callers.

CPL could be extended in a way that enables users to individually charge
callers. 

Non-repudiation is an issue here. 

Comments?

--
Wolfgang Beck
T-Systems Nova GmbH 

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