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From: "Steffen Fries" <steffen.fries@siemens.com>
Organization: Siemens AG
To: sipping@ietf.org
Date: Mon, 03 Feb 2003 10:12:36 +0100
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Subject: [Sipping] SIP, SRTP and MIKEY
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Hi,

I'm not quite sure, which working group fits best for this 
question, thus I'm posting this question here too. I already 
had a post in the the SIP WG but got only one reply. 

When SIP and SRTP are used in conjunction, an appropriate
key management is necessary for SRTP. Within the MSEC WG 
MIKEY has been defined, which is thought to be used (also) 
for SRTP. 

MIKEY and an related draft (draft-ietf-msec-MIKEY-DHHMAC-
01.txt) offer 4 different key management methods based on:
- pre-shared secrets
- public key encryption
- Diffie Hellman protected with signatures
- Diffie Hellman protected with pre-shared secrets

When MIKEY and SRTP are to be used in a SIP environment, 
what would be a suitable choice out of the four options?

I'm not sure if the pre-shared secret based methods are 
suitable, since this would assume, that all users who want to 
communicate need to exchange a shared secret before. Well, this 
could be done by puting a shared secret in the SIP message and 
securing this by S/MIME, but then MIKEY would be protected by 
symmetric methods, although asymmetric technology was used to 
secure the shared secret transport, namely S/MIME. One could 
also use the certificates and private keys to secure MIKEY 
right from the beginning. 

There might be scenarios where the symmetric case is 
appropriate, but I'm not sure if this is a rather general case.

Is MIKEY generally considered for key management in SIP or will 
this rather be done using draft-baugher-mmusic-sdpmediasec-
00.txt secured by S/MIME?

The usage of MIKEY and SRTP is especially interesting in 
conjunction with other multimedia protocols like H.323. Voice 
encryption and associated key management could be performed 
across the different signaling protocols.

Was there already a discussion related to this question, which 
I may be missed? 

Regards
        Steffen


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From mailnull@www1.ietf.org  Mon Feb  3 07:53:55 2003
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From: "Fredrik Lindholm (EAB)" <Fredrik.Lindholm@era.ericsson.se>
To: "'Steffen Fries'" <steffen.fries@siemens.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>,
        "Elisabetta Carrara (EAB)"
	 <Elisabetta.Carrara@era.ericsson.se>
Subject: RE: [Sipping] SIP, SRTP and MIKEY
Date: Mon, 3 Feb 2003 10:51:53 +0100 
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Hi Steffen,

I'll guess Elisabetta's mail got lost when she tried 
to answer you the last time (please also cc: AVT and/or 
MSEC/MMUSIC if you have questions regarding SRTP 
and/or MIKEY).

Best,
Fredrik

-----Original Message-----
From: Elisabetta Carrara (EAB) 
Sent: den 17 januari 2003 16:41
To: 'Steffen Fries'; sip@ietf.org
Cc: Fredrik Lindholm (EAB)
Subject: RE: SRTP and MIKEY usage in SIP 


Hi Steffen
the public-key based methods are most suitable for
client-to-client communications. The pre-shared could
be for ex a good choice for client(s)-to-server 
communications.
So, for SIP the public-key based may be more suitable,
generally speaking.

Cheers,
/E

 



> -----Original Message-----
> From: Steffen Fries [mailto:steffen.fries@siemens.com]
> Sent: den 17 januari 2003 16:20
> To: sip@ietf.org
> Cc: Elisabetta.Carrara@era.ericsson.se
> Subject: SRTP and MIKEY usage in SIP 
> 
> 
> Hi,
> 
> I'm not quite sure, which working group fits best for this 
> question, thus I'm starting with the SIP WG ;-)
> 
> When SIP and SRTP are used in conjunction, an appropriate
> key management is necessary for SRTP. Within the MSEC WG 
> MIKEY has been defined, which is thought to be used (also) 
> for SRTP. 
> 
> MIKEY and an related draft (draft-ietf-msec-MIKEY-DHHMAC-
> 01.txt) offer 4 different key management methods based on:
> - pre-shared secrets
> - public key encryption
> - Diffie Hellman protected with signatures
> - Diffie Hellman protected with pre-shared secrets
> 
> When MIKEY and SRTP are to be used in a SIP environment, 
> what would be a suitable choice out of the four options?
> 
> I'm not sure if the pre-shared secret pased methods are 
> suitable, since this would assume, that all users who want to 
> communicate need to exchange a shared secret before. Well, this 
> could be done by puting a shared secret in the SIP message and 
> securing this by S/MIME, but then MIKEY would be protected by 
> symmetric methods, although asymmetric technology was used to 
> secure the shared secret transport, namely S/MIME. One could 
> also use the certificates and private keys to secure MIKEY 
> right from the beginning. 
> 
> There might be scenarios where the symmetric case is 
> appropriate, but I'm not sure if this is a rather general case.
> 
> Is MIKEY generally considered for key management in SIP or will 
> this rather be done using draft-baugher-mmusic-sdpmediasec-
> 00.txt secured by S/MIME?
> 
> The usage of MIKEY and SRTP is especially interesting in 
> conjunction with other multimedia protocols like H.323. Voice 
> encryption and associated key management could be performed 
> across the different signaling protocols.
> 
> Was there already a discussion related to this question, which 
> I may be missed? 
> 
> Regards
>         Steffen
> 
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Use sip@ietf.org for new developments of core SIP



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Subject: [sipping] draft-burger-sipping-netann-03.txt (Early media announcement)
To: "Eric Burger" <eburger@snowshore.com>
Cc: David.Rio@alcatel.fr, sipping@ietf.org
Date: Mon, 3 Feb 2003 17:17:46 +0100
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Eric,
In early media announcement scenario, you suggest to send a 487 REQUEST
TERMINATED, once the anouncement is played.
But in RFC 3261, the 487 response is said to be used only when the previous
request has been terminated with a BYE or a CANCEL.
It is not the case in early media  announcement scenario.
So my question is ca:n we consider that the early media annoucement
scenario is compliant with RFC 3261?

We've faced a case in which a softphone refuses to close a call (screen
still active) because it has received a 487 without any previous
BYE or CANCEL. Is this terminal not compliant with RFC3261?(below is the
call flow of the call)



Softphone           PROXY (SER)             MS
  INVITE ---------->          -------------->
  <-----------------          ---------------183
                      PLAY of 10 s
  <------------------         ---------------487
  ACK--------------->         ---------------->

After the ACK is sent, the call is still active.



pascal



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Until it hits the archive:

http://flyingfox.snowshore.com/i-d/sip/draft-vandyke-mscml-01.html
http://flyingfox.snowshore.com/i-d/sip/draft-vandyke-mscml-01.txt
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List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=subscribe>

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: MIME media types for DPNSS Objects
	Author(s)	: R. Mukundan et al.
	Filename	: draft-mukundan-sipping-dpnss-01.txt
	Pages		: 9
	Date		: 2003-2-3
	
This document describes Multipart Internet Mail Extensions (MIME)
type for application/Digital Private Network Signaling System (DPNSS)
objects for use in Session Initiation Protocol (SIP) applications,
according to the rules defined in RFC 2048. These types can be used
to identify Digital Private Network Signaling System 1 (DPNSS 1)
objects within a SIP message such as INVITE or INFO, as might be
implemented when using SIP in an environment where part of the call
involves interworking to DPNSS 1 based Private Networks. RFC 3204
addresses a similar need for Integrated Service Digital Network
User Part (ISUP) and Q Signaling (QSIG).

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-mukundan-sipping-dpnss-01.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<2003-2-3125419.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-mukundan-sipping-dpnss-01.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-2-3125419.I-D@ietf.org>

--OtherAccess--

--NextPart--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Tue Feb  4 06:48:37 2003
Received: from www1.ietf.org (ietf.org [132.151.1.19] (may be forged))
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA17616
	for <sipping-archive@odin.ietf.org>; Tue, 4 Feb 2003 06:48:37 -0500 (EST)
Received: (from mailnull@localhost)
	by www1.ietf.org (8.11.6/8.11.6) id h14Bs2h19975
	for sipping-archive@odin.ietf.org; Tue, 4 Feb 2003 06:54:02 -0500
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	for <sipping-web-archive@optimus.ietf.org>; Tue, 4 Feb 2003 06:54:02 -0500
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	Tue, 4 Feb 2003 06:53:34 -0500
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	by www1.ietf.org (8.11.6/8.11.6) with ESMTP id h14BpUJ19732
	for <sipping@optimus.ietf.org>; Tue, 4 Feb 2003 06:51:30 -0500
Received: from CNRI.Reston.VA.US (localhost [127.0.0.1])
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA17371;
	Tue, 4 Feb 2003 06:45:32 -0500 (EST)
Message-Id: <200302041145.GAA17371@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
Cc: sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 04 Feb 2003 06:45:32 -0500
Subject: [Sipping] I-D ACTION:draft-ietf-sipping-overlap-04.txt
Sender: sipping-admin@ietf.org
Errors-To: sipping-admin@ietf.org
X-BeenThere: sipping@ietf.org
X-Mailman-Version: 2.0.12
Precedence: bulk
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=unsubscribe>
List-Id: SIPPING Working Group (applications of SIP) <sipping.ietf.org>
List-Post: <mailto:sipping@ietf.org>
List-Help: <mailto:sipping-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=subscribe>

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Mapping of of Integrated Services Digital Network 
                          (ISUP) Overlap Signalling to the Session Initiation 
                          Protocol
	Author(s)	: G. Camarillo et al.
	Filename	: draft-ietf-sipping-overlap-04.txt
	Pages		: 12
	Date		: 2003-2-3
	
This document describes a way to map ISUP overlap signalling to SIP.
This mechanism might be implemented when using SIP in an environment
where part of the call involves interworking with the Public Switched
Telephone Network (PSTN).

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sipping-overlap-04.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-sipping-overlap-04.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<2003-2-3125647.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-sipping-overlap-04.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-sipping-overlap-04.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-2-3125647.I-D@ietf.org>

--OtherAccess--

--NextPart--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Tue Feb  4 06:56:11 2003
Received: from www1.ietf.org (ietf.org [132.151.1.19] (may be forged))
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA17826
	for <sipping-archive@odin.ietf.org>; Tue, 4 Feb 2003 06:56:11 -0500 (EST)
Received: (from mailnull@localhost)
	by www1.ietf.org (8.11.6/8.11.6) id h14C1bQ20349
	for sipping-archive@odin.ietf.org; Tue, 4 Feb 2003 07:01:37 -0500
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	for <sipping-web-archive@optimus.ietf.org>; Tue, 4 Feb 2003 07:01:37 -0500
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	Tue, 4 Feb 2003 07:01:08 -0500
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	by www1.ietf.org (8.11.6/8.11.6) with ESMTP id h14C0tJ20292
	for <sipping@optimus.ietf.org>; Tue, 4 Feb 2003 07:00:55 -0500
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	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA17791
	for <sipping@ietf.org>; Tue, 4 Feb 2003 06:54:53 -0500 (EST)
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	by wiprom2mx1.wipro.com (8.11.3/8.11.3) with SMTP id h14BwQv06052
	for <sipping@ietf.org>; Tue, 4 Feb 2003 17:28:26 +0530 (IST)
Received: from blr-ec-msg01.wipro.com ([10.200.50.99]) by blr-ec-bh2.wipro.com with Microsoft SMTPSVC(5.0.2195.5329);
	 Tue, 4 Feb 2003 17:28:21 +0530
Received: from ranjith2 ([192.168.178.234]) by blr-ec-msg01.wipro.com with Microsoft SMTPSVC(5.0.2195.5329);
	 Tue, 4 Feb 2003 17:28:21 +0530
From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
To: <sipping@ietf.org>
Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
Date: Tue, 4 Feb 2003 17:32:10 +0530
Message-ID: <NHBBLCDDNKFBAALPHPGFCEGICCAA.ranjith.mukundan@wipro.com>
MIME-Version: 1.0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
X-Priority: 3 (Normal)
X-MSMail-Priority: Normal
X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2910.0)
Importance: Normal
X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400
In-Reply-To: <200302041144.GAA17152@ietf.org>
X-OriginalArrivalTime: 04 Feb 2003 11:58:21.0188 (UTC) FILETIME=[B6861C40:01C2CC44]
Content-Transfer-Encoding: 7bit
Sender: sipping-admin@ietf.org
Errors-To: sipping-admin@ietf.org
X-BeenThere: sipping@ietf.org
X-Mailman-Version: 2.0.12
Precedence: bulk
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=unsubscribe>
List-Id: SIPPING Working Group (applications of SIP) <sipping.ietf.org>
List-Post: <mailto:sipping@ietf.org>
List-Help: <mailto:sipping-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=subscribe>
Content-Transfer-Encoding: 7bit
Content-Transfer-Encoding: 7bit

Hi List,
We would like you take a look at this draft and revert with any
comments/queries that you might have.

SIPPING Chairs,
The authors feel that the requirement and the need for this MIME type is
quite acute. Hence we'd like you to consider this for inclusion as a WG I-D
and suggest the way forward.

- ranjith..

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Tuesday, February 04, 2003 5:14 PM
To: IETF-Announce:
Cc: sipping@ietf.org
Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: MIME media types for DPNSS Objects
	Author(s)	: R. Mukundan et al.
	Filename	: draft-mukundan-sipping-dpnss-01.txt
	Pages		: 9
	Date		: 2003-2-3

This document describes Multipart Internet Mail Extensions (MIME)
type for application/Digital Private Network Signaling System (DPNSS)
objects for use in Session Initiation Protocol (SIP) applications,
according to the rules defined in RFC 2048. These types can be used
to identify Digital Private Network Signaling System 1 (DPNSS 1)
objects within a SIP message such as INVITE or INFO, as might be
implemented when using SIP in an environment where part of the call
involves interworking to DPNSS 1 based Private Networks. RFC 3204
addresses a similar need for Integrated Service Digital Network
User Part (ISUP) and Q Signaling (QSIG).

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt

To remove yourself from the IETF Announcement list, send a message to
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-mukundan-sipping-dpnss-01.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".

NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.


Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb  5 06:49:16 2003
Received: from www1.ietf.org (ietf.org [132.151.1.19] (may be forged))
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA19370
	for <sipping-archive@odin.ietf.org>; Wed, 5 Feb 2003 06:49:16 -0500 (EST)
Received: (from mailnull@localhost)
	by www1.ietf.org (8.11.6/8.11.6) id h15BtCj18092
	for sipping-archive@odin.ietf.org; Wed, 5 Feb 2003 06:55:12 -0500
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	for <sipping@optimus.ietf.org>; Wed, 5 Feb 2003 06:51:25 -0500
Received: from CNRI.Reston.VA.US (localhost [127.0.0.1])
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA19104;
	Wed, 5 Feb 2003 06:44:57 -0500 (EST)
Message-Id: <200302051144.GAA19104@ietf.org>
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary="NextPart"
To: IETF-Announce: ;
Cc: sip@ietf.org, sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Wed, 05 Feb 2003 06:44:57 -0500
Subject: [Sipping] I-D ACTION:draft-ietf-sip-scvrtdisco-03.txt
Sender: sipping-admin@ietf.org
Errors-To: sipping-admin@ietf.org
X-BeenThere: sipping@ietf.org
X-Mailman-Version: 2.0.12
Precedence: bulk
List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=unsubscribe>
List-Id: SIPPING Working Group (applications of SIP) <sipping.ietf.org>
List-Post: <mailto:sipping@ietf.org>
List-Help: <mailto:sipping-request@ietf.org?subject=help>
List-Subscribe: <https://www1.ietf.org/mailman/listinfo/sipping>,
	<mailto:sipping-request@ietf.org?subject=subscribe>

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Session Initiation Protocol Extension Header Field for
                          Service Route Discovery During Registration
	Author(s)	: D. Willis, B. Hoeneisen
	Filename	: draft-ietf-sip-scvrtdisco-03.txt
	Pages		: 18
	Date		: 2003-2-4
	
This document defines a SIP extension header field used in
conjunction with responses to REGISTER requests to provide a
mechanism by which a registrar may inform a registering UA of a
service route that the UA may use to request outbound services from
the registrar's domain.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-scvrtdisco-03.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sip-scvrtdisco-03.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-sip-scvrtdisco-03.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<2003-2-4134059.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-sip-scvrtdisco-03.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-sip-scvrtdisco-03.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-2-4134059.I-D@ietf.org>

--OtherAccess--

--NextPart--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb  5 14:57:31 2003
Received: from www1.ietf.org (ietf.org [132.151.1.19] (may be forged))
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA08340
	for <sipping-archive@odin.ietf.org>; Wed, 5 Feb 2003 14:57:31 -0500 (EST)
Received: (from mailnull@localhost)
	by www1.ietf.org (8.11.6/8.11.6) id h15K3aj18609
	for sipping-archive@odin.ietf.org; Wed, 5 Feb 2003 15:03:36 -0500
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From: "Garland Sharratt" <gsharratt@convedia.com>
To: "Eric Burger" <eburger@snowshore.com>
Cc: "Sipping@Ietf. Org" <sipping@ietf.org>
Subject: RE: [Sipping] Updated Version of MSCML
Date: Wed, 5 Feb 2003 11:58:39 -0800
Keywords: .aaa
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Eric:

I took a look at draft-vandyke-mscml-01 and saw no substantive changes.

MSCML isn't perfect but my only real concern with it (both 01 and 00) is
the fact that SnowShore has claimed IPR on it.  This doesn't work for an
interface that is supposed to be -- and otherwise likely would be --
adopted by all call agent and media server vendors who need its type of
media server control.

I did notice that the following paragraph in 00 --

   "The IETF has been notified of intellectual property rights claimed
in regard to some or all of the specification contained in this
document.  For more information consult the online list of claimed
rights."

-- has been removed in 01.  But the SnowShore notice on the IETF Page of
Intellectual Property Rights Notices is still present ("SnowShore Patent
Statement pertaining to draft-vandyke-mscml").

Is there any chance of getting MSCML released into the public domain so
that it can be widely and safely adopted by everyone?

Thanks,

Garland


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Eric Burger
Sent: Monday, February 03, 2003 11:54
To: Sipping (E-mail)
Subject: [Sipping] Updated Version of MSCML


Until it hits the archive:

http://flyingfox.snowshore.com/i-d/sip/draft-vandyke-mscml-01.html
http://flyingfox.snowshore.com/i-d/sip/draft-vandyke-mscml-01.txt
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Folks,

I am updating the design team web pages for SIPPING and SIP.
http://www.softarmor.com/sipwg/teams
http://www.softarmor.com/sipping/teams/

I intend to consider the following design teams terminated:

SIP MIB 
SIP Security 
PacketCable DCS Convergence 
Call Control 
SIP-T
Call Flow 
SIP-H.323 


If somebody believes that any of these design teams is alive and,
therefore, it should not be considered terminated, let me know.
Otherwise, I will "execute" all of them.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Authentication, Authorization and Accounting Requirements 
			  for the Session Initiation Protocol
	Author(s)	: J. Loughney, G. Camarillo
	Filename	: draft-ietf-sipping-aaa-req-02.txt,.pdf
	Pages		: 14
	Date		: 2003-2-5
	
As SIP services are deployed on the Internet, there is a need for
authentication, authorization and accounting of SIP sessions. This
document sets out the basic requirements for this work.

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From mailnull@www1.ietf.org  Fri Feb  7 08:34:07 2003
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gonzalo,

please don't terminate the SIP MIB team. 
we are still here, but have been having
a hard time devoting time to a new revision (-05)
of the I-D.   we are working on it and 
we are targetting the end of the month to publish.

the previous version of the draft (-04) has 
expired.  that version had passed last call in sip wg.
we took a moment to consider updating the mib to
comply with rfc3261 - as it was riddled with references
to 2543 at that time - before forwarding to iesg.  
we decided it prudent and thus the need for -05.   
however, other project priorities and some personal 
issues with the team have made it difficult to devote 
time/resources to -05 of the draft,  but i hope that 
we are moving forward at this point.

kevin
Gonzalo Camarillo wrote:
> 
> Folks,
> 
> I am updating the design team web pages for SIPPING and SIP.
> http://www.softarmor.com/sipwg/teams
> http://www.softarmor.com/sipping/teams/
> 
> I intend to consider the following design teams terminated:
> 
> SIP MIB
> SIP Security
> PacketCable DCS Convergence
> Call Control
> SIP-T
> Call Flow
> SIP-H.323
> 
> If somebody believes that any of these design teams is alive and,
> therefore, it should not be considered terminated, let me know.
> Otherwise, I will "execute" all of them.
> 
> Thanks,
> 
> Gonzalo
> _______________________________________________
> 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

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
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Thread-Topic: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
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<P><FONT SIZE=3D2>Hi,<BR>
<BR>
posting WRT <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR>
<BR>
<BR>
&gt;&gt;<BR>
&gt;&gt; Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd prefer<BR>
&gt;&gt; that folks implement a translation to SIP, where SIP can =
natively carry<BR>
&gt;&gt; all the information that DPNSS is trying to convey.<BR>
&gt;&gt;<BR>
<BR>
To put things in perspective:<BR>
<BR>
1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and =
parts of NA) very similar to QSIG (deployed primarily Europe) for =
interconnecting PBXs or centrex customers hosted off LEs. DPNSS (like =
QSIG) finds enormous application towards realizing a local VPN or =
network-wide VPN services.<BR>
<BR>
2- This draft aims to do exactly what RFC 3204 (MIME objects for =
QSIG/ISUP) does for QSIG with appropriate/relevant customization for =
DPNSS - i.e., to provide end-to-end &amp; *complete*, DPNSS Service =
transparency between pockets of DPNSS based private networks over a SIP =
based core.<BR>
<BR>
3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS =
tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant translation/interworking mechanisms to ISUP. The =
translation/interworking mechanism to ISUP is limited to the basic call =
and certain ISUP supplementary services where there is a direct/easy =
equivalent.<BR>
<BR>
4- H.323 also provides equivalent tunneling mechanisms (via the H.450 =
Annexe procedures) for transparently carrying DSS1 based signaling =
between DSS1-based network elements/gateways, over a H.323 core.<BR>
<BR>
Some specifics:<BR>
<BR>
1- DPNSS supports, as specified in BTNR 188, approximately 240+ =
supplementary features/services (like Call Back When Free, Call Back =
When Next Used, Executive Intrusion, Travelling Class of Service, Do Not =
Disturb, Night Service, Private Network Route Optimization, Centralized =
Operator, Three Party Takeover, Remote Diversion Registration etc). It =
would be almost impossible to interwork to equivalent mechanisms in SIP, =
without losing the service transparency when using an intervening SIP =
based core.<BR>
<BR>
2- Yes, we definitely need an interworking/translation mechanism to map =
all the basic call/service paremeters, and procedures for mapping the =
messages. But this does not preclude the need for a DPNSS MIME type, but =
would rather be complimentary to the MIME type mechanism. Similar to <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A> , for QSIG, we are in the process of putting together an =
interworking draft for DPNSS as well - but rather than making it DPNSS =
specific, we are trying to make it a common draft to cover PRI, DPNSS =
&amp; DASS2 [as Q.SIG interworking is already a SIPPING WG ID]. Note =
that, PRI and DASS2 will not need a MIME type as these are, as one might =
be aware, for PSTN access, unlike DPNSS and QSIG (which are private =
network protocols).<BR>
<BR>
3- From the SIP perspective, carrying non-SIP applications as MIME =
objects to ensure end to end service transparency is architecturally =
less appalling than, say for e.g., tunneling DSS1 protocols in H.323 =
based network - so the SIP DPNSS MIME proposition as per this I-D is the =
obvious way forward for DPNSS service transparency over a SIP based =
core, especially now that we see acute deployment needs to be met.<BR>
<BR>
cheers,<BR>
ranjith..<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On Behalf<BR>
&gt; Of<BR>
&gt; Ranjith Mukundan<BR>
&gt; Sent: Tuesday, February 04, 2003 5:32 PM<BR>
&gt; To: sipping@ietf.org<BR>
&gt; Subject: RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Hi List,<BR>
&gt; We would like you take a look at this draft and revert with any<BR>
&gt; comments/queries that you might have.<BR>
&gt;<BR>
&gt; SIPPING Chairs,<BR>
&gt; The authors feel that the requirement and the need for this MIME =
type<BR>
&gt; is<BR>
&gt; quite acute. Hence we'd like you to consider this for inclusion as =
a<BR>
&gt; WG I-D<BR>
&gt; and suggest the way forward.<BR>
&gt;<BR>
&gt; - ranjith..<BR>
&gt;<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On Behalf<BR>
&gt; Of<BR>
&gt; Internet-Drafts@ietf.org<BR>
&gt; Sent: Tuesday, February 04, 2003 5:14 PM<BR>
&gt; To: IETF-Announce:<BR>
&gt; Cc: sipping@ietf.org<BR>
&gt; Subject: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; A New Internet-Draft is available from the on-line =
Internet-Drafts<BR>
&gt; directories.<BR>
&gt;<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME media types for DPNSS =
Objects<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-mukundan-sipping-dpnss-01.txt<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 9<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-3<BR>
&gt;<BR>
&gt; This document describes Multipart Internet Mail Extensions (MIME) =
type<BR>
&gt; for application/Digital Private Network Signaling System =
(DPNSS)<BR>
&gt; objects for use in Session Initiation Protocol (SIP) =
applications,<BR>
&gt; according to the rules defined in RFC 2048. These types can be used =
to<BR>
&gt; identify Digital Private Network Signaling System 1 (DPNSS 1) =
objects<BR>
&gt; within a SIP message such as INVITE or INFO, as might be =
implemented<BR>
&gt; when using SIP in an environment where part of the call =
involves<BR>
&gt; interworking to DPNSS 1 based Private Networks. RFC 3204 addresses =
a<BR>
&gt; similar need for Integrated Service Digital Network User Part =
(ISUP)<BR>
&gt; and Q Signaling (QSIG).<BR>
&gt;<BR>
&gt; A URL for this Internet-Draft is:<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A><BR>
&gt; t<BR>
&gt;<BR>
&gt; To remove yourself from the IETF Announcement list, send a message =
to<BR>
&gt; ietf-announce-request with the word unsubscribe in the body of =
the<BR>
&gt; message.<BR>
&gt;<BR>
&gt; Internet-Drafts are also available by anonymous FTP. Login with =
the<BR>
&gt; username<BR>
&gt; &quot;anonymous&quot; and a password of your e-mail address. After =
logging in,<BR>
&gt; type &quot;cd internet-drafts&quot; and then<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; A list of Internet-Drafts directories can be found in<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A> or<BR>
&gt; <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A><BR>
&gt;<BR>
&gt;<BR>
&gt; Internet-Drafts can also be obtained by e-mail.<BR>
&gt;<BR>
&gt; Send a message to:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mailserv@ietf.org.<BR>
&gt; In the body type:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;FILE =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; NOTE: The mail server at ietf.org can return the document in<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
&quot;mpack&quot; utility.&nbsp; To use this<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, insert the command =
&quot;ENCODING mime&quot; before the &quot;FILE&quot;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you will need &quot;munpack&quot; or<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant mail reader.&nbsp; =
Different MIME-compliant mail readers<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, =
especially when dealing with<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;multipart&quot; MIME messages =
(i.e. documents which have been split<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so check =
your local documentation on<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these =
messages.<BR>
&gt;<BR>
&gt;<BR>
&gt; Below is the data which will enable a MIME compliant mail =
reader<BR>
&gt; implementation to automatically retrieve the ASCII version of =
the<BR>
&gt; Internet-Draft.<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A><BR>
&gt; This list is for NEW development of the application of SIP Use<BR>
&gt; sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>
&gt; sip@ietf.org for new developments of core SIP<BR>
<BR>
<BR>
</FONT>
</P>

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From: "Veda Hegde" <vhegde@wave3software.com>
To: <sipping@ietf.org>
Date: Fri, 7 Feb 2003 11:50:32 -0800
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Subject: [Sipping] dynamic payload type
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Hi:
Section 8.3.2 in RFC 3264 gives an example about the dynamic RTP map.
 
========================================================================
================
However, in the case of RTP, the mapping from a particular dynamic
payload type
number to a particular codec within that media stream MUST NOT change
for the 
duration of a session.  For example, if A generates an offer with G.711
assigned 
to dynamic payload type number 46, payload type number 46 MUST refer to
G.711 from 
that point forward in any offers or answers for that media stream within
the session.  
However, it is acceptable for multiple payload type numbers to be mapped
to the same
codec, so that an updated offer could also use payload type number 72
for G.711.
========================================================================
=================
 
 
Is it allowed to use 46 for G.723 in the next m line?  
 
Thanks,
Veda
 
 

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Hi:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Section 8.3.2 in RFC 3264 gives an example about the =
dynamic
RTP map.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p></=
span></font></p>

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>However, in the case of RTP, the mapping from =
a particular dynamic payload =
type<o:p></o:p></span></font></pre><pre><span
class=3DGramE><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>number</span></font></span> to a particular =
codec within that media stream MUST NOT change for the =
<o:p></o:p></pre><pre><span
class=3DGramE><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>duration</span></font></span> of a =
session.<span style=3D'mso-spacerun:yes'>&nbsp; </span>For example, if A =
generates an offer with G.711 assigned <o:p></o:p></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>to =
dynamic payload type number 46, payload type number 46 MUST refer to =
G.711 from <o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>that =
point forward in any offers or answers for that media stream within the =
session.<span style=3D'mso-spacerun:yes'>&nbsp; =
</span><o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>However, =
it is acceptable for multiple payload type numbers to be mapped to the =
same<o:p></o:p></span></font></pre><pre><span
class=3DGramE><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>codec</span></font></span>, so that an =
updated offer could also use payload type number 72 for =
G.711.<o:p></o:p></pre>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<o:p></o:p=
></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Is it allowed to use 46 for G.723 in the next m line? =
<span
style=3D'mso-spacerun:yes'>&nbsp;</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks,<br>
Veda<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

------=_NextPart_000_0013_01C2CE9F.21500750--


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From mailnull@www1.ietf.org  Mon Feb 10 01:10:04 2003
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Subject: RE: [sipping] draft-burger-sipping-netann-03.txt (Early media announcement)
Date: Mon, 10 Feb 2003 01:10:51 -0500
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Thread-Topic: [sipping] draft-burger-sipping-netann-03.txt (Early media announcement)
Thread-Index: AcLLn9NPTQVIgkqTT/+LT3TQ3hOD6AE+wdoQ
From: "Eric Burger" <eburger@snowshore.com>
To: <Pascal.Jalet@alcatel.fr>
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The idea of an early media announcement is that there is an implicit CANCEL.  That reasoning still holds for RFC 3261.  Do you have a better thought for a more appropriate result code?

In the case of the soft phone you mention, I would offer the soft phone is not compliant with RFC 3261.  It should tear down the dialog on any 4xx response.

> -----Original Message-----
> From: Pascal.Jalet@alcatel.fr [mailto:Pascal.Jalet@alcatel.fr]
> Sent: Monday, February 03, 2003 11:18 AM
> To: Eric Burger
> Cc: David.Rio@alcatel.fr; sipping@ietf.org
> Subject: [sipping] draft-burger-sipping-netann-03.txt (Early media
> announcement)
> 
> 
> 
> Eric,
> In early media announcement scenario, you suggest to send a 
> 487 REQUEST
> TERMINATED, once the anouncement is played.
> But in RFC 3261, the 487 response is said to be used only 
> when the previous
> request has been terminated with a BYE or a CANCEL.
> It is not the case in early media  announcement scenario.
> So my question is ca:n we consider that the early media annoucement
> scenario is compliant with RFC 3261?
> 
> We've faced a case in which a softphone refuses to close a 
> call (screen
> still active) because it has received a 487 without any previous
> BYE or CANCEL. Is this terminal not compliant with 
> RFC3261?(below is the
> call flow of the call)
> 
> 
> 
> Softphone           PROXY (SER)             MS
>   INVITE ---------->          -------------->
>   <-----------------          ---------------183
>                       PLAY of 10 s
>   <------------------         ---------------487
>   ACK--------------->         ---------------->
> 
> After the ACK is sent, the call is still active.
> 
> 
> 
> pascal
> 
> 
> 
> 
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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CC: Dean Willis <dwillis@softarmor.com>, Rohan Mahy <rohan@cisco.com>,
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Subject: [Sipping] SIP-H.323 design team
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Folks,

The requirements document for SIP-H.323 interworking is advancing:

http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-reqs-03.txt

Now, it is time to start thinking about the solution document. If there
are enough people willing to contribute, we intend to form a design team
in SIPPING.

The scope of the work will be most likely limited to basic call set-up.

Those of you who are interested in being part of such a design team,
please let me know.

Thanks,

Gonzalo
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Subject: [Sipping] Expert review for draft-burger-sipping-netann-04.txt
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Folks,

We would like the SIPPING working group to provide expert review of:

http://www.ietf.org/internet-drafts/draft-burger-sipping-netann-04.txt

The authors have requested individual publication of this document. 
Expert Review will close on March 10th, 2003

Thanks,

Gonzalo
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Subject: [Sipping] New version of the early media draft
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Folks,

I have just sumitted a new version of the early media draft. Until it
appears in the archives, you can fetch it from:

http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-early-media-01.txt

I have included all the (long and useful) discussions we had in the
mailing list. Therefore, the draft now keeps separate the "gateway" and
the "application server" models. I have also removed the parts we did
not agree on.

Feedback is, as usual, appreciated.

Thanks,

Gonzalo
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sipping] New version of the SIP-AAA reqs draft
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Hi,

we have released a new version of the SIP-AAA requirements draft. Based
on the feedback we have received, it seems that the draft now includes
all the requirements people had in mind. (Although further comments are,
of course, always welcome).

We will now try to get the work on the diameter-based solution going.

Regards,

Gonzalo



-------- Original Message --------
Subject: [Sipping] I-D ACTION:draft-ietf-sipping-aaa-req-02.txt,.pdf
Date: Thu, 06 Feb 2003 09:30:09 -0500
From: Internet-Drafts@ietf.org
Reply-To: Internet-Drafts@ietf.org
To: IETF-Announce: ;
CC: sipping@ietf.org

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Proposal
Investigation Working Group of the IETF.

	Title		: Authentication, Authorization and Accounting Requirements 
			  for the Session Initiation Protocol
	Author(s)	: J. Loughney, G. Camarillo
	Filename	: draft-ietf-sipping-aaa-req-02.txt,.pdf
	Pages		: 14
	Date		: 2003-2-5
	
As SIP services are deployed on the Internet, there is a need for
authentication, authorization and accounting of SIP sessions. This
document sets out the basic requirements for this work.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-aaa-req-02.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the
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Internet-Drafts are also available by anonymous FTP. Login with the
username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sipping-aaa-req-02.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
CC: sipping <sipping@ietf.org>, Dean Willis <dwillis@softarmor.com>,
        Rohan Mahy <rohan@cisco.com>, Eric Burger <eburger@snowshore.com>
Subject: Re: [Sipping] Expert review for draft-burger-sipping-netann-04.txt
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Hi,

Some after-having-a-first-look-at-the-draft comments.

1.

The 487 Request Terminated response code is used after the announcement
has been played. There is already a discussion about this on the list, but
I also think it is an issue. I don't know which code would be better to
use, but RFC3261 does say that 487 is sent only when the request has been
terminated by CANCEL, and I am not sure I agree on the "implicit CANCEL"
thing... So, I think we should use another code. Either we come up with a
new one, or we use whatever existing (does it really matter that much
which code we use?).


2.

Chapter 3 says that 183, with SDP shall be sent, to enable early media. I
think it shall be said that 183 must be sent RELIABLE (using 100rel), if
the SDP is to provide some functionality.


3.

I don't like the "early" URI parameter. We should have a GENERIC early
media mechanism for SIP, and not come up with solutions for specific
applications.

Also, the UAC may not even know it will contact a media server. It may
send a "normal" SIP-URI, which is converted by a proxy into a ANNC-URL.
How does the proxy know if the UAC support early media or not?


Regards,

Christer Holmberg
Ericsson Finland




Gonzalo Camarillo wrote:

> Folks,
>
> We would like the SIPPING working group to provide expert review of:
>
> http://www.ietf.org/internet-drafts/draft-burger-sipping-netann-04.txt
>
> The authors have requested individual publication of this document.
> Expert Review will close on March 10th, 2003
>
> Thanks,
>
> Gonzalo
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "sipping" <sipping@ietf.org>
Cc: "Dean Willis" <dwillis@softarmor.com>, "Rohan Mahy" <rohan@cisco.com>,
        <kns10@cs.columbia.edu>, "Henning Schulzrinne" <hgs@cs.columbia.edu>,
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Gonzalo:

The design team of H.323-SIP Interworking has also the same members like
that of the requirements teams. We have provided the first draft long
time ago last year and was reviewed in the SIP-H.323 mailing list (later
on SIP Implementer's list). We are also working for the second draft.

At this point, I would also suggest, like requirements document, let
Henning Schulzrinne should be the lead editor for the SIP-H.323
interworking standard. I have the draft copy of the second draft of the
SIP-H.323 Interworking draft. I will send to all design team members so
that Henning can start editing the document.

Best regards,
Radhika R. Roy
rrroy@att.com

On behalf of the SIP-H.323 Design Team

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Monday, February 10, 2003 3:17 AM
To: sipping
Cc: Dean Willis; Rohan Mahy; kns10@cs.columbia.edu; Henning Schulzrinne;
alan.johnston@wcom.com; Roy, Radhika R, ALABS
Subject: SIP-H.323 design team


Folks,

The requirements document for SIP-H.323 interworking is advancing:

http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-
reqs-03.txt

Now, it is time to start thinking about the solution document. If there
are enough people willing to contribute, we intend to form a design team
in SIPPING.

The scope of the work will be most likely limited to basic call set-up.

Those of you who are interested in being part of such a design team,
please let me know.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Mon, 10 Feb 2003 20:19:29 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <3E47912B.CE58BFB5@lmf.ericsson.se>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'" <sipping@ietf.org>
Cc: "'John Loughney'" <john.loughney@nokia.com>
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> We will now try to get the work on the diameter-based solution going.

As has been mentioned here, 99.999% (five nines :-0>) of existing
implementations and deployment are RADIUS. After RADIUS come the web
services. 

I suggest therefore to start with RADIUS instead.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Gonzalo Camarillo
> Sent: Monday, February 10, 2003 5:47 AM
> To: sipping
> Cc: John Loughney
> Subject: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Hi,
> 
> we have released a new version of the SIP-AAA requirements 
> draft. Based on the feedback we have received, it seems that 
> the draft now includes all the requirements people had in 
> mind. (Although further comments are, of course, always welcome).
> 
> We will now try to get the work on the diameter-based solution going.
> 
> Regards,
> 
> Gonzalo
> 
> 
> 
> -------- Original Message --------
> Subject: [Sipping] I-D ACTION:draft-ietf-sipping-aaa-req-02.txt,.pdf
> Date: Thu, 06 Feb 2003 09:30:09 -0500
> From: Internet-Drafts@ietf.org
> Reply-To: Internet-Drafts@ietf.org
> To: IETF-Announce: ;
> CC: sipping@ietf.org
> 
> --NextPart
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories. This draft is a work item of the 
> Session Initiation Proposal Investigation Working Group of the IETF.
> 
> 	Title		: Authentication, Authorization and 
> Accounting Requirements 
> 			  for the Session Initiation Protocol
> 	Author(s)	: J. Loughney, G. Camarillo
> 	Filename	: draft-ietf-sipping-aaa-req-02.txt,.pdf
> 	Pages		: 14
> 	Date		: 2003-2-5
> 	
> As SIP services are deployed on the Internet, there is a need 
> for authentication, authorization and accounting of SIP 
> sessions. This document sets out the basic requirements for this work.
> 
> A URL for this Internet-Draft is: 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-aaa-req-02.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body 
> of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login 
> with the username "anonymous" and a password of your e-mail 
> address. After logging in, type "cd internet-drafts" and then
> 	"get draft-ietf-sipping-aaa-req-02.txt".
> 
> A list of Internet-Drafts directories can be found in 
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sipping-aaa-req-02.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant 
> mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail 
> reader implementation to automatically retrieve the ASCII 
> version of the Internet-Draft.
> 
> 
> --NextPart
> Content-Type: Multipart/Alternative; Boundary="OtherAccess"
> 
> --OtherAccess
> Content-Type: Message/External-body;
> 	access-type="mail-server";
> 	server="mailserv@ietf.org"
> 
> Content-Type: text/plain
> Content-ID:	<2003-2-5140318.I-D@ietf.org>
> 
> ENCODING mime
> FILE /internet-drafts/draft-ietf-sipping-aaa-req-02.txt
> 
> --OtherAccess
> Content-Type: Message/External-body;
> 	name="draft-ietf-sipping-aaa-req-02.txt";
> 	site="ftp.ietf.org";
> 	access-type="anon-ftp";
> 	directory="internet-drafts"
> 
> 
> --OtherAccess--
> 
> --NextPart--
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
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<P><FONT SIZE=3D2>we'd appreciate some views on this draft from atleast =
those who fairly understand QSIG.<BR>
<BR>
of course, others who feel that this MIME type collides directly with =
the purpose of SIP are encouraged to provide alternatives bearing in =
mind that providing complete end-to-end service transparency over the =
intervening SIP based core would be essential, if we are to provide a =
seamless migration path from H.323 based core networks - which already =
provide equivalent mechanisms for DSS1 based protocols.<BR>
<BR>
- ranjith..<BR>
<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Ranjith Mukundan<BR>
Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09 AM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
<BR>
Hi,<BR>
<BR>
posting WRT <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR>
<BR>
<BR>
&gt;&gt;<BR>
&gt;&gt; Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd prefer<BR>
&gt;&gt; that folks implement a translation to SIP, where SIP can =
natively carry<BR>
&gt;&gt; all the information that DPNSS is trying to convey.<BR>
&gt;&gt;<BR>
<BR>
To put things in perspective:<BR>
<BR>
1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and =
parts of NA) very similar to QSIG (deployed primarily Europe) for =
interconnecting PBXs or centrex customers hosted off LEs. DPNSS (like =
QSIG) finds enormous application towards realizing a local VPN or =
network-wide VPN services.<BR>
<BR>
2- This draft aims to do exactly what RFC 3204 (MIME objects for =
QSIG/ISUP) does for QSIG with appropriate/relevant customization for =
DPNSS - i.e., to provide end-to-end &amp; *complete*, DPNSS Service =
transparency between pockets of DPNSS based private networks over a SIP =
based core.<BR>
<BR>
3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS =
tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant translation/interworking mechanisms to ISUP. The =
translation/interworking mechanism to ISUP is limited to the basic call =
and certain ISUP supplementary services where there is a direct/easy =
equivalent.<BR>
<BR>
4- H.323 also provides equivalent tunneling mechanisms (via the H.450 =
Annexe procedures) for transparently carrying DSS1 based signaling =
between DSS1-based network elements/gateways, over a H.323 core.<BR>
<BR>
Some specifics:<BR>
<BR>
1- DPNSS supports, as specified in BTNR 188, approximately 240+ =
supplementary features/services (like Call Back When Free, Call Back =
When Next Used, Executive Intrusion, Travelling Class of Service, Do Not =
Disturb, Night Service, Private Network Route Optimization, Centralized =
Operator, Three Party Takeover, Remote Diversion Registration etc). It =
would be almost impossible to interwork to equivalent mechanisms in SIP, =
without losing the service transparency when using an intervening SIP =
based core.<BR>
<BR>
2- Yes, we definitely need an interworking/translation mechanism to map =
all the basic call/service paremeters, and procedures for mapping the =
messages. But this does not preclude the need for a DPNSS MIME type, but =
would rather be complimentary to the MIME type mechanism. Similar to <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A> , for QSIG, we are in the process of putting together an =
interworking draft for DPNSS as well - but rather than making it DPNSS =
specific, we are trying to make it a common draft to cover PRI, DPNSS =
&amp; DASS2 [as Q.SIG interworking is already a SIPPING WG ID]. Note =
that, PRI and DASS2 will not need a MIME type as these are, as one might =
be aware, for PSTN access, unlike DPNSS and QSIG (which are private =
network protocols).<BR>
<BR>
3- From the SIP perspective, carrying non-SIP applications as MIME =
objects to ensure end to end service transparency is architecturally =
less appalling than, say for e.g., tunneling DSS1 protocols in H.323 =
based network - so the SIP DPNSS MIME proposition as per this I-D is the =
obvious way forward for DPNSS service transparency over a SIP based =
core, especially now that we see acute deployment needs to be met.<BR>
<BR>
cheers,<BR>
ranjith..<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On Behalf<BR>
&gt; Of<BR>
&gt; Ranjith Mukundan<BR>
&gt; Sent: Tuesday, February 04, 2003 5:32 PM<BR>
&gt; To: sipping@ietf.org<BR>
&gt; Subject: RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Hi List,<BR>
&gt; We would like you take a look at this draft and revert with any<BR>
&gt; comments/queries that you might have.<BR>
&gt;<BR>
&gt; SIPPING Chairs,<BR>
&gt; The authors feel that the requirement and the need for this MIME =
type<BR>
&gt; is<BR>
&gt; quite acute. Hence we'd like you to consider this for inclusion as =
a<BR>
&gt; WG I-D<BR>
&gt; and suggest the way forward.<BR>
&gt;<BR>
&gt; - ranjith..<BR>
&gt;<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On Behalf<BR>
&gt; Of<BR>
&gt; Internet-Drafts@ietf.org<BR>
&gt; Sent: Tuesday, February 04, 2003 5:14 PM<BR>
&gt; To: IETF-Announce:<BR>
&gt; Cc: sipping@ietf.org<BR>
&gt; Subject: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; A New Internet-Draft is available from the on-line =
Internet-Drafts<BR>
&gt; directories.<BR>
&gt;<BR>
&gt;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME media types for DPNSS =
Objects<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-mukundan-sipping-dpnss-01.txt<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 9<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-3<BR>
&gt;<BR>
&gt; This document describes Multipart Internet Mail Extensions (MIME) =
type<BR>
&gt; for application/Digital Private Network Signaling System =
(DPNSS)<BR>
&gt; objects for use in Session Initiation Protocol (SIP) =
applications,<BR>
&gt; according to the rules defined in RFC 2048. These types can be used =
to<BR>
&gt; identify Digital Private Network Signaling System 1 (DPNSS 1) =
objects<BR>
&gt; within a SIP message such as INVITE or INFO, as might be =
implemented<BR>
&gt; when using SIP in an environment where part of the call =
involves<BR>
&gt; interworking to DPNSS 1 based Private Networks. RFC 3204 addresses =
a<BR>
&gt; similar need for Integrated Service Digital Network User Part =
(ISUP)<BR>
&gt; and Q Signaling (QSIG).<BR>
&gt;<BR>
&gt; A URL for this Internet-Draft is:<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A><BR>
&gt; t<BR>
&gt;<BR>
&gt; To remove yourself from the IETF Announcement list, send a message =
to<BR>
&gt; ietf-announce-request with the word unsubscribe in the body of =
the<BR>
&gt; message.<BR>
&gt;<BR>
&gt; Internet-Drafts are also available by anonymous FTP. Login with =
the<BR>
&gt; username<BR>
&gt; &quot;anonymous&quot; and a password of your e-mail address. After =
logging in,<BR>
&gt; type &quot;cd internet-drafts&quot; and then<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; A list of Internet-Drafts directories can be found in<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A> or<BR>
&gt; <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A><BR>
&gt;<BR>
&gt;<BR>
&gt; Internet-Drafts can also be obtained by e-mail.<BR>
&gt;<BR>
&gt; Send a message to:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mailserv@ietf.org.<BR>
&gt; In the body type:<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;FILE =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; NOTE: The mail server at ietf.org can return the document in<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
&quot;mpack&quot; utility.&nbsp; To use this<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, insert the command =
&quot;ENCODING mime&quot; before the &quot;FILE&quot;<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you will need &quot;munpack&quot; or<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant mail reader.&nbsp; =
Different MIME-compliant mail readers<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, =
especially when dealing with<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;multipart&quot; MIME messages =
(i.e. documents which have been split<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so check =
your local documentation on<BR>
&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these =
messages.<BR>
&gt;<BR>
&gt;<BR>
&gt; Below is the data which will enable a MIME compliant mail =
reader<BR>
&gt; implementation to automatically retrieve the ASCII version of =
the<BR>
&gt; Internet-Draft.<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A><BR>
&gt; This list is for NEW development of the application of SIP Use<BR>
&gt; sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>
&gt; sip@ietf.org for new developments of core SIP<BR>
<BR>
<BR>
<BR>
<BR>
</FONT>
</P>

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Early Media and Ringback Tone Generation in the 
                          Session Initiation Protocol
	Author(s)	: G. Camarillo, H. Schulzrinne
	Filename	: draft-camarillo-sipping-early-media-01.txt
	Pages		: 10
	Date		: 2003-2-10
	
This document describes how to manage early media in SIP using two
models; the gateway model and the application server model. It also
describes which inputs need to be taken into consideration to define
local policies for ringback tone generation.

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Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Tue, 11 Feb 2003 16:58:21 +0200
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Thread-Topic: [Sipping] New version of the SIP-AAA reqs draft
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To: <Henry.Sinnreich@wcom.com>, <Gonzalo.Camarillo@lmf.ericsson.se>,
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Hi Henry,


> > We will now try to get the work on the diameter-based 
> solution going.
> 
> As has been mentioned here, 99.999% (five nines :-0>) of existing
> implementations and deployment are RADIUS. After RADIUS come the web
> services. 
> 
> I suggest therefore to start with RADIUS instead.

RADIUS is extremely limited in a number of cases - security, transport
issues, reliability and is most likely illegal in many jurisdictions
for use with token-based accounting (due to the non-reliable nature
of RADIUS).  

What I think you are more concerned for the solution to work with
RADIUS, correct?  It may be pre-mature to discuss solutions, but
one option is develop a solution using Diameter in RADIUS-compatibility
mode.  The alert reader then may be able to determine how to use RADIUS
in such circumstances.  However, it is unlikely that the IESG will
approve using RADIUS for SIP-AAA needs because of the well-documented
short comings of RADIUS.

br,
John
_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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X-Sybari-Space: 00000000 00000000 00000000
From: "Mark Watson" <mwatson@nortelnetworks.com>
To: sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Tue, 11 Feb 2003 17:29:06 -0000
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All,
 
I really can't see any difference between DPNSS and Q.SIG when it comes to
the need for a MIME type for carriage of this within SIP. Can anyone come up
with an argument for/against this draft which does not apply to Q.SIG just
as well ?
 
If not, then I certainly think this should be progressed.
 
I have some comments on the draft:
1) DPNSS has now been published by the UK's Network Interoperability
Consultative Committee as NICC Document 1301:2001/03. You can find it at
http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm
<http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.ht
m>  along with lots of other gems.
 
You should reference this version, rather than the somewhat harder to obtain
BTNR.
 
2) Why do you need a length octet at the front of the message ? Can't you
(or shouldn't you) put a Content-Length in the MIME headers for this MIME
part ?
3) I think given the short length of the messages you could get away with
mandating that segmented DPNSS messages should always be reassembled before
encapsulating in SIP and that receivers must be able to receive such
reassembled messages. This removes some unnecessary complexity.
4) Section 3, under table 1, you say 'Floating Point' when you mean
'Decimal'. I think the words 'contiguously increasing' could probably be
removed here too. You will have trouble finding the next real number after
6.00!
 
Regards...Mark
 

-----Original Message-----
From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
Sent: 11 February 2003 08:59
To: sipping@ietf.org
Subject: [Sipping] QSIG Vs DPNSS MIME



we'd appreciate some views on this draft from atleast those who fairly
understand QSIG.

of course, others who feel that this MIME type collides directly with the
purpose of SIP are encouraged to provide alternatives bearing in mind that
providing complete end-to-end service transparency over the intervening SIP
based core would be essential, if we are to provide a seamless migration
path from H.323 based core networks - which already provide equivalent
mechanisms for DSS1 based protocols.

- ranjith..


-----Original Message-----
From:   Ranjith Mukundan
Sent:   Sat 2/8/2003 12:09 AM
To:     sipping@ietf.org
Cc:    
Subject:        RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt

Hi,

posting WRT
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt> 


>>
>> Why is it so important to be able to tunnel DPNSS in SIP?  I'd prefer
>> that folks implement a translation to SIP, where SIP can natively carry
>> all the information that DPNSS is trying to convey.
>>

To put things in perspective:

1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and parts of
NA) very similar to QSIG (deployed primarily Europe) for interconnecting
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous
application towards realizing a local VPN or network-wide VPN services.

2- This draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP)
does for QSIG with appropriate/relevant customization for DPNSS - i.e., to
provide end-to-end & *complete*, DPNSS Service transparency between pockets
of DPNSS based private networks over a SIP based core.

3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS
tunneling mechanisms through (ETSI & ANSI) ISUP *apart from* the concomitant
translation/interworking mechanisms to ISUP. The translation/interworking
mechanism to ISUP is limited to the basic call and certain ISUP
supplementary services where there is a direct/easy equivalent.

4- H.323 also provides equivalent tunneling mechanisms (via the H.450 Annexe
procedures) for transparently carrying DSS1 based signaling between
DSS1-based network elements/gateways, over a H.323 core.

Some specifics:

1- DPNSS supports, as specified in BTNR 188, approximately 240+
supplementary features/services (like Call Back When Free, Call Back When
Next Used, Executive Intrusion, Travelling Class of Service, Do Not Disturb,
Night Service, Private Network Route Optimization, Centralized Operator,
Three Party Takeover, Remote Diversion Registration etc). It would be almost
impossible to interwork to equivalent mechanisms in SIP, without losing the
service transparency when using an intervening SIP based core.

2- Yes, we definitely need an interworking/translation mechanism to map all
the basic call/service paremeters, and procedures for mapping the messages.
But this does not preclude the need for a DPNSS MIME type, but would rather
be complimentary to the MIME type mechanism. Similar to
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt>  ,
for QSIG, we are in the process of putting together an interworking draft
for DPNSS as well - but rather than making it DPNSS specific, we are trying
to make it a common draft to cover PRI, DPNSS & DASS2 [as Q.SIG interworking
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME
type as these are, as one might be aware, for PSTN access, unlike DPNSS and
QSIG (which are private network protocols).

3- From the SIP perspective, carrying non-SIP applications as MIME objects
to ensure end to end service transparency is architecturally less appalling
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so the
SIP DPNSS MIME proposition as per this I-D is the obvious way forward for
DPNSS service transparency over a SIP based core, especially now that we see
acute deployment needs to be met.

cheers,
ranjith..

> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Ranjith Mukundan
> Sent: Tuesday, February 04, 2003 5:32 PM
> To: sipping@ietf.org
> Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> Hi List,
> We would like you take a look at this draft and revert with any
> comments/queries that you might have.
>
> SIPPING Chairs,
> The authors feel that the requirement and the need for this MIME type
> is
> quite acute. Hence we'd like you to consider this for inclusion as a
> WG I-D
> and suggest the way forward.
>
> - ranjith..
>
> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Internet-Drafts@ietf.org
> Sent: Tuesday, February 04, 2003 5:14 PM
> To: IETF-Announce:
> Cc: sipping@ietf.org
> Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>       Title           : MIME media types for DPNSS Objects
>       Author(s)       : R. Mukundan et al.
>       Filename        : draft-mukundan-sipping-dpnss-01.txt
>       Pages           : 9
>       Date            : 2003-2-3
>
> This document describes Multipart Internet Mail Extensions (MIME) type
> for application/Digital Private Network Signaling System (DPNSS)
> objects for use in Session Initiation Protocol (SIP) applications,
> according to the rules defined in RFC 2048. These types can be used to
> identify Digital Private Network Signaling System 1 (DPNSS 1) objects
> within a SIP message such as INVITE or INFO, as might be implemented
> when using SIP in an environment where part of the call involves
> interworking to DPNSS 1 based Private Networks. RFC 3204 addresses a
> similar need for Integrated Service Digital Network User Part (ISUP)
> and Q Signaling (QSIG).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx> 
> t
>
> To remove yourself from the IETF Announcement list, send a message to
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>
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>
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>
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>       exhibit different behavior, especially when dealing with
>       "multipart" MIME messages (i.e. documents which have been split
>       up into multiple messages), so check your local documentation on
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>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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> This list is for NEW development of the application of SIP Use
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_______________________________________________ Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
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SIP 


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<TITLE>QSIG Vs DPNSS MIME</TITLE>

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<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003>All,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>I=20
really can't see any difference between DPNSS and Q.SIG when it comes =
to the=20
need for a MIME type for carriage of this within SIP. Can anyone come =
up with an=20
argument for/against this draft which does not apply to Q.SIG just as =
well=20
?</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>If=20
not, then I certainly think this should be =
progressed.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>I=20
have some comments on the draft:</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>1)=20
DPNSS has now been published by the UK's Network Interoperability =
Consultative=20
Committee as NICC Document 1301:2001/03. You can find it at <A=20
href=3D"http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstand=
ards/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnec=
tstandards/isc.htm</A>&nbsp;along=20
with lots of other gems.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>You=20
should reference this version, rather than the somewhat harder to =
obtain=20
BTNR.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>2)=20
Why do you need a length octet at the front of the message ? Can't you =
(or=20
shouldn't you) put a Content-Length in the MIME headers for this MIME =
part=20
?</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>3) I=20
think given the short length of the messages you could get away with =
mandating=20
that segmented DPNSS messages should always be reassembled before =
encapsulating=20
in SIP and that receivers must be able to receive such reassembled =
messages.=20
This removes some unnecessary complexity.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN =
class=3D145285416-11022003>4)=20
Section 3, under table 1, you say 'Floating Point' when you mean =
'Decimal'. I=20
think the words 'contiguously increasing' could probably be removed =
here too.=20
You will have trouble finding the next real number after=20
6.00!</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003>Regards...Mark</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DVerdana size=3D1><SPAN=20
class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; =
PADDING-LEFT: 5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Ranjith Mukundan=20
  [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 11 February 2003=20
  08:59<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] =
QSIG Vs=20
  DPNSS MIME<BR><BR></DIV></FONT><!-- Converted from text/plain format =
-->
  <P><FONT size=3D2>we'd appreciate some views on this draft from =
atleast those=20
  who fairly understand QSIG.<BR><BR>of course, others who feel that =
this MIME=20
  type collides directly with the purpose of SIP are encouraged to =
provide=20
  alternatives bearing in mind that providing complete end-to-end =
service=20
  transparency over the intervening SIP based core would be essential, =
if we are=20
  to provide a seamless migration path from H.323 based core networks - =
which=20
  already provide equivalent mechanisms for DSS1 based =
protocols.<BR><BR>-=20
  ranjith..<BR><BR><BR>-----Original Message-----<BR>From:&nbsp;&nbsp; =
Ranjith=20
  Mukundan<BR>Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09=20
  AM<BR>To:&nbsp;&nbsp;&nbsp;&nbsp;=20
  =
sipping@ietf.org<BR>Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>Subject:&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  RE: [Sipping] I-D=20
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR><BR>Hi,<BR><BR>posting =
WRT <A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpns=
s-01.txt</A><BR><BR><BR>&gt;&gt;<BR>&gt;&gt;=20
  Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; I'd=20
  prefer<BR>&gt;&gt; that folks implement a translation to SIP, where =
SIP can=20
  natively carry<BR>&gt;&gt; all the information that DPNSS is trying =
to=20
  convey.<BR>&gt;&gt;<BR><BR>To put things in perspective:<BR><BR>1- =
DPNSS is a=20
  protocol (deployed primarily in UK, NZ, Australia and parts of NA) =
very=20
  similar to QSIG (deployed primarily Europe) for interconnecting PBXs =
or=20
  centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous =
application=20
  towards realizing a local VPN or network-wide VPN services.<BR><BR>2- =
This=20
  draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP) =
does for=20
  QSIG with appropriate/relevant customization for DPNSS - i.e., to =
provide=20
  end-to-end &amp; *complete*, DPNSS Service transparency between =
pockets of=20
  DPNSS based private networks over a SIP based core.<BR><BR>3- =
Currently,=20
  circuit-switched SS7 based cores, provide equivalent DPNSS tunneling=20
  mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant=20
  translation/interworking mechanisms to ISUP. The =
translation/interworking=20
  mechanism to ISUP is limited to the basic call and certain ISUP =
supplementary=20
  services where there is a direct/easy equivalent.<BR><BR>4- H.323 =
also=20
  provides equivalent tunneling mechanisms (via the H.450 Annexe =
procedures) for=20
  transparently carrying DSS1 based signaling between DSS1-based =
network=20
  elements/gateways, over a H.323 core.<BR><BR>Some =
specifics:<BR><BR>1- DPNSS=20
  supports, as specified in BTNR 188, approximately 240+ supplementary=20
  features/services (like Call Back When Free, Call Back When Next =
Used,=20
  Executive Intrusion, Travelling Class of Service, Do Not Disturb, =
Night=20
  Service, Private Network Route Optimization, Centralized Operator, =
Three Party=20
  Takeover, Remote Diversion Registration etc). It would be almost =
impossible to=20
  interwork to equivalent mechanisms in SIP, without losing the service =

  transparency when using an intervening SIP based core.<BR><BR>2- Yes, =
we=20
  definitely need an interworking/translation mechanism to map all the =
basic=20
  call/service paremeters, and procedures for mapping the messages. But =
this=20
  does not preclude the need for a DPNSS MIME type, but would rather be =

  complimentary to the MIME type mechanism. Similar to <A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-=
00.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-=
00.txt</A>=20
  , for QSIG, we are in the process of putting together an interworking =
draft=20
  for DPNSS as well - but rather than making it DPNSS specific, we are =
trying to=20
  make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG =
interworking=20
  is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a =
MIME=20
  type as these are, as one might be aware, for PSTN access, unlike =
DPNSS and=20
  QSIG (which are private network protocols).<BR><BR>3- From the SIP=20
  perspective, carrying non-SIP applications as MIME objects to ensure =
end to=20
  end service transparency is architecturally less appalling than, say =
for e.g.,=20
  tunneling DSS1 protocols in H.323 based network - so the SIP DPNSS =
MIME=20
  proposition as per this I-D is the obvious way forward for DPNSS =
service=20
  transparency over a SIP based core, especially now that we see acute=20
  deployment needs to be met.<BR><BR>cheers,<BR>ranjith..<BR><BR>&gt;=20
  -----Original Message-----<BR>&gt; From: sipping-admin@ietf.org [<A=20
  =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
]On=20
  Behalf<BR>&gt; Of<BR>&gt; Ranjith Mukundan<BR>&gt; Sent: Tuesday, =
February 04,=20
  2003 5:32 PM<BR>&gt; To: sipping@ietf.org<BR>&gt; Subject: RE: =
[Sipping] I-D=20
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; Hi =

  List,<BR>&gt; We would like you take a look at this draft and revert =
with=20
  any<BR>&gt; comments/queries that you might have.<BR>&gt;<BR>&gt; =
SIPPING=20
  Chairs,<BR>&gt; The authors feel that the requirement and the need =
for this=20
  MIME type<BR>&gt; is<BR>&gt; quite acute. Hence we'd like you to =
consider this=20
  for inclusion as a<BR>&gt; WG I-D<BR>&gt; and suggest the way=20
  forward.<BR>&gt;<BR>&gt; - ranjith..<BR>&gt;<BR>&gt; -----Original=20
  Message-----<BR>&gt; From: sipping-admin@ietf.org [<A=20
  =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
]On=20
  Behalf<BR>&gt; Of<BR>&gt; Internet-Drafts@ietf.org<BR>&gt; Sent: =
Tuesday,=20
  February 04, 2003 5:14 PM<BR>&gt; To: IETF-Announce:<BR>&gt; Cc:=20
  sipping@ietf.org<BR>&gt; Subject: [Sipping] I-D=20
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; A =
New=20
  Internet-Draft is available from the on-line Internet-Drafts<BR>&gt;=20
  directories.<BR>&gt;<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME =
media=20
  types for DPNSS Objects<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et =
al.<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :=20
  draft-mukundan-sipping-dpnss-01.txt<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-3<BR>&gt;<BR>&gt; =
This=20
  document describes Multipart Internet Mail Extensions (MIME) =
type<BR>&gt; for=20
  application/Digital Private Network Signaling System (DPNSS)<BR>&gt; =
objects=20
  for use in Session Initiation Protocol (SIP) applications,<BR>&gt; =
according=20
  to the rules defined in RFC 2048. These types can be used to<BR>&gt; =
identify=20
  Digital Private Network Signaling System 1 (DPNSS 1) objects<BR>&gt; =
within a=20
  SIP message such as INVITE or INFO, as might be implemented<BR>&gt; =
when using=20
  SIP in an environment where part of the call involves<BR>&gt; =
interworking to=20
  DPNSS 1 based Private Networks. RFC 3204 addresses a<BR>&gt; similar =
need for=20
  Integrated Service Digital Network User Part (ISUP)<BR>&gt; and Q =
Signaling=20
  (QSIG).<BR>&gt;<BR>&gt; A URL for this Internet-Draft is:<BR>&gt; <A=20
  =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.tx</A><BR>&gt;=20
  t<BR>&gt;<BR>&gt; To remove yourself from the IETF Announcement list, =
send a=20
  message to<BR>&gt; ietf-announce-request with the word unsubscribe in =
the body=20
  of the<BR>&gt; message.<BR>&gt;<BR>&gt; Internet-Drafts are also =
available by=20
  anonymous FTP. Login with the<BR>&gt; username<BR>&gt; "anonymous" =
and a=20
  password of your e-mail address. After logging in,<BR>&gt; type "cd=20
  internet-drafts" and then<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "get =

  draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; A list of=20
  Internet-Drafts directories can be found in<BR>&gt; <A=20
  =
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html=
</A>=20
  or<BR>&gt; <A=20
  =
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ie=
tf/1shadow-sites.txt</A><BR>&gt;<BR>&gt;<BR>&gt;=20
  Internet-Drafts can also be obtained by e-mail.<BR>&gt;<BR>&gt; Send =
a message=20
  to:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mailserv@ietf.org.<BR>&gt; =
In the=20
  body type:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "FILE=20
  =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; =
NOTE:=20
  The mail server at ietf.org can return the document in<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the "mpack" =

  utility.&nbsp; To use this<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
feature,=20
  insert the command "ENCODING mime" before the "FILE"<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you=20
  will need "munpack" or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a =
MIME-compliant=20
  mail reader.&nbsp; Different MIME-compliant mail readers<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, especially =
when=20
  dealing with<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "multipart" MIME =
messages=20
  (i.e. documents which have been split<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  up into multiple messages), so check your local documentation =
on<BR>&gt;=20
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these=20
  messages.<BR>&gt;<BR>&gt;<BR>&gt; Below is the data which will enable =
a MIME=20
  compliant mail reader<BR>&gt; implementation to automatically =
retrieve the=20
  ASCII version of the<BR>&gt; Internet-Draft.<BR>&gt;<BR>&gt;=20
  _______________________________________________<BR>&gt; Sipping =
mailing=20
  list&nbsp; <A=20
  =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.iet=
f.org/mailman/listinfo/sipping</A><BR>&gt;=20
  This list is for NEW development of the application of SIP =
Use<BR>&gt;=20
  sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>&gt;=20
  sip@ietf.org for new developments of core=20
  =
SIP<BR><BR><BR><BR><BR></FONT></P>______________________________________=
_________=20
  Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping =
This list=20
  is for NEW development of the application of SIP Use=20
  sip-implementors@cs.columbia.edu for questions on current sip Use =
sip@ietf.org=20
  for new developments of core SIP </BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C2D1F3.1418CB24--
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] QSIG Vs DPNSS MIME
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<P><FONT SIZE=3D2>Mark,<BR>
<BR>
&gt;&gt; 2) Why do you need a length octet at the front of the message ? =
Can't you<BR>
&gt;&gt; (or shouldn't you) put a Content-Length in the MIME headers for =
this MIME<BR>
&gt;&gt; part ?<BR>
<BR>
[ranjith]<BR>
The MIME Content-length is for the entire MIME object and represents the =
cumulative lenght of the MIME content. There is no MIME length field =
specifically for one of the many potential MIME objects that could have =
been encapsulated. For e.g., in the examples illustrated in the draft, =
the content length would be indicative of the SDP and the DPNSS PDU put =
together. Given this, for implementation ease, the length field will be =
highly desirable. We are also drawing upon our implementation experience =
from an equivalent mechanism that is employed while carrying DPNSS in =
ANSI ISUP - where, a length field is (logically) prepended along the =
encapsulated DPNSS PDU.<BR>
<BR>
<BR>
&gt;&gt; 3) I think given the short length of the messages you could get =
away with<BR>
&gt;&gt; mandating that segmented DPNSS messages should always be =
reassembled before<BR>
&gt;&gt; encapsulating in SIP and that receivers must be able to receive =
such<BR>
&gt;&gt; reassembled messages. This removes some unnecessary =
complexity.<BR>
<BR>
[ranjith]<BR>
The carriage of segmented DPNSS messages was retained, albeit leading to =
inefficient network usage, as an *option* to primarily, accomodate =
existing DPNSS layer 3 gateway implementations, where the DPNSS message =
processing (either as transit or end-node function) is primarily based =
on segmented DPNSS messages. Even in DPNSS - H.323 gateway =
implementations that exist, segmented DPNSS messages are not =
re-assembled. In order to keep the migration (towards SIP based core) =
for the existing DPNSS gateway implementations seamless, it would indeed =
be better to retain carriage of segmented DPNSS messages as an *option* =
BUT *strongly recommend* to reassemble DPNSS messages before =
encapsulation in SIP. We'll modify the text in the draft =
accordingly.<BR>
<BR>
cheers,<BR>
- ranjith..<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Mark Watson [<A =
HREF=3D"mailto:mwatson@nortelnetworks.com">mailto:mwatson@nortelnetworks.=
com</A>]<BR>
Sent:&nbsp;&nbsp; Tue 2/11/2003 10:59 PM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] QSIG Vs =
DPNSS MIME<BR>
All,<BR>
<BR>
I really can't see any difference between DPNSS and Q.SIG when it comes =
to<BR>
the need for a MIME type for carriage of this within SIP. Can anyone =
come up<BR>
with an argument for/against this draft which does not apply to Q.SIG =
just<BR>
as well ?<BR>
<BR>
If not, then I certainly think this should be progressed.<BR>
<BR>
I have some comments on the draft:<BR>
1) DPNSS has now been published by the UK's Network Interoperability<BR>
Consultative Committee as NICC Document 1301:2001/03. You can find it =
at<BR>
<A =
HREF=3D"http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstanda=
rds/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnects=
tandards/isc.htm</A><BR>
&lt;http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/=
isc.ht<BR>
m&gt;&nbsp; along with lots of other gems.<BR>
<BR>
You should reference this version, rather than the somewhat harder to =
obtain<BR>
BTNR.<BR>
<BR>
2) Why do you need a length octet at the front of the message ? Can't =
you<BR>
(or shouldn't you) put a Content-Length in the MIME headers for this =
MIME<BR>
part ?<BR>
3) I think given the short length of the messages you could get away =
with<BR>
mandating that segmented DPNSS messages should always be reassembled =
before<BR>
encapsulating in SIP and that receivers must be able to receive such<BR>
reassembled messages. This removes some unnecessary complexity.<BR>
4) Section 3, under table 1, you say 'Floating Point' when you mean<BR>
'Decimal'. I think the words 'contiguously increasing' could probably =
be<BR>
removed here too. You will have trouble finding the next real number =
after<BR>
6.00!<BR>
<BR>
Regards...Mark<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Ranjith Mukundan [<A =
HREF=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro.=
com</A>]<BR>
Sent: 11 February 2003 08:59<BR>
To: sipping@ietf.org<BR>
Subject: [Sipping] QSIG Vs DPNSS MIME<BR>
<BR>
<BR>
<BR>
we'd appreciate some views on this draft from atleast those who =
fairly<BR>
understand QSIG.<BR>
<BR>
of course, others who feel that this MIME type collides directly with =
the<BR>
purpose of SIP are encouraged to provide alternatives bearing in mind =
that<BR>
providing complete end-to-end service transparency over the intervening =
SIP<BR>
based core would be essential, if we are to provide a seamless =
migration<BR>
path from H.323 based core networks - which already provide =
equivalent<BR>
mechanisms for DSS1 based protocols.<BR>
<BR>
- ranjith..<BR>
<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Ranjith Mukundan<BR>
Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09 AM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Cc:&nbsp;&nbsp;&nbsp;<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
<BR>
Hi,<BR>
<BR>
posting WRT<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A>&gt;<BR>
<BR>
<BR>
&gt;&gt;<BR>
&gt;&gt; Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd prefer<BR>
&gt;&gt; that folks implement a translation to SIP, where SIP can =
natively carry<BR>
&gt;&gt; all the information that DPNSS is trying to convey.<BR>
&gt;&gt;<BR>
<BR>
To put things in perspective:<BR>
<BR>
1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and =
parts of<BR>
NA) very similar to QSIG (deployed primarily Europe) for =
interconnecting<BR>
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds =
enormous<BR>
application towards realizing a local VPN or network-wide VPN =
services.<BR>
<BR>
2- This draft aims to do exactly what RFC 3204 (MIME objects for =
QSIG/ISUP)<BR>
does for QSIG with appropriate/relevant customization for DPNSS - i.e., =
to<BR>
provide end-to-end &amp; *complete*, DPNSS Service transparency between =
pockets<BR>
of DPNSS based private networks over a SIP based core.<BR>
<BR>
3- Currently, circuit-switched SS7 based cores, provide equivalent =
DPNSS<BR>
tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant<BR>
translation/interworking mechanisms to ISUP. The =
translation/interworking<BR>
mechanism to ISUP is limited to the basic call and certain ISUP<BR>
supplementary services where there is a direct/easy equivalent.<BR>
<BR>
4- H.323 also provides equivalent tunneling mechanisms (via the H.450 =
Annexe<BR>
procedures) for transparently carrying DSS1 based signaling between<BR>
DSS1-based network elements/gateways, over a H.323 core.<BR>
<BR>
Some specifics:<BR>
<BR>
1- DPNSS supports, as specified in BTNR 188, approximately 240+<BR>
supplementary features/services (like Call Back When Free, Call Back =
When<BR>
Next Used, Executive Intrusion, Travelling Class of Service, Do Not =
Disturb,<BR>
Night Service, Private Network Route Optimization, Centralized =
Operator,<BR>
Three Party Takeover, Remote Diversion Registration etc). It would be =
almost<BR>
impossible to interwork to equivalent mechanisms in SIP, without losing =
the<BR>
service transparency when using an intervening SIP based core.<BR>
<BR>
2- Yes, we definitely need an interworking/translation mechanism to map =
all<BR>
the basic call/service paremeters, and procedures for mapping the =
messages.<BR>
But this does not preclude the need for a DPNSS MIME type, but would =
rather<BR>
be complimentary to the MIME type mechanism. Similar to<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A>&gt;&nbsp; ,<BR>
for QSIG, we are in the process of putting together an interworking =
draft<BR>
for DPNSS as well - but rather than making it DPNSS specific, we are =
trying<BR>
to make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG =
interworking<BR>
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a =
MIME<BR>
type as these are, as one might be aware, for PSTN access, unlike DPNSS =
and<BR>
QSIG (which are private network protocols).<BR>
<BR>
3- From the SIP perspective, carrying non-SIP applications as MIME =
objects<BR>
to ensure end to end service transparency is architecturally less =
appalling<BR>
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so =
the<BR>
SIP DPNSS MIME proposition as per this I-D is the obvious way forward =
for<BR>
DPNSS service transparency over a SIP based core, especially now that we =
see<BR>
acute deployment needs to be met.<BR>
<BR>
cheers,<BR>
ranjith..<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Ranjith Mukundan<BR>
&gt; Sent: Tuesday, February 04, 2003 5:32 PM<BR>
&gt; To: sipping@ietf.org<BR>
&gt; Subject: RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Hi List,<BR>
&gt; We would like you take a look at this draft and revert with any<BR>
&gt; comments/queries that you might have.<BR>
&gt;<BR>
&gt; SIPPING Chairs,<BR>
&gt; The authors feel that the requirement and the need for this MIME =
type<BR>
&gt; is<BR>
&gt; quite acute. Hence we'd like you to consider this for inclusion as =
a<BR>
&gt; WG I-D<BR>
&gt; and suggest the way forward.<BR>
&gt;<BR>
&gt; - ranjith..<BR>
&gt;<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Internet-Drafts@ietf.org<BR>
&gt; Sent: Tuesday, February 04, 2003 5:14 PM<BR>
&gt; To: IETF-Announce:<BR>
&gt; Cc: sipping@ietf.org<BR>
&gt; Subject: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; A New Internet-Draft is available from the on-line =
Internet-Drafts<BR>
&gt; directories.<BR>
&gt;<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME =
media types for DPNSS Objects<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-2-3<BR>
&gt;<BR>
&gt; This document describes Multipart Internet Mail Extensions (MIME) =
type<BR>
&gt; for application/Digital Private Network Signaling System =
(DPNSS)<BR>
&gt; objects for use in Session Initiation Protocol (SIP) =
applications,<BR>
&gt; according to the rules defined in RFC 2048. These types can be used =
to<BR>
&gt; identify Digital Private Network Signaling System 1 (DPNSS 1) =
objects<BR>
&gt; within a SIP message such as INVITE or INFO, as might be =
implemented<BR>
&gt; when using SIP in an environment where part of the call =
involves<BR>
&gt; interworking to DPNSS 1 based Private Networks. RFC 3204 addresses =
a<BR>
&gt; similar need for Integrated Service Digital Network User Part =
(ISUP)<BR>
&gt; and Q Signaling (QSIG).<BR>
&gt;<BR>
&gt; A URL for this Internet-Draft is:<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A>&gt;<BR>
&gt; t<BR>
&gt;<BR>
&gt; To remove yourself from the IETF Announcement list, send a message =
to<BR>
&gt; ietf-announce-request with the word unsubscribe in the body of =
the<BR>
&gt; message.<BR>
&gt;<BR>
&gt; Internet-Drafts are also available by anonymous FTP. Login with =
the<BR>
&gt; username<BR>
&gt; &quot;anonymous&quot; and a password of your e-mail address. After =
logging in,<BR>
&gt; type &quot;cd internet-drafts&quot; and then<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; A list of Internet-Drafts directories can be found in<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A> &lt;<A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A>&gt;&nbsp; or<BR>
&gt; <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A><BR>
&lt;<A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A>&gt;<BR>
&gt;<BR>
&gt;<BR>
&gt; Internet-Drafts can also be obtained by e-mail.<BR>
&gt;<BR>
&gt; Send a message to:<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mailserv@ietf.org.<BR>
&gt; In the body type:<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;FILE =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; NOTE: The mail server at ietf.org can return the document in<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
&quot;mpack&quot; utility.&nbsp; To use this<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, insert the command =
&quot;ENCODING mime&quot; before the &quot;FILE&quot;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you will need &quot;munpack&quot; or<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant mail =
reader.&nbsp; Different MIME-compliant mail readers<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, =
especially when dealing with<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;multipart&quot; MIME =
messages (i.e. documents which have been split<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so =
check your local documentation on<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these =
messages.<BR>
&gt;<BR>
&gt;<BR>
&gt; Below is the data which will enable a MIME compliant mail =
reader<BR>
&gt; implementation to automatically retrieve the ASCII version of =
the<BR>
&gt; Internet-Draft.<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A><BR>
&lt;<A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A>&gt;<BR>
&gt; This list is for NEW development of the application of SIP Use<BR>
&gt; sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>
&gt; sip@ietf.org for new developments of core SIP<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
_______________________________________________ Sipping mailing list<BR>
<A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A> This list is for NEW<BR>
development of the application of SIP Use =
sip-implementors@cs.columbia.edu<BR>
for questions on current sip Use sip@ietf.org for new developments of =
core<BR>
SIP<BR>
<BR>
<BR>
<BR>
</FONT>
</P>

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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Tue, 11 Feb 2003 12:38:07 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <A16A3EE4D4CA124FADC7987B1AC89FE440ED07@esebe022.ntc.nokia.com>
To: john.loughney@nokia.com, Gonzalo.Camarillo@lmf.ericsson.se,
        sipping@ietf.org
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>but one option is develop a solution using 
> Diameter in RADIUS-compatibility mode.

OK, this is better than nothing. We'll take what we can :-)

As mentioned, the installed base is RADIUS, no matter how voluminous the
DIAMETER literature may be.

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of john.loughney@nokia.com
> Sent: Tuesday, February 11, 2003 8:58 AM
> To: Henry.Sinnreich@wcom.com; 
> Gonzalo.Camarillo@lmf.ericsson.se; sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Hi Henry,
> 
> 
> > > We will now try to get the work on the diameter-based
> > solution going.
> > 
> > As has been mentioned here, 99.999% (five nines :-0>) of existing 
> > implementations and deployment are RADIUS. After RADIUS 
> come the web 
> > services.
> > 
> > I suggest therefore to start with RADIUS instead.
> 
> RADIUS is extremely limited in a number of cases - security, 
> transport issues, reliability and is most likely illegal in 
> many jurisdictions for use with token-based accounting (due 
> to the non-reliable nature of RADIUS).  
> 
> What I think you are more concerned for the solution to work 
> with RADIUS, correct?  It may be pre-mature to discuss 
> solutions, but one option is develop a solution using 
> Diameter in RADIUS-compatibility mode.  The alert reader then 
> may be able to determine how to use RADIUS in such 
> circumstances.  However, it is unlikely that the IESG will 
> approve using RADIUS for SIP-AAA needs because of the 
> well-documented short comings of RADIUS.
> 
> br,
> John
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Tue, 11 Feb 2003 20:46:01 +0200
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Thread-Topic: [Sipping] New version of the SIP-AAA reqs draft
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To: <Henry.Sinnreich@wcom.com>, <Gonzalo.Camarillo@lmf.ericsson.se>,
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Hi Henry,

> >but one option is develop a solution using 
> > Diameter in RADIUS-compatibility mode.
> 
> OK, this is better than nothing. We'll take what we can :-)
> 
> As mentioned, the installed base is RADIUS, no matter how 
> voluminous the DIAMETER literature may be.

I think the issues is that the IETF should recommend what is the
right thing to do & why.  If people choose to do something else,
well, that is life.  

I understand what you are asking but I do note that some
people have thought that the potential for AAA is much greater
than the existing installed base.

br,
John
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re : [Sipping] QSIG Vs DPNSS MIME
Date: Wed, 12 Feb 2003 00:27:17 +0530
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Thread-Topic: Re : [Sipping] QSIG Vs DPNSS MIME
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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
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<P><FONT SIZE=3D2>John,<BR>
<BR>
<BR>
&gt;&gt; 4. During this work we found it necessary to specify that a =
QSIG-SIP gateway<BR>
&gt;&gt; should behave as a transit PBX from the perspective of the QSIG =
protocol.<BR>
<BR>
[ranjith]<BR>
Yes, as in the case of QSIG, the DPNSS-SIP gateway too is expected to =
behave as a 'transit' from the DPNSS point of view. Thats a good point =
you raised. We need to look harder at this aspect and incorporate the =
required text in the draft (esp regarding the transit timers etc).<BR>
<BR>
<BR>
&gt;&gt; 6. The message buffering option may not be suitable for =
buffering ISRM(I)<BR>
&gt;&gt; and SSRM(I) messages. These messages are not only used to =
overcome the 45<BR>
&gt;&gt; octet message limit but are also used for overlap sending. =
Consequently a<BR>
&gt;&gt; SSRM(C) will not necessarily be expected.<BR>
<BR>
[ranjith]<BR>
Agreed. The buffering option makes sense for segmented messages alone. =
Extending it to overlap signaling will unnecessarily complicate the =
gateway implementation and also defeats the purpose of overlap =
signaling. As you suggest, I guess we should restrict the buffering =
option to segmented messages alone.<BR>
<BR>
&gt;&gt; I suspect that you will wish to limit DPNSS tunnelling to<BR>
&gt;&gt; option 1 - otherwise you will need to carry additional =
information that is<BR>
&gt;&gt; normally available from DPNSS layer 2. This needs to be =
stated.<BR>
<BR>
[ranjith]<BR>
Currently, Yes. But you've raised an interesting point. We'll explore =
the possibility of option 2; unless it complicates the protocol aspects =
vis-a-vis the advantages.<BR>
<BR>
Thanks for the inputs.<BR>
<BR>
cheers,<BR>
- ranjith..<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Elwell, John [<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</A=
>]<BR>
Sent: Tuesday, February 11, 2003 4:59 PM<BR>
To: 'Ranjith Mukundan'<BR>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME<BR>
<BR>
<BR>
Ranjith,<BR>
<BR>
Just a few remarks.<BR>
<BR>
1. DPNSS is no longer published by BT. The latest version (version 7) =
is<BR>
DPNSS[188] published by Oftel sub-committee NICC (Network =
Interoperability<BR>
Consultative Committee). This has also has an internal reference =
number<BR>
ND1301:2001/03. The current URL is:<BR>
<BR>
<A =
HREF=3D"http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstanda=
rds/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnects=
tandards/isc.htm</A><BR>
&lt;http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/=
isc.ht<BR>
m&gt;<BR>
<BR>
but this is liable to change (e.g., Oftel will change to Ofcom) and =
should<BR>
not be quoted in any long-term document.<BR>
<BR>
I don't think previous versions (1 to 6) are available any more, but =
of<BR>
course there will be implementations. However, most people will be =
aware<BR>
only of the BTNRs, so it would still be worth mentioning &quot;formerly =
BTNR<BR>
188&quot;.<BR>
<BR>
2. If version number is to be carried, then version 7 should be =
added.<BR>
<BR>
3. For QSIG tunneling over SIP we found that the RFC specifying the =
MIME<BR>
type is insufficient for achieving correct operation in multi-vendor<BR>
networks. Therefore in ECMA we are working on an additional =
specification<BR>
that will detail procedures to be adopted at the QSIG-SIP gateway.<BR>
<BR>
4. During this work we found it necessary to specify that a QSIG-SIP =
gateway<BR>
should behave as a transit PBX from the perspective of the QSIG =
protocol. I<BR>
think this was mainly to make sure that all timers worked correctly. =
You<BR>
might wish to explore whether the same statement needs to be made for =
DPNSS<BR>
gateways. For example, by making a gateway behave as a transit PBX, it =
would<BR>
be required to respond with a CIM when a CRM is received, rather than =
simply<BR>
passing on the CRM.<BR>
<BR>
5. This may also have impact on the message buffering option. A gateway =
may<BR>
need to run a timer if it receives EEM(I) to guard against failure =
to<BR>
receive EEM(C).<BR>
<BR>
6. The message buffering option may not be suitable for buffering =
ISRM(I)<BR>
and SSRM(I) messages. These messages are not only used to overcome the =
45<BR>
octet message limit but are also used for overlap sending. Consequently =
a<BR>
SSRM(C) will not necessarily be expected.<BR>
<BR>
7. For QSIG we are considering two options: option 1 is to tunnel only =
one<BR>
QSIG call over a single SIP dialogue. Option 2 is to tunnel an =
arbitrary<BR>
number of QSIG calls in series and/or in parallel over a semi-permanent =
SIP<BR>
dialogue. Option 2 uses the QSIG call reference to identify the call and =
the<BR>
QSIG channel identification information element to identify the =
particular<BR>
RTP session. I suspect that you will wish to limit DPNSS tunnelling =
to<BR>
option 1 - otherwise you will need to carry additional information that =
is<BR>
normally available from DPNSS layer 2. This needs to be stated.<BR>
<BR>
I hope this is of use to you.<BR>
<BR>
John<BR>
<BR>
John Elwell - Technical Consultant<BR>
Tel: +44 (0)115 943 4989<BR>
Fax: +44 (0)115 943 4969<BR>
e-mail: <A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</A=
> &lt;<A =
HREF=3D"mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</A=
>&gt;<BR>
web: <A =
HREF=3D"http://www.siemens-design-services.com">http://www.siemens-design=
-services.com</A><BR>
&lt;<A =
HREF=3D"http://www.siemens-design-services.com/">http://www.siemens-desig=
n-services.com/</A>&gt;<BR>
<BR>
Siemens Communications<BR>
Technology Drive, Beeston, Nottingham, NG9 1LA.&nbsp; England.<BR>
<BR>
&quot;Siemens Communications - a division of Siemens plc,&nbsp; =
Registered No: 727817,<BR>
England. Registered office: Siemens House, Oldbury, Bracknell, =
Berkshire,<BR>
RG12 8FZ.<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: Ranjith Mukundan [<A =
HREF=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro.=
com</A>]<BR>
Sent: 11 February 2003 08:59<BR>
To: sipping@ietf.org<BR>
Subject: [Sipping] QSIG Vs DPNSS MIME<BR>
<BR>
<BR>
<BR>
we'd appreciate some views on this draft from atleast those who =
fairly<BR>
understand QSIG.<BR>
<BR>
of course, others who feel that this MIME type collides directly with =
the<BR>
purpose of SIP are encouraged to provide alternatives bearing in mind =
that<BR>
providing complete end-to-end service transparency over the intervening =
SIP<BR>
based core would be essential, if we are to provide a seamless =
migration<BR>
path from H.323 based core networks - which already provide =
equivalent<BR>
mechanisms for DSS1 based protocols.<BR>
<BR>
- ranjith..<BR>
<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Ranjith Mukundan<BR>
Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09 AM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Cc:<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
<BR>
Hi,<BR>
<BR>
posting WRT<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A>&gt;<BR>
<BR>
<BR>
&gt;&gt;<BR>
&gt;&gt; Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd prefer<BR>
&gt;&gt; that folks implement a translation to SIP, where SIP can =
natively carry<BR>
&gt;&gt; all the information that DPNSS is trying to convey.<BR>
&gt;&gt;<BR>
<BR>
To put things in perspective:<BR>
<BR>
1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and =
parts of<BR>
NA) very similar to QSIG (deployed primarily Europe) for =
interconnecting<BR>
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds =
enormous<BR>
application towards realizing a local VPN or network-wide VPN =
services.<BR>
<BR>
2- This draft aims to do exactly what RFC 3204 (MIME objects for =
QSIG/ISUP)<BR>
does for QSIG with appropriate/relevant customization for DPNSS - i.e., =
to<BR>
provide end-to-end &amp; *complete*, DPNSS Service transparency between =
pockets<BR>
of DPNSS based private networks over a SIP based core.<BR>
<BR>
3- Currently, circuit-switched SS7 based cores, provide equivalent =
DPNSS<BR>
tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant<BR>
translation/interworking mechanisms to ISUP. The =
translation/interworking<BR>
mechanism to ISUP is limited to the basic call and certain ISUP<BR>
supplementary services where there is a direct/easy equivalent.<BR>
<BR>
4- H.323 also provides equivalent tunneling mechanisms (via the H.450 =
Annexe<BR>
procedures) for transparently carrying DSS1 based signaling between<BR>
DSS1-based network elements/gateways, over a H.323 core.<BR>
<BR>
Some specifics:<BR>
<BR>
1- DPNSS supports, as specified in BTNR 188, approximately 240+<BR>
supplementary features/services (like Call Back When Free, Call Back =
When<BR>
Next Used, Executive Intrusion, Travelling Class of Service, Do Not =
Disturb,<BR>
Night Service, Private Network Route Optimization, Centralized =
Operator,<BR>
Three Party Takeover, Remote Diversion Registration etc). It would be =
almost<BR>
impossible to interwork to equivalent mechanisms in SIP, without losing =
the<BR>
service transparency when using an intervening SIP based core.<BR>
<BR>
2- Yes, we definitely need an interworking/translation mechanism to map =
all<BR>
the basic call/service paremeters, and procedures for mapping the =
messages.<BR>
But this does not preclude the need for a DPNSS MIME type, but would =
rather<BR>
be complimentary to the MIME type mechanism. Similar to<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A>&gt;&nbsp; ,<BR>
for QSIG, we are in the process of putting together an interworking =
draft<BR>
for DPNSS as well - but rather than making it DPNSS specific, we are =
trying<BR>
to make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG =
interworking<BR>
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a =
MIME<BR>
type as these are, as one might be aware, for PSTN access, unlike DPNSS =
and<BR>
QSIG (which are private network protocols).<BR>
<BR>
3- From the SIP perspective, carrying non-SIP applications as MIME =
objects<BR>
to ensure end to end service transparency is architecturally less =
appalling<BR>
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so =
the<BR>
SIP DPNSS MIME proposition as per this I-D is the obvious way forward =
for<BR>
DPNSS service transparency over a SIP based core, especially now that we =
see<BR>
acute deployment needs to be met.<BR>
<BR>
cheers,<BR>
ranjith..<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Ranjith Mukundan<BR>
&gt; Sent: Tuesday, February 04, 2003 5:32 PM<BR>
&gt; To: sipping@ietf.org<BR>
&gt; Subject: RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Hi List,<BR>
&gt; We would like you take a look at this draft and revert with any<BR>
&gt; comments/queries that you might have.<BR>
&gt;<BR>
&gt; SIPPING Chairs,<BR>
&gt; The authors feel that the requirement and the need for this MIME =
type<BR>
&gt; is<BR>
&gt; quite acute. Hence we'd like you to consider this for inclusion as =
a<BR>
&gt; WG I-D<BR>
&gt; and suggest the way forward.<BR>
&gt;<BR>
&gt; - ranjith..<BR>
&gt;<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Internet-Drafts@ietf.org<BR>
&gt; Sent: Tuesday, February 04, 2003 5:14 PM<BR>
&gt; To: IETF-Announce:<BR>
&gt; Cc: sipping@ietf.org<BR>
&gt; Subject: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; A New Internet-Draft is available from the on-line =
Internet-Drafts<BR>
&gt; directories.<BR>
&gt;<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME =
media types for DPNSS Objects<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-2-3<BR>
&gt;<BR>
&gt; This document describes Multipart Internet Mail Extensions (MIME) =
type<BR>
&gt; for application/Digital Private Network Signaling System =
(DPNSS)<BR>
&gt; objects for use in Session Initiation Protocol (SIP) =
applications,<BR>
&gt; according to the rules defined in RFC 2048. These types can be used =
to<BR>
&gt; identify Digital Private Network Signaling System 1 (DPNSS 1) =
objects<BR>
&gt; within a SIP message such as INVITE or INFO, as might be =
implemented<BR>
&gt; when using SIP in an environment where part of the call =
involves<BR>
&gt; interworking to DPNSS 1 based Private Networks. RFC 3204 addresses =
a<BR>
&gt; similar need for Integrated Service Digital Network User Part =
(ISUP)<BR>
&gt; and Q Signaling (QSIG).<BR>
&gt;<BR>
&gt; A URL for this Internet-Draft is:<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
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Date: Tue, 11 Feb 2003 13:19:47 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <A16A3EE4D4CA124FADC7987B1AC89FE440ED18@esebe022.ntc.nokia.com>
To: john.loughney@nokia.com, Gonzalo.Camarillo@lmf.ericsson.se,
        sipping@ietf.org
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>potential for AAA is much 
> greater than the existing installed base.

Yes, and it will be in the XML based web services architecture. Nobody
will buy anything, including IP services using anything else. No bank IT
department will ever know what DIAMETER is.

I rest my case and will not repeat this argument any further. Who can
predict the future anyway...

Henry

> -----Original Message-----
> From: john.loughney@nokia.com [mailto:john.loughney@nokia.com] 
> Sent: Tuesday, February 11, 2003 12:46 PM
> To: Henry.Sinnreich@wcom.com; 
> Gonzalo.Camarillo@lmf.ericsson.se; sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Hi Henry,
> 
> > >but one option is develop a solution using
> > > Diameter in RADIUS-compatibility mode.
> > 
> > OK, this is better than nothing. We'll take what we can :-)
> > 
> > As mentioned, the installed base is RADIUS, no matter how
> > voluminous the DIAMETER literature may be.
> 
> I think the issues is that the IETF should recommend what is 
> the right thing to do & why.  If people choose to do 
> something else, well, that is life.  
> 
> I understand what you are asking but I do note that some
> people have thought that the potential for AAA is much 
> greater than the existing installed base.
> 
> br,
> John
> 

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>, sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Tue, 11 Feb 2003 14:57:46 -0500
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01C2D207.D8D90320
Content-Type: text/plain;
	charset="iso-8859-1"

 
I begin to sympathize with those who argued that the ISUP/QSIG MIME type
should have been a more open-ended envelope for telephony signaling
information. I am reluctant to standardize every conceivable sort of
telephony signaling as a MIME type with some profile information for
encapsulating it in SIP.
 
I suppose the question really is one of how great the need is to standardize
this DPNSS encapsulation. I'm not really familiar with it, so I don't have
much of an opinion either way.
 
In any event, as another nit, this document should not define the
Content-Disposition header, it should reference the definition and BNF in
RFC3204.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 11, 2003 9:29 AM
To: sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME


All,
 
I really can't see any difference between DPNSS and Q.SIG when it comes to
the need for a MIME type for carriage of this within SIP. Can anyone come up
with an argument for/against this draft which does not apply to Q.SIG just
as well ?
 
If not, then I certainly think this should be progressed.
 
I have some comments on the draft:
1) DPNSS has now been published by the UK's Network Interoperability
Consultative Committee as NICC Document 1301:2001/03. You can find it at
http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm
<http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.ht
m>  along with lots of other gems.
 
You should reference this version, rather than the somewhat harder to obtain
BTNR.
 
2) Why do you need a length octet at the front of the message ? Can't you
(or shouldn't you) put a Content-Length in the MIME headers for this MIME
part ?
3) I think given the short length of the messages you could get away with
mandating that segmented DPNSS messages should always be reassembled before
encapsulating in SIP and that receivers must be able to receive such
reassembled messages. This removes some unnecessary complexity.
4) Section 3, under table 1, you say 'Floating Point' when you mean
'Decimal'. I think the words 'contiguously increasing' could probably be
removed here too. You will have trouble finding the next real number after
6.00!
 
Regards...Mark
 

-----Original Message-----
From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
Sent: 11 February 2003 08:59
To: sipping@ietf.org
Subject: [Sipping] QSIG Vs DPNSS MIME



we'd appreciate some views on this draft from atleast those who fairly
understand QSIG.

of course, others who feel that this MIME type collides directly with the
purpose of SIP are encouraged to provide alternatives bearing in mind that
providing complete end-to-end service transparency over the intervening SIP
based core would be essential, if we are to provide a seamless migration
path from H.323 based core networks - which already provide equivalent
mechanisms for DSS1 based protocols.

- ranjith..


-----Original Message-----
From:   Ranjith Mukundan
Sent:   Sat 2/8/2003 12:09 AM
To:     sipping@ietf.org
Cc:    
Subject:        RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt

Hi,

posting WRT
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt> 


>>
>> Why is it so important to be able to tunnel DPNSS in SIP?  I'd prefer
>> that folks implement a translation to SIP, where SIP can natively carry
>> all the information that DPNSS is trying to convey.
>>

To put things in perspective:

1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and parts of
NA) very similar to QSIG (deployed primarily Europe) for interconnecting
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous
application towards realizing a local VPN or network-wide VPN services.

2- This draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP)
does for QSIG with appropriate/relevant customization for DPNSS - i.e., to
provide end-to-end & *complete*, DPNSS Service transparency between pockets
of DPNSS based private networks over a SIP based core.

3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS
tunneling mechanisms through (ETSI & ANSI) ISUP *apart from* the concomitant
translation/interworking mechanisms to ISUP. The translation/interworking
mechanism to ISUP is limited to the basic call and certain ISUP
supplementary services where there is a direct/easy equivalent.

4- H.323 also provides equivalent tunneling mechanisms (via the H.450 Annexe
procedures) for transparently carrying DSS1 based signaling between
DSS1-based network elements/gateways, over a H.323 core.

Some specifics:

1- DPNSS supports, as specified in BTNR 188, approximately 240+
supplementary features/services (like Call Back When Free, Call Back When
Next Used, Executive Intrusion, Travelling Class of Service, Do Not Disturb,
Night Service, Private Network Route Optimization, Centralized Operator,
Three Party Takeover, Remote Diversion Registration etc). It would be almost
impossible to interwork to equivalent mechanisms in SIP, without losing the
service transparency when using an intervening SIP based core.

2- Yes, we definitely need an interworking/translation mechanism to map all
the basic call/service paremeters, and procedures for mapping the messages.
But this does not preclude the need for a DPNSS MIME type, but would rather
be complimentary to the MIME type mechanism. Similar to
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt>  ,
for QSIG, we are in the process of putting together an interworking draft
for DPNSS as well - but rather than making it DPNSS specific, we are trying
to make it a common draft to cover PRI, DPNSS & DASS2 [as Q.SIG interworking
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME
type as these are, as one might be aware, for PSTN access, unlike DPNSS and
QSIG (which are private network protocols).

3- From the SIP perspective, carrying non-SIP applications as MIME objects
to ensure end to end service transparency is architecturally less appalling
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so the
SIP DPNSS MIME proposition as per this I-D is the obvious way forward for
DPNSS service transparency over a SIP based core, especially now that we see
acute deployment needs to be met.

cheers,
ranjith..

> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Ranjith Mukundan
> Sent: Tuesday, February 04, 2003 5:32 PM
> To: sipping@ietf.org
> Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> Hi List,
> We would like you take a look at this draft and revert with any
> comments/queries that you might have.
>
> SIPPING Chairs,
> The authors feel that the requirement and the need for this MIME type
> is
> quite acute. Hence we'd like you to consider this for inclusion as a
> WG I-D
> and suggest the way forward.
>
> - ranjith..
>
> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Internet-Drafts@ietf.org
> Sent: Tuesday, February 04, 2003 5:14 PM
> To: IETF-Announce:
> Cc: sipping@ietf.org
> Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>       Title           : MIME media types for DPNSS Objects
>       Author(s)       : R. Mukundan et al.
>       Filename        : draft-mukundan-sipping-dpnss-01.txt
>       Pages           : 9
>       Date            : 2003-2-3
>
> This document describes Multipart Internet Mail Extensions (MIME) type
> for application/Digital Private Network Signaling System (DPNSS)
> objects for use in Session Initiation Protocol (SIP) applications,
> according to the rules defined in RFC 2048. These types can be used to
> identify Digital Private Network Signaling System 1 (DPNSS 1) objects
> within a SIP message such as INVITE or INFO, as might be implemented
> when using SIP in an environment where part of the call involves
> interworking to DPNSS 1 based Private Networks. RFC 3204 addresses a
> similar need for Integrated Service Digital Network User Part (ISUP)
> and Q Signaling (QSIG).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx> 
> t
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the
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>
> Internet-Drafts are also available by anonymous FTP. Login with the
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>       "get draft-mukundan-sipping-dpnss-01.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html>  or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
<ftp://ftp.ietf.org/ietf/1shadow-sites.txt> 
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
>       mailserv@ietf.org.
> In the body type:
>       "FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
>
> NOTE: The mail server at ietf.org can return the document in
>       MIME-encoded form by using the "mpack" utility.  To use this
>       feature, insert the command "ENCODING mime" before the "FILE"
>       command.  To decode the response(s), you will need "munpack" or
>       a MIME-compliant mail reader.  Different MIME-compliant mail readers
>       exhibit different behavior, especially when dealing with
>       "multipart" MIME messages (i.e. documents which have been split
>       up into multiple messages), so check your local documentation on
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>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
<https://www1.ietf.org/mailman/listinfo/sipping> 
> This list is for NEW development of the application of SIP Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sip@ietf.org for new developments of core SIP






_______________________________________________ Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
development of the application of SIP Use sip-implementors@cs.columbia.edu
for questions on current sip Use sip@ietf.org for new developments of core
SIP 


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<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>QSIG Vs DPNSS MIME</TITLE>

<META content=3D"MSHTML 5.50.4807.2300" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>I=20
begin to sympathize with those who argued that the ISUP/QSIG MIME type =
should=20
have been a more open-ended envelope for telephony signaling =
information. I am=20
reluctant to standardize every conceivable sort of telephony signaling =
as a MIME=20
type with some profile information for encapsulating it in=20
SIP.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D798504719-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>I=20
suppose the question really is one of how great the need is to =
standardize this=20
DPNSS encapsulation. I'm not really familiar with it, so I don't have =
much of an=20
opinion either way.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D798504719-11022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>In any=20
event,&nbsp;as another nit, this document should not define the=20
Content-Disposition header, it should reference the definition and BNF =
in=20
RFC3204.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D798504719-11022003><FONT face=3DArial =
color=3D#0000ff size=3D2>Jon=20
Peterson</FONT></SPAN></DIV>
<DIV><SPAN class=3D798504719-11022003><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>NeuStar, Inc.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Tuesday, February =
11, 2003=20
  9:29 AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> RE: =
[Sipping] QSIG=20
  Vs DPNSS MIME<BR><BR></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003>All,</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>I=20
  really can't see any difference between DPNSS and Q.SIG when it comes =
to the=20
  need for a MIME type for carriage of this within SIP. Can anyone come =
up with=20
  an argument for/against this draft which does not apply to Q.SIG just =
as well=20
  ?</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>If=20
  not, then I certainly think this should be =
progressed.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>I=20
  have some comments on the draft:</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>1)=20
  DPNSS has now been published by the UK's Network Interoperability =
Consultative=20
  Committee as NICC Document 1301:2001/03. You can find it at <A=20
  =
href=3D"http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstand=
ards/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnec=
tstandards/isc.htm</A>&nbsp;along=20
  with lots of other gems.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003>You should reference this version, rather =
than the=20
  somewhat harder to obtain BTNR.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>2)=20
  Why do you need a length octet at the front of the message ? Can't =
you (or=20
  shouldn't you) put a Content-Length in the MIME headers for this MIME =
part=20
  ?</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>3)=20
  I think given the short length of the messages you could get away =
with=20
  mandating that segmented DPNSS messages should always be reassembled =
before=20
  encapsulating in SIP and that receivers must be able to receive such=20
  reassembled messages. This removes some unnecessary=20
  complexity.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D145285416-11022003>4)=20
  Section 3, under table 1, you say 'Floating Point' when you mean =
'Decimal'. I=20
  think the words 'contiguously increasing' could probably be removed =
here too.=20
  You will have trouble finding the next real number after=20
  6.00!</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003>Regards...Mark</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Ranjith =
Mukundan=20
    [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 11 February =
2003=20
    08:59<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] =
QSIG Vs=20
    DPNSS MIME<BR><BR></DIV></FONT><!-- Converted from text/plain =
format -->
    <P><FONT size=3D2>we'd appreciate some views on this draft from =
atleast those=20
    who fairly understand QSIG.<BR><BR>of course, others who feel that =
this MIME=20
    type collides directly with the purpose of SIP are encouraged to =
provide=20
    alternatives bearing in mind that providing complete end-to-end =
service=20
    transparency over the intervening SIP based core would be =
essential, if we=20
    are to provide a seamless migration path from H.323 based core =
networks -=20
    which already provide equivalent mechanisms for DSS1 based=20
    protocols.<BR><BR>- ranjith..<BR><BR><BR>-----Original=20
    Message-----<BR>From:&nbsp;&nbsp; Ranjith =
Mukundan<BR>Sent:&nbsp;&nbsp; Sat=20
    2/8/2003 12:09 AM<BR>To:&nbsp;&nbsp;&nbsp;&nbsp;=20
    =
sipping@ietf.org<BR>Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>Subject:&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    RE: [Sipping] I-D=20
    =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR><BR>Hi,<BR><BR>posting =
WRT <A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpns=
s-01.txt</A><BR><BR><BR>&gt;&gt;<BR>&gt;&gt;=20
    Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; I'd =

    prefer<BR>&gt;&gt; that folks implement a translation to SIP, where =
SIP can=20
    natively carry<BR>&gt;&gt; all the information that DPNSS is trying =
to=20
    convey.<BR>&gt;&gt;<BR><BR>To put things in perspective:<BR><BR>1- =
DPNSS is=20
    a protocol (deployed primarily in UK, NZ, Australia and parts of =
NA) very=20
    similar to QSIG (deployed primarily Europe) for interconnecting =
PBXs or=20
    centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous=20
    application towards realizing a local VPN or network-wide VPN=20
    services.<BR><BR>2- This draft aims to do exactly what RFC 3204 =
(MIME=20
    objects for QSIG/ISUP) does for QSIG with appropriate/relevant =
customization=20
    for DPNSS - i.e., to provide end-to-end &amp; *complete*, DPNSS =
Service=20
    transparency between pockets of DPNSS based private networks over a =
SIP=20
    based core.<BR><BR>3- Currently, circuit-switched SS7 based cores, =
provide=20
    equivalent DPNSS tunneling mechanisms through (ETSI &amp; ANSI) =
ISUP *apart=20
    from* the concomitant translation/interworking mechanisms to ISUP. =
The=20
    translation/interworking mechanism to ISUP is limited to the basic =
call and=20
    certain ISUP supplementary services where there is a direct/easy=20
    equivalent.<BR><BR>4- H.323 also provides equivalent tunneling =
mechanisms=20
    (via the H.450 Annexe procedures) for transparently carrying DSS1 =
based=20
    signaling between DSS1-based network elements/gateways, over a =
H.323=20
    core.<BR><BR>Some specifics:<BR><BR>1- DPNSS supports, as specified =
in BTNR=20
    188, approximately 240+ supplementary features/services (like Call =
Back When=20
    Free, Call Back When Next Used, Executive Intrusion, Travelling =
Class of=20
    Service, Do Not Disturb, Night Service, Private Network Route =
Optimization,=20
    Centralized Operator, Three Party Takeover, Remote Diversion =
Registration=20
    etc). It would be almost impossible to interwork to equivalent =
mechanisms in=20
    SIP, without losing the service transparency when using an =
intervening SIP=20
    based core.<BR><BR>2- Yes, we definitely need an =
interworking/translation=20
    mechanism to map all the basic call/service paremeters, and =
procedures for=20
    mapping the messages. But this does not preclude the need for a =
DPNSS MIME=20
    type, but would rather be complimentary to the MIME type mechanism. =
Similar=20
    to <A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-=
00.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-=
00.txt</A>=20
    , for QSIG, we are in the process of putting together an =
interworking draft=20
    for DPNSS as well - but rather than making it DPNSS specific, we =
are trying=20
    to make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG =

    interworking is already a SIPPING WG ID]. Note that, PRI and DASS2 =
will not=20
    need a MIME type as these are, as one might be aware, for PSTN =
access,=20
    unlike DPNSS and QSIG (which are private network =
protocols).<BR><BR>3- From=20
    the SIP perspective, carrying non-SIP applications as MIME objects =
to ensure=20
    end to end service transparency is architecturally less appalling =
than, say=20
    for e.g., tunneling DSS1 protocols in H.323 based network - so the =
SIP DPNSS=20
    MIME proposition as per this I-D is the obvious way forward for =
DPNSS=20
    service transparency over a SIP based core, especially now that we =
see acute=20
    deployment needs to be met.<BR><BR>cheers,<BR>ranjith..<BR><BR>&gt; =

    -----Original Message-----<BR>&gt; From: sipping-admin@ietf.org [<A =

    =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
]On=20
    Behalf<BR>&gt; Of<BR>&gt; Ranjith Mukundan<BR>&gt; Sent: Tuesday, =
February=20
    04, 2003 5:32 PM<BR>&gt; To: sipping@ietf.org<BR>&gt; Subject: RE: =
[Sipping]=20
    I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; Hi=20
    List,<BR>&gt; We would like you take a look at this draft and =
revert with=20
    any<BR>&gt; comments/queries that you might have.<BR>&gt;<BR>&gt; =
SIPPING=20
    Chairs,<BR>&gt; The authors feel that the requirement and the need =
for this=20
    MIME type<BR>&gt; is<BR>&gt; quite acute. Hence we'd like you to =
consider=20
    this for inclusion as a<BR>&gt; WG I-D<BR>&gt; and suggest the way=20
    forward.<BR>&gt;<BR>&gt; - ranjith..<BR>&gt;<BR>&gt; -----Original=20
    Message-----<BR>&gt; From: sipping-admin@ietf.org [<A=20
    =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
]On=20
    Behalf<BR>&gt; Of<BR>&gt; Internet-Drafts@ietf.org<BR>&gt; Sent: =
Tuesday,=20
    February 04, 2003 5:14 PM<BR>&gt; To: IETF-Announce:<BR>&gt; Cc:=20
    sipping@ietf.org<BR>&gt; Subject: [Sipping] I-D=20
    ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; =
A New=20
    Internet-Draft is available from the on-line =
Internet-Drafts<BR>&gt;=20
    directories.<BR>&gt;<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =

    Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME =
media=20
    types for DPNSS Objects<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et =
al.<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :=20
    draft-mukundan-sipping-dpnss-01.txt<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-2-3<BR>&gt;<BR>&gt; This=20
    document describes Multipart Internet Mail Extensions (MIME) =
type<BR>&gt;=20
    for application/Digital Private Network Signaling System =
(DPNSS)<BR>&gt;=20
    objects for use in Session Initiation Protocol (SIP) =
applications,<BR>&gt;=20
    according to the rules defined in RFC 2048. These types can be used =

    to<BR>&gt; identify Digital Private Network Signaling System 1 =
(DPNSS 1)=20
    objects<BR>&gt; within a SIP message such as INVITE or INFO, as =
might be=20
    implemented<BR>&gt; when using SIP in an environment where part of =
the call=20
    involves<BR>&gt; interworking to DPNSS 1 based Private Networks. =
RFC 3204=20
    addresses a<BR>&gt; similar need for Integrated Service Digital =
Network User=20
    Part (ISUP)<BR>&gt; and Q Signaling (QSIG).<BR>&gt;<BR>&gt; A URL =
for this=20
    Internet-Draft is:<BR>&gt; <A=20
    =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss=
-01.tx</A><BR>&gt;=20
    t<BR>&gt;<BR>&gt; To remove yourself from the IETF Announcement =
list, send a=20
    message to<BR>&gt; ietf-announce-request with the word unsubscribe =
in the=20
    body of the<BR>&gt; message.<BR>&gt;<BR>&gt; Internet-Drafts are =
also=20
    available by anonymous FTP. Login with the<BR>&gt; username<BR>&gt; =

    "anonymous" and a password of your e-mail address. After logging =
in,<BR>&gt;=20
    type "cd internet-drafts" and then<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    "get draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; A list =
of=20
    Internet-Drafts directories can be found in<BR>&gt; <A=20
    =
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html=
</A>=20
    or<BR>&gt; <A=20
    =
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ie=
tf/1shadow-sites.txt</A><BR>&gt;<BR>&gt;<BR>&gt;=20
    Internet-Drafts can also be obtained by e-mail.<BR>&gt;<BR>&gt; =
Send a=20
    message to:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
    mailserv@ietf.org.<BR>&gt; In the body type:<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "FILE=20
    =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; =
NOTE:=20
    The mail server at ietf.org can return the document in<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
"mpack"=20
    utility.&nbsp; To use this<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
feature,=20
    insert the command "ENCODING mime" before the "FILE"<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you=20
    will need "munpack" or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=20
    MIME-compliant mail reader.&nbsp; Different MIME-compliant mail=20
    readers<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different =
behavior,=20
    especially when dealing with<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =

    "multipart" MIME messages (i.e. documents which have been =
split<BR>&gt;=20
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so check =
your=20
    local documentation on<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how =
to=20
    manipulate these messages.<BR>&gt;<BR>&gt;<BR>&gt; Below is the =
data which=20
    will enable a MIME compliant mail reader<BR>&gt; implementation to=20
    automatically retrieve the ASCII version of the<BR>&gt;=20
    Internet-Draft.<BR>&gt;<BR>&gt;=20
    _______________________________________________<BR>&gt; Sipping =
mailing=20
    list&nbsp; <A=20
    =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.iet=
f.org/mailman/listinfo/sipping</A><BR>&gt;=20
    This list is for NEW development of the application of SIP =
Use<BR>&gt;=20
    sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>&gt;=20
    sip@ietf.org for new developments of core=20
    =
SIP<BR><BR><BR><BR><BR></FONT></P>______________________________________=
_________=20
    Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping =
This=20
    list is for NEW development of the application of SIP Use=20
    sip-implementors@cs.columbia.edu for questions on current sip Use=20
    sip@ietf.org for new developments of core SIP=20
</BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Miao, Kai" <kai.miao@intel.com>
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Subject: [Sipping] Question about call flow in draft-ietf-sipping-service-examples-0
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Could someone provide an answer to the following question:
 
In the first example, Call Hold, given in
draft-ietf-sipping-service-examples-03.txt, why F10, i.e., INVITE(hold), has
a=sendonly in its SDP message and F12, i.e., 200 OK, has no "a=" attribute
at all while "No RTP Sent!" is marked in the call flow diagram? 
 
Shoudn't F12 contain a=recvonly or a=inactive, following the Offer/Answer
model? Why there is "no RTP sent" in the diagram when there is "a=sendonly"
in F10?
 
Thanks,
Kai Miao
Intel   

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<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=481480723-11022003>Could 
someone provide an answer to the following question:</SPAN></FONT></DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003>In the first example, Call Hold, given 
in&nbsp;</SPAN>draft-ietf-sipping-service-examples-03.tx<SPAN 
class=481480723-11022003>t, why F10, i.e., INVITE(hold), has a=sendonly&nbsp;in 
its SDP message and F12, i.e., 200 OK, has no "a=" attribute at all while "No 
RTP Sent!" is marked in the call flow 
diagram?&nbsp;</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003>Shoudn't F12 contain a=recvonly or a=inactive, 
following the Offer/Answer model? Why there is "no RTP sent" in the diagram when 
there is "a=sendonly" in&nbsp;F10?</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003>Thanks,</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003>Kai Miao</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT size=2><FONT color=#0000ff><FONT face=Arial><SPAN 
class=481480723-11022003>Intel&nbsp;&nbsp;&nbsp;</SPAN></FONT></FONT></FONT></DIV></BODY></HTML>

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Miao, Kai would like to recall the message, "Question about call flow in
draft-ietf-sipping-service-examples-03".
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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>, <sipping@ietf.org>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
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QSIG Vs DPNSS MIMEin so far as inter-PBX/Centrex connectivity goes, amongst
the DSS1 derived protocols, DPNSS is, IMO, the *only*  (big) missing piece.
Can anyone think of another, standards-based, telephony signaling, the might
require a MIME carriage?? (atleast, i can't think of any - not even an
arcane, minimally deployed one)

As I mentioned earlier, we dont need MIME types for protocols like DASS2,
PRI and all the other 'conceivable sort' of telephony signaling thats used
for PSTN connectivity (vis-a-vis inter-PBX/centrex connectivity).  Even in
the case of telephony protocols used for inter-PBX/centrex connectivity,
unless it requires the same end to end, complete service transparency as in
the case of private networks interconnected by QSIG or DPNSS, it does not
mandate carriage as MIME type (a simple interworking function to SIP would
suffice).

So I dont think we'll a slew of MIME types...

>> In any event, as another nit, this document should not define the
Content-Disposition header, it should reference the definition and BNF in
RFC3204.
Should've known. thanks - it was an oversight.

cheers,
- ranjith..
  -----Original Message-----
  From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Peterson, Jon
  Sent: Wednesday, February 12, 2003 1:28 AM
  To: 'Mark Watson'; sipping@ietf.org
  Subject: RE: [Sipping] QSIG Vs DPNSS MIME



  I begin to sympathize with those who argued that the ISUP/QSIG MIME type
should have been a more open-ended envelope for telephony signaling
information. I am reluctant to standardize every conceivable sort of
telephony signaling as a MIME type with some profile information for
encapsulating it in SIP.

  I suppose the question really is one of how great the need is to
standardize this DPNSS encapsulation. I'm not really familiar with it, so I
don't have much of an opinion either way.

  In any event, as another nit, this document should not define the
Content-Disposition header, it should reference the definition and BNF in
RFC3204.

  Jon Peterson
  NeuStar, Inc.
    -----Original Message-----
    From: Mark Watson [mailto:mwatson@nortelnetworks.com]
    Sent: Tuesday, February 11, 2003 9:29 AM
    To: sipping@ietf.org
    Subject: RE: [Sipping] QSIG Vs DPNSS MIME


    All,

    I really can't see any difference between DPNSS and Q.SIG when it comes
to the need for a MIME type for carriage of this within SIP. Can anyone come
up with an argument for/against this draft which does not apply to Q.SIG
just as well ?

    If not, then I certainly think this should be progressed.

    I have some comments on the draft:
    1) DPNSS has now been published by the UK's Network Interoperability
Consultative Committee as NICC Document 1301:2001/03. You can find it at
http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm
along with lots of other gems.

    You should reference this version, rather than the somewhat harder to
obtain BTNR.

    2) Why do you need a length octet at the front of the message ? Can't
you (or shouldn't you) put a Content-Length in the MIME headers for this
MIME part ?
    3) I think given the short length of the messages you could get away
with mandating that segmented DPNSS messages should always be reassembled
before encapsulating in SIP and that receivers must be able to receive such
reassembled messages. This removes some unnecessary complexity.
    4) Section 3, under table 1, you say 'Floating Point' when you mean
'Decimal'. I think the words 'contiguously increasing' could probably be
removed here too. You will have trouble finding the next real number after
6.00!

    Regards...Mark

      -----Original Message-----
      From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
      Sent: 11 February 2003 08:59
      To: sipping@ietf.org
      Subject: [Sipping] QSIG Vs DPNSS MIME


      we'd appreciate some views on this draft from atleast those who fairly
understand QSIG.

      of course, others who feel that this MIME type collides directly with
the purpose of SIP are encouraged to provide alternatives bearing in mind
that providing complete end-to-end service transparency over the intervening
SIP based core would be essential, if we are to provide a seamless migration
path from H.323 based core networks - which already provide equivalent
mechanisms for DSS1 based protocols.

      - ranjith..


      -----Original Message-----
      From:   Ranjith Mukundan
      Sent:   Sat 2/8/2003 12:09 AM
      To:     sipping@ietf.org
      Cc:
      Subject:        RE: [Sipping] I-D
ACTION:draft-mukundan-sipping-dpnss-01.txt

      Hi,

      posting WRT
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt


      >>
      >> Why is it so important to be able to tunnel DPNSS in SIP?  I'd
prefer
      >> that folks implement a translation to SIP, where SIP can natively
carry
      >> all the information that DPNSS is trying to convey.
      >>

      To put things in perspective:

      1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and
parts of NA) very similar to QSIG (deployed primarily Europe) for
interconnecting PBXs or centrex customers hosted off LEs. DPNSS (like QSIG)
finds enormous application towards realizing a local VPN or network-wide VPN
services.

      2- This draft aims to do exactly what RFC 3204 (MIME objects for
QSIG/ISUP) does for QSIG with appropriate/relevant customization for DPNSS -
i.e., to provide end-to-end & *complete*, DPNSS Service transparency between
pockets of DPNSS based private networks over a SIP based core.

      3- Currently, circuit-switched SS7 based cores, provide equivalent
DPNSS tunneling mechanisms through (ETSI & ANSI) ISUP *apart from* the
concomitant translation/interworking mechanisms to ISUP. The
translation/interworking mechanism to ISUP is limited to the basic call and
certain ISUP supplementary services where there is a direct/easy equivalent.

      4- H.323 also provides equivalent tunneling mechanisms (via the H.450
Annexe procedures) for transparently carrying DSS1 based signaling between
DSS1-based network elements/gateways, over a H.323 core.

      Some specifics:

      1- DPNSS supports, as specified in BTNR 188, approximately 240+
supplementary features/services (like Call Back When Free, Call Back When
Next Used, Executive Intrusion, Travelling Class of Service, Do Not Disturb,
Night Service, Private Network Route Optimization, Centralized Operator,
Three Party Takeover, Remote Diversion Registration etc). It would be almost
impossible to interwork to equivalent mechanisms in SIP, without losing the
service transparency when using an intervening SIP based core.

      2- Yes, we definitely need an interworking/translation mechanism to
map all the basic call/service paremeters, and procedures for mapping the
messages. But this does not preclude the need for a DPNSS MIME type, but
would rather be complimentary to the MIME type mechanism. Similar to
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt , for
QSIG, we are in the process of putting together an interworking draft for
DPNSS as well - but rather than making it DPNSS specific, we are trying to
make it a common draft to cover PRI, DPNSS & DASS2 [as Q.SIG interworking is
already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME type
as these are, as one might be aware, for PSTN access, unlike DPNSS and QSIG
(which are private network protocols).

      3- From the SIP perspective, carrying non-SIP applications as MIME
objects to ensure end to end service transparency is architecturally less
appalling than, say for e.g., tunneling DSS1 protocols in H.323 based
network - so the SIP DPNSS MIME proposition as per this I-D is the obvious
way forward for DPNSS service transparency over a SIP based core, especially
now that we see acute deployment needs to be met.

      cheers,
      ranjith..

      > -----Original Message-----
      > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On
Behalf
      > Of
      > Ranjith Mukundan
      > Sent: Tuesday, February 04, 2003 5:32 PM
      > To: sipping@ietf.org
      > Subject: RE: [Sipping] I-D
ACTION:draft-mukundan-sipping-dpnss-01.txt
      >
      >
      > Hi List,
      > We would like you take a look at this draft and revert with any
      > comments/queries that you might have.
      >
      > SIPPING Chairs,
      > The authors feel that the requirement and the need for this MIME
type
      > is
      > quite acute. Hence we'd like you to consider this for inclusion as a
      > WG I-D
      > and suggest the way forward.
      >
      > - ranjith..
      >
      > -----Original Message-----
      > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On
Behalf
      > Of
      > Internet-Drafts@ietf.org
      > Sent: Tuesday, February 04, 2003 5:14 PM
      > To: IETF-Announce:
      > Cc: sipping@ietf.org
      > Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
      >
      >
      > A New Internet-Draft is available from the on-line Internet-Drafts
      > directories.
      >
      >
      >       Title           : MIME media types for DPNSS Objects
      >       Author(s)       : R. Mukundan et al.
      >       Filename        : draft-mukundan-sipping-dpnss-01.txt
      >       Pages           : 9
      >       Date            : 2003-2-3
      >
      > This document describes Multipart Internet Mail Extensions (MIME)
type
      > for application/Digital Private Network Signaling System (DPNSS)
      > objects for use in Session Initiation Protocol (SIP) applications,
      > according to the rules defined in RFC 2048. These types can be used
to
      > identify Digital Private Network Signaling System 1 (DPNSS 1)
objects
      > within a SIP message such as INVITE or INFO, as might be implemented
      > when using SIP in an environment where part of the call involves
      > interworking to DPNSS 1 based Private Networks. RFC 3204 addresses a
      > similar need for Integrated Service Digital Network User Part (ISUP)
      > and Q Signaling (QSIG).
      >
      > A URL for this Internet-Draft is:
      >
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
      > t
      >
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      >       command.  To decode the response(s), you will need "munpack"
or
      >       a MIME-compliant mail reader.  Different MIME-compliant mail
readers
      >       exhibit different behavior, especially when dealing with
      >       "multipart" MIME messages (i.e. documents which have been
split
      >       up into multiple messages), so check your local documentation
on
      >       how to manipulate these messages.
      >
      >
      > Below is the data which will enable a MIME compliant mail reader
      > implementation to automatically retrieve the ASCII version of the
      > Internet-Draft.
      >
      > _______________________________________________
      > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
      > This list is for NEW development of the application of SIP Use
      > sip-implementors@cs.columbia.edu for questions on current sip Use
      > sip@ietf.org for new developments of core SIP






      _______________________________________________ Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
development of the application of SIP Use sip-implementors@cs.columbia.edu
for questions on current sip Use sip@ietf.org for new developments of core
SIP

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<HTML><HEAD><TITLE>QSIG Vs DPNSS MIME</TITLE>
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<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =
size=3D2>in so=20
far as inter-PBX/Centrex connectivity goes, amongst the DSS1 derived =
protocols,=20
DPNSS is, IMO, the *only*&nbsp; (big) missing piece. Can anyone think of =

another, standards-based, telephony signaling, the might require a MIME=20
carriage?? (atleast, i can't think of any - not even an arcane, =
minimally=20
deployed one) </FONT></SPAN></DIV>
<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =
size=3D2>As I=20
mentioned earlier, we dont need MIME types for protocols like DASS2, PRI =
and all=20
the other 'conceivable sort' of telephony signaling thats used for PSTN=20
connectivity (vis-a-vis inter-PBX/centrex connectivity). &nbsp;Even in =
the case=20
of telephony protocols used for inter-PBX/centrex connectivity, unless =
it=20
requires the same end to end, complete service transparency as in the =
case of=20
private networks&nbsp;interconnected by&nbsp;QSIG or DPNSS, it does not=20
mandate&nbsp;carriage as&nbsp;MIME type (a simple interworking function =
to SIP=20
would suffice).&nbsp;&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =
size=3D2>So I=20
dont think we'll a slew of MIME types...</FONT></SPAN></DIV>
<DIV><SPAN class=3D058362703-12022003><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D058362703-12022003>
<DIV><SPAN class=3D798504719-11022003><FONT face=3DArial><FONT =
size=3D2><SPAN=20
class=3D058362703-12022003>&gt;&gt; </SPAN>In any event,&nbsp;as another =
nit, this=20
document should not define the Content-Disposition header, it should =
reference=20
the definition and BNF in RFC3204.</FONT></FONT></SPAN></DIV>
<DIV><SPAN class=3D798504719-11022003><SPAN =
class=3D058362703-12022003><FONT=20
face=3DArial color=3D#0000ff size=3D2>Should've known. thanks - it was =
an=20
oversight.</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D798504719-11022003><SPAN =
class=3D058362703-12022003><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D798504719-11022003><SPAN =
class=3D058362703-12022003><FONT=20
face=3DArial color=3D#0000ff size=3D2>cheers,</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=3D798504719-11022003><SPAN =
class=3D058362703-12022003><FONT=20
face=3DArial color=3D#0000ff size=3D2>-=20
ranjith..</FONT></SPAN></SPAN></DIV></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
sipping-admin@ietf.org=20
  [mailto:sipping-admin@ietf.org]<B>On Behalf Of </B>Peterson,=20
  Jon<BR><B>Sent:</B> Wednesday, February 12, 2003 1:28 AM<BR><B>To:</B> =
'Mark=20
  Watson'; sipping@ietf.org<BR><B>Subject:</B> RE: [Sipping] QSIG Vs =
DPNSS=20
  MIME<BR><BR></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>I=20
  begin to sympathize with those who argued that the ISUP/QSIG MIME type =
should=20
  have been a more open-ended envelope for telephony signaling =
information. I am=20
  reluctant to standardize every conceivable sort of telephony signaling =
as a=20
  MIME type with some profile information for encapsulating it in=20
  SIP.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D798504719-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>I=20
  suppose the question really is one of how great the need is to =
standardize=20
  this DPNSS encapsulation. I'm not really familiar with it, so I don't =
have=20
  much of an opinion either way.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
  class=3D798504719-11022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D798504719-11022003>In=20
  any event,&nbsp;as another nit, this document should not define the=20
  Content-Disposition header, it should reference the definition and BNF =
in=20
  RFC3204.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D798504719-11022003><FONT face=3DArial =
color=3D#0000ff size=3D2>Jon=20
  Peterson</FONT></SPAN></DIV>
  <DIV><SPAN class=3D798504719-11022003><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>NeuStar, Inc.</FONT></SPAN></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Mark Watson=20
    [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Tuesday, =
February 11,=20
    2003 9:29 AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> RE: =
[Sipping]=20
    QSIG Vs DPNSS MIME<BR><BR></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>All,</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>I really can't see any difference between =
DPNSS and=20
    Q.SIG when it comes to the need for a MIME type for carriage of this =
within=20
    SIP. Can anyone come up with an argument for/against this draft =
which does=20
    not apply to Q.SIG just as well ?</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>If not, then I certainly think this =
should be=20
    progressed.</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>I have some comments on the=20
    draft:</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>1) DPNSS has now been published by the =
UK's Network=20
    Interoperability Consultative Committee as NICC Document =
1301:2001/03. You=20
    can find it at <A=20
    =
href=3D"http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstanda=
rds/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnects=
tandards/isc.htm</A>&nbsp;along=20
    with lots of other gems.</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>You should reference this version, rather =
than the=20
    somewhat harder to obtain BTNR.</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>2) Why do you need a length octet at the =
front of=20
    the message ? Can't you (or shouldn't you) put a Content-Length in =
the MIME=20
    headers for this MIME part ?</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>3) I think given the short length of the =
messages=20
    you could get away with mandating that segmented DPNSS messages =
should=20
    always be reassembled before encapsulating in SIP and that receivers =
must be=20
    able to receive such reassembled messages. This removes some =
unnecessary=20
    complexity.</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>4) Section 3, under table 1, you say =
'Floating=20
    Point' when you mean 'Decimal'. I think the words 'contiguously =
increasing'=20
    could probably be removed here too. You will have trouble finding =
the next=20
    real number after 6.00!</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003>Regards...Mark</SPAN></FONT></DIV>
    <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
    class=3D145285416-11022003></SPAN></FONT>&nbsp;</DIV>
    <BLOCKQUOTE=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid">
      <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
      size=3D2>-----Original Message-----<BR><B>From:</B> Ranjith =
Mukundan=20
      [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 11 February =
2003=20
      08:59<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] =
QSIG Vs=20
      DPNSS MIME<BR><BR></DIV></FONT><!-- Converted from text/plain =
format -->
      <P><FONT size=3D2>we'd appreciate some views on this draft from =
atleast=20
      those who fairly understand QSIG.<BR><BR>of course, others who =
feel that=20
      this MIME type collides directly with the purpose of SIP are =
encouraged to=20
      provide alternatives bearing in mind that providing complete =
end-to-end=20
      service transparency over the intervening SIP based core would be=20
      essential, if we are to provide a seamless migration path from =
H.323 based=20
      core networks - which already provide equivalent mechanisms for =
DSS1 based=20
      protocols.<BR><BR>- ranjith..<BR><BR><BR>-----Original=20
      Message-----<BR>From:&nbsp;&nbsp; Ranjith =
Mukundan<BR>Sent:&nbsp;&nbsp;=20
      Sat 2/8/2003 12:09 AM<BR>To:&nbsp;&nbsp;&nbsp;&nbsp;=20
      =
sipping@ietf.org<BR>Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>Subject:&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      RE: [Sipping] I-D=20
      =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR><BR>Hi,<BR><BR>posting WRT =

      <A=20
      =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR><BR><BR>&gt;&gt;<BR>&gt;&gt;=20
      Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd=20
      prefer<BR>&gt;&gt; that folks implement a translation to SIP, =
where SIP=20
      can natively carry<BR>&gt;&gt; all the information that DPNSS is =
trying to=20
      convey.<BR>&gt;&gt;<BR><BR>To put things in perspective:<BR><BR>1- =
DPNSS=20
      is a protocol (deployed primarily in UK, NZ, Australia and parts =
of NA)=20
      very similar to QSIG (deployed primarily Europe) for =
interconnecting PBXs=20
      or centrex customers hosted off LEs. DPNSS (like QSIG) finds =
enormous=20
      application towards realizing a local VPN or network-wide VPN=20
      services.<BR><BR>2- This draft aims to do exactly what RFC 3204 =
(MIME=20
      objects for QSIG/ISUP) does for QSIG with appropriate/relevant=20
      customization for DPNSS - i.e., to provide end-to-end &amp; =
*complete*,=20
      DPNSS Service transparency between pockets of DPNSS based private =
networks=20
      over a SIP based core.<BR><BR>3- Currently, circuit-switched SS7 =
based=20
      cores, provide equivalent DPNSS tunneling mechanisms through (ETSI =
&amp;=20
      ANSI) ISUP *apart from* the concomitant translation/interworking=20
      mechanisms to ISUP. The translation/interworking mechanism to ISUP =
is=20
      limited to the basic call and certain ISUP supplementary services =
where=20
      there is a direct/easy equivalent.<BR><BR>4- H.323 also provides=20
      equivalent tunneling mechanisms (via the H.450 Annexe procedures) =
for=20
      transparently carrying DSS1 based signaling between DSS1-based =
network=20
      elements/gateways, over a H.323 core.<BR><BR>Some =
specifics:<BR><BR>1-=20
      DPNSS supports, as specified in BTNR 188, approximately 240+ =
supplementary=20
      features/services (like Call Back When Free, Call Back When Next =
Used,=20
      Executive Intrusion, Travelling Class of Service, Do Not Disturb, =
Night=20
      Service, Private Network Route Optimization, Centralized Operator, =
Three=20
      Party Takeover, Remote Diversion Registration etc). It would be =
almost=20
      impossible to interwork to equivalent mechanisms in SIP, without =
losing=20
      the service transparency when using an intervening SIP based=20
      core.<BR><BR>2- Yes, we definitely need an =
interworking/translation=20
      mechanism to map all the basic call/service paremeters, and =
procedures for=20
      mapping the messages. But this does not preclude the need for a =
DPNSS MIME=20
      type, but would rather be complimentary to the MIME type =
mechanism.=20
      Similar to <A=20
      =
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A>=20
      , for QSIG, we are in the process of putting together an =
interworking=20
      draft for DPNSS as well - but rather than making it DPNSS =
specific, we are=20
      trying to make it a common draft to cover PRI, DPNSS &amp; DASS2 =
[as Q.SIG=20
      interworking is already a SIPPING WG ID]. Note that, PRI and DASS2 =
will=20
      not need a MIME type as these are, as one might be aware, for PSTN =
access,=20
      unlike DPNSS and QSIG (which are private network =
protocols).<BR><BR>3-=20
      From the SIP perspective, carrying non-SIP applications as MIME =
objects to=20
      ensure end to end service transparency is architecturally less =
appalling=20
      than, say for e.g., tunneling DSS1 protocols in H.323 based =
network - so=20
      the SIP DPNSS MIME proposition as per this I-D is the obvious way =
forward=20
      for DPNSS service transparency over a SIP based core, especially =
now that=20
      we see acute deployment needs to be=20
      met.<BR><BR>cheers,<BR>ranjith..<BR><BR>&gt; -----Original=20
      Message-----<BR>&gt; From: sipping-admin@ietf.org [<A=20
      =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On=20
      Behalf<BR>&gt; Of<BR>&gt; Ranjith Mukundan<BR>&gt; Sent: Tuesday, =
February=20
      04, 2003 5:32 PM<BR>&gt; To: sipping@ietf.org<BR>&gt; Subject: RE: =

      [Sipping] I-D=20
      ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; =
Hi=20
      List,<BR>&gt; We would like you take a look at this draft and =
revert with=20
      any<BR>&gt; comments/queries that you might have.<BR>&gt;<BR>&gt; =
SIPPING=20
      Chairs,<BR>&gt; The authors feel that the requirement and the need =
for=20
      this MIME type<BR>&gt; is<BR>&gt; quite acute. Hence we'd like you =
to=20
      consider this for inclusion as a<BR>&gt; WG I-D<BR>&gt; and =
suggest the=20
      way forward.<BR>&gt;<BR>&gt; - ranjith..<BR>&gt;<BR>&gt; =
-----Original=20
      Message-----<BR>&gt; From: sipping-admin@ietf.org [<A=20
      =
href=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
On=20
      Behalf<BR>&gt; Of<BR>&gt; Internet-Drafts@ietf.org<BR>&gt; Sent: =
Tuesday,=20
      February 04, 2003 5:14 PM<BR>&gt; To: IETF-Announce:<BR>&gt; Cc:=20
      sipping@ietf.org<BR>&gt; Subject: [Sipping] I-D=20
      ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; =
A New=20
      Internet-Draft is available from the on-line =
Internet-Drafts<BR>&gt;=20
      directories.<BR>&gt;<BR>&gt;<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
MIME media=20
      types for DPNSS Objects<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et =
al.<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :=20
      draft-mukundan-sipping-dpnss-01.txt<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-2-3<BR>&gt;<BR>&gt; This=20
      document describes Multipart Internet Mail Extensions (MIME) =
type<BR>&gt;=20
      for application/Digital Private Network Signaling System =
(DPNSS)<BR>&gt;=20
      objects for use in Session Initiation Protocol (SIP) =
applications,<BR>&gt;=20
      according to the rules defined in RFC 2048. These types can be =
used=20
      to<BR>&gt; identify Digital Private Network Signaling System 1 =
(DPNSS 1)=20
      objects<BR>&gt; within a SIP message such as INVITE or INFO, as =
might be=20
      implemented<BR>&gt; when using SIP in an environment where part of =
the=20
      call involves<BR>&gt; interworking to DPNSS 1 based Private =
Networks. RFC=20
      3204 addresses a<BR>&gt; similar need for Integrated Service =
Digital=20
      Network User Part (ISUP)<BR>&gt; and Q Signaling =
(QSIG).<BR>&gt;<BR>&gt; A=20
      URL for this Internet-Draft is:<BR>&gt; <A=20
      =
href=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A><BR>&gt;=20
      t<BR>&gt;<BR>&gt; To remove yourself from the IETF Announcement =
list, send=20
      a message to<BR>&gt; ietf-announce-request with the word =
unsubscribe in=20
      the body of the<BR>&gt; message.<BR>&gt;<BR>&gt; Internet-Drafts =
are also=20
      available by anonymous FTP. Login with the<BR>&gt; =
username<BR>&gt;=20
      "anonymous" and a password of your e-mail address. After logging=20
      in,<BR>&gt; type "cd internet-drafts" and then<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "get=20
      draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; A list of=20
      Internet-Drafts directories can be found in<BR>&gt; <A=20
      =
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A>=20
      or<BR>&gt; <A=20
      =
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A><BR>&gt;<BR>&gt;<BR>&gt;=20
      Internet-Drafts can also be obtained by e-mail.<BR>&gt;<BR>&gt; =
Send a=20
      message to:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      mailserv@ietf.org.<BR>&gt; In the body type:<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "FILE=20
      =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt;=20
      NOTE: The mail server at ietf.org can return the document =
in<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
"mpack"=20
      utility.&nbsp; To use this<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
feature,=20
      insert the command "ENCODING mime" before the "FILE"<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s),=20
      you will need "munpack" or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=20
      MIME-compliant mail reader.&nbsp; Different MIME-compliant mail=20
      readers<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different =
behavior,=20
      especially when dealing with<BR>&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      "multipart" MIME messages (i.e. documents which have been =
split<BR>&gt;=20
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so =
check your=20
      local documentation on<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how =
to=20
      manipulate these messages.<BR>&gt;<BR>&gt;<BR>&gt; Below is the =
data which=20
      will enable a MIME compliant mail reader<BR>&gt; implementation to =

      automatically retrieve the ASCII version of the<BR>&gt;=20
      Internet-Draft.<BR>&gt;<BR>&gt;=20
      _______________________________________________<BR>&gt; Sipping =
mailing=20
      list&nbsp; <A=20
      =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A><BR>&gt;=20
      This list is for NEW development of the application of SIP =
Use<BR>&gt;=20
      sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>&gt;=20
      sip@ietf.org for new developments of core=20
      =
SIP<BR><BR><BR><BR><BR></FONT></P>_______________________________________=
________=20
      Sipping mailing list =
https://www1.ietf.org/mailman/listinfo/sipping This=20
      list is for NEW development of the application of SIP Use=20
      sip-implementors@cs.columbia.edu for questions on current sip Use=20
      sip@ietf.org for new developments of core SIP=20
</BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Subject: Re: [Sipping] Question about call flow in
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Hi,

I think there is an error in the call flows.

Either A has to respond with an "a=inactive", so that there is no RTP
flow between the UAs, or else there should be unidirectional RTP flow
between the 2 UAs.

Best Regards
- Ajit

"Miao, Kai" wrote:

>  Could someone provide an answer to the following question:In the
> first example, Call Hold, given
> in draft-ietf-sipping-service-examples-03.txt, why F10, i.e.,
> INVITE(hold), has a=sendonly in its SDP message and F12, i.e., 200 OK,
> has no "a=" attribute at all while "No RTP Sent!" is marked in the
> call flow diagram? Shoudn't F12 contain a=recvonly or a=inactive,
> following the Offer/Answer model? Why there is "no RTP sent" in the
> diagram when there is "a=sendonly" in F10?Thanks,Kai MiaoIntel

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi,
<p>I think there is an error in the call flows.
<p>Either A has to respond with an "a=inactive", so that there is no RTP
flow between the UAs, or else there should be unidirectional RTP flow between
the 2 UAs.
<p>Best Regards
<br>- Ajit
<p>"Miao, Kai" wrote:
<blockquote TYPE=CITE>&nbsp;<span class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>Could
someone provide an answer to the following question:</font></font></font></span><span
class=481480723-11022003></span><span
class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>In
the first example, Call Hold, given in&nbsp;</span>draft-ietf-sipping-service-examples-03.tx<span
class=481480723-11022003>t,
why F10, i.e., INVITE(hold), has a=sendonly in its SDP message and F12,
i.e., 200 OK, has no "a=" attribute at all while "No RTP Sent!" is marked
in the call flow diagram?&nbsp;</font></font></font></span><span
class=481480723-11022003></span><span
class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>Shoudn't
F12 contain a=recvonly or a=inactive, following the Offer/Answer model?
Why there is "no RTP sent" in the diagram when there is "a=sendonly" in
F10?</font></font></font></span><span
class=481480723-11022003></span><span
class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>Thanks,</font></font></font></span><span
class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>Kai
Miao</font></font></font></span><span
class=481480723-11022003><font face="Arial"><font color="#0000FF"><font size=-1>Intel&nbsp;</font></font></font></span></blockquote>
</html>

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_______________________________________________
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Subject: Re: [Sipping] QSIG Vs DPNSS MIME
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My two cents on this stuff (applies to QSIG too).

The idea behind sip-t was that carriage of the isup bodies made sense 
because the semantics of the sip messages matched the isup; the isup was 
providing "extra" information that was simply carried in its native 
form. Having the semantics match is important for use of services in the 
SIP network.

If these DPNSS bodies have a semantic different than the SIP message 
they are carried in (i.e., call transfer is carried in an INFO), then I 
think thats a bad thing and not in line with the sip-t architecture.

Now, I don't know DPNSS really, so I don't know if this is what you are 
proposing. It is just a comment on the general direction.

-Jonathan R.


Peterson, Jon wrote:
>  
> I begin to sympathize with those who argued that the ISUP/QSIG MIME type 
> should have been a more open-ended envelope for telephony signaling 
> information. I am reluctant to standardize every conceivable sort of 
> telephony signaling as a MIME type with some profile information for 
> encapsulating it in SIP.
>  
> I suppose the question really is one of how great the need is to 
> standardize this DPNSS encapsulation. I'm not really familiar with it, 
> so I don't have much of an opinion either way.
>  
> In any event, as another nit, this document should not define the 
> Content-Disposition header, it should reference the definition and BNF 
> in RFC3204.
>  
> Jon Peterson
> NeuStar, Inc.
> 
>     -----Original Message-----
>     *From:* Mark Watson [mailto:mwatson@nortelnetworks.com]
>     *Sent:* Tuesday, February 11, 2003 9:29 AM
>     *To:* sipping@ietf.org
>     *Subject:* RE: [Sipping] QSIG Vs DPNSS MIME
> 
>     All,
>      
>     I really can't see any difference between DPNSS and Q.SIG when it
>     comes to the need for a MIME type for carriage of this within SIP.
>     Can anyone come up with an argument for/against this draft which
>     does not apply to Q.SIG just as well ?
>      
>     If not, then I certainly think this should be progressed.
>      
>     I have some comments on the draft:
>     1) DPNSS has now been published by the UK's Network Interoperability
>     Consultative Committee as NICC Document 1301:2001/03. You can find
>     it at
>     http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm along
>     with lots of other gems.
>      
>     You should reference this version, rather than the somewhat harder
>     to obtain BTNR.
>      
>     2) Why do you need a length octet at the front of the message ?
>     Can't you (or shouldn't you) put a Content-Length in the MIME
>     headers for this MIME part ?
>     3) I think given the short length of the messages you could get away
>     with mandating that segmented DPNSS messages should always be
>     reassembled before encapsulating in SIP and that receivers must be
>     able to receive such reassembled messages. This removes some
>     unnecessary complexity.
>     4) Section 3, under table 1, you say 'Floating Point' when you mean
>     'Decimal'. I think the words 'contiguously increasing' could
>     probably be removed here too. You will have trouble finding the next
>     real number after 6.00!
>      
>     Regards...Mark
>      
> 
>         -----Original Message-----
>         *From:* Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
>         *Sent:* 11 February 2003 08:59
>         *To:* sipping@ietf.org
>         *Subject:* [Sipping] QSIG Vs DPNSS MIME
> 
>         we'd appreciate some views on this draft from atleast those who
>         fairly understand QSIG.
> 
>         of course, others who feel that this MIME type collides directly
>         with the purpose of SIP are encouraged to provide alternatives
>         bearing in mind that providing complete end-to-end service
>         transparency over the intervening SIP based core would be
>         essential, if we are to provide a seamless migration path from
>         H.323 based core networks - which already provide equivalent
>         mechanisms for DSS1 based protocols.
> 
>         - ranjith..
> 
> 
>         -----Original Message-----
>         From:   Ranjith Mukundan
>         Sent:   Sat 2/8/2003 12:09 AM
>         To:     sipping@ietf.org
>         Cc:    
>         Subject:        RE: [Sipping] I-D
>         ACTION:draft-mukundan-sipping-dpnss-01.txt
> 
>         Hi,
> 
>         posting WRT
>         http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
> 
> 
>          >>
>          >> Why is it so important to be able to tunnel DPNSS in SIP? 
>         I'd prefer
>          >> that folks implement a translation to SIP, where SIP can
>         natively carry
>          >> all the information that DPNSS is trying to convey.
>          >>
> 
>         To put things in perspective:
> 
>         1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia
>         and parts of NA) very similar to QSIG (deployed primarily
>         Europe) for interconnecting PBXs or centrex customers hosted off
>         LEs. DPNSS (like QSIG) finds enormous application towards
>         realizing a local VPN or network-wide VPN services.
> 
>         2- This draft aims to do exactly what RFC 3204 (MIME objects for
>         QSIG/ISUP) does for QSIG with appropriate/relevant customization
>         for DPNSS - i.e., to provide end-to-end & *complete*, DPNSS
>         Service transparency between pockets of DPNSS based private
>         networks over a SIP based core.
> 
>         3- Currently, circuit-switched SS7 based cores, provide
>         equivalent DPNSS tunneling mechanisms through (ETSI & ANSI) ISUP
>         *apart from* the concomitant translation/interworking mechanisms
>         to ISUP. The translation/interworking mechanism to ISUP is
>         limited to the basic call and certain ISUP supplementary
>         services where there is a direct/easy equivalent.
> 
>         4- H.323 also provides equivalent tunneling mechanisms (via the
>         H.450 Annexe procedures) for transparently carrying DSS1 based
>         signaling between DSS1-based network elements/gateways, over a
>         H.323 core.
> 
>         Some specifics:
> 
>         1- DPNSS supports, as specified in BTNR 188, approximately 240+
>         supplementary features/services (like Call Back When Free, Call
>         Back When Next Used, Executive Intrusion, Travelling Class of
>         Service, Do Not Disturb, Night Service, Private Network Route
>         Optimization, Centralized Operator, Three Party Takeover, Remote
>         Diversion Registration etc). It would be almost impossible to
>         interwork to equivalent mechanisms in SIP, without losing the
>         service transparency when using an intervening SIP based core.
> 
>         2- Yes, we definitely need an interworking/translation mechanism
>         to map all the basic call/service paremeters, and procedures for
>         mapping the messages. But this does not preclude the need for a
>         DPNSS MIME type, but would rather be complimentary to the MIME
>         type mechanism. Similar to
>         http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
>         , for QSIG, we are in the process of putting together an
>         interworking draft for DPNSS as well - but rather than making it
>         DPNSS specific, we are trying to make it a common draft to cover
>         PRI, DPNSS & DASS2 [as Q.SIG interworking is already a SIPPING
>         WG ID]. Note that, PRI and DASS2 will not need a MIME type as
>         these are, as one might be aware, for PSTN access, unlike DPNSS
>         and QSIG (which are private network protocols).
> 
>         3- From the SIP perspective, carrying non-SIP applications as
>         MIME objects to ensure end to end service transparency is
>         architecturally less appalling than, say for e.g., tunneling
>         DSS1 protocols in H.323 based network - so the SIP DPNSS MIME
>         proposition as per this I-D is the obvious way forward for DPNSS
>         service transparency over a SIP based core, especially now that
>         we see acute deployment needs to be met.
> 
>         cheers,
>         ranjith..
> 
>          > -----Original Message-----
>          > From: sipping-admin@ietf.org
>         [mailto:sipping-admin@ietf.org]On Behalf
>          > Of
>          > Ranjith Mukundan
>          > Sent: Tuesday, February 04, 2003 5:32 PM
>          > To: sipping@ietf.org
>          > Subject: RE: [Sipping] I-D
>         ACTION:draft-mukundan-sipping-dpnss-01.txt
>          >
>          >
>          > Hi List,
>          > We would like you take a look at this draft and revert with any
>          > comments/queries that you might have.
>          >
>          > SIPPING Chairs,
>          > The authors feel that the requirement and the need for this
>         MIME type
>          > is
>          > quite acute. Hence we'd like you to consider this for
>         inclusion as a
>          > WG I-D
>          > and suggest the way forward.
>          >
>          > - ranjith..
>          >
>          > -----Original Message-----
>          > From: sipping-admin@ietf.org
>         [mailto:sipping-admin@ietf.org]On Behalf
>          > Of
>          > Internet-Drafts@ietf.org
>          > Sent: Tuesday, February 04, 2003 5:14 PM
>          > To: IETF-Announce:
>          > Cc: sipping@ietf.org
>          > Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>          >
>          >
>          > A New Internet-Draft is available from the on-line
>         Internet-Drafts
>          > directories.
>          >
>          >
>          >       Title           : MIME media types for DPNSS Objects
>          >       Author(s)       : R. Mukundan et al.
>          >       Filename        : draft-mukundan-sipping-dpnss-01.txt
>          >       Pages           : 9
>          >       Date            : 2003-2-3
>          >
>          > This document describes Multipart Internet Mail Extensions
>         (MIME) type
>          > for application/Digital Private Network Signaling System (DPNSS)
>          > objects for use in Session Initiation Protocol (SIP)
>         applications,
>          > according to the rules defined in RFC 2048. These types can
>         be used to
>          > identify Digital Private Network Signaling System 1 (DPNSS 1)
>         objects
>          > within a SIP message such as INVITE or INFO, as might be
>         implemented
>          > when using SIP in an environment where part of the call involves
>          > interworking to DPNSS 1 based Private Networks. RFC 3204
>         addresses a
>          > similar need for Integrated Service Digital Network User Part
>         (ISUP)
>          > and Q Signaling (QSIG).
>          >
>          > A URL for this Internet-Draft is:
>          >
>         http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
>          > t
>          >
>          > To remove yourself from the IETF Announcement list, send a
>         message to
>          > ietf-announce-request with the word unsubscribe in the body
>         of the
>          > message.
>          >
>          > Internet-Drafts are also available by anonymous FTP. Login
>         with the
>          > username
>          > "anonymous" and a password of your e-mail address. After
>         logging in,
>          > type "cd internet-drafts" and then
>          >       "get draft-mukundan-sipping-dpnss-01.txt".
>          >
>          > A list of Internet-Drafts directories can be found in
>          > http://www.ietf.org/shadow.html or
>          > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>          >
>          >
>          > Internet-Drafts can also be obtained by e-mail.
>          >
>          > Send a message to:
>          >       mailserv@ietf.org.
>          > In the body type:
>          >       "FILE
>         /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
>          >
>          > NOTE: The mail server at ietf.org can return the document in
>          >       MIME-encoded form by using the "mpack" utility.  To use
>         this
>          >       feature, insert the command "ENCODING mime" before the
>         "FILE"
>          >       command.  To decode the response(s), you will need
>         "munpack" or
>          >       a MIME-compliant mail reader.  Different MIME-compliant
>         mail readers
>          >       exhibit different behavior, especially when dealing with
>          >       "multipart" MIME messages (i.e. documents which have
>         been split
>          >       up into multiple messages), so check your local
>         documentation on
>          >       how to manipulate these messages.
>          >
>          >
>          > Below is the data which will enable a MIME compliant mail reader
>          > implementation to automatically retrieve the ASCII version of the
>          > Internet-Draft.
>          >
>          > _______________________________________________
>          > Sipping mailing list 
>         https://www1.ietf.org/mailman/listinfo/sipping
>          > This list is for NEW development of the application of SIP Use
>          > sip-implementors@cs.columbia.edu for questions on current sip Use
>          > sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
>         _______________________________________________ Sipping mailing
>         list https://www1.ietf.org/mailman/listinfo/sipping This list is
>         for NEW development of the application of SIP Use
>         sip-implementors@cs.columbia.edu for questions on current sip
>         Use sip@ietf.org for new developments of core SIP 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb 12 00:51:17 2003
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Date: Wed, 12 Feb 2003 00:53:29 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] updated conferencing framework
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Folks,

I've submitted an update to the conferencing framework document. THis is 
the result of a lot of work of the conferencing design team to make 
things consistent across documents, and to resolve hard problems like 
how sidebars are handled. Until it appears in the archives you can pick 
up a copy at:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-conferencing-framework-01.txt

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb 12 01:05:31 2003
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Message-ID: <3E49E4B5.8010608@dynamicsoft.com>
Date: Wed, 12 Feb 2003 01:07:49 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] New I-D on URI leasing
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Folks,

I've just submitted an I-D on the concept of "URI leasing". THis is my 
proposed technique for solving the "globally routable UA URI" problem. 
SPecifically, for cases like REFER, how do we construct a URI that 
routes to a specific UA instance in a domain? At the last ietf meeting, 
this issue came up in several contexts. I suggested a single document to 
explore the problem. The chairs asked for volunteers, and I said I'd 
write one, time permitting, if no one else stepped up. Since no one did, 
well, here it is.

Until it appears in the archives, you can pick it up at:
http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.txt
http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.html

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb 12 02:17:26 2003
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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
To: "Peterson, Jon" <jon.peterson@neustar.biz>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sipping@ietf.org>, "'Mark Watson'" <mwatson@nortelnetworks.com>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Wed, 12 Feb 2003 12:52:28 +0530
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In case of DPNSS too the message semantics matches SIP messages - atleast to
the same extent as it does for ISUP. The primary driver for carrying the
DPNSS objects is to seamlessly transfer the DPNSS supplementary service
information across pockets of DPNSS private networks (including DPNSS based
centrex customer groups) over an intervening SIP based core, in its native
form.

In fact, from the message semantics point of view (as opposed to the
supplementary service procedures and info elements point of view); as far as
the session setup, control and release messages go, DPNSS messages (ISRM -
for setup, NAM-for alerting, CRM/CIM- for rel/rel-ack, ANS, EEM - for end to
end 'pass along' messages etc) are more aligned to ISUP message semantics
than perhaps to QSIG message semantics. For e.g., in case of QSIG some of
the session related service messages like NOTIFY, INFORMATION, FAC - would
all get carried in SIP INFO message, while in the case of DPNSS, only EEM
(equivalent of PAM in ANSI ISUP) would get carried in the SIP INFO and hence
finds a better semantic match to SIP messages - perhaps a better match when
compared to QSIG.

It's only wrt the DPNSS overlap messages and layer 3 maintenance messages
(like LLM, LRM etc) that finding a semantically good message fit to SIP
becomes diffucult - but equivalent constraints exists in case of ISUP
overlap and maintenance messages (Blocking, Reset, Cirtuit Validation Test
etc) too.

We could include some typical service flows in the appendix of the this
draft to reinforce the fact that DPNSS layer 3 messages does in fact render
itself for a better semantic message fit with SIP.

cheers,
- ranjith..



-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Jonathan Rosenberg
Sent: Wednesday, February 12, 2003 11:18 AM
To: Peterson, Jon
Cc: 'Mark Watson'; sipping@ietf.org
Subject: Re: [Sipping] QSIG Vs DPNSS MIME


My two cents on this stuff (applies to QSIG too).

The idea behind sip-t was that carriage of the isup bodies made sense
because the semantics of the sip messages matched the isup; the isup was
providing "extra" information that was simply carried in its native
form. Having the semantics match is important for use of services in the
SIP network.

If these DPNSS bodies have a semantic different than the SIP message
they are carried in (i.e., call transfer is carried in an INFO), then I
think thats a bad thing and not in line with the sip-t architecture.

Now, I don't know DPNSS really, so I don't know if this is what you are
proposing. It is just a comment on the general direction.

-Jonathan R.


Peterson, Jon wrote:
>
> I begin to sympathize with those who argued that the ISUP/QSIG MIME type
> should have been a more open-ended envelope for telephony signaling
> information. I am reluctant to standardize every conceivable sort of
> telephony signaling as a MIME type with some profile information for
> encapsulating it in SIP.
>
> I suppose the question really is one of how great the need is to
> standardize this DPNSS encapsulation. I'm not really familiar with it,
> so I don't have much of an opinion either way.
>
> In any event, as another nit, this document should not define the
> Content-Disposition header, it should reference the definition and BNF
> in RFC3204.
>
> Jon Peterson
> NeuStar, Inc.
>
>     -----Original Message-----
>     *From:* Mark Watson [mailto:mwatson@nortelnetworks.com]
>     *Sent:* Tuesday, February 11, 2003 9:29 AM
>     *To:* sipping@ietf.org
>     *Subject:* RE: [Sipping] QSIG Vs DPNSS MIME
>
>     All,
>
>     I really can't see any difference between DPNSS and Q.SIG when it
>     comes to the need for a MIME type for carriage of this within SIP.
>     Can anyone come up with an argument for/against this draft which
>     does not apply to Q.SIG just as well ?
>
>     If not, then I certainly think this should be progressed.
>
>     I have some comments on the draft:
>     1) DPNSS has now been published by the UK's Network Interoperability
>     Consultative Committee as NICC Document 1301:2001/03. You can find
>     it at
>
http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm
along
>     with lots of other gems.
>
>     You should reference this version, rather than the somewhat harder
>     to obtain BTNR.
>
>     2) Why do you need a length octet at the front of the message ?
>     Can't you (or shouldn't you) put a Content-Length in the MIME
>     headers for this MIME part ?
>     3) I think given the short length of the messages you could get away
>     with mandating that segmented DPNSS messages should always be
>     reassembled before encapsulating in SIP and that receivers must be
>     able to receive such reassembled messages. This removes some
>     unnecessary complexity.
>     4) Section 3, under table 1, you say 'Floating Point' when you mean
>     'Decimal'. I think the words 'contiguously increasing' could
>     probably be removed here too. You will have trouble finding the next
>     real number after 6.00!
>
>     Regards...Mark
>
>
>         -----Original Message-----
>         *From:* Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
>         *Sent:* 11 February 2003 08:59
>         *To:* sipping@ietf.org
>         *Subject:* [Sipping] QSIG Vs DPNSS MIME
>
>         we'd appreciate some views on this draft from atleast those who
>         fairly understand QSIG.
>
>         of course, others who feel that this MIME type collides directly
>         with the purpose of SIP are encouraged to provide alternatives
>         bearing in mind that providing complete end-to-end service
>         transparency over the intervening SIP based core would be
>         essential, if we are to provide a seamless migration path from
>         H.323 based core networks - which already provide equivalent
>         mechanisms for DSS1 based protocols.
>
>         - ranjith..
>
>
>         -----Original Message-----
>         From:   Ranjith Mukundan
>         Sent:   Sat 2/8/2003 12:09 AM
>         To:     sipping@ietf.org
>         Cc:
>         Subject:        RE: [Sipping] I-D
>         ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>         Hi,
>
>         posting WRT
>
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
>
>
>          >>
>          >> Why is it so important to be able to tunnel DPNSS in SIP?
>         I'd prefer
>          >> that folks implement a translation to SIP, where SIP can
>         natively carry
>          >> all the information that DPNSS is trying to convey.
>          >>
>
>         To put things in perspective:
>
>         1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia
>         and parts of NA) very similar to QSIG (deployed primarily
>         Europe) for interconnecting PBXs or centrex customers hosted off
>         LEs. DPNSS (like QSIG) finds enormous application towards
>         realizing a local VPN or network-wide VPN services.
>
>         2- This draft aims to do exactly what RFC 3204 (MIME objects for
>         QSIG/ISUP) does for QSIG with appropriate/relevant customization
>         for DPNSS - i.e., to provide end-to-end & *complete*, DPNSS
>         Service transparency between pockets of DPNSS based private
>         networks over a SIP based core.
>
>         3- Currently, circuit-switched SS7 based cores, provide
>         equivalent DPNSS tunneling mechanisms through (ETSI & ANSI) ISUP
>         *apart from* the concomitant translation/interworking mechanisms
>         to ISUP. The translation/interworking mechanism to ISUP is
>         limited to the basic call and certain ISUP supplementary
>         services where there is a direct/easy equivalent.
>
>         4- H.323 also provides equivalent tunneling mechanisms (via the
>         H.450 Annexe procedures) for transparently carrying DSS1 based
>         signaling between DSS1-based network elements/gateways, over a
>         H.323 core.
>
>         Some specifics:
>
>         1- DPNSS supports, as specified in BTNR 188, approximately 240+
>         supplementary features/services (like Call Back When Free, Call
>         Back When Next Used, Executive Intrusion, Travelling Class of
>         Service, Do Not Disturb, Night Service, Private Network Route
>         Optimization, Centralized Operator, Three Party Takeover, Remote
>         Diversion Registration etc). It would be almost impossible to
>         interwork to equivalent mechanisms in SIP, without losing the
>         service transparency when using an intervening SIP based core.
>
>         2- Yes, we definitely need an interworking/translation mechanism
>         to map all the basic call/service paremeters, and procedures for
>         mapping the messages. But this does not preclude the need for a
>         DPNSS MIME type, but would rather be complimentary to the MIME
>         type mechanism. Similar to
>
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
>         , for QSIG, we are in the process of putting together an
>         interworking draft for DPNSS as well - but rather than making it
>         DPNSS specific, we are trying to make it a common draft to cover
>         PRI, DPNSS & DASS2 [as Q.SIG interworking is already a SIPPING
>         WG ID]. Note that, PRI and DASS2 will not need a MIME type as
>         these are, as one might be aware, for PSTN access, unlike DPNSS
>         and QSIG (which are private network protocols).
>
>         3- From the SIP perspective, carrying non-SIP applications as
>         MIME objects to ensure end to end service transparency is
>         architecturally less appalling than, say for e.g., tunneling
>         DSS1 protocols in H.323 based network - so the SIP DPNSS MIME
>         proposition as per this I-D is the obvious way forward for DPNSS
>         service transparency over a SIP based core, especially now that
>         we see acute deployment needs to be met.
>
>         cheers,
>         ranjith..
>
>          > -----Original Message-----
>          > From: sipping-admin@ietf.org
>         [mailto:sipping-admin@ietf.org]On Behalf
>          > Of
>          > Ranjith Mukundan
>          > Sent: Tuesday, February 04, 2003 5:32 PM
>          > To: sipping@ietf.org
>          > Subject: RE: [Sipping] I-D
>         ACTION:draft-mukundan-sipping-dpnss-01.txt
>          >
>          >
>          > Hi List,
>          > We would like you take a look at this draft and revert with any
>          > comments/queries that you might have.
>          >
>          > SIPPING Chairs,
>          > The authors feel that the requirement and the need for this
>         MIME type
>          > is
>          > quite acute. Hence we'd like you to consider this for
>         inclusion as a
>          > WG I-D
>          > and suggest the way forward.
>          >
>          > - ranjith..
>          >
>          > -----Original Message-----
>          > From: sipping-admin@ietf.org
>         [mailto:sipping-admin@ietf.org]On Behalf
>          > Of
>          > Internet-Drafts@ietf.org
>          > Sent: Tuesday, February 04, 2003 5:14 PM
>          > To: IETF-Announce:
>          > Cc: sipping@ietf.org
>          > Subject: [Sipping] I-D
ACTION:draft-mukundan-sipping-dpnss-01.txt
>          >
>          >
>          > A New Internet-Draft is available from the on-line
>         Internet-Drafts
>          > directories.
>          >
>          >
>          >       Title           : MIME media types for DPNSS Objects
>          >       Author(s)       : R. Mukundan et al.
>          >       Filename        : draft-mukundan-sipping-dpnss-01.txt
>          >       Pages           : 9
>          >       Date            : 2003-2-3
>          >
>          > This document describes Multipart Internet Mail Extensions
>         (MIME) type
>          > for application/Digital Private Network Signaling System
(DPNSS)
>          > objects for use in Session Initiation Protocol (SIP)
>         applications,
>          > according to the rules defined in RFC 2048. These types can
>         be used to
>          > identify Digital Private Network Signaling System 1 (DPNSS 1)
>         objects
>          > within a SIP message such as INVITE or INFO, as might be
>         implemented
>          > when using SIP in an environment where part of the call
involves
>          > interworking to DPNSS 1 based Private Networks. RFC 3204
>         addresses a
>          > similar need for Integrated Service Digital Network User Part
>         (ISUP)
>          > and Q Signaling (QSIG).
>          >
>          > A URL for this Internet-Draft is:
>          >
>
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
>          > t
>          >
>          > To remove yourself from the IETF Announcement list, send a
>         message to
>          > ietf-announce-request with the word unsubscribe in the body
>         of the
>          > message.
>          >
>          > Internet-Drafts are also available by anonymous FTP. Login
>         with the
>          > username
>          > "anonymous" and a password of your e-mail address. After
>         logging in,
>          > type "cd internet-drafts" and then
>          >       "get draft-mukundan-sipping-dpnss-01.txt".
>          >
>          > A list of Internet-Drafts directories can be found in
>          > http://www.ietf.org/shadow.html or
>          > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>          >
>          >
>          > Internet-Drafts can also be obtained by e-mail.
>          >
>          > Send a message to:
>          >       mailserv@ietf.org.
>          > In the body type:
>          >       "FILE
>         /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
>          >
>          > NOTE: The mail server at ietf.org can return the document in
>          >       MIME-encoded form by using the "mpack" utility.  To use
>         this
>          >       feature, insert the command "ENCODING mime" before the
>         "FILE"
>          >       command.  To decode the response(s), you will need
>         "munpack" or
>          >       a MIME-compliant mail reader.  Different MIME-compliant
>         mail readers
>          >       exhibit different behavior, especially when dealing with
>          >       "multipart" MIME messages (i.e. documents which have
>         been split
>          >       up into multiple messages), so check your local
>         documentation on
>          >       how to manipulate these messages.
>          >
>          >
>          > Below is the data which will enable a MIME compliant mail
reader
>          > implementation to automatically retrieve the ASCII version of
the
>          > Internet-Draft.
>          >
>          > _______________________________________________
>          > Sipping mailing list
>         https://www1.ietf.org/mailman/listinfo/sipping
>          > This list is for NEW development of the application of SIP Use
>          > sip-implementors@cs.columbia.edu for questions on current sip
Use
>          > sip@ietf.org for new developments of core SIP
>
>
>
>
>         _______________________________________________ Sipping mailing
>         list https://www1.ietf.org/mailman/listinfo/sipping This list is
>         for NEW development of the application of SIP Use
>         sip-implementors@cs.columbia.edu for questions on current sip
>         Use sip@ietf.org for new developments of core SIP
>

--
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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http://www.dynamicsoft.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Ranjith Mukundan'" <ranjith.mukundan@wipro.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>, sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
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I can't think of any other telephony signalling protocols which would
warrent MIME encapsulation in SIP either. Service Transparency is only an
issue when it's a peer-to-peer protocol that you are considering (if it's
not peer-to-peer e.g. DSS1, then there's no transparency concept in the
original protocol to be carried over into SIP - every node needs to
understand every feature, in which case a SIP gateway can interwork to
appropriate SIP features or correctly terminate the feature on the telephony
side).
 
National legacy PSTN signalling aside, the only standard public network
signalling is ISUP.
 
The only standard private network signalling systems that I know of are
Q.SIG and DPNSS. Of coure there are lot of proprietary ones, but I don't
think we should consider MIME types for these.
 
...Mark

-----Original Message-----
From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
Sent: 12 February 2003 03:40
To: Peterson, Jon; Watson, Mark [MOP:EP10:EXCH]; sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME


in so far as inter-PBX/Centrex connectivity goes, amongst the DSS1 derived
protocols, DPNSS is, IMO, the *only*  (big) missing piece. Can anyone think
of another, standards-based, telephony signaling, the might require a MIME
carriage?? (atleast, i can't think of any - not even an arcane, minimally
deployed one) 
 
As I mentioned earlier, we dont need MIME types for protocols like DASS2,
PRI and all the other 'conceivable sort' of telephony signaling thats used
for PSTN connectivity (vis-a-vis inter-PBX/centrex connectivity).  Even in
the case of telephony protocols used for inter-PBX/centrex connectivity,
unless it requires the same end to end, complete service transparency as in
the case of private networks interconnected by QSIG or DPNSS, it does not
mandate carriage as MIME type (a simple interworking function to SIP would
suffice).  
 
So I dont think we'll a slew of MIME types...
 

>> In any event, as another nit, this document should not define the
Content-Disposition header, it should reference the definition and BNF in
RFC3204.
Should've known. thanks - it was an oversight.
 
cheers,
- ranjith..

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Peterson, Jon
Sent: Wednesday, February 12, 2003 1:28 AM
To: 'Mark Watson'; sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME


 
I begin to sympathize with those who argued that the ISUP/QSIG MIME type
should have been a more open-ended envelope for telephony signaling
information. I am reluctant to standardize every conceivable sort of
telephony signaling as a MIME type with some profile information for
encapsulating it in SIP.
 
I suppose the question really is one of how great the need is to standardize
this DPNSS encapsulation. I'm not really familiar with it, so I don't have
much of an opinion either way.
 
In any event, as another nit, this document should not define the
Content-Disposition header, it should reference the definition and BNF in
RFC3204.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 11, 2003 9:29 AM
To: sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME


All,
 
I really can't see any difference between DPNSS and Q.SIG when it comes to
the need for a MIME type for carriage of this within SIP. Can anyone come up
with an argument for/against this draft which does not apply to Q.SIG just
as well ?
 
If not, then I certainly think this should be progressed.
 
I have some comments on the draft:
1) DPNSS has now been published by the UK's Network Interoperability
Consultative Committee as NICC Document 1301:2001/03. You can find it at
http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm
<http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.ht
m>  along with lots of other gems.
 
You should reference this version, rather than the somewhat harder to obtain
BTNR.
 
2) Why do you need a length octet at the front of the message ? Can't you
(or shouldn't you) put a Content-Length in the MIME headers for this MIME
part ?
3) I think given the short length of the messages you could get away with
mandating that segmented DPNSS messages should always be reassembled before
encapsulating in SIP and that receivers must be able to receive such
reassembled messages. This removes some unnecessary complexity.
4) Section 3, under table 1, you say 'Floating Point' when you mean
'Decimal'. I think the words 'contiguously increasing' could probably be
removed here too. You will have trouble finding the next real number after
6.00!
 
Regards...Mark
 

-----Original Message-----
From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
Sent: 11 February 2003 08:59
To: sipping@ietf.org
Subject: [Sipping] QSIG Vs DPNSS MIME



we'd appreciate some views on this draft from atleast those who fairly
understand QSIG.

of course, others who feel that this MIME type collides directly with the
purpose of SIP are encouraged to provide alternatives bearing in mind that
providing complete end-to-end service transparency over the intervening SIP
based core would be essential, if we are to provide a seamless migration
path from H.323 based core networks - which already provide equivalent
mechanisms for DSS1 based protocols.

- ranjith..


-----Original Message-----
From:   Ranjith Mukundan
Sent:   Sat 2/8/2003 12:09 AM
To:     sipping@ietf.org
Cc:    
Subject:        RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt

Hi,

posting WRT
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt> 


>>
>> Why is it so important to be able to tunnel DPNSS in SIP?  I'd prefer
>> that folks implement a translation to SIP, where SIP can natively carry
>> all the information that DPNSS is trying to convey.
>>

To put things in perspective:

1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and parts of
NA) very similar to QSIG (deployed primarily Europe) for interconnecting
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous
application towards realizing a local VPN or network-wide VPN services.

2- This draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP)
does for QSIG with appropriate/relevant customization for DPNSS - i.e., to
provide end-to-end & *complete*, DPNSS Service transparency between pockets
of DPNSS based private networks over a SIP based core.

3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS
tunneling mechanisms through (ETSI & ANSI) ISUP *apart from* the concomitant
translation/interworking mechanisms to ISUP. The translation/interworking
mechanism to ISUP is limited to the basic call and certain ISUP
supplementary services where there is a direct/easy equivalent.

4- H.323 also provides equivalent tunneling mechanisms (via the H.450 Annexe
procedures) for transparently carrying DSS1 based signaling between
DSS1-based network elements/gateways, over a H.323 core.

Some specifics:

1- DPNSS supports, as specified in BTNR 188, approximately 240+
supplementary features/services (like Call Back When Free, Call Back When
Next Used, Executive Intrusion, Travelling Class of Service, Do Not Disturb,
Night Service, Private Network Route Optimization, Centralized Operator,
Three Party Takeover, Remote Diversion Registration etc). It would be almost
impossible to interwork to equivalent mechanisms in SIP, without losing the
service transparency when using an intervening SIP based core.

2- Yes, we definitely need an interworking/translation mechanism to map all
the basic call/service paremeters, and procedures for mapping the messages.
But this does not preclude the need for a DPNSS MIME type, but would rather
be complimentary to the MIME type mechanism. Similar to
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt>  ,
for QSIG, we are in the process of putting together an interworking draft
for DPNSS as well - but rather than making it DPNSS specific, we are trying
to make it a common draft to cover PRI, DPNSS & DASS2 [as Q.SIG interworking
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME
type as these are, as one might be aware, for PSTN access, unlike DPNSS and
QSIG (which are private network protocols).

3- From the SIP perspective, carrying non-SIP applications as MIME objects
to ensure end to end service transparency is architecturally less appalling
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so the
SIP DPNSS MIME proposition as per this I-D is the obvious way forward for
DPNSS service transparency over a SIP based core, especially now that we see
acute deployment needs to be met.

cheers,
ranjith..

> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Ranjith Mukundan
> Sent: Tuesday, February 04, 2003 5:32 PM
> To: sipping@ietf.org
> Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> Hi List,
> We would like you take a look at this draft and revert with any
> comments/queries that you might have.
>
> SIPPING Chairs,
> The authors feel that the requirement and the need for this MIME type
> is
> quite acute. Hence we'd like you to consider this for inclusion as a
> WG I-D
> and suggest the way forward.
>
> - ranjith..
>
> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Internet-Drafts@ietf.org
> Sent: Tuesday, February 04, 2003 5:14 PM
> To: IETF-Announce:
> Cc: sipping@ietf.org
> Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>       Title           : MIME media types for DPNSS Objects
>       Author(s)       : R. Mukundan et al.
>       Filename        : draft-mukundan-sipping-dpnss-01.txt
>       Pages           : 9
>       Date            : 2003-2-3
>
> This document describes Multipart Internet Mail Extensions (MIME) type
> for application/Digital Private Network Signaling System (DPNSS)
> objects for use in Session Initiation Protocol (SIP) applications,
> according to the rules defined in RFC 2048. These types can be used to
> identify Digital Private Network Signaling System 1 (DPNSS 1) objects
> within a SIP message such as INVITE or INFO, as might be implemented
> when using SIP in an environment where part of the call involves
> interworking to DPNSS 1 based Private Networks. RFC 3204 addresses a
> similar need for Integrated Service Digital Network User Part (ISUP)
> and Q Signaling (QSIG).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx> 
> t
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the
> message.
>
> Internet-Drafts are also available by anonymous FTP. Login with the
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>       "get draft-mukundan-sipping-dpnss-01.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html>  or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
<ftp://ftp.ietf.org/ietf/1shadow-sites.txt> 
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
>       mailserv@ietf.org.
> In the body type:
>       "FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
>
> NOTE: The mail server at ietf.org can return the document in
>       MIME-encoded form by using the "mpack" utility.  To use this
>       feature, insert the command "ENCODING mime" before the "FILE"
>       command.  To decode the response(s), you will need "munpack" or
>       a MIME-compliant mail reader.  Different MIME-compliant mail readers
>       exhibit different behavior, especially when dealing with
>       "multipart" MIME messages (i.e. documents which have been split
>       up into multiple messages), so check your local documentation on
>       how to manipulate these messages.
>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
<https://www1.ietf.org/mailman/listinfo/sipping> 
> This list is for NEW development of the application of SIP Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sip@ietf.org for new developments of core SIP






_______________________________________________ Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
development of the application of SIP Use sip-implementors@cs.columbia.edu
for questions on current sip Use sip@ietf.org for new developments of core
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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=518111009-12022003>I 
can't think of any other telephony signalling protocols which would warrent MIME 
encapsulation in SIP either. Service Transparency is only an issue when it's a 
peer-to-peer protocol that you are considering (if it's not peer-to-peer e.g. 
DSS1, then there's no transparency concept in the original protocol to be 
carried over into SIP - every node needs to understand every feature, in which 
case a SIP gateway can interwork to appropriate SIP features or correctly 
terminate the feature on the telephony side).</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=518111009-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=518111009-12022003>National legacy PSTN signalling aside, the only 
standard public network signalling is ISUP.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=518111009-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=518111009-12022003>The 
only standard private network signalling systems that I know of are Q.SIG and 
DPNSS. Of coure there are lot of proprietary ones, but I don't think we should 
consider MIME types for these.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=518111009-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=518111009-12022003>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ranjith Mukundan 
  [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 12 February 2003 
  03:40<BR><B>To:</B> Peterson, Jon; Watson, Mark [MOP:EP10:EXCH]; 
  sipping@ietf.org<BR><B>Subject:</B> RE: [Sipping] QSIG Vs DPNSS 
  MIME<BR><BR></DIV></FONT>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial size=2>in 
  so far as inter-PBX/Centrex connectivity goes, amongst the DSS1 derived 
  protocols, DPNSS is, IMO, the *only*&nbsp; (big) missing piece. Can anyone 
  think of another, standards-based, telephony signaling, the might require a 
  MIME carriage?? (atleast, i can't think of any - not even an arcane, minimally 
  deployed one) </FONT></SPAN></DIV>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial size=2>As I 
  mentioned earlier, we dont need MIME types for protocols like DASS2, PRI and 
  all the other 'conceivable sort' of telephony signaling thats used for PSTN 
  connectivity (vis-a-vis inter-PBX/centrex connectivity). &nbsp;Even in the 
  case of telephony protocols used for inter-PBX/centrex connectivity, unless it 
  requires the same end to end, complete service transparency as in the case of 
  private networks&nbsp;interconnected by&nbsp;QSIG or DPNSS, it does not 
  mandate&nbsp;carriage as&nbsp;MIME type (a simple interworking function to SIP 
  would suffice).&nbsp;&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial size=2>So I 
  dont think we'll a slew of MIME types...</FONT></SPAN></DIV>
  <DIV><SPAN class=058362703-12022003><FONT color=#0000ff face=Arial 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=058362703-12022003>
  <DIV><SPAN class=798504719-11022003><FONT face=Arial><FONT size=2><SPAN 
  class=058362703-12022003>&gt;&gt; </SPAN>In any event,&nbsp;as another nit, 
  this document should not define the Content-Disposition header, it should 
  reference the definition and BNF in RFC3204.</FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=798504719-11022003><SPAN class=058362703-12022003><FONT 
  color=#0000ff face=Arial size=2>Should've known. thanks - it was an 
  oversight.</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=798504719-11022003><SPAN class=058362703-12022003><FONT 
  color=#0000ff face=Arial size=2></FONT></SPAN></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=798504719-11022003><SPAN class=058362703-12022003><FONT 
  color=#0000ff face=Arial size=2>cheers,</FONT></SPAN></SPAN></DIV>
  <DIV><SPAN class=798504719-11022003><SPAN class=058362703-12022003><FONT 
  color=#0000ff face=Arial size=2>- 
  ranjith..</FONT></SPAN></SPAN></DIV></SPAN></DIV>
  <BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
    <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
    size=2>-----Original Message-----<BR><B>From:</B> sipping-admin@ietf.org 
    [mailto:sipping-admin@ietf.org]<B>On Behalf Of </B>Peterson, 
    Jon<BR><B>Sent:</B> Wednesday, February 12, 2003 1:28 AM<BR><B>To:</B> 'Mark 
    Watson'; sipping@ietf.org<BR><B>Subject:</B> RE: [Sipping] QSIG Vs DPNSS 
    MIME<BR><BR></FONT></DIV>
    <DIV><FONT color=#0000ff face=Arial size=2></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=798504719-11022003>I 
    begin to sympathize with those who argued that the ISUP/QSIG MIME type 
    should have been a more open-ended envelope for telephony signaling 
    information. I am reluctant to standardize every conceivable sort of 
    telephony signaling as a MIME type with some profile information for 
    encapsulating it in SIP.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
    class=798504719-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=798504719-11022003>I 
    suppose the question really is one of how great the need is to standardize 
    this DPNSS encapsulation. I'm not really familiar with it, so I don't have 
    much of an opinion either way.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN 
    class=798504719-11022003></SPAN></FONT>&nbsp;</DIV>
    <DIV><FONT color=#0000ff face=Arial size=2><SPAN class=798504719-11022003>In 
    any event,&nbsp;as another nit, this document should not define the 
    Content-Disposition header, it should reference the definition and BNF in 
    RFC3204.</SPAN></FONT></DIV>
    <DIV><FONT color=#0000ff face=Arial size=2></FONT>&nbsp;</DIV>
    <DIV><SPAN class=798504719-11022003><FONT color=#0000ff face=Arial 
    size=2>Jon Peterson</FONT></SPAN></DIV>
    <DIV><SPAN class=798504719-11022003><FONT color=#0000ff face=Arial 
    size=2>NeuStar, Inc.</FONT></SPAN></DIV>
    <BLOCKQUOTE dir=ltr 
    style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
      <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
      size=2>-----Original Message-----<BR><B>From:</B> Mark Watson 
      [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Tuesday, February 11, 
      2003 9:29 AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> RE: 
      [Sipping] QSIG Vs DPNSS MIME<BR><BR></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>All,</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>I really can't see any difference between DPNSS 
      and Q.SIG when it comes to the need for a MIME type for carriage of this 
      within SIP. Can anyone come up with an argument for/against this draft 
      which does not apply to Q.SIG just as well ?</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>If not, then I certainly think this should be 
      progressed.</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>I have some comments on the 
      draft:</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>1) DPNSS has now been published by the UK's 
      Network Interoperability Consultative Committee as NICC Document 
      1301:2001/03. You can find it at <A 
      href="http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm">http://www.oftel.gov.uk/ind_groups/nicc/Public/interconnectstandards/isc.htm</A>&nbsp;along 
      with lots of other gems.</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>You should reference this version, rather than 
      the somewhat harder to obtain BTNR.</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>2) Why do you need a length octet at the front of 
      the message ? Can't you (or shouldn't you) put a Content-Length in the 
      MIME headers for this MIME part ?</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>3) I think given the short length of the messages 
      you could get away with mandating that segmented DPNSS messages should 
      always be reassembled before encapsulating in SIP and that receivers must 
      be able to receive such reassembled messages. This removes some 
      unnecessary complexity.</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>4) Section 3, under table 1, you say 'Floating 
      Point' when you mean 'Decimal'. I think the words 'contiguously 
      increasing' could probably be removed here too. You will have trouble 
      finding the next real number after 6.00!</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003>Regards...Mark</SPAN></FONT></DIV>
      <DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
      class=145285416-11022003></SPAN></FONT>&nbsp;</DIV>
      <BLOCKQUOTE 
      style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; PADDING-LEFT: 5px">
        <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
        size=2>-----Original Message-----<BR><B>From:</B> Ranjith Mukundan 
        [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 11 February 2003 
        08:59<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] QSIG 
        Vs DPNSS MIME<BR><BR></DIV></FONT><!-- Converted from text/plain format -->
        <P><FONT size=2>we'd appreciate some views on this draft from atleast 
        those who fairly understand QSIG.<BR><BR>of course, others who feel that 
        this MIME type collides directly with the purpose of SIP are encouraged 
        to provide alternatives bearing in mind that providing complete 
        end-to-end service transparency over the intervening SIP based core 
        would be essential, if we are to provide a seamless migration path from 
        H.323 based core networks - which already provide equivalent mechanisms 
        for DSS1 based protocols.<BR><BR>- ranjith..<BR><BR><BR>-----Original 
        Message-----<BR>From:&nbsp;&nbsp; Ranjith Mukundan<BR>Sent:&nbsp;&nbsp; 
        Sat 2/8/2003 12:09 AM<BR>To:&nbsp;&nbsp;&nbsp;&nbsp; 
        sipping@ietf.org<BR>Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        RE: [Sipping] I-D 
        ACTION:draft-mukundan-sipping-dpnss-01.txt<BR><BR>Hi,<BR><BR>posting WRT 
        <A 
        href="http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt</A><BR><BR><BR>&gt;&gt;<BR>&gt;&gt; 
        Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; I'd 
        prefer<BR>&gt;&gt; that folks implement a translation to SIP, where SIP 
        can natively carry<BR>&gt;&gt; all the information that DPNSS is trying 
        to convey.<BR>&gt;&gt;<BR><BR>To put things in perspective:<BR><BR>1- 
        DPNSS is a protocol (deployed primarily in UK, NZ, Australia and parts 
        of NA) very similar to QSIG (deployed primarily Europe) for 
        interconnecting PBXs or centrex customers hosted off LEs. DPNSS (like 
        QSIG) finds enormous application towards realizing a local VPN or 
        network-wide VPN services.<BR><BR>2- This draft aims to do exactly what 
        RFC 3204 (MIME objects for QSIG/ISUP) does for QSIG with 
        appropriate/relevant customization for DPNSS - i.e., to provide 
        end-to-end &amp; *complete*, DPNSS Service transparency between pockets 
        of DPNSS based private networks over a SIP based core.<BR><BR>3- 
        Currently, circuit-switched SS7 based cores, provide equivalent DPNSS 
        tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the 
        concomitant translation/interworking mechanisms to ISUP. The 
        translation/interworking mechanism to ISUP is limited to the basic call 
        and certain ISUP supplementary services where there is a direct/easy 
        equivalent.<BR><BR>4- H.323 also provides equivalent tunneling 
        mechanisms (via the H.450 Annexe procedures) for transparently carrying 
        DSS1 based signaling between DSS1-based network elements/gateways, over 
        a H.323 core.<BR><BR>Some specifics:<BR><BR>1- DPNSS supports, as 
        specified in BTNR 188, approximately 240+ supplementary 
        features/services (like Call Back When Free, Call Back When Next Used, 
        Executive Intrusion, Travelling Class of Service, Do Not Disturb, Night 
        Service, Private Network Route Optimization, Centralized Operator, Three 
        Party Takeover, Remote Diversion Registration etc). It would be almost 
        impossible to interwork to equivalent mechanisms in SIP, without losing 
        the service transparency when using an intervening SIP based 
        core.<BR><BR>2- Yes, we definitely need an interworking/translation 
        mechanism to map all the basic call/service paremeters, and procedures 
        for mapping the messages. But this does not preclude the need for a 
        DPNSS MIME type, but would rather be complimentary to the MIME type 
        mechanism. Similar to <A 
        href="http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt</A> 
        , for QSIG, we are in the process of putting together an interworking 
        draft for DPNSS as well - but rather than making it DPNSS specific, we 
        are trying to make it a common draft to cover PRI, DPNSS &amp; DASS2 [as 
        Q.SIG interworking is already a SIPPING WG ID]. Note that, PRI and DASS2 
        will not need a MIME type as these are, as one might be aware, for PSTN 
        access, unlike DPNSS and QSIG (which are private network 
        protocols).<BR><BR>3- From the SIP perspective, carrying non-SIP 
        applications as MIME objects to ensure end to end service transparency 
        is architecturally less appalling than, say for e.g., tunneling DSS1 
        protocols in H.323 based network - so the SIP DPNSS MIME proposition as 
        per this I-D is the obvious way forward for DPNSS service transparency 
        over a SIP based core, especially now that we see acute deployment needs 
        to be met.<BR><BR>cheers,<BR>ranjith..<BR><BR>&gt; -----Original 
        Message-----<BR>&gt; From: sipping-admin@ietf.org [<A 
        href="mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]On 
        Behalf<BR>&gt; Of<BR>&gt; Ranjith Mukundan<BR>&gt; Sent: Tuesday, 
        February 04, 2003 5:32 PM<BR>&gt; To: sipping@ietf.org<BR>&gt; Subject: 
        RE: [Sipping] I-D 
        ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; Hi 
        List,<BR>&gt; We would like you take a look at this draft and revert 
        with any<BR>&gt; comments/queries that you might have.<BR>&gt;<BR>&gt; 
        SIPPING Chairs,<BR>&gt; The authors feel that the requirement and the 
        need for this MIME type<BR>&gt; is<BR>&gt; quite acute. Hence we'd like 
        you to consider this for inclusion as a<BR>&gt; WG I-D<BR>&gt; and 
        suggest the way forward.<BR>&gt;<BR>&gt; - ranjith..<BR>&gt;<BR>&gt; 
        -----Original Message-----<BR>&gt; From: sipping-admin@ietf.org [<A 
        href="mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]On 
        Behalf<BR>&gt; Of<BR>&gt; Internet-Drafts@ietf.org<BR>&gt; Sent: 
        Tuesday, February 04, 2003 5:14 PM<BR>&gt; To: IETF-Announce:<BR>&gt; 
        Cc: sipping@ietf.org<BR>&gt; Subject: [Sipping] I-D 
        ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; A New 
        Internet-Draft is available from the on-line Internet-Drafts<BR>&gt; 
        directories.<BR>&gt;<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME 
        media types for DPNSS Objects<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et 
        al.<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 
        draft-mukundan-sipping-dpnss-01.txt<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 9<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-3<BR>&gt;<BR>&gt; 
        This document describes Multipart Internet Mail Extensions (MIME) 
        type<BR>&gt; for application/Digital Private Network Signaling System 
        (DPNSS)<BR>&gt; objects for use in Session Initiation Protocol (SIP) 
        applications,<BR>&gt; according to the rules defined in RFC 2048. These 
        types can be used to<BR>&gt; identify Digital Private Network Signaling 
        System 1 (DPNSS 1) objects<BR>&gt; within a SIP message such as INVITE 
        or INFO, as might be implemented<BR>&gt; when using SIP in an 
        environment where part of the call involves<BR>&gt; interworking to 
        DPNSS 1 based Private Networks. RFC 3204 addresses a<BR>&gt; similar 
        need for Integrated Service Digital Network User Part (ISUP)<BR>&gt; and 
        Q Signaling (QSIG).<BR>&gt;<BR>&gt; A URL for this Internet-Draft 
        is:<BR>&gt; <A 
        href="http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx</A><BR>&gt; 
        t<BR>&gt;<BR>&gt; To remove yourself from the IETF Announcement list, 
        send a message to<BR>&gt; ietf-announce-request with the word 
        unsubscribe in the body of the<BR>&gt; message.<BR>&gt;<BR>&gt; 
        Internet-Drafts are also available by anonymous FTP. Login with 
        the<BR>&gt; username<BR>&gt; "anonymous" and a password of your e-mail 
        address. After logging in,<BR>&gt; type "cd internet-drafts" and 
        then<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "get 
        draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; A list of 
        Internet-Drafts directories can be found in<BR>&gt; <A 
        href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</A> 
        or<BR>&gt; <A 
        href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A><BR>&gt;<BR>&gt;<BR>&gt; 
        Internet-Drafts can also be obtained by e-mail.<BR>&gt;<BR>&gt; Send a 
        message to:<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        mailserv@ietf.org.<BR>&gt; In the body type:<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "FILE 
        /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; 
        NOTE: The mail server at ietf.org can return the document in<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the "mpack" 
        utility.&nbsp; To use this<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
        feature, insert the command "ENCODING mime" before the "FILE"<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the response(s), 
        you will need "munpack" or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a 
        MIME-compliant mail reader.&nbsp; Different MIME-compliant mail 
        readers<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different 
        behavior, especially when dealing with<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "multipart" MIME messages (i.e. documents 
        which have been split<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into 
        multiple messages), so check your local documentation on<BR>&gt; 
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these 
        messages.<BR>&gt;<BR>&gt;<BR>&gt; Below is the data which will enable a 
        MIME compliant mail reader<BR>&gt; implementation to automatically 
        retrieve the ASCII version of the<BR>&gt; 
        Internet-Draft.<BR>&gt;<BR>&gt; 
        _______________________________________________<BR>&gt; Sipping mailing 
        list&nbsp; <A 
        href="https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf.org/mailman/listinfo/sipping</A><BR>&gt; 
        This list is for NEW development of the application of SIP Use<BR>&gt; 
        sip-implementors@cs.columbia.edu for questions on current sip 
        Use<BR>&gt; sip@ietf.org for new developments of core 
        SIP<BR><BR><BR><BR><BR></FONT></P>_______________________________________________ 
        Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This 
        list is for NEW development of the application of SIP Use 
        sip-implementors@cs.columbia.edu for questions on current sip Use 
        sip@ietf.org for new developments of core SIP 
  </BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: Marjou Xavier <xavier.marjou@siemens.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] updated conferencing framework
Date: Wed, 12 Feb 2003 12:05:15 +0100
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Hi,

It seems that a sentence is not finished at the end of Page 30:
If the focus in the application server receives a conference policy control command from ...

Xavier Marjou

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, February 12, 2003 6:53 AM
To: sipping@ietf.org
Subject: [Sipping] updated conferencing framework


Folks,

I've submitted an update to the conferencing framework document. THis is 
the result of a lot of work of the conferencing design team to make 
things consistent across documents, and to resolve hard problems like 
how sidebars are handled. Until it appears in the archives you can pick 
up a copy at:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-conferencing-framework-01.txt

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: SIP Generic Request History Capability û Requirements
	Author(s)	: M. Barnes, M. Watson, C. Jennings, J. Peterson
	Filename	: draft-ietf-sipping-req-history-02.txt
	Pages		: 11
	Date		: 2003-2-11
	
Many services that SIP is anticipated to support require the ability 
to determine why and how the call arrived at a specific application.  
Examples of such services include (but are not limited to) sessions 
initiated to call centers via 'click to talk' SIP URLs on a web page, 
'call history/logging' style services within intelligent 'call 
management' software for SIP UAs and calls to voicemail servers and 
call centers.  While SIP implicitly provides the redirect/retarget 
capabilities that enable calls to be routed to chosen applications, 
there is currently no standard mechanism within SIP for communicating 
the history of such a request. This 'request history' information 
allows the receiving application to determine hints about how and why 
the call arrived at the application/user.     
This draft discusses the motivations in support of a mechanism for 
recording the 'request history', and proposes detailed requirements 
for such a generic 'request history' capability.

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	 <kai.miao@intel.com>
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Subject: RE: [Sipping] Question about call flow in draft-ietf-sipping-serv
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Hi Ajit,
 
I had noticed in RFC 3264 (right after I sent my email):
 
   "If the stream to be placed on hold was previously a sendrecv media
   stream, it is placed on hold by marking it as sendonly.  If the
   stream to be placed on hold was previously a recvonly media stream,
   it is placed on hold by marking it inactive.

   .........

   Typically, when a user "presses" hold, the agent will generate an
   offer with all streams in the SDP indicating a direction of sendonly,
   and it will also locally mute, so that no media is sent to the far
   end, and no media is played out."
 

This should explains why the example was that way.........basically it
doesn't matter what the answer is because there is no stream sent from User
B...... 
 
But, my question is why not make things more consistent by using
"a=recvonly" (or "a=inactive") instead of the default "a=sendrecv" in the
answer? What happen if the case is not stream on hold, for example, if it is
changing a stream from "sendrecv" to "sendonly"? Wouldn't this type of
answer cause confusion and violate the "rules" stated in RFC3264?
 
Thanks,
Kai   

-----Original Message-----
From: Ajit Kalele [mailto:kalele@mahindrabt.com]
Sent: Wednesday, February 12, 2003 12:27 AM
To: Miao, Kai
Cc: sipping@ietf.org
Subject: Re: [Sipping] Question about call flow in
draft-ietf-sipping-service-examples-03


Hi, 

I think there is an error in the call flows. 


Either A has to respond with an "a=inactive", so that there is no RTP flow
between the UAs, or else there should be unidirectional RTP flow between the
2 UAs. 


Best Regards 
- Ajit 


"Miao, Kai" wrote: 


 Could someone provide an answer to the following question:In the first
example, Call Hold, given in draft-ietf-sipping-service-examples-03.txt, why
F10, i.e., INVITE(hold), has a=sendonly in its SDP message and F12, i.e.,
200 OK, has no "a=" attribute at all while "No RTP Sent!" is marked in the
call flow diagram? Shoudn't F12 contain a=recvonly or a=inactive, following
the Offer/Answer model? Why there is "no RTP sent" in the diagram when there
is "a=sendonly" in F10?Thanks,Kai MiaoIntel 


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<BODY>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=272141114-12022003>Hi 
Ajit,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=272141114-12022003>I had 
noticed in RFC 3264 (right after I sent&nbsp;my email):</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003>&nbsp;&nbsp; "If the stream to be placed on hold was 
previously a sendrecv media<BR>&nbsp;&nbsp; stream, it is placed on hold by 
marking it as sendonly.&nbsp; If the<BR>&nbsp;&nbsp; stream to be placed on hold 
was previously a recvonly media stream,<BR>&nbsp;&nbsp; it is placed on hold by 
marking it inactive.<BR><BR>&nbsp;&nbsp; .........<BR><BR>&nbsp;&nbsp; 
Typically, when a user "presses" hold, the agent will generate 
an<BR>&nbsp;&nbsp; offer with all streams in the SDP indicating a direction of 
sendonly,<BR>&nbsp;&nbsp; and it will also locally mute, so that no media is 
sent to the far<BR>&nbsp;&nbsp; end, and no media is played 
out."</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003><BR>This should explains why the example was that 
way.........basically it doesn't matter&nbsp;what the answer is because 
there&nbsp;is no stream sent from User B...... </SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=272141114-12022003>But, 
my question is why not make&nbsp;things more consistent by using "a=recvonly" 
(or "a=inactive") instead of the default "a=sendrecv" in the answer? 
What&nbsp;happen if the case is not stream on hold,&nbsp;for example,&nbsp;if it 
is changing a stream from "sendrecv" to "sendonly"? Wouldn't this type of answer 
cause confusion and violate the "rules" stated 
in&nbsp;RFC3264?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003>Thanks,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=272141114-12022003>Kai&nbsp;&nbsp; </DIV></SPAN></FONT>
<BLOCKQUOTE>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ajit Kalele 
  [mailto:kalele@mahindrabt.com]<BR><B>Sent:</B> Wednesday, February 12, 2003 
  12:27 AM<BR><B>To:</B> Miao, Kai<BR><B>Cc:</B> 
  sipping@ietf.org<BR><B>Subject:</B> Re: [Sipping] Question about call flow in 
  draft-ietf-sipping-service-examples-03<BR><BR></DIV></FONT>Hi, 
  <P>I think there is an error in the call flows. 
  <P>Either A has to respond with an "a=inactive", so that there is no RTP flow 
  between the UAs, or else there should be unidirectional RTP flow between the 2 
  UAs. 
  <P>Best Regards <BR>- Ajit 
  <P>"Miao, Kai" wrote: 
  <BLOCKQUOTE TYPE="CITE">&nbsp;<SPAN class=481480723-11022003><FONT 
    face=Arial><FONT color=#0000ff><FONT size=-1>Could someone provide an answer 
    to the following question:</FONT></FONT></FONT></SPAN><SPAN 
    class=481480723-11022003></SPAN><SPAN class=481480723-11022003><FONT 
    face=Arial><FONT color=#0000ff><FONT size=-1>In the first example, Call 
    Hold, given in&nbsp;</SPAN>draft-ietf-sipping-service-examples-03.tx<SPAN 
    class=481480723-11022003>t, why F10, i.e., INVITE(hold), has a=sendonly in 
    its SDP message and F12, i.e., 200 OK, has no "a=" attribute at all while 
    "No RTP Sent!" is marked in the call flow 
    diagram?&nbsp;</FONT></FONT></FONT></SPAN><SPAN 
    class=481480723-11022003></SPAN><SPAN class=481480723-11022003><FONT 
    face=Arial><FONT color=#0000ff><FONT size=-1>Shoudn't F12 contain a=recvonly 
    or a=inactive, following the Offer/Answer model? Why there is "no RTP sent" 
    in the diagram when there is "a=sendonly" in 
    F10?</FONT></FONT></FONT></SPAN><SPAN class=481480723-11022003></SPAN><SPAN 
    class=481480723-11022003><FONT face=Arial><FONT color=#0000ff><FONT 
    size=-1>Thanks,</FONT></FONT></FONT></SPAN><SPAN 
    class=481480723-11022003><FONT face=Arial><FONT color=#0000ff><FONT 
    size=-1>Kai Miao</FONT></FONT></FONT></SPAN><SPAN 
    class=481480723-11022003><FONT face=Arial><FONT color=#0000ff><FONT 
    size=-1>Intel&nbsp;</FONT></FONT></FONT></SPAN></BLOCKQUOTE><BR>*********************************************************<BR>Disclaimer<BR><BR>This 
  message (including any attachments) contains <BR>confidential information 
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  and are hereby notified that <BR>any disclosure, copying, or distribution of 
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  us at http://www.mahindrabt.com<BR><BR></BLOCKQUOTE></BODY></HTML>

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Date: Wed, 12 Feb 2003 09:18:26 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: RE: [Sipping] Question about call flow in draft-ietf-sipping-serv
 ice-examples-03
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To: "Miao, Kai" <kai.miao@intel.com>, "'Ajit Kalele'" <kalele@mahindrabt.com>,
        "Miao, Kai" <kai.miao@intel.com>
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Hi,

Thanks for the comments on the draft.

You are correct - the answer to the hold offer (a=sendonly) should be 
a=recvonly instead of what is currently shown as sendrecv in the draft.

This will be fixed in the next version of the document.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

At 07:38 AM 2/12/2003 -0700, Miao, Kai wrote:
>Hi Ajit,
>
>I had noticed in RFC 3264 (right after I sent my email):
>
>    "If the stream to be placed on hold was previously a sendrecv media
>    stream, it is placed on hold by marking it as sendonly.  If the
>    stream to be placed on hold was previously a recvonly media stream,
>    it is placed on hold by marking it inactive.
>
>    .........
>
>    Typically, when a user "presses" hold, the agent will generate an
>    offer with all streams in the SDP indicating a direction of sendonly,
>    and it will also locally mute, so that no media is sent to the far
>    end, and no media is played out."
>
>
>This should explains why the example was that way.........basically it 
>doesn't matter what the answer is because there is no stream sent from 
>User B......
>
>But, my question is why not make things more consistent by using 
>"a=recvonly" (or "a=inactive") instead of the default "a=sendrecv" in the 
>answer? What happen if the case is not stream on hold, for example, if it 
>is changing a stream from "sendrecv" to "sendonly"? Wouldn't this type of 
>answer cause confusion and violate the "rules" stated in RFC3264?
>
>Thanks,
>Kai
>-----Original Message-----
>From: Ajit Kalele [mailto:kalele@mahindrabt.com]
>Sent: Wednesday, February 12, 2003 12:27 AM
>To: Miao, Kai
>Cc: sipping@ietf.org
>Subject: Re: [Sipping] Question about call flow in 
>draft-ietf-sipping-service-examples-03
>
>Hi,
>
>I think there is an error in the call flows.
>
>Either A has to respond with an "a=inactive", so that there is no RTP flow 
>between the UAs, or else there should be unidirectional RTP flow between 
>the 2 UAs.
>
>Best Regards
>- Ajit
>
>"Miao, Kai" wrote:
>>  Could someone provide an answer to the following question:In the first 
>> example, Call Hold, given in draft-ietf-sipping-service-examples-03.txt, 
>> why F10, i.e., INVITE(hold), has a=sendonly in its SDP message and F12, 
>> i.e., 200 OK, has no "a=" attribute at all while "No RTP Sent!" is 
>> marked in the call flow diagram? Shoudn't F12 contain a=recvonly or 
>> a=inactive, following the Offer/Answer model? Why there is "no RTP sent" 
>> in the diagram when there is "a=sendonly" in F10?Thanks,Kai MiaoIntel
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Date: Wed, 12 Feb 2003 10:07:05 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sipping] New I-D on URI leasing
In-reply-to: <3E49E4B5.8010608@dynamicsoft.com>
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Hi Jonathan,

This is a useful discussion of the problem of constructing globally 
routable Contact URIs.

However, if you re-read 
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-00.txt 
you will see that your proposed solution is almost identical to the "device 
AOR" proposed in Section 7.5, with the exception of the LEASE method, 
which, as you mention can (and perhaps should) be done outside of SIP.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

At 01:07 AM 2/12/2003 -0500, Jonathan Rosenberg wrote:
>Folks,
>
>I've just submitted an I-D on the concept of "URI leasing". THis is my 
>proposed technique for solving the "globally routable UA URI" problem. 
>SPecifically, for cases like REFER, how do we construct a URI that routes 
>to a specific UA instance in a domain? At the last ietf meeting, this 
>issue came up in several contexts. I suggested a single document to 
>explore the problem. The chairs asked for volunteers, and I said I'd write 
>one, time permitting, if no one else stepped up. Since no one did, well, 
>here it is.
>
>Until it appears in the archives, you can pick it up at:
>http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.txt
>http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.html
>
>Thanks,
>Jonathan R.
>--
>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

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To: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: FW: [Sipping] I-D ACTION:draft-ietf-sipping-req-history-02.txt
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Hi all,

There's really only one minor change to this document:
- The last paragraph in section 5 was re-worded based on Mark suggestion per
the email discussion on the capturing of request history in outgoing
messages:
http://www.ietf.org/mail-archive/working-groups/sipping/current/msg03695.htm
l

John E. had suggested additional wording, but I think the one sentence
captures the essence of the requirement and the additional wording is more
appropriately addressed in the solution draft (which I think it now is per
the updated solution draft - per my next email). 

There was also some minor changes to correct "funny characters", but I can
see that there is still one in the title that I'll fix the next round.  But,
other than that, this document should be complete.

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com
972-684-5432

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Wednesday, February 12, 2003 8:22 AM
Cc: sipping@ietf.org
Subject: [Sipping] I-D ACTION:draft-ietf-sipping-req-history-02.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Proposal Investigation
Working Group of the IETF.

	Title		: SIP Generic Request History Capability û
Requirements
	Author(s)	: M. Barnes, M. Watson, C. Jennings, J. Peterson
	Filename	: draft-ietf-sipping-req-history-02.txt
	Pages		: 11
	Date		: 2003-2-11
	
Many services that SIP is anticipated to support require the ability 
to determine why and how the call arrived at a specific application.  
Examples of such services include (but are not limited to) sessions 
initiated to call centers via 'click to talk' SIP URLs on a web page, 
'call history/logging' style services within intelligent 'call 
management' software for SIP UAs and calls to voicemail servers and 
call centers.  While SIP implicitly provides the redirect/retarget 
capabilities that enable calls to be routed to chosen applications, 
there is currently no standard mechanism within SIP for communicating 
the history of such a request. This 'request history' information 
allows the receiving application to determine hints about how and why 
the call arrived at the application/user.     
This draft discusses the motivations in support of a mechanism for 
recording the 'request history', and proposes detailed requirements 
for such a generic 'request history' capability.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-02.txt

To remove yourself from the IETF Announcement list, send a message to 
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type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
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	"multipart" MIME messages (i.e. documents which have been split
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Internet-Draft.


-----
To: 
Subject: 
Date: Wed, 12 Feb 2003 09:25:18 -0600
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begin 600 ATT12027
M0V]N=&5N="UT>7!E.B!M97-S86=E+V5X=&5R;F%L+6)O9'D[#0H)86-C97-S
M+71Y<&4](FUA:6PM<V5R=F5R(CL-"@ES97)V97(](FUA:6QS97)V0&EE=&8N
M;W)G(@T*#0I#;VYT96YT+51Y<&4Z('1E>'0O<&QA:6X-"D-O;G1E;G0M240Z
M"3PR,#`S+3(M,3$Q-#$S-#DN22U$0&EE=&8N;W)G/@T*#0I%3D-/1$E.1R!M
M:6UE#0I&24Q%("]I;G1E<FYE="UD<F%F=',O9')A9G0M:65T9BUS:7!P:6YG
5+7)E<2UH:7-T;W)Y+3`R+G1X=`T*
`
end

begin 600 draft-ietf-sipping-req-history-02.url
M6TEN=&5R;F5T4VAO<G1C=71=#0I54DP]9G1P.B\O9G1P+FEE=&8N;W)G+VEN
M=&5R;F5T+61R869T<R]D<F%F="UI971F+7-I<'!I;F<M<F5Q+6AI<W1O<GDM
$,#(-"@==
`
end
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Ranjith,
 
Seems like a reasonable thing to do. A few comments which I hope are
helpful...
 
1) The discussion of handling segmented information might be clearer if you
were to use the terminology of the DPNSS specification such as Selection
Block/Selection Field and Information Block/Information Field to refer to
the items being manipulated. In the case of the ISRM/SSRM, as others have
already commented, you would need to distinguish between the Supplementary
Information and the Destination Address which make up the Selection Block;
the Supplementary Information can be safely buffered but the Destination
Address needs to be handled according to overlap procedures.
 
2) Using that terminology might make clearer how you propose to perform the
buffering. I think you are proposing to concatenate the Information Fields
from a sequence of messages to assemble a complete Information Block (or
segment thereof if it exceeds your 255 octet limit) with a single message
type octet, although I don't see this stated explicitly. If so, then I think
your example is slightly inaccurate: combining a sequence of five EEMs each
of 45 octets would produce an EEM of 221 octets (as four message type octets
are removed)?
 
3) I'm not sure there is really any need for a version number, since DPNSS
has comprehensive compatibility mechanisms?
 
Regards,
Mark.

-----Original Message-----
From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
Sent: 11 February 2003 08:59
To: sipping@ietf.org
Subject: [Sipping] QSIG Vs DPNSS MIME



we'd appreciate some views on this draft from atleast those who fairly
understand QSIG.

of course, others who feel that this MIME type collides directly with the
purpose of SIP are encouraged to provide alternatives bearing in mind that
providing complete end-to-end service transparency over the intervening SIP
based core would be essential, if we are to provide a seamless migration
path from H.323 based core networks - which already provide equivalent
mechanisms for DSS1 based protocols.

- ranjith..


-----Original Message-----
From:   Ranjith Mukundan
Sent:   Sat 2/8/2003 12:09 AM
To:     sipping@ietf.org
Cc:    
Subject:        RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt

Hi,

posting WRT
http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt> 


>>
>> Why is it so important to be able to tunnel DPNSS in SIP?  I'd prefer
>> that folks implement a translation to SIP, where SIP can natively carry
>> all the information that DPNSS is trying to convey.
>>

To put things in perspective:

1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and parts of
NA) very similar to QSIG (deployed primarily Europe) for interconnecting
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous
application towards realizing a local VPN or network-wide VPN services.

2- This draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP)
does for QSIG with appropriate/relevant customization for DPNSS - i.e., to
provide end-to-end & *complete*, DPNSS Service transparency between pockets
of DPNSS based private networks over a SIP based core.

3- Currently, circuit-switched SS7 based cores, provide equivalent DPNSS
tunneling mechanisms through (ETSI & ANSI) ISUP *apart from* the concomitant
translation/interworking mechanisms to ISUP. The translation/interworking
mechanism to ISUP is limited to the basic call and certain ISUP
supplementary services where there is a direct/easy equivalent.

4- H.323 also provides equivalent tunneling mechanisms (via the H.450 Annexe
procedures) for transparently carrying DSS1 based signaling between
DSS1-based network elements/gateways, over a H.323 core.

Some specifics:

1- DPNSS supports, as specified in BTNR 188, approximately 240+
supplementary features/services (like Call Back When Free, Call Back When
Next Used, Executive Intrusion, Travelling Class of Service, Do Not Disturb,
Night Service, Private Network Route Optimization, Centralized Operator,
Three Party Takeover, Remote Diversion Registration etc). It would be almost
impossible to interwork to equivalent mechanisms in SIP, without losing the
service transparency when using an intervening SIP based core.

2- Yes, we definitely need an interworking/translation mechanism to map all
the basic call/service paremeters, and procedures for mapping the messages.
But this does not preclude the need for a DPNSS MIME type, but would rather
be complimentary to the MIME type mechanism. Similar to
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
<http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt>  ,
for QSIG, we are in the process of putting together an interworking draft
for DPNSS as well - but rather than making it DPNSS specific, we are trying
to make it a common draft to cover PRI, DPNSS & DASS2 [as Q.SIG interworking
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME
type as these are, as one might be aware, for PSTN access, unlike DPNSS and
QSIG (which are private network protocols).

3- From the SIP perspective, carrying non-SIP applications as MIME objects
to ensure end to end service transparency is architecturally less appalling
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so the
SIP DPNSS MIME proposition as per this I-D is the obvious way forward for
DPNSS service transparency over a SIP based core, especially now that we see
acute deployment needs to be met.

cheers,
ranjith..

> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Ranjith Mukundan
> Sent: Tuesday, February 04, 2003 5:32 PM
> To: sipping@ietf.org
> Subject: RE: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> Hi List,
> We would like you take a look at this draft and revert with any
> comments/queries that you might have.
>
> SIPPING Chairs,
> The authors feel that the requirement and the need for this MIME type
> is
> quite acute. Hence we'd like you to consider this for inclusion as a
> WG I-D
> and suggest the way forward.
>
> - ranjith..
>
> -----Original Message-----
> From: sipping-admin@ietf.org [ mailto:sipping-admin@ietf.org
<mailto:sipping-admin@ietf.org> ]On Behalf
> Of
> Internet-Drafts@ietf.org
> Sent: Tuesday, February 04, 2003 5:14 PM
> To: IETF-Announce:
> Cc: sipping@ietf.org
> Subject: [Sipping] I-D ACTION:draft-mukundan-sipping-dpnss-01.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>       Title           : MIME media types for DPNSS Objects
>       Author(s)       : R. Mukundan et al.
>       Filename        : draft-mukundan-sipping-dpnss-01.txt
>       Pages           : 9
>       Date            : 2003-2-3
>
> This document describes Multipart Internet Mail Extensions (MIME) type
> for application/Digital Private Network Signaling System (DPNSS)
> objects for use in Session Initiation Protocol (SIP) applications,
> according to the rules defined in RFC 2048. These types can be used to
> identify Digital Private Network Signaling System 1 (DPNSS 1) objects
> within a SIP message such as INVITE or INFO, as might be implemented
> when using SIP in an environment where part of the call involves
> interworking to DPNSS 1 based Private Networks. RFC 3204 addresses a
> similar need for Integrated Service Digital Network User Part (ISUP)
> and Q Signaling (QSIG).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx
<http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx> 
> t
>
> To remove yourself from the IETF Announcement list, send a message to
> ietf-announce-request with the word unsubscribe in the body of the
> message.
>
> Internet-Drafts are also available by anonymous FTP. Login with the
> username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
>       "get draft-mukundan-sipping-dpnss-01.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html <http://www.ietf.org/shadow.html>  or
> ftp://ftp.ietf.org/ietf/1shadow-sites.txt
<ftp://ftp.ietf.org/ietf/1shadow-sites.txt> 
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
>       mailserv@ietf.org.
> In the body type:
>       "FILE /internet-drafts/draft-mukundan-sipping-dpnss-01.txt".
>
> NOTE: The mail server at ietf.org can return the document in
>       MIME-encoded form by using the "mpack" utility.  To use this
>       feature, insert the command "ENCODING mime" before the "FILE"
>       command.  To decode the response(s), you will need "munpack" or
>       a MIME-compliant mail reader.  Different MIME-compliant mail readers
>       exhibit different behavior, especially when dealing with
>       "multipart" MIME messages (i.e. documents which have been split
>       up into multiple messages), so check your local documentation on
>       how to manipulate these messages.
>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
<https://www1.ietf.org/mailman/listinfo/sipping> 
> This list is for NEW development of the application of SIP Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sip@ietf.org for new developments of core SIP






_______________________________________________ Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
development of the application of SIP Use sip-implementors@cs.columbia.edu
for questions on current sip Use sip@ietf.org for new developments of core
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<TITLE>QSIG Vs DPNSS MIME</TITLE>

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<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=766265214-12022003>Ranjith,</SPAN></FONT></DIV>
<DIV><SPAN class=766265214-12022003></SPAN>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=766265214-12022003>Seems 
like a reasonable thing to do. A few comments which I hope are 
helpful...</SPAN></FONT></DIV>
<DIV><SPAN class=766265214-12022003></SPAN>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=766265214-12022003>1) The 
discussion of handling segmented information might be clearer if you were to use 
the terminology of the DPNSS specification such as Selection Block/Selection 
Field and Information Block/Information Field to refer to the items being 
manipulated. In the case of the ISRM/SSRM, as others have already commented, you 
would need to distinguish between the Supplementary Information and the 
Destination Address which make up the Selection Block; the Supplementary 
Information can be safely buffered but the Destination Address needs to be 
handled according to overlap procedures.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=766265214-12022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=766265214-12022003>2) 
Using that terminology might make clearer how you propose to perform the 
buffering. I think you are proposing to concatenate the Information Fields from 
a sequence of messages to assemble a complete Information Block (or segment 
thereof if it exceeds your 255 octet limit) with a single message type octet, 
although I don't see this stated explicitly. If so, then I think your example is 
slightly inaccurate: combining a sequence of five EEMs each of 45 octets would 
produce an EEM of 221 octets (as four message type octets are 
removed)?</SPAN></FONT></DIV>
<DIV><SPAN class=766265214-12022003></SPAN>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=766265214-12022003>3) I'm 
not sure there is really any need for a version number, since DPNSS&nbsp;has 
comprehensive compatibility mechanisms?</SPAN></FONT></DIV>
<DIV><SPAN class=766265214-12022003></SPAN>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=766265214-12022003>Regards,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=766265214-12022003>Mark.</SPAN></FONT></DIV>
<BLOCKQUOTE>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Ranjith Mukundan 
  [mailto:ranjith.mukundan@wipro.com]<BR><B>Sent:</B> 11 February 2003 
  08:59<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] QSIG Vs 
  DPNSS MIME<BR><BR></DIV></FONT><!-- Converted from text/plain format -->
  <P><FONT size=2>we'd appreciate some views on this draft from atleast those 
  who fairly understand QSIG.<BR><BR>of course, others who feel that this MIME 
  type collides directly with the purpose of SIP are encouraged to provide 
  alternatives bearing in mind that providing complete end-to-end service 
  transparency over the intervening SIP based core would be essential, if we are 
  to provide a seamless migration path from H.323 based core networks - which 
  already provide equivalent mechanisms for DSS1 based protocols.<BR><BR>- 
  ranjith..<BR><BR><BR>-----Original Message-----<BR>From:&nbsp;&nbsp; Ranjith 
  Mukundan<BR>Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09 
  AM<BR>To:&nbsp;&nbsp;&nbsp;&nbsp; 
  sipping@ietf.org<BR>Cc:&nbsp;&nbsp;&nbsp;&nbsp;<BR>Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  RE: [Sipping] I-D 
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR><BR>Hi,<BR><BR>posting WRT <A 
  href="http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.txt</A><BR><BR><BR>&gt;&gt;<BR>&gt;&gt; 
  Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; I'd 
  prefer<BR>&gt;&gt; that folks implement a translation to SIP, where SIP can 
  natively carry<BR>&gt;&gt; all the information that DPNSS is trying to 
  convey.<BR>&gt;&gt;<BR><BR>To put things in perspective:<BR><BR>1- DPNSS is a 
  protocol (deployed primarily in UK, NZ, Australia and parts of NA) very 
  similar to QSIG (deployed primarily Europe) for interconnecting PBXs or 
  centrex customers hosted off LEs. DPNSS (like QSIG) finds enormous application 
  towards realizing a local VPN or network-wide VPN services.<BR><BR>2- This 
  draft aims to do exactly what RFC 3204 (MIME objects for QSIG/ISUP) does for 
  QSIG with appropriate/relevant customization for DPNSS - i.e., to provide 
  end-to-end &amp; *complete*, DPNSS Service transparency between pockets of 
  DPNSS based private networks over a SIP based core.<BR><BR>3- Currently, 
  circuit-switched SS7 based cores, provide equivalent DPNSS tunneling 
  mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the concomitant 
  translation/interworking mechanisms to ISUP. The translation/interworking 
  mechanism to ISUP is limited to the basic call and certain ISUP supplementary 
  services where there is a direct/easy equivalent.<BR><BR>4- H.323 also 
  provides equivalent tunneling mechanisms (via the H.450 Annexe procedures) for 
  transparently carrying DSS1 based signaling between DSS1-based network 
  elements/gateways, over a H.323 core.<BR><BR>Some specifics:<BR><BR>1- DPNSS 
  supports, as specified in BTNR 188, approximately 240+ supplementary 
  features/services (like Call Back When Free, Call Back When Next Used, 
  Executive Intrusion, Travelling Class of Service, Do Not Disturb, Night 
  Service, Private Network Route Optimization, Centralized Operator, Three Party 
  Takeover, Remote Diversion Registration etc). It would be almost impossible to 
  interwork to equivalent mechanisms in SIP, without losing the service 
  transparency when using an intervening SIP based core.<BR><BR>2- Yes, we 
  definitely need an interworking/translation mechanism to map all the basic 
  call/service paremeters, and procedures for mapping the messages. But this 
  does not preclude the need for a DPNSS MIME type, but would rather be 
  complimentary to the MIME type mechanism. Similar to <A 
  href="http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt</A> 
  , for QSIG, we are in the process of putting together an interworking draft 
  for DPNSS as well - but rather than making it DPNSS specific, we are trying to 
  make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG interworking 
  is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a MIME 
  type as these are, as one might be aware, for PSTN access, unlike DPNSS and 
  QSIG (which are private network protocols).<BR><BR>3- From the SIP 
  perspective, carrying non-SIP applications as MIME objects to ensure end to 
  end service transparency is architecturally less appalling than, say for e.g., 
  tunneling DSS1 protocols in H.323 based network - so the SIP DPNSS MIME 
  proposition as per this I-D is the obvious way forward for DPNSS service 
  transparency over a SIP based core, especially now that we see acute 
  deployment needs to be met.<BR><BR>cheers,<BR>ranjith..<BR><BR>&gt; 
  -----Original Message-----<BR>&gt; From: sipping-admin@ietf.org [<A 
  href="mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]On 
  Behalf<BR>&gt; Of<BR>&gt; Ranjith Mukundan<BR>&gt; Sent: Tuesday, February 04, 
  2003 5:32 PM<BR>&gt; To: sipping@ietf.org<BR>&gt; Subject: RE: [Sipping] I-D 
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; Hi 
  List,<BR>&gt; We would like you take a look at this draft and revert with 
  any<BR>&gt; comments/queries that you might have.<BR>&gt;<BR>&gt; SIPPING 
  Chairs,<BR>&gt; The authors feel that the requirement and the need for this 
  MIME type<BR>&gt; is<BR>&gt; quite acute. Hence we'd like you to consider this 
  for inclusion as a<BR>&gt; WG I-D<BR>&gt; and suggest the way 
  forward.<BR>&gt;<BR>&gt; - ranjith..<BR>&gt;<BR>&gt; -----Original 
  Message-----<BR>&gt; From: sipping-admin@ietf.org [<A 
  href="mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]On 
  Behalf<BR>&gt; Of<BR>&gt; Internet-Drafts@ietf.org<BR>&gt; Sent: Tuesday, 
  February 04, 2003 5:14 PM<BR>&gt; To: IETF-Announce:<BR>&gt; Cc: 
  sipping@ietf.org<BR>&gt; Subject: [Sipping] I-D 
  ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>&gt;<BR>&gt;<BR>&gt; A New 
  Internet-Draft is available from the on-line Internet-Drafts<BR>&gt; 
  directories.<BR>&gt;<BR>&gt;<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME media 
  types for DPNSS Objects<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 
  draft-mukundan-sipping-dpnss-01.txt<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 9<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-3<BR>&gt;<BR>&gt; This 
  document describes Multipart Internet Mail Extensions (MIME) type<BR>&gt; for 
  application/Digital Private Network Signaling System (DPNSS)<BR>&gt; objects 
  for use in Session Initiation Protocol (SIP) applications,<BR>&gt; according 
  to the rules defined in RFC 2048. These types can be used to<BR>&gt; identify 
  Digital Private Network Signaling System 1 (DPNSS 1) objects<BR>&gt; within a 
  SIP message such as INVITE or INFO, as might be implemented<BR>&gt; when using 
  SIP in an environment where part of the call involves<BR>&gt; interworking to 
  DPNSS 1 based Private Networks. RFC 3204 addresses a<BR>&gt; similar need for 
  Integrated Service Digital Network User Part (ISUP)<BR>&gt; and Q Signaling 
  (QSIG).<BR>&gt;<BR>&gt; A URL for this Internet-Draft is:<BR>&gt; <A 
  href="http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-01.tx</A><BR>&gt; 
  t<BR>&gt;<BR>&gt; To remove yourself from the IETF Announcement list, send a 
  message to<BR>&gt; ietf-announce-request with the word unsubscribe in the body 
  of the<BR>&gt; message.<BR>&gt;<BR>&gt; Internet-Drafts are also available by 
  anonymous FTP. Login with the<BR>&gt; username<BR>&gt; "anonymous" and a 
  password of your e-mail address. After logging in,<BR>&gt; type "cd 
  internet-drafts" and then<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "get 
  draft-mukundan-sipping-dpnss-01.txt".<BR>&gt;<BR>&gt; A list of 
  Internet-Drafts directories can be found in<BR>&gt; <A 
  href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</A> 
  or<BR>&gt; <A 
  href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A><BR>&gt;<BR>&gt;<BR>&gt; 
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  The mail server at ietf.org can return the document in<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the "mpack" 
  utility.&nbsp; To use this<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, 
  insert the command "ENCODING mime" before the "FILE"<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the response(s), you 
  will need "munpack" or<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant 
  mail reader.&nbsp; Different MIME-compliant mail readers<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, especially when 
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  (i.e. documents which have been split<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  up into multiple messages), so check your local documentation on<BR>&gt; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these 
  messages.<BR>&gt;<BR>&gt;<BR>&gt; Below is the data which will enable a MIME 
  compliant mail reader<BR>&gt; implementation to automatically retrieve the 
  ASCII version of the<BR>&gt; Internet-Draft.<BR>&gt;<BR>&gt; 
  _______________________________________________<BR>&gt; Sipping mailing 
  list&nbsp; <A 
  href="https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf.org/mailman/listinfo/sipping</A><BR>&gt; 
  This list is for NEW development of the application of SIP Use<BR>&gt; 
  sip-implementors@cs.columbia.edu for questions on current sip Use<BR>&gt; 
  sip@ietf.org for new developments of core 
  SIP<BR><BR><BR><BR><BR></FONT></P>_______________________________________________ 
  Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list 
  is for NEW development of the application of SIP Use 
  sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org 
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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Wed, 12 Feb 2003 11:26:46 -0600
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Subject: [Sipping] FW: I-D ACTION:draft-barnes-sipping-history-info-02.txt
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Hi all,

This document has had several changes as a result of the feedback on the
mailing list:

- Section 2.1 - corrected the terminology to be consistent with the
capturing of the single targeted-to-URI and corrected the syntax of the
index as highlighted by Paul K. 

- In section 2.3, clarified that History-Info can be included in provisional
responses per John E.'s comment

- Additional normative text around the maintenance of the Index added to
section 2.3.1 and 2.3.3 per Paul K's comment.
 
- Added additional sections to 2.3.3 to further clarify the Request and
Response processing 

- Updated the example in section 2.5 to include the use of the Index

- Added a high level flow to Appendix C, highlighting a scenario whereby
local policy prevents the History-Info from being sent in the Responses.

Per my previous posting on the -01 updates, the Security solution required
for History-Info will be addressed in a separate draft (which hopefully can
be put forth before the deadline) and the call flows (although one
additional one has been added) still do need more work, but again, I'd like
agreement on the syntax before adding too much more. 

One area that perhaps require further consideration is whether the Index
should be mandatory since we are now only capturing the Targeted-to-URI, it
won't be possible to really interpret a series of captured Request URIs
without an Index.  Right now, the inclusion of an Index is a SHOULD, with
the caveat that if there is no History-Info in a request received by a
Proxy, that it adds an entry based upon the received Request before it
retargets.  For this scenario, an INDEX MUST not be added because there's no
way to know what it should be AND the lack of an index highlights a missing
entry, which might be useful to an end application.  

Please do review and provide feedback.

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Wednesday, February 12, 2003 8:20 AM
Subject: I-D ACTION:draft-barnes-sipping-history-info-02.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: An Extension to the Session Initiation Protocol 
			  for Request History Information
	Author(s)	: M. Barnes, M. Watson
	Filename	: draft-barnes-sipping-history-info-02.txt
	Pages		: 20
	Date		: 2003-2-11
	
This draft defines a standard mechanism for capturing the history 
information associated with a SIP request.  This capability enables 
many enhanced services by providing the information as to how and 
why a call arrives at a specific application or user.  This draft 
defines a new optional SIP header, History-Info, for capturing the 
history information in requests. A new option tag, HistInfo, to be 
included in the Supported header is defined to allow UAs to 
indicate whether the HistInfo should be returned in responses to a 
request which has captured the history information.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-barnes-sipping-history-info-02.txt

To remove yourself from the IETF Announcement list, send a message to 
ietf-announce-request with the word unsubscribe in the body of the message.

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-barnes-sipping-history-info-02.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-barnes-sipping-history-info-02.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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Internet-Draft.


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<P><FONT SIZE=3D2>Hi all,</FONT>
</P>

<P><FONT SIZE=3D2>This document has had several changes as a result of =
the feedback on the mailing list:</FONT>
</P>

<P><FONT SIZE=3D2>- Section 2.1 - corrected the terminology to be =
consistent with the capturing of the single targeted-to-URI and =
corrected the syntax of the index as highlighted by Paul K. </FONT></P>

<P><FONT SIZE=3D2>- In section 2.3, clarified that History-Info can be =
included in provisional responses per John E.'s comment</FONT>
</P>

<P><FONT SIZE=3D2>- Additional normative text around the maintenance of =
the Index added to section 2.3.1 and 2.3.3 per Paul K's comment.</FONT>
<BR><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>- Added additional sections to 2.3.3 to further =
clarify the Request and Response processing </FONT>
</P>

<P><FONT SIZE=3D2>- Updated the example in section 2.5 to include the =
use of the Index</FONT>
</P>

<P><FONT SIZE=3D2>- Added a high level flow to Appendix C, highlighting =
a scenario whereby local policy prevents the History-Info from being =
sent in the Responses.</FONT></P>

<P><FONT SIZE=3D2>Per my previous posting on the -01 updates, the =
Security solution required for History-Info will be addressed in a =
separate draft (which hopefully can be put forth before the deadline) =
and the call flows (although one additional one has been added) still =
do need more work, but again, I'd like agreement on the syntax before =
adding too much more. </FONT></P>

<P><FONT SIZE=3D2>One area that perhaps require further consideration =
is whether the Index should be mandatory since we are now only =
capturing the Targeted-to-URI, it won't be possible to really interpret =
a series of captured Request URIs without an Index.&nbsp; Right now, =
the inclusion of an Index is a SHOULD, with the caveat that if there is =
no History-Info in a request received by a Proxy, that it adds an entry =
based upon the received Request before it retargets.&nbsp; For this =
scenario, an INDEX MUST not be added because there's no way to know =
what it should be AND the lack of an index highlights a missing entry, =
which might be useful to an end application.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>Please do review and provide feedback.</FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>mbarnes@nortelnetworks.com</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Internet-Drafts@ietf.org [<A =
HREF=3D"mailto:Internet-Drafts@ietf.org">mailto:Internet-Drafts@ietf.org=
</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, February 12, 2003 8:20 AM</FONT>
<BR><FONT SIZE=3D2>Subject: I-D =
ACTION:draft-barnes-sipping-history-info-02.txt</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>A New Internet-Draft is available from the on-line =
Internet-Drafts directories.</FONT>
</P>
<BR>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Title&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
An Extension to the Session Initiation Protocol </FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&nbsp; for =
Request History Information</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : M. Barnes, M. =
Watson</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-barnes-sipping-history-info-02.txt</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Pages&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
20</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2003-2-11</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>This draft defines a standard mechanism for =
capturing the history </FONT>
<BR><FONT SIZE=3D2>information associated with a SIP request.&nbsp; =
This capability enables </FONT>
<BR><FONT SIZE=3D2>many enhanced services by providing the information =
as to how and </FONT>
<BR><FONT SIZE=3D2>why a call arrives at a specific application or =
user.&nbsp; This draft </FONT>
<BR><FONT SIZE=3D2>defines a new optional SIP header, History-Info, for =
capturing the </FONT>
<BR><FONT SIZE=3D2>history information in requests. A new option tag, =
HistInfo, to be </FONT>
<BR><FONT SIZE=3D2>included in the Supported header is defined to allow =
UAs to </FONT>
<BR><FONT SIZE=3D2>indicate whether the HistInfo should be returned in =
responses to a </FONT>
<BR><FONT SIZE=3D2>request which has captured the history =
information.</FONT>
</P>

<P><FONT SIZE=3D2>A URL for this Internet-Draft is:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-barnes-sipping-history=
-info-02.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-barnes-sippi=
ng-history-info-02.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>To remove yourself from the IETF Announcement list, =
send a message to </FONT>
<BR><FONT SIZE=3D2>ietf-announce-request with the word unsubscribe in =
the body of the message.</FONT>
</P>

<P><FONT SIZE=3D2>Internet-Drafts are also available by anonymous FTP. =
Login with the username</FONT>
<BR><FONT SIZE=3D2>&quot;anonymous&quot; and a password of your e-mail =
address. After logging in,</FONT>
<BR><FONT SIZE=3D2>type &quot;cd internet-drafts&quot; and then</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>&quot;get =
draft-barnes-sipping-history-info-02.txt&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>A list of Internet-Drafts directories can be found =
in</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A> </FONT>
<BR><FONT SIZE=3D2>or <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Internet-Drafts can also be obtained by =
e-mail.</FONT>
</P>

<P><FONT SIZE=3D2>Send a message to:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>mailserv@ietf.org.</FONT>
<BR><FONT SIZE=3D2>In the body type:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;FILE =
/internet-drafts/draft-barnes-sipping-history-info-02.txt&quot;.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>NOTE:&nbsp;&nbsp; The mail server at ietf.org can =
return the document in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>MIME-encoded form by using the &quot;mpack&quot; =
utility.&nbsp; To use this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>feature, =
insert the command &quot;ENCODING mime&quot; before the =
&quot;FILE&quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>command.&nbsp; To decode the response(s), you will need =
&quot;munpack&quot; or</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>a =
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readers</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>exhibit =
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<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&quot;multipart&quot; MIME messages (i.e. documents which have =
been split</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>up into =
multiple messages), so check your local documentation on</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>how to =
manipulate these messages.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2>Below is the data which will enable a MIME compliant =
mail reader</FONT>
<BR><FONT SIZE=3D2>implementation to automatically retrieve the ASCII =
version of the</FONT>
<BR><FONT SIZE=3D2>Internet-Draft.</FONT>
</P>

<P><FONT FACE=3D"Arial" SIZE=3D2 COLOR=3D"#000000"></FONT>&nbsp;

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<P><FONT SIZE=3D2>Mark,<BR>
<BR>
OK, 1) &amp; 2) seems to be a good suggestion as it would indeed help =
clarify the buffering mechanism better. Yes, you are right, the example =
is inaccurate - it would add up to 221 instead of 225.<BR>
<BR>
regarding 3), though the version field would seem apparently redundant =
(as you've aptly noted the protocol is, spec-wise, perfectly compatible =
with other versions), we believe that it would help debug the protocol =
better - esp, when connecting networks implementing different versions. =
We've known, in the legacy implementations, some debugging issues that =
cropped up, whilst interwoking issue 4 PBX to an issue 6 PBX. So, we'd =
suggest that we retain this parameter. But as Mark (Watson) pointed out, =
the text describing the version field type needs to be made more =
accurate<BR>
<BR>
cheers,<BR>
- ranjith..<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Ashworth, Mark [MOP:MA01:EXCH] [<A =
HREF=3D"mailto:mashwort@europem01.nt.com">mailto:mashwort@europem01.nt.co=
m</A>]<BR>
Sent:&nbsp;&nbsp; Wed 2/12/2003 10:37 PM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; Ranjith Mukundan<BR>
Cc:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] QSIG Vs =
DPNSS MIME<BR>
Ranjith,<BR>
<BR>
Seems like a reasonable thing to do. A few comments which I hope are<BR>
helpful...<BR>
<BR>
1) The discussion of handling segmented information might be clearer if =
you<BR>
were to use the terminology of the DPNSS specification such as =
Selection<BR>
Block/Selection Field and Information Block/Information Field to refer =
to<BR>
the items being manipulated. In the case of the ISRM/SSRM, as others =
have<BR>
already commented, you would need to distinguish between the =
Supplementary<BR>
Information and the Destination Address which make up the Selection =
Block;<BR>
the Supplementary Information can be safely buffered but the =
Destination<BR>
Address needs to be handled according to overlap procedures.<BR>
<BR>
2) Using that terminology might make clearer how you propose to perform =
the<BR>
buffering. I think you are proposing to concatenate the Information =
Fields<BR>
from a sequence of messages to assemble a complete Information Block =
(or<BR>
segment thereof if it exceeds your 255 octet limit) with a single =
message<BR>
type octet, although I don't see this stated explicitly. If so, then I =
think<BR>
your example is slightly inaccurate: combining a sequence of five EEMs =
each<BR>
of 45 octets would produce an EEM of 221 octets (as four message type =
octets<BR>
are removed)?<BR>
<BR>
3) I'm not sure there is really any need for a version number, since =
DPNSS<BR>
has comprehensive compatibility mechanisms?<BR>
<BR>
Regards,<BR>
Mark.<BR>
<BR>
-----Original Message-----<BR>
From: Ranjith Mukundan [<A =
HREF=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro.=
com</A>]<BR>
Sent: 11 February 2003 08:59<BR>
To: sipping@ietf.org<BR>
Subject: [Sipping] QSIG Vs DPNSS MIME<BR>
<BR>
<BR>
<BR>
we'd appreciate some views on this draft from atleast those who =
fairly<BR>
understand QSIG.<BR>
<BR>
of course, others who feel that this MIME type collides directly with =
the<BR>
purpose of SIP are encouraged to provide alternatives bearing in mind =
that<BR>
providing complete end-to-end service transparency over the intervening =
SIP<BR>
based core would be essential, if we are to provide a seamless =
migration<BR>
path from H.323 based core networks - which already provide =
equivalent<BR>
mechanisms for DSS1 based protocols.<BR>
<BR>
- ranjith..<BR>
<BR>
<BR>
-----Original Message-----<BR>
From:&nbsp;&nbsp; Ranjith Mukundan<BR>
Sent:&nbsp;&nbsp; Sat 2/8/2003 12:09 AM<BR>
To:&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org<BR>
Cc:&nbsp;&nbsp;&nbsp;<BR>
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
<BR>
Hi,<BR>
<BR>
posting WRT<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.txt</A>&gt;<BR>
<BR>
<BR>
&gt;&gt;<BR>
&gt;&gt; Why is it so important to be able to tunnel DPNSS in SIP?&nbsp; =
I'd prefer<BR>
&gt;&gt; that folks implement a translation to SIP, where SIP can =
natively carry<BR>
&gt;&gt; all the information that DPNSS is trying to convey.<BR>
&gt;&gt;<BR>
<BR>
To put things in perspective:<BR>
<BR>
1- DPNSS is a protocol (deployed primarily in UK, NZ, Australia and =
parts of<BR>
NA) very similar to QSIG (deployed primarily Europe) for =
interconnecting<BR>
PBXs or centrex customers hosted off LEs. DPNSS (like QSIG) finds =
enormous<BR>
application towards realizing a local VPN or network-wide VPN =
services.<BR>
<BR>
2- This draft aims to do exactly what RFC 3204 (MIME objects for =
QSIG/ISUP)<BR>
does for QSIG with appropriate/relevant customization for DPNSS - i.e., =
to<BR>
provide end-to-end &amp; *complete*, DPNSS Service transparency between =
pockets<BR>
of DPNSS based private networks over a SIP based core.<BR>
<BR>
3- Currently, circuit-switched SS7 based cores, provide equivalent =
DPNSS<BR>
tunneling mechanisms through (ETSI &amp; ANSI) ISUP *apart from* the =
concomitant<BR>
translation/interworking mechanisms to ISUP. The =
translation/interworking<BR>
mechanism to ISUP is limited to the basic call and certain ISUP<BR>
supplementary services where there is a direct/easy equivalent.<BR>
<BR>
4- H.323 also provides equivalent tunneling mechanisms (via the H.450 =
Annexe<BR>
procedures) for transparently carrying DSS1 based signaling between<BR>
DSS1-based network elements/gateways, over a H.323 core.<BR>
<BR>
Some specifics:<BR>
<BR>
1- DPNSS supports, as specified in BTNR 188, approximately 240+<BR>
supplementary features/services (like Call Back When Free, Call Back =
When<BR>
Next Used, Executive Intrusion, Travelling Class of Service, Do Not =
Disturb,<BR>
Night Service, Private Network Route Optimization, Centralized =
Operator,<BR>
Three Party Takeover, Remote Diversion Registration etc). It would be =
almost<BR>
impossible to interwork to equivalent mechanisms in SIP, without losing =
the<BR>
service transparency when using an intervening SIP based core.<BR>
<BR>
2- Yes, we definitely need an interworking/translation mechanism to map =
all<BR>
the basic call/service paremeters, and procedures for mapping the =
messages.<BR>
But this does not preclude the need for a DPNSS MIME type, but would =
rather<BR>
be complimentary to the MIME type mechanism. Similar to<BR>
<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-0=
0.txt">http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00=
.txt</A>&gt;&nbsp; ,<BR>
for QSIG, we are in the process of putting together an interworking =
draft<BR>
for DPNSS as well - but rather than making it DPNSS specific, we are =
trying<BR>
to make it a common draft to cover PRI, DPNSS &amp; DASS2 [as Q.SIG =
interworking<BR>
is already a SIPPING WG ID]. Note that, PRI and DASS2 will not need a =
MIME<BR>
type as these are, as one might be aware, for PSTN access, unlike DPNSS =
and<BR>
QSIG (which are private network protocols).<BR>
<BR>
3- From the SIP perspective, carrying non-SIP applications as MIME =
objects<BR>
to ensure end to end service transparency is architecturally less =
appalling<BR>
than, say for e.g., tunneling DSS1 protocols in H.323 based network - so =
the<BR>
SIP DPNSS MIME proposition as per this I-D is the obvious way forward =
for<BR>
DPNSS service transparency over a SIP based core, especially now that we =
see<BR>
acute deployment needs to be met.<BR>
<BR>
cheers,<BR>
ranjith..<BR>
<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Ranjith Mukundan<BR>
&gt; Sent: Tuesday, February 04, 2003 5:32 PM<BR>
&gt; To: sipping@ietf.org<BR>
&gt; Subject: RE: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; Hi List,<BR>
&gt; We would like you take a look at this draft and revert with any<BR>
&gt; comments/queries that you might have.<BR>
&gt;<BR>
&gt; SIPPING Chairs,<BR>
&gt; The authors feel that the requirement and the need for this MIME =
type<BR>
&gt; is<BR>
&gt; quite acute. Hence we'd like you to consider this for inclusion as =
a<BR>
&gt; WG I-D<BR>
&gt; and suggest the way forward.<BR>
&gt;<BR>
&gt; - ranjith..<BR>
&gt;<BR>
&gt; -----Original Message-----<BR>
&gt; From: sipping-admin@ietf.org [ <A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A><=
BR>
&lt;<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>&=
gt; ]On Behalf<BR>
&gt; Of<BR>
&gt; Internet-Drafts@ietf.org<BR>
&gt; Sent: Tuesday, February 04, 2003 5:14 PM<BR>
&gt; To: IETF-Announce:<BR>
&gt; Cc: sipping@ietf.org<BR>
&gt; Subject: [Sipping] I-D =
ACTION:draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;<BR>
&gt;<BR>
&gt; A New Internet-Draft is available from the on-line =
Internet-Drafts<BR>
&gt; directories.<BR>
&gt;<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : MIME =
media types for DPNSS Objects<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : R. Mukundan et al.<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-mukundan-sipping-dpnss-01.txt<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
9<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2003-2-3<BR>
&gt;<BR>
&gt; This document describes Multipart Internet Mail Extensions (MIME) =
type<BR>
&gt; for application/Digital Private Network Signaling System =
(DPNSS)<BR>
&gt; objects for use in Session Initiation Protocol (SIP) =
applications,<BR>
&gt; according to the rules defined in RFC 2048. These types can be used =
to<BR>
&gt; identify Digital Private Network Signaling System 1 (DPNSS 1) =
objects<BR>
&gt; within a SIP message such as INVITE or INFO, as might be =
implemented<BR>
&gt; when using SIP in an environment where part of the call =
involves<BR>
&gt; interworking to DPNSS 1 based Private Networks. RFC 3204 addresses =
a<BR>
&gt; similar need for Integrated Service Digital Network User Part =
(ISUP)<BR>
&gt; and Q Signaling (QSIG).<BR>
&gt;<BR>
&gt; A URL for this Internet-Draft is:<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A><BR>
&lt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-=
01.tx">http://www.ietf.org/internet-drafts/draft-mukundan-sipping-dpnss-0=
1.tx</A>&gt;<BR>
&gt; t<BR>
&gt;<BR>
&gt; To remove yourself from the IETF Announcement list, send a message =
to<BR>
&gt; ietf-announce-request with the word unsubscribe in the body of =
the<BR>
&gt; message.<BR>
&gt;<BR>
&gt; Internet-Drafts are also available by anonymous FTP. Login with =
the<BR>
&gt; username<BR>
&gt; &quot;anonymous&quot; and a password of your e-mail address. After =
logging in,<BR>
&gt; type &quot;cd internet-drafts&quot; and then<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;get =
draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; A list of Internet-Drafts directories can be found in<BR>
&gt; <A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A> &lt;<A =
HREF=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/A>&gt;&nbsp; or<BR>
&gt; <A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</A><BR>
&lt;<A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
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&gt;<BR>
&gt;<BR>
&gt; Internet-Drafts can also be obtained by e-mail.<BR>
&gt;<BR>
&gt; Send a message to:<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; mailserv@ietf.org.<BR>
&gt; In the body type:<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;FILE =
/internet-drafts/draft-mukundan-sipping-dpnss-01.txt&quot;.<BR>
&gt;<BR>
&gt; NOTE: The mail server at ietf.org can return the document in<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MIME-encoded form by using the =
&quot;mpack&quot; utility.&nbsp; To use this<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; feature, insert the command =
&quot;ENCODING mime&quot; before the &quot;FILE&quot;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; command.&nbsp; To decode the =
response(s), you will need &quot;munpack&quot; or<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a MIME-compliant mail =
reader.&nbsp; Different MIME-compliant mail readers<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; exhibit different behavior, =
especially when dealing with<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;multipart&quot; MIME =
messages (i.e. documents which have been split<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; up into multiple messages), so =
check your local documentation on<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; how to manipulate these =
messages.<BR>
&gt;<BR>
&gt;<BR>
&gt; Below is the data which will enable a MIME compliant mail =
reader<BR>
&gt; implementation to automatically retrieve the ASCII version of =
the<BR>
&gt; Internet-Draft.<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A><BR>
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&gt; This list is for NEW development of the application of SIP Use<BR>
&gt; sip-implementors@cs.columbia.edu for questions on current sip =
Use<BR>
&gt; sip@ietf.org for new developments of core SIP<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
<BR>
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To: "'Ranjith Mukundan'" <ranjith.mukundan@wipro.com>,
        "Peterson, Jon"
	 <jon.peterson@neustar.biz>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sipping@ietf.org, "'Mark Watson'" <mwatson@nortelnetworks.com>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Wed, 12 Feb 2003 13:48:39 -0600
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> -----Original Message-----
> From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
> 
> In case of DPNSS too the message semantics matches SIP 
> messages - atleast to
> the same extent as it does for ISUP.

I think you mean "at least to the same extent as it does
for QSIG."

DPNSS, QSIG, and ISDN are all UNI protocols. ISUP is
an NNI protocol. SIP is end-to-end, which really
ends up acting more like an NNI protocol than a UNI
protocol.

/a
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Date: Wed, 12 Feb 2003 20:05:38 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sipping] New I-D on URI leasing
In-reply-to: <3E49E4B5.8010608@dynamicsoft.com>
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Just to clarify and expand on my earlier short comments...

I'm not unhappy that the solution proposed in the draft  is similar to the 
one discussed in cc-transfer.   The method does work,  and we can  discuss 
whether it should be two REGISTERs or one LEASE and one REGISTER.  However, 
both approaches are multi-step, which others objected to when it was 
discussed at IETF-55.

I had hoped that a new proposal was coming, or a way to make Rohan's 
escaped Route header in a Contact work.    This approach, (4.4 in this 
draft) does not actually suffer from the problems listed - if multiple 
proxies must be traversed, multiple URIs can be escaped into the Route 
header.  The real problem with this approach is that it violates Table 1 in 
RFC 3261 so would be ignored by all RFC  3261 compliant UAs.  The backwards 
compatibility problem with pre-bis-04 UAs is even worse.  Since under the 
old Record-Route rules the Contact is included in the Route header, you 
could potentially have a URI in a Route header with  an escaped Route 
header - yuk!  However, there might be a clever solution to work around 
this problem.  This is the only solution that I know of that meets REQ-8.

REQ-9 is a bit strange - the anonymity of a Contact is a very small part of 
the total anonymity  picture.   It seems this requirement could simply be 
met by having the user portion  of the Contact be a random string.  The 
privacy discussion in 4.2 about the call center operator is simplistic - 
for privacy, much more than just fixing Contact is required and would be 
used in practice.

I'm not sure understand the motivation for REQ-11.  I don't know why a UA 
would want to generate a unique Contact for each dialog - it seems like 
quite a burden on the UA and a difficult requirement for a solution to 
meet.  It seems like there is some sort of security  reason for this, but 
it is not stated.

One minor correction - the transfer problem is with  attended transfer 
only, not blind transfer or transfer in general.

As this is an important topic for many  applications, it is important that 
we get some good discussion going on the requirements and possible solutions.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

At 01:07 AM 2/12/2003 -0500, Jonathan Rosenberg wrote:
>Folks,
>
>I've just submitted an I-D on the concept of "URI leasing". THis is my 
>proposed technique for solving the "globally routable UA URI" problem. 
>SPecifically, for cases like REFER, how do we construct a URI that routes 
>to a specific UA instance in a domain? At the last ietf meeting, this 
>issue came up in several contexts. I suggested a single document to 
>explore the problem. The chairs asked for volunteers, and I said I'd write 
>one, time permitting, if no one else stepped up. Since no one did, well, 
>here it is.
>
>Until it appears in the archives, you can pick it up at:
>http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.txt
>http://www.jdrosen.net/papers/draft-rosenberg-sipping-lease-00.html
>
>Thanks,
>Jonathan R.
>--
>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
To: "Adam Roach" <adam@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sipping@ietf.org>, "'Mark Watson'" <mwatson@nortelnetworks.com>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Thu, 13 Feb 2003 10:21:45 +0530
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PRI, DASS2 etc are all 'true' UNI protocols - but not QSIG and DPNSS. I
would qualifiy DPNSS & QSIG as UUI (user-to-user) protocols as they are used
between PBXs (PINX in QSIG terminology) in a private network (PISN in QSIG
terminology). I mean, you could view QSIG and DPNSS as an
ISUP-protocol-equivalent for private networks. QSIG and ISDN are
peer-to-peer/end-to-end protocols unlike PRI (and DASS2) - making it both
SIP-like (atleast to the same extent as ISUP) from the *protocol
architecture* point of view.

From the protocol message semantic point of view, DPNSS matches SIP messages
to a larger extent than QSIG - esp wrt the session setup/progress/tear-down
messages (as opposed to the link/circuit maintenance messages) used for the
basic call *and* supplementary services like call transfer, three way
conference etc. As an example that I cited in an earlier mail, mid call QSIG
messages like INFO, NOTIFY & FAC would *all* get carried in SIP INFO
method - where as, in case of DPNSS, the only mid call message, and which
has a direct parlance to SIP INFO is EEM (end to end message; equivalent to
ISUP PAM and to some extent ISUP APM). For this reason, I'd think, DPNSS
message semantics matches SIP atleast to the same extent as it does for
ISUP.

cheers,
- ranjith..

-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Thursday, February 13, 2003 1:19 AM
To: 'Ranjith Mukundan'; Peterson, Jon; Jonathan Rosenberg
Cc: sipping@ietf.org; 'Mark Watson'
Subject: RE: [Sipping] QSIG Vs DPNSS MIME


> -----Original Message-----
> From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
>
> In case of DPNSS too the message semantics matches SIP
> messages - atleast to
> the same extent as it does for ISUP.

I think you mean "at least to the same extent as it does
for QSIG."

DPNSS, QSIG, and ISDN are all UNI protocols. ISUP is
an NNI protocol. SIP is end-to-end, which really
ends up acting more like an NNI protocol than a UNI
protocol.

/a

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From: margaretmary <margaret_mary@dexceldesigns.com>
To: <sipping@ietf.org>
Date: Thu, 13 Feb 2003 10:11:53 +0530
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This is a multi-part message in MIME format.

------=_NextPart_000_00D6_01C2D348.54C4B9C0
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Hello sir,

The draft specifed below for conference server model framework

SIPPING Working Group =20
Internet Draft =20
Document: draft-ietf-sipping-cc-framework-01.txt     =20
June 2002                                                                =
                                  =20
Expires: December 2002                              =20

This statement is taken form the above mentioned draft.
Please some one elaborate this statement

"Below are listed several call control "actions" which establish or
    modify dialogs and relate the participants in a conversation space.
    The names of the actions listed are for descriptive purposes only
    (they are not normative).  This list of actions is not meant to be
    exhaustive."

This section 5.0 of the draft is only theoritical or practical =
implementation can be done.

please try to provide a solution for this



Regards,
Margaret

------=_NextPart_000_00D6_01C2D348.54C4B9C0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello sir,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>The draft specifed below for conference =
server=20
model framework</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>SIPPING Working =
Group&nbsp;&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Internet Draft&nbsp;&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Document:=20
draft-ietf-sipping-cc-framework-01.txt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>June=20
2002&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Expires: December=20
2002&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>This statement is taken form the above =
mentioned=20
draft.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Please some one elaborate this=20
statement</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;</DIV></FONT>
<DIV><FONT face=3DArial size=3D2><STRONG>"Below are listed several call =
control=20
"actions" which establish or<BR>&nbsp;&nbsp;&nbsp; modify dialogs and =
relate the=20
participants in a conversation space.<BR>&nbsp;&nbsp;&nbsp; The names of =
the=20
actions listed are for descriptive purposes only<BR>&nbsp;&nbsp;&nbsp; =
(they are=20
not normative).&nbsp; This list of actions is not meant to=20
be<BR>&nbsp;&nbsp;&nbsp; exhaustive."</STRONG></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>This section 5.0 of the draft is only =
theoritical=20
or practical implementation can be done.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>please try to provide a solution for=20
this</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial =
size=3D2>Regards,<BR>Margaret</FONT></DIV></BODY></HTML>

<br>--------------------------------------------------------------<br>Dexcel Electronics Designs (P) Ltd., Bangalore, India<br>
------=_NextPart_000_00D6_01C2D348.54C4B9C0--

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From mailnull@www1.ietf.org  Thu Feb 13 01:47:37 2003
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for conference framework, refer this draft
http://www.jdrosen.net/papers/draft-rosenberg-sipping-conferencing-framework
-01.txt

Ranjit



  -----Original Message-----
  From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
margaretmary
  Sent: Thursday, February 13, 2003 10:12 AM
  To: sipping@ietf.org
  Subject: [Sipping] query on conference server


  Hello sir,

  The draft specifed below for conference server model framework

  SIPPING Working Group
  Internet Draft
  Document: draft-ietf-sipping-cc-framework-01.txt
  June 2002
  Expires: December 2002

  This statement is taken form the above mentioned draft.
  Please some one elaborate this statement

  "Below are listed several call control "actions" which establish or
      modify dialogs and relate the participants in a conversation space.
      The names of the actions listed are for descriptive purposes only
      (they are not normative).  This list of actions is not meant to be
      exhaustive."

  This section 5.0 of the draft is only theoritical or practical
implementation can be done.

  please try to provide a solution for this



  Regards,
  Margaret

  --------------------------------------------------------------
  Dexcel Electronics Designs (P) Ltd., Bangalore, India


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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D859564506-13022003>for=20
conference framework, refer this draft</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff><SPAN class=3D859564506-13022003>
<P><A=20
href=3D"http://www.jdrosen.net/papers/draft-rosenberg-sipping-conferencin=
g-framework-01.txt"><FONT=20
face=3DArial=20
size=3D2>http://www.jdrosen.net/papers/draft-rosenberg-sipping-conferenci=
ng-framework-01.txt</FONT></A></P>
<P><FONT face=3DArial size=3D2><SPAN=20
class=3D859564506-13022003>Ranjit</SPAN></FONT></P>
<P>&nbsp;</P></SPAN></FONT></DIV>
<BLOCKQUOTE>
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> =
sipping-admin@ietf.org=20
  [mailto:sipping-admin@ietf.org]<B>On Behalf Of=20
  </B>margaretmary<BR><B>Sent:</B> Thursday, February 13, 2003 10:12=20
  AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] query =
on=20
  conference server<BR><BR></DIV></FONT>
  <DIV><FONT face=3DArial size=3D2>Hello sir,</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>The draft specifed below for =
conference server=20
  model framework</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>SIPPING Working =
Group&nbsp;&nbsp;</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>Internet =
Draft&nbsp;&nbsp;</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>Document:=20
  draft-ietf-sipping-cc-framework-01.txt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>June=20
  =
2002&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;=20
  </FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>Expires: December=20
  =
2002&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  </FONT></DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>This statement is taken form the =
above mentioned=20
  draft.</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>Please some one elaborate this=20
  statement</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>&nbsp;</DIV></FONT>
  <DIV><FONT face=3DArial size=3D2><STRONG>"Below are listed several =
call control=20
  "actions" which establish or<BR>&nbsp;&nbsp;&nbsp; modify dialogs and =
relate=20
  the participants in a conversation space.<BR>&nbsp;&nbsp;&nbsp; The =
names of=20
  the actions listed are for descriptive purposes =
only<BR>&nbsp;&nbsp;&nbsp;=20
  (they are not normative).&nbsp; This list of actions is not meant to=20
  be<BR>&nbsp;&nbsp;&nbsp; exhaustive."</STRONG></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>This section 5.0 of the draft is only =
theoritical=20
  or practical implementation can be done.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>please try to provide a solution for=20
  this</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial=20
  =
size=3D2>Regards,<BR>Margaret</FONT></DIV><BR>---------------------------=
-----------------------------------<BR>Dexcel=20
  Electronics Designs (P) Ltd., Bangalore, =
India<BR></BLOCKQUOTE></BODY></HTML>

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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Ranjith Mukundan'" <ranjith.mukundan@wipro.com>,
        Adam Roach
	 <adam@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sipping@ietf.org
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Thu, 13 Feb 2003 09:25:42 -0000
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Q.SIG and DPNSS are private NNI protocols, certainly not UNIs. They can both
be used for connection of private networks to virtual private network
services offered by the public network, but in these cases the public
network 'emulates' a private network exchange, so although there is a
physical connection between private and public networks, it is still
logically a private NNI.

...Mark

> -----Original Message-----
> From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
> Sent: 13 February 2003 04:52
> To: Adam Roach; Peterson, Jon; Jonathan Rosenberg
> Cc: sipping@ietf.org; Watson, Mark [MOP:EP10:EXCH]
> Subject: RE: [Sipping] QSIG Vs DPNSS MIME
> 
> 
> PRI, DASS2 etc are all 'true' UNI protocols - but not QSIG 
> and DPNSS. I
> would qualifiy DPNSS & QSIG as UUI (user-to-user) protocols 
> as they are used
> between PBXs (PINX in QSIG terminology) in a private network 
> (PISN in QSIG
> terminology). I mean, you could view QSIG and DPNSS as an
> ISUP-protocol-equivalent for private networks. QSIG and ISDN are
> peer-to-peer/end-to-end protocols unlike PRI (and DASS2) - 
> making it both
> SIP-like (atleast to the same extent as ISUP) from the *protocol
> architecture* point of view.
> 
> From the protocol message semantic point of view, DPNSS 
> matches SIP messages
> to a larger extent than QSIG - esp wrt the session 
> setup/progress/tear-down
> messages (as opposed to the link/circuit maintenance 
> messages) used for the
> basic call *and* supplementary services like call transfer, three way
> conference etc. As an example that I cited in an earlier 
> mail, mid call QSIG
> messages like INFO, NOTIFY & FAC would *all* get carried in SIP INFO
> method - where as, in case of DPNSS, the only mid call 
> message, and which
> has a direct parlance to SIP INFO is EEM (end to end message; 
> equivalent to
> ISUP PAM and to some extent ISUP APM). For this reason, I'd 
> think, DPNSS
> message semantics matches SIP atleast to the same extent as 
> it does for
> ISUP.
> 
> cheers,
> - ranjith..
> 
> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Thursday, February 13, 2003 1:19 AM
> To: 'Ranjith Mukundan'; Peterson, Jon; Jonathan Rosenberg
> Cc: sipping@ietf.org; 'Mark Watson'
> Subject: RE: [Sipping] QSIG Vs DPNSS MIME
> 
> 
> > -----Original Message-----
> > From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
> >
> > In case of DPNSS too the message semantics matches SIP
> > messages - atleast to
> > the same extent as it does for ISUP.
> 
> I think you mean "at least to the same extent as it does
> for QSIG."
> 
> DPNSS, QSIG, and ISDN are all UNI protocols. ISUP is
> an NNI protocol. SIP is end-to-end, which really
> ends up acting more like an NNI protocol than a UNI
> protocol.
> 
> /a
> 
> 

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<TITLE>RE: [Sipping] QSIG Vs DPNSS MIME</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Q.SIG and DPNSS are private NNI protocols, certainly =
not UNIs. They can both be used for connection of private networks to =
virtual private network services offered by the public network, but in =
these cases the public network 'emulates' a private network exchange, =
so although there is a physical connection between private and public =
networks, it is still logically a private NNI.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Ranjith Mukundan [<A =
HREF=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro=
.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 13 February 2003 04:52</FONT>
<BR><FONT SIZE=3D2>&gt; To: Adam Roach; Peterson, Jon; Jonathan =
Rosenberg</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sipping@ietf.org; Watson, Mark =
[MOP:EP10:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sipping] QSIG Vs DPNSS =
MIME</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; PRI, DASS2 etc are all 'true' UNI protocols - =
but not QSIG </FONT>
<BR><FONT SIZE=3D2>&gt; and DPNSS. I</FONT>
<BR><FONT SIZE=3D2>&gt; would qualifiy DPNSS &amp; QSIG as UUI =
(user-to-user) protocols </FONT>
<BR><FONT SIZE=3D2>&gt; as they are used</FONT>
<BR><FONT SIZE=3D2>&gt; between PBXs (PINX in QSIG terminology) in a =
private network </FONT>
<BR><FONT SIZE=3D2>&gt; (PISN in QSIG</FONT>
<BR><FONT SIZE=3D2>&gt; terminology). I mean, you could view QSIG and =
DPNSS as an</FONT>
<BR><FONT SIZE=3D2>&gt; ISUP-protocol-equivalent for private networks. =
QSIG and ISDN are</FONT>
<BR><FONT SIZE=3D2>&gt; peer-to-peer/end-to-end protocols unlike PRI =
(and DASS2) - </FONT>
<BR><FONT SIZE=3D2>&gt; making it both</FONT>
<BR><FONT SIZE=3D2>&gt; SIP-like (atleast to the same extent as ISUP) =
from the *protocol</FONT>
<BR><FONT SIZE=3D2>&gt; architecture* point of view.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; From the protocol message semantic point of =
view, DPNSS </FONT>
<BR><FONT SIZE=3D2>&gt; matches SIP messages</FONT>
<BR><FONT SIZE=3D2>&gt; to a larger extent than QSIG - esp wrt the =
session </FONT>
<BR><FONT SIZE=3D2>&gt; setup/progress/tear-down</FONT>
<BR><FONT SIZE=3D2>&gt; messages (as opposed to the link/circuit =
maintenance </FONT>
<BR><FONT SIZE=3D2>&gt; messages) used for the</FONT>
<BR><FONT SIZE=3D2>&gt; basic call *and* supplementary services like =
call transfer, three way</FONT>
<BR><FONT SIZE=3D2>&gt; conference etc. As an example that I cited in =
an earlier </FONT>
<BR><FONT SIZE=3D2>&gt; mail, mid call QSIG</FONT>
<BR><FONT SIZE=3D2>&gt; messages like INFO, NOTIFY &amp; FAC would =
*all* get carried in SIP INFO</FONT>
<BR><FONT SIZE=3D2>&gt; method - where as, in case of DPNSS, the only =
mid call </FONT>
<BR><FONT SIZE=3D2>&gt; message, and which</FONT>
<BR><FONT SIZE=3D2>&gt; has a direct parlance to SIP INFO is EEM (end =
to end message; </FONT>
<BR><FONT SIZE=3D2>&gt; equivalent to</FONT>
<BR><FONT SIZE=3D2>&gt; ISUP PAM and to some extent ISUP APM). For this =
reason, I'd </FONT>
<BR><FONT SIZE=3D2>&gt; think, DPNSS</FONT>
<BR><FONT SIZE=3D2>&gt; message semantics matches SIP atleast to the =
same extent as </FONT>
<BR><FONT SIZE=3D2>&gt; it does for</FONT>
<BR><FONT SIZE=3D2>&gt; ISUP.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; cheers,</FONT>
<BR><FONT SIZE=3D2>&gt; - ranjith..</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Adam Roach [<A =
HREF=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, February 13, 2003 1:19 =
AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Ranjith Mukundan'; Peterson, Jon; Jonathan =
Rosenberg</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sipping@ietf.org; 'Mark Watson'</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sipping] QSIG Vs DPNSS =
MIME</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: Ranjith Mukundan [<A =
HREF=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro=
.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; In case of DPNSS too the message semantics =
matches SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; messages - atleast to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the same extent as it does for =
ISUP.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I think you mean &quot;at least to the same =
extent as it does</FONT>
<BR><FONT SIZE=3D2>&gt; for QSIG.&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; DPNSS, QSIG, and ISDN are all UNI protocols. =
ISUP is</FONT>
<BR><FONT SIZE=3D2>&gt; an NNI protocol. SIP is end-to-end, which =
really</FONT>
<BR><FONT SIZE=3D2>&gt; ends up acting more like an NNI protocol than a =
UNI</FONT>
<BR><FONT SIZE=3D2>&gt; protocol.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; /a</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From: "Ranjith Mukundan" <ranjith.mukundan@wipro.com>
To: "Watson, Mark [MOP:EP10:EXCH]" <mwatson@europem01.nt.com>,
        "Adam Roach" <adam@dynamicsoft.com>,
        "Peterson, Jon" <jon.peterson@neustar.biz>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] QSIG Vs DPNSS MIME
Date: Thu, 13 Feb 2003 15:35:17 +0530
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RE: [Sipping] QSIG Vs DPNSS MIMEyes. this *private NNI* was what i was
refering to as "UUI".
  -----Original Message-----
  From: Watson, Mark [MOP:EP10:EXCH] [mailto:mwatson@europem01.nt.com]
  Sent: Thursday, February 13, 2003 2:56 PM
  To: 'Ranjith Mukundan'; Adam Roach; Peterson, Jon; Jonathan Rosenberg
  Cc: sipping@ietf.org
  Subject: RE: [Sipping] QSIG Vs DPNSS MIME


  Q.SIG and DPNSS are private NNI protocols, certainly not UNIs. They can
both be used for connection of private networks to virtual private network
services offered by the public network, but in these cases the public
network 'emulates' a private network exchange, so although there is a
physical connection between private and public networks, it is still
logically a private NNI.

  ...Mark

  > -----Original Message-----
  > From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
  > Sent: 13 February 2003 04:52
  > To: Adam Roach; Peterson, Jon; Jonathan Rosenberg
  > Cc: sipping@ietf.org; Watson, Mark [MOP:EP10:EXCH]
  > Subject: RE: [Sipping] QSIG Vs DPNSS MIME
  >
  >
  > PRI, DASS2 etc are all 'true' UNI protocols - but not QSIG
  > and DPNSS. I
  > would qualifiy DPNSS & QSIG as UUI (user-to-user) protocols
  > as they are used
  > between PBXs (PINX in QSIG terminology) in a private network
  > (PISN in QSIG
  > terminology). I mean, you could view QSIG and DPNSS as an
  > ISUP-protocol-equivalent for private networks. QSIG and ISDN are
  > peer-to-peer/end-to-end protocols unlike PRI (and DASS2) -
  > making it both
  > SIP-like (atleast to the same extent as ISUP) from the *protocol
  > architecture* point of view.
  >
  > From the protocol message semantic point of view, DPNSS
  > matches SIP messages
  > to a larger extent than QSIG - esp wrt the session
  > setup/progress/tear-down
  > messages (as opposed to the link/circuit maintenance
  > messages) used for the
  > basic call *and* supplementary services like call transfer, three way
  > conference etc. As an example that I cited in an earlier
  > mail, mid call QSIG
  > messages like INFO, NOTIFY & FAC would *all* get carried in SIP INFO
  > method - where as, in case of DPNSS, the only mid call
  > message, and which
  > has a direct parlance to SIP INFO is EEM (end to end message;
  > equivalent to
  > ISUP PAM and to some extent ISUP APM). For this reason, I'd
  > think, DPNSS
  > message semantics matches SIP atleast to the same extent as
  > it does for
  > ISUP.
  >
  > cheers,
  > - ranjith..
  >
  > -----Original Message-----
  > From: Adam Roach [mailto:adam@dynamicsoft.com]
  > Sent: Thursday, February 13, 2003 1:19 AM
  > To: 'Ranjith Mukundan'; Peterson, Jon; Jonathan Rosenberg
  > Cc: sipping@ietf.org; 'Mark Watson'
  > Subject: RE: [Sipping] QSIG Vs DPNSS MIME
  >
  >
  > > -----Original Message-----
  > > From: Ranjith Mukundan [mailto:ranjith.mukundan@wipro.com]
  > >
  > > In case of DPNSS too the message semantics matches SIP
  > > messages - atleast to
  > > the same extent as it does for ISUP.
  >
  > I think you mean "at least to the same extent as it does
  > for QSIG."
  >
  > DPNSS, QSIG, and ISDN are all UNI protocols. ISUP is
  > an NNI protocol. SIP is end-to-end, which really
  > ends up acting more like an NNI protocol than a UNI
  > protocol.
  >
  > /a
  >
  >


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<HTML><HEAD><TITLE>RE: [Sipping] QSIG Vs DPNSS MIME</TITLE>
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<DIV><SPAN class=3D542480210-13022003><FONT face=3DArial color=3D#0000ff =
size=3D2>yes.=20
this *private NNI* was what i was refering to as "UUI". =
</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Watson, Mark =
[MOP:EP10:EXCH]=20
  [mailto:mwatson@europem01.nt.com]<BR><B>Sent:</B> Thursday, February =
13, 2003=20
  2:56 PM<BR><B>To:</B> 'Ranjith Mukundan'; Adam Roach; Peterson, Jon; =
Jonathan=20
  Rosenberg<BR><B>Cc:</B> sipping@ietf.org<BR><B>Subject:</B> RE: =
[Sipping] QSIG=20
  Vs DPNSS MIME<BR><BR></FONT></DIV>
  <P><FONT size=3D2>Q.SIG and DPNSS are private NNI protocols, certainly =
not UNIs.=20
  They can both be used for connection of private networks to virtual =
private=20
  network services offered by the public network, but in these cases the =
public=20
  network 'emulates' a private network exchange, so although there is a =
physical=20
  connection between private and public networks, it is still logically =
a=20
  private NNI.</FONT></P>
  <P><FONT size=3D2>...Mark</FONT> </P>
  <P><FONT size=3D2>&gt; -----Original Message-----</FONT> <BR><FONT =
size=3D2>&gt;=20
  From: Ranjith Mukundan [<A=20
  =
href=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro.=
com</A>]</FONT>=20
  <BR><FONT size=3D2>&gt; Sent: 13 February 2003 04:52</FONT> <BR><FONT=20
  size=3D2>&gt; To: Adam Roach; Peterson, Jon; Jonathan Rosenberg</FONT> =
<BR><FONT=20
  size=3D2>&gt; Cc: sipping@ietf.org; Watson, Mark =
[MOP:EP10:EXCH]</FONT>=20
  <BR><FONT size=3D2>&gt; Subject: RE: [Sipping] QSIG Vs DPNSS =
MIME</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; PRI, DASS2 etc are all 'true' UNI protocols - but not =
QSIG=20
  </FONT><BR><FONT size=3D2>&gt; and DPNSS. I</FONT> <BR><FONT =
size=3D2>&gt; would=20
  qualifiy DPNSS &amp; QSIG as UUI (user-to-user) protocols =
</FONT><BR><FONT=20
  size=3D2>&gt; as they are used</FONT> <BR><FONT size=3D2>&gt; between =
PBXs (PINX=20
  in QSIG terminology) in a private network </FONT><BR><FONT =
size=3D2>&gt; (PISN=20
  in QSIG</FONT> <BR><FONT size=3D2>&gt; terminology). I mean, you could =
view QSIG=20
  and DPNSS as an</FONT> <BR><FONT size=3D2>&gt; =
ISUP-protocol-equivalent for=20
  private networks. QSIG and ISDN are</FONT> <BR><FONT size=3D2>&gt;=20
  peer-to-peer/end-to-end protocols unlike PRI (and DASS2) - =
</FONT><BR><FONT=20
  size=3D2>&gt; making it both</FONT> <BR><FONT size=3D2>&gt; SIP-like =
(atleast to=20
  the same extent as ISUP) from the *protocol</FONT> <BR><FONT =
size=3D2>&gt;=20
  architecture* point of view.</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; From the protocol message semantic point of view, DPNSS=20
  </FONT><BR><FONT size=3D2>&gt; matches SIP messages</FONT> <BR><FONT =
size=3D2>&gt;=20
  to a larger extent than QSIG - esp wrt the session </FONT><BR><FONT=20
  size=3D2>&gt; setup/progress/tear-down</FONT> <BR><FONT size=3D2>&gt; =
messages (as=20
  opposed to the link/circuit maintenance </FONT><BR><FONT size=3D2>&gt; =
messages)=20
  used for the</FONT> <BR><FONT size=3D2>&gt; basic call *and* =
supplementary=20
  services like call transfer, three way</FONT> <BR><FONT size=3D2>&gt; =
conference=20
  etc. As an example that I cited in an earlier </FONT><BR><FONT =
size=3D2>&gt;=20
  mail, mid call QSIG</FONT> <BR><FONT size=3D2>&gt; messages like INFO, =
NOTIFY=20
  &amp; FAC would *all* get carried in SIP INFO</FONT> <BR><FONT =
size=3D2>&gt;=20
  method - where as, in case of DPNSS, the only mid call =
</FONT><BR><FONT=20
  size=3D2>&gt; message, and which</FONT> <BR><FONT size=3D2>&gt; has a =
direct=20
  parlance to SIP INFO is EEM (end to end message; </FONT><BR><FONT =
size=3D2>&gt;=20
  equivalent to</FONT> <BR><FONT size=3D2>&gt; ISUP PAM and to some =
extent ISUP=20
  APM). For this reason, I'd </FONT><BR><FONT size=3D2>&gt; think, =
DPNSS</FONT>=20
  <BR><FONT size=3D2>&gt; message semantics matches SIP atleast to the =
same extent=20
  as </FONT><BR><FONT size=3D2>&gt; it does for</FONT> <BR><FONT =
size=3D2>&gt;=20
  ISUP.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
cheers,</FONT>=20
  <BR><FONT size=3D2>&gt; - ranjith..</FONT> <BR><FONT size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; -----Original Message-----</FONT> =
<BR><FONT=20
  size=3D2>&gt; From: Adam Roach [<A=20
  =
href=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</FO=
NT>=20
  <BR><FONT size=3D2>&gt; Sent: Thursday, February 13, 2003 1:19 =
AM</FONT>=20
  <BR><FONT size=3D2>&gt; To: 'Ranjith Mukundan'; Peterson, Jon; =
Jonathan=20
  Rosenberg</FONT> <BR><FONT size=3D2>&gt; Cc: sipping@ietf.org; 'Mark=20
  Watson'</FONT> <BR><FONT size=3D2>&gt; Subject: RE: [Sipping] QSIG Vs =
DPNSS=20
  MIME</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; &gt; -----Original Message-----</FONT> =
<BR><FONT=20
  size=3D2>&gt; &gt; From: Ranjith Mukundan [<A=20
  =
href=3D"mailto:ranjith.mukundan@wipro.com">mailto:ranjith.mukundan@wipro.=
com</A>]</FONT>=20
  <BR><FONT size=3D2>&gt; &gt;</FONT> <BR><FONT size=3D2>&gt; &gt; In =
case of DPNSS=20
  too the message semantics matches SIP</FONT> <BR><FONT size=3D2>&gt; =
&gt;=20
  messages - atleast to</FONT> <BR><FONT size=3D2>&gt; &gt; the same =
extent as it=20
  does for ISUP.</FONT> <BR><FONT size=3D2>&gt; </FONT><BR><FONT =
size=3D2>&gt; I=20
  think you mean "at least to the same extent as it does</FONT> =
<BR><FONT=20
  size=3D2>&gt; for QSIG."</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; DPNSS, QSIG, and ISDN are all UNI protocols. ISUP =
is</FONT>=20
  <BR><FONT size=3D2>&gt; an NNI protocol. SIP is end-to-end, which =
really</FONT>=20
  <BR><FONT size=3D2>&gt; ends up acting more like an NNI protocol than =
a=20
  UNI</FONT> <BR><FONT size=3D2>&gt; protocol.</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; /a</FONT> <BR><FONT size=3D2>&gt; =
</FONT><BR><FONT=20
  size=3D2>&gt; </FONT></P></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
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From: "Brierre, Jean Joseph" <Jean.Joseph.Brierre@wcom.com>
To: "Sinnreich, Henry" <Henry.Sinnreich@wcom.com>,
        "'sipforum-discussion@lists.su.se'" <sipforum-discussion@lists.su.se>,
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Subject: [Sipping] RE: [Sipforum-discussion] SIP Telephony Device Requirements,
 Conf	iguration and Data
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Henry, 

below are my comments to the Somefolks draft. 



Integrated Switch/Hub SIP devices designed for wired Ethernet SHOULD have an
uplink port such that another IP device, such as a personal computer, MAY
share the network connection. SIP clients MUST prioritize the transmission
of RTP traffic over the shared network connection. 

comment 1: 
Wish you would remove the Hub reference since this is a shared collision
domain concept which I do not recommend for sip client that has real time
requirements sharing this shared Ethernet environment with a host device
that may also have real time data on the same hub. Switch is the only way to
go since ethernet HUB collisions between 2 devices can effect throughput

"IP Requirements IP-1: SIP telephony devices MUST be able to acquire an IP
address by: Automatic IP address configuration using DHCP, or Manual IP
address entry from the device. "

Comment 2 
Why did you not reference RFC3361 DHCP option for sip
ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt this a great MAY RFC to
support for sip clients to learn SIP proxy servers

Comment 3 
No mention was made of support for Visual and Hearing Impaired Special
Equipment or even RFC3351 ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt

Comment 4 
You mention management but only for provisioning, why can we not have SNMP
v3 support for network LAN statistics, SIP traffic stats or even RTCP info
on the SIP phone.






> -----Original Message-----
> From: sipforum-discussion-admin@lists.su.se 
> [mailto:sipforum-discussion-admin@lists.su.se] On Behalf Of 
> Henry Sinnreich
> Sent: Monday, February 10, 2003 8:49 AM
> To: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> Subject: [Sipforum-discussion] SIP Telephony Device 
> Requirements, Configuration and Data
> 
> 
> For everyone interested SIP phones and other SIP 
> communication devices, and the 56 IETF  coming next month, 
> please see if you like or dislike or don't care about the draft:
> 
http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt

The idea is to have discussions on the lists, and if there is interest
and rough consensus to move ahead with an informational RFC. Would
appreciate your comments to 

sipping@ietf.org , and also to

sipforum-discussion@lists.su.se 

Thanks, Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA

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Date: Thu, 13 Feb 2003 09:27:41 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
In-reply-to: <4E4F4B7EF2A53842971777670C0E513C01C88869@DGEXCH05.wcomnet.com>
To: "'Brierre, Jean Joseph'" <Jean.Joseph.Brierre@wcom.com>,
        sipforum-discussion@lists.su.se, sipping@ietf.org
Message-id: <000001c2d374$7367e150$02212aa6@hsinnreich2>
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 Conf	iguration and Data
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Comments 1, 2, and 3 seem obvious - will edit in the next version.
Thanks for pointing out.

As for comment 4: "why can we not have SNMP v3 support for network LAN
statistics, SIP traffic stats or even RTCP info on the SIP phone", 

 it would be interesting to see if SIP phone manufacturers think this is
OK for any type of SIP device, or only for certain devices, such as SIP
telephony adaptors that support key sets. I believe it makes sense only
for latter. Opinions?

Thanks, Henry

> -----Original Message-----
> From: Brierre, Jean Joseph [mailto:Jean.Joseph.Brierre@wcom.com] 
> Sent: Thursday, February 13, 2003 5:43 AM
> To: Sinnreich, Henry; 'sipforum-discussion@lists.su.se'; 
> 'sipping@ietf.org'
> Subject: RE: [Sipforum-discussion] SIP Telephony Device 
> Requirements, Conf iguration and Data
> 
> 
> Henry, 
> 
> below are my comments to the Somefolks draft. 
> 
> 
> 
> Integrated Switch/Hub SIP devices designed for wired Ethernet 
> SHOULD have an uplink port such that another IP device, such 
> as a personal computer, MAY share the network connection. SIP 
> clients MUST prioritize the transmission of RTP traffic over 
> the shared network connection. 
> 
> comment 1: 
> Wish you would remove the Hub reference since this is a 
> shared collision domain concept which I do not recommend for 
> sip client that has real time requirements sharing this 
> shared Ethernet environment with a host device that may also 
> have real time data on the same hub. Switch is the only way 
> to go since ethernet HUB collisions between 2 devices can 
> effect throughput
> 
> "IP Requirements IP-1: SIP telephony devices MUST be able to 
> acquire an IP address by: Automatic IP address configuration 
> using DHCP, or Manual IP address entry from the device. "
> 
> Comment 2 
> Why did you not reference RFC3361 DHCP option for sip 
> ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt this a great 
> MAY RFC to support for sip clients to learn SIP proxy servers
> 
> Comment 3 
> No mention was made of support for Visual and Hearing 
> Impaired Special Equipment or even RFC3351 
ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt

Comment 4 
You mention management but only for provisioning, why can we not have
SNMP v3 support for network LAN statistics, SIP traffic stats or even
RTCP info on the SIP phone.






> -----Original Message-----
> From: sipforum-discussion-admin@lists.su.se
> [mailto:sipforum-discussion-admin@lists.su.se] On Behalf Of 
> Henry Sinnreich
> Sent: Monday, February 10, 2003 8:49 AM
> To: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> Subject: [Sipforum-discussion] SIP Telephony Device 
> Requirements, Configuration and Data
> 
> 
> For everyone interested SIP phones and other SIP
> communication devices, and the 56 IETF  coming next month, 
> please see if you like or dislike or don't care about the draft:
> 
http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt

The idea is to have discussions on the lists, and if there is interest
and rough consensus to move ahead with an informational RFC. Would
appreciate your comments to 

sipping@ietf.org , and also to

sipforum-discussion@lists.su.se 

Thanks, Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Thu Feb 13 10:39:01 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>,
        "'Brierre, Jean Joseph'"
	 <Jean.Joseph.Brierre@wcom.com>,
        sipforum-discussion@lists.su.se, sipping@ietf.org
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Subject: [Sipping] RE: [Sipforum-discussion] SIP Telephony Device Requirements, Conf
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I think it makes sense for any network device.  I'm not sure how
far I would go, but our network management systems want to
know the devices that are out there, and have the basic
info any device gives you.

Certainly we've been asked by customers for tools that network
management can use to determine what is happening when users
say "I got a call, but I couldn't hear them".  We've offered
syslogs and they got amused.  

I'll admit that my "sipphone" engineers think this is silly.
My customers beg to differ.

I think the only thing there is to discuss is:
	How much info is available
	Are there alarms/traps

And, of course, it MUST be SNMPv3

Brian

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: Thursday, February 13, 2003 10:28 AM
> To: 'Brierre, Jean Joseph'; sipforum-discussion@lists.su.se;
> sipping@ietf.org
> Subject: RE: [Sipforum-discussion] SIP Telephony Device Requirements,
> Conf iguration and Data
> 
> 
> Comments 1, 2, and 3 seem obvious - will edit in the next version.
> Thanks for pointing out.
> 
> As for comment 4: "why can we not have SNMP v3 support for network LAN
> statistics, SIP traffic stats or even RTCP info on the SIP phone", 
> 
>  it would be interesting to see if SIP phone manufacturers 
> think this is
> OK for any type of SIP device, or only for certain devices, 
> such as SIP
> telephony adaptors that support key sets. I believe it makes 
> sense only
> for latter. Opinions?
> 
> Thanks, Henry
> 
> > -----Original Message-----
> > From: Brierre, Jean Joseph [mailto:Jean.Joseph.Brierre@wcom.com] 
> > Sent: Thursday, February 13, 2003 5:43 AM
> > To: Sinnreich, Henry; 'sipforum-discussion@lists.su.se'; 
> > 'sipping@ietf.org'
> > Subject: RE: [Sipforum-discussion] SIP Telephony Device 
> > Requirements, Conf iguration and Data
> > 
> > 
> > Henry, 
> > 
> > below are my comments to the Somefolks draft. 
> > 
> > 
> > 
> > Integrated Switch/Hub SIP devices designed for wired Ethernet 
> > SHOULD have an uplink port such that another IP device, such 
> > as a personal computer, MAY share the network connection. SIP 
> > clients MUST prioritize the transmission of RTP traffic over 
> > the shared network connection. 
> > 
> > comment 1: 
> > Wish you would remove the Hub reference since this is a 
> > shared collision domain concept which I do not recommend for 
> > sip client that has real time requirements sharing this 
> > shared Ethernet environment with a host device that may also 
> > have real time data on the same hub. Switch is the only way 
> > to go since ethernet HUB collisions between 2 devices can 
> > effect throughput
> > 
> > "IP Requirements IP-1: SIP telephony devices MUST be able to 
> > acquire an IP address by: Automatic IP address configuration 
> > using DHCP, or Manual IP address entry from the device. "
> > 
> > Comment 2 
> > Why did you not reference RFC3361 DHCP option for sip 
> > ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt this a great 
> > MAY RFC to support for sip clients to learn SIP proxy servers
> > 
> > Comment 3 
> > No mention was made of support for Visual and Hearing 
> > Impaired Special Equipment or even RFC3351 
> ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt
> 
> Comment 4 
> You mention management but only for provisioning, why can we not have
> SNMP v3 support for network LAN statistics, SIP traffic stats or even
> RTCP info on the SIP phone.
> 
> 
> 
> 
> 
> 
> > -----Original Message-----
> > From: sipforum-discussion-admin@lists.su.se
> > [mailto:sipforum-discussion-admin@lists.su.se] On Behalf Of 
> > Henry Sinnreich
> > Sent: Monday, February 10, 2003 8:49 AM
> > To: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> > Subject: [Sipforum-discussion] SIP Telephony Device 
> > Requirements, Configuration and Data
> > 
> > 
> > For everyone interested SIP phones and other SIP
> > communication devices, and the 56 IETF  coming next month, 
> > please see if you like or dislike or don't care about the draft:
> > 
> http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt
> 
> The idea is to have discussions on the lists, and if there is interest
> and rough consensus to move ahead with an informational RFC. Would
> appreciate your comments to 
> 
> sipping@ietf.org , and also to
> 
> sipforum-discussion@lists.su.se 
> 
> Thanks, Henry
> 
> Henry Sinnreich
> WorldCom
> 400 International Parkway
> Richardson, Texas 75081
> USA
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: URI Leasing in the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-rosenberg-sipping-lease-00.txt
	Pages		: 23
	Date		: 2003-2-12
	
Several applications of the Session Initiation Protocol (SIP) require
a user agent (UA) to construct and distribute a URI which can be used
by anyone on the Internet to route a call to that specific UA
instance.  An example of such an application is call transfer, based
on the REFER method.  To date, a variety of techniques have been
proposed for the contruction of such URI.  All of them have fatal
drawbacks.  This document proposes that the problem is actually one
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creation of an address-of-record (AOR) within a domain.  Requirements
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: A Framework for Conferencing with the Session 
                            Initiation Protocol
	Author(s)	: J. Rosenberg
	Filename	: draft-rosenberg-sipping-conferencing-framework-01.txt
	Pages		: 39
	Date		: 2003-2-12
	
The Session Initiation Protocol (SIP) supports the initiation,
modification, and termination of media sessions between user agents.
These sessions are managed by SIP dialogs, which represent a SIP
relationship between a pair of user agents. Because dialogs are
between pairs of user agents, SIP's usage for two-party
communications (such as a phone call), is obvious. Communications
sessions with multiple participants, generally known as conferencing,
are more complicated. This document defines a framework for how such
conferencing can occur. This framework describes the overall
architecture, terminology, and protocol components needed for multi-
party conferencing.

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http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-conferencing-framework-01.txt

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Session Initiation Protocol Call Control - Transfer
	Author(s)	: R. Sparks, A. Johnston
	Filename	: draft-ietf-sipping-cc-transfer-01.txt
	Pages		: 32
	Date		: 2003-2-12
	
This document describes providing Call Transfer capabilities in the
Session Initiation Protocol (SIP).  This work is part of the SIP
Multiparty Call Control Framework.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-01.txt

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From mailnull@www1.ietf.org  Thu Feb 13 13:37:35 2003
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Date: Thu, 13 Feb 2003 12:28:20 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-cc-transfer-01.txt
In-reply-to: <200302131747.MAA19487@ietf.org>
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All,

The major changes from the previous draft are the addition of:

  - Section on use of Referred-By header.
  - Selected message details.
  - Flow for attended transfer with non-globally routable Contact URI
  - Flow for attended transfer fallback to unattended transfer.
  - Security Considerations Section.

Otherwise, the draft is quite stable and has no open issues or To Do items 
that the authors are aware of.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

At 12:47 PM 2/13/2003 -0500, Internet-Drafts@ietf.org wrote:
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>This draft is a work item of the Session Initiation Proposal Investigation 
>Working Group of the IETF.
>
>         Title           : Session Initiation Protocol Call Control - Transfer
>         Author(s)       : R. Sparks, A. Johnston
>         Filename        : draft-ietf-sipping-cc-transfer-01.txt
>         Pages           : 32
>         Date            : 2003-2-12
>
>This document describes providing Call Transfer capabilities in the
>Session Initiation Protocol (SIP).  This work is part of the SIP
>Multiparty Call Control Framework.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-01.txt
>
>To remove yourself from the IETF Announcement list, send a message to
>ietf-announce-request with the word unsubscribe in the body of the message.
>
>Internet-Drafts are also available by anonymous FTP. Login with the username
>"anonymous" and a password of your e-mail address. After logging in,
>type "cd internet-drafts" and then
>         "get draft-ietf-sipping-cc-transfer-01.txt".
>
>A list of Internet-Drafts directories can be found in
>http://www.ietf.org/shadow.html
>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
>Internet-Drafts can also be obtained by e-mail.
>
>Send a message to:
>         mailserv@ietf.org.
>In the body type:
>         "FILE /internet-drafts/draft-ietf-sipping-cc-transfer-01.txt".
>
>NOTE:   The mail server at ietf.org can return the document in
>         MIME-encoded form by using the "mpack" utility.  To use this
>         feature, insert the command "ENCODING mime" before the "FILE"
>         command.  To decode the response(s), you will need "munpack" or
>         a MIME-compliant mail reader.  Different MIME-compliant mail readers
>         exhibit different behavior, especially when dealing with
>         "multipart" MIME messages (i.e. documents which have been split
>         up into multiple messages), so check your local documentation on
>         how to manipulate these messages.
>
>
>Below is the data which will enable a MIME compliant mail reader
>implementation to automatically retrieve the ASCII version of the
>Internet-Draft.
>Content-Type: text/plain
>Content-ID:     <2003-2-13130639.I-D@ietf.org>
>
>ENCODING mime
>FILE /internet-drafts/draft-ietf-sipping-cc-transfer-01.txt
>
><ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-01.txt>

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From mailnull@www1.ietf.org  Thu Feb 13 14:25:47 2003
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        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'" <sipping@ietf.org>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Cc: "'John Loughney'" <john.loughney@nokia.com>
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At 03:19 AM 2/11/2003, Henry Sinnreich wrote:
>> We will now try to get the work on the diameter-based solution going.
>
>As has been mentioned here, 99.999% (five nines :-0>) of existing
>implementations and deployment are RADIUS. After RADIUS come the web
>services. 
>
>I suggest therefore to start with RADIUS instead.

I'm sharing the same deployment experience and second the same plan: RADIUS.

-Jiri 

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        <Gonzalo.Camarillo@lmf.ericsson.se>, <sipping@ietf.org>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
>Hi Henry,
>
>> >but one option is develop a solution using 
>> > Diameter in RADIUS-compatibility mode.
>> 
>> OK, this is better than nothing. We'll take what we can :-)
>> 
>> As mentioned, the installed base is RADIUS, no matter how 
>> voluminous the DIAMETER literature may be.
>
>I think the issues is that the IETF should recommend what is the
>right thing to do & why.  If people choose to do something else,
>well, that is life.  

Remember that running code does not necessarily mean something known
to compile -- that is something known to be deployed.

>I understand what you are asking but I do note that some
>people have thought that the potential for AAA is much greater
>than the existing installed base.

I actually think that the deployment argument is a very important one.

-Jiri 

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Date: Thu, 13 Feb 2003 14:27:27 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
In-reply-to: <NFBBJHOFMDBFILNPMNGKOENADLAA.gunnar.hellstrom@omnitor.se>
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        "=?iso-8859-1?Q?'J=F6rgen_Bj=F6rkner'?=" <jorgen.bjorkner@hotsip.com>,
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Subject: [Sipping] RE: [Sipforum-discussion] SIP Telephony Device Requirements,
 Configuration and Data
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Gunnar Hellström wrote:

> SIP-xx, analogue text support:

> SIP-zz, digital text support:

> In 4.11 Ringer Behaviour 

> Add to References
> -----------------
> [y] G. Hellstrom: "RTP Payload for Text Conversation.", RFC 
> 2793, IETF, May 2000. [z] ITU-T Recommendation V.18 

Are all excellent points that we will add. Thanks for the comments!

Henry

> -----Original Message-----
> From: Gunnar Hellström [mailto:gunnar.hellstrom@omnitor.se] 
> Sent: Thursday, February 13, 2003 2:22 PM
> To: Henry Sinnreich; 'Jörgen Björkner'; 
> sipforum-discussion@lists.su.se
> Cc: sipping@ietf.org
> Subject: Re: [Sipforum-discussion] SIP Telephony Device 
> Requirements, Configuration and Data
> 
> 
> Henry,
> 
> It is mentioned in the introduction to the document that 
> "text shall be kept in mind".
> 
> I suggest some specific text items to be entered that are in 
> line with existing standardisation, and accessibility needs 
> and useful for all:
> 
> --------------------------------------------------------------
> --------------
> -------------------
> SIP-xx, analogue text support: SIP adapter devices (for 
> analog phone lines)
>   MUST support conversion between real time text transmission 
> using RFC 2793 [y]
>   and text telephones according to ITU-T V.18 [x] allowing 
> alternating use of
>   text and voice. SIP clients MUST fallback to G.711 if 
> RFC2793 connection fails.
> 
> 
> SIP-zz, digital text support: SIP telephone devices MUST support
>   real time text conversation using RFC 2793 [y] for the text stream.
>   It MUST be possible to use text simultaneously with voice.
> 
> 
> 
> ---------------------------------------------
> In 4.11 Ringer Behaviour  : add the section below for the 
> purpose of deaf, hearing impaired, deaf-blind people and 
> people in the garden and noisy
> locations:
> 
> SIP Telephony devices MUST provide an external interface to
> an external alerting system.
> 
> 
> 
> Add to References
> -----------------
> [y] G. Hellstrom: "RTP Payload for Text Conversation.", RFC 
> 2793, IETF, May 2000. [z] ITU-T Recommendation V.18 
> "Operational and interworking requirements for DCEs operating 
> in the text telephone mode", November 2000 with amendment 
> November 2002.
> 
> --------------------------------------------------------------
> --------------
> -------
> 
> 
> 
> 
> Thanks
> -------------------------------------------
> Gunnar Hellström
> Omnitor AB
> Renathvägen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> > -----Ursprungligt meddelande-----
> > Från: sipforum-discussion-admin@lists.su.se
> > [mailto:sipforum-discussion-admin@lists.su.se]För Henry Sinnreich
> > Skickat: den 10 februari 2003 15:49
> > Till: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> > Ämne: [Sipforum-discussion] SIP Telephony Device Requirements, 
> > Configuration and Data
> >
> >
> > For everyone interested SIP phones and other SIP communication 
> > devices, and the 56 IETF  coming next month, please see if 
> you like or 
> > dislike or don't care about the draft:
> >
> > 
> http://www.ietf.org/internet->
drafts/draft-sinnreich-sipdev-req-00.txt
> >
> > The idea is to have discussions on the lists, and if there 
> is interest 
> > and rough consensus to move ahead with an informational RFC. Would 
> > appreciate your comments to
> >
> > sipping@ietf.org , and also to
> >
> > sipforum-discussion@lists.su.se
> >
> > Thanks, Henry
> >
> > Henry Sinnreich
> > WorldCom
> > 400 International Parkway
> > Richardson, Texas 75081
> > USA
> >
> >
> >
> 
> 
> 

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Alan Johnston wrote:
> Just to clarify and expand on my earlier short comments...
> 
> I'm not unhappy that the solution proposed in the draft  is similar to 
> the one discussed in cc-transfer.   The method does work,  and we can  
> discuss whether it should be two REGISTERs or one LEASE and one 
> REGISTER.  However, both approaches are multi-step, which others 
> objected to when it was discussed at IETF-55.

I don't see a problem in adding steps if its done asynchronously with 
calls. That is, with the mechanism I discuss, you would be allocated a 
namespace from which you can choose URI. THerefore, for each call, there 
is zero additional protocol operations needed.

> 
> I had hoped that a new proposal was coming, or a way to make Rohan's 
> escaped Route header in a Contact work.

I'm not sure whether you are trying to imply that there is nothign new 
in my draft?

> This approach, (4.4 in this 
> draft) does not actually suffer from the problems listed - if multiple 
> proxies must be traversed, multiple URIs can be escaped into the Route 
> header. 

How does a UA learn the route headers to include? If you say that they 
are taken from the route set of the dialog, then how does a UA know 
which ones need to be included? How does a UA know whether a different 
caller is supposed to use that same set of proxies? The route set from 
the dialog is also purposefully scoped to that dialog, so that proxies 
can include information in it that is related to the dialog. So, would a 
UA discard that user part? The URI might not work without that. A smart 
proxy that embeds dialog information in the user part, so that it can 
reject calls with route headers it never generated, would cause that to 
fail.





> The real problem with this approach is that it violates Table 1 
> in RFC 3261 so would be ignored by all RFC  3261 compliant UAs. 

There are some additional problems, as above.


> The 
> backwards compatibility problem with pre-bis-04 UAs is even worse.  
> Since under the old Record-Route rules the Contact is included in the 
> Route header, you could potentially have a URI in a Route header with  
> an escaped Route header - yuk!  However, there might be a clever 
> solution to work around this problem.  This is the only solution that I 
> know of that meets REQ-8.

No, thats not true. The mechanism I proposed, if the lease and register 
operations are combined, can be made to work without network state.


> 
> REQ-9 is a bit strange - the anonymity of a Contact is a very small part 
> of the total anonymity  picture. 

We have already discussed, in the context of rfc3323, several different 
notions of what encompasses privacy. This requirement is meant to rule 
out solutions that, for example, would require the user's AOR to be used.


   It seems this requirement could
> simply be met by having the user portion  of the Contact be a random 
> string. 

Yes.

  The privacy discussion in 4.2 about the call center operator is
> simplistic - for privacy, much more than just fixing Contact is required 
> and would be used in practice.

As I mentioned, there are various levels of privacy that we have 
recognized as useful.

> 
> I'm not sure understand the motivation for REQ-11.  I don't know why a 
> UA would want to generate a unique Contact for each dialog - it seems 
> like quite a burden on the UA and a difficult requirement for a solution 
> to meet.  It seems like there is some sort of security  reason for this, 
> but it is not stated.

This is what I had in mind. You and I are talking. I want to transfer 
you to Robert. You don't want anyone to generally be able to call your 
device - they should go through your normal SIP AOR that identifies you 
as a user. However, for the transfer to work, I need a URI for your 
device. If you could create those contact URIs, you could create one 
that contains a hash of the dialog ID with me, or a timestamp embedded. 
This way, if you should receive an INVITE to that URI, you could check 
it to make sure it still valid. If not, you reject the INVITE.

> 
> One minor correction - the transfer problem is with  attended transfer 
> only, not blind transfer or transfer in general.

Yes, that is right. My error.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Session Initiation Protocol Call Control -  
                          Conferencing for User Agents
	Author(s)	: A. Johnston, O. Levin
	Filename	: draft-johnston-sipping-cc-conferencing-01.txt
	Pages		: 24
	Date		: 2003-2-11
	
This document defines conferencing call control features for the
Session Initiation Protocol (SIP).  This document builds on the
Conferencing Requirements and Framework documents to define how a
tightly coupled SIP conference works.  The approach is explored from
different user agent (UA) types perspective: conference-unaware,
conference-aware and focus UAs.  The use of URIs in conferencing,
OPTIONS for capabilities discovery, and call control using REFER are
covered in detail with example call flow diagrams.

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From: William Marshall <wtm@research.att.com>
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To: sipping@ietf.org
Subject: Re: [Sipping] Expert review for draft-burger-sipping-netann-04.txt
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There is one issue in dealing with announcements and announcement
servers that doesn't arise with normal sessions - the timing of the
session termination.

In a normal media session, the media is being generated in real time
and transmitted via RTP to the UAC.  When the source of the media
stops, the session is terminated with a BYE, and the UAC closes
both the signaling and media paths.  While there is potential for
a race, it is not significant.

However, when the media is an announcement, contained likely within
a single (possibly large) file, it is conceivable that the server
will send the (large) RTP packets to the UAC, and then send a BYE
while the UAC has most of the announcement still unplayed.

The basic architectural question is whether it is the responsibility
of the UAC to drain any pending media before closing a session.  I
do not know of any such requirement in RFC3261.  Is it considered
a common implementation practice to do so?

Alternatively, it could be the responsibility of the announcement
server to delay sending the BYE (or 4xx response) based on the 
expected playout time, though it is not clear how that would be
calculated.

One choice or the other needs to be stated, and required, in order
to achieve interoperability.

I also have concerns about the use of 487 to terminate early media
sessions.  This has been raised by others.

Bill Marshall
wtm@research.att.com

-----original message-----
Date: Mon, 10 Feb 2003 12:39:56 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
To: sipping <sipping@ietf.org>
Subject: [Sipping] Expert review for draft-burger-sipping-netann-04.txt

Folks,

We would like the SIPPING working group to provide expert review of:

http://www.ietf.org/internet-drafts/draft-burger-sipping-netann-04.txt

The authors have requested individual publication of this document. 
Expert Review will close on March 10th, 2003

Thanks,

Gonzalo
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Date: Fri, 14 Feb 2003 16:33:38 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sipping] New I-D on URI leasing
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At 02:54 AM 2/14/2003 -0500, Jonathan Rosenberg wrote:


>Alan Johnston wrote:
>>Just to clarify and expand on my earlier short comments...
>>I'm not unhappy that the solution proposed in the draft  is similar to 
>>the one discussed in cc-transfer.   The method does work,  and we can
>>discuss whether it should be two REGISTERs or one LEASE and one 
>>REGISTER.  However, both approaches are multi-step, which others objected 
>>to when it was discussed at IETF-55.
>
>I don't see a problem in adding steps if its done asynchronously with 
>calls. That is, with the mechanism I discuss, you would be allocated a 
>namespace from which you can choose URI. THerefore, for each call, there 
>is zero additional protocol operations needed.

>>I had hoped that a new proposal was coming, or a way to make Rohan's 
>>escaped Route header in a Contact work.
>
>I'm not sure whether you are trying to imply that there is nothign new in 
>my draft?

Of course not.

>>This approach, (4.4 in this draft) does not actually suffer from the 
>>problems listed - if multiple proxies must be traversed, multiple URIs 
>>can be escaped into the Route header.
>
>How does a UA learn the route headers to include? If you say that they are 
>taken from the route set of the dialog, then how does a UA know which ones 
>need to be included? How does a UA know whether a different caller is 
>supposed to use that same set of proxies? The route set from the dialog is 
>also purposefully scoped to that dialog, so that proxies can include 
>information in it that is related to the dialog. So, would a UA discard 
>that user part? The URI might not work without that. A smart proxy that 
>embeds dialog information in the user part, so that it can reject calls 
>with route headers it never generated, would cause that to fail.
>

The Route URI wouldn't be extracted from a dialog, it would have to be 
learned or provisioned similar to a pre-loaded Route URI. The Route header 
used also would be similar to a pre-loaded Route in that it would never 
contain any specific dialog information.

Think of how a normal user's AOR is routed.  Usually, a DNS SRV lookup (or 
similar) is performed on the host name which initially routes the request 
to a set of proxies.  These proxies then take care of all the routing 
between the proxy and the endpoint.  If multiple proxies need to be 
traversed, the proxy first proxy inserts them.   All an embedded Route 
header in a Contact needs to do is to cause essentially the same DNS 
lookups to occur so the request reaches the same initial set of 
proxies.  This another way of stating your requirement that the device URI 
be in the same domain as the AOR URI for the user - they must resolve to a 
similar set of proxies, unless the topology is trivial in which case it 
doesn't matter.




>>The real problem with this approach is that it violates Table 1 in RFC 
>>3261 so would be ignored by all RFC  3261 compliant UAs.
>
>There are some additional problems, as above.
>
>
>>The backwards compatibility problem with pre-bis-04 UAs is even worse.
>>Since under the old Record-Route rules the Contact is included in the 
>>Route header, you could potentially have a URI in a Route header with
>>an escaped Route header - yuk!  However, there might be a clever solution 
>>to work around this problem.  This is the only solution that I know of 
>>that meets REQ-8.
>
>No, thats not true. The mechanism I proposed, if the lease and register 
>operations are combined, can be made to work without network state.

There is still a binding between the device URI template and the device 
itself.  Regardless of whether this is done through a REGISTER or a LEASE, 
it is still state information and is effectively the same as a registration.


>>REQ-9 is a bit strange - the anonymity of a Contact is a very small part 
>>of the total anonymity  picture.
>
>We have already discussed, in the context of rfc3323, several different 
>notions of what encompasses privacy. This requirement is meant to rule out 
>solutions that, for example, would require the user's AOR to be used.

The use of the user's AOR is already ruled out by the most important 
requirement that it resolve uniquely to a device without forking.  By the 
way, that requirement is mentioned in the definition section 2 but not in 
the requirements section 6.  A user's AOR in a Contact would also cause all 
sorts of other problems with respect to forking of re-INVITEs, etc.


>   It seems this requirement could
>>simply be met by having the user portion  of the Contact be a random string.
>
>Yes.
>
>  The privacy discussion in 4.2 about the call center operator is
>>simplistic - for privacy, much more than just fixing Contact is required 
>>and would be used in practice.
>
>As I mentioned, there are various levels of privacy that we have 
>recognized as useful.
>
>>I'm not sure understand the motivation for REQ-11.  I don't know why a UA 
>>would want to generate a unique Contact for each dialog - it seems like 
>>quite a burden on the UA and a difficult requirement for a solution to 
>>meet.  It seems like there is some sort of security  reason for this, but 
>>it is not stated.
>
>This is what I had in mind. You and I are talking. I want to transfer you 
>to Robert. You don't want anyone to generally be able to call your device 
>- they should go through your normal SIP AOR that identifies you as a 
>user. However, for the transfer to work, I need a URI for your device. If 
>you could create those contact URIs, you could create one that contains a 
>hash of the dialog ID with me, or a timestamp embedded. This way, if you 
>should receive an INVITE to that URI, you could check it to make sure it 
>still valid. If not, you reject the INVITE.

Usually, the use of a Contact URI is involves in a dialog.  For example, in 
an attended transfer, there is a Replaces header.  The UA then knows which 
dialog is being replaced - there is no requirement for the Contact to be 
limited to the duration of the dialog - the rejection is based on the 
Replaces header and whether the requester is permitted to replace 
dialogs.  The same thing for a conference - the focus will know if the 
conference is still valid and whether the requester may join based on the 
conference policy for that URI - not based on the URI itself which just 
references the conference instance.

In SIP, we have always said that a Contact URI may not be valid outside the 
dialog.  It has never been a requirement that it never be valid outside a 
dialog.  I'm still not seeing enough justification for this requirement - 
seems like a lot of pain for little gain, but I'd be interested in other's 
opinions.  If this requirement could be met easily, I'd be happy with it, 
but it introduce a whole set of name space requirements on a UA - seems 
like a lot of complexity.

>>One minor correction - the transfer problem is with  attended transfer 
>>only, not blind transfer or transfer in general.
>
>Yes, that is right. My error.
>
>-Jonathan R.
>--
>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

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To: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
cc: john.loughney@nokia.com, Henry.Sinnreich@wcom.com,
        Gonzalo.Camarillo@lmf.ericsson.se, sipping@ietf.org,
        aboba@internaut.com
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft 
In-Reply-To: Message from Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de> 
   of "Thu, 13 Feb 2003 19:56:42 +0100." <5.2.0.9.0.20030213195458.04501ea0@mailhost.fokus.gmd.de> 
Date: Fri, 14 Feb 2003 21:58:40 -0800
From: Allison Mankin <mankin@psg.com>
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Jiri,

It always seems good to be able to stay with installed base, but
there's a factor against our supporting RADIUS drafts for SIP's
accounting - safety.  Even though there is deployment, there are
some extremely bad cases of damage in production, and the IETF 
has concluded that RADIUS is not suitable for usage-based accounting
(RFC2975). 

Unless there is no intention of having in SIP accounting usage-based,
monetary, auditable applications, RADIUS is not the solution.  The
protocol just does not preserve a complete stream of reliable
accounting records that are auditable on any level, especially with
any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
the use of RADIUS may even be illegal in situations where there is an
authority over the billing.  Records are too easily lost by the
protocol. This is stated clearly by RFC2975.  This is the reason why
the Operations and Mgt Area developed Diameter.  On these technical*
grounds, that the RADIUS protocol does not have enough reliability,
the work must be based on the Diameter protocol, which is now
approved.  Alternatively, work could be done that does not include
usage-based accounting, but this does not seem to be what is 
envisioned in the SIPPING AAA requirements.  

Allison

P.S. There's a current investigation of RADIUS compatibility mode
in Diameter for this - more needs to be thought about it.
> 
> At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> >Hi Henry,
> >
> >> >but one option is develop a solution using 
> >> > Diameter in RADIUS-compatibility mode.
> >> 
> >> OK, this is better than nothing. We'll take what we can :-)
> >> 
> >> As mentioned, the installed base is RADIUS, no matter how 
> >> voluminous the DIAMETER literature may be.
> >
> >I think the issues is that the IETF should recommend what is the
> >right thing to do & why.  If people choose to do something else,
> >well, that is life.  
> 
> Remember that running code does not necessarily mean something known
> to compile -- that is something known to be deployed.
> 
> >I understand what you are asking but I do note that some
> >people have thought that the potential for AAA is much greater
> >than the existing installed base.
> 
> I actually think that the deployment argument is a very important one.
> 
> -Jiri 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
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Folks,

the transcoding design team has produced a new version of the draft
below:

http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sip-deaf-02.txt

http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sip-deaf-02.pdf

Comments and suggestions are welcome.

Regards,

Gonzalo
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Date: Mon, 17 Feb 2003 11:00:37 -0500
From: "David R. Oran" <oran@cisco.com>
To: sipping@ietf.org
Subject: Re: [Sipping] I-D ACTION:draft-rosenberg-sipping-lease-00.txt
Message-ID: <269039778.1045479637@[10.32.254.183]>
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> 	Title		: URI Leasing in the Session Initiation Protocol (SIP)
> 	Author(s)	: J. Rosenberg
> 	Filename	: draft-rosenberg-sipping-lease-00.txt
> 	Pages		: 23
> 	Date		: 2003-2-12
> 	
In general I like the direction this is going. I have a query. Did you 
consider making the creation and communication of the GRUU simply a 
side-effect of a REGISTER operation as opposed to a separate method? In 
other words, the REGISTER would optionally request the construction of a 
(range of/template for) GRUUs for the UA, and the response would contain 
the constructed values. This would seem to meet all of the requirements, 
including, at least provisionally, the requirement for not substantially 
increasing the domain-level state stored by proxies.

One small point bears upon the above. In the text you say that a lease 
should not extend beyond the registration expiration of a registered UA. 
However, the requirement is weaker; it only says that the domain has final 
say over the lease duration. Was this intentional? If so I would think this 
complicates the life of a registrar somewhat, compared with managing GRUUs 
just like registrations, or state directly bound to registrations. I can 
see a rather straightforward implementation if leases are simply extra 
stuff associated with a registration. I have a harder time seeing a simple 
implementation if a new method or some completely separate mechanism is 
used.

Dave.


Dave.
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Date: Mon, 17 Feb 2003 10:07:46 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <E18jvLW-000HAk-00@psg.com>
To: "'Allison Mankin'" <mankin@psg.com>,
        "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>
Cc: john.loughney@nokia.com, Gonzalo.Camarillo@lmf.ericsson.se,
        sipping@ietf.org, aboba@internaut.com,
        Henning Schulzrinne <hgs@cs.columbia.edu>,
        "'John R. Vollbrecht'" <jrv@interlinknetworks.com>,
        "'Cees deLaat'" <C.T.A.M.deLaat@phys.uu.nl>,
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Why not get help from the aaaarch RG on what the next generation of AAA
should be?

I have taken the liberty to copy here John Vollbrecht and Cees deLat to
see if some consultation with the aaarch folks is doable during the
March IETF. The aaaarch charter addresses exactly the dilemma we are
discussing here: The NG AAA.

http://www.aaaarch.org/charter.html

Note: RADIUS may not be perfect but is used for 100's of millions of
Internet dial-up users all over the world, so maybe it is just good
enough -  valid IETF principle if I remember well.

This is the reason, several of us like the I-D on SIP and RADIUS by
Henning Schulzrinne. It is for real and I suggest it should be advanced
to an informational RFC. See also

http://www.cs.columbia.edu/~hgs/sip/drafts/draft-byerly-sip-radius-00.tx
t

http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sterman-sip-radius-00.t
xt

http://www.cs.columbia.edu/~hgs/research/IRT/cinema/release/radius_notes
.html

But let's see what we can learn from aaaarch.

What do you all think?

Thanks, Henry


> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> On Behalf Of Allison Mankin
> Sent: Friday, February 14, 2003 11:59 PM
> To: Jiri Kuthan
> Cc: john.loughney@nokia.com; Henry.Sinnreich@wcom.com; 
> Gonzalo.Camarillo@lmf.ericsson.se; sipping@ietf.org; 
> aboba@internaut.com
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> 
> Jiri,
> 
> It always seems good to be able to stay with installed base,
> but there's a factor against our supporting RADIUS drafts for 
> SIP's accounting - safety.  Even though there is deployment, 
> there are some extremely bad cases of damage in production, 
> and the IETF 
> has concluded that RADIUS is not suitable for usage-based 
> accounting (RFC2975). 
> 
> Unless there is no intention of having in SIP accounting
> usage-based, monetary, auditable applications, RADIUS is not 
> the solution.  The protocol just does not preserve a complete 
> stream of reliable accounting records that are auditable on 
> any level, especially with any roaming.  As Bernard Aboba 
> stated to this WG, quoting Mike O'Dell, the use of RADIUS may 
> even be illegal in situations where there is an authority 
> over the billing.  Records are too easily lost by the 
> protocol. This is stated clearly by RFC2975.  This is the 
> reason why the Operations and Mgt Area developed Diameter.  
> On these technical* grounds, that the RADIUS protocol does 
> not have enough reliability, the work must be based on the 
> Diameter protocol, which is now approved.  Alternatively, 
> work could be done that does not include usage-based 
> accounting, but this does not seem to be what is 
> envisioned in the SIPPING AAA requirements.  
> 
> Allison
> 
> P.S. There's a current investigation of RADIUS compatibility
> mode in Diameter for this - more needs to be thought about it.
> > 
> > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > >Hi Henry,
> > >
> > >> >but one option is develop a solution using
> > >> > Diameter in RADIUS-compatibility mode.
> > >> 
> > >> OK, this is better than nothing. We'll take what we can :-)
> > >> 
> > >> As mentioned, the installed base is RADIUS, no matter how 
> > >> voluminous the DIAMETER literature may be.
> > >
> > >I think the issues is that the IETF should recommend what is the
> > >right thing to do & why.  If people choose to do something else, 
> > >well, that is life.
> > 
> > Remember that running code does not necessarily mean
> something known
> > to compile -- that is something known to be deployed.
> > 
> > >I understand what you are asking but I do note that some
> people have
> > >thought that the potential for AAA is much greater than
> the existing
> > >installed base.
> > 
> > I actually think that the deployment argument is a very
> important one.
> > 
> > -Jiri
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sip@ietf.org for new developments of core SIP
> 

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Date: Mon, 17 Feb 2003 14:23:15 -0500
From: "David R. Oran" <oran@cisco.com>
To: William Marshall <wtm@research.att.com>, sipping@ietf.org
Subject: Re: [Sipping] Expert review for draft-burger-sipping-netann-04.txt
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--On Friday, February 14, 2003 10:41 AM -0500 William Marshall 
<wtm@research.att.com> wrote:

> There is one issue in dealing with announcements and announcement
> servers that doesn't arise with normal sessions - the timing of the
> session termination.
>
> In a normal media session, the media is being generated in real time
> and transmitted via RTP to the UAC.  When the source of the media
> stops, the session is terminated with a BYE, and the UAC closes
> both the signaling and media paths.  While there is potential for
> a race, it is not significant.
>
> However, when the media is an announcement, contained likely within
> a single (possibly large) file, it is conceivable that the server
> will send the (large) RTP packets to the UAC, and then send a BYE
> while the UAC has most of the announcement still unplayed.
>
> The basic architectural question is whether it is the responsibility
> of the UAC to drain any pending media before closing a session.  I
> do not know of any such requirement in RFC3261.  Is it considered
> a common implementation practice to do so?
>
In the real world, don't announcment servers just replay the message over 
and over until the user hangs up? Or, if they don't hang up for a longish 
time, they just time out and send a BYE to conserve resources or avoid the 
"John Clayton memorial PBX bug".

> Alternatively, it could be the responsibility of the announcement
> server to delay sending the BYE (or 4xx response) based on the
> expected playout time, though it is not clear how that would be
> calculated.
>
Well, you could be conservative.

> One choice or the other needs to be stated, and required, in order
> to achieve interoperability.
>
> I also have concerns about the use of 487 to terminate early media
> sessions.  This has been raised by others.
>
> Bill Marshall
> wtm@research.att.com
>
> -----original message-----
> Date: Mon, 10 Feb 2003 12:39:56 +0200
> From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
> To: sipping <sipping@ietf.org>
> Subject: [Sipping] Expert review for draft-burger-sipping-netann-04.txt
>
> Folks,
>
> We would like the SIPPING working group to provide expert review of:
>
> http://www.ietf.org/internet-drafts/draft-burger-sipping-netann-04.txt
>
> The authors have requested individual publication of this document.
> Expert Review will close on March 10th, 2003
>
> Thanks,
>
> Gonzalo
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
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Date: Tue, 18 Feb 2003 20:50:04 +0100
To: Allison Mankin <mankin@psg.com>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft 
Cc: john.loughney@nokia.com, Henry.Sinnreich@wcom.com,
        Gonzalo.Camarillo@lmf.ericsson.se, sipping@ietf.org,
        aboba@internaut.com
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Allison,

I am no way opposed to the critique of RADIUS -- but where is the
DIAMETER running code?

-jiri

At 06:58 AM 2/15/2003, Allison Mankin wrote:

>Jiri,
>
>It always seems good to be able to stay with installed base, but
>there's a factor against our supporting RADIUS drafts for SIP's
>accounting - safety.  Even though there is deployment, there are
>some extremely bad cases of damage in production, and the IETF 
>has concluded that RADIUS is not suitable for usage-based accounting
>(RFC2975). 
>
>Unless there is no intention of having in SIP accounting usage-based,
>monetary, auditable applications, RADIUS is not the solution.  The
>protocol just does not preserve a complete stream of reliable
>accounting records that are auditable on any level, especially with
>any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
>the use of RADIUS may even be illegal in situations where there is an
>authority over the billing.  Records are too easily lost by the
>protocol. This is stated clearly by RFC2975.  This is the reason why
>the Operations and Mgt Area developed Diameter.  On these technical*
>grounds, that the RADIUS protocol does not have enough reliability,
>the work must be based on the Diameter protocol, which is now
>approved.  Alternatively, work could be done that does not include
>usage-based accounting, but this does not seem to be what is 
>envisioned in the SIPPING AAA requirements.  
>
>Allison
>
>P.S. There's a current investigation of RADIUS compatibility mode
>in Diameter for this - more needs to be thought about it.
>> 
>> At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
>> >Hi Henry,
>> >
>> >> >but one option is develop a solution using 
>> >> > Diameter in RADIUS-compatibility mode.
>> >> 
>> >> OK, this is better than nothing. We'll take what we can :-)
>> >> 
>> >> As mentioned, the installed base is RADIUS, no matter how 
>> >> voluminous the DIAMETER literature may be.
>> >
>> >I think the issues is that the IETF should recommend what is the
>> >right thing to do & why.  If people choose to do something else,
>> >well, that is life.  
>> 
>> Remember that running code does not necessarily mean something known
>> to compile -- that is something known to be deployed.
>> 
>> >I understand what you are asking but I do note that some
>> >people have thought that the potential for AAA is much greater
>> >than the existing installed base.
>> 
>> I actually think that the deployment argument is a very important one.
>> 
>> -Jiri 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP 

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Henry Sinnreich writes:

 > I suggest therefore to start with RADIUS instead.

i fully agree, juha

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Date: Wed, 19 Feb 2003 07:47:47 +0200
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        <sipping@ietf.org>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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john.loughney@nokia.com writes:

 > RADIUS is extremely limited in a number of cases - security, transport
 > issues, reliability and is most likely illegal in many jurisdictions
 > for use with token-based accounting (due to the non-reliable nature
 > of RADIUS).  

how does that differ from sip itself?  one can always run radius over
ipsec.

-- juha

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Date: Wed, 19 Feb 2003 00:03:36 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <15955.6047.468360.246276@harjus.eng.song.fi>
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        "'sipping'" <sipping@ietf.org>,
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I still would like to hear if people think a discussion with the AAA
architecture RG is a good way to get out of this impasse. Please browse
the work at http://www.aaaarch.org/charter.html

It appears there is enough push-back on DIAMETER in this WG that this
should be given some time for discussion at the next meeting. All
messages from the 'audience' favored RADIUS, contrary to the proposed
work agenda.

I have yet to hear an argument why some form of XML documents over some
secure transport, as in Web Services should not be the choice after
RADIUS.

Or are the authors so committed to a DIAMETER project that this
discussion is futile?

Thanks, Henry



> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi] 
> Sent: Tuesday, February 18, 2003 11:35 PM
> To: Henry Sinnreich
> Cc: 'Gonzalo Camarillo'; 'sipping'; 'John Loughney'
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Henry Sinnreich writes:
> 
>  > I suggest therefore to start with RADIUS instead.
> 
> i fully agree, juha
> 

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From mailnull@www1.ietf.org  Wed Feb 19 03:40:23 2003
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is there any open source diameter server implementations similar to
freeradius?  

is there a good number of commercial diameter server implementations for
me to choose from?

is there any open source diameter client libraries similar to
radiusclient?

is there any commercial diameter client libraries for me to choose from?

i don't mind using diameter for sip once answers to most of the above
questions is "yes".  as long it isn't, it makes no sense to base sip aaa
work on diameter.

-- juha

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Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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Hi Juha,

Isn't this a chicken-and-egg problem? Diameter just got to proposed standard - back in 1999 if you had asked the same set of questions about RFC 2543, I bet we'd all be talking about H.323 or something... ;)

Cheers,
Aki

 > -----Original Message-----
 > From: ext jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
 > Sent: 19 February, 2003 10:37
 > To: jari.arkko@piuha.net
 > Cc: Henry Sinnreich; 'Gonzalo Camarillo'; 'sipping'; Loughney John
 > (NRC/Helsinki)
 > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
 > 
 > 
 > is there any open source diameter server implementations similar to
 > freeradius?  
 > 
 > is there a good number of commercial diameter server 
 > implementations for
 > me to choose from?
 > 
 > is there any open source diameter client libraries similar to
 > radiusclient?
 > 
 > is there any commercial diameter client libraries for me to 
 > choose from?
 > 
 > i don't mind using diameter for sip once answers to most of the above
 > questions is "yes".  as long it isn't, it makes no sense to 
 > base sip aaa
 > work on diameter.
 > 
 > -- juha
 > 
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 > 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Hi,

I agree with Aki.

Perhaps some answer to Juha question could be found
from the Open source Diameter project (see attached mail).

http://opendiameter.org.

Br
Marco=20



> -----Original Message-----
> From: ext aki.niemi@nokia.com [mailto:aki.niemi@nokia.com]
> Sent: 19 February, 2003 11:35
> To: jh@lohi.eng.song.fi
> Cc: Henry.Sinnreich@wcom.com; Gonzalo.Camarillo@lmf.ericsson.se;
> sipping@ietf.org; Loughney John (NRC/Helsinki); jari.arkko@piuha.net
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
>=20
>=20
> Hi Juha,
>=20
> Isn't this a chicken-and-egg problem? Diameter just got to=20
> proposed standard - back in 1999 if you had asked the same=20
> set of questions about RFC 2543, I bet we'd all be talking=20
> about H.323 or something... ;)
>=20
> Cheers,
> Aki
>=20
>  > -----Original Message-----
>  > From: ext jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
>  > Sent: 19 February, 2003 10:37
>  > To: jari.arkko@piuha.net
>  > Cc: Henry Sinnreich; 'Gonzalo Camarillo'; 'sipping'; Loughney John
>  > (NRC/Helsinki)
>  > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
>  >=20
>  >=20
>  > is there any open source diameter server implementations similar to
>  > freeradius? =20
>  >=20
>  > is there a good number of commercial diameter server=20
>  > implementations for
>  > me to choose from?
>  >=20
>  > is there any open source diameter client libraries similar to
>  > radiusclient?
>  >=20
>  > is there any commercial diameter client libraries for me to=20
>  > choose from?
>  >=20
>  > i don't mind using diameter for sip once answers to most=20
> of the above
>  > questions is "yes".  as long it isn't, it makes no sense to=20
>  > base sip aaa
>  > work on diameter.
>  >=20
>  > -- juha
>  >=20
>  > _______________________________________________
>  > Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
>  > This list is for NEW development of the application of SIP
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sip@ietf.org for new developments of core SIP
>  >=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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Hi,

Let me allow to post this announce on the AAA mailing list.

A new project for develpping open source Diameter software has been
created.  A Diameter message parser library is available in the first
release.  We will actively work on releasing new stuff and updating 
them.  If you are interested in this project, visit 
http://opendiameter.org.  

For further questions, please send email to
diameter-devel@opendiameter.org, not to this list.

Thanks,
Yoshihiro Ohba


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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Wed Feb 19 05:16:44 2003
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        <sipping@ietf.org>, <john.loughney@nokia.com>, <jari.arkko@piuha.net>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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aki.niemi@nokia.com writes:

 > Isn't this a chicken-and-egg problem? Diameter just got to proposed
 > standard - back in 1999 if you had asked the same set of questions
 > about RFC 2543, I bet we'd all be talking about H.323 or
 > something... ;)

sure it is ok to develop new protocols.  however, nobody takes a
protocol seriously that doesn't have a good number of interoperable
implementations.  thus i consider it a mistake to make the sip community
a guinea pig for the diameter protocol and would stick to radius until
diameter has proven itself in products and real world deployments.

-- juha


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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I don't think the AAA architecture RG is an engineering alternative
the SIP WG can consider, assuming we want to get this thing deployed
and running in a relatively short time period with many vendors
supporting it.

My preference is DIAMETER. I do agree that we probably need to
look at RADIUS compatibility as well. Perhaps this is the compromise
we need for us to continue? (I don't see a major technical obstacle
to the compatibility; the DIAMETER NASREQ application has been defined
to work with existing RADIUS NASes. If SIP AAA is defined with this
in mind, we can get at least some amount of functionality over
RADIUS. The main problem is perhaps finding standardized RADIUS
attribute numbers, of which we have almost run out... Also naturally,
some new functions of DIAMETER can not be realized through translation
to RADIUS, such as the what requirement 2.3.4 in the SIP AAA requirements
draft indicates. And I would reserve RADIUS mode mainly for
compatibility with old products, not as the mandatory-to-implement
feature in some future RFC. It is seen as the weakest component
in the security of planned network access systems, for instance.)

Perhaps I'm biased for being involved in the DIAMETER work myself.
However, I am also involved with some planned very large scale
applications of SIP, and those folks intend to use DIAMETER --
they are also building products in this space.

Finally, a few comments to the SIP-AAA requirements:

- The security requirements might be expanded a bit.
   Are proxies going to be needed? Is end-to-end
   security through proxies necessary for some reason?

- The document does not mention anything about reliability
   of AAA transactions, accounting in particular. I would
   think this is of high interest, as Allison also noted.

- Do we really need both cumulative and non-cumulative
   accounting?

- 2.4.2 and 2.4.3 are unclear to me. What is really
   required here?

- Section 3.2 example is a bit thin. I suspect there's
   more to it than this.

- John's address has a non-ASCII character.

Jari

Henry Sinnreich wrote:
 > I still would like to hear if people think a discussion with the AAA
 > architecture RG is a good way to get out of this impasse. Please browse
 > the work at http://www.aaaarch.org/charter.html
 >
 > It appears there is enough push-back on DIAMETER in this WG that this
 > should be given some time for discussion at the next meeting. All
 > messages from the 'audience' favored RADIUS, contrary to the proposed
 > work agenda.
 >
 > I have yet to hear an argument why some form of XML documents over some
 > secure transport, as in Web Services should not be the choice after
 > RADIUS.
 >
 > Or are the authors so committed to a DIAMETER project that this
 > discussion is futile?
 >
 > Thanks, Henry
 >
 >
 >
 >
 >>-----Original Message-----
 >>From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
 >>Sent: Tuesday, February 18, 2003 11:35 PM
 >>To: Henry Sinnreich
 >>Cc: 'Gonzalo Camarillo'; 'sipping'; 'John Loughney'
 >>Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
 >>
 >>
 >>Henry Sinnreich writes:
 >>
 >> > I suggest therefore to start with RADIUS instead.
 >>
 >>i fully agree, juha
 >>
 >
 >
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 >
 >




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Date: Wed, 19 Feb 2003 19:44:54 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sipping] I-D ACTION:draft-rosenberg-sipping-lease-00.txt
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At 11:00 AM 2/17/2003 -0500, David R. Oran wrote:
>>         Title           : URI Leasing in the Session Initiation Protocol 
>> (SIP)
>>         Author(s)       : J. Rosenberg
>>         Filename        : draft-rosenberg-sipping-lease-00.txt
>>         Pages           : 23
>>         Date            : 2003-2-12
>>
>In general I like the direction this is going. I have a query. Did you 
>consider making the creation and communication of the GRUU simply a 
>side-effect of a REGISTER operation as opposed to a separate method? In 
>other words, the REGISTER would optionally request the construction of a 
>(range of/template for) GRUUs for the UA, and the response would contain 
>the constructed values. This would seem to meet all of the requirements, 
>including, at least provisionally, the requirement for not substantially 
>increasing the domain-level state stored by proxies.

I like this approach.  It also solves the problem of how a UA finds out if 
it needs to lease a URI or not and whether the registrar supports 
it.  Otherwise, it must either send an OPTIONS to the registrar or find out 
with a 501 that the registrar does not support a LEASE method.  I also 
don't think a UA can tell if it's Contact is routeable, but the registrar 
can (or should).    As a result UA might lease a URI that it in fact 
doesn't need if its Contact is already globally routeable which would 
needless increase stored state.

If the leased URI is returned in a new header or Contact parameter, the 
registrar would only do so if the UA needs it - if the Contact URI in the 
REGISTER is already globally routeable (no screening proxies, 
firewalls/NATs, etc.) then the registrar would not return any value.  UAs 
that didn't support the extension would simply ignore.  A UA supporting the 
extension would use the returned leased URI instead of the one that it 
registered with.


>One small point bears upon the above. In the text you say that a lease 
>should not extend beyond the registration expiration of a registered UA. 
>However, the requirement is weaker; it only says that the domain has final 
>say over the lease duration. Was this intentional? If so I would think 
>this complicates the life of a registrar somewhat, compared with managing 
>GRUUs just like registrations, or state directly bound to registrations. I 
>can see a rather straightforward implementation if leases are simply extra 
>stuff associated with a registration. I have a harder time seeing a simple 
>implementation if a new method or some completely separate mechanism is used.

And the separation can cause problems - a UA that wishes to watch the state 
of its registration and lease would have to watch both because if either 
one fails, then requests will fail.  Seems better to combine the two for 
the reasons Dave mentions.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

>Dave.
>
>
>Dave.
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

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Comments on draft-ietf-sipping-overlap-04:

General:

The ability of one call to result in multiple IAMs being sent into the PSTN 
continues to be a serious problem for overlap which must be solved.

Section 3.1:

All references to one number being a "prefix" of another should be deleted. 
This text indicates that overlap can not be used where this "prefix" 
situation can occur, however, one of the purposes of overlap is to handle the 
situation in which there are varying lengths of numbers in some destination 
area, something that would naturally exist if one number really could be a 
"prefix" of another. This creates a major contradiction. However, there is 
still no identified case in the world-wide numbering plan in which one number 
is a "prefix" of another. If there were, overlap signaling to get to it would 
certainly be allowed just as it would be in the existing PSTN. If one 
believes that this "prefix" situation really exists, then the overlap 
procedures defined in this draft must correctly support it.

Section 3.3 contains the statement:

   As a general rule, a media stream corresponding to the response to an
   INVITE with a greater number of digits should be given more priority
   than media streams from responses with less digits.

This continues to be a problem, since it is not always the case. If the 
originator dialed more digits than were needed when overlap was being used, 
the media stream corresponding to the response with the fewer digits would be 
the one to use. The PSTN had no problem with this. SIP is going to have to 
figure out a way to correctly handle it.

Section 3.5 contains the statement:

   If the gateway is attached to the PSTN in an area where en-bloc
   signalling is used, a REL for the previous IAM and a new IAM should
   be generated.

This causes confusion since all areas would allow en-bloc, although some do 
not allow overlap, for example, an area where US SS#7 according to T1.113 is 
used. In those areas, since overlap is not allowed, the gateway or switch 
which provides interworking into that non-overlap area must perform the 
overlap to en-bloc procedures (by time outs, detecting enough digits, etc.) 
There can not be a "previous IAM" sent to the PSTN, since the gateway or 
switch would not have sent one until it knew it had enough digits to send a 
complete IAM, after which it would stop accepting or sending any further 
digits.

This paragraph should be worded as:

   If the gateway is attached to the PSTN in an area where overlap signalling
   is not allowed, it must not send an IAM until it determines that it has 
   received the full number in one or more INVITEs, using a process similar 
to  
   that described in Section 2. It must not send SAMs or additional IAMs after
   that one IAM.

Mike Pierce


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Content-Type: text/html; charset="US-ASCII"
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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>Comments on draft-ietf-si=
pping-overlap-04:
<BR>
<BR>General:
<BR>
<BR>The ability of one call to result in multiple IAMs being sent into the P=
STN continues to be a serious problem for overlap which must be solved.
<BR>
<BR>Section 3.1:
<BR>
<BR>All references to one number being a "prefix" of another should be delet=
ed. This text indicates that overlap can not be used where this "prefix" sit=
uation can occur, however, one of the purposes of overlap is to handle the s=
ituation in which there are varying lengths of numbers in some destination a=
rea, something that would naturally exist if one number really could be a "p=
refix" of another. This creates a major contradiction. However, there is sti=
ll no identified case in the world-wide numbering plan in which one number i=
s a "prefix" of another. If there were, overlap signaling to get to it would=
 certainly be allowed just as it would be in the existing PSTN. If one belie=
ves that this "prefix" situation really exists, then the overlap procedures=20=
defined in this draft must correctly support it.
<BR>
<BR>Section 3.3 contains the statement:
<BR>
<BR> &nbsp;&nbsp;As a general rule, a media stream corresponding to the resp=
onse to an
<BR> &nbsp;&nbsp;INVITE with a greater number of digits should be given more=
 priority
<BR> &nbsp;&nbsp;than media streams from responses with less digits.
<BR>
<BR>This continues to be a problem, since it is not always the case. If the=20=
originator dialed more digits than were needed when overlap was being used,=20=
the media stream corresponding to the response with the fewer digits would b=
e the one to use. The PSTN had no problem with this. SIP is going to have to=
 figure out a way to correctly handle it.
<BR>
<BR>Section 3.5 contains the statement:
<BR>
<BR> &nbsp;&nbsp;If the gateway is attached to the PSTN in an area where en-=
bloc
<BR> &nbsp;&nbsp;signalling is used, a REL for the previous IAM and a new IA=
M should
<BR> &nbsp;&nbsp;be generated.
<BR>
<BR>This causes confusion since all areas would allow en-bloc, although some=
 do not allow overlap, for example, an area where US SS#7 according to T1.11=
3 is used. In those areas, since overlap is not allowed, the gateway or swit=
ch which provides interworking into that non-overlap area must perform the o=
verlap to en-bloc procedures (by time outs, detecting enough digits, etc.) T=
here can not be a "previous IAM" sent to the PSTN, since the gateway or swit=
ch would not have sent one until it knew it had enough digits to send a comp=
lete IAM, after which it would stop accepting or sending any further digits.
<BR>
<BR>This paragraph should be worded as:
<BR>
<BR> &nbsp;&nbsp;If the gateway is attached to the PSTN in an area where ove=
rlap signalling
<BR> &nbsp;&nbsp;is not allowed, it must not send an IAM until it determines=
 that it has=20
<BR> &nbsp;&nbsp;received the full number in one or more INVITEs, using a pr=
ocess similar to &nbsp;
<BR> &nbsp;&nbsp;that described in Section 2. It must not send SAMs or addit=
ional IAMs after
<BR> &nbsp;&nbsp;that one IAM.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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I agree!  SNMP v3 support for network LAN statistics etc. might just 
about make sense for SIP telphony adaptors, but it is unlikely that you 
would convince SIP telephone manufacturers of the need for such support 
in their products.

Adian Lewis

Henry Sinnreich wrote:

>Comments 1, 2, and 3 seem obvious - will edit in the next version.
>Thanks for pointing out.
>
>As for comment 4: "why can we not have SNMP v3 support for network LAN
>statistics, SIP traffic stats or even RTCP info on the SIP phone", 
>
> it would be interesting to see if SIP phone manufacturers think this is
>OK for any type of SIP device, or only for certain devices, such as SIP
>telephony adaptors that support key sets. I believe it makes sense only
>for latter. Opinions?
>
>Thanks, Henry
>
>  
>
>>-----Original Message-----
>>From: Brierre, Jean Joseph [mailto:Jean.Joseph.Brierre@wcom.com] 
>>Sent: Thursday, February 13, 2003 5:43 AM
>>To: Sinnreich, Henry; 'sipforum-discussion@lists.su.se'; 
>>'sipping@ietf.org'
>>Subject: RE: [Sipforum-discussion] SIP Telephony Device 
>>Requirements, Conf iguration and Data
>>
>>
>>Henry, 
>>
>>below are my comments to the Somefolks draft. 
>>
>>
>>
>>Integrated Switch/Hub SIP devices designed for wired Ethernet 
>>SHOULD have an uplink port such that another IP device, such 
>>as a personal computer, MAY share the network connection. SIP 
>>clients MUST prioritize the transmission of RTP traffic over 
>>the shared network connection. 
>>
>>comment 1: 
>>Wish you would remove the Hub reference since this is a 
>>shared collision domain concept which I do not recommend for 
>>sip client that has real time requirements sharing this 
>>shared Ethernet environment with a host device that may also 
>>have real time data on the same hub. Switch is the only way 
>>to go since ethernet HUB collisions between 2 devices can 
>>effect throughput
>>
>>"IP Requirements IP-1: SIP telephony devices MUST be able to 
>>acquire an IP address by: Automatic IP address configuration 
>>using DHCP, or Manual IP address entry from the device. "
>>
>>Comment 2 
>>Why did you not reference RFC3361 DHCP option for sip 
>>ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt this a great 
>>MAY RFC to support for sip clients to learn SIP proxy servers
>>
>>Comment 3 
>>No mention was made of support for Visual and Hearing 
>>Impaired Special Equipment or even RFC3351 
>>    
>>
>ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt
>
>Comment 4 
>You mention management but only for provisioning, why can we not have
>SNMP v3 support for network LAN statistics, SIP traffic stats or even
>RTCP info on the SIP phone.
>
>
>
>
>
>
>  
>
>>-----Original Message-----
>>From: sipforum-discussion-admin@lists.su.se
>>[mailto:sipforum-discussion-admin@lists.su.se] On Behalf Of 
>>Henry Sinnreich
>>Sent: Monday, February 10, 2003 8:49 AM
>>To: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
>>Subject: [Sipforum-discussion] SIP Telephony Device 
>>Requirements, Configuration and Data
>>
>>
>>For everyone interested SIP phones and other SIP
>>communication devices, and the 56 IETF  coming next month, 
>>please see if you like or dislike or don't care about the draft:
>>
>>    
>>
>http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt
>
>The idea is to have discussions on the lists, and if there is interest
>and rough consensus to move ahead with an informational RFC. Would
>appreciate your comments to 
>
>sipping@ietf.org , and also to
>
>sipforum-discussion@lists.su.se 
>
>Thanks, Henry
>
>Henry Sinnreich
>WorldCom
>400 International Parkway
>Richardson, Texas 75081
>USA
>
>
>  
>


--------------010700070200030407060707
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Content-Transfer-Encoding: 7bit

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
  <title></title>
</head>
<body>
I agree! &nbsp;SNMP v3 support for network LAN statistics etc. might just about
make sense for SIP telphony adaptors, but it is unlikely that you would convince
SIP telephone manufacturers of the need for such support in their products.<br>
<br>
Adian Lewis<br>
<br>
Henry Sinnreich wrote:<br>
<blockquote type="cite"
 cite="mid000001c2d374$7367e150$02212aa6@hsinnreich2">
  <pre wrap="">Comments 1, 2, and 3 seem obvious - will edit in the next version.
Thanks for pointing out.

As for comment 4: "why can we not have SNMP v3 support for network LAN
statistics, SIP traffic stats or even RTCP info on the SIP phone", 

 it would be interesting to see if SIP phone manufacturers think this is
OK for any type of SIP device, or only for certain devices, such as SIP
telephony adaptors that support key sets. I believe it makes sense only
for latter. Opinions?

Thanks, Henry

  </pre>
  <blockquote type="cite">
    <pre wrap="">-----Original Message-----
From: Brierre, Jean Joseph [<a class="moz-txt-link-freetext" href="mailto:Jean.Joseph.Brierre@wcom.com">mailto:Jean.Joseph.Brierre@wcom.com</a>] 
Sent: Thursday, February 13, 2003 5:43 AM
To: Sinnreich, Henry; '<a class="moz-txt-link-abbreviated" href="mailto:sipforum-discussion@lists.su.se">sipforum-discussion@lists.su.se</a>'; 
'<a class="moz-txt-link-abbreviated" href="mailto:sipping@ietf.org">sipping@ietf.org</a>'
Subject: RE: [Sipforum-discussion] SIP Telephony Device 
Requirements, Conf iguration and Data


Henry, 

below are my comments to the Somefolks draft. 



Integrated Switch/Hub SIP devices designed for wired Ethernet 
SHOULD have an uplink port such that another IP device, such 
as a personal computer, MAY share the network connection. SIP 
clients MUST prioritize the transmission of RTP traffic over 
the shared network connection. 

comment 1: 
Wish you would remove the Hub reference since this is a 
shared collision domain concept which I do not recommend for 
sip client that has real time requirements sharing this 
shared Ethernet environment with a host device that may also 
have real time data on the same hub. Switch is the only way 
to go since ethernet HUB collisions between 2 devices can 
effect throughput

"IP Requirements IP-1: SIP telephony devices MUST be able to 
acquire an IP address by: Automatic IP address configuration 
using DHCP, or Manual IP address entry from the device. "

Comment 2 
Why did you not reference RFC3361 DHCP option for sip 
<a class="moz-txt-link-freetext" href="ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt">ftp://ftp.rfc-editor.org/in-notes/rfc3361.txt</a> this a great 
MAY RFC to support for sip clients to learn SIP proxy servers

Comment 3 
No mention was made of support for Visual and Hearing 
Impaired Special Equipment or even RFC3351 
    </pre>
  </blockquote>
  <pre wrap=""><!----><a class="moz-txt-link-freetext" href="ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt">ftp://ftp.rfc-editor.org/in-notes/rfc3351.txt</a>

Comment 4 
You mention management but only for provisioning, why can we not have
SNMP v3 support for network LAN statistics, SIP traffic stats or even
RTCP info on the SIP phone.






  </pre>
  <blockquote type="cite">
    <pre wrap="">-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:sipforum-discussion-admin@lists.su.se">sipforum-discussion-admin@lists.su.se</a>
[<a class="moz-txt-link-freetext" href="mailto:sipforum-discussion-admin@lists.su.se">mailto:sipforum-discussion-admin@lists.su.se</a>] On Behalf Of 
Henry Sinnreich
Sent: Monday, February 10, 2003 8:49 AM
To: 'J&ouml;rgen Bj&ouml;rkner'; <a class="moz-txt-link-abbreviated" href="mailto:sipforum-discussion@lists.su.se">sipforum-discussion@lists.su.se</a>
Subject: [Sipforum-discussion] SIP Telephony Device 
Requirements, Configuration and Data


For everyone interested SIP phones and other SIP
communication devices, and the 56 IETF  coming next month, 
please see if you like or dislike or don't care about the draft:

    </pre>
  </blockquote>
  <pre wrap=""><!----><a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt">http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt</a>

The idea is to have discussions on the lists, and if there is interest
and rough consensus to move ahead with an informational RFC. Would
appreciate your comments to 

<a class="moz-txt-link-abbreviated" href="mailto:sipping@ietf.org">sipping@ietf.org</a> , and also to

<a class="moz-txt-link-abbreviated" href="mailto:sipforum-discussion@lists.su.se">sipforum-discussion@lists.su.se</a> 

Thanks, Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA


  </pre>
</blockquote>
<br>
</body>
</html>

--------------010700070200030407060707--

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From: =?iso-8859-1?Q?Gunnar_Hellstr=F6m?= <gunnar.hellstrom@omnitor.se>
To: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>,
        "=?iso-8859-1?Q?'J=F6rgen_Bj=F6rkner'?=" <jorgen.bjorkner@hotsip.com>,
        <sipforum-discussion@lists.su.se>
Cc: <sipping@ietf.org>
Date: Thu, 13 Feb 2003 21:21:41 +0100
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Subject: [Sipping] Re: [Sipforum-discussion] SIP Telephony Device Requirements, Configuration and Data
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Henry,

It is mentioned in the introduction to the document that "text shall be kept
in mind".

I suggest some specific text items to be entered that are in line with
existing standardisation, and accessibility needs and useful for all:

----------------------------------------------------------------------------
-------------------
SIP-xx, analogue text support: SIP adapter devices (for analog phone lines)
  MUST support conversion between real time text transmission using RFC 2793
[y]
  and text telephones according to ITU-T V.18 [x] allowing alternating use
of
  text and voice. SIP clients MUST fallback to G.711 if RFC2793 connection
fails.


SIP-zz, digital text support: SIP telephone devices MUST support
  real time text conversation using RFC 2793 [y] for the text stream.
  It MUST be possible to use text simultaneously with voice.



---------------------------------------------
In 4.11 Ringer Behaviour  : add the section below for the purpose of deaf,
hearing impaired, deaf-blind people and people in the garden and noisy
locations:

SIP Telephony devices MUST provide an external interface to
an external alerting system.



Add to References
-----------------
[y] G. Hellstrom: "RTP Payload for Text Conversation.", RFC 2793, IETF, May
2000.
[z] ITU-T Recommendation V.18 "Operational and interworking requirements for
DCEs operating in the text telephone mode", November 2000 with amendment
November 2002.

----------------------------------------------------------------------------
-------




Thanks
-------------------------------------------
Gunnar Hellström
Omnitor AB
Renathvägen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------

> -----Ursprungligt meddelande-----
> Från: sipforum-discussion-admin@lists.su.se
> [mailto:sipforum-discussion-admin@lists.su.se]För Henry Sinnreich
> Skickat: den 10 februari 2003 15:49
> Till: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> Ämne: [Sipforum-discussion] SIP Telephony Device Requirements,
> Configuration and Data
>
>
> For everyone interested SIP phones and other SIP communication devices,
> and the 56 IETF  coming next month, please see if you like or dislike or
> don't care about the draft:
>
> http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt
>
> The idea is to have discussions on the lists, and if there is interest
> and rough consensus to move ahead with an informational RFC. Would
> appreciate your comments to
>
> sipping@ietf.org , and also to
>
> sipforum-discussion@lists.su.se
>
> Thanks, Henry
>
> Henry Sinnreich
> WorldCom
> 400 International Parkway
> Richardson, Texas 75081
> USA
>
>
>


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "PRATI,LUCA (HP-Italy,ex1)" <luca.prati@hp.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>,
        sipping <sipping@ietf.org>
Cc: Dean Willis <dwillis@softarmor.com>, Rohan Mahy <rohan@cisco.com>,
        Eric Burger <eburger@snowshore.com>
Subject: RE: [Sipping] Expert review for draft-burger-sipping-netann-04.tx
	t
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Folks,

in case of pre-paid service the we should also support the Advice of
Charging. AOC information could be presented at the start of the call, or
during a call, or at the end of the call. AOC may be presented using an
audible tone or announcement (e.g. remaining balance = 33 minutes). 

The call flow in that specific case is not clear for me especially in case
of Mid-Call Interruption...could help me?

Thanks,

Luca

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Monday, February 10, 2003 11:40 AM
To: sipping
Cc: Dean Willis; Rohan Mahy; Eric Burger
Subject: [Sipping] Expert review for draft-burger-sipping-netann-04.txt


Folks,

We would like the SIPPING working group to provide expert review of:

http://www.ietf.org/internet-drafts/draft-burger-sipping-netann-04.txt

The authors have requested individual publication of this document. 
Expert Review will close on March 10th, 2003

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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I don't think the AAA architecture RG is an engineering alternative
the SIP WG can consider, assuming we want to get this thing deployed
and running in a relatively short time period with many vendors
supporting it.

My preference is DIAMETER. I do agree that we probably need to
look at RADIUS compatibility as well. Perhaps this is the compromise
we need for us to continue? (I don't see a major technical obstacle
to the compatibility; the DIAMETER NASREQ application has been defined
to work with existing RADIUS NASes. If SIP AAA is defined with this
in mind, we can get at least some amount of functionality over
RADIUS. The main problem is perhaps finding standardized RADIUS
attribute numbers, of which we have almost run out... Also naturally,
some new functions of DIAMETER can not be realized through translation
to RADIUS, such as the what requirement 2.3.4 in the SIP AAA requirements
draft indicates. And I would reserve RADIUS mode mainly for
compatibility with old products, not as the mandatory-to-implement
feature in some future RFC. It is seen as the weakest component
in the security of planned network access systems, for instance.)

Perhaps I'm biased for being involved in the DIAMETER work myself.
However, I am also involved with some planned very large scale
applications of SIP, and those folks intend to use DIAMETER --
they are also building products in this space.

Finally, a few comments to the SIP-AAA requirements:

- The security requirements might be expanded a bit.
   Are proxies going to be needed? Is end-to-end
   security through proxies necessary for some reason?

- The document does not mention anything about reliability
   of AAA transactions, accounting in particular. I would
   think this is of high interest, as Allison also noted.

- Do we really need both cumulative and non-cumulative
   accounting?

- 2.4.2 and 2.4.3 are unclear to me. What is really
   required here?

- Section 3.2 example is a bit thin. I suspect there's
   more to it than this.

- John's address has a non-ASCII character.

Jari

Henry Sinnreich wrote:
> I still would like to hear if people think a discussion with the AAA
> architecture RG is a good way to get out of this impasse. Please browse
> the work at http://www.aaaarch.org/charter.html
> 
> It appears there is enough push-back on DIAMETER in this WG that this
> should be given some time for discussion at the next meeting. All
> messages from the 'audience' favored RADIUS, contrary to the proposed
> work agenda.
> 
> I have yet to hear an argument why some form of XML documents over some
> secure transport, as in Web Services should not be the choice after
> RADIUS.
> 
> Or are the authors so committed to a DIAMETER project that this
> discussion is futile?
> 
> Thanks, Henry
> 
> 
> 
> 
>>-----Original Message-----
>>From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi] 
>>Sent: Tuesday, February 18, 2003 11:35 PM
>>To: Henry Sinnreich
>>Cc: 'Gonzalo Camarillo'; 'sipping'; 'John Loughney'
>>Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
>>
>>
>>Henry Sinnreich writes:
>>
>> > I suggest therefore to start with RADIUS instead.
>>
>>i fully agree, juha
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 



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This list is for NEW development of the application of SIP
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Date: Wed, 19 Feb 2003 13:44:29 -0500
To: Allison Mankin <mankin@psg.com>
From: Stuart Jacobs <stu.jacobs@labs.gte.com>
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft 
Cc: sipping@ietf.org
In-Reply-To: <E18jvLW-000HAk-00@psg.com>
References: <Message from Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
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Allison,

Within Verizon, accounting of SIP based services will most likely not use 
Diameter in our Next Generation Network (NGN) VoIP Services.  We will be 
doing our usage accounting within SIP Proxies and intend to transfer said 
accounting "call detail records" back to our billing OSSes via other 
protocols with IPsec providing authentication and confidentiality for these 
transfers.  We currently do not plan on any deployment of Diameter within 
Verizon's wireline NGN architecture.  We will probably continue to use 
Radius for remote access authentication at the application layer for a 
number or years into the future.

Stu Jacobs

At 2/14/03 09:58 PM, you wrote:

>Jiri,
>
>It always seems good to be able to stay with installed base, but
>there's a factor against our supporting RADIUS drafts for SIP's
>accounting - safety.  Even though there is deployment, there are
>some extremely bad cases of damage in production, and the IETF
>has concluded that RADIUS is not suitable for usage-based accounting
>(RFC2975).
>
>Unless there is no intention of having in SIP accounting usage-based,
>monetary, auditable applications, RADIUS is not the solution.  The
>protocol just does not preserve a complete stream of reliable
>accounting records that are auditable on any level, especially with
>any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
>the use of RADIUS may even be illegal in situations where there is an
>authority over the billing.  Records are too easily lost by the
>protocol. This is stated clearly by RFC2975.  This is the reason why
>the Operations and Mgt Area developed Diameter.  On these technical*
>grounds, that the RADIUS protocol does not have enough reliability,
>the work must be based on the Diameter protocol, which is now
>approved.  Alternatively, work could be done that does not include
>usage-based accounting, but this does not seem to be what is
>envisioned in the SIPPING AAA requirements.
>
>Allison
>
>P.S. There's a current investigation of RADIUS compatibility mode
>in Diameter for this - more needs to be thought about it.
> >
> > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > >Hi Henry,
> > >
> > >> >but one option is develop a solution using
> > >> > Diameter in RADIUS-compatibility mode.
> > >>
> > >> OK, this is better than nothing. We'll take what we can :-)
> > >>
> > >> As mentioned, the installed base is RADIUS, no matter how
> > >> voluminous the DIAMETER literature may be.
> > >
> > >I think the issues is that the IETF should recommend what is the
> > >right thing to do & why.  If people choose to do something else,
> > >well, that is life.
> >
> > Remember that running code does not necessarily mean something known
> > to compile -- that is something known to be deployed.
> >
> > >I understand what you are asking but I do note that some
> > >people have thought that the potential for AAA is much greater
> > >than the existing installed base.
> >
> > I actually think that the deployment argument is a very important one.
> >
> > -Jiri
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

==========================
Stuart Jacobs CISSP
PMTS - Sr. Technologist
Verizon Laboratories
40 Sylvan Road Waltham, MA 02451-1128     USA
telephone: (781) 466-3076   fax: (781) 466-2838
stu.jacobs@labs.gte.com sjj0@labs.gte.com  stu.jacobs@verizon.com
==========================

_______________________________________________
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This list is for NEW development of the application of SIP
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Date: Thu, 20 Feb 2003 08:06:29 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] Re: [Sipforum-discussion] SIP Telephony Device
 Requirements, Configuration and Data
In-reply-to: <NFBBJHOFMDBFILNPMNGKOENADLAA.gunnar.hellstrom@omnitor.se>
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Gunnar Hellström writes:

> SIP-xx, analogue text support: SIP adapter devices (for 
> analog phone lines)
>   MUST support conversion between real time text transmission 
> using RFC 2793 [y]
>   and text telephones according to ITU-T V.18 [x] allowing 
> alternating use of
>   text and voice. SIP clients MUST fallback to G.711 if 
> RFC2793 connection fails.

This and the other below are excellent points! Thanks for the
contribution. Will add in the next version of 

http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Gunnar Hellström
> Sent: Thursday, February 13, 2003 2:22 PM
> To: Henry Sinnreich; 'Jörgen Björkner'; 
> sipforum-discussion@lists.su.se
> Cc: sipping@ietf.org
> Subject: [Sipping] Re: [Sipforum-discussion] SIP Telephony 
> Device Requirements, Configuration and Data
> 
> 
> Henry,
> 
> It is mentioned in the introduction to the document that 
> "text shall be kept in mind".
> 
> I suggest some specific text items to be entered that are in 
> line with existing standardisation, and accessibility needs 
> and useful for all:
> 
> --------------------------------------------------------------
> --------------
> -------------------
> SIP-xx, analogue text support: SIP adapter devices (for 
> analog phone lines)
>   MUST support conversion between real time text transmission 
> using RFC 2793 [y]
>   and text telephones according to ITU-T V.18 [x] allowing 
> alternating use of
>   text and voice. SIP clients MUST fallback to G.711 if 
> RFC2793 connection fails.
> 
> 
> SIP-zz, digital text support: SIP telephone devices MUST support
>   real time text conversation using RFC 2793 [y] for the text stream.
>   It MUST be possible to use text simultaneously with voice.
> 
> 
> 
> ---------------------------------------------
> In 4.11 Ringer Behaviour  : add the section below for the 
> purpose of deaf, hearing impaired, deaf-blind people and 
> people in the garden and noisy
> locations:
> 
> SIP Telephony devices MUST provide an external interface to
> an external alerting system.
> 
> 
> 
> Add to References
> -----------------
> [y] G. Hellstrom: "RTP Payload for Text Conversation.", RFC 
> 2793, IETF, May 2000. [z] ITU-T Recommendation V.18 
> "Operational and interworking requirements for DCEs operating 
> in the text telephone mode", November 2000 with amendment 
> November 2002.
> 
> --------------------------------------------------------------
> --------------
> -------
> 
> 
> 
> 
> Thanks
> -------------------------------------------
> Gunnar Hellström
> Omnitor AB
> Renathvägen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> > -----Ursprungligt meddelande-----
> > Från: sipforum-discussion-admin@lists.su.se
> > [mailto:sipforum-discussion-admin@lists.su.se]För Henry Sinnreich
> > Skickat: den 10 februari 2003 15:49
> > Till: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
> > Ämne: [Sipforum-discussion] SIP Telephony Device Requirements, 
> > Configuration and Data
> >
> >
> > For everyone interested SIP phones and other SIP communication 
> > devices, and the 56 IETF  coming next month, please see if 
> you like or 
> > dislike or don't care about the draft:
> >
> > 
> http://www.ietf.org/internet->
drafts/draft-sinnreich-sipdev-req-00.txt
> >
> > The idea is to have discussions on the lists, and if there 
> is interest 
> > and rough consensus to move ahead with an informational RFC. Would 
> > appreciate your comments to
> >
> > sipping@ietf.org , and also to
> >
> > sipforum-discussion@lists.su.se
> >
> > Thanks, Henry
> >
> > Henry Sinnreich
> > WorldCom
> > 400 International Parkway
> > Richardson, Texas 75081
> > USA
> >
> >
> >
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'"
	 <sipping@ietf.org>
Cc: "'Dean Willis'" <dwillis@softarmor.com>, "'Rohan Mahy'" <rohan@cisco.com>,
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        "'Henning Schulzrinne'"
	 <hgs@cs.columbia.edu>,
        "'alan.johnston@wcom.com'"
	 <alan.johnston@wcom.com>,
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Subject: RE: [Sipping] SIP-H.323 design team
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Gonzallo and all,

I have a few comments/concerns about this requirement document.

- Section 6 states that the IWF shall use the H.323 version 2
Recommendation. I don't believe
  this is a valid requirement. Many (if not most) implementations are up to
versions 3 and 4
  these days. We could either say nothing at all about H.323 versions, or
require version 2
  or above. In H.323, new versions need to be backward compatible with older
versions, so this
  shouldn't be an issue. 

- Many requirements are extremely vague. For example, things like 6.1.2 a
"the IWF shall support
  all the addressing shcemes of both H.323 and SIP". I'm not sure if this is
even feasible. 
  Requirement 8.1.1 is also very vague I also really doubt that it is even
possible to do so: 
  I'd recommend writing examples instead of a clear mapping).

- Certain requirements seem to modify the behavior that and H.323 endpoint
or GK may have. We
  can not impose system-wide behavioral changes on an existing H.323 (or
SIP) network. For example, 
  6.4 says that the pre-granted ARQ shall be supported by the IWF. It is
entirely up to the
  GK to decide which mode (pre-granted or not) will be used, not the IWF.
Perhaps I'm not reading
  the requirement properly and this requirement is just saying that the IWF
needs to support
  both pre-Granted ARQ and normal ARQ (if it is the case, then this bullet
should just be 
  merger with the previous requirement about vague requirements). One of the
reason I am
  thinking that the intent is to prescribe a specific behavior on the H.323
(and SIP) protocol
  is that the now-expired draft-agrawal-sip-h323-interworking-01.txt was
doing so. Are we
  thinking of ressucitating this draft, or are we thinking of starting from
scratch with
  a different approach?

- Section 6.5 uses the INFO method for sending DTMF information. I will
point out that H.323v4
  supports RFC2833 (the procedures for exchanging the necessary dynamic
payload type are 
  similar to the procedures used in SIP).

- Section 8.2 I like.

- Section 8.4 needs to be clarified (what is "invisible support"?).

- Section 12.4 is probably one of the most important to look into. It should
explain how the
  offer/answer model maps to H.323 (both fastStart and mid/call or slow
start). It should 
  talk about re-INVITE, UPDATE, OpenLogicalChannnel, a=sendonly/inactive,
TCS=0,etc. 
  I am pretty sure that there is no simple mapping as the models are quite
different.

- There should also be a section on in-band and out-of band media (early
media) and how they 
  map.

Lastly, we need to make sure that this document is meant to be informative
at best. I would
much prefer to have document explaining the various mismatches between SIP
and H.323 and how 
to minimize them than a document trying to describe a fixed mapping between
the two: more 
a free-form description of the various "gotchas" that may be encountered
when trying to 
interwork the two protocols.

For example, it could explain how opening logical channels in H.323 and the
terminal 
capability set negotiation process differ from the offer/answer model. I see
very little 
benefits in creating a document that would describe only a mapping for
"basic call setup" 
as this is not where people are going to have difficulties.

I am convinced that a "simple protocol mapping" approach such as the one
defined in the now
expired draft-agrawal-sip-h323-interworking-01.txt is not possible and that
instead the IWF 
will have to maintain full call control/state information in order to map
properly between 
the protocols. I think we would be doing developpers and service providers a
great disservice 
by issuing an RFC implying that a simple "protocol mapping" box might be
sufficient to allow 
for interworking between SIP and H.323. I'd rather see no interworking
document at all than 
a simplistic mapping document.

Comments?

François Audet



> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Monday, February 10, 2003 12:17 AM
> To: sipping
> Cc: Dean Willis; Rohan Mahy; kns10@cs.columbia.edu; Henning
> Schulzrinne; alan.johnston@wcom.com; rrroy@att.com
> Subject: [Sipping] SIP-H.323 design team
> 
> 
> Folks,
> 
> The requirements document for SIP-H.323 interworking is advancing:
> 
http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-reqs
-03.txt

Now, it is time to start thinking about the solution document. If there are
enough people willing to contribute, we intend to form a design team in
SIPPING.

The scope of the work will be most likely limited to basic call set-up.

Those of you who are interested in being part of such a design team, please
let me know.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
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<P><FONT SIZE=2>Gonzallo and all,</FONT>
</P>

<P><FONT SIZE=2>I have a few comments/concerns about this requirement document.</FONT>
</P>

<P><FONT SIZE=2>- Section 6 states that the IWF shall use the H.323 version 2 Recommendation. I don't believe</FONT>
<BR><FONT SIZE=2>&nbsp; this is a valid requirement. Many (if not most) implementations are up to versions 3 and 4</FONT>
<BR><FONT SIZE=2>&nbsp; these days. We could either say nothing at all about H.323 versions, or require version 2</FONT>
<BR><FONT SIZE=2>&nbsp; or above. In H.323, new versions need to be backward compatible with older versions, so this</FONT>
<BR><FONT SIZE=2>&nbsp; shouldn't be an issue. </FONT>
</P>

<P><FONT SIZE=2>- Many requirements are extremely vague. For example, things like 6.1.2 a &quot;the IWF shall support</FONT>
<BR><FONT SIZE=2>&nbsp; all the addressing shcemes of both H.323 and SIP&quot;. I'm not sure if this is even feasible. </FONT>
<BR><FONT SIZE=2>&nbsp; Requirement 8.1.1 is also very vague I also really doubt that it is even possible to do so: </FONT>
<BR><FONT SIZE=2>&nbsp; I'd recommend writing examples instead of a clear mapping).</FONT>
</P>

<P><FONT SIZE=2>- Certain requirements seem to modify the behavior that and H.323 endpoint or GK may have. We</FONT>
<BR><FONT SIZE=2>&nbsp; can not impose system-wide behavioral changes on an existing H.323 (or SIP) network. For example, </FONT>
<BR><FONT SIZE=2>&nbsp; 6.4 says that the pre-granted ARQ shall be supported by the IWF. It is entirely up to the</FONT>
<BR><FONT SIZE=2>&nbsp; GK to decide which mode (pre-granted or not) will be used, not the IWF. Perhaps I'm not reading</FONT>
<BR><FONT SIZE=2>&nbsp; the requirement properly and this requirement is just saying that the IWF needs to support</FONT>
<BR><FONT SIZE=2>&nbsp; both pre-Granted ARQ and normal ARQ (if it is the case, then this bullet should just be </FONT>
<BR><FONT SIZE=2>&nbsp; merger with the previous requirement about vague requirements). One of the reason I am</FONT>
<BR><FONT SIZE=2>&nbsp; thinking that the intent is to prescribe a specific behavior on the H.323 (and SIP) protocol</FONT>
<BR><FONT SIZE=2>&nbsp; is that the now-expired draft-agrawal-sip-h323-interworking-01.txt was doing so. Are we</FONT>
<BR><FONT SIZE=2>&nbsp; thinking of ressucitating this draft, or are we thinking of starting from scratch with</FONT>
<BR><FONT SIZE=2>&nbsp; a different approach?</FONT>
</P>

<P><FONT SIZE=2>- Section 6.5 uses the INFO method for sending DTMF information. I will point out that H.323v4</FONT>
<BR><FONT SIZE=2>&nbsp; supports RFC2833 (the procedures for exchanging the necessary dynamic payload type are </FONT>
<BR><FONT SIZE=2>&nbsp; similar to the procedures used in SIP).</FONT>
</P>

<P><FONT SIZE=2>- Section 8.2 I like.</FONT>
</P>

<P><FONT SIZE=2>- Section 8.4 needs to be clarified (what is &quot;invisible support&quot;?).</FONT>
</P>

<P><FONT SIZE=2>- Section 12.4 is probably one of the most important to look into. It should explain how the</FONT>
<BR><FONT SIZE=2>&nbsp; offer/answer model maps to H.323 (both fastStart and mid/call or slow start). It should </FONT>
<BR><FONT SIZE=2>&nbsp; talk about re-INVITE, UPDATE, OpenLogicalChannnel, a=sendonly/inactive, TCS=0,etc. </FONT>
<BR><FONT SIZE=2>&nbsp; I am pretty sure that there is no simple mapping as the models are quite different.</FONT>
</P>

<P><FONT SIZE=2>- There should also be a section on in-band and out-of band media (early media) and how they </FONT>
<BR><FONT SIZE=2>&nbsp; map.</FONT>
</P>

<P><FONT SIZE=2>Lastly, we need to make sure that this document is meant to be informative at best. I would</FONT>
<BR><FONT SIZE=2>much prefer to have document explaining the various mismatches between SIP and H.323 and how </FONT>
<BR><FONT SIZE=2>to minimize them than a document trying to describe a fixed mapping between the two: more </FONT>
<BR><FONT SIZE=2>a free-form description of the various &quot;gotchas&quot; that may be encountered when trying to </FONT>
<BR><FONT SIZE=2>interwork the two protocols.</FONT>
</P>

<P><FONT SIZE=2>For example, it could explain how opening logical channels in H.323 and the terminal </FONT>
<BR><FONT SIZE=2>capability set negotiation process differ from the offer/answer model. I see very little </FONT>
<BR><FONT SIZE=2>benefits in creating a document that would describe only a mapping for &quot;basic call setup&quot; </FONT>
<BR><FONT SIZE=2>as this is not where people are going to have difficulties.</FONT>
</P>

<P><FONT SIZE=2>I am convinced that a &quot;simple protocol mapping&quot; approach such as the one defined in the now</FONT>
<BR><FONT SIZE=2>expired draft-agrawal-sip-h323-interworking-01.txt is not possible and that instead the IWF </FONT>
<BR><FONT SIZE=2>will have to maintain full call control/state information in order to map properly between </FONT>
<BR><FONT SIZE=2>the protocols. I think we would be doing developpers and service providers a great disservice </FONT>
<BR><FONT SIZE=2>by issuing an RFC implying that a simple &quot;protocol mapping&quot; box might be sufficient to allow </FONT>
<BR><FONT SIZE=2>for interworking between SIP and H.323. I'd rather see no interworking document at all than </FONT>
<BR><FONT SIZE=2>a simplistic mapping document.</FONT>
</P>

<P><FONT SIZE=2>Comments?</FONT>
</P>

<P><FONT SIZE=2>François Audet</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Gonzalo Camarillo [<A HREF="mailto:Gonzalo.Camarillo@lmf.ericsson.se">mailto:Gonzalo.Camarillo@lmf.ericsson.se</A>]</FONT>
<BR><FONT SIZE=2>&gt; Sent: Monday, February 10, 2003 12:17 AM</FONT>
<BR><FONT SIZE=2>&gt; To: sipping</FONT>
<BR><FONT SIZE=2>&gt; Cc: Dean Willis; Rohan Mahy; kns10@cs.columbia.edu; Henning</FONT>
<BR><FONT SIZE=2>&gt; Schulzrinne; alan.johnston@wcom.com; rrroy@att.com</FONT>
<BR><FONT SIZE=2>&gt; Subject: [Sipping] SIP-H.323 design team</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Folks,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; The requirements document for SIP-H.323 interworking is advancing:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2><A HREF="http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-reqs-03.txt" TARGET="_blank">http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-reqs-03.txt</A></FONT>
</P>

<P><FONT SIZE=2>Now, it is time to start thinking about the solution document. If there are enough people willing to contribute, we intend to form a design team in SIPPING.</FONT></P>

<P><FONT SIZE=2>The scope of the work will be most likely limited to basic call set-up.</FONT>
</P>

<P><FONT SIZE=2>Those of you who are interested in being part of such a design team, please let me know.</FONT>
</P>

<P><FONT SIZE=2>Thanks,</FONT>
</P>

<P><FONT SIZE=2>Gonzalo</FONT>
<BR><FONT SIZE=2>_______________________________________________</FONT>
<BR><FONT SIZE=2>Sipping mailing list&nbsp; <A HREF="https://www1.ietf.org/mailman/listinfo/sipping" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FONT>
<BR><FONT SIZE=2>This list is for NEW development of the application of SIP</FONT>
<BR><FONT SIZE=2>Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP</FONT>
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Date: Thu, 20 Feb 2003 17:36:13 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <3E5339E4.7030204@piuha.net>
To: jari.arkko@piuha.net
Cc: jh@lohi.eng.song.fi,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'" <sipping@ietf.org>,
        "'John Loughney'" <john.loughney@nokia.com>
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> I don't think the AAA architecture RG is an engineering 
> alternative the SIP WG can consider,

It would be only fair to include the aaaarch folks in this discussion.
They have done serious work - applicable to the real world.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Jari Arkko
> Sent: Wednesday, February 19, 2003 2:02 AM
> To: Henry Sinnreich
> Cc: jh@lohi.eng.song.fi; 'Gonzalo Camarillo'; 'sipping'; 
> 'John Loughney'
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> 
> I don't think the AAA architecture RG is an engineering 
> alternative the SIP WG can consider, assuming we want to get 
> this thing deployed and running in a relatively short time 
> period with many vendors supporting it.
> 
> My preference is DIAMETER. I do agree that we probably need 
> to look at RADIUS compatibility as well. Perhaps this is the 
> compromise we need for us to continue? (I don't see a major 
> technical obstacle to the compatibility; the DIAMETER NASREQ 
> application has been defined to work with existing RADIUS 
> NASes. If SIP AAA is defined with this in mind, we can get at 
> least some amount of functionality over RADIUS. The main 
> problem is perhaps finding standardized RADIUS attribute 
> numbers, of which we have almost run out... Also naturally, 
> some new functions of DIAMETER can not be realized through 
> translation to RADIUS, such as the what requirement 2.3.4 in 
> the SIP AAA requirements draft indicates. And I would reserve 
> RADIUS mode mainly for compatibility with old products, not 
> as the mandatory-to-implement feature in some future RFC. It 
> is seen as the weakest component in the security of planned 
> network access systems, for instance.)
> 
> Perhaps I'm biased for being involved in the DIAMETER work 
> myself. However, I am also involved with some planned very 
> large scale applications of SIP, and those folks intend to 
> use DIAMETER -- they are also building products in this space.
> 
> Finally, a few comments to the SIP-AAA requirements:
> 
> - The security requirements might be expanded a bit.
>    Are proxies going to be needed? Is end-to-end
>    security through proxies necessary for some reason?
> 
> - The document does not mention anything about reliability
>    of AAA transactions, accounting in particular. I would
>    think this is of high interest, as Allison also noted.
> 
> - Do we really need both cumulative and non-cumulative
>    accounting?
> 
> - 2.4.2 and 2.4.3 are unclear to me. What is really
>    required here?
> 
> - Section 3.2 example is a bit thin. I suspect there's
>    more to it than this.
> 
> - John's address has a non-ASCII character.
> 
> Jari
> 
> Henry Sinnreich wrote:
> > I still would like to hear if people think a discussion 
> with the AAA 
> > architecture RG is a good way to get out of this impasse. Please 
> > browse the work at http://www.aaaarch.org/charter.html
> > 
> > It appears there is enough push-back on DIAMETER in this WG 
> that this 
> > should be given some time for discussion at the next meeting. All 
> > messages from the 'audience' favored RADIUS, contrary to 
> the proposed 
> > work agenda.
> > 
> > I have yet to hear an argument why some form of XML documents over 
> > some secure transport, as in Web Services should not be the choice 
> > after RADIUS.
> > 
> > Or are the authors so committed to a DIAMETER project that this 
> > discussion is futile?
> > 
> > Thanks, Henry
> > 
> > 
> > 
> > 
> >>-----Original Message-----
> >>From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> >>Sent: Tuesday, February 18, 2003 11:35 PM
> >>To: Henry Sinnreich
> >>Cc: 'Gonzalo Camarillo'; 'sipping'; 'John Loughney'
> >>Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> >>
> >>
> >>Henry Sinnreich writes:
> >>
> >> > I suggest therefore to start with RADIUS instead.
> >>
> >>i fully agree, juha
> >>
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> > 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

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To: Francois Audet <audet@nortelnetworks.com>
CC: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'" <sipping@ietf.org>,
        "'Dean Willis'" <dwillis@softarmor.com>,
        "'Rohan Mahy'" <rohan@cisco.com>,
        "'kns10@cs.columbia.edu'" <kns10@cs.columbia.edu>,
        "'alan.johnston@wcom.com'" <alan.johnston@wcom.com>,
        "'rrroy@att.com'" <rrroy@att.com>
Subject: Re: [Sipping] SIP-H.323 design team
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Francois Audet wrote:
> Gonzallo and all,
> 
> I have a few comments/concerns about this requirement document.

Before we go there: the requirements document was last-called sometime 
in the last millenium and is now at the RFC editor. (The text will 
change a bit due to some last-minute editing to make it a bit more 
readable, but it's too late to make substantive changes).

draft-agrawal-sip-h323-interworking-reqs-03 was issued in early 2002, I 
think and was IESG approved for informational on April 18, 2002. Thus, 
your comments would have been somewhat more helpful about 18 months ago :-)

> 
> - Section 6 states that the IWF shall use the H.323 version 2 
> Recommendation. I don't believe
>   this is a valid requirement. Many (if not most) implementations are up 
> to versions 3 and 4
>   these days. We could either say nothing at all about H.323 versions, 
> or require version 2
>   or above. In H.323, new versions need to be backward compatible with 
> older versions, so this
>   shouldn't be an issue.
> 
> - Many requirements are extremely vague. For example, things like 6.1.2 
> a "the IWF shall support
>   all the addressing shcemes of both H.323 and SIP". I'm not sure if 
> this is even feasible.
>   Requirement 8.1.1 is also very vague I also really doubt that it is 
> even possible to do so:
>   I'd recommend writing examples instead of a clear mapping).
> 
> - Certain requirements seem to modify the behavior that and H.323 
> endpoint or GK may have. We
>   can not impose system-wide behavioral changes on an existing H.323 (or 
> SIP) network. For example,
>   6.4 says that the pre-granted ARQ shall be supported by the IWF. It is 
> entirely up to the
>   GK to decide which mode (pre-granted or not) will be used, not the 
> IWF. Perhaps I'm not reading
>   the requirement properly and this requirement is just saying that the 
> IWF needs to support
>   both pre-Granted ARQ and normal ARQ (if it is the case, then this 
> bullet should just be
>   merger with the previous requirement about vague requirements). One of 
> the reason I am
>   thinking that the intent is to prescribe a specific behavior on the 
> H.323 (and SIP) protocol
>   is that the now-expired draft-agrawal-sip-h323-interworking-01.txt was 
> doing so. Are we
>   thinking of ressucitating this draft, or are we thinking of starting 
> from scratch with
>   a different approach?
> 
> - Section 6.5 uses the INFO method for sending DTMF information. I will 
> point out that H.323v4
>   supports RFC2833 (the procedures for exchanging the necessary dynamic 
> payload type are
>   similar to the procedures used in SIP).

The final version has removed any mentioning of INFO. I caught this at 
the last minute :-)

> 
> - Section 8.2 I like.
> 
> - Section 8.4 needs to be clarified (what is "invisible support"?).
> 
> - Section 12.4 is probably one of the most important to look into. It 
> should explain how the
>   offer/answer model maps to H.323 (both fastStart and mid/call or slow 
> start). It should
>   talk about re-INVITE, UPDATE, OpenLogicalChannnel, 
> a=sendonly/inactive, TCS=0,etc.
>   I am pretty sure that there is no simple mapping as the models are 
> quite different.
> 
> - There should also be a section on in-band and out-of band media (early 
> media) and how they
>   map.
> 
> Lastly, we need to make sure that this document is meant to be 
> informative at best. I would

It will be, like all requirements documents.

> much prefer to have document explaining the various mismatches between 
> SIP and H.323 and how
> to minimize them than a document trying to describe a fixed mapping 
> between the two: more
> a free-form description of the various "gotchas" that may be encountered 
> when trying to
> interwork the two protocols.

Your contribution to such a document would be much appreciated. 
Unfortunately, it doesn't write itself...


> 
> For example, it could explain how opening logical channels in H.323 and 
> the terminal
> capability set negotiation process differ from the offer/answer model. I 
> see very little
> benefits in creating a document that would describe only a mapping for 
> "basic call setup"
> as this is not where people are going to have difficulties.
> 
> I am convinced that a "simple protocol mapping" approach such as the one 
> defined in the now
> expired draft-agrawal-sip-h323-interworking-01.txt is not possible and 
> that instead the IWF
> will have to maintain full call control/state information in order to 
> map properly between
> the protocols. I think we would be doing developpers and service 
> providers a great disservice
> by issuing an RFC implying that a simple "protocol mapping" box might be 
> sufficient to allow
> for interworking between SIP and H.323. I'd rather see no interworking 
> document at all than
> a simplistic mapping document.
> 
> Comments?
> 
> François Audet
> 

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Date: Thu, 20 Feb 2003 17:45:38 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <5.1.1.6.0.20030219133601.041fa380@pophost.gte.com>
To: "'Stuart Jacobs'" <stu.jacobs@labs.gte.com>,
        "'Allison Mankin'" <mankin@psg.com>
Cc: sipping@ietf.org, Henning Schulzrinne <hgs@cs.columbia.edu>
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Is there any service provider planning to use SIP-DIAMETER? If yes,
please speak up.

(Though a count for SIP-RADIUS would be just as interesting)

It looks not only is RADIUS the installed base, but it will continue to
be so.

The design group should therefore give priority to the SIP-RADIUS work.
Actually, the old I-D by Henning Schulzrinne is pretty solid and
probably could be used 'as is'.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Stuart Jacobs
> Sent: Wednesday, February 19, 2003 12:44 PM
> To: Allison Mankin
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Allison,
> 
> Within Verizon, accounting of SIP based services will most 
> likely not use 
> Diameter in our Next Generation Network (NGN) VoIP Services.  
> We will be 
> doing our usage accounting within SIP Proxies and intend to 
> transfer said 
> accounting "call detail records" back to our billing OSSes via other 
> protocols with IPsec providing authentication and 
> confidentiality for these 
> transfers.  We currently do not plan on any deployment of 
> Diameter within 
> Verizon's wireline NGN architecture.  We will probably 
> continue to use 
> Radius for remote access authentication at the application 
> layer for a 
> number or years into the future.
> 
> Stu Jacobs
> 
> At 2/14/03 09:58 PM, you wrote:
> 
> >Jiri,
> >
> >It always seems good to be able to stay with installed base, but 
> >there's a factor against our supporting RADIUS drafts for SIP's 
> >accounting - safety.  Even though there is deployment, there 
> are some 
> >extremely bad cases of damage in production, and the IETF 
> has concluded 
> >that RADIUS is not suitable for usage-based accounting (RFC2975).
> >
> >Unless there is no intention of having in SIP accounting 
> usage-based, 
> >monetary, auditable applications, RADIUS is not the solution.  The 
> >protocol just does not preserve a complete stream of reliable 
> >accounting records that are auditable on any level, 
> especially with any 
> >roaming.  As Bernard Aboba stated to this WG, quoting Mike 
> O'Dell, the 
> >use of RADIUS may even be illegal in situations where there is an 
> >authority over the billing.  Records are too easily lost by the 
> >protocol. This is stated clearly by RFC2975.  This is the reason why 
> >the Operations and Mgt Area developed Diameter.  On these technical* 
> >grounds, that the RADIUS protocol does not have enough 
> reliability, the 
> >work must be based on the Diameter protocol, which is now approved.  
> >Alternatively, work could be done that does not include usage-based 
> >accounting, but this does not seem to be what is envisioned in the 
> >SIPPING AAA requirements.
> >
> >Allison
> >
> >P.S. There's a current investigation of RADIUS compatibility mode in 
> >Diameter for this - more needs to be thought about it.
> > >
> > > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > > >Hi Henry,
> > > >
> > > >> >but one option is develop a solution using
> > > >> > Diameter in RADIUS-compatibility mode.
> > > >>
> > > >> OK, this is better than nothing. We'll take what we can :-)
> > > >>
> > > >> As mentioned, the installed base is RADIUS, no matter how 
> > > >> voluminous the DIAMETER literature may be.
> > > >
> > > >I think the issues is that the IETF should recommend what is the 
> > > >right thing to do & why.  If people choose to do something else, 
> > > >well, that is life.
> > >
> > > Remember that running code does not necessarily mean 
> something known 
> > > to compile -- that is something known to be deployed.
> > >
> > > >I understand what you are asking but I do note that some people 
> > > >have thought that the potential for AAA is much greater than the 
> > > >existing installed base.
> > >
> > > I actually think that the deployment argument is a very important 
> > > one.
> > >
> > > -Jiri
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listi> nfo/sipping
> > > This list 
> is for NEW development of the 
> application of SIP Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core SIP
> >_______________________________________________
> >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >This list is for NEW development of the application of SIP
> >Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> >sip@ietf.org for new developments of core SIP
> 
> ==========================
> Stuart Jacobs CISSP
> PMTS - Sr. Technologist
> Verizon Laboratories
> 40 Sylvan Road Waltham, MA 02451-1128     USA
> telephone: (781) 466-3076   fax: (781) 466-2838
> stu.jacobs@labs.gte.com sjj0@labs.gte.com  
> stu.jacobs@verizon.com ==========================
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

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To: Henry Sinnreich <Henry.Sinnreich@wcom.com>
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        "'Allison Mankin'" <mankin@psg.com>, sipping@ietf.org,
        Henning Schulzrinne <hgs@cs.columbia.edu>
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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Henry Sinnreich wrote:
> Is there any service provider planning to use SIP-DIAMETER? If yes,
> please speak up.

Any of the 3G operators count for this? I'm not sure how many
of the subscribe to this group though, even if their vendors
do...

Jari


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        "'Allison Mankin'" <mankin@psg.com>, sipping@ietf.org,
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Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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Jari Arkko writes:

 > Henry Sinnreich wrote:
 > > Is there any service provider planning to use SIP-DIAMETER? If yes,
 > > please speak up.
 > 
 > Any of the 3G operators count for this? I'm not sure how many
 > of the subscribe to this group though, even if their vendors
 > do...

it depends.  will the vendors really give 3g operators a free choice
which diameter server they want to use in their 3g network?  which
diameter servers has nokia tested interoperability against?  does any of
these servers come from non-3g vendors?  or is the use of diameter in 3g
networks similar fake as use of sip in 3g im?

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Jari Arkko <jari.arkko@piuha.net>
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Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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jh@lohi.eng.song.fi wrote:
> Jari Arkko writes:
> 
>  > Henry Sinnreich wrote:
>  > > Is there any service provider planning to use SIP-DIAMETER? If yes,
>  > > please speak up.
>  > 
>  > Any of the 3G operators count for this? I'm not sure how many
>  > of the subscribe to this group though, even if their vendors
>  > do...
> 
> it depends.  will the vendors really give 3g operators a free choice
> which diameter server they want to use in their 3g network?  which

I'm not quite sure what you mean by free choice in this context, but
perhaps you are after standardized interfaces? These would be publicly
available in the form of RFCs (eg DIAMETER base) and 3G specifications
on how DIAMETER is used for multimedia application.

> diameter servers has nokia tested interoperability against?  does any of

I don't know but there's been Diameter interoperability events. Perhaps
you John could dig up the Nokia information from those if you participated?
I know Ericsson particpated at least.

> these servers come from non-3g vendors?  or is the use of diameter in 3g
> networks similar fake as use of sip in 3g im?

The question was if anyone was going to deploy and I answered ;-)

I don't fully agree with the way 3G sets up their AAA. But we are
discussing the creation of an IETF standard for SIP AAA. 3G folks
had to create a multimedia AAA application because the IETF does
not have one. My opinion is that we need an IETF standard in this
space, so that the 3G folks, SIP vendor groups, or other parties
don't all diverge on how to do this. IMHO, this is really crucial
for SIP. My belief is also that the IETF standard will be better than
any of these groups could do themselves. As far as 3G goes, my
understanding was that they are happy to switch to an IETF standard
when it becomes available.

Jari

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 21 Feb 2003 08:45:57 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <3E55EB63.7080605@piuha.net>
To: jari.arkko@piuha.net, jh@lohi.eng.song.fi
Cc: "'John Loughney'" <john.loughney@nokia.com>, sipping@ietf.org
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>  > Any of the 3G operators count for this? I'm not sure how many
>  > of the subscribe to this group though, even if their vendors
>  > do...

I doubt any service providers will agree vendors are entitled to speak
for them, wired or wireless. The service providers on this list have
given solid arguments why we prefer RADIUS at present. As for what comes
beyond RADIUS, we have the OSP token I-D

http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-token-04.txt

 and the IETF work at

http://www.aaaarch.org/charter.html

DIAMETER is just too little, too late IMHO, when looking at the options
in these links.

Question: Has anyone predetermined in SIPPING that DIAMETER good for
service providers? Interesting...

Thanks, Henry



> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Jari Arkko
> Sent: Friday, February 21, 2003 3:04 AM
> To: jh@lohi.eng.song.fi
> Cc: 'John Loughney'; sipping@ietf.org
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> jh@lohi.eng.song.fi wrote:
> > Jari Arkko writes:
> > 
> >  > Henry Sinnreich wrote:
> >  > > Is there any service provider planning to use SIP-DIAMETER? If 
> > yes,  > > please speak up.  >
> >  > Any of the 3G operators count for this? I'm not sure how many
> >  > of the subscribe to this group though, even if their vendors
> >  > do...
> > 
> > it depends.  will the vendors really give 3g operators a 
> free choice 
> > which diameter server they want to use in their 3g network?  which
> 
> I'm not quite sure what you mean by free choice in this 
> context, but perhaps you are after standardized interfaces? 
> These would be publicly available in the form of RFCs (eg 
> DIAMETER base) and 3G specifications on how DIAMETER is used 
> for multimedia application.
> 
> > diameter servers has nokia tested interoperability against? 
>  does any 
> > of
> 
> I don't know but there's been Diameter interoperability 
> events. Perhaps you John could dig up the Nokia information 
> from those if you participated? I know Ericsson particpated at least.
> 
> > these servers come from non-3g vendors?  or is the use of 
> diameter in 
> > 3g networks similar fake as use of sip in 3g im?
> 
> The question was if anyone was going to deploy and I answered ;-)
> 
> I don't fully agree with the way 3G sets up their AAA. But we 
> are discussing the creation of an IETF standard for SIP AAA. 
> 3G folks had to create a multimedia AAA application because 
> the IETF does not have one. My opinion is that we need an 
> IETF standard in this space, so that the 3G folks, SIP vendor 
> groups, or other parties don't all diverge on how to do this. 
> IMHO, this is really crucial for SIP. My belief is also that 
> the IETF standard will be better than any of these groups 
> could do themselves. As far as 3G goes, my understanding was 
> that they are happy to switch to an IETF standard when it 
> becomes available.
> 
> Jari
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Fri, 21 Feb 2003 17:16:04 +0200
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Hi,

> I doubt any service providers will agree vendors are entitled to speak
> for them, wired or wireless.

I guess Jari was not speaking on behalf of wireless operators, rather He was
just reporting what has been standardized in 3GPP by both vendors and
operators i.e. AAA for IP multimedia services based on Diameter.

Consequently 3G wireless service providers will deploy Diameter.
I guess an IETF standard on SIP-AAA based on Diameter is needed to avoid
divergent solutions across multiple networks, which perhaps end up
in interoperability problems. 

Best Regards
Marco

> -----Original Message-----
> From: ext Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: 21 February, 2003 16:46
> To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> Cc: Loughney John (NRC/Helsinki); sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> >  > Any of the 3G operators count for this? I'm not sure how many
> >  > of the subscribe to this group though, even if their vendors
> >  > do...
> 
> I doubt any service providers will agree vendors are entitled to speak
> for them, wired or wireless. The service providers on this list have
> given solid arguments why we prefer RADIUS at present. As for 
> what comes
> beyond RADIUS, we have the OSP token I-D
> 
> http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> en-04.txt
> 
>  and the IETF work at
> 
> http://www.aaaarch.org/charter.html
> 
> DIAMETER is just too little, too late IMHO, when looking at 
> the options
> in these links.
> 
> Question: Has anyone predetermined in SIPPING that DIAMETER good for
> service providers? Interesting...
> 
> Thanks, Henry
> 
> 
> 
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> > On Behalf Of Jari Arkko
> > Sent: Friday, February 21, 2003 3:04 AM
> > To: jh@lohi.eng.song.fi
> > Cc: 'John Loughney'; sipping@ietf.org
> > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > jh@lohi.eng.song.fi wrote:
> > > Jari Arkko writes:
> > > 
> > >  > Henry Sinnreich wrote:
> > >  > > Is there any service provider planning to use 
> SIP-DIAMETER? If 
> > > yes,  > > please speak up.  >
> > >  > Any of the 3G operators count for this? I'm not sure how many
> > >  > of the subscribe to this group though, even if their vendors
> > >  > do...
> > > 
> > > it depends.  will the vendors really give 3g operators a 
> > free choice 
> > > which diameter server they want to use in their 3g network?  which
> > 
> > I'm not quite sure what you mean by free choice in this 
> > context, but perhaps you are after standardized interfaces? 
> > These would be publicly available in the form of RFCs (eg 
> > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > for multimedia application.
> > 
> > > diameter servers has nokia tested interoperability against? 
> >  does any 
> > > of
> > 
> > I don't know but there's been Diameter interoperability 
> > events. Perhaps you John could dig up the Nokia information 
> > from those if you participated? I know Ericsson particpated 
> at least.
> > 
> > > these servers come from non-3g vendors?  or is the use of 
> > diameter in 
> > > 3g networks similar fake as use of sip in 3g im?
> > 
> > The question was if anyone was going to deploy and I answered ;-)
> > 
> > I don't fully agree with the way 3G sets up their AAA. But we 
> > are discussing the creation of an IETF standard for SIP AAA. 
> > 3G folks had to create a multimedia AAA application because 
> > the IETF does not have one. My opinion is that we need an 
> > IETF standard in this space, so that the 3G folks, SIP vendor 
> > groups, or other parties don't all diverge on how to do this. 
> > IMHO, this is really crucial for SIP. My belief is also that 
> > the IETF standard will be better than any of these groups 
> > could do themselves. As far as 3G goes, my understanding was 
> > that they are happy to switch to an IETF standard when it 
> > becomes available.
> > 
> > Jari
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current 
> > sip Use sip@ietf.org for new developments of core SIP
> > 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 21 Feb 2003 09:36:38 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: <142DC466081D9B409AAD1DDCA4B21E1D9E44C5@esebe016.ntc.nokia.com>
To: marco.stura@nokia.com, jari.arkko@piuha.net, jh@lohi.eng.song.fi
Cc: john.loughney@nokia.com, sipping@ietf.org,
        Henning Schulzrinne <hgs@cs.columbia.edu>
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> Diameter is needed to avoid divergent solutions across 
> multiple networks, which perhaps end up in interoperability problems. 

Yes, DIAMETER will be divergent with the existing RADIUS base serving
100's of millions Internet users. 

That's why we should make

http://www.watersprings.org/pub/id/draft-schulzrinne-sipping-radius-acco
unting-00.txt

 and

http://www.watersprings.org/pub/id/draft-byerly-sip-radius-00.txt

 should be the first WG design items.

Henry

> -----Original Message-----
> From: marco.stura@nokia.com [mailto:marco.stura@nokia.com] 
> Sent: Friday, February 21, 2003 9:16 AM
> To: Henry.Sinnreich@wcom.com; jari.arkko@piuha.net; 
> jh@lohi.eng.song.fi
> Cc: john.loughney@nokia.com; sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Hi,
> 
> > I doubt any service providers will agree vendors are 
> entitled to speak 
> > for them, wired or wireless.
> 
> I guess Jari was not speaking on behalf of wireless 
> operators, rather He was just reporting what has been 
> standardized in 3GPP by both vendors and operators i.e. AAA 
> for IP multimedia services based on Diameter.
> 
> Consequently 3G wireless service providers will deploy 
> Diameter. I guess an IETF standard on SIP-AAA based on 
> Diameter is needed to avoid divergent solutions across 
> multiple networks, which perhaps end up in interoperability problems. 
> 
> Best Regards
> Marco
> 
> > -----Original Message-----
> > From: ext Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > Sent: 21 February, 2003 16:46
> > To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> > Cc: Loughney John (NRC/Helsinki); sipping@ietf.org
> > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > >  > Any of the 3G operators count for this? I'm not sure 
> how many  > 
> > > of the subscribe to this group though, even if their vendors  > 
> > > do...
> > 
> > I doubt any service providers will agree vendors are 
> entitled to speak 
> > for them, wired or wireless. The service providers on this 
> list have 
> > given solid arguments why we prefer RADIUS at present. As for what 
> > comes beyond RADIUS, we have the OSP token I-D
> > 
> > http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> > en-04.txt
> > 
> >  and the IETF work at
> > 
> > http://www.aaaarch.org/charter.html
> > 
> > DIAMETER is just too little, too late IMHO, when looking at
> > the options
> > in these links.
> > 
> > Question: Has anyone predetermined in SIPPING that DIAMETER 
> good for 
> > service providers? Interesting...
> > 
> > Thanks, Henry
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > On Behalf Of Jari Arkko
> > > Sent: Friday, February 21, 2003 3:04 AM
> > > To: jh@lohi.eng.song.fi
> > > Cc: 'John Loughney'; sipping@ietf.org
> > > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > > 
> > > 
> > > jh@lohi.eng.song.fi wrote:
> > > > Jari Arkko writes:
> > > > 
> > > >  > Henry Sinnreich wrote:
> > > >  > > Is there any service provider planning to use
> > SIP-DIAMETER? If
> > > > yes,  > > please speak up.  >
> > > >  > Any of the 3G operators count for this? I'm not sure 
> how many  
> > > > > of the subscribe to this group though, even if their 
> vendors  > 
> > > > do...
> > > > 
> > > > it depends.  will the vendors really give 3g operators a
> > > free choice
> > > > which diameter server they want to use in their 3g 
> network?  which
> > > 
> > > I'm not quite sure what you mean by free choice in this
> > > context, but perhaps you are after standardized interfaces? 
> > > These would be publicly available in the form of RFCs (eg 
> > > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > > for multimedia application.
> > > 
> > > > diameter servers has nokia tested interoperability against?
> > >  does any
> > > > of
> > > 
> > > I don't know but there's been Diameter interoperability
> > > events. Perhaps you John could dig up the Nokia information 
> > > from those if you participated? I know Ericsson particpated 
> > at least.
> > > 
> > > > these servers come from non-3g vendors?  or is the use of
> > > diameter in
> > > > 3g networks similar fake as use of sip in 3g im?
> > > 
> > > The question was if anyone was going to deploy and I answered ;-)
> > > 
> > > I don't fully agree with the way 3G sets up their AAA. But we
> > > are discussing the creation of an IETF standard for SIP AAA. 
> > > 3G folks had to create a multimedia AAA application because 
> > > the IETF does not have one. My opinion is that we need an 
> > > IETF standard in this space, so that the 3G folks, SIP vendor 
> > > groups, or other parties don't all diverge on how to do this. 
> > > IMHO, this is really crucial for SIP. My belief is also that 
> > > the IETF standard will be better than any of these groups 
> > > could do themselves. As far as 3G goes, my understanding was 
> > > that they are happy to switch to an IETF standard when it 
> > > becomes available.
> > > 
> > > Jari
> > > 
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listi> nfo/sipping
> > > This list 
> is for NEW development of the 
> application of SIP Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core SIP
> > > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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> Yes, DIAMETER will be divergent with the existing RADIUS base serving
> 100's of millions Internet users. 

Yes, but RADIUS will be divergent with the wireless environment
currently serving 700's millions of GSM/GPRS users.

That's why we should make Diameter base AAA-SIP solution with
special care to RADIUS interoperability.

Br
Marco

> -----Original Message-----
> From: ext Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: 21 February, 2003 17:37
> To: Stura Marco (NET/Helsinki); jari.arkko@piuha.net;
> jh@lohi.eng.song.fi
> Cc: Loughney John (NRC/Helsinki); sipping@ietf.org; Henning 
> Schulzrinne
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> > Diameter is needed to avoid divergent solutions across 
> > multiple networks, which perhaps end up in interoperability 
> problems. 
> 
> Yes, DIAMETER will be divergent with the existing RADIUS base serving
> 100's of millions Internet users. 
> 
> That's why we should make
> 
> http://www.watersprings.org/pub/id/draft-schulzrinne-sipping-r
> adius-acco
> unting-00.txt
> 
>  and
> 
> http://www.watersprings.org/pub/id/draft-byerly-sip-radius-00.txt
> 
>  should be the first WG design items.
> 
> Henry
> 
> > -----Original Message-----
> > From: marco.stura@nokia.com [mailto:marco.stura@nokia.com] 
> > Sent: Friday, February 21, 2003 9:16 AM
> > To: Henry.Sinnreich@wcom.com; jari.arkko@piuha.net; 
> > jh@lohi.eng.song.fi
> > Cc: john.loughney@nokia.com; sipping@ietf.org
> > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > Hi,
> > 
> > > I doubt any service providers will agree vendors are 
> > entitled to speak 
> > > for them, wired or wireless.
> > 
> > I guess Jari was not speaking on behalf of wireless 
> > operators, rather He was just reporting what has been 
> > standardized in 3GPP by both vendors and operators i.e. AAA 
> > for IP multimedia services based on Diameter.
> > 
> > Consequently 3G wireless service providers will deploy 
> > Diameter. I guess an IETF standard on SIP-AAA based on 
> > Diameter is needed to avoid divergent solutions across 
> > multiple networks, which perhaps end up in interoperability 
> problems. 
> > 
> > Best Regards
> > Marco
> > 
> > > -----Original Message-----
> > > From: ext Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > > Sent: 21 February, 2003 16:46
> > > To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> > > Cc: Loughney John (NRC/Helsinki); sipping@ietf.org
> > > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > > 
> > > 
> > > >  > Any of the 3G operators count for this? I'm not sure 
> > how many  > 
> > > > of the subscribe to this group though, even if their vendors  > 
> > > > do...
> > > 
> > > I doubt any service providers will agree vendors are 
> > entitled to speak 
> > > for them, wired or wireless. The service providers on this 
> > list have 
> > > given solid arguments why we prefer RADIUS at present. As 
> for what 
> > > comes beyond RADIUS, we have the OSP token I-D
> > > 
> > > http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> > > en-04.txt
> > > 
> > >  and the IETF work at
> > > 
> > > http://www.aaaarch.org/charter.html
> > > 
> > > DIAMETER is just too little, too late IMHO, when looking at
> > > the options
> > > in these links.
> > > 
> > > Question: Has anyone predetermined in SIPPING that DIAMETER 
> > good for 
> > > service providers? Interesting...
> > > 
> > > Thanks, Henry
> > > 
> > > 
> > > 
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > On Behalf Of Jari Arkko
> > > > Sent: Friday, February 21, 2003 3:04 AM
> > > > To: jh@lohi.eng.song.fi
> > > > Cc: 'John Loughney'; sipping@ietf.org
> > > > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > > > 
> > > > 
> > > > jh@lohi.eng.song.fi wrote:
> > > > > Jari Arkko writes:
> > > > > 
> > > > >  > Henry Sinnreich wrote:
> > > > >  > > Is there any service provider planning to use
> > > SIP-DIAMETER? If
> > > > > yes,  > > please speak up.  >
> > > > >  > Any of the 3G operators count for this? I'm not sure 
> > how many  
> > > > > > of the subscribe to this group though, even if their 
> > vendors  > 
> > > > > do...
> > > > > 
> > > > > it depends.  will the vendors really give 3g operators a
> > > > free choice
> > > > > which diameter server they want to use in their 3g 
> > network?  which
> > > > 
> > > > I'm not quite sure what you mean by free choice in this
> > > > context, but perhaps you are after standardized interfaces? 
> > > > These would be publicly available in the form of RFCs (eg 
> > > > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > > > for multimedia application.
> > > > 
> > > > > diameter servers has nokia tested interoperability against?
> > > >  does any
> > > > > of
> > > > 
> > > > I don't know but there's been Diameter interoperability
> > > > events. Perhaps you John could dig up the Nokia information 
> > > > from those if you participated? I know Ericsson particpated 
> > > at least.
> > > > 
> > > > > these servers come from non-3g vendors?  or is the use of
> > > > diameter in
> > > > > 3g networks similar fake as use of sip in 3g im?
> > > > 
> > > > The question was if anyone was going to deploy and I 
> answered ;-)
> > > > 
> > > > I don't fully agree with the way 3G sets up their AAA. But we
> > > > are discussing the creation of an IETF standard for SIP AAA. 
> > > > 3G folks had to create a multimedia AAA application because 
> > > > the IETF does not have one. My opinion is that we need an 
> > > > IETF standard in this space, so that the 3G folks, SIP vendor 
> > > > groups, or other parties don't all diverge on how to do this. 
> > > > IMHO, this is really crucial for SIP. My belief is also that 
> > > > the IETF standard will be better than any of these groups 
> > > > could do themselves. As far as 3G goes, my understanding was 
> > > > that they are happy to switch to an IETF standard when it 
> > > > becomes available.
> > > > 
> > > > Jari
> > > > 
> > > > _______________________________________________
> > > > Sipping mailing list  
> > https://www1.ietf.org/mailman/listi> nfo/sipping
> > > > This list 
> > is for NEW development of the 
> > application of SIP Use 
> > > > sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > > > sip@ietf.org for new developments of core SIP
> > > > 
> > > 
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core SIP
> > > 
> > 
> 
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
Cc: sipping@ietf.org
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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Henry, you seem to have reached the point where you have totally confused two questions that ought to be taken independently.

1)	Should IETF specify RADIUS for use with SIP. This is apparently what you want. It is not required by the operators or vendors represented in 3GPP, but nobody from 3GPP is telling you not to do it. 

2)	Should IETF specify DIAMETER for use with SIP. That decision to use DIAMETER in association with SIP has been made in 3GPP in an open forum for all 3GPP interested vendors and operators, and that decision is well known throughout the industry. Those decisions are documented in meeting reports from 3GPP SA2, CN4, SA and CN, and specified in approved 3GPP specifications all of which are publically available. If you wish to change that decision, then by all means join 3GPP and make the appropriate proposal. To the best of my knowledge nobody has suggested that RADIUS is even suitable for use in the 3GPP environment here (despite RADIUS already appearing elsewhere in the 3GPP architecture), and IMS cannot be deployed without these capabilities. Further, while 3GPP release 5 uses proprietary extensions to DIAMETER to provide service codes (these are provided in an informational RFC), 3GPP has undertaken to replace these in release 6 with an IETF worked proposal. If IETF does n!
ot do that work (with appropriate 3GPP support), then 3GPP cannot meet that agreement.

Therefore from a 3GPP perspective the point is that for question 1, we have no position and for question 2, we need the work from IETF in order to complete agreements we have already made with IETF.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: 21 February 2003 14:46
> To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> Cc: 'John Loughney'; sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> >  > Any of the 3G operators count for this? I'm not sure how many
> >  > of the subscribe to this group though, even if their vendors
> >  > do...
> 
> I doubt any service providers will agree vendors are entitled to speak
> for them, wired or wireless. The service providers on this list have
> given solid arguments why we prefer RADIUS at present. As for 
> what comes
> beyond RADIUS, we have the OSP token I-D
> 
> http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> en-04.txt
> 
>  and the IETF work at
> 
> http://www.aaaarch.org/charter.html
> 
> DIAMETER is just too little, too late IMHO, when looking at 
> the options
> in these links.
> 
> Question: Has anyone predetermined in SIPPING that DIAMETER good for
> service providers? Interesting...
> 
> Thanks, Henry
> 
> 
> 
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> > On Behalf Of Jari Arkko
> > Sent: Friday, February 21, 2003 3:04 AM
> > To: jh@lohi.eng.song.fi
> > Cc: 'John Loughney'; sipping@ietf.org
> > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > jh@lohi.eng.song.fi wrote:
> > > Jari Arkko writes:
> > > 
> > >  > Henry Sinnreich wrote:
> > >  > > Is there any service provider planning to use 
> SIP-DIAMETER? If 
> > > yes,  > > please speak up.  >
> > >  > Any of the 3G operators count for this? I'm not sure how many
> > >  > of the subscribe to this group though, even if their vendors
> > >  > do...
> > > 
> > > it depends.  will the vendors really give 3g operators a 
> > free choice 
> > > which diameter server they want to use in their 3g network?  which
> > 
> > I'm not quite sure what you mean by free choice in this 
> > context, but perhaps you are after standardized interfaces? 
> > These would be publicly available in the form of RFCs (eg 
> > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > for multimedia application.
> > 
> > > diameter servers has nokia tested interoperability against? 
> >  does any 
> > > of
> > 
> > I don't know but there's been Diameter interoperability 
> > events. Perhaps you John could dig up the Nokia information 
> > from those if you participated? I know Ericsson particpated 
> at least.
> > 
> > > these servers come from non-3g vendors?  or is the use of 
> > diameter in 
> > > 3g networks similar fake as use of sip in 3g im?
> > 
> > The question was if anyone was going to deploy and I answered ;-)
> > 
> > I don't fully agree with the way 3G sets up their AAA. But we 
> > are discussing the creation of an IETF standard for SIP AAA. 
> > 3G folks had to create a multimedia AAA application because 
> > the IETF does not have one. My opinion is that we need an 
> > IETF standard in this space, so that the 3G folks, SIP vendor 
> > groups, or other parties don't all diverge on how to do this. 
> > IMHO, this is really crucial for SIP. My belief is also that 
> > the IETF standard will be better than any of these groups 
> > could do themselves. As far as 3G goes, my understanding was 
> > that they are happy to switch to an IETF standard when it 
> > becomes available.
> > 
> > Jari
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current 
> > sip Use sip@ietf.org for new developments of core SIP
> > 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Cc: <Henry.Sinnreich@wcom.com>, <jari.arkko@piuha.net>,
        <john.loughney@nokia.com>, <sipping@ietf.org>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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marco.stura@nokia.com writes:

 > Consequently 3G wireless service providers will deploy Diameter.

that is	fine with me although i'm not sure what	the walled garden of
3gpp im has to do with this working group.

 > I guess an IETF standard on SIP-AAA based on Diameter is needed to avoid
 > divergent solutions across multiple networks, which perhaps end up
 > in interoperability problems.

other operators than 3g are already using radius for sip aaa and this
cannot be ignored by this working group.  we and many others are using
radius authentication according to the sterman draft are very happy with
it.

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 21 Feb 2003 10:50:33 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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 <475FF955A05DD411980D00508B6D5FB00439EBFF@en0033exch001u.uk.lucent.com>
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Keith, the confusion stems from the fact: Why if 3GPP has decided on
DIAMETER, this should not be treated in the WG as all the other "p"
labels for SIP used by 3GPP? No problem with what 3GPP folks decide, but
that should be labeled "p" (standing here for proprietary, private and I
bet, also for provisional, since I don't believe 3GPP operetors will not
eventually deploy some form of web services XML etc. for AAA).

As for quoting 3GPP SA2, CN4, SA and CN, we could equally qoute T1, etc.
The world is bigger than the 3GPP universe :-)

Henry

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com] 
> Sent: Friday, February 21, 2003 10:25 AM
> To: 'Henry Sinnreich'
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Henry, you seem to have reached the point where you have 
> totally confused two questions that ought to be taken independently.
> 
> 1)	Should IETF specify RADIUS for use with SIP. This is 
> apparently what you want. It is not required by the operators 
> or vendors represented in 3GPP, but nobody from 3GPP is 
> telling you not to do it. 
> 
> 2)	Should IETF specify DIAMETER for use with SIP. That 
> decision to use DIAMETER in association with SIP has been 
> made in 3GPP in an open forum for all 3GPP interested vendors 
> and operators, and that decision is well known throughout the 
> industry. Those decisions are documented in meeting reports 
> from 3GPP SA2, CN4, SA and CN, and specified in approved 3GPP 
> specifications all of which are publically available. If you 
> wish to change that decision, then by all means join 3GPP and 
> make the appropriate proposal. To the best of my knowledge 
> nobody has suggested that RADIUS is even suitable for use in 
> the 3GPP environment here (despite RADIUS already appearing 
> elsewhere in the 3GPP architecture), and IMS cannot be 
> deployed without these capabilities. Further, while 3GPP 
> release 5 uses proprietary extensions to DIAMETER to provide 
> service codes (these are provided in an informational RFC), 
> 3GPP has undertaken to replace these in release 6 with an 
> IETF worked proposal. If IETF does n!
> ot do that work (with appropriate 3GPP support), then 3GPP 
> cannot meet that agreement.
> 
> Therefore from a 3GPP perspective the point is that for 
> question 1, we have no position and for question 2, we need 
> the work from IETF in order to complete agreements we have 
> already made with IETF.
> 
> Keith
> 
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com 
> 
> > -----Original Message-----
> > From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > Sent: 21 February 2003 14:46
> > To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> > Cc: 'John Loughney'; sipping@ietf.org
> > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > >  > Any of the 3G operators count for this? I'm not sure 
> how many  > 
> > > of the subscribe to this group though, even if their vendors  > 
> > > do...
> > 
> > I doubt any service providers will agree vendors are 
> entitled to speak 
> > for them, wired or wireless. The service providers on this 
> list have 
> > given solid arguments why we prefer RADIUS at present. As for what 
> > comes beyond RADIUS, we have the OSP token I-D
> > 
> > http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> > en-04.txt
> > 
> >  and the IETF work at
> > 
> > http://www.aaaarch.org/charter.html
> > 
> > DIAMETER is just too little, too late IMHO, when looking at
> > the options
> > in these links.
> > 
> > Question: Has anyone predetermined in SIPPING that DIAMETER 
> good for 
> > service providers? Interesting...
> > 
> > Thanks, Henry
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > On Behalf Of Jari Arkko
> > > Sent: Friday, February 21, 2003 3:04 AM
> > > To: jh@lohi.eng.song.fi
> > > Cc: 'John Loughney'; sipping@ietf.org
> > > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > > 
> > > 
> > > jh@lohi.eng.song.fi wrote:
> > > > Jari Arkko writes:
> > > > 
> > > >  > Henry Sinnreich wrote:
> > > >  > > Is there any service provider planning to use
> > SIP-DIAMETER? If
> > > > yes,  > > please speak up.  >
> > > >  > Any of the 3G operators count for this? I'm not sure 
> how many  
> > > > > of the subscribe to this group though, even if their 
> vendors  > 
> > > > do...
> > > > 
> > > > it depends.  will the vendors really give 3g operators a
> > > free choice
> > > > which diameter server they want to use in their 3g 
> network?  which
> > > 
> > > I'm not quite sure what you mean by free choice in this
> > > context, but perhaps you are after standardized interfaces? 
> > > These would be publicly available in the form of RFCs (eg 
> > > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > > for multimedia application.
> > > 
> > > > diameter servers has nokia tested interoperability against?
> > >  does any
> > > > of
> > > 
> > > I don't know but there's been Diameter interoperability
> > > events. Perhaps you John could dig up the Nokia information 
> > > from those if you participated? I know Ericsson particpated 
> > at least.
> > > 
> > > > these servers come from non-3g vendors?  or is the use of
> > > diameter in
> > > > 3g networks similar fake as use of sip in 3g im?
> > > 
> > > The question was if anyone was going to deploy and I answered ;-)
> > > 
> > > I don't fully agree with the way 3G sets up their AAA. But we
> > > are discussing the creation of an IETF standard for SIP AAA. 
> > > 3G folks had to create a multimedia AAA application because 
> > > the IETF does not have one. My opinion is that we need an 
> > > IETF standard in this space, so that the 3G folks, SIP vendor 
> > > groups, or other parties don't all diverge on how to do this. 
> > > IMHO, this is really crucial for SIP. My belief is also that 
> > > the IETF standard will be better than any of these groups 
> > > could do themselves. As far as 3G goes, my understanding was 
> > > that they are happy to switch to an IETF standard when it 
> > > becomes available.
> > > 
> > > Jari
> > > 
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listi> nfo/sipping
> > > This list 
> is for NEW development of the 
> application of SIP Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core SIP
> > > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Message-ID: <15958.31368.133191.629532@lohi.eng.song.fi>
Date: Fri, 21 Feb 2003 21:14:16 +0200
To: sipping@ietf.org
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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just to add to henry's list ofdrafts, this is what many open source and
commercial radius servers and sip proxies support for digest
authentication with sip and which is in use by several sip providers:

ftp://lohi.eng.song.fi/tmp/draft-sterman-aaa-sip-00.txt

(the directory is not listable).

-- juha






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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: [Sipping] is there a way to authenticate bye?
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related to sip aaa requirements, is there a way to figure out that a bye
that terminates a session/dialog was sent by a local user that the proxy
is responsible for (and thus should be authenticated by the proxy)?  if
the matching invite was originated by a non-local user, the to and from
headers may not tell anything about the matter.

-- juha
_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
Cc: sipping@ietf.org
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Fri, 21 Feb 2003 17:48:16 -0000
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If DIAMETER was a 3GPP protocol, then treating the work with the same status as P-headers would be a perfectly valid statement, but DIAMETER is an approved IETF protocol as draft standard currently sitting in the RFC editors queue and therefore is on the same IETF standards track as RADIUS. 

Further, DIAMETER was chosen by 3GPP because RADIUS does not meet the 3GPP requirements for that particular interface. Available implementations comes secondary to those requirements (after all 3GPP have specified RFC 3261 rather than RFC 2543 and there are not many deployed implementations of that either).

What you appear to be saying is that DIAMETER should not have been created in the first place (presumably with a view that RADIUS should have been enhanced to deal with its shortcomings). That is an issue you must take up with the IESG and the WG responsible, not try and blame 3GPP for it.

And finally to reiterate the point of my previous mail, 3GPP is asking IETF to provide DIAMETER support. In my view, that has zero impact on whether RADIUS is done as well. The input that work should not be done on RADIUS is coming from within the IETF, and is nothing to do with 3GPP.

Keith

P.S. I cannot quote T1 yet because the release 5 specifications have not yet been endorsed by the SDOs, of which T1 is one (that stage in the process will occur shortly). And the reason for quoting those groups was not to say that my group is better that yours, it was to make clear that the choice of DIAMETER is well known in 3GPP (and therefore by the nature of the 3G market by all operators deploying 3GPP) by publically available information from those groups.


Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: 21 February 2003 16:51
> To: 'Drage, Keith (Keith)'
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Keith, the confusion stems from the fact: Why if 3GPP has decided on
> DIAMETER, this should not be treated in the WG as all the other "p"
> labels for SIP used by 3GPP? No problem with what 3GPP folks 
> decide, but
> that should be labeled "p" (standing here for proprietary, 
> private and I
> bet, also for provisional, since I don't believe 3GPP 
> operetors will not
> eventually deploy some form of web services XML etc. for AAA).
> 
> As for quoting 3GPP SA2, CN4, SA and CN, we could equally 
> qoute T1, etc.
> The world is bigger than the 3GPP universe :-)
> 
> Henry
> 
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com] 
> > Sent: Friday, February 21, 2003 10:25 AM
> > To: 'Henry Sinnreich'
> > Cc: sipping@ietf.org
> > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > Henry, you seem to have reached the point where you have 
> > totally confused two questions that ought to be taken independently.
> > 
> > 1)	Should IETF specify RADIUS for use with SIP. This is 
> > apparently what you want. It is not required by the operators 
> > or vendors represented in 3GPP, but nobody from 3GPP is 
> > telling you not to do it. 
> > 
> > 2)	Should IETF specify DIAMETER for use with SIP. That 
> > decision to use DIAMETER in association with SIP has been 
> > made in 3GPP in an open forum for all 3GPP interested vendors 
> > and operators, and that decision is well known throughout the 
> > industry. Those decisions are documented in meeting reports 
> > from 3GPP SA2, CN4, SA and CN, and specified in approved 3GPP 
> > specifications all of which are publically available. If you 
> > wish to change that decision, then by all means join 3GPP and 
> > make the appropriate proposal. To the best of my knowledge 
> > nobody has suggested that RADIUS is even suitable for use in 
> > the 3GPP environment here (despite RADIUS already appearing 
> > elsewhere in the 3GPP architecture), and IMS cannot be 
> > deployed without these capabilities. Further, while 3GPP 
> > release 5 uses proprietary extensions to DIAMETER to provide 
> > service codes (these are provided in an informational RFC), 
> > 3GPP has undertaken to replace these in release 6 with an 
> > IETF worked proposal. If IETF does n!
> > ot do that work (with appropriate 3GPP support), then 3GPP 
> > cannot meet that agreement.
> > 
> > Therefore from a 3GPP perspective the point is that for 
> > question 1, we have no position and for question 2, we need 
> > the work from IETF in order to complete agreements we have 
> > already made with IETF.
> > 
> > Keith
> > 
> > Keith Drage
> > Lucent Technologies
> > Tel: +44 1793 776249
> > Email: drage@lucent.com 
> > 
> > > -----Original Message-----
> > > From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > > Sent: 21 February 2003 14:46
> > > To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> > > Cc: 'John Loughney'; sipping@ietf.org
> > > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > > 
> > > 
> > > >  > Any of the 3G operators count for this? I'm not sure 
> > how many  > 
> > > > of the subscribe to this group though, even if their vendors  > 
> > > > do...
> > > 
> > > I doubt any service providers will agree vendors are 
> > entitled to speak 
> > > for them, wired or wireless. The service providers on this 
> > list have 
> > > given solid arguments why we prefer RADIUS at present. As 
> for what 
> > > comes beyond RADIUS, we have the OSP token I-D
> > > 
> > > http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> > > en-04.txt
> > > 
> > >  and the IETF work at
> > > 
> > > http://www.aaaarch.org/charter.html
> > > 
> > > DIAMETER is just too little, too late IMHO, when looking at
> > > the options
> > > in these links.
> > > 
> > > Question: Has anyone predetermined in SIPPING that DIAMETER 
> > good for 
> > > service providers? Interesting...
> > > 
> > > Thanks, Henry
> > > 
> > > 
> > > 
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > On Behalf Of Jari Arkko
> > > > Sent: Friday, February 21, 2003 3:04 AM
> > > > To: jh@lohi.eng.song.fi
> > > > Cc: 'John Loughney'; sipping@ietf.org
> > > > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > > > 
> > > > 
> > > > jh@lohi.eng.song.fi wrote:
> > > > > Jari Arkko writes:
> > > > > 
> > > > >  > Henry Sinnreich wrote:
> > > > >  > > Is there any service provider planning to use
> > > SIP-DIAMETER? If
> > > > > yes,  > > please speak up.  >
> > > > >  > Any of the 3G operators count for this? I'm not sure 
> > how many  
> > > > > > of the subscribe to this group though, even if their 
> > vendors  > 
> > > > > do...
> > > > > 
> > > > > it depends.  will the vendors really give 3g operators a
> > > > free choice
> > > > > which diameter server they want to use in their 3g 
> > network?  which
> > > > 
> > > > I'm not quite sure what you mean by free choice in this
> > > > context, but perhaps you are after standardized interfaces? 
> > > > These would be publicly available in the form of RFCs (eg 
> > > > DIAMETER base) and 3G specifications on how DIAMETER is used 
> > > > for multimedia application.
> > > > 
> > > > > diameter servers has nokia tested interoperability against?
> > > >  does any
> > > > > of
> > > > 
> > > > I don't know but there's been Diameter interoperability
> > > > events. Perhaps you John could dig up the Nokia information 
> > > > from those if you participated? I know Ericsson particpated 
> > > at least.
> > > > 
> > > > > these servers come from non-3g vendors?  or is the use of
> > > > diameter in
> > > > > 3g networks similar fake as use of sip in 3g im?
> > > > 
> > > > The question was if anyone was going to deploy and I 
> answered ;-)
> > > > 
> > > > I don't fully agree with the way 3G sets up their AAA. But we
> > > > are discussing the creation of an IETF standard for SIP AAA. 
> > > > 3G folks had to create a multimedia AAA application because 
> > > > the IETF does not have one. My opinion is that we need an 
> > > > IETF standard in this space, so that the 3G folks, SIP vendor 
> > > > groups, or other parties don't all diverge on how to do this. 
> > > > IMHO, this is really crucial for SIP. My belief is also that 
> > > > the IETF standard will be better than any of these groups 
> > > > could do themselves. As far as 3G goes, my understanding was 
> > > > that they are happy to switch to an IETF standard when it 
> > > > becomes available.
> > > > 
> > > > Jari
> > > > 
> > > > _______________________________________________
> > > > Sipping mailing list  
> > https://www1.ietf.org/mailman/listi> nfo/sipping
> > > > This list 
> > is for NEW development of the 
> > application of SIP Use 
> > > > sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > > > sip@ietf.org for new developments of core SIP
> > > > 
> > > 
> > > _______________________________________________
> > > Sipping mailing list  
https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 21 Feb 2003 12:02:34 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
In-reply-to: 
 <475FF955A05DD411980D00508B6D5FB00439EC00@en0033exch001u.uk.lucent.com>
To: "'Drage, Keith (Keith)'" <drage@lucent.com>
Cc: sipping@ietf.org
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> What you appear to be saying is that DIAMETER should not have 
> been created in the first place (presumably with a view that 
> RADIUS should have been enhanced to deal with its 
> shortcomings). That is an issue you must take up with the 
> IESG and the WG responsible, not try and blame 3GPP for it.

On this we can agree. 3GPP is not the problem for us. DIAMETER is.

Henry

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com] 
> Sent: Friday, February 21, 2003 11:48 AM
> To: 'Henry Sinnreich'
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> If DIAMETER was a 3GPP protocol, then treating the work with 
> the same status as P-headers would be a perfectly valid 
> statement, but DIAMETER is an approved IETF protocol as draft 
> standard currently sitting in the RFC editors queue and 
> therefore is on the same IETF standards track as RADIUS. 
> 
> Further, DIAMETER was chosen by 3GPP because RADIUS does not 
> meet the 3GPP requirements for that particular interface. 
> Available implementations comes secondary to those 
> requirements (after all 3GPP have specified RFC 3261 rather 
> than RFC 2543 and there are not many deployed implementations 
> of that either).
> 
> What you appear to be saying is that DIAMETER should not have 
> been created in the first place (presumably with a view that 
> RADIUS should have been enhanced to deal with its 
> shortcomings). That is an issue you must take up with the 
> IESG and the WG responsible, not try and blame 3GPP for it.
> 
> And finally to reiterate the point of my previous mail, 3GPP 
> is asking IETF to provide DIAMETER support. In my view, that 
> has zero impact on whether RADIUS is done as well. The input 
> that work should not be done on RADIUS is coming from within 
> the IETF, and is nothing to do with 3GPP.
> 
> Keith
> 
> P.S. I cannot quote T1 yet because the release 5 
> specifications have not yet been endorsed by the SDOs, of 
> which T1 is one (that stage in the process will occur 
> shortly). And the reason for quoting those groups was not to 
> say that my group is better that yours, it was to make clear 
> that the choice of DIAMETER is well known in 3GPP (and 
> therefore by the nature of the 3G market by all operators 
> deploying 3GPP) by publically available information from those groups.
> 
> 
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com 
> 
> > -----Original Message-----
> > From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > Sent: 21 February 2003 16:51
> > To: 'Drage, Keith (Keith)'
> > Cc: sipping@ietf.org
> > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > 
> > 
> > Keith, the confusion stems from the fact: Why if 3GPP has 
> decided on 
> > DIAMETER, this should not be treated in the WG as all the other "p" 
> > labels for SIP used by 3GPP? No problem with what 3GPP 
> folks decide, 
> > but that should be labeled "p" (standing here for proprietary,
> > private and I
> > bet, also for provisional, since I don't believe 3GPP 
> > operetors will not
> > eventually deploy some form of web services XML etc. for AAA).
> > 
> > As for quoting 3GPP SA2, CN4, SA and CN, we could equally
> > qoute T1, etc.
> > The world is bigger than the 3GPP universe :-)
> > 
> > Henry
> > 
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Friday, February 21, 2003 10:25 AM
> > > To: 'Henry Sinnreich'
> > > Cc: sipping@ietf.org
> > > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > > 
> > > 
> > > Henry, you seem to have reached the point where you have
> > > totally confused two questions that ought to be taken 
> independently.
> > > 
> > > 1)	Should IETF specify RADIUS for use with SIP. This is 
> > > apparently what you want. It is not required by the operators
> > > or vendors represented in 3GPP, but nobody from 3GPP is 
> > > telling you not to do it. 
> > > 
> > > 2)	Should IETF specify DIAMETER for use with SIP. That 
> > > decision to use DIAMETER in association with SIP has been
> > > made in 3GPP in an open forum for all 3GPP interested vendors 
> > > and operators, and that decision is well known throughout the 
> > > industry. Those decisions are documented in meeting reports 
> > > from 3GPP SA2, CN4, SA and CN, and specified in approved 3GPP 
> > > specifications all of which are publically available. If you 
> > > wish to change that decision, then by all means join 3GPP and 
> > > make the appropriate proposal. To the best of my knowledge 
> > > nobody has suggested that RADIUS is even suitable for use in 
> > > the 3GPP environment here (despite RADIUS already appearing 
> > > elsewhere in the 3GPP architecture), and IMS cannot be 
> > > deployed without these capabilities. Further, while 3GPP 
> > > release 5 uses proprietary extensions to DIAMETER to provide 
> > > service codes (these are provided in an informational RFC), 
> > > 3GPP has undertaken to replace these in release 6 with an 
> > > IETF worked proposal. If IETF does n!
> > > ot do that work (with appropriate 3GPP support), then 3GPP 
> > > cannot meet that agreement.
> > > 
> > > Therefore from a 3GPP perspective the point is that for
> > > question 1, we have no position and for question 2, we need 
> > > the work from IETF in order to complete agreements we have 
> > > already made with IETF.
> > > 
> > > Keith
> > > 
> > > Keith Drage
> > > Lucent Technologies
> > > Tel: +44 1793 776249
> > > Email: drage@lucent.com
> > > 
> > > > -----Original Message-----
> > > > From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > > > Sent: 21 February 2003 14:46
> > > > To: jari.arkko@piuha.net; jh@lohi.eng.song.fi
> > > > Cc: 'John Loughney'; sipping@ietf.org
> > > > Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> > > > 
> > > > 
> > > > >  > Any of the 3G operators count for this? I'm not sure
> > > how many  >
> > > > > of the subscribe to this group though, even if their 
> vendors  >
> > > > > do...
> > > > 
> > > > I doubt any service providers will agree vendors are
> > > entitled to speak
> > > > for them, wired or wireless. The service providers on this
> > > list have
> > > > given solid arguments why we prefer RADIUS at present. As
> > for what
> > > > comes beyond RADIUS, we have the OSP token I-D
> > > > 
> > > > http://www.ietf.org/internet-drafts/draft-johnston-sip-osp-tok
> > > > en-04.txt
> > > > 
> > > >  and the IETF work at
> > > > 
> > > > http://www.aaaarch.org/charter.html
> > > > 
> > > > DIAMETER is just too little, too late IMHO, when looking at the 
> > > > options in these links.
> > > > 
> > > > Question: Has anyone predetermined in SIPPING that DIAMETER
> > > good for
> > > > service providers? Interesting...
> > > > 
> > > > Thanks, Henry
> > > > 
> > > > 
> > > > 
> > > > > -----Original Message-----
> > > > > From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org] On 
> > > > > Behalf Of Jari Arkko
> > > > > Sent: Friday, February 21, 2003 3:04 AM
> > > > > To: jh@lohi.eng.song.fi
> > > > > Cc: 'John Loughney'; sipping@ietf.org
> > > > > Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> > > > > 
> > > > > 
> > > > > jh@lohi.eng.song.fi wrote:
> > > > > > Jari Arkko writes:
> > > > > > 
> > > > > >  > Henry Sinnreich wrote:
> > > > > >  > > Is there any service provider planning to use
> > > > SIP-DIAMETER? If
> > > > > > yes,  > > please speak up.  >
> > > > > >  > Any of the 3G operators count for this? I'm not sure
> > > how many
> > > > > > > of the subscribe to this group though, even if their
> > > vendors  >
> > > > > > do...
> > > > > > 
> > > > > > it depends.  will the vendors really give 3g operators a
> > > > > free choice
> > > > > > which diameter server they want to use in their 3g
> > > network?  which
> > > > > 
> > > > > I'm not quite sure what you mean by free choice in 
> this context, 
> > > > > but perhaps you are after standardized interfaces? 
> These would 
> > > > > be publicly available in the form of RFCs (eg 
> DIAMETER base) and 
> > > > > 3G specifications on how DIAMETER is used for multimedia 
> > > > > application.
> > > > > 
> > > > > > diameter servers has nokia tested interoperability against?
> > > > >  does any
> > > > > > of
> > > > > 
> > > > > I don't know but there's been Diameter 
> interoperability events. 
> > > > > Perhaps you John could dig up the Nokia information 
> from those 
> > > > > if you participated? I know Ericsson particpated
> > > > at least.
> > > > > 
> > > > > > these servers come from non-3g vendors?  or is the use of
> > > > > diameter in
> > > > > > 3g networks similar fake as use of sip in 3g im?
> > > > > 
> > > > > The question was if anyone was going to deploy and I
> > answered ;-)
> > > > > 
> > > > > I don't fully agree with the way 3G sets up their AAA. But we 
> > > > > are discussing the creation of an IETF standard for 
> SIP AAA. 3G 
> > > > > folks had to create a multimedia AAA application because the 
> > > > > IETF does not have one. My opinion is that we need an IETF 
> > > > > standard in this space, so that the 3G folks, SIP 
> vendor groups, 
> > > > > or other parties don't all diverge on how to do this. 
> IMHO, this 
> > > > > is really crucial for SIP. My belief is also that the IETF 
> > > > > standard will be better than any of these groups could do 
> > > > > themselves. As far as 3G goes, my understanding was that they 
> > > > > are happy to switch to an IETF standard when it becomes 
> > > > > available.
> > > > > 
> > > > > Jari
> > > > > 
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > > https://www1.ietf.org/mailman/listi> nfo/sipping
> > > > > This list
> > > is for NEW development of the
> > > application of SIP Use 
> > > > > sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > > > sip@ietf.org for new developments of core SIP
> > > > > 
> > > > 
> > > > _______________________________________________
> > > > Sipping mailing list
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sip@ietf.org for new developments of core SIP
> > > 
> > 
> 

_______________________________________________
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Henry,

> > What you appear to be saying is that DIAMETER should not have 
> > been created in the first place (presumably with a view that 
> > RADIUS should have been enhanced to deal with its 
> > shortcomings). That is an issue you must take up with the 
> > IESG and the WG responsible, not try and blame 3GPP for it.
> 
> On this we can agree. 3GPP is not the problem for us. DIAMETER is.

Do you have technical reasons for this?  Diameter running in 
RADIUS compatability mode should allow NAS's and RADIUS servers
to talk to Diameter servers.

I understand your reasons for not wanting to upgrade everything
to Diameter, but I still don't see what reasons you have for
being dismissive of Diameter.

br,
John
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This list is for NEW development of the application of SIP
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Hi all,

So, it seems that there are the following needs:

1) Compatability with existing RADIUS infrastructure.

2) Support for Diameter for the 3G community
  (I believe both 3GPP and 3GPP2 will be using Diameter).

3) Some vague handwaving about Web Services.

It seems that there are some expired drafts on using
RADIUS for at least 2 of the AA's in AAA.  Expired
drafts don't usually count for much, however.

A way forward would be to have the authors of the expired
drafts to update them and re-submit them.  A solution
sufficient for 3G is in the works, so I have heard.
I have seen no concrete proposals for web services, so 
until I see one in a draft, I am going to discount it
entirely.

So, after all of the possible solutions are re-issued in
draft form, probably the SIPPING WG chairs, AAA WG chairs
and relevent IESG members could sort out how to go forward,
perhaps forming a small team to make recommendations on
how to procede.

I would like to say that at present, the SIP-AAA interface
is an unstandardized interface, so no matter what the
final protoocl solution is, everyone want to support the
standard will need to implement it from scatch, so backward
compatability may not be a huge barrier, as long as there
is some way to interwork with existing RADIUS servers and
NASes.

br,
John
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From: "Ejzak, Richard P (Richard)" <ejzak@lucent.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        sipping
	 <sipping@ietf.org>
Subject: RE: [Sipping] New version of the early media draft
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Gonzalo,
It seems the single greatest difficulty with the gateway model (as you point out in your draft) is the difficulty of depending on the arrival of media packets to determine whether or not to generate local ringing.  I have previously (last summer) suggested a solution to this (use 180 exclusively to indicate that local ringing should override any inband media and 183 otherwise) that is unacceptable since it does not satisfy the requirement that 180 be used to indicate alerting of the called party.

Have you considered using the Alert-Info header in the following way? Since the real problem is that we would like to be able to discriminate between cases when 180 should cause local ringing and when 180 should not (since inband media is available), and it is the UAS that knows conclusively which case applies, let's agree on an encoding of the Alert-Info header that indicates that local ringing should not be applied.  The following line is a syntactically valid header:

Alert-Info: <data:,>

This header identifies the Alert-Info as the null string, which seems unambiguously and efficiently interpretable within a 180 response as "do not play local ringing even though the called party is being alerted."  If the UAC does not recognize it or chooses not to interpret it for security reasons, it can be safely ignored.

This modest addition to your draft (or an equivalent mechanism) would make it unnecessary for the gateway model to depend on local policy or incoming media packet detection and is completely consistent with all of the constraints you have documented.

Another approach may be to actually specify the rtp stream as the source of the alert info (although this would require the definition of a new URI scheme type).  It might look like:

Alert-Info: <rtp:/12.34.56.78:90>

where the Alert-Info specifies the address and port of the rtp stream.  I don't like this mechanism since it is redundant with SDP and prone to error.  It is also ambiguous in the event of any subsequent media change using UPDATE (Do I begin to generate local ringing if the rtp stream source address changes?  Or do I continue to listen to the address specified in the Alert-Info and ignore any UPDATEs?  No to both.)

May I suggest including such a mechanism (preferably the first one) in your draft.

Richard Ejzak
Lucent Technologies

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
Sent: Monday, February 10, 2003 5:41 AM
To: sipping
Subject: [Sipping] New version of the early media draft


Folks,

I have just sumitted a new version of the early media draft. Until it
appears in the archives, you can fetch it from:

http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-early-media-01.txt

I have included all the (long and useful) discussions we had in the
mailing list. Therefore, the draft now keeps separate the "gateway" and
the "application server" models. I have also removed the parts we did
not agree on.

Feedback is, as usual, appreciated.

Thanks,

Gonzalo
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Thank you John for your summary. Let's stop the previous
thread and continue with the technical discussion -- I
think it was a fair question to ask who is going to deploy
what, but it doesn't sound very productive to start arguing
the details on who can speak for who, which deployment
examples should not be counted, specific other SDO plans, etc.

But back to the technical issues. I think the *needs* are
discussed in the requirements document (which I think could
still be improved *). The main solutions that have been proposed
are RADIUS, DIAMETER, and Web Services. These solutions
appear to be targeted at slightly different purposes, however,
and I suspect this is the root cause of the argument. Do
we just want a quick scheme to transport Digest authentication
from a SIP server to a central authentication box? If yes,
its very easy to explain why RADIUS looks so promising (even
if there are folks who don't its security).

Or do we want more, such as reliable accounting, ability for a
server to perform "user de-authorization", ability to carry many
pieces of information in standardized attributes, etc. (I think we
want these things). RADIUS doesn't have these features and
extending it to do them would result in something resembling
DIAMETER so if you want any of this DIAMETER looks like
a better solution.

By the way, I continue to wonder why we need to have
such a controversial debate when I suspect we might be able to
cater to the needs of both groups. One possible idea is that
the digest-carrying AAA protocol extension would be defined
using RADIUS attributes in a manner which makes it possible
run it over either basic RADIUS or DIAMETER NASREQ **). The
remaining AAA functionality for SIP would be covered only
in DIAMETER commands and AVPs, mainly because I suspect that
it can't be done in RADIUS. Is this a potential compromise?

Jari

*) One thing that wasn't clear from the requirements
specification was whether we expect to deal just with
SIP proxies, or do we have support also end systems, such
as some sort of a media server which not just runs SIP
but also offers some function?

**) Details to be worked out. One problem is that 3G
folks might actually want to get session keys back from the
authentication server as their auth methods generate these.
Perhaps this part could only be supported in DIAMETER.
RADIUS has key transport as well, but based on Microsoft's
vendor specific attributes, which are pretty outdated at
this point.

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Date: Fri, 21 Feb 2003 14:43:45 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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> I understand your reasons for not wanting to upgrade
> everything to Diameter, but I still don't see what reasons 
> you have for being dismissive of Diameter.

Thanks for clarifying the fact that we have two issues. 

The 2nd item, why DIAMETER is not the best option for NG: It is too
little, too late.

AAA is a natural part of e-commerce, that is of the web services
architecture, as defined in W3C and numerous industry consortiums. It
makes little sense for all the world to make payments and conduct
transactions in the 'web services currency' and only phone calls to be
paid in 'DIAMETER currency'. Actually two types of phones call
'currencies': Wired using RADIUS and mobile using DIAMETER. Not that
DIAMETER is inherently evil or poorly engineered. It is just too little
too late, when looking beyond RADIUS and its limitations. My two cents.

I would see the design jobs therefore this way:

1-Near term: Use RADIUS-SIP to accommodate the huge isntalled base. The
I-D's mentioned here. Including the OSP token for SIP.

2-Longer-term: The Web services architecture.

It appears the present discussion stems from the fact that we only have
volonteers for a DIAMETER-SIP draft. None of the mentioned RADIUS-SIP
authors have volunteered any work here, and this is the problem. So
unless the no RADIUS or web services volonteer steps up, we will have
neither 1 nor 2.

And this is not any fault of 3GPP :-)

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> On Behalf Of john.loughney@nokia.com
> Sent: Friday, February 21, 2003 12:14 PM
> To: Henry.Sinnreich@wcom.com; drage@lucent.com
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Henry,
> 
> > > What you appear to be saying is that DIAMETER should not have been

> > > created in the first place (presumably with a view that RADIUS 
> > > should have been enhanced to deal with its shortcomings). That is 
> > > an issue you must take up with the IESG and the WG responsible, 
> > > not try and blame 3GPP for it.
> > 
> > On this we can agree. 3GPP is not the problem for us. DIAMETER is.
> 
> Do you have technical reasons for this?  Diameter running in
> RADIUS compatability mode should allow NAS's and RADIUS 
> servers to talk to Diameter servers.
> 
> I understand your reasons for not wanting to upgrade
> everything to Diameter, but I still don't see what reasons 
> you have for being dismissive of Diameter.
> 
> br,
> John
> _______________________________________________
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> This list is for NEW development of the application of SIP Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sip@ietf.org for new developments of core SIP
> 

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Date: Fri, 21 Feb 2003 15:55:55 -0500
To: Allison Mankin <mankin@psg.com>, sipping@ietf.org
From: Stu Jacobs <Stu.Jacobs@Verizon.com>
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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Allison,

This material is part of ongoing efforts of Verizon and Verizon management 
to engage in thoughtful considerations of the fundamental changes and 
challenges facing the telecommunications industry. To meet its fiduciary 
responsibilities, management must explore all alternatives, even those that 
may appear highly speculative and hypothetical. Statements and 
representations contained herein are preliminary and/or tentative and 
should not be relied on unless approved by the appropriate Verizon 
governing body.  The following represent the anticipated approach Verizon 
most likely will use.  I cannot be more firm at this time as our 
architectural plans are not finalized.

Verizon's current direction for our Next Generation Network (NGN) VoIP 
Services will likely use a challenge/response approach (ala. Radius) for 
authentication and access control between our SIP service customers and SIP 
Proxies.  All our SIP service customers will probably go through SIP 
Proxies.  We will not support SIP UA to SIP UA within our operational 
domain.  Our SIP Proxies will likely authenticate themselves to other SIP 
Proxies and other back-office systems (OSSs, Application Servers, EMSs, 
etc.) by use of IPsec, specifically IKE combined with X.509v3 server 
certificates and ESP transport mode Null-encryption using 
HMAC-SHA1-96.  Access controls (rights/policies) between our intermediate 
systems (incl. SIP Proxies, Gateways , etc.) are expected to be based on 
use of policy certificates.  All policy and identity certificates are 
expected to be distributed via a redundant hierarchy of LDAP servers 
(repositories).  Accounting data will most likely be compiled by our SIP 
Proxies and transferred to billing systems with authentication provided by 
IPsec.  The actual billing data will probably be captured in the form of 
XML in conjunction with XML signatures (as defined by the W3C).

Stu

At 2/20/03 05:24 PM, you wrote:
>Stu,
>
>You are not doing the usage accounting with Radius, then.
>
>It is only doing the remote access authentication?
>
>We can work with this scenario.  Could you speak to this in
>the open mailing list?  I would like to rule out Radius for
>call detail and usage-based accounting, but it is ok for the
>other A's.
>
>Allison
>
> > Diameter in our Next Generation Network (NGN) VoIP Services.  We will be
> > doing our usage accounting within SIP Proxies and intend to transfer said
> > accounting "call detail records" back to our billing OSSes via other
> > protocols with IPsec providing authentication and confidentiality for 
> these
> > transfers.  We currently do not plan on any deployment of Diameter within
> > Verizon's wireline NGN architecture.  We will probably continue to use
> > Radius for remote access authentication at the application layer for a
> > number or years into the future.
> >
> > Stu Jacobs
> >
> > At 2/14/03 09:58 PM, you wrote:
> >
> > >Jiri,
> > >
> > >It always seems good to be able to stay with installed base, but
> > >there's a factor against our supporting RADIUS drafts for SIP's
> > >accounting - safety.  Even though there is deployment, there are
> > >some extremely bad cases of damage in production, and the IETF
> > >has concluded that RADIUS is not suitable for usage-based accounting
> > >(RFC2975).
> > >
> > >Unless there is no intention of having in SIP accounting usage-based,
> > >monetary, auditable applications, RADIUS is not the solution.  The
> > >protocol just does not preserve a complete stream of reliable
> > >accounting records that are auditable on any level, especially with
> > >any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
> > >the use of RADIUS may even be illegal in situations where there is an
> > >authority over the billing.  Records are too easily lost by the
> > >protocol. This is stated clearly by RFC2975.  This is the reason why
> > >the Operations and Mgt Area developed Diameter.  On these technical*
> > >grounds, that the RADIUS protocol does not have enough reliability,
> > >the work must be based on the Diameter protocol, which is now
> > >approved.  Alternatively, work could be done that does not include
> > >usage-based accounting, but this does not seem to be what is
> > >envisioned in the SIPPING AAA requirements.
> > >
> > >Allison
> > >
> > >P.S. There's a current investigation of RADIUS compatibility mode
> > >in Diameter for this - more needs to be thought about it.
> > > >
> > > > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > > > >Hi Henry,
> > > > >
> > > > >> >but one option is develop a solution using
> > > > >> > Diameter in RADIUS-compatibility mode.
> > > > >>
> > > > >> OK, this is better than nothing. We'll take what we can :-)
> > > > >>
> > > > >> As mentioned, the installed base is RADIUS, no matter how
> > > > >> voluminous the DIAMETER literature may be.
> > > > >
> > > > >I think the issues is that the IETF should recommend what is the
> > > > >right thing to do & why.  If people choose to do something else,
> > > > >well, that is life.
> > > >
> > > > Remember that running code does not necessarily mean something known
> > > > to compile -- that is something known to be deployed.
> > > >
> > > > >I understand what you are asking but I do note that some
> > > > >people have thought that the potential for AAA is much greater
> > > > >than the existing installed base.
> > > >
> > > > I actually think that the deployment argument is a very important one.
> > > >
> > > > -Jiri
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > >_______________________________________________
> > >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > >This list is for NEW development of the application of SIP
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sip@ietf.org for new developments of core SIP
> >
> > ==========================
> > Stuart Jacobs CISSP
> > PMTS - Sr. Technologist
> > Verizon Laboratories
> > 40 Sylvan Road Waltham, MA 02451-1128     USA
> > telephone: (781) 466-3076   fax: (781) 466-2838
> > stu.jacobs@labs.gte.com sjj0@labs.gte.com  stu.jacobs@verizon.com
> > ==========================
> >
==========================
Stuart Jacobs CISSP
PMTS - Sr. Technologist
Verizon Laboratories
40 Sylvan Road Waltham, MA 02451-1128   USA
telephone: (781) 466-3076 fax: (781) 466-2838
stu.jacobs@labs.gte.com sjj0@labs.gte.com stu.jacobs@verizon.com
==========================

==========================
Stuart Jacobs CISSP
PMTS - Sr. Technologist
Verizon Laboratories
40 Sylvan Road Waltham, MA 02451-1128     USA
telephone: (781) 466-3076   fax: (781) 466-2838
stu.jacobs@labs.gte.com sjj0@labs.gte.com  stu.jacobs@verizon.com
==========================

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Subject: [Sipping] Attempted summary of SIP-AAA discussion
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john.loughney@nokia.com writes:

 > It seems that there are some expired drafts on using
 > RADIUS for at least 2 of the AA's in AAA.  Expired
 > drafts don't usually count for much, however.

what counts is that there are several implementations of the sterman
draft (both servers and proxies), which are in active use by sip service
providers.  

 > A way forward would be to have the authors of the expired
 > drafts to update them and re-submit them.  

i have asked the athors of the sterman draft to re-submit it, but for
some reason they have not done so.  perhaps running code is enough for
them.

 > I would like to say that at present, the SIP-AAA interface
 > is an unstandardized interface, so no matter what the
 > final protoocl solution is, everyone want to support the
 > standard will need to implement it from scatch, so backward
 > compatability may not be a huge barrier, as long as there
 > is some way to interwork with existing RADIUS servers and
 > NASes.

if sterman draft becomes an rfc, there is no need to re-implement
anything for digest authentication.  i agree that accounting requires
more work.

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: john.loughney@nokia.com
Cc: <Henry.Sinnreich@wcom.com>, <drage@lucent.com>, <sipping@ietf.org>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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john.loughney@nokia.com writes:

 > I understand your reasons for not wanting to upgrade everything
 > to Diameter, but I still don't see what reasons you have for
 > being dismissive of Diameter.

henry may have points why he doesn't like diameter, but for me, it is
nto really that big deal IF diameter AAA infrastructure some day becomes
wide spread.  today, however, that is not the case.  instead everybody
is using radius as the protocol between NASes (includifn sip proxies)
and AAA servers and also between AAA servers themselves.

examples of the latter are gric and ipass, which provide roaming service
for hundreds of isps literally all over the world.  if radius is so bad
that what you say IESG claims, i don't understand how this is possible.
the roaming services run radius over ipsec which solves the security
issues.

-- juha


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From: "Shiva" <shaj@cisco.com>
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Date: Sat, 22 Feb 2003 12:27:46 +0530
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Hello,
 
Overall the draft looks good.  Here are some of my comments on this
draft
 
The idea of having a unique URI for each conference is a good one.
However a more scalable approach would be to have a master Focus or a
group of them that register with a standard URI. This way users use a
standard URI with a Conference ID as the payload in the Invite message.
This is directed to the master Focus that does the following
* Retrieve the Conference ID
* Check if a Focus corresponding to the conference exists
* If the Conference specific Focus  does not exist create a new Focus
with a unique URI derived from the Conference ID
* Forward the call using the standard Refer mechanism to the new Focus
with the Unique URI.
 
The advantage of doing this is that I could have multiple Conference
servers that are running that are registered their master focus. 
 
We can implement Load Balancing across Conference servers without the UA
realizing it as it is not connecting to a specific Focus on a specific
server. A second advantage is that if in the course of a meeting if the
focus were to go down or the server went down the participants can
attempt to rejoin the call. In such a case the Master Focus would see
that the Conference specific focus is down and recreate it allowing
users to quickly rejoin the conference.
 
Of course one needs to work out the details of the communication and
keepalive between the Master Focus and the Conference focus it creates.
 
The other advantage is if there are global Conference policies these can
be applied at the master and the Individual Conference focus needs to
worry about the specifics of the conference it is dealing with.
 
The other comment was in the area of the Conference policy control
protocol. Is this something like the SDP which is carried as a payload
in SIP or does it have specifics of it's own. Currently most of the
operations that are part of the CPCP are also possible through SIP based
mechanisms. Is there a need to define two parallel mechanisms within the
same framework. 
 
 
Instead of dealing with Conference aware UA's and Conference Unaware
UA's we could assume all participants in the Conference to be Conference
Aware. The Conference Unaware participants are routed through a Gateway
(e.g. an IVR ) that provides a primitive interface to the conference
unaware participants and a Conference aware interface to the Conference
itself.  This would keep the model simpler and it can allow Conference
aware UAs a way to control the conference.
 
Regards
Shiva
Cisco Systems
 
 
 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1141" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D937210006-22022003>Hello,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D937210006-22022003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT><SPAN class=3D937210006-22022003>
<DIV><FONT face=3DArial><FONT size=3D2><SPAN =
class=3D937210006-22022003>Overall the=20
draft looks good.&nbsp; </SPAN>Here are some of my comments<SPAN=20
class=3D937210006-22022003> </SPAN>on this draft</FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></SPAN></FONT>
<DIV><SPAN class=3D937210006-22022003>The idea of having a unique URI =
for each=20
conference is a good one. However a more scalable approach would be to=20
have&nbsp;a master Focus or a group of them&nbsp;that register with a =
standard=20
URI. This way users&nbsp;use a standard URI with a Conference ID as the =
payload=20
in the&nbsp;Invite message. &nbsp;This is directed to the =
master&nbsp;Focus=20
that&nbsp;does the following</SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>* =
Retrieve the=20
Conference ID</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>* =
Check if&nbsp;a=20
Focus corresponding to the conference exists</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>* If =
the Conference=20
specific Focus&nbsp;&nbsp;does not exist c</FONT></SPAN><SPAN=20
class=3D937210006-22022003>reate a new Focus with&nbsp;a unique =
URI&nbsp;derived=20
from the Conference ID</SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003>* Forward the call using the =
standard Refer=20
mechanism to the new Focus with the Unique URI.</SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>The =
advantage of=20
doing this is that I could have multiple Conference servers that are =
running=20
that are registered their master focus. </FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>We can =
implement=20
Load Balancing across Conference servers without the UA realizing =
it&nbsp;as it=20
is not&nbsp;connecting to a specific Focus on a specific server. A =
second=20
advantage is that if in the course of a meeting if the focus were to go =
down or=20
the server went down the participants can attempt to rejoin the call. In =
such a=20
case the Master Focus would see that the Conference specific focus is =
down and=20
recreate it allowing users to quickly rejoin the =
conference.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>Of =
course one needs=20
to work out the details of the communication and keepalive between the =
Master=20
Focus and the Conference focus it creates.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>The =
other advantage=20
is if there are global Conference policies these can be applied at the =
master=20
and the Individual Conference focus needs to worry about the specifics =
of the=20
conference it is dealing with.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>The =
other comment=20
was in the area of the Conference policy control protocol. Is this =
something=20
like the SDP which is carried as a payload in SIP or does it have =
specifics of=20
it's own. Currently most of the operations that are part of the CPCP are =
also=20
possible through SIP based mechanisms. Is there a need to define two =
parallel=20
mechanisms within the same framework. </FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial =
size=3D2></FONT></SPAN><SPAN=20
class=3D937210006-22022003><FONT face=3DArial =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial =
size=3D2>Instead of dealing=20
with Conference aware UA's and Conference Unaware UA's we could assume =
all=20
participants in the Conference to be Conference Aware. The Conference =
Unaware=20
participants are routed through a Gateway (e.g.&nbsp;an IVR )&nbsp;that =
provides=20
a primitive interface to the conference unaware participants and a =
Conference=20
aware interface to the Conference itself.&nbsp; This would keep the =
model=20
simpler and it can allow Conference aware UAs a way to control the=20
conference.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2>Shiva</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial size=3D2>Cisco=20
Systems</FONT></SPAN></DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937210006-22022003><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV></DIV></BODY></HTML>

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jh@lohi.eng.song.fi wrote:
> john.loughney@nokia.com writes:
> 
>  > I understand your reasons for not wanting to upgrade everything
>  > to Diameter, but I still don't see what reasons you have for
>  > being dismissive of Diameter.
> 
> henry may have points why he doesn't like diameter, but for me, it is
> nto really that big deal IF diameter AAA infrastructure some day becomes
> wide spread.  today, however, that is not the case.  instead everybody
> is using radius as the protocol between NASes (includifn sip proxies)
> and AAA servers and also between AAA servers themselves.

Right. That's why DIAMETER has been designed to be able to work
with RADIUS. Naturally new features in DIAMETER are not necessarily
accessible in RADIUS, but things that run on RADIUS should also run
on DIAMETER, such as current NAS or AP authentication.

(This is by the way one reason why I don't believe web services
will be successful in this space; I see the RADIUS - DIAMETER line
continuing for network access, telephony even if some other apps do
something else.)

> examples of the latter are gric and ipass, which provide roaming service
> for hundreds of isps literally all over the world.  if radius is so bad
> that what you say IESG claims, i don't understand how this is possible.
> the roaming services run radius over ipsec which solves the security
> issues.

The limitations of RADIUS have been documented in various places.
Running IPsec does actually help in some of the security-related
limitations, though not all. Everyone knows RADIUS has problems,
but it was the only thing people had.

To recap: Let's plan for a solution that provides the functions
required in the requirements document (using DIAMETER), but
allows for some amount of compatibility with RADIUS.

Jari

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Hi Henry,

> Thanks for clarifying the fact that we have two issues. 
> 
> The 2nd item, why DIAMETER is not the best option for NG: It is too
> little, too late.
> 
> AAA is a natural part of e-commerce, that is of the web services
> architecture, as defined in W3C and numerous industry consortiums. It
> makes little sense for all the world to make payments and conduct
> transactions in the 'web services currency' and only phone calls to be
> paid in 'DIAMETER currency'. Actually two types of phones call
> 'currencies': Wired using RADIUS and mobile using DIAMETER. Not that
> DIAMETER is inherently evil or poorly engineered. It is just too little
> too late, when looking beyond RADIUS and its limitations. My 
> two cents.

Well, I would debate that there is a significant difference between Diameter
currencies and Web services' currencies.  Diameter, ultimately, is a transport
protocol for AAA, it does not define the format, etc.
 
> I would see the design jobs therefore this way:
> 
> 1-Near term: Use RADIUS-SIP to accommodate the huge isntalled base. The
> I-D's mentioned here. Including the OSP token for SIP.
>
> 2-Longer-term: The Web services architecture.

I would think that Web services are not in scope for the IETF, it is
much more of something for W3C, OMA or IRTF to take up, in my opinion.

I think the near term is what we should be working on, and I see
the solution to be RADIUS compatability & Diameter support - 
meaning that RADIUS may be able to support Authorization & Authentication,
but Accounting might not be possible, that is why Diameter should be 
considered.  There is as well a request from the 3G's that a Diameter
based application be supported.

> It appears the present discussion stems from the fact that we only have
> volonteers for a DIAMETER-SIP draft. None of the mentioned RADIUS-SIP
> authors have volunteered any work here, and this is the problem. So
> unless the no RADIUS or web services volonteer steps up, we will have
> neither 1 nor 2.

Well, what I suggest is that for the folks to whom RADIUS-SIP is important,
they should get together and get the resources to work on a solution.  The
work doesn't do it by itself.

br,
John
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Subject: Re: [Sipping] Comments on draft-rosenberg-sipping-conferencing-framework-01.txt
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Inline prefixed by <ARC>

The idea of having a unique URI for each conference is a good one. However 
a more scalable approach would be to have a master Focus or a group of 
them that register with a standard URI. This way users use a standard URI 
with a Conference ID as the payload in the Invite message.  This is 
directed to the master Focus that does the following
* Retrieve the Conference ID
* Check if a Focus corresponding to the conference exists
* If the Conference specific Focus  does not exist create a new Focus with a unique URI derived from the Conference ID
* Forward the call using the standard Refer mechanism to the new Focus 
with the Unique URI.
 
<ARC>
 You may also want to read 
http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sipping-cc-conferencing-01.txt.
Its provides the definition of a conference Factory URI, which could be 
used to create new conferences and their Focii.
Note that when you send an INVITE to a factory URI, that entity returns 
the actual Conference-URI in the Contact header.
Also, you may be using a central 'conference enabling application' that 
asks you lots of questions before allowing you to create/join  a new
conference. See an example of that in 4.3 of alan/orit's draft.
Note however , that when you want to use a "standard wellknown URI" that 
has not been published to you by the conference server,
you are assuming that there is some agreement for that domain on 
conference naming policies.
I am not sure I like your idea of a payload contaning conf-id (what use is 
that ?)
Also, note that for scalability (if thats your primary concern), this 
mechanism allows a way for you to start with a 'guessed URI' and from
then on communicate directly with you assigned focus without routing 
everthing to what you call your 'master focus'. 
</ARC>


Instead of dealing with Conference aware UA's and Conference Unaware UA's 
we could assume all participants in the Conference to be Conference Aware. 
The Conference Unaware participants are routed through a Gateway (e.g. an 
IVR ) that provides a primitive interface to the conference unaware 
participants and a Conference aware interface to the Conference itself. 
This would keep the model simpler and it can allow Conference aware UAs a 
way to control the conference.

<ARC>
 It is extremely important to acknowledge conference-unaware endpoints. 
You cannot assume an IVR or any other application in all 
conference scenarios who 'aware' on behalf of an unaware UA. Again, 
reading both Jonathan and Alan/Orit's draft in conjunction may motivate
you more on the requirement of conference unaware UAs. Remeber also , that 
most of the EPs deployed in SIP have zilch idea of what isFocus is
as well as will not be aware of protocols to moderate a conference for a 
while.
</ARC>


The other comment was in the area of the Conference policy control 
protocol. Is this something like the SDP which is carried as a payload in 
SIP or does it have specifics of it's own. Currently most of the 
operations that are part of the CPCP are also possible through SIP based 
mechanisms. Is there a need to define two parallel mechanisms within the 
same framework. 

<ARC>
I think what has been identifed here is a need for a conference policy 
control protocol. It is not ascertained yet which protocol will deliver 
these
requirements. Thats as far as I know. The existing SDP specs do not meet 
all control requirements that come to my mind. 

Alan - would you mind reposting that link of conferencing work in progress 
 ? I thought I noticed some drafts attempting to address this in more 
detail

Now a question again for the authors of 
draft-johnston-sipping-cc-conferencing-01.txt:

* section 4.5 callflow - you show alice sending an invite to a 
conf-factory URI but the ladder diagram shows it being received by a 
'Focus'.
The definition of a focus 


3.1 Focus UA

   A focus, as defined in the framework, hosts a SIP conference and
   maintains a SIP signaling relationship with each participant in the
   conference. A focus contains a conference-aware user agent that
   supports the conferencing call control conventions as defined in this
   document.

Since a 'Focus' hosts 'a specific' conference, I find it a little 
disconcerting
that the call flows show messages with a factory-URI reaching a focus. I 
thought
it may be clearer, if we limit a 'focus' to be the target of a specific 
URI meant
for a specific conference. If a person sends an INVITE to a factory-URI or
a 'conf-app-URI' < like an IVR etc which then contacts a Factory after 
auth etc>
it should not be received by a Focus, should it ?

</ARC>

regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860  (M): 240 997 0066{@vtext.com}




"Shiva" <shaj@cisco.com>
Sent by: sipping-admin@ietf.org
02/22/2003 01:57 AM
Please respond to shaj

 
        To:     <sipping@ietf.org>
        cc: 
        Subject:        [Sipping] Comments on 
draft-rosenberg-sipping-conferencing-framework-01.txt


Hello,
 
Overall the draft looks good.  Here are some of my comments on this draft
 

The advantage of doing this is that I could have multiple Conference 
servers that are running that are registered their master focus. 
 
We can implement Load Balancing across Conference servers without the UA 
realizing it as it is not connecting to a specific Focus on a specific 
server. A second advantage is that if in the course of a meeting if the 
focus were to go down or the server went down the participants can attempt 
to rejoin the call. In such a case the Master Focus would see that the 
Conference specific focus is down and recreate it allowing users to 
quickly rejoin the conference.
 
Of course one needs to work out the details of the communication and 
keepalive between the Master Focus and the Conference focus it creates.
 
The other advantage is if there are global Conference policies these can 
be applied at the master and the Individual Conference focus needs to 
worry about the specifics of the conference it is dealing with.
 
The other comment was in the area of the Conference policy control 
protocol. Is this something like the SDP which is carried as a payload in 
SIP or does it have specifics of it's own. Currently most of the 
operations that are part of the CPCP are also possible through SIP based 
mechanisms. Is there a need to define two parallel mechanisms within the 
same framework. 
 
 
Instead of dealing with Conference aware UA's and Conference Unaware UA's 
we could assume all participants in the Conference to be Conference Aware. 
The Conference Unaware participants are routed through a Gateway (e.g. an 
IVR ) that provides a primitive interface to the conference unaware 
participants and a Conference aware interface to the Conference itself. 
This would keep the model simpler and it can allow Conference aware UAs a 
way to control the conference.
 
Regards
Shiva
Cisco Systems
 
 
 


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<br><font size=2 face="sans-serif">Inline prefixed by &lt;ARC&gt;</font>
<br>
<br><font size=3 face="Times New Roman">The idea of having a unique URI for each conference is a good one. However a more scalable approach would be to have a master Focus or a group of them that register with a standard URI. This way users use a standard URI with a Conference ID as the payload in the Invite message. &nbsp;This is directed to the master Focus that does the following</font>
<br><font size=2 face="Arial">* Retrieve the Conference ID</font>
<br><font size=2 face="Arial">* Check if a Focus corresponding to the conference exists</font>
<br><font size=2 face="Arial">* If the Conference specific Focus &nbsp;does not exist c</font><font size=3 face="Times New Roman">reate a new Focus with a unique URI derived from the Conference ID</font>
<br><font size=3 face="Times New Roman">* Forward the call using the standard Refer mechanism to the new Focus with the Unique URI.</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="sans-serif">&lt;ARC&gt;</font>
<br><font size=2 face="sans-serif">&nbsp;You may also want to read http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sipping-cc-conferencing-01.txt.</font>
<br><font size=2 face="sans-serif">Its provides the definition of a conference Factory URI, which could be used to create new conferences and their Focii.</font>
<br><font size=2 face="sans-serif">Note that when you send an INVITE to a factory URI, that entity returns the actual Conference-URI in the Contact header.</font>
<br><font size=2 face="sans-serif">Also, you may be using a central 'conference enabling application' that &nbsp;asks you lots of questions before allowing you to create/join &nbsp;a new</font>
<br><font size=2 face="sans-serif">conference. See an example of that in 4.3 of alan/orit's draft.</font>
<br><font size=2 face="sans-serif">Note however , that when you want to use a &quot;standard wellknown URI&quot; that has not been published to you by the conference server,</font>
<br><font size=2 face="sans-serif">you are assuming that there is some agreement for that domain on conference naming policies.</font>
<br><font size=2 face="sans-serif">I am not sure I like your idea of a payload contaning conf-id (what use is that ?)</font>
<br><font size=2 face="sans-serif">Also, note that for scalability (if thats your primary concern), this mechanism allows a way for you to start with a 'guessed URI' and from</font>
<br><font size=2 face="sans-serif">then on communicate directly with you assigned focus without routing everthing to what you call your 'master focus'. </font>
<br><font size=2 face="sans-serif">&lt;/ARC&gt;</font>
<br>
<br>
<br><font size=2 face="Arial">Instead of dealing with Conference aware UA's and Conference Unaware UA's we could assume all participants in the Conference to be Conference Aware. The Conference Unaware participants are routed through a Gateway (e.g. an IVR ) that provides a primitive interface to the conference unaware participants and a Conference aware interface to the Conference itself. &nbsp;This would keep the model simpler and it can allow Conference aware UAs a way to control the conference.</font>
<br>
<br><font size=2 face="sans-serif">&lt;ARC&gt;</font>
<br><font size=2 face="sans-serif">&nbsp;It is extremely important to acknowledge conference-unaware endpoints. You cannot assume an IVR or any other application in all </font>
<br><font size=2 face="sans-serif">conference scenarios who 'aware' on behalf of an unaware UA. Again, reading both Jonathan and Alan/Orit's draft in conjunction may motivate</font>
<br><font size=2 face="sans-serif">you more on the requirement of conference unaware UAs. Remeber also , that most of the EPs deployed in SIP have zilch idea of what isFocus is</font>
<br><font size=2 face="sans-serif">as well as will not be aware of protocols to moderate a conference for a while.</font>
<br><font size=2 face="sans-serif">&lt;/ARC&gt;</font>
<br>
<br>
<br><font size=2 face="Arial">The other comment was in the area of the Conference policy control protocol. Is this something like the SDP which is carried as a payload in SIP or does it have specifics of it's own. Currently most of the operations that are part of the CPCP are also possible through SIP based mechanisms. Is there a need to define two parallel mechanisms within the same framework. </font>
<br>
<br><font size=2 face="sans-serif">&lt;ARC&gt;</font>
<br><font size=2 face="sans-serif">I think what has been identifed here is a need for a conference policy control protocol. It is not ascertained yet which protocol will deliver these</font>
<br><font size=2 face="sans-serif">requirements. Thats as far as I know. The existing SDP specs do not meet all control requirements that come to my mind. </font>
<br>
<br><font size=2 face="sans-serif">Alan - would you mind reposting that link of conferencing work in progress &nbsp;? I thought I noticed some drafts attempting to address this in more detail</font>
<br>
<br><font size=2 face="sans-serif">Now a question again for the authors of &nbsp;draft-johnston-sipping-cc-conferencing-01.txt:</font>
<br>
<br><font size=2 face="sans-serif">* section 4.5 callflow - you show alice sending an invite to a conf-factory URI but the ladder diagram shows it being received by a 'Focus'.</font>
<br><font size=2 face="sans-serif">The definition of a focus </font>
<br>
<br>
<br><font size=2 face="Arial">3.1 Focus UA<br>
<br>
 &nbsp; A focus, as defined in the framework, hosts a SIP conference and<br>
 &nbsp; maintains a SIP signaling relationship with each participant in the<br>
 &nbsp; conference. A focus contains a conference-aware user agent that<br>
 &nbsp; supports the conferencing call control conventions as defined in this<br>
 &nbsp; document.</font>
<br>
<br><font size=2 face="Arial">Since a 'Focus' hosts 'a specific' conference, I find it a little disconcerting</font>
<br><font size=2 face="Arial">that the call flows show messages with a factory-URI reaching a focus. I thought</font>
<br><font size=2 face="Arial">it may be clearer, if we limit a 'focus' to be the target of a specific URI meant</font>
<br><font size=2 face="Arial">for a specific conference. If a person sends an INVITE to a factory-URI or</font>
<br><font size=2 face="Arial">a 'conf-app-URI' &lt; like an IVR etc which then contacts a Factory after auth etc&gt;</font>
<br><font size=2 face="Arial">it should not be received by a Focus, should it ?</font>
<br>
<br><font size=2 face="Arial">&lt;/ARC&gt;</font>
<br>
<br><font size=3 face="Courier New">regds</font>
<br><font size=3 face="Courier New">Arjun<br>
</font>
<br><font size=2 face="sans-serif">--<br>
Arjun Roychowdhury @ Hughes Software Systems<br>
11717 Exploration Lane, Germantown MD 20876<br>
(O): 301 212 7860 &nbsp;(M): 240 997 0066{@vtext.com}</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td><font size=1 face="sans-serif"><b>&quot;Shiva&quot; &lt;shaj@cisco.com&gt;</b></font>
<br><font size=1 face="sans-serif">Sent by: sipping-admin@ietf.org</font>
<p><font size=1 face="sans-serif">02/22/2003 01:57 AM</font>
<br><font size=1 face="sans-serif">Please respond to shaj</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&lt;sipping@ietf.org&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[Sipping] Comments on draft-rosenberg-sipping-conferencing-framework-01.txt</font></table>
<br>
<br>
<br><font size=2 face="Arial">Hello,</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">Overall the draft looks good. &nbsp;Here are some of my comments on this draft</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br>
<br><font size=2 face="Arial">The advantage of doing this is that I could have multiple Conference servers that are running that are registered their master focus. </font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">We can implement Load Balancing across Conference servers without the UA realizing it as it is not connecting to a specific Focus on a specific server. A second advantage is that if in the course of a meeting if the focus were to go down or the server went down the participants can attempt to rejoin the call. In such a case the Master Focus would see that the Conference specific focus is down and recreate it allowing users to quickly rejoin the conference.</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">Of course one needs to work out the details of the communication and keepalive between the Master Focus and the Conference focus it creates.</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">The other advantage is if there are global Conference policies these can be applied at the master and the Individual Conference focus needs to worry about the specifics of the conference it is dealing with.</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">The other comment was in the area of the Conference policy control protocol. Is this something like the SDP which is carried as a payload in SIP or does it have specifics of it's own. Currently most of the operations that are part of the CPCP are also possible through SIP based mechanisms. Is there a need to define two parallel mechanisms within the same framework. </font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">Instead of dealing with Conference aware UA's and Conference Unaware UA's we could assume all participants in the Conference to be Conference Aware. The Conference Unaware participants are routed through a Gateway (e.g. an IVR ) that provides a primitive interface to the conference unaware participants and a Conference aware interface to the Conference itself. &nbsp;This would keep the model simpler and it can allow Conference aware UAs a way to control the conference.</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=2 face="Arial">Regards</font>
<br><font size=2 face="Arial">Shiva</font>
<br><font size=2 face="Arial">Cisco Systems</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br><font size=3 face="Times New Roman">&nbsp;</font>
<br>
<br>
--=_alternative 0055D6DB85256CD5_=--
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Date: Sun, 23 Feb 2003 20:52:54 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
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To: jari.arkko@piuha.net, jh@lohi.eng.song.fi
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> To recap: Let's plan for a solution that provides the 
> functions required in the requirements document (using 
> DIAMETER), but allows for some amount of compatibility with RADIUS.

We may live to see neither in the near future, but a new one, based on
web services, a la .NET or Liberty Alliance. As mentioned, it makes no
sense for telephone calls to be AAA-ed and purchased in some exotic
RADIUS or DIAMTER "currency", while everything else is purchased and
AAA-ed with web services.

The problem in this WG at present is there are no volunteers for this
alternative. I will desist trying to stop DIAMETER if it meets someone's
needs, but just wanted to share the possibility mentioned above.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Jari Arkko
> Sent: Saturday, February 22, 2003 2:54 AM
> To: jh@lohi.eng.song.fi
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> jh@lohi.eng.song.fi wrote:
> > john.loughney@nokia.com writes:
> > 
> >  > I understand your reasons for not wanting to upgrade 
> everything  > 
> > to Diameter, but I still don't see what reasons you have 
> for  > being 
> > dismissive of Diameter.
> > 
> > henry may have points why he doesn't like diameter, but for 
> me, it is 
> > nto really that big deal IF diameter AAA infrastructure some day 
> > becomes wide spread.  today, however, that is not the case. 
>  instead 
> > everybody is using radius as the protocol between NASes 
> (includifn sip 
> > proxies) and AAA servers and also between AAA servers themselves.
> 
> Right. That's why DIAMETER has been designed to be able to 
> work with RADIUS. Naturally new features in DIAMETER are not 
> necessarily accessible in RADIUS, but things that run on 
> RADIUS should also run on DIAMETER, such as current NAS or AP 
> authentication.
> 
> (This is by the way one reason why I don't believe web 
> services will be successful in this space; I see the RADIUS - 
> DIAMETER line continuing for network access, telephony even 
> if some other apps do something else.)
> 
> > examples of the latter are gric and ipass, which provide roaming 
> > service for hundreds of isps literally all over the world.  
> if radius 
> > is so bad that what you say IESG claims, i don't understand 
> how this 
> > is possible. the roaming services run radius over ipsec 
> which solves 
> > the security issues.
> 
> The limitations of RADIUS have been documented in various 
> places. Running IPsec does actually help in some of the 
> security-related limitations, though not all. Everyone knows 
> RADIUS has problems, but it was the only thing people had.
> 
> To recap: Let's plan for a solution that provides the 
> functions required in the requirements document (using 
> DIAMETER), but allows for some amount of compatibility with RADIUS.
> 
> Jari
> 
> _______________________________________________
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Diameter Multimedia Application
	Author(s)	: M. Belinchon et al.
	Filename	: draft-belinchon-aaa-diameter-mm-app-00.txt
	Pages		: 37
	Date		: 2003-2-21
	
This document specifies a Diameter application that supports the
authentication, authorization, and collection of accounting
information for Session Initiation Protocol (SIP) services rendered
to a client node.  This application, combined with the base Diameter
protocol and appropriate SIP extensions, allows SIP User Agents (UAs)
to obtain services from SIP servers that are connected to a Diameter
infrastructure.  When combined with the Inter-Domain capability of
the base protocol, service may even be obtained from SIP servers that
belong to foreign domains, as would be encountered by roaming mobile
nodes.
Note that the specification defined here may be used independently of
the authentication technique used for authenticating a node's link
layer or network-layer access.  In particular, we do not require the
client node to be authenticated for access with the use of either the
Mobile IP [3] or NASREQ [11] Diameter application.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-belinchon-aaa-diameter-mm-app-00.txt

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A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
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	"FILE /internet-drafts/draft-belinchon-aaa-diameter-mm-app-00.txt".
	
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--------------030109090601070907060601--

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From mailnull@www1.ietf.org  Mon Feb 24 10:48:03 2003
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Date: Mon, 24 Feb 2003 09:39:16 -0600
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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Subject: [Sipping] I-D on persistent connections
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Folks:

A while ago we had a discussion on persistent connections in SIP.
We have submitted an I-D to the IETF outlining the problem, some
requirements and some solutions.  Until it appears in the archives, you
can get it from:

http://www.iit.edu/~gurbvij/I-D/draft-jain-sipping-persistent-conn-reqs-00.txt

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From mailnull@www1.ietf.org  Mon Feb 24 11:24:35 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "Jonathan Rosenberg (E-mail)" <jdrosen@dynamicsoft.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Mon, 24 Feb 2003 11:26:09 -0500
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Subject: [Sipping] Sidebars
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In the latest conferencing framework draft, sidebars are
proposed as a separate conference URI with special properties.
In particular there is this text:
   A sidebar is represented by a separate conference URI. This URI is a
   type of "alias" for the main conference URI. Both route to the same
   focus. Like any other conference, the sidebar conference URI has a
   conference policy and a media policy associated with it. Like any
   other conference, one can join it by sending an INVITE to this URI,
   or ask others to join by referring them to it. However, it differs
   from a normal conference URI in several ways. First, users in the
   main conference do not need to establish a separate dialog to the
   sidebar conference. The focus recognizes the sidebar as a special
   URI, and knows to use the existing dialog to the main conference as a
   "virtual" connection to the sidebar URI.

   The second difference is the way in which conference and media
   policies are implemented. If the conference policy control protocol
   is used to add a user to a normal conference, the focus will
   typically send an INVITE to the participant to ask them to join. For
   a sidebar conference, it is done differently. If the conference
   policy control protocol is used to add a user to it, and that user is
   already part of the main conference, the focus will use the
   conference notification service to alert the existing participant
   that they have been asked to join the sidebar. The invited user can
   then make use of the CPCP to formally add themselves to the sidebar.

I'm interested in how a non-conference aware UA participates in a sidebar.
Clearly, an out-of-band mechanism can be used to manipulate the CPCP
to add such a UA to a sidebar, and similar means to control media
policy.  This is very encouraging.

I'm confused, however, on exactly how this special sidebar URI
will work with a non-conference aware UA.

Specifically:
The text says that an INVITE with the sidebar URI will not be sent
by the focus to request the user to join.  Is there any circumstance
in which the focus would use the sidebar URI?  I'm thinking of,
for example, what would normally be a re-invite to add a media
stream when another user, who is not a main conference participant
already, joins the sidebar.  Any use of the sidebar URI with a non
conference aware UA would confuse the heck out of it.

So far, I'm thinking that the only real use of this URI is for
a UA which is not a main conference participant to get added to
the sidebar, but not the main conference.  The wording in the framework
document could offer more clarity on the use of this URI.

Brian
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, sipping@ietf.org
Subject: RE: [Sipping] Comments on Overlap-04
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Some responses below.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Wednesday, February 19, 2003 7:41 PM
To: sipping@ietf.org
Subject: [Sipping] Comments on Overlap-04


Comments on draft-ietf-sipping-overlap-04: 

General: 

The ability of one call to result in multiple IAMs being sent into the PSTN
continues to be a serious problem for overlap which must be solved.  
 
<JFP> This concern has been the subject of whole libraries of previous mail;
I think we need to agree to disagree on this point. As we've said in the
past, the draft allows this condition to arise only if operators choose to
configure their networks in a fashion that allows it. </JFP> 

Section 3.1: 

All references to one number being a "prefix" of another should be deleted.
This text indicates that overlap can not be used where this "prefix"
situation can occur, however, one of the purposes of overlap is to handle
the situation in which there are varying lengths of numbers in some
destination area, something that would naturally exist if one number really
could be a "prefix" of another. This creates a major contradiction.  
 
<JFP> Not necessarily a contradiction, actually. There are varying lengths
of numbers in some numbering plans. That does not entail, however, that one
number can necessarily be a prefix of another. The numbers +9990123 and
+99912345 could exist in the same numbering plan without creating any
problems for overlap signaling. The concern is if the numbers +9990123 and
+99901234 were to exist in the same numbering plan.    
 
 However, there is still no identified case in the world-wide numbering plan
in which one number is a "prefix" of another. If there were, overlap
signaling to get to it would certainly be allowed just as it would be in the
existing PSTN. If one believes that this "prefix" situation really exists,
then the overlap procedures defined in this draft must correctly support it.

 
<JFP> We can't necessarily verify that it's the case, but people keep
telling us that this happens in practice in the PSTN. As usual, it is tough
to provide a counter-existence proof. I think we still need this text.
</JFP>

Section 3.3 contains the statement: 

  As a general rule, a media stream corresponding to the response to an 
  INVITE with a greater number of digits should be given more priority 
  than media streams from responses with less digits. 

This continues to be a problem, since it is not always the case. If the
originator dialed more digits than were needed when overlap was being used,
the media stream corresponding to the response with the fewer digits would
be the one to use. The PSTN had no problem with this. SIP is going to have
to figure out a way to correctly handle it. 
 
<JFP> Our text satisfies particular early-media concerns, and is valid, in
my opinion. If fewer digits than the minimum length of a number in the
numbering plan are dialed, a gateway could still attempt to pass backwards
media to the UAC in order to provide early media data. The behavior that
occurs when more digits are dialed than necessary is, no doubt, specific to
different nationalities, but it seems reasonable to assume that superfluous
digits will be discarded. No one, to date, has asserted anything to the
contrary anyway. </JFP>

Section 3.5 contains the statement: 

  If the gateway is attached to the PSTN in an area where en-bloc 
  signalling is used, a REL for the previous IAM and a new IAM should 
  be generated. 

This causes confusion since all areas would allow en-bloc, although some do
not allow overlap, for example, an area where US SS#7 according to T1.113 is
used. In those areas, since overlap is not allowed, the gateway or switch
which provides interworking into that non-overlap area must perform the
overlap to en-bloc procedures (by time outs, detecting enough digits, etc.)
There can not be a "previous IAM" sent to the PSTN, since the gateway or
switch would not have sent one until it knew it had enough digits to send a
complete IAM, after which it would stop accepting or sending any further
digits. 
 
<JFP> First of all, if you send an incomplete number into an en-bloc
signaling network, that are ISUP failure responses (REL with cause 28) that
will be sent in response. Our text identifies what one should do if the call
has not already been rejected. Gateway very well might be stupid enough to
send incomplete addresses into a network that cannot accept them, and it
seems dangerous to assume otherwise. </JFP>

This paragraph should be worded as: 

  If the gateway is attached to the PSTN in an area where overlap signalling

  is not allowed, it must not send an IAM until it determines that it has 
  received the full number in one or more INVITEs, using a process similar
to   
  that described in Section 2. It must not send SAMs or additional IAMs
after 
  that one IAM. 

Mike Pierce 



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<DIV><SPAN class=268460804-20022003><FONT face=Arial color=#0000ff size=2>Some 
responses below.</FONT></SPAN></DIV>
<DIV><SPAN class=268460804-20022003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=268460804-20022003><FONT face=Arial color=#0000ff size=2>Jon 
Peterson</FONT></SPAN></DIV>
<DIV><SPAN class=268460804-20022003><FONT face=Arial color=#0000ff 
size=2>NeuStar, Inc.</FONT></SPAN></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> Wednesday, February 19, 2003 7:41 
  PM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] Comments on 
  Overlap-04<BR><BR></FONT></DIV>
  <DIV><FONT face=Arial size=2>Comments on draft-ietf-sipping-overlap-04: 
  <BR><BR>General: <BR><BR>The ability of one call to result in multiple IAMs 
  being sent into the PSTN continues to be a serious problem for overlap which 
  must be solved.&nbsp;<SPAN class=268460804-20022003><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=268460804-20022003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN class=268460804-20022003><FONT 
  color=#0000ff>&lt;JFP&gt; This&nbsp;concern&nbsp;has been the subject of whole 
  libraries of previous mail;&nbsp;I think we need to agree to disagree on this 
  point. As we've said in the past, the draft allows this condition to arise 
  only if operators choose to configure their networks in a fashion that allows 
  it. &lt;/JFP&gt;</FONT>&nbsp;</SPAN><BR><BR>Section 3.1: <BR><BR>All 
  references to one number being a "prefix" of another should be deleted. This 
  text indicates that overlap can not be used where this "prefix" situation can 
  occur, however, one of the purposes of overlap is to handle the situation in 
  which there are varying lengths of numbers in some destination area, something 
  that would naturally exist if one number really could be a "prefix" of 
  another. This creates a major contradiction.&nbsp;<SPAN 
  class=268460804-20022003><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT size=2><SPAN 
  class=268460804-20022003></SPAN></FONT>&nbsp;</DIV><SPAN 
  class=268460804-20022003><FONT face=Arial color=#0000ff 
  size=2>&lt;JFP&gt;&nbsp;Not necessarily a contradiction, actually. There are 
  varying lengths of numbers in some numbering plans.&nbsp;That does not entail, 
  however, that one number can necessarily be a prefix of another. The numbers 
  +9990123 and +99912345 could exist in the same numbering plan without creating 
  any problems for overlap signaling. The concern is if the numbers +9990123 and 
  +99901234 were to exist in the same numbering plan.</FONT></SPAN>&nbsp;<SPAN 
  class=268460804-20022003><FONT face=Arial size=2>&nbsp;&nbsp;</FONT></SPAN>
  <DIV><SPAN class=268460804-20022003></SPAN>&nbsp;</DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN 
  class=268460804-20022003>&nbsp;</SPAN>However, there is still no identified 
  case in the world-wide numbering plan in which one number is a "prefix" of 
  another. If there were, overlap signaling to get to it would certainly be 
  allowed just as it would be in the existing PSTN. If one believes that this 
  "prefix" situation really exists, then the overlap procedures defined in this 
  draft must correctly support it.&nbsp;<BR><SPAN class=268460804-20022003><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN class=268460804-20022003><SPAN 
  class=268460804-20022003><FONT color=#0000ff>&lt;JFP&gt; We can't necessarily 
  verify that it's the case, but people keep telling us that this happens in 
  practice in the PSTN. As usual, it is tough to provide a counter-existence 
  proof. I think we still need this 
  text.&nbsp;&lt;/JFP&gt;</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN 
  class=268460804-20022003></SPAN><BR>Section 3.3 contains the statement: 
  <BR><BR>&nbsp;&nbsp;As a general rule, a media stream corresponding to the 
  response to an <BR>&nbsp;&nbsp;INVITE with a greater number of digits should 
  be given more priority <BR>&nbsp;&nbsp;than media streams from responses with 
  less digits. <BR><BR>This continues to be a problem, since it is not always 
  the case. If the originator dialed more digits than were needed when overlap 
  was being used, the media stream corresponding to the response with the fewer 
  digits would be the one to use. The PSTN had no problem with this. SIP is 
  going to have to figure out a way to correctly handle it.&nbsp;<BR><SPAN 
  class=268460804-20022003><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN class=268460804-20022003><SPAN 
  class=268460804-20022003><FONT color=#0000ff>&lt;JFP&gt; Our text satisfies 
  particular early-media concerns, and is valid, in my opinion. If fewer digits 
  than the minimum length of a number in the numbering plan are dialed, a 
  gateway could still attempt to pass backwards media to the UAC in order to 
  provide early media data. The behavior that occurs when more digits are dialed 
  than necessary is, no doubt, specific to different nationalities, but it seems 
  reasonable to assume that superfluous digits will be discarded. No one, to 
  date, has asserted anything to the contrary anyway. 
  &lt;/JFP&gt;</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN 
  class=268460804-20022003></SPAN><BR>Section 3.5 contains the statement: 
  <BR><BR>&nbsp;&nbsp;If the gateway is attached to the PSTN in an area where 
  en-bloc <BR>&nbsp;&nbsp;signalling is used, a REL for the previous IAM and a 
  new IAM should <BR>&nbsp;&nbsp;be generated. <BR><BR>This causes confusion 
  since all areas would allow en-bloc, although some do not allow overlap, for 
  example, an area where US SS#7 according to T1.113 is used. In those areas, 
  since overlap is not allowed, the gateway or switch which provides 
  interworking into that non-overlap area must perform the overlap to en-bloc 
  procedures (by time outs, detecting enough digits, etc.) There can not be a 
  "previous IAM" sent to the PSTN, since the gateway or switch would not have 
  sent one until it knew it had enough digits to send a complete IAM, after 
  which it would stop accepting or sending any further digits.&nbsp;<BR><SPAN 
  class=268460804-20022003><FONT 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN class=268460804-20022003><SPAN 
  class=268460804-20022003><FONT color=#0000ff>&lt;JFP&gt; First of all, if you 
  send an incomplete number into an en-bloc signaling network, that are ISUP 
  failure responses (REL with cause 28) that&nbsp;will be sent in response. Our 
  text identifies what one should do if the call has not already been rejected. 
  Gateway very well might be stupid enough to send incomplete addresses into a 
  network that cannot accept them, and it seems dangerous to assume otherwise. 
  </FONT></SPAN></SPAN></FONT></FONT><FONT face=Arial><FONT size=2><SPAN 
  class=268460804-20022003><SPAN class=268460804-20022003><FONT 
  color=#0000ff>&lt;/JFP&gt;</FONT></SPAN></SPAN></FONT></FONT></DIV>
  <DIV><FONT face=Arial><FONT size=2><SPAN 
  class=268460804-20022003></SPAN><BR>This paragraph should be worded as: 
  <BR><BR>&nbsp;&nbsp;If the gateway is attached to the PSTN in an area where 
  overlap signalling <BR>&nbsp;&nbsp;is not allowed, it must not send an IAM 
  until it determines that it has <BR>&nbsp;&nbsp;received the full number in 
  one or more INVITEs, using a process similar to &nbsp; <BR>&nbsp;&nbsp;that 
  described in Section 2. It must not send SAMs or additional IAMs after 
  <BR>&nbsp;&nbsp;that one IAM. <BR><BR>Mike Pierce 
<BR></DIV></FONT></FONT></BLOCKQUOTE></BODY></HTML>

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To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] New version of the early media draft
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Gonzallo, 

I finally had a chance to read the new version of early media draft. Kind of
interesting
that this version didn't get much feedback at all.

It is a very good document. I think you have captured very well the very
many issues that
were discussed last year. I am very comfortable with this new version.

I have only one suggestion (and a minor one). In section 2.2 on page 7, in
the paragraph that
starts with "Note" just below point 3, I would change the lst sentence to
the following:

   This type of gateway could choose to generate local ringback upon
reception 
   of a 180 (Ringing) response, and override or mix it with any incoming
media that 
   happens to arrive (if it does at all).



> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se] 
> Sent: Monday, February 10, 2003 3:41 AM
> To: sipping
> Subject: [Sipping] New version of the early media draft
> 
> 
> Folks,
> 
> I have just sumitted a new version of the early media draft. 
> Until it appears in the archives, you can fetch it from:
> 
http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-early-m
edia-01.txt

I have included all the (long and useful) discussions we had in the mailing
list. Therefore, the draft now keeps separate the "gateway" and the
"application server" models. I have also removed the parts we did not agree
on.

Feedback is, as usual, appreciated.

Thanks,

Gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use
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<TITLE>RE: [Sipping] New version of the early media draft</TITLE>
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<P><FONT SIZE=3D2>Gonzallo, </FONT>
</P>

<P><FONT SIZE=3D2>I finally had a chance to read the new version of =
early media draft. Kind of interesting</FONT>
<BR><FONT SIZE=3D2>that this version didn't get much feedback at =
all.</FONT>
</P>

<P><FONT SIZE=3D2>It is a very good document. I think you have captured =
very well the very many issues that</FONT>
<BR><FONT SIZE=3D2>were discussed last year. I am very comfortable with =
this new version.</FONT>
</P>

<P><FONT SIZE=3D2>I have only one suggestion (and a minor one). In =
section 2.2 on page 7, in the paragraph that</FONT>
<BR><FONT SIZE=3D2>starts with &quot;Note&quot; just below point 3, I =
would change the lst sentence to the following:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp; This type of gateway could choose to =
generate local ringback upon reception </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; of a 180 (Ringing) response, and =
override or mix it with any incoming media that </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; happens to arrive (if it does at =
all).</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Gonzalo Camarillo [<A =
HREF=3D"mailto:Gonzalo.Camarillo@lmf.ericsson.se">mailto:Gonzalo.Camaril=
lo@lmf.ericsson.se</A>] </FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Monday, February 10, 2003 3:41 AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: sipping</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: [Sipping] New version of the early =
media draft</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I have just sumitted a new version of the early =
media draft. </FONT>
<BR><FONT SIZE=3D2>&gt; Until it appears in the archives, you can fetch =
it from:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sip=
ping-early-media-01.txt" =
TARGET=3D"_blank">http://standards.ericsson.net/gonzalo/papers/draft-cam=
arillo-sipping-early-media-01.txt</A></FONT>
</P>

<P><FONT SIZE=3D2>I have included all the (long and useful) discussions =
we had in the mailing list. Therefore, the draft now keeps separate the =
&quot;gateway&quot; and the &quot;application server&quot; models. I =
have also removed the parts we did not agree on.</FONT></P>

<P><FONT SIZE=3D2>Feedback is, as usual, appreciated.</FONT>
</P>

<P><FONT SIZE=3D2>Thanks,</FONT>
</P>

<P><FONT SIZE=3D2>Gonzalo</FONT>
<BR><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>This list is for NEW development of the application =
of SIP</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip Use sip@ietf.org for new developments of core SIP</FONT>
</P>

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Francois Audet'" <audet@nortelnetworks.com>,
        "'Gonzalo Camarillo'"
	 <Gonzalo.Camarillo@lmf.ericsson.se>,
        "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] New version of the early media draft
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Though I didn't say so to this list, I also thought this document was a
great improvement, and I'm also very comfortable with it.
 
Jon Peterson
NeuStar, Inc.

-----Original Message-----
From: Francois Audet [mailto:audet@nortelnetworks.com]
Sent: Monday, February 24, 2003 3:28 PM
To: 'Gonzalo Camarillo'; 'sipping'
Subject: RE: [Sipping] New version of the early media draft



Gonzallo, 

I finally had a chance to read the new version of early media draft. Kind of
interesting 
that this version didn't get much feedback at all. 

It is a very good document. I think you have captured very well the very
many issues that 
were discussed last year. I am very comfortable with this new version. 

I have only one suggestion (and a minor one). In section 2.2 on page 7, in
the paragraph that 
starts with "Note" just below point 3, I would change the lst sentence to
the following: 

   This type of gateway could choose to generate local ringback upon
reception 
   of a 180 (Ringing) response, and override or mix it with any incoming
media that 
   happens to arrive (if it does at all). 



> -----Original Message----- 
> From: Gonzalo Camarillo [ mailto:Gonzalo.Camarillo@lmf.ericsson.se
<mailto:Gonzalo.Camarillo@lmf.ericsson.se> ] 
> Sent: Monday, February 10, 2003 3:41 AM 
> To: sipping 
> Subject: [Sipping] New version of the early media draft 
> 
> 
> Folks, 
> 
> I have just sumitted a new version of the early media draft. 
> Until it appears in the archives, you can fetch it from: 
> 
http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-early-m
edia-01.txt
<http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sipping-early-
media-01.txt>  

I have included all the (long and useful) discussions we had in the mailing
list. Therefore, the draft now keeps separate the "gateway" and the
"application server" models. I have also removed the parts we did not agree
on.

Feedback is, as usual, appreciated. 

Thanks, 

Gonzalo 
_______________________________________________ 
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
<https://www1.ietf.org/mailman/listinfo/sipping>  
This list is for NEW development of the application of SIP 
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP 


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<BODY>
<DIV><SPAN class=3D058465923-24022003><FONT face=3DArial =
color=3D#0000ff size=3D2>Though=20
I didn't say so to this list,&nbsp;I also thought this document was a =
great=20
improvement, and I'm also very comfortable with it.</FONT></SPAN></DIV>
<DIV><SPAN class=3D058465923-24022003><FONT face=3DArial =
color=3D#0000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D058465923-24022003><FONT face=3DArial =
color=3D#0000ff size=3D2>Jon=20
Peterson</FONT></SPAN></DIV>
<DIV><SPAN class=3D058465923-24022003><FONT face=3DArial =
color=3D#0000ff=20
size=3D2>NeuStar, Inc.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Francois Audet=20
  [mailto:audet@nortelnetworks.com]<BR><B>Sent:</B> Monday, February =
24, 2003=20
  3:28 PM<BR><B>To:</B> 'Gonzalo Camarillo'; =
'sipping'<BR><B>Subject:</B> RE:=20
  [Sipping] New version of the early media draft<BR><BR></FONT></DIV>
  <P><FONT size=3D2>Gonzallo, </FONT></P>
  <P><FONT size=3D2>I finally had a chance to read the new version of =
early media=20
  draft. Kind of interesting</FONT> <BR><FONT size=3D2>that this =
version didn't=20
  get much feedback at all.</FONT> </P>
  <P><FONT size=3D2>It is a very good document. I think you have =
captured very=20
  well the very many issues that</FONT> <BR><FONT size=3D2>were =
discussed last=20
  year. I am very comfortable with this new version.</FONT> </P>
  <P><FONT size=3D2>I have only one suggestion (and a minor one). In =
section 2.2=20
  on page 7, in the paragraph that</FONT> <BR><FONT size=3D2>starts =
with "Note"=20
  just below point 3, I would change the lst sentence to the =
following:</FONT>=20
  </P>
  <P><FONT size=3D2>&nbsp;&nbsp; This type of gateway could choose to =
generate=20
  local ringback upon reception </FONT><BR><FONT size=3D2>&nbsp;&nbsp; =
of a 180=20
  (Ringing) response, and override or mix it with any incoming media =
that=20
  </FONT><BR><FONT size=3D2>&nbsp;&nbsp; happens to arrive (if it does =
at=20
  all).</FONT> </P><BR><BR>
  <P><FONT size=3D2>&gt; -----Original Message-----</FONT> <BR><FONT =
size=3D2>&gt;=20
  From: Gonzalo Camarillo [<A=20
  =
href=3D"mailto:Gonzalo.Camarillo@lmf.ericsson.se">mailto:Gonzalo.Camaril=
lo@lmf.ericsson.se</A>]=20
  </FONT><BR><FONT size=3D2>&gt; Sent: Monday, February 10, 2003 3:41 =
AM</FONT>=20
  <BR><FONT size=3D2>&gt; To: sipping</FONT> <BR><FONT size=3D2>&gt; =
Subject:=20
  [Sipping] New version of the early media draft</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; =
Folks,</FONT>=20
  <BR><FONT size=3D2>&gt; </FONT><BR><FONT size=3D2>&gt; I have just =
sumitted a new=20
  version of the early media draft. </FONT><BR><FONT size=3D2>&gt; =
Until it=20
  appears in the archives, you can fetch it from:</FONT> <BR><FONT =
size=3D2>&gt;=20
  </FONT><BR><FONT size=3D2><A target=3D_blank=20
  =
href=3D"http://standards.ericsson.net/gonzalo/papers/draft-camarillo-sip=
ping-early-media-01.txt">http://standards.ericsson.net/gonzalo/papers/dr=
aft-camarillo-sipping-early-media-01.txt</A></FONT>=20
  </P>
  <P><FONT size=3D2>I have included all the (long and useful) =
discussions we had=20
  in the mailing list. Therefore, the draft now keeps separate the =
"gateway" and=20
  the "application server" models. I have also removed the parts we did =
not=20
  agree on.</FONT></P>
  <P><FONT size=3D2>Feedback is, as usual, appreciated.</FONT> </P>
  <P><FONT size=3D2>Thanks,</FONT> </P>
  <P><FONT size=3D2>Gonzalo</FONT> <BR><FONT=20
  size=3D2>_______________________________________________</FONT> =
<BR><FONT=20
  size=3D2>Sipping mailing list&nbsp; <A target=3D_blank=20
  =
href=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.iet=
f.org/mailman/listinfo/sipping</A></FONT>=20
  <BR><FONT size=3D2>This list is for NEW development of the =
application of=20
  SIP</FONT> <BR><FONT size=3D2>Use sip-implementors@cs.columbia.edu =
for questions=20
  on current sip Use sip@ietf.org for new developments of core =
SIP</FONT>=20
</P></BLOCKQUOTE></BODY></HTML>

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Requirements for conference policy data
	Author(s)	: P. Koskelainen
	Filename	: draft-koskelainen-sipping-conf-policy-req-00.txt
	Pages		: 9
	Date		: 2003-2-24
	
The conference participants may communicate with the conference
policy server, using a conference policy control protocol (CPCP)
which is a strictly client-server transactional protocol. This
document describes the requirements for conference policy data. Media
policy related requirements are beyond the scope of this document.
CPCP protocol is not mandatory and the only mechanism to manipulate
conference policy data in a conference. For example, web interface
can be used as well to perform conference policy manipulation.
However, for automata a protocol is needed.

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http://www.ietf.org/internet-drafts/draft-koskelainen-sipping-conf-policy-req-00.txt

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From mailnull@www1.ietf.org  Tue Feb 25 08:38:42 2003
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Date: Fri, 21 Feb 2003 11:22:08 -0500
To: Allison Mankin <mankin@psg.com>
From: Stuart Jacobs <stu.jacobs@labs.gte.com>
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft 
Cc: sipping@ietf.org, sip@ietf.org
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Allison,

This material is part of ongoing efforts of Verizon and Verizon management 
to engage in thoughtful considerations of the fundamental changes and 
challenges facing the telecommunications industry. To meet its fiduciary 
responsibilities, management must explore all alternatives, even those that 
may appear highly speculative and hypothetical. Statements and 
representations contained herein are preliminary and/or tentative and 
should not be relied on unless approved by the appropriate Verizon 
governing body.  The following represent the anticipated approach Verizon 
most likely will use.  I cannot be more firm at this time as our 
architectural plans are not finalized.

Verizon's current direction for our Next Generation Network (NGN) VoIP 
Services will likely use a challenge/response approach (ala. Radius) for 
authentication and access control between our SIP service customers and SIP 
Proxies.  All our SIP service customers will probably go through SIP 
Proxies.  We will not support SIP UA to SIP UA within our operational 
domain.  Our SIP Proxies will likely authenticate themselves to other SIP 
Proxies and other back-office systems (OSSs, Application Servers, EMSs, 
etc.) by use of IPsec, specifically IKE combined with X.509v3 server 
certificates and ESP transport mode Null-encryption using 
HMAC-SHA1-96.  Access controls (rights/policies) between our intermediate 
systems (incl. SIP Proxies, Gateways , etc.) are expected to be based on 
use of policy certificates.  All policy and identity certificates are 
expected to be distributed via a redundant hierarchy of LDAP servers 
(repositories).  Accounting data will most likely be compiled by our SIP 
Proxies and transferred to billing systems with authentication provided by 
IPsec.  The actual billing data will probably be captured in the form of 
XML in conjunction with XML signatures (as defined by the W3C).

Stu

At 2/20/03 05:24 PM, you wrote:
>Stu,
>
>You are not doing the usage accounting with Radius, then.
>
>It is only doing the remote access authentication?
>
>We can work with this scenario.  Could you speak to this in
>the open mailing list?  I would like to rule out Radius for
>call detail and usage-based accounting, but it is ok for the
>other A's.
>
>Allison
>
> > Diameter in our Next Generation Network (NGN) VoIP Services.  We will be
> > doing our usage accounting within SIP Proxies and intend to transfer said
> > accounting "call detail records" back to our billing OSSes via other
> > protocols with IPsec providing authentication and confidentiality for 
> these
> > transfers.  We currently do not plan on any deployment of Diameter within
> > Verizon's wireline NGN architecture.  We will probably continue to use
> > Radius for remote access authentication at the application layer for a
> > number or years into the future.
> >
> > Stu Jacobs
> >
> > At 2/14/03 09:58 PM, you wrote:
> >
> > >Jiri,
> > >
> > >It always seems good to be able to stay with installed base, but
> > >there's a factor against our supporting RADIUS drafts for SIP's
> > >accounting - safety.  Even though there is deployment, there are
> > >some extremely bad cases of damage in production, and the IETF
> > >has concluded that RADIUS is not suitable for usage-based accounting
> > >(RFC2975).
> > >
> > >Unless there is no intention of having in SIP accounting usage-based,
> > >monetary, auditable applications, RADIUS is not the solution.  The
> > >protocol just does not preserve a complete stream of reliable
> > >accounting records that are auditable on any level, especially with
> > >any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
> > >the use of RADIUS may even be illegal in situations where there is an
> > >authority over the billing.  Records are too easily lost by the
> > >protocol. This is stated clearly by RFC2975.  This is the reason why
> > >the Operations and Mgt Area developed Diameter.  On these technical*
> > >grounds, that the RADIUS protocol does not have enough reliability,
> > >the work must be based on the Diameter protocol, which is now
> > >approved.  Alternatively, work could be done that does not include
> > >usage-based accounting, but this does not seem to be what is
> > >envisioned in the SIPPING AAA requirements.
> > >
> > >Allison
> > >
> > >P.S. There's a current investigation of RADIUS compatibility mode
> > >in Diameter for this - more needs to be thought about it.
> > > >
> > > > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > > > >Hi Henry,
> > > > >
> > > > >> >but one option is develop a solution using
> > > > >> > Diameter in RADIUS-compatibility mode.
> > > > >>
> > > > >> OK, this is better than nothing. We'll take what we can :-)
> > > > >>
> > > > >> As mentioned, the installed base is RADIUS, no matter how
> > > > >> voluminous the DIAMETER literature may be.
> > > > >
> > > > >I think the issues is that the IETF should recommend what is the
> > > > >right thing to do & why.  If people choose to do something else,
> > > > >well, that is life.
> > > >
> > > > Remember that running code does not necessarily mean something known
> > > > to compile -- that is something known to be deployed.
> > > >
> > > > >I understand what you are asking but I do note that some
> > > > >people have thought that the potential for AAA is much greater
> > > > >than the existing installed base.
> > > >
> > > > I actually think that the deployment argument is a very important one.
> > > >
> > > > -Jiri
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > >_______________________________________________
> > >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > >This list is for NEW development of the application of SIP
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sip@ietf.org for new developments of core SIP
> >
> > ==========================
> > Stuart Jacobs CISSP
> > PMTS - Sr. Technologist
> > Verizon Laboratories
> > 40 Sylvan Road Waltham, MA 02451-1128     USA
> > telephone: (781) 466-3076   fax: (781) 466-2838
> > stu.jacobs@labs.gte.com sjj0@labs.gte.com  stu.jacobs@verizon.com
> > ==========================
> >

==========================
Stuart Jacobs CISSP
PMTS - Sr. Technologist
Verizon Laboratories
40 Sylvan Road Waltham, MA 02451-1128    USA
telephone: (781) 466-3076  fax: (781) 466-2838
stu.jacobs@labs.gte.com sjj0@labs.gte.com stu.jacobs@verizon.com
==========================

_______________________________________________
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From mailnull@www1.ietf.org  Tue Feb 25 09:16:26 2003
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To: Stuart Jacobs <stu.jacobs@labs.gte.com>
Cc: Allison Mankin <mankin@psg.com>, sipping@ietf.org, sip@ietf.org
Subject: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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nobody replied to my bye question, but after some more thinking i have
myself come to the conclusion that it is impossible for a regular sip
proxy without a dialog state or a crypto identification of all requests
from its clients to do any accounting at all.

so it would be important for a sip-aaa requirements draft to list the
preconditions for sip accounting.

-- juha


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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Dean Willis" <dean.willis@softarmor.com>
To: <jari.arkko@piuha.net>, "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
Cc: "'Stuart Jacobs'" <stu.jacobs@labs.gte.com>,
        "'Allison Mankin'" <mankin@psg.com>, <sipping@ietf.org>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Tue, 25 Feb 2003 09:17:41 -0600
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Jari asked:
> Henry Sinnreich wrote:
> > Is there any service provider planning to use SIP-DIAMETER? If yes, 
> > please speak up.
> 
> Any of the 3G operators count for this? I'm not sure how many 
> of the subscribe to this group though, even if their vendors do...

Of the two 3G deployments I have my hands in daily, neither uses DIAMETER
for authentication. One uses a database accessible by a proxy (using
conventional database interfaces), and the other uses RADIUS. Both use
database methods for accounting.

This is not to say that they might not use DIAMETER someday. It's just not
in their existing networks, and most of todays 3G deployments are coupled to
older infrastructure elements, because that makes business sense. I think
most of the mobile operators plan on using DIAMETER in the future, some as
soon as within the next six months. Making DIAMETER work right will make
supporting a couple of hundred million subscribers much easier for them.

However, I think we can show existence proofs that RADIUS can be made to do
authentication for SIP. Since there IS a constituency, tackling this effort
might be a reasonable thing to do in the IETF. 

Having done RADIUS accounting for several large ISPs, I can also say that
only someone with a far higher pain tolerance than mine would try to use
RADIUS for time-based usage accounting. Something about unreliable record
delivery causes many annoying glitches. I don't want to go there.

--
Dean

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Jim Fleming" <JimFleming@ameritech.net>
To: "Stuart Jacobs" <stu.jacobs@labs.gte.com>
Cc: <sipping@ietf.org>, <sip@ietf.org>
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Subject: [Sipping] "...management must explore all alternatives..." ??
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From: "Stuart Jacobs" <stu.jacobs@labs.gte.com>
"...management must explore all alternatives..."
====

Do you mean "explore all alternatives" provided by the I* society...(small s...aka The Big Lie Society)...
...after ethnic cleansing has been done by the I* society thugs...?


----- Original Message ----- 
From: "Stuart Jacobs" <stu.jacobs@labs.gte.com>
To: "Allison Mankin" <mankin@psg.com>
Cc: <sipping@ietf.org>; <sip@ietf.org>
Sent: Friday, February 21, 2003 10:22 AM
Subject: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft


> Allison,
> 
> This material is part of ongoing efforts of Verizon and Verizon management 
> to engage in thoughtful considerations of the fundamental changes and 
> challenges facing the telecommunications industry. To meet its fiduciary 
> responsibilities, management must explore all alternatives, even those that 
> may appear highly speculative and hypothetical. Statements and 
> representations contained herein are preliminary and/or tentative and 
> should not be relied on unless approved by the appropriate Verizon 
> governing body.  The following represent the anticipated approach Verizon 
> most likely will use.  I cannot be more firm at this time as our 
> architectural plans are not finalized.
> 
> Verizon's current direction for our Next Generation Network (NGN) VoIP 
> Services will likely use a challenge/response approach (ala. Radius) for 
> authentication and access control between our SIP service customers and SIP 
> Proxies.  All our SIP service customers will probably go through SIP 
> Proxies.  We will not support SIP UA to SIP UA within our operational 
> domain.  Our SIP Proxies will likely authenticate themselves to other SIP 
> Proxies and other back-office systems (OSSs, Application Servers, EMSs, 
> etc.) by use of IPsec, specifically IKE combined with X.509v3 server 
> certificates and ESP transport mode Null-encryption using 
> HMAC-SHA1-96.  Access controls (rights/policies) between our intermediate 
> systems (incl. SIP Proxies, Gateways , etc.) are expected to be based on 
> use of policy certificates.  All policy and identity certificates are 
> expected to be distributed via a redundant hierarchy of LDAP servers 
> (repositories).  Accounting data will most likely be compiled by our SIP 
> Proxies and transferred to billing systems with authentication provided by 
> IPsec.  The actual billing data will probably be captured in the form of 
> XML in conjunction with XML signatures (as defined by the W3C).
> 
> Stu
> 
> At 2/20/03 05:24 PM, you wrote:
> >Stu,
> >
> >You are not doing the usage accounting with Radius, then.
> >
> >It is only doing the remote access authentication?
> >
> >We can work with this scenario.  Could you speak to this in
> >the open mailing list?  I would like to rule out Radius for
> >call detail and usage-based accounting, but it is ok for the
> >other A's.
> >
> >Allison
> >
> > > Diameter in our Next Generation Network (NGN) VoIP Services.  We will be
> > > doing our usage accounting within SIP Proxies and intend to transfer said
> > > accounting "call detail records" back to our billing OSSes via other
> > > protocols with IPsec providing authentication and confidentiality for 
> > these
> > > transfers.  We currently do not plan on any deployment of Diameter within
> > > Verizon's wireline NGN architecture.  We will probably continue to use
> > > Radius for remote access authentication at the application layer for a
> > > number or years into the future.
> > >
> > > Stu Jacobs
> > >
> > > At 2/14/03 09:58 PM, you wrote:
> > >
> > > >Jiri,
> > > >
> > > >It always seems good to be able to stay with installed base, but
> > > >there's a factor against our supporting RADIUS drafts for SIP's
> > > >accounting - safety.  Even though there is deployment, there are
> > > >some extremely bad cases of damage in production, and the IETF
> > > >has concluded that RADIUS is not suitable for usage-based accounting
> > > >(RFC2975).
> > > >
> > > >Unless there is no intention of having in SIP accounting usage-based,
> > > >monetary, auditable applications, RADIUS is not the solution.  The
> > > >protocol just does not preserve a complete stream of reliable
> > > >accounting records that are auditable on any level, especially with
> > > >any roaming.  As Bernard Aboba stated to this WG, quoting Mike O'Dell,
> > > >the use of RADIUS may even be illegal in situations where there is an
> > > >authority over the billing.  Records are too easily lost by the
> > > >protocol. This is stated clearly by RFC2975.  This is the reason why
> > > >the Operations and Mgt Area developed Diameter.  On these technical*
> > > >grounds, that the RADIUS protocol does not have enough reliability,
> > > >the work must be based on the Diameter protocol, which is now
> > > >approved.  Alternatively, work could be done that does not include
> > > >usage-based accounting, but this does not seem to be what is
> > > >envisioned in the SIPPING AAA requirements.
> > > >
> > > >Allison
> > > >
> > > >P.S. There's a current investigation of RADIUS compatibility mode
> > > >in Diameter for this - more needs to be thought about it.
> > > > >
> > > > > At 07:46 PM 2/11/2003, john.loughney@nokia.com wrote:
> > > > > >Hi Henry,
> > > > > >
> > > > > >> >but one option is develop a solution using
> > > > > >> > Diameter in RADIUS-compatibility mode.
> > > > > >>
> > > > > >> OK, this is better than nothing. We'll take what we can :-)
> > > > > >>
> > > > > >> As mentioned, the installed base is RADIUS, no matter how
> > > > > >> voluminous the DIAMETER literature may be.
> > > > > >
> > > > > >I think the issues is that the IETF should recommend what is the
> > > > > >right thing to do & why.  If people choose to do something else,
> > > > > >well, that is life.
> > > > >
> > > > > Remember that running code does not necessarily mean something known
> > > > > to compile -- that is something known to be deployed.
> > > > >
> > > > > >I understand what you are asking but I do note that some
> > > > > >people have thought that the potential for AAA is much greater
> > > > > >than the existing installed base.
> > > > >
> > > > > I actually think that the deployment argument is a very important one.
> > > > >
> > > > > -Jiri
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > >_______________________________________________
> > > >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > >This list is for NEW development of the application of SIP
> > > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > > >Use sip@ietf.org for new developments of core SIP
> > >
> > > ==========================
> > > Stuart Jacobs CISSP
> > > PMTS - Sr. Technologist
> > > Verizon Laboratories
> > > 40 Sylvan Road Waltham, MA 02451-1128     USA
> > > telephone: (781) 466-3076   fax: (781) 466-2838
> > > stu.jacobs@labs.gte.com sjj0@labs.gte.com  stu.jacobs@verizon.com
> > > ==========================
> > >
> 
> ==========================
> Stuart Jacobs CISSP
> PMTS - Sr. Technologist
> Verizon Laboratories
> 40 Sylvan Road Waltham, MA 02451-1128    USA
> telephone: (781) 466-3076  fax: (781) 466-2838
> stu.jacobs@labs.gte.com sjj0@labs.gte.com stu.jacobs@verizon.com
> ==========================
> 
> _______________________________________________
> 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

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Orit Levin <orit@radvision.com>
To: "'aroychow@hss.hns.com'" <aroychow@hss.hns.com>, shaj@cisco.com
Cc: sipping@ietf.org, sipping-admin@ietf.org
Subject: RE: [Sipping] Comments on draft-rosenberg-sipping-conferencing-fr
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I agree with most of Arjun's analysis. My comments are inline (OL:).
 
Orit.
 
-----Original Message-----
From: aroychow@hss.hns.com [mailto:aroychow@hss.hns.com] 
Sent: Saturday, February 22, 2003 10:38 AM
To: shaj@cisco.com
Cc: sipping@ietf.org; sipping-admin@ietf.org
Subject: Re: [Sipping] Comments on
draft-rosenberg-sipping-conferencing-framework-01.txt
 
---skip
 * Forward the call using the standard Refer mechanism to the new Focus with
the Unique URI. 
  
<ARC> 
 You may also want to read
http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sipping-c
c-conferencing-01.txt. 
Its provides the definition of a conference Factory URI, which could be used
to create new conferences and their Focii. 
Note that when you send an INVITE to a factory URI, that entity returns the
actual Conference-URI in the Contact header. 
---skip
</ARC> 
OL: The cc-conferencing does take into consideration this scenario i.e.
Referring to a focus by a conference factory. The statement in 3.2 "the
Conference Factory URI and the newly created focus URI MAY resolve to
different physical devices" targets the exact case. We even had this
scenario included in one of the intermediate versions of the draft. We can
restore the example if the readers find it helpful. 
 
The advantage of doing this is that I could have multiple Conference servers
that are running that are registered their master focus. 
 
We can implement Load Balancing across Conference servers without the UA
realizing it as it is not connecting to a specific Focus on a specific
server. A second advantage is that if in the course of a meeting if the
focus were to go down or the server went down the participants can attempt
to rejoin the call. In such a case the Master Focus would see that the
Conference specific focus is down and recreate it allowing users to quickly
rejoin the conference.
 
Of course one needs to work out the details of the communication and
keepalive between the Master Focus and the Conference focus it creates.
 
 OL: I agree that this is one of the possible ways to implement Load
Balancing. The Factory-Focus "decomposition" allows for that without
requiring the factory to forward SIP messages during stable sessions. I am
much less sure about the second advantage - redundancy. It is still can be
implemented with additional interface between the factory and its focuses
(as Shiva points out) but I wouldn't impose standard SIP
procedures/conventions in order to implement it - there are so many
(proprietary and standard) ways to address redundancy that I wouldn't put it
as a requirement for SIP conferencing.
 
...skip
Instead of dealing with Conference aware UA's and Conference Unaware UA's we
could assume all participants in the Conference to be Conference Aware. 
...skip
<ARC> 
 It is extremely important to acknowledge conference-unaware endpoints. 
...skip
</ARC> 
OL: One of the first requirements (in the conferencing-requirements draft)
is to be able to accommodate basic UAs in a SIP conference. That being said
based on the chosen approach (and the examined scenarios) on the SIP level
there is no need for differentiation between conference-aware and
conference-unaware participants neither for a conference factory nor for a
focus. 
...skip
<ARC> 
...skip
 Now a question again for the authors of
draft-johnston-sipping-cc-conferencing-01.txt: 

* section 4.5 callflow - you show alice sending an invite to a conf-factory
URI but the ladder diagram shows it being received by a 'Focus'. 
The definition of a focus 


3.1 Focus UA

  A focus, as defined in the framework, hosts a SIP conference and
  maintains a SIP signaling relationship with each participant in the
  conference. A focus contains a conference-aware user agent that
  supports the conferencing call control conventions as defined in this
  document. 

Since a 'Focus' hosts 'a specific' conference, I find it a little
disconcerting 
that the call flows show messages with a factory-URI reaching a focus. I
thought 
it may be clearer, if we limit a 'focus' to be the target of a specific URI
meant 
for a specific conference. If a person sends an INVITE to a factory-URI or 
a 'conf-app-URI' < like an IVR etc which then contacts a Factory after auth
etc> 
it should not be received by a Focus, should it ? 

</ARC>
OL:  I agree with this comment - the picture on its own is "disconcerting".
In the shown scenario the factory and the focus are logical entities that
located in a single physical device (called "focus" in the picture). We will
either rearrange the drawing or add a text clarifying the case.
Thanks you, guys, for all your comments,
Orit. 

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<body lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:.5in'>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I agree with most of <span =
class=3DSpellE>Arjun's</span>
analysis. My comments are inline (<span =
class=3DGramE>OL:</span>).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>=


<div>

<p class=3DMsoAutoSig><i><font size=3D3 color=3Dnavy face=3D"Courier =
New"><span
style=3D'font-size:12.0pt;font-family:"Courier =
New";color:navy;font-style:italic;
mso-no-proof:yes'>Orit.</span></font></i><font color=3Dnavy><span
style=3D'color:navy;mso-no-proof:yes'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>=


<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<p class=3DMsoNormal><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> =
aroychow@hss.hns.com
[mailto:aroychow@hss.hns.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Saturday, February =
22, 2003
10:38 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> shaj@cisco.com<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> =
</span></font><st1:PersonName><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>sipping@ietf.org</span></f=
ont></st1:PersonName><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>; =
</span></font><st1:PersonName><font
 size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>sipping-admin@ietf.org</sp=
an></font></st1:PersonName><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [Sipping] =
Comments on
draft-rosenberg-sipping-conferencing-framework-01.txt</span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 =
color=3Dnavy
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:navy'><o:p>&nbsp;</o:p></span></font></p=
>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>---skip</span></=
font><br>
<span style=3D'mso-spacerun:yes'>&nbsp;</span>* Forward the call using =
the
standard Refer mechanism to the new Focus with the Unique URI. <br>
&nbsp; <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&lt;ARC&gt;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;<span
class=3DGramE>You</span> may also want to read
http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sippi=
ng-cc-conferencing-01.txt.</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Its
provides the definition of a conference Factory URI, which could be =
used to
create new conferences and their Focii.</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Note
that when you send an INVITE to a factory URI, that entity returns the =
actual
Conference-URI in the Contact header.</span></font> <font =
color=3Dnavy><span
style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>---skip<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>&lt;/ARC&gt;</span></font> =
<font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:
10.0pt'>OL: The cc-conferencing does take into consideration this =
scenario i.e.
<span class=3DGramE>Referring</span> to a focus by a conference =
factory. The
statement in 3.2 "the Conference Factory URI and the newly created =
focus
URI MAY resolve to different physical devices" targets the exact case. =
We
even had this scenario included in one of the intermediate versions of =
the draft.
We can restore the example if the readers find it helpful. =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The advantage of doing this is that I could have =
multiple
Conference servers that are running that are registered their master =
focus. </span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>We can implement Load Balancing across Conference =
servers
without the UA realizing it&nbsp;as it is not&nbsp;connecting to a =
specific
Focus on a specific server. A second advantage is that if in the course =
of a
meeting if the focus were to go down or the server went down the =
participants
can attempt to rejoin the call. In such a case the Master Focus would =
see that
the Conference specific focus is down and recreate it allowing users to =
quickly
rejoin the conference.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Of course one needs to work out the details of the
communication and <span class=3DSpellE>keepalive</span> between the =
Master Focus
and the Conference focus it creates.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;OL: I agree that this is one of the possible ways to =
implement
Load Balancing. The Factory-Focus "decomposition" allows for that
without requiring the factory to forward SIP messages during stable =
sessions. I
am much less sure about the second advantage - redundancy. It is still
can be implemented with additional interface between the factory and =
its focuses
(as Shiva points out) but I wouldn't impose standard SIP =
procedures/conventions
in order to implement it - there are so many (proprietary and standard) =
ways
to address redundancy that I wouldn't put it as a requirement for SIP
conferencing.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>...skip<br>
</span></font><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial'>Instead of dealing with Conference aware UA's and Conference =
Unaware
UA's we could assume all participants in the Conference to be =
Conference Aware.
<font color=3Dnavy><span =
style=3D'color:navy'><o:p></o:p></span></font></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>...skip<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>&lt;ARC&gt;</span></font> =
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;<span
class=3DGramE>It</span> is extremely important to acknowledge =
conference-unaware
endpoints. <font color=3Dnavy><span =
style=3D'color:navy'><o:p></o:p></span></font></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>...skip<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>&lt;/ARC&gt;</span></font> =
<font
color=3Dnavy><span style=3D'color:navy'><o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>OL: One
of the first requirements (in the conferencing-requirements draft) is =
to be
able to accommodate basic <span class=3DSpellE>UAs</span> in a SIP =
conference.
That being said based on the chosen approach (and the examined =
scenarios) on
the SIP level there is no need for differentiation between =
conference-aware and
conference-unaware participants neither for a conference factory nor =
for a focus.
<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
color=3Dnavy
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:navy'>...skip<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>&lt;ARC&gt;</span></font> =
<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>...skip<br>
<span style=3D'mso-spacerun:yes'>&nbsp;</span></span></font><font =
size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>Now a =
question
again for the authors of =
&nbsp;draft-johnston-sipping-cc-conferencing-01.txt:</span></font>
<br>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>*
section 4.5 callflow - you show alice sending an invite to a =
conf-factory URI
but the ladder diagram shows it being received by a =
'Focus'.</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>The
definition of a focus </span></font><br>
<br>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>3.1
Focus UA<br>
<br>
&nbsp; A focus, as defined in the framework, hosts a SIP conference =
and<br>
&nbsp; maintains a SIP signaling relationship with each participant in =
the<br>
&nbsp; conference. A focus contains a conference-aware user agent =
that<br>
&nbsp; supports the conferencing call control conventions as defined in =
this<br>
&nbsp; document.</span></font> <br>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Since
a 'Focus' hosts 'a specific' conference, I find it a little =
disconcerting</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>that
the call flows show messages with a factory-URI reaching a focus. I =
thought</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>it may
be clearer, if we limit a 'focus' to be the target of a specific URI =
meant</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>for a
specific conference. If a person sends an INVITE to a factory-URI =
or</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>a
'conf-app-URI' &lt; like an IVR etc which then contacts a Factory after =
auth
etc&gt;</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>it
should not be received by a Focus, should it ?</span></font> <br>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&lt;/ARC&gt;<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>OL:</span></font> <span
style=3D'mso-spacerun:yes'>&nbsp;</span>I agree with this comment - the =
picture
on its own is "disconcerting". In the shown scenario the factory
and the focus are logical entities that located in a single physical =
device (called
"focus" in the picture). We will either rearrange the drawing or
add a text clarifying the case.<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Thanks you, =
guys, for all
your comments,<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Orit. =
<o:p></o:p></span></font></p>

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Subject: [Sipping] new I-D on the "grand unified theory of nat traversal"
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Folks,

If you take a look at draft-ietf-sipping-nat-scenarios, the general 
conclusoin you come to is "what a mess!". There are a million different 
solutions, and some of them work in some cases some times. There is no 
unifying mechanism which works very well in almost all of the cases. I 
have been working on a methodology for integrating the various pieces 
into a semi-coherent story about what clients need to do in order to 
handle a wide variety of situations. I have written it up, and submitted 
as an I-D. Until it appears in the archives, you can pick it up at:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-ice-00.txt
http://www.jdrosen.net/papers/draft-rosenberg-sipping-ice-00.html

Its called Interactive Connectivity Establishment (ICE). Its a bit rough 
around the edges, and needs a lot more examples and use cases, but 
hopefully you will get the idea.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Sipping] I-D ACTION:draft-wu-sipping-floor-control-03.txt
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Hi,

the definition of the FL semantics for the grouping framework needs more
elaboration. You need to say whether or not only one control line is
allowed per group and include an IANA considerations section as follows:

IANA Considerations

   IANA needs to register the following new "semantics" attribute for
   the SDP grouping framework [RFC3388]:

   Semantics                  Token      Reference
   -------------------        -----      ---------
   Floor control              FL         [RFCxxxx]


   It should be registered in the SDP parameters registry
   (http://www.iana.org/assignments/sdp-parameters) under Semantics for
   the "group" SDP Attribute.



You should also add IANA considerations for a=moderate


In Section 8.1, you write: "the conference server uses a=moderated line
to mute all media tools of the participant". Does the conference server
set to inactive the SDP direction attributes as well or it leaves them
as they are (e.g., sendrecv)?

Thanks,

Gonzalo



Internet-Drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> 
>         Title           : Use of Session Initiation Protocol (SIP) and Simple
>                           Object Access
>                           Protocol (SOAP) for Conference Floor Control
>         Author(s)       : X. Wu et al.
>         Filename        : draft-wu-sipping-floor-control-03.txt
>         Pages           : 33
>         Date            : 2003-1-15
> 
> During a conference, floor control coordinates simultaneous access to
> shared resource in multimedia conferences. Floor control allows
> applications and users to gain safe and mutually exclusive or non-
> exclusive access to the shared resources. This document defines an
> approach of using Session Initiation Protocol (SIP) event
> notification mechanism and Simple Object Access Protocol (SOAP) to
> perform floor control.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-wu-sipping-floor-control-03.txt
> 
> To remove yourself from the IETF Announcement list, send a message to
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> 
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-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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Hello,

I have noticed that the SIP-QSIG draft describes how to use overlap
signalling in a SIP network.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt

I believe that this draft should not describe overlap procedures. It
should reference the overlap mapping draft instead:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt

A short section explaining the differences between overlap in QSIG and
in ISUP (e.g., INFORMATION message instead of SAM) is OK, but I would
not like to see the whole overlap mapping explained here again. We have
gone thru some pain adding several considerations and motivational text
to the overlap draft so that we have rough consensus on it, and we do
not need to re-create that here again.

Thanks,

Gonzalo
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From: "Elwell, John" <john.elwell@siemens.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@lmf.ericsson.se>,
        frank.derks@philips.com, olivier.rousseau@col.bsf.alcatel.fr,
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Gonzallo,

Well, we tried to do this originally, but the overlap draft was a moving
target and, as you point out, was ISUP-specific rather than generic. Even
now the overlap draft is still work in progress rather than an RFC - when do
you envisage completion?

John Elwell - Technical Consultant
Tel: +44 (0)115 943 4989
Fax: +44 (0)115 943 4969
e-mail: mailto:john.elwell@siemens.com
web: http://www.siemens-design-services.com

Siemens Communications
Technology Drive, Beeston, Nottingham, NG9 1LA.  England.

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> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: 26 February 2003 11:25
> To: john.elwell@siemens.com; frank.derks@philips.com;
> olivier.rousseau@col.bsf.alcatel.fr; patrick.mourot@sxb.bsf.alcatel.fr
> Cc: Jon Peterson; sipping
> Subject: Overlap in QSIG draft
> 
> 
> Hello,
> 
> I have noticed that the SIP-QSIG draft describes how to use overlap
> signalling in a SIP network.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
> 
> I believe that this draft should not describe overlap procedures. It
> should reference the overlap mapping draft instead:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt
> 
> A short section explaining the differences between overlap in QSIG and
> in ISUP (e.g., INFORMATION message instead of SAM) is OK, but I would
> not like to see the whole overlap mapping explained here 
> again. We have
> gone thru some pain adding several considerations and 
> motivational text
> to the overlap draft so that we have rough consensus on it, and we do
> not need to re-create that here again.
> 
> Thanks,
> 
> Gonzalo
> 
_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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To: "Elwell, John" <john.elwell@siemens.com>
CC: frank.derks@philips.com, olivier.rousseau@col.bsf.alcatel.fr,
        patrick.mourot@sxb.bsf.alcatel.fr,
        Jon Peterson <Jon.Peterson@neustar.com>, sipping <sipping@ietf.org>
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Subject: [Sipping] Re: Overlap in QSIG draft
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John,

the last version of the draft (which was released two or three weeks
ago) will be presented to the IESG shortly. I believe that we have
addressed all the issues that were raised... the document could be
approved in a reasonably short period of time. 

We'll keep you updated.

Gonzalo



"Elwell, John" wrote:
> 
> Gonzallo,
> 
> Well, we tried to do this originally, but the overlap draft was a moving
> target and, as you point out, was ISUP-specific rather than generic. Even
> now the overlap draft is still work in progress rather than an RFC - when do
> you envisage completion?
> 
> John Elwell - Technical Consultant
> Tel: +44 (0)115 943 4989
> Fax: +44 (0)115 943 4969
> e-mail: mailto:john.elwell@siemens.com
> web: http://www.siemens-design-services.com
> 
> Siemens Communications
> Technology Drive, Beeston, Nottingham, NG9 1LA.  England.
> 
> "Siemens Communications - a division of Siemens plc,  Registered No:
> 727817, England. Registered office: Siemens House, Oldbury, Bracknell,
> Berkshire, RG12 8FZ.
> This communication contains information which is confidential and may
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> prohibited. If you have received this communication in error, please
> contact us immediately and also delete the communication from your
> computer. We accept no liability for any loss or damage suffered by any
> person arising from use of this e-mail."
> 
> > -----Original Message-----
> > From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> > Sent: 26 February 2003 11:25
> > To: john.elwell@siemens.com; frank.derks@philips.com;
> > olivier.rousseau@col.bsf.alcatel.fr; patrick.mourot@sxb.bsf.alcatel.fr
> > Cc: Jon Peterson; sipping
> > Subject: Overlap in QSIG draft
> >
> >
> > Hello,
> >
> > I have noticed that the SIP-QSIG draft describes how to use overlap
> > signalling in a SIP network.
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
> >
> > I believe that this draft should not describe overlap procedures. It
> > should reference the overlap mapping draft instead:
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt
> >
> > A short section explaining the differences between overlap in QSIG and
> > in ISUP (e.g., INFORMATION message instead of SAM) is OK, but I would
> > not like to see the whole overlap mapping explained here
> > again. We have
> > gone thru some pain adding several considerations and
> > motivational text
> > to the overlap draft so that we have rough consensus on it, and we do
> > not need to re-create that here again.
> >
> > Thanks,
> >
> > Gonzalo
> >

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Requirements for Persistent Connection Management in 
                          the Session Initiation Protocol (SIP)
	Author(s)	: R. Jain, V. Gurbani
	Filename	: draft-jain-sipping-persistent-conn-reqs-00.txt
	Pages		: 15
	Date		: 2003-2-25
	
SIP over connection-oriented transport protocol based systems are 
likely to face certain distinct performance and behavioral issues 
that are not manifest when SIP is transported over connectionless 
protocols. Allowing SIP entities to mutually conserve connections 
over a predictable, extended period of time is one of the leading 
requirements to help SIP entities deliver their optimal 
performance in the network. Overall, this document contemplates 
transport layer connection management issues relating to SIP. 
Requirements and potential solutions for introducing a backward 
compatible notion of persistent connections in SIP are presented.

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Date: Wed, 26 Feb 2003 10:04:22 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
In-reply-to: <3E5CC3DB.20706@polytech.univ-nantes.fr>
To: "'Antoine Duval'" <antoine.duval@polytech.univ-nantes.fr>, sip@ietf.org,
        sipping@ietf.org
Cc: Alan Johnston <alan.johnston@wcom.com>, Rohan Mahy <rohan@cisco.com>,
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Subject: [Sipping] RE: [Sip] whiteboard
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I believe the shared whiteboard is of special interest to the
conferencing design team of the SIPPING WG - where this belongs BTW.

Thanks, Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA
 

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Antoine Duval
> Sent: Wednesday, February 26, 2003 7:41 AM
> To: sip@ietf.org
> Subject: [Sip] whiteboard
> 
> 
> Hi everybody !
> 
> I'm searching information for developping a shared whiteboard 
> (protocol 
> and application) with sip, but i'm not satisfied of results...
> 
> That why, i'm posting this email here, in case you can help me.
> 
> Any idea ?
> 
> Antoine
> 	
> 
> _______________________________________________
> 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
> 

_______________________________________________
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Thread-Topic: [Sipforum-discussion] SIP Telephony Device Requirements, Configuration and Data
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Hi Henry,

Here is a somewhat lengthy email with a few comments about the SIP Telephony Device Requirements draft:


  SIP-2, DNS SRV: SIP clients MUST t support RFC2782 [3] and MAY 
  support the SIP service examples [4] for basic PBX-like or Centrex-
  like functions.  SIP clients MUST support DNS SRV for locating a SIP 
  Server. When making a SRV query, the client MUST t use the DNS 
  additional information in the response that contains the IP addresses 
  for the A records. 
     
  If the DNS additional information is not present, the client MUST 
  make DNS A record queries to resolve the hostnames.

[EricT]
I suggest referencing RFC 3263 (Locating SIP Servers) in the above section. I don't know what services are being referred to when saying "and MAY support the SIP service examples [4] for basic PBX-like or Centrex-like functions.".  This section is about DNS SRV, but I found nothing in the latest SIP Services Examples draft about DNS SRV...


  SIP-3, Do Not Disturb: SIP clients MUST be able to set the state of 
  the client as Do Not Disturb.  Clients MUST respond to new INVITES 
  with a ô486 Busy Hereö.  Clients MUST respond to re-INVITES on 
  existing call legs as normal behavior [4]. 

[EricT]
I suggest rewording to something like "It must be possible for the SIP device user to configure the device state to Do Not Disturb. ...", as it is generally the user that sets the DND state ;)

  SIP-4, call hold resume: SIP clients MUST be able to place a call on 
  hold/resume [4].

[EricT]
Should it focus on the usage of RFC 3264, with the use of a=sendonly instead of c=0.0.0.0? Also with a focus that the answer does not automatically comes with a "held" sdp? I suggest adding something like "SIP clients MUST follow RFC 3264 when placing a call on hold. More specifically, the a=sendonly attribute MUST be used. The SDP answer of SIP clients that are being placed on hold MUST NOT contain "held" SDP, unless the user session was originally on hold."

  SIP-7, message waiting indicator: SIP clients MUST support SIP 
  message waiting  and an indicator for integration with voicemail 
  platforms [5]. 

[EricT]
This is kind of hard, since there is not RFC yet for this and different vendor/integrators are using different draft versions which sometimes do not interoperate.  Should maybe add a note to this effect... Maybe something like "No RFC exist yet for this service. SIP clients MUST implement this service in its RFC form when it becomes available."

  SIP-9, transfer: SIP devices MUST support REFER and NOTIFY as 
  required to support the transfer [6].  SIP clients MAY support 
  escaped headers in the Refer-To: header.

[EricT]
Same problem as SIP-7...  I suggest upgrading the "MAY support escaped headers..." to MUST, as not supporting this has impact on attended call transfer.


  SIP-11, attended call transfer: SIP devices MUST support attended 
  call transfer [3], [4].  SIP clients MAY support escaped headers in 
  the Refer-To: header. 

[EricT]
Reference [3] makes no sense here ;) Should probably be [6]. 

  SIP-12, music on hold: SIP devices MUST support music on hold [4]. 

[EricT]
Not sure why this service is a MUST? Maybe in enterprise environments, but what about homes?


  SIP-15: DTMF RTP payload: SIP clients MUST be able to send DTMF 
  specified by RFC2833 [7].   
   
  Payload type negotiation MUST comply with RFC 3264 [8]. 
   
  Payload type MUST remain constant throughout the session.  For 
  example, if an endpoint decides to renegotiate codecs or put the call 
  on hold, the payload type for the re-invite MUST be the same as 
  initial payload type. 

[EricT]
Should probably refer to RFC 3388. Maybe add "SIP devices SHOULD support Flow Identification (FID) as defined in RFC 3388."

  SIP-20, reason phrase display: If the ôReason Phraseö of a response 
  message is displayed, the client MUST use ôReason Phraseö in the 
  response packet.  The client MUST not use an internal ôStatus Codeö 
  table. 
   
  Another SIP device MAY respond with ô403 Forbiddenö, or various other 
  messages. 

[EricT]
I am not sure about the above two MUST. Something tells me that the client could rely on an internal table if there was a problem with language negotiation...  I don't know what the last sentence means??? "Another SIP device MAY ..." It makes no sense to me...

  SIP-25, Dynamic login/logout for user mobility: SIP devices MUST 
  support user mobility.  SIP clients MUST store a static profile in 
  non-volatile memory so that this information is available during the 
  power up sequence.  SIP clients MUST allow a user to walk up to a 
  client, login, and be able to send and receive calls with his profile 
  information. 
  For emergency numbers (e.g. 911) the client MUST send the credentials 
  (username/password) of the static profile. 

[EricT]
We are developing analog adapters and don't remember receiving such a request. I suggest either changing the MUST to SHOULD or adding some text to specify that it is a MUST for devices with the proper interface to do it and a SHOULD for devices with a limited interface. I know we could possibly configure this through a web interface, but the point is that there are a lot of vendore that currently just want to replace the wire, not the phone and its interface...

  App-5, SIMPLE Integration: SIP devices MUST provide a ôbuddy-
  list/addressbookö and use SIP extensions to leverage presence [21]. 

[EricT]
This also does not applies well to black phones applications. I suggest changing this to "MAY".


  FW/NAT-2, NAT capable configuration: SIP devices MUST be able to 
  operate behind a static NAT/PAT (Network Address Translation/Port 
  Address Translation) device using the STUN protocol [x22].  SIP 
  clients MUST be able to populate SIP messages with the public, 
  external address of the NAT/PAT device and use specific port ranges 
  for RTP. 
   
  FW/NAT-3, UPnP capable configuration 
  SIP devices MUST be able to operate with a UPnP 
  (http://www.upnp.org/) firewall device. 

[EricT]
I suggest using a MUST on only one of these, and a MAY on the other. I am not fully aware of all details about these two protocols, but I think a device could very well work behind a firewall by using only one mechanism. I think the draft should clearly state which mechanism is preferred.

As a last point, I suggest splitting this draft into two or more drafts. One about generic requirements for SIP devices, and one (or more) about the configuration mechanism along with the data that can be configured. The first draft could refer the second one.  The draft, as it is right now, defines basic requirements, then some configuration framework and its requirement, and then data (and data representation). I don't think those mix well. Focusing would probably make the documents easier to maintain ;)

I might come back later with more comments on configuration/data.

Best Regards,

EricT



-----Original Message-----
From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
Sent: Monday, February 10, 2003 9:49
To: 'Jörgen Björkner'; sipforum-discussion@lists.su.se
Subject: [Sipforum-discussion] SIP Telephony Device Requirements,
Configuration and Data


For everyone interested SIP phones and other SIP communication devices,
and the 56 IETF  coming next month, please see if you like or dislike or
don't care about the draft:

http://www.ietf.org/internet-drafts/draft-sinnreich-sipdev-req-00.txt

The idea is to have discussions on the lists, and if there is interest
and rough consensus to move ahead with an informational RFC. Would
appreciate your comments to 

sipping@ietf.org , and also to

sipforum-discussion@lists.su.se 

Thanks, Henry

Henry Sinnreich
WorldCom
400 International Parkway
Richardson, Texas 75081
USA


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: jh@lohi.eng.song.fi, Stuart Jacobs <stu.jacobs@labs.gte.com>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
Cc: Allison Mankin <mankin@psg.com>, sipping@ietf.org, sip@ietf.org
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Sorry for latency -- yes, I think that is a very important issue,
and mentioning it in the draft would be very good. (Has anyone else 
made the observation too that operationally-related questions are 
rarely replied here?)

I agree too it takes awareness of dialog state. It doesn't imply though,
a proxy needs to be stateful -- record-routing can help, most likely
along with some message integrity check if the dialog information
is sensitive (which is most likely the case with AAA).

-jiri

At 03:16 PM 2/25/2003, jh@lohi.eng.song.fi wrote:
>nobody replied to my bye question, but after some more thinking i have
>myself come to the conclusion that it is impossible for a regular sip
>proxy without a dialog state or a crypto identification of all requests
>from its clients to do any accounting at all.
>
>so it would be important for a sip-aaa requirements draft to list the
>preconditions for sip accounting.
>
>-- juha
>
>
>_______________________________________________
>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 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        Allison Mankin <mankin@psg.com>, sipping@ietf.org
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
References: <5.1.1.6.0.20030221110917.04229140@pophost.gte.com>
	 <5.1.1.6.0.20030221110917.04229140@pophost.gte.com> <5.2.0.9.0.20030226125028.00b3e288@mailhost.fokus.gmd.de>
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Juha,

[removing SIP (but keeping SIPPING) from the cc: ]

Do you mean that you would like to see a note along these lines added to
the draft?

"Note that, in order to perform accounting, a SIP proxy server should
see and authenticate every message sent over the dialog."

Gonzalo

Jiri Kuthan wrote:
> 
> Sorry for latency -- yes, I think that is a very important issue,
> and mentioning it in the draft would be very good. (Has anyone else
> made the observation too that operationally-related questions are
> rarely replied here?)
> 
> I agree too it takes awareness of dialog state. It doesn't imply though,
> a proxy needs to be stateful -- record-routing can help, most likely
> along with some message integrity check if the dialog information
> is sensitive (which is most likely the case with AAA).
> 
> -jiri
> 
> At 03:16 PM 2/25/2003, jh@lohi.eng.song.fi wrote:
> >nobody replied to my bye question, but after some more thinking i have
> >myself come to the conclusion that it is impossible for a regular sip
> >proxy without a dialog state or a crypto identification of all requests
> >from its clients to do any accounting at all.
> >
> >so it would be important for a sip-aaa requirements draft to list the
> >preconditions for sip accounting.
> >
> >-- juha
> >
> >
> >_______________________________________________
> >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
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Cc: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>,
        Stuart Jacobs <stu.jacobs@labs.gte.com>,
        Allison Mankin <mankin@psg.com>, sipping@ietf.org
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Gonzalo Camarillo writes:

 > Do you mean that you would like to see a note along these lines added to
 > the draft?
 > 
 > "Note that, in order to perform accounting, a SIP proxy server should
 > see and authenticate every message sent over the dialog."

unfortunately in most common cases the sip proxy cannot authenticate
every message.  in order to be able to account a request, the proxy
should be able to either authenticate it (if it was send by its own
user) or know that i came from a trusted another proxy that has already
authenticated the request.

in order to know that the requstest was sent by its own user, the proxy
should either be dialog aware or have somekind of security association
with its user.  neither of these is possible with products that are in
common use today.

-- juha

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: sipping@ietf.org
Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
Date: Thu, 27 Feb 2003 13:21:06 +0100
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 From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Thursday, February 27, 2003 9:28 AM
> To: Jiri Kuthan
>
> Juha,
>
> Do you mean that you would like to see a note along these 
> lines added to
> the draft?
> 
> "Note that, in order to perform accounting, a SIP proxy server should
> see and authenticate every message sent over the dialog."
> 
I don't think it's possible to do reliable accounting within a proxy
if any or both of the UAs are cheating. The caller can deliberately
ignore Record-Route and bypass the proxy when sending an INVITE. The
callee can send broken BYEs that will be discarded by the caller but
will be accepted by the accounting proxy. 

What kind of accounting information do we want to collect: The number
of SIP messages per user? This is possible with an accounting proxy.
Call minutes? How does the proxy know what kind of session is being
established if the SDP is encrypted? Only the endpoints and the bandwidth
reservation infrastructure can do this kind of accounting.

The only way I see is to include some form of micropayment tokens ("postage stamps")
in SIP messages. draft-johnston-sip-osp-token-04.txt describes such a token.
Periodical session updates containing tokens prevent disputes about the actual
length of the call. If the callee has tokens, he can charge for the call. The
trusted third party that generates those tokens can be an OSP server or may be
a proxy that inserts some header.

(see http://www.ietf.org/internet-drafts/draft-beck-sipping-svc-charging-req-01.txt)


--
Wolfgang Beck
T-Systems Nova GmbH 

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Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
Date: Thu, 27 Feb 2003 14:36:48 +0200
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Thread-Topic: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Hi,

In the end, doesn't this also depend heavily on what the purpose for collecting accounting data is? If you're only doing accounting for the purposes of traffic analysis or logging, i.e., event based accounting at the proxy, I doubt that authentication or origin of the messages is real critical. 

On the other hand, if you're collecting data for charging purposes, then authentication and integrity of the SIP messages is critical. However, hasn't billing based on accounting data collected by proxies been discouraged on this list several times before?

Cheers,
Aki

 > -----Original Message-----
 > From: ext jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
 > Sent: 27 February, 2003 13:58
 > To: Gonzalo Camarillo
 > Cc: Jiri Kuthan; Stuart Jacobs; Allison Mankin; sipping@ietf.org
 > Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA 
 > reqs draft
 > 
 > 
 > Gonzalo Camarillo writes:
 > 
 >  > Do you mean that you would like to see a note along these 
 > lines added to
 >  > the draft?
 >  > 
 >  > "Note that, in order to perform accounting, a SIP proxy 
 > server should
 >  > see and authenticate every message sent over the dialog."
 > 
 > unfortunately in most common cases the sip proxy cannot authenticate
 > every message.  in order to be able to account a request, the proxy
 > should be able to either authenticate it (if it was send by its own
 > user) or know that i came from a trusted another proxy that 
 > has already
 > authenticated the request.
 > 
 > in order to know that the requstest was sent by its own 
 > user, the proxy
 > should either be dialog aware or have somekind of security 
 > association
 > with its user.  neither of these is possible with products 
 > that are in
 > common use today.
 > 
 > -- juha
 > 
 > _______________________________________________
 > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
 > This list is for NEW development of the application of SIP
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sip@ietf.org for new developments of core SIP
 > 
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Beck01, Wolfgang writes:

 > What kind of accounting information do we want to collect: The number
 > > of SIP messages per user? This is possible with an accounting
 > proxy.

unfortunately not.

-- juha

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	Title		: Certificate Discover for SIP
	Author(s)	: C. Jennings
	Filename	: draft-jennings-sipping-certs-00.txt
	Pages		: 12
	Date		: 2003-2-26
	
This draft describes a scheme where SIP user agents can create self
signed certificates for use with the SIP S/MIME mechanism and can
then store the certificate on a web server associated with the
address of record (AOR) for the user. Other user agents that want to
call that AOR can retrieve these certificates from the web server.
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possible to have a reasonable degree of confidence of the identities
of the parties in a SIP session.

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	Title		: Directed Transcoding
	Author(s)	: E. Burger
	Filename	: draft-burger-sipping-dir-transcode-00.txt
	Pages		: 17
	Date		: 2003-2-26
	
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CC: sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Thu, 27 Feb 2003 07:43:01 -0500
Subject: [Sipping] I-D ACTION:draft-peterson-sipping-ieprep-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Considerations on the IEPREP requirements for SIP
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sipping-ieprep-00.txt
	Pages		: 10
	Date		: 2003-2-26
	
The IEPREP working group has provided the SIPPING WG with a framework
and requirements draft for prioritized communications services in
SIP.  The considerations raised in that draft are examined here, and
some feedback and implementation recommendations based on its
requirements are provided.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-peterson-sipping-ieprep-00.txt

To remove yourself from the IETF Announcement list, send a message to 
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Internet-Drafts are also available by anonymous FTP. Login with the username
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	"get draft-peterson-sipping-ieprep-00.txt".

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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
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	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
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--NextPart
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--OtherAccess
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Thu Feb 27 07:47:00 2003
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CC: sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Thu, 27 Feb 2003 07:43:25 -0500
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Interactive Connectivity Establishment (ICE): A 
                          Methodology for Nettwork Address Translator (NAT)    
                          Traversal for the Session Initiation
                          Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-rosenberg-sipping-ice-00.txt
	Pages		: 35
	Date		: 2003-2-26
	
This document describes a methodology for Network Address Translator
(NAT) traversal for the Session Initiation Protocol (SIP). This
methodology is called Interactive Connectivity Establishment (ICE).
ICE is not a new protocol, but rather makes use of existing
protocols, such as Simple Traversal of UDP Through NAT (STUN),
Traversal Using Relay NAT (TURN) and even Real Specific IP (RSIP).
ICE works through the mutual cooperation of both endpoints in a SIP
dialog. By having the endpoints work together in NAT traversal, a
number of important properties are obtained. ICE always works,
independent of the types or number of NATs. It always represents the
cheapest solution for a carrier. It always results in the minimum
voice latency. It can be done with no increase in call setup delays.
It is far less brittle than STUN. ICE also facilitates the transition
of the Internet from IPv4 to IPv6, supporting calls between
dual-stack and v6 clients behind a 4to6 NAT. Preconditions can be
used in conjunction with ICE, to guarantee that the phone never rings
unless the users will both hear and see each other when they pick up.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-ice-00.txt

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type "cd internet-drafts" and then
	"get draft-rosenberg-sipping-ice-00.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Send a message to:
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In the body type:
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NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
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--NextPart
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--OtherAccess
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Content-Type: text/plain
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--OtherAccess
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	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

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--NextPart--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Thu Feb 27 07:49:19 2003
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From: Internet-Drafts@ietf.org
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Date: Thu, 27 Feb 2003 07:43:30 -0500
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Interworking between Digital Subscriber Signalling 
                          System No. one (DSS1) and Session Initiation
	Author(s)	: P. Kotar
	Filename	: draft-kotar-sipping-dss1-sip-iw-00.txt
	Pages		: 36
	Date		: 2003-2-26
	
This document specifies the interworking between the Digital Subscriber Signalling System No.one (DSS1) as specified in [1] and the Session Initiation Protocol (SIP) as specified in [2]. 
SIP is an Internet application-layer control protocol for creating,     modifying, and terminating sessions with one or more participants. These sessions include, in particular, telephone calls. DSS1 is a signalling protocol for establishing, modifying and terminating circuit-switched calls, in particular telephone calls, within Public Integrated Services Digital Networks (ISDN)and between the Public network and Private Networks. DSS1 is specified in a number of ETSI Standards and published also as ITU-T standards. 
The interworking SHALL be described from DSS1 viewpoint for the basic call procedures. 
The interworking between the above signalling protocols occurs in a Media Gateway Controller (MGC), also called Softswitch which contains the protocol  functionality for the ISDN subscribers and the SIP protocol functionality.
In case of contradictions between the mapping described in the present document and the requirements specified in the corresponding DSS1 standards and SIP RFC , the procedures in the DSS1 standards and SIP RFC applies.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-kotar-sipping-dss1-sip-iw-00.txt

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Send a message to:
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NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
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--NextPart
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--OtherAccess
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Content-Type: text/plain
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ENCODING mime
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--OtherAccess
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	site="ftp.ietf.org";
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	directory="internet-drafts"

Content-Type: text/plain
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--NextPart--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From mailnull@www1.ietf.org  Thu Feb 27 07:49:37 2003
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CC: sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Thu, 27 Feb 2003 07:44:54 -0500
Subject: [Sipping] I-D ACTION:draft-rosenberg-sipping-callerprefs-usecases-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Use Cases for Session Initiation Protocol (SIP) Caller
                          Preferences
	Author(s)	: J. Rosenberg, P. Kyzivat
	Filename	: draft-rosenberg-sipping-callerprefs-usecases-00.txt
	Pages		: 24
	Date		: 2003-2-26
	
This document describes a set of use cases for the SIP Caller
Preferences and Callee Capabilities extension. Each use case is
presented as a desired routing decision, followed by the actual
headers and processing logic which would result in that decision.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-callerprefs-usecases-00.txt

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type "cd internet-drafts" and then
	"get draft-rosenberg-sipping-callerprefs-usecases-00.txt".

A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
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NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
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--OtherAccess
Content-Type: Message/External-body;
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Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-rosenberg-sipping-callerprefs-usecases-00.txt

--OtherAccess
Content-Type: Message/External-body;
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	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: jh@lohi.eng.song.fi
Cc: sipping@ietf.org
Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> Sent: Thursday, February 27, 2003 1:39 PM
> To: Beck, Wolfgang
> Cc: sipping@ietf.org; stu.jacobs@labs.gte.com; mankin@psg.com
> Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Beck01, Wolfgang writes:
> 
>  > What kind of accounting information do we want to collect: 
> The number
>  > > of SIP messages per user? This is possible with an accounting
>  > proxy.
> 
> unfortunately not.
well, at least in theory: the UAs have TLS connections to their proxies, the
proxies trust each other and the ingress proxies insert P-Asserted-Identity
headers.

I wonder if it is really useful to count SIP messages per user on all proxies
in the path. Can you give some examples? 

In the router world, we only do accounting on the first hop where there
is some kind of security association, the 'sealed CPE' or PPP link. We don't
do IP accounting in the core.

--
Wolfgang Beck
T-Systems Nova GmbH 
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Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Beck01, Wolfgang writes:

 > well, at least in theory: the UAs have TLS connections to their
 > proxies, the  > proxies trust each other and the ingress proxies
 > insert P-Asserted-Identity > headers.

how does the proxy know that a ua that has tls connection to it, is
its user? tls alone is not enough. 

-- juha


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Jari Arkko <jari.arkko@piuha.net>
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Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
References: <5.1.1.6.0.20030221110917.04229140@pophost.gte.com> <5.1.1.6.0.20030221110917.04229140@pophost.gte.com> <5.2.0.9.0.20030226125028.00b3e288@mailhost.fokus.gmd.de>
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There are different reasons for collecting accounting data. Only
one of these reasons is billing based on it. Even if you bill
based on usage, it probably depends on the particular business
models what "usage" is. It could be a one-time fee per call,
not tracking of session lengths. I agree that billing based
on what proxies know about bandwidths, codecs, session lengths
is troublesome (at least in the general case, maybe possible
in some specific networks).

But one key question is whether SIP AAA is only for proxies or
for all types of SIP nodes. I think it should not be just
for proxies. SIP user agents of course would also know
session lengths and other parameters. (Not for regular user
terminals of course but for some services offered by the
network.)

Jari

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This list is for NEW development of the application of SIP
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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: jh@lohi.eng.song.fi
Cc: sipping@ietf.org
Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> Sent: Thursday, February 27, 2003 2:50 PM
> To: Beck, Wolfgang
> Cc: sipping@ietf.org
> Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Beck01, Wolfgang writes:
> 
>  > well, at least in theory: the UAs have TLS connections to their
>  > proxies, the  > proxies trust each other and the ingress proxies
>  > insert P-Asserted-Identity > headers.
> 
> how does the proxy know that a ua that has tls connection to it, is
> its user? tls alone is not enough. 
> 
During TLS handshake, the proxy obtains a X509 certificate from the UA.
The cert. contains a subject name that identifies the peer. This can be
used to create a P-Asserted-Identity header. Only the proxy that is directly
connected via TLS to the customer can do that of, of course. All following
proxies have to rely on P-Asserted-Identity (or OSP-token?). You can use
digest authentication on top of TLS to discriminate SIP users that share a
TLS connection using a proxy.

--
Wolfgang Beck
T-Systems Nova GmbH 
_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: "Beck01, Wolfgang" <BeckW@t-systems.com>
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Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
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Beck01, Wolfgang writes:

 > During TLS handshake, the proxy obtains a X509 certificate from the
 > UA.

so you assume somekind of pki and mutual authentication of ua and proxy?
if so, we don't have any of it in real products and practices. 

 > The cert. contains a subject name that identifies the peer. This can be
 > used to create a P-Asserted-Identity header. Only the proxy that is directly
 > connected via TLS to the customer can do that of, of course. All following
 > proxies have to rely on P-Asserted-Identity (or OSP-token?). You can use
 > digest authentication on top of TLS to discriminate SIP users that share a
 > TLS connection using a proxy.

if you don't have the certificate system that you outlined above, the
proxy can't do digest authentication, because you don't know if the user
sending the request is your user or not.  dialog state would help.
another thing that would help is always using as fro uri the uri of the
sending user, which would allow the proxy to know if the user can be
challenged or not. 

-- juha

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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re: [Sipping] Comments on Overlap-04
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In a message dated 2/24/2003 2:23:44 PM Eastern Standard Time, 
jon.peterson@neustar.biz writes:


> [MIke's] Comments on draft-ietf-sipping-overlap-04: 
> 
> General: 
> 
> The ability of one call to result in multiple IAMs being sent into the PSTN 
> continues to be a serious problem for overlap which must be solved.  
>  
> <JFP> This concern has been the subject of whole libraries of previous 
> mail; I think we need to agree to disagree on this point. As we've said in 
> the past, the draft allows this condition to arise only if operators choose 
> to configure their networks in a fashion that allows it. </JFP>


<MP>Yes, that's why I didn't go into detail. We'll just agree that we 
disagree on this and will never convince the other.</MP>

> 
> Section 3.1: 
> 
> All references to one number being a "prefix" of another should be deleted. 
> This text indicates that overlap can not be used where this "prefix" 
> situation can occur, however, one of the purposes of overlap is to handle 
> the situation in which there are varying lengths of numbers in some 
> destination area, something that would naturally exist if one number really 
> could be a "prefix" of another. This creates a major contradiction.  
>  
> <JFP> Not necessarily a contradiction, actually. There are varying lengths 
> of numbers in some numbering plans. That does not entail, however, that one 
> number can necessarily be a prefix of another. The numbers +9990123 and 
> +99912345 could exist in the same numbering plan without creating any 
> problems for overlap signaling. The concern is if the numbers +9990123 and 
> +99901234 were to exist in the same numbering plan.     
> <MP>However, there is still no identified case in the world-wide numbering 
> plan in which one number is a "prefix" of another. If there were, overlap 
> signaling to get to it would certainly be allowed just as it would be in 
> the existing PSTN. If one believes that this "prefix" situation really 
> exists, then the overlap procedures defined in this draft must correctly 
> support it. </MP}
>  
> <JFP> We can't necessarily verify that it's the case, but people keep 
> telling us that this happens in practice in the PSTN. As usual, it is tough 
> to provide a counter-existence proof. I think we still need this text. </JFP
> >


<MP> One doesn't have to come up with a "counter-existence proof" for 
unsubstantiated claims. But anyone who claims that this situation exists 
should be able to come up with an existence proof. However, my point is, if 
you believe this situation exists, and the text of the draft says it does, 
then the overlap procedures must handle it properly. They can not say that 
overlap can't be used in this case. Note that any gateway trying to decide 
not to use overlap would be unaware that this situation exists.</MP>


> Section 3.3 contains the statement: 
> 
>   As a general rule, a media stream corresponding to the response to an 
>   INVITE with a greater number of digits should be given more priority 
>   than media streams from responses with less digits. 
> 
> This continues to be a problem, since it is not always the case. If the 
> originator dialed more digits than were needed when overlap was being used, 
> the media stream corresponding to the response with the fewer digits would 
> be the one to use. The PSTN had no problem with this. SIP is going to have 
> to figure out a way to correctly handle it. 
>  
> <JFP> Our text satisfies particular early-media concerns, and is valid, in 
> my opinion. If fewer digits than the minimum length of a number in the 
> numbering plan are dialed, a gateway could still attempt to pass backwards 
> media to the UAC in order to provide early media data. The behavior that 
> occurs when more digits are dialed than necessary is, no doubt, specific to 
> different nationalities, but it seems reasonable to assume that superfluous 
> digits will be discarded. No one, to date, has asserted anything to the 
> contrary anyway. </JFP>
> 


<MP>Yes, superfluous digits are discarded. In the overlap case, they are 
discarded by the destination exchange that doesn't need them. Any previous 
exchange or gateway would not know they were superfluous. It would give 
"precedence" to the wrong call setup (with more digits) and release the 
desired one, which the called party has actually answered. I don't think this 
issue is related to "early media".</MP>


> Section 3.5 contains the statement: 
> 
>   If the gateway is attached to the PSTN in an area where en-bloc 
>   signalling is used, a REL for the previous IAM and a new IAM should 
>   be generated. 
> 
> This causes confusion since all areas would allow en-bloc, although some do 
> not allow overlap, for example, an area where US SS#7 according to T1.113 
> is used. In those areas, since overlap is not allowed, the gateway or 
> switch which provides interworking into that non-overlap area must perform 
> the overlap to en-bloc procedures (by time outs, detecting enough digits, 
> etc.) There can not be a "previous IAM" sent to the PSTN, since the gateway 
> or switch would not have sent one until it knew it had enough digits to 
> send a complete IAM, after which it would stop accepting or sending any 
> further digits. 
>  
> <JFP> First of all, if you send an incomplete number into an en-bloc 
> signaling network, that are ISUP failure responses (REL with cause 28) that 
> will be sent in response. Our text identifies what one should do if the 
> call has not already been rejected. Gateway very well might be stupid 
> enough to send incomplete addresses into a network that cannot accept them, 
> and it seems dangerous to assume otherwise. </JFP>
> 

This paragraph should be worded as: 
> 
>   If the gateway is attached to the PSTN in an area where overlap 
> signalling 
>   is not allowed, it must not send an IAM until it determines that it has 
>   received the full number in one or more INVITEs, using a process similar 
> to   
>   that described in Section 2. It must not send SAMs or additional IAMs 
> after 
>   


<MP>My comment stands.</MP>


> Mike Pierce 
> 
> 





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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 2/24/2=
003 2:23:44 PM Eastern Standard Time, jon.peterson@neustar.biz writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">[MIke's] Comments on draft-=
ietf-sipping-overlap-04:=20
<BR>
<BR>General:=20
<BR>
<BR>The ability of one call to result in multiple IAMs being sent into the P=
STN continues to be a serious problem for overlap which must be solved. </FO=
NT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LAN=
G=3D"0">&nbsp;</FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF"=20=
FACE=3D"Arial" LANG=3D"0">
<BR>=20
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">&lt;JFP&gt; This concern has been the subject of whole libra=
ries of previous mail; I think we need to agree to disagree on this point. A=
s we've said in the past, the draft allows this condition to arise only if o=
perators choose to configure their networks in a fashion that allows it. &lt=
;/JFP&gt;</FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=
=3D"Arial" LANG=3D"0"> </BLOCKQUOTE>
<BR>
<BR>
<BR>&lt;MP&gt;Yes, that's why I didn't go into detail. We'll just agree that=
 we disagree on this and will never convince the other.&lt;/MP&gt;
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>Section 3.1:=20
<BR>
<BR>All references to one number being a "prefix" of another should be delet=
ed. This text indicates that overlap can not be used where this "prefix" sit=
uation can occur, however, one of the purposes of overlap is to handle the s=
ituation in which there are varying lengths of numbers in some destination a=
rea, something that would naturally exist if one number really could be a "p=
refix" of another. This creates a major contradiction. </FONT><FONT  COLOR=
=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0">&nbsp;<=
/FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial"=20=
LANG=3D"0">
<BR>=20
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">&lt;JFP&gt; Not necessarily a contradiction, actually. There=
 are varying lengths of numbers in some numbering plans. That does not entai=
l, however, that one number can necessarily be a prefix of another. The numb=
ers +9990123 and +99912345 could exist in the same numbering plan without cr=
eating any problems for overlap signaling. The concern is if the numbers +99=
90123 and +99901234 were to exist in the same numbering plan.</FONT><FONT  C=
OLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0"> &n=
bsp;&nbsp;&nbsp;&nbsp;
<BR>&lt;MP&gt;However, there is still no identified case in the world-wide n=
umbering plan in which one number is a "prefix" of another. If there were, o=
verlap signaling to get to it would certainly be allowed just as it would be=
 in the existing PSTN. If one believes that this "prefix" situation really e=
xists, then the overlap procedures defined in this draft must correctly supp=
ort it. &lt;/MP}
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"> </FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERI=
F" FACE=3D"Arial" LANG=3D"0">
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">&lt;JFP&gt; We can't necessarily verify that it's the case,=20=
but people keep telling us that this happens in practice in the PSTN. As usu=
al, it is tough to provide a counter-existence proof. I think we still need=20=
this text. &lt;/JFP&gt;</BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>&lt;MP&gt; One doesn't have to come up with a "counter-existence proof"=20=
for unsubstantiated claims. But anyone who claims that this situation exists=
 should be able to come up with an existence proof. However, my point is, if=
 you believe this situation exists, and the text of the draft says it does,=20=
then the overlap procedures must handle it properly. They can not say that o=
verlap can't be used in this case. Note that any gateway trying to decide no=
t to use overlap would be unaware that this situation exists.&lt;/MP&gt;
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Section 3.3 contains the st=
atement:=20
<BR>
<BR> &nbsp;As a general rule, a media stream corresponding to the response t=
o an=20
<BR> &nbsp;INVITE with a greater number of digits should be given more prior=
ity=20
<BR> &nbsp;than media streams from responses with less digits.=20
<BR>
<BR>This continues to be a problem, since it is not always the case. If the=20=
originator dialed more digits than were needed when overlap was being used,=20=
the media stream corresponding to the response with the fewer digits would b=
e the one to use. The PSTN had no problem with this. SIP is going to have to=
 figure out a way to correctly handle it.=20
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"> </FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERI=
F" FACE=3D"Arial" LANG=3D"0">
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">&lt;JFP&gt; Our text satisfies particular early-media concer=
ns, and is valid, in my opinion. If fewer digits than the minimum length of=20=
a number in the numbering plan are dialed, a gateway could still attempt to=20=
pass backwards media to the UAC in order to provide early media data. The be=
havior that occurs when more digits are dialed than necessary is, no doubt,=20=
specific to different nationalities, but it seems reasonable to assume that=20=
superfluous digits will be discarded. No one, to date, has asserted anything=
 to the contrary anyway. &lt;/JFP&gt;</FONT><FONT  COLOR=3D"#000000" SIZE=
=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0">
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>&lt;MP&gt;Yes, superfluous digits are discarded. In the overlap case, th=
ey are discarded by the destination exchange that doesn't need them. Any pre=
vious exchange or gateway would not know they were superfluous. It would giv=
e "precedence" to the wrong call setup (with more digits) and release the de=
sired one, which the called party has actually answered. I don't think this=20=
issue is related to "early media".&lt;/MP&gt;
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Section 3.5 contains the st=
atement:=20
<BR>
<BR> &nbsp;If the gateway is attached to the PSTN in an area where en-bloc=20
<BR> &nbsp;signalling is used, a REL for the previous IAM and a new IAM shou=
ld=20
<BR> &nbsp;be generated.=20
<BR>
<BR>This causes confusion since all areas would allow en-bloc, although some=
 do not allow overlap, for example, an area where US SS#7 according to T1.11=
3 is used. In those areas, since overlap is not allowed, the gateway or swit=
ch which provides interworking into that non-overlap area must perform the o=
verlap to en-bloc procedures (by time outs, detecting enough digits, etc.) T=
here can not be a "previous IAM" sent to the PSTN, since the gateway or swit=
ch would not have sent one until it knew it had enough digits to send a comp=
lete IAM, after which it would stop accepting or sending any further digits.=
=20
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"> </FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERI=
F" FACE=3D"Arial" LANG=3D"0">
<BR></FONT><FONT  COLOR=3D"#0000ff" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">&lt;JFP&gt; First of all, if you send an incomplete number i=
nto an en-bloc signaling network, that are ISUP failure responses (REL with=20=
cause 28) that will be sent in response. Our text identifies what one should=
 do if the call has not already been rejected. Gateway very well might be st=
upid enough to send incomplete addresses into a network that cannot accept t=
hem, and it seems dangerous to assume otherwise. &lt;/JFP&gt;</FONT><FONT  C=
OLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0">
<BR></BLOCKQUOTE>
<BR>
<BR>This paragraph should be worded as:=20
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR> &nbsp;If the gateway is attached to the PSTN in an area where overlap s=
ignalling=20
<BR> &nbsp;is not allowed, it must not send an IAM until it determines that=20=
it has=20
<BR> &nbsp;received the full number in one or more INVITEs, using a process=20=
similar to &nbsp;&nbsp;
<BR> &nbsp;that described in Section 2. It must not send SAMs or additional=20=
IAMs after=20
<BR> &nbsp;that one IAM. </BLOCKQUOTE>
<BR>
<BR>
<BR>&lt;MP&gt;My comment stands.&lt;/MP&gt;
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Mike Pierce=20
<BR>
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>
<BR></FONT></HTML>

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Date: Thu, 27 Feb 2003 11:02:09 EST
Subject: Re: [Sipping] Overlap in QSIG draft
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In a message dated 2/26/2003 8:07:06 AM Eastern Standard Time, 
Gonzalo.Camarillo@lmf.ericsson.se writes:


> I have noticed that the SIP-QSIG draft describes how to use overlap
> signalling in a SIP network.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
> 
> I believe that this draft should not describe overlap procedures. It
> should reference the overlap mapping draft instead:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt
> 



In a message dated 2/26/2003 8:17:56 AM Eastern Standard Time, 
Gonzalo.Camarillo@lmf.ericsson.se writes:


> the last version of the draft (which was released two or three weeks
> ago) will be presented to the IESG shortly. I believe that we have
> addressed all the issues that were raised... the document could be
> approved in a reasonably short period of time. 
> 



For the record, while I agree that the QSIG draft should reference rather 
then repeat what is specified elsewhere, I do not believe that all the issues 
that have been raised concerning the overlap draft have been addressed or 
resolved. I have commented extensively on that draft, and I believe that many 
of those comments have been ignored. If these issues are not addressed, I am 
sure that others (existing network operators) will specify that interworking 
with the PSTN must be done according to Q.1912.SIP, which is currently being 
developed by ITU-T SG11.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 2/26/2=
003 8:07:06 AM Eastern Standard Time, Gonzalo.Camarillo@lmf.ericsson.se writ=
es:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I have noticed that the SIP=
-QSIG draft describes how to use overlap
<BR>signalling in a SIP network.
<BR>
<BR>http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-00.txt
<BR>
<BR>I believe that this draft should not describe overlap procedures. It
<BR>should reference the overlap mapping draft instead:
<BR>
<BR>http://www.ietf.org/internet-drafts/draft-ietf-sipping-overlap-04.txt
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>
<BR>In a message dated 2/26/2003 8:17:56 AM Eastern Standard Time, Gonzalo.C=
amarillo@lmf.ericsson.se writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">the last version of the dra=
ft (which was released two or three weeks
<BR>ago) will be presented to the IESG shortly. I believe that we have
<BR>addressed all the issues that were raised... the document could be
<BR>approved in a reasonably short period of time.=20
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>
<BR>For the record, while I agree that the QSIG draft should reference rathe=
r then repeat what is specified elsewhere, I do not believe that all the iss=
ues that have been raised concerning the overlap draft have been addressed o=
r resolved. I have commented extensively on that draft, and I believe that m=
any of those comments have been ignored. If these issues are not addressed,=20=
I am sure that others (existing network operators) will specify that interwo=
rking with the PSTN must be done according to Q.1912.SIP, which is currently=
 being developed by ITU-T SG11.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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Subject: Re: [Sipping] Overlap in QSIG draft
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Mike,

>For the record, while I agree that the QSIG draft should reference 
>rather then repeat what is specified elsewhere, I do not believe
>that all the issues that have been raised concerning the overlap draft 
>have been addressed or resolved. I have commented
>extensively on that draft, and I believe that many of those comments 
>have been ignored. If these issues are not addressed, I am
>sure that others (existing network operators) will specify that 
>interworking with the PSTN must be done according to Q.1912.SIP,
>which is currently being developed by ITU-T SG11. 

Jon wrote you a note the other day explaining in which points we agreed
with you and in which ones we did not (and why). We have discussed the
same issues several times and explained you why we cannot specify some
of the things you proposed. I do not think we need to discuss them again
now.

Gonzalo
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Date: Thu, 27 Feb 2003 15:42:32 -0500 (EST)
From: Xiaotao Wu <xiaotaow@cs.columbia.edu>
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
cc: <sipping@ietf.org>, Henning Schulzrinne <hgs@cs.columbia.edu>,
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Subject: Re: [Sipping] I-D ACTION:draft-wu-sipping-floor-control-03.txt
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inline. thanks!

On Wed, 26 Feb 2003, Gonzalo Camarillo wrote:

> Hi,
>
> the definition of the FL semantics for the grouping framework needs more
> elaboration. You need to say whether or not only one control line is
> allowed per group and include an IANA considerations section as follows:
>
> IANA Considerations
>
>    IANA needs to register the following new "semantics" attribute for
>    the SDP grouping framework [RFC3388]:
>
>    Semantics                  Token      Reference
>    -------------------        -----      ---------
>    Floor control              FL         [RFCxxxx]
>
>
>    It should be registered in the SDP parameters registry
>    (http://www.iana.org/assignments/sdp-parameters) under Semantics for
>    the "group" SDP Attribute.
>
> You should also add IANA considerations for a=moderate
>

Thanks! Will add this in.

>
> In Section 8.1, you write: "the conference server uses a=moderated line
> to mute all media tools of the participant". Does the conference server
> set to inactive the SDP direction attributes as well or it leaves them
> as they are (e.g., sendrecv)?

I think the conference server should leave them as they are. The
'a=type:moderated' is for conference attribute in session level. It should
not change the attributes of any media.

Thanks!

-Xiaotao

>
> Thanks,
>
> Gonzalo
>
>
>
> Internet-Drafts@ietf.org wrote:
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts directories.
> >
> >         Title           : Use of Session Initiation Protocol (SIP) and Simple
> >                           Object Access
> >                           Protocol (SOAP) for Conference Floor Control
> >         Author(s)       : X. Wu et al.
> >         Filename        : draft-wu-sipping-floor-control-03.txt
> >         Pages           : 33
> >         Date            : 2003-1-15
> >
> > During a conference, floor control coordinates simultaneous access to
> > shared resource in multimedia conferences. Floor control allows
> > applications and users to gain safe and mutually exclusive or non-
> > exclusive access to the shared resources. This document defines an
> > approach of using Session Initiation Protocol (SIP) event
> > notification mechanism and Simple Object Access Protocol (SOAP) to
> > perform floor control.
> >
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-wu-sipping-floor-control-03.txt
> >
> > To remove yourself from the IETF Announcement list, send a message to
> > ietf-announce-request with the word unsubscribe in the body of the message.
> >
> > Internet-Drafts are also available by anonymous FTP. Login with the username
> > "anonymous" and a password of your e-mail address. After logging in,
> > type "cd internet-drafts" and then
> >         "get draft-wu-sipping-floor-control-03.txt".
> >
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> > Internet-Drafts can also be obtained by e-mail.
> >
> > Send a message to:
> >         mailserv@ietf.org.
> > In the body type:
> >         "FILE /internet-drafts/draft-wu-sipping-floor-control-03.txt".
> >
> > NOTE:   The mail server at ietf.org can return the document in
> >         MIME-encoded form by using the "mpack" utility.  To use this
> >         feature, insert the command "ENCODING mime" before the "FILE"
> >         command.  To decode the response(s), you will need "munpack" or
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> >
> >
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >
> >   ------------------------------------------------------------------------
>
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
>



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I've looked through this draft and have several comments.

SIP-8, local dial plan:
  I'm unhappy with requiring a local dialing plan.
  Many of my customers wish to force all calls through the
    local proxy

SIP-13, client-based conferencing: 
  I don't know what the sentence "SIP devices MUST be able to 
  support handset based conferencing" means.  I understand the
  next sentence.  Are you proposing that each handset follow
  the SIP conferencing requirements?  If not, what, exactly,
  is the requirement?

SIP-15: DTMF RTP payload
  I'm confused why this is a MUST.  If the device only supports
  G.711 or better (i.e. no low bandwidth, highly compressed) codecs,
  why is it a requirement to generate 2833?

SIP-22 through SIP-26, multi-line support
  I'm having problems with the terminology.  "Multiline" to me
  means the ability to handle more than one call at a time (all but
  one on hold).  Here you mean "multiple users on the same UA"
  I think the requirements are okay (at MAY), but we need
  a new term that really means multiple users registered at one UA.

  I'll note that there is a possible problem with our basic standards.
  To allow multiple users at the same UA, the UA, the registrar and
  the incoming proxy all have to support such a thing.  There is no
  way for the UA to know if that is possible.

SIP-24, multi-line call waiting indicator
  I don't understand "If call waiting is set for an identity, the 
  client MUST respond with "486 Busy Here" when an incoming call 
  to that identity is received and the client has an existing 
  call with any of the identities"

  Are you disallowing the possibility of more than one waiting
  call for an identity?  How else do I understand "call waiting
  is set"?

SIP-25, Dynamic login/logout 
  I disagree with the architecture assumption made here.  I think
  emergency calls should NOT depend on any kind of authentication.
  Thus, I think it should not be necessary to register before
  making an emergency call, and thus there is no need for a default
  identity.  

  Specifically, if your proposal was implemented, and, for whatever
  reason, authentication failed, and a user could not make an emergency
  call, I think someone is in DEEP doo-doo.

  I think we need some convention for identity (ie what is in the From)
  to aid law enforcement in tracking malicious calls to emergency
  centers, but, please, don't depend on authentication.  As far as I
  know, emergency centers would rather get some "anonymous" calls than
  miss a call because someone configured something wrong.

SIPsec-2 password protection
  I dislike this requirement.  I think the problems it generates outweigh
  the security it provides.  Specifically, to make this work, you have
  to put a default password in the device, so you can get in the first
  time, and you need some kind of a reset function when people forget
  the settings.  The former almost always gets left in this state, which
  opens up a huge attack, the latter always eventually gets known, and
  exploited.

  Our systems have VERY limited configuration done at the terminal -
  basically, just how you get an IP address.  Everything else we store
  on a server and download.  This works well for us, and putting a password
  on the IP settings is just not justified.

SIPsec-4, INVITE user portion acceptance
  You don't have the corresponding output requirements - SIP devices must
  have a setting that forces all outbound messages to a specific proxy.
  This is a clear requirement for many operators.

International-1, language support
  I have a very minor problem with "The setting for this header MUST 
  be provisioned".  My problem is that it could be interpreted as
  disallowing a separate language preference for each user who is
  registered when multi-user support is provided.  We certainly would
  not provision the PHONE with language.  We would have a user profile
  setting for this.

App-2, hold ring-back
  Do you expect the phone to send any SIP message when it starts to
  ring?  Maybe you should say what you expect.

App-5, SIMPLE Integration
  I think this is not well specified.  SIMPLE has so many possible
  variations, I don't see what we say.  I also think there is no
  "address book" spec yet, and specifically, an address book may
  have many more addresses in it than you get presence for.

Web-1
  I object to this requirement.  We provide a very nice
  UI on our device to specify things like ring tones and forwarding
  preferences.  It happens that we also have a web UI that can do this,
  but we really don't need it.  I'll note that the wording is
  awkward.  The first sentence says "SIP devices MUST support a web server",
  while the second sentence says "Web pages ... MAY be [in] ..centralized 
  web servers"  The latter is more correct than the former.

Web-2
  Object in that not all such settings make sense to be remotely settable.
  Forwarding behavior may be, ringtones may not be, at least with respect
  to a user.

FW/NAT-1, outbound proxy support
  I think this is too tightly specified.  Specifically, I think the
  device needs to accept an outbound proxy.  It must be configurable
  to always use this proxy, and for each user, there must be a full
  uri that may or may not be in the same domain as the outbound proxy.
  There is no need to explicitly know the domain; you need to know the
  URI.  It may be that your implementation affixes a registration name
  to a default domain.  Ours doesn't.  We authenticate with a username
  and then return an AOR.  I think that is quite reasonable.

  That means the first MUST (default domain) can go away.

Int-1: (buttons)
  I would not want this to be interpreted as requiring any specific
  hard buttons.  Any/all of these could be in some soft UI.

Int-3 
  There are several paragraphs after the first one in Int-3 that I
  can't make any sense of.  Is some of it supposed to be an Int-4?
  
  The fragment "End user options..." is not a sentence.

Lines-1 This requirement gets confused with the prior multi-line
  section above.  Consolidation and terminology change is needed.

Section 3
  First of all, at least at the moment, there is no SIP mechanism to
  do any of this, and I suspect it won't be a sip mechanism that
  does it, although it could be a form of PUBLISH.  So, I think 
  using terms like "User Agent" is incorrect.  Perhaps use Profile 
  Server and Profile Client.

  There are several cases where the word "may" became "MAY"
  inappropriately.  

  I also think there are "device profiles" that are associated with
  a particular UA, "user profiles" that are associated with a particular
  user, and a hybrid, where there is a profile for a particular
  user at a particular device.  These need names, we need a name
  for any of them, and the text has to be edited to use the
  correct one of these.

  I have some problems with:
    Change Notification is the means by which the profile server tells 
    the user agent that profiles of interest have changed.  Typically, 
    the intention would be for the user agent to get those changes or the 
    updated profiles.  

  My problem is that this is a database, the terminal may change it,
  or an external web accessed page may change it.  In the former, the
  device has to inform the database that it is changed.  In the latter,
  the database has to inform the device that it is changed.  Both of
  these are change notifications.

OpenExt-1: and OpenExt-2: are the same.
  I think extensibility of the data object is more important than
  "intelligence" at the server, whatever that means.

ENR-5:
  Also is needed to support multiple users on the same UA.

CN-3: 
  Could we have some justification for this?  I'm not objecting
  to a MAY, I want the device to MAY support a scheduled change.

SEC-n
  I have some problems with authentication, because I envision this
  to be close to the first thing you do after you have an IP address.
  The device is either provisioned with, or learns the profile
  server from DHCP.  It contacts it first, before any registration,
  ....  You don't have a lot to authenticate either end with, except
  the aforementioned password which I think is worse than nothing.
  
Section 4
  I am confused over what this intended to be.  Are we talking about
  settings that are made manually at the device, not covered by
  a device profile described in Section 3?  I personally think we should
  be minimizing such things.  Right now, ideally, I think you learn
  the profile server from DHCP and get everything from that except the
  IP address/mask stuff that you also get from DHCP.  It's a tossup
  whether you learn the sip proxy from DHCP or the profile server.
  You certainly shouldn't be having a "Session Timer" setting that
  doesn't come from the device profile.

  Also has the confusion on the term "line"

  There is a typo in the chapter number in the first sentence.

Network-2
  We should be clear whether we intend this wording to imply you have
  more than one setting, or whether there is one setting and it is
  applied to both media and signalling.  I think there are potentially
  many settings, one for each media type and one for signalling.

Dial-1: Dial Plan. 
  I object.  We SHOULD recommend a "Dial" button and not rely on
  a dial plan.  A dial plan should at best be a MAY.

Regional-4: 
  I recognize that some sites will not support DHCP.  DHCP should
  be the preferred way to find the time server.  The text should say
  that.  If you don't have DHCP, it has to be provisioned, and thus
  we need this in the list, I just think its use should be discouraged.
  
Phonebook-1:
  The text seems to not allow a phonebook to be stored on the profile
  server.  It should allow that.  The problem is the word "stored".

PhoneServers-1:
  I don't understand why you would encourage phonebooks to be stored
  in someplace other than the profile server.  I can see some difficulty
  in storing history there, although we would do it that way.

4.11 Ringer Behavior  
   
  The text "These devices SHOULD ignore all line related settings 
  that do no affect the default outbound line settings they receive"
  has a typo, and it confuses how "default" should work.  They are
  line settings.  There is a default line.  It should state something
  like "... all line settings other than the default line's settings"

  The last paragraph of this section should be deleted.  It doesn't
  help, and it ignores the most common choice of ringtone selection
  by caller.

4.12 User Related Settings
  This text is very confusing to me.  I'm certainly agreeable to supporting
  multiple users registered at a UA.  I'm interested in supporting
  multiple call appearances per user (or per UA if you prefer, they are
  the same as long as you don't restrict one user to a SINGLE appearance)

  The picture you seem to have is a physical or UI button per line,
  and you want to assign a user to a line.  Okay, if that's what you
  want as long as I can assign more than one line to one user.

  Our system works a bit differently but the effect is the same with
  regard to allowing multiple users registered at one UA, and multiple
  call appearances per user.

  The only thing I can make out of this text is that you want to be
  able to have some prior configuration that has some default association
  between a line button and a user.  While we wouldn't do that, I understand
  that such a facility may be useful for some vendors.  The text is far
  from clear about that, if indeed that is what you meant.

  Roaming means that a user walks up to a phone and does something that
  causes him to be registered.  In your picture, it causes one or more
  line buttons to be associated with him.  That isn't a configuration
  mechanism, that's a run time dynamic assignment mechanism.  Registering
  on a UA may cause a new user profile to be downloaded, I get that,
  but this section is on device configuration, as best I can figure out,
  and this text doesn't make much sense to me here.

4.13 Line Related Settings 
  
  Another very confusing section for me.  My problem is that I don't see
  a 1to1 correspondence between a "line" and an address of record.
  The whole "line" concept is POTS specific.  There are users and there
  are call appearances.  You can have one or more users, and each user
  can have zero or more call appearances.  Each user has an AoR, but
  you can have more than one active call at this AoR at a particular
  device.  

  It could be that if you changed "line" to "registered user", the text
  could make some sense, but it might need more work than that.
  You clearly don't have a separate registration if a user has more
  than one "line" in your sense of "line"; The first call to the user's
  AoR would show up on his first "line", the second would show at his
  second "line".  There doesn't have to be any knowledge of this in
  anything except the device.  There doesn't have to be a URI for each
  line.

4.16 through 4.20
  These are, or should at least have the option to be, PER USER settings,
  not per device settings.  I'm having a little conflicting views
  problems, because we implement these things in the proxy server
  not the device, but we're having some problems with the idea of 
  "ring once then forward".  Generally, we think that forwarding
  behavior is best done at the server since you have to do it anyway
  when the user is not logged in.  On the other hand, not all servers
  will do that, and so maybe requiring the phone to be able to do
  it may be necessary.

I did not review sections 5 and higher.

Brian






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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'jh@lohi.eng.song.fi'" <jh@lohi.eng.song.fi>,
        "Beck01, Wolfgang"
	 <BeckW@t-systems.com>
Cc: sipping@ietf.org
Subject: RE: [Sipping] New version of the SIP-AAA reqs draft
Date: Thu, 27 Feb 2003 16:19:49 -0500
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I agree with Juha that it's pretty unlikely that a UA will have an X.509
cert it can use to do mutual TLS auth with a proxy. The proxy should
definitely authenticate itself to the UA with a certificate, but it is much
more plausible that the UA will authenticate itself to the proxy with
Digest. This is the model presented in RFC3261,  anyway.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
> Sent: Thursday, February 27, 2003 7:25 AM
> To: Beck01, Wolfgang
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] New version of the SIP-AAA reqs draft
> 
> 
> Beck01, Wolfgang writes:
> 
>  > During TLS handshake, the proxy obtains a X509 certificate from the
>  > UA.
> 
> so you assume somekind of pki and mutual authentication of ua 
> and proxy?
> if so, we don't have any of it in real products and practices. 
> 
>  > The cert. contains a subject name that identifies the 
> peer. This can be
>  > used to create a P-Asserted-Identity header. Only the 
> proxy that is directly
>  > connected via TLS to the customer can do that of, of 
> course. All following
>  > proxies have to rely on P-Asserted-Identity (or 
> OSP-token?). You can use
>  > digest authentication on top of TLS to discriminate SIP 
> users that share a
>  > TLS connection using a proxy.
> 
> if you don't have the certificate system that you outlined above, the
> proxy can't do digest authentication, because you don't know 
> if the user
> sending the request is your user or not.  dialog state would help.
> another thing that would help is always using as fro uri the 
> uri of the
> sending user, which would allow the proxy to know if the user can be
> challenged or not. 
> 
> -- juha
> 
> _______________________________________________
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> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
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Subject: Re: [Sipping] Comments on draft-sinnreich-sipdev-req-00.txt (long)
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> 
> SIP-15: DTMF RTP payload
>   I'm confused why this is a MUST.  If the device only supports
>   G.711 or better (i.e. no low bandwidth, highly compressed) codecs,
>   why is it a requirement to generate 2833?

It's a lot easier if you don't have to recognize DTMF from tones. (Doing 
reliable DTMF recognition is surprisingly tricky; in particular, it's 
hard to do this without a full reconstruction of the audio signal, i.e., 
playout delay compensation.) Thus, it seems silly to waste CPU resources 
on the client to generate a sine waveform just waste more of them trying 
to detect which one it was on the ACD server or gateway. It's trivial to 
implement 2833 sending, so is there a reason *NOT* to require it?

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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: sipping@ietf.org
Cc: jh@lohi.eng.song.fi, stu.jacobs@labs.gte.com, mankin@psg.com
Subject: RE: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@lmf.ericsson.se]
> Sent: Thursday, February 27, 2003 9:28 AM
> To: Jiri Kuthan
>
> Juha,
>
> Do you mean that you would like to see a note along these 
> lines added to
> the draft?
> 
> "Note that, in order to perform accounting, a SIP proxy server should
> see and authenticate every message sent over the dialog."
> 
I don't think it's possible to do reliable accounting within a proxy
if any or both of the UAs are cheating. The caller can deliberately
ignore Record-Route and bypass the proxy when sending an INVITE. The
callee can send broken BYEs that will be discarded by the caller but
will be accepted by the accounting proxy. 

What kind of accounting information do we want to collect: The number
of SIP messages per user? This is possible with an accounting proxy.
Call minutes? How does the proxy know what kind of session is being
established if the SDP is encrypted? Only the endpoints and the bandwidth
reservation infrastructure can do this kind of accounting.

The only way I see is to include some form of micropayment tokens ("postage stamps")
in SIP messages. draft-johnston-sip-osp-token-04.txt describes such a token.
Periodical session updates containing tokens prevent disputes about the actual
length of the call. If the callee has tokens, he can charge for the call. The
trusted third party that generates those tokens can be an OSP server or may be
a proxy that inserts some header.

(see http://www.ietf.org/internet-drafts/draft-beck-sipping-svc-charging-req-01.txt)


--
Wolfgang Beck
T-Systems Nova GmbH 


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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: jon.peterson@neustar.biz, jh@lohi.eng.song.fi
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Subject: RE: [Sipping] proxy accounting, was:  New version of the SIP-AAA 
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> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> Sent: Thursday, February 27, 2003 10:20 PM
> 
> I agree with Juha that it's pretty unlikely that a UA will 
> have an X.509
> cert it can use to do mutual TLS auth with a proxy. The proxy should
> definitely authenticate itself to the UA with a certificate, 
> but it is much
> more plausible that the UA will authenticate itself to the proxy with
> Digest. This is the model presented in RFC3261,  anyway.
> 

I was referring to theoretical possibility, deliberately ignoring
ugly real-world details. Btw, RfC3261 proposes S/MIME certificates
stored in UAs to identify end users.

Maybe Juha and I are talking about different things. The question
was: can any proxy on the path do a per-user accounting? If it is
not used for billing, there is no incentive for users to tamper
with the SIP headers and a certain percentage of incorrect
accounting records is tolerable. If it is used for billing - proxy
usage, not call duration - then we need some proof of origin. We
have:

- Digest + P-Asserted-Identity -> Problems with BYE and CANCEL, untrusted proxies
- TLS + P-Asserted-Identity -> no certs on today's UAs, only within one trust domain
- S/MIME signatures -> no implementations in sight
- OSP-Token -> not a standard yet, no implementations (?)



--
Wolfgang Beck
T-Systems Nova GmbH 
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> - Digest + P-Asserted-Identity -> Problems with BYE and CANCEL, untrusted proxies
> - TLS + P-Asserted-Identity -> no certs on today's UAs, only within one trust domain
> - S/MIME signatures -> no implementations in sight
> - OSP-Token -> not a standard yet, no implementations (?)

Yes, though I would modify this a bit. I wouldn't group Digest and
P-Asserted-Identity under the same heading; the difference is that
in the first the proxy itself is doing the checking whereas in the
other you are relying on someone else to have done it for you.

Also, what is "proxy usage"? Is it routing an INVITE, REGISTER
through the proxy, or is it counting every SIP protocol message?
If former, limitations you mentioned above may not matter.
Aren't the digest problems with ACK and CANCEL, not with BYE?
And I believe there's limited support for authentication of ACK,
just that no new challenge can be generated.

Thus, I think the table should be:

- P-Asserted-Identity -> Only reliable in an environment with trusted proxies
- Digest -> Ok in some cases, Digest limitations for CANCEL maybe ACK in
             some other cases
- TLS + P-Asserted-Identity -> no certs on today's UAs, only within one trust domain
- S/MIME signatures -> no implementations in sight
- OSP-Token -> not a standard yet, no implementations (?)

(This table applies only when talking about accounting for the purpose
of billing.)

Jari

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Beck01, Wolfgang writes:

 > - Digest + P-Asserted-Identity -> Problems with BYE and CANCEL,
 > - untrusted proxies

as i said, it would help quite a bit if we would change the rule how
from header is constructed when sending a request within a dialog.  i
propose that in the next version of the sip rfc, the uri of the sender
of the request is always used.  that would help the proxy to figure out
if it can challenge the request.

-- juha
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Jari Arkko writes:

 > Aren't the digest problems with ACK and CANCEL, not with BYE?

no the problem is with any request within a dialog.

-- juha

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Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Juha,

I still do not quite understand your arguments. Could you please
elaborate on why a proxy cannot know that a message comes from a
particular UA?

Thanks,

Gonzalo


jh@lohi.eng.song.fi wrote:
> 
> Beck01, Wolfgang writes:
> 
>  > What kind of accounting information do we want to collect: The number
>  > > of SIP messages per user? This is possible with an accounting
>  > proxy.
> 
> unfortunately not.
> 
> -- juha
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sipping] WGLC on draft-ietf-sipping-aaa-req-02.txt
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Folks,

we have had some discussions about the SIP-AAA work in the mailing list
lately. In order to take advantage of this interest, I would like to
WGLC the SIP-AAA requirements draft.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-aaa-req-02.txt

By last calling this document, we want to try to keep the focus of the
WG on the requirements, before we jump into discussing solutions.

This WGLC will end on April 4th.

Thanks,

Gonzalo
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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: jari.arkko@piuha.net
Cc: jon.peterson@neustar.biz, jh@lohi.eng.song.fi, sipping@ietf.org
Subject: RE: [Sipping] proxy accounting
Date: Fri, 28 Feb 2003 11:45:43 +0100
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Jari,
> From: Jari Arkko [mailto:jari.arkko@piuha.net]
> Sent: Friday, February 28, 2003 10:33 AM
> 
> > - Digest + P-Asserted-Identity -> Problems with BYE and 
> CANCEL, untrusted proxies
> > - TLS + P-Asserted-Identity -> no certs on today's UAs, 
> only within one trust domain
> > - S/MIME signatures -> no implementations in sight
> > - OSP-Token -> not a standard yet, no implementations (?)
> 
> Yes, though I would modify this a bit. I wouldn't group Digest and
> P-Asserted-Identity under the same heading; the difference is that
> in the first the proxy itself is doing the checking whereas in the
> other you are relying on someone else to have done it for you.
I was thinking of a combination where the first proxy authenticates
with digest and inserts a P-Asserted-Identity header afterwards.

taking your comment into account, the list looks like this:

Digest: proxy relies on itself

P-Asserted-Identity: proxy relies on the first proxy to do the checking (using
  digest or TLS) and on intermediate proxies not discarding P-Asserted-Identity.

S/MIME: proxy relies on the CA to do the checking

OSP-Token: proxy relies on the OSP-Server to do the checking

 > Also, what is "proxy usage"? Is it routing an INVITE, REGISTER
> through the proxy, or is it counting every SIP protocol message?
I meant counting protocol messages.

> Aren't the digest problems with ACK and CANCEL, not with BYE?
> And I believe there's limited support for authentication of ACK,
> just that no new challenge can be generated.
You are right, I was mixing things up, Digest has other limitations
here. Imagine a scenario where all proxies in a path subsequently
challenge all requests of a UA. Call setup times will be bad.

--
Wolfgang Beck
T-Systems Nova GmbH 

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Cc: "Beck01, Wolfgang" <BeckW@t-systems.com>, sipping@ietf.org,
        stu.jacobs@labs.gte.com, mankin@psg.com,
        Jari Arkko <Jari.Arkko@lmf.ericsson.se>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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        <15966.1788.184832.423324@harjus.eng.song.fi>
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Date: Fri, 28 Feb 2003 14:44:58 +0200
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Gonzalo Camarillo writes:

 > I still do not quite understand your arguments. Could you please
 > elaborate on why a proxy cannot know that a message comes from a
 > particular UA?

for example, i can send an invite where To header looks like this:

To: <sip:33983900@sip.song.fi>

my proxy can then map it via enum, for example, to a request uri

sip:jh@tutpro.com

which the proxy of tutpro.com then delivers to me.

when i send a bye to my proxy to close this call, from header has the
above To uri that my proxy can't use to determine that it should
challenge the request,

my current understanding is that in order to be able to challenge the
request, the proxy should keep dialog state.  if you know some other way
for the proxy to determine from the bye request alone that its own user
has sent it, please let me know and i'll instantly add that capability
to my proxy.

-- juha
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        stu.jacobs@labs.gte.com, mankin@psg.com,
        Jari Arkko <Jari.Arkko@lmf.ericsson.se>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
References: <C3F9C806AEC6D5119643000347055E322089AE@G9JNW.mgb01.telekom.de>
	        <15966.1788.184832.423324@harjus.eng.song.fi>
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Juha,

Here you have a figure of the scenario you have just described:

                       alice.com   . . .   bob.com
                 .       proxy              proxy     .
               .                                       .
         Alice  . . . . . . . . . . . . . . . . . . . .  Bob
      
We have an INVITE from alice@alice.com to bob@bob.com

Now, you have a BYE from Bob to Alice. In your scenario, which proxy is
doing accounting, alice.com or bob.com?

Thanks for the clarification,

Gonzalo



Juha Heinanen wrote:
> 
> Gonzalo Camarillo writes:
> 
>  > I still do not quite understand your arguments. Could you please
>  > elaborate on why a proxy cannot know that a message comes from a
>  > particular UA?
> 
> for example, i can send an invite where To header looks like this:
> 
> To: <sip:33983900@sip.song.fi>
> 
> my proxy can then map it via enum, for example, to a request uri
> 
> sip:jh@tutpro.com
> 
> which the proxy of tutpro.com then delivers to me.
> 
> when i send a bye to my proxy to close this call, from header has the
> above To uri that my proxy can't use to determine that it should
> challenge the request,
> 
> my current understanding is that in order to be able to challenge the
> request, the proxy should keep dialog state.  if you know some other way
> for the proxy to determine from the bye request alone that its own user
> has sent it, please let me know and i'll instantly add that capability
> to my proxy.
> 
> -- juha
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: "Beck01, Wolfgang" <BeckW@t-systems.com>, sipping@ietf.org,
        stu.jacobs@labs.gte.com, mankin@psg.com,
        Jari Arkko <Jari.Arkko@lmf.ericsson.se>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Gonzalo Camarillo writes:

 > Here you have a figure of the scenario you have just described:
 > 
 >                        alice.com   . . .   bob.com
 >                  .       proxy              proxy     .
 >                .                                       .
 >          Alice  . . . . . . . . . . . . . . . . . . . .  Bob
 >       
 > We have an INVITE from alice@alice.com to bob@bob.com

no, originally, we have an invite from alice@alice.com to
33983900@sip.song.fi that gets translated by enum to bob@bob.com.
 > 
 > Now, you have a BYE from Bob to Alice. In your scenario, which proxy is
 > doing accounting, alice.com or bob.com?

both proxies account both placed and received calls of their own users.

-- juha

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Thread-Topic: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Subject: [Sipping] Introduction to Accounting Management (from IETF 53)
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Hi all,

Those of you at IETF 53 may remember that Bernard Aboba gave a presentation
about some of the shortcomings of RADIUS, especially with regards to
accounting.  I searched the procedings pages and noticed that the presentation
was not there.  

I've put a copy of the presentation here: http://www-nrc.nokia.com/sua/Accounting.ppt,
if anyone is interested in reading it.  It may be relevent to the
current discussion.

best regards,
John
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Subject: Re: [Sipping] WGLC on draft-ietf-sipping-aaa-req-02.txt
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Juha,

I was not refering to you. I was talking about the Diameter-Radious-Web
services discussions. Your point falls perfectly within the scope of
this WGLC.

Thanks,

Gonzalo

jh@lohi.eng.song.fi wrote:
> 
> Gonzalo Camarillo writes:
> 
>  > By last calling this document, we want to try to keep the focus of the
>  > WG on the requirements, before we jump into discussing solutions.
> 
> i have been talking about requirements, i.e., what is required to make
> accounting possible at all, and i don't think the current document
> really addresses that question.
> 
> - juha
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Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Juha,

OK, now I understand what you mean. Your proxy decides to perform or to
not perform accounting for a particular request based on the From header
field of the request. That is a poor choice. The proxy should challenge
the request (e.g., using digest). This way, the proxy can ensure that
the request comes from that particular user.

Your proxy can keep from challenging a particular request if the request
carries a P-asserted-identity, for instance.

Your proxy can refuse to route any request unless the user has provided
proper credentials or the network has provided credentials for the user.

Gonzalo


jh@lohi.eng.song.fi wrote:
> 
> Gonzalo Camarillo writes:
> 
>  > Here you have a figure of the scenario you have just described:
>  >
>  >                        alice.com   . . .   bob.com
>  >                  .       proxy              proxy     .
>  >                .                                       .
>  >          Alice  . . . . . . . . . . . . . . . . . . . .  Bob
>  >
>  > We have an INVITE from alice@alice.com to bob@bob.com
> 
> no, originally, we have an invite from alice@alice.com to
> 33983900@sip.song.fi that gets translated by enum to bob@bob.com.
>  >
>  > Now, you have a BYE from Bob to Alice. In your scenario, which proxy is
>  > doing accounting, alice.com or bob.com?
> 
> both proxies account both placed and received calls of their own users.
> 
> -- juha

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Media Policy Manipulation in the Conference Policy 
                          Control Protocol
	Author(s)	: R. Mahy, N. Ismail
	Filename	: draft-mahy-sipping-media-policy-control-00.txt
	Pages		: 19
	Date		: 2003-2-27
	
The SIP conferencing framework defines a model for tightly-coupled
conferencing in the Session Initiation Protocol (SIP), in which a
Conference Policy Control Protocol is used to manipulate policies
relevant to a specific conference, such as conference membership
policy, authorization policy, and media policy. This document
describes a logical model to describe media processing in a SIP
conference. It also defines specific protocol semantics and a
specific syntax to manipulate that model.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Role-based Authorization Requirements for the Session 
                          Initiation Protocol
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sipping-role-authz-00.txt
	Pages		: 10
	Date		: 2003-2-27
	
This document lays out a set of requirements related to role-based
authorization for the Session Initiation Protocol.  While numerous
authentication mechanisms are described in the base SIP
specification, role-based authorization provides information used to
make policy decisions based on the attributes of a participant in a
session.  This approach provides a richer framework for
authorization, as well as allow greater privacy for users of an
identity system.

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Date: Fri, 28 Feb 2003 13:45:48 +0200
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: sipping <sipping@ietf.org>, John Loughney <john.loughney@nokia.com>,
        Dean Willis <dean.willis@softarmor.com>, Rohan Mahy <rohan@cisco.com>,
        Allison Mankin <mankin@psg.com>
Subject: [Sipping] WGLC on draft-ietf-sipping-aaa-req-02.txt
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Gonzalo Camarillo writes:

 > By last calling this document, we want to try to keep the focus of the
 > WG on the requirements, before we jump into discussing solutions.

i have been talking about requirements, i.e., what is required to make
accounting possible at all, and i don't think the current document
really addresses that question.

- juha

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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: "Beck01, Wolfgang" <BeckW@t-systems.com>, sipping@ietf.org,
        stu.jacobs@labs.gte.com, mankin@psg.com,
        Jari Arkko <Jari.Arkko@lmf.ericsson.se>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Gonzalo Camarillo writes:

 > OK, now I understand what you mean. Your proxy decides to perform or to
 > not perform accounting for a particular request based on the From header
 > field of the request. That is a poor choice. The proxy should challenge
 > the request (e.g., using digest). This way, the proxy can ensure that
 > the request comes from that particular user.

so are you saying that a user cannot place a call to the user of my
proxy unless the user is also a user of the same proxy or has been
challenged by someone else that my proxy trusts?  if so, we'll have very
little sip traffic in our networks.  it would be the same as disallowing
a user of an email server from receiving email unless it was sent by
another email server that the email server trusts.

also, if the proxy serves multiple realms, how can it know which realm
the user that send the request within a dialog belongs to so that it can
insert the proper realm in the digest challenge?  today there is no
header in the request where the user could tell that it belongs to a
certain realm.  the from header is the only one, but it only works for
sure for initial requests.

 > Your proxy can keep from challenging a particular request if the request
 > carries a P-asserted-identity, for instance.

as i said, there would be very little sip traffic if you impose such a
rule.  besides it is a p header and not part of sip.  if you remember i
tried to make it a real header, but failed to convince people.  now you
make the whole accounting process to depend on it.  i don't think we
should accept as working group documents anything that depends on p
headers.

 > Your proxy can refuse to route any request unless the user has provided
 > proper credentials or the network has provided credentials for the user.

see above.  with today's means, i don't even know which realm the user
is supposed to belong to.

-- juha

_______________________________________________
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This list is for NEW development of the application of SIP
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        Jari Arkko <Jari.Arkko@lmf.ericsson.se>
Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Juha,

> so are you saying that a user cannot place a call to the user of my
> proxy unless the user is also a user of the same proxy or has been
> challenged by someone else that my proxy trusts?  if so, we'll have very
> little sip traffic in our networks.  it would be the same as disallowing
> a user of an email server from receiving email unless it was sent by
> another email server that the email server trusts.

Everything depends on your trust model. If your proxy is performing some
type of accounting that you (the user) wants it to perform (e.g., some
type of service), the proxy will accept "anonymous" and no password as
credentials, because the proxy is not concerned with UAs trying to
cheat. Therefore, external users will be able to call you.

And if you are not concerned about security either and you want to base
your service in the contents of the To (or From) fields, then your UA
can actually modify the To field. What it cannot modify is the tag.

If your proxy does not trust the UA because it is a billing proxy, for
instance, your proxy will only accept requests with proper credentials
or requests that are going to be routed to a known (by the proxy) UA.
But ensuring that the UA does not cheat is outside of the scope of the
requirements, and it can be done in several ways.


> also, if the proxy serves multiple realms, how can it know which realm
> the user that send the request within a dialog belongs to so that it can
> insert the proper realm in the digest challenge?  today there is no
> header in the request where the user could tell that it belongs to a
> certain realm.  the from header is the only one, but it only works for
> sure for initial requests.

You can change the contents of the From field. No problem. In fact, one
of the reasons we removed the To and From URIs from the dialog
identifiers (only the tags remain) was to address this type of
situation.

Gonzalo
_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sipping] proxy accounting, was:  New version of the SIP-AAAreqs 
 draft
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Juha,

> i
> propose that in the next version of the sip rfc, the uri of the sender
> of the request is always used.  that would help the proxy to figure out
> if it can challenge the request.

This is what I said in one of my previous mails. Per RFC 3261, you are
free to change your URL if you want. No change to the spec needed.

Gonzalo
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To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Cc: "Beck01, Wolfgang" <BeckW@t-systems.com>, sipping@ietf.org,
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Subject: Re: [Sip] Re: [Sipping] New version of the SIP-AAA reqs draft
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Gonzalo Camarillo writes:

 > You can change the contents of the From field. No problem. In fact, one
 > of the reasons we removed the To and From URIs from the dialog
 > identifiers (only the tags remain) was to address this type of
 > situation.

currently your text says:

The From URI of the request MUST be set to the local URI from the dialog
state.

...

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.

so before that is done and the text is changed so that the from uri MUST
be the from uri of the user sending the request, there can't be any
challenging or accounting of requests within the dialog.

alternatively, we must add another header or field to the request that
tells what realm the ua sending the request claims to belong to.

for example, take a look at eap/tls.  there the user has a means to tell
what realm he/she wants to authenticate with.  currently we are lacking
this possibility in sip, which is a major shortcoming.

-- juha

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Gonzalo Camarillo writes:

 > This is what I said in one of my previous mails. Per RFC 3261, you are
 > free to change your URL if you want. No change to the spec needed.

the text that i quoted, doesn't say so.  in order to get this capability
implemented in UAs, it must be a MUST in the text, or. alternatively, we
must have a new header/field where the realm can be placed.  the same
goes for the P header.  i don't know any general purpose sip ua that is
implementing it.

-- juha

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Subject: Re: [Sipping] Comments on draft-rosenberg-sipping-conferencing-framework-01.txt
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Overall the draft looks good.  I wish this conference frame work can be
extended to cover the following enterprise/service conferencing scenario:

When an application need a conference service, it only creates a conference
ID (and its associated policy).  By the time the application does not know
(does not want to or can not specify) which conference service ( Focus) to
use.   Moreover,  it is desired for the performance and scalability that a
set of Focuses should work collaboratively to perform the distributed
conference service.  For example, 20 participants in a conference,   10 in
CA and 10 in NY,  it is better to have two Focuses (one in CA and one in
NY) to jointly perform the service.   Protocols among Focuses are required.

Also, most of VoIP phone does support ad-hoc conference capability and is
an useful media mixing resource.  It can be useful (for the resource usage
and scalability) that the Focus can delegate some of the media mixing
function to be performed in these phones of the conference participants.
Protocols among Focus and participants are required.

There are some components are required in the above scenario to achieve the
dynamic conference configuration.     .

1.  A conference is identified by a conference ID.  A conference might be
collaboratively serviced by more than one Focuses.  Since URI is used for
routing to only one UA, it would be more appropriate to carry the
conference ID (and its associated policy) in the payload with standard URI.

2. The default Focus for a UA can be pre-configured or discovered using
similar procedure of outbound proxy (or proxy server).

3. Protocols among Focuses (and conference policy server) are required.
The Subscribe and Notify SIP mechanism with extended SDP can be used among
Focuses to establish a collaboratively distributed service for a
conference.

4. Protocols between Focus (and conference policy server) and participants
are required.   In Figure 2, the conference policy control protocol is only
one way (from participant to conference policy server), it can be two way
protocol to make use of the mixing resources in the participants (e.g.,
VoIP phone).   One possibility is to extend REFER mechanism using
(extended) SDP for carrying conferencing policy.

Regards,

zon-yin shae
IBM Watson Research Center


aroychow@hss.hns.com@ietf.org on 02/22/2003 10:37:44 AM

Sent by:    sipping-admin@ietf.org


To:    <shaj@cisco.com>
cc:    sipping@ietf.org, sipping-admin@ietf.org
Subject:    Re: [Sipping] Comments on
       draft-rosenberg-sipping-conferencing-framework-01.txt




Inline prefixed by <ARC>

The idea of having a unique URI for each conference is a good one. However
a more scalable approach would be to have a master Focus or a group of them
that register with a standard URI. This way users use a standard URI with a
Conference ID as the payload in the Invite message.  This is directed to
the master Focus that does the following
* Retrieve the Conference ID
* Check if a Focus corresponding to the conference exists
* If the Conference specific Focus  does not exist create a new Focus with
a unique URI derived from the Conference ID
* Forward the call using the standard Refer mechanism to the new Focus with
the Unique URI.

<ARC>
 You may also want to read
http://www.iptel.org/info/players/ietf/conferencing/draft-johnston-sipping-cc-conferencing-01.txt
.
Its provides the definition of a conference Factory URI, which could be
used to create new conferences and their Focii.
Note that when you send an INVITE to a factory URI, that entity returns the
actual Conference-URI in the Contact header.
Also, you may be using a central 'conference enabling application'
that  asks you lots of questions before allowing you to create/join  a new
conference. See an example of that in 4.3 of alan/orit's draft.
Note however , that when you want to use a "standard wellknown URI" that
has not been published to you by the conference server,
you are assuming that there is some agreement for that domain on conference
naming policies.
I am not sure I like your idea of a payload contaning conf-id (what use is
that ?)
Also, note that for scalability (if thats your primary concern), this
mechanism allows a way for you to start with a 'guessed URI' and from
then on communicate directly with you assigned focus without routing
everthing to what you call your 'master focus'.
</ARC>


Instead of dealing with Conference aware UA's and Conference Unaware UA's
we could assume all participants in the Conference to be Conference Aware.
The Conference Unaware participants are routed through a Gateway (e.g. an
IVR ) that provides a primitive interface to the conference unaware
participants and a Conference aware interface to the Conference
itself.  This would keep the model simpler and it can allow Conference
aware UAs a way to control the conference.

<ARC>
 It is extremely important to acknowledge conference-unaware endpoints. You
cannot assume an IVR or any other application in all
conference scenarios who 'aware' on behalf of an unaware UA. Again, reading
both Jonathan and Alan/Orit's draft in conjunction may motivate
you more on the requirement of conference unaware UAs. Remeber also , that
most of the EPs deployed in SIP have zilch idea of what isFocus is
as well as will not be aware of protocols to moderate a conference for a
while.
</ARC>


The other comment was in the area of the Conference policy control
protocol. Is this something like the SDP which is carried as a payload in
SIP or does it have specifics of it's own. Currently most of the operations
that are part of the CPCP are also possible through SIP based mechanisms.
Is there a need to define two parallel mechanisms within the same
framework.

<ARC>
I think what has been identifed here is a need for a conference policy
control protocol. It is not ascertained yet which protocol will deliver
these
requirements. Thats as far as I know. The existing SDP specs do not meet
all control requirements that come to my mind.

Alan - would you mind reposting that link of conferencing work in
progress  ? I thought I noticed some drafts attempting to address this in
more detail

Now a question again for the authors
of  draft-johnston-sipping-cc-conferencing-01.txt:

* section 4.5 callflow - you show alice sending an invite to a conf-factory
URI but the ladder diagram shows it being received by a 'Focus'.
The definition of a focus


3.1 Focus UA

  A focus, as defined in the framework, hosts a SIP conference and
  maintains a SIP signaling relationship with each participant in the
  conference. A focus contains a conference-aware user agent that
  supports the conferencing call control conventions as defined in this
  document.

Since a 'Focus' hosts 'a specific' conference, I find it a little
disconcerting
that the call flows show messages with a factory-URI reaching a focus. I
thought
it may be clearer, if we limit a 'focus' to be the target of a specific URI
meant
for a specific conference. If a person sends an INVITE to a factory-URI or
a 'conf-app-URI' < like an IVR etc which then contacts a Factory after auth
etc>
it should not be received by a Focus, should it ?

</ARC>

regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860  (M): 240 997 0066{@vtext.com}


                                                                           
     "Shiva"                                                               
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     Sent by:                            cc:                               
     sipping-admin@ietf.o                Subject:        [Sipping]         
     rg                          Comments on                               
                                 draft-rosenberg-sipping-conferencing-fram 
     02/22/2003 01:57 AM         ework-01.txt                              
     Please respond to                                                     
     shaj                                                                  
                                                                           
                                                                           




Hello,

Overall the draft looks good.  Here are some of my comments on this draft


The advantage of doing this is that I could have multiple Conference
servers that are running that are registered their master focus.

We can implement Load Balancing across Conference servers without the UA
realizing it as it is not connecting to a specific Focus on a specific
server. A second advantage is that if in the course of a meeting if the
focus were to go down or the server went down the participants can attempt
to rejoin the call. In such a case the Master Focus would see that the
Conference specific focus is down and recreate it allowing users to quickly
rejoin the conference.

Of course one needs to work out the details of the communication and
keepalive between the Master Focus and the Conference focus it creates.

The other advantage is if there are global Conference policies these can be
applied at the master and the Individual Conference focus needs to worry
about the specifics of the conference it is dealing with.

The other comment was in the area of the Conference policy control
protocol. Is this something like the SDP which is carried as a payload in
SIP or does it have specifics of it's own. Currently most of the operations
that are part of the CPCP are also possible through SIP based mechanisms.
Is there a need to define two parallel mechanisms within the same
framework.


Instead of dealing with Conference aware UA's and Conference Unaware UA's
we could assume all participants in the Conference to be Conference Aware.
The Conference Unaware participants are routed through a Gateway (e.g. an
IVR ) that provides a primitive interface to the conference unaware
participants and a Conference aware interface to the Conference
itself.  This would keep the model simpler and it can allow Conference
aware UAs a way to control the conference.

Regards
Shiva
Cisco Systems







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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Date: Fri, 28 Feb 2003 13:45:05 -0500
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Subject: [Sipping] Facing Up to Realities in SIP Deployments in Carrier networks
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I urge you to read:
http://www.vasaforum.org/Docs/SIPRFIAnalysisFinal-Web1202.doc

It contains a sobering assessment of where we are in SIP deployment
in carrier networks.  Of course, carrier networks are not the
only place SIP has a place, but this assessment still rings of
truth to me beyond this specific market, and spurs me to ask what 
we in the IETF SIPish community can do to address the issues this 
report cites.

First, read it.  Then we can talk.

Brian
_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 28 Feb 2003 14:46:26 -0500
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
From: Richard Shockey <richard@shockey.us>
Subject: Re: [Sipping] Facing Up to Realities in SIP Deployments in
  Carrier networks
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At 01:45 PM 2/28/2003 -0500, Rosen, Brian wrote:
>I urge you to read:
>http://www.vasaforum.org/Docs/SIPRFIAnalysisFinal-Web1202.doc
>
>It contains a sobering assessment of where we are in SIP deployment
>in carrier networks.  Of course, carrier networks are not the
>only place SIP has a place, but this assessment still rings of
>truth to me beyond this specific market, and spurs me to ask what
>we in the IETF SIPish community can do to address the issues this
>report cites.
>
>First, read it.  Then we can talk.
>
>Brian


Brian ..why would anything in this report suprise you ..its essentially 
written by and for existing carriers who essentially look at VoIP as 
cannibalistic to their existing revenue streams. Where SIP is deploying in 
competitive greenfield carrier environments such as Yahoo/Japan, Vonage, 
China etc its working very well and we all have heard of extremely positive 
user experiences from ordinary consumers.

I can almost agree with some the LEC's, to them there is no justification 
for SIP. Why should they engineer themselves out of monopoly profits ( and 
jobs ) to provide real and tangeable consumer benefits of lower costs and 
greater choice of services when the existing regulatory environment does 
not reward them for doing so..

I agree there are still major issues here ..DTMF,  E911 and other 
government mandated services..

and If I hear any more whining about OoS from people who have never made a 
SIP phone call I'm just going to scream..


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Richard Shockey'" <richard@shockey.us>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] Facing Up to Realities in SIP Deployments in Carrie
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Richard

I don't think getting into speculation on what motivates carriers
to deploy SIP is germane.  What is germane is:
·	If SIP-related standardization efforts has been slowing deployment.
·	If there are significant difficulties in interworking different
vendors' products in laboratory trials of both SIP to SIP and SIP to legacy
network element interoperability.  
·	If Efforts to provide lawful intercept and emergency services will
be hampered by the nature of SIP-based networks. 
·	If potential users have difficulty comparing products because
everyone uses different scalability metrics
·	If NAT and firewall traversal is a major issue limiting the
deployment of SIP
·	If a standard framework for basic authentication and privacy of
SIP-based communication is required
·	If there are significant new security issues relating to:
-	Interception of signaling and media traffic
-	Introduction of malicious traffic through open interfaces to SIP
endpoints or other networks
-	The need to authenticate endpoints through insecure networks
-	Securing network and service management interfaces based on open
standards.

Note that the above list is largely text lifted from the document.

If I think through these issues, I think most of them are just flat out
true.
It may be that some service providers don't care about them, but I think
most will.  I think that it is well worth our while to sort through this
list and figure out if the problems can be solved, and how they should be
solved.  It may be that IETF is not the right place to solve all of them.

Brian

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Friday, February 28, 2003 2:46 PM
> To: Rosen, Brian; 'sipping@ietf.org'
> Subject: Re: [Sipping] Facing Up to Realities in SIP Deployments in
> Carrier networks
> 
> 
> At 01:45 PM 2/28/2003 -0500, Rosen, Brian wrote:
> >I urge you to read:
> >http://www.vasaforum.org/Docs/SIPRFIAnalysisFinal-Web1202.doc
> >
> >It contains a sobering assessment of where we are in SIP deployment
> >in carrier networks.  Of course, carrier networks are not the
> >only place SIP has a place, but this assessment still rings of
> >truth to me beyond this specific market, and spurs me to ask what
> >we in the IETF SIPish community can do to address the issues this
> >report cites.
> >
> >First, read it.  Then we can talk.
> >
> >Brian
> 
> 
> Brian ..why would anything in this report suprise you ..its 
> essentially 
> written by and for existing carriers who essentially look at VoIP as 
> cannibalistic to their existing revenue streams. Where SIP is 
> deploying in 
> competitive greenfield carrier environments such as 
> Yahoo/Japan, Vonage, 
> China etc its working very well and we all have heard of 
> extremely positive 
> user experiences from ordinary consumers.
> 
> I can almost agree with some the LEC's, to them there is no 
> justification 
> for SIP. Why should they engineer themselves out of monopoly 
> profits ( and 
> jobs ) to provide real and tangeable consumer benefits of 
> lower costs and 
> greater choice of services when the existing regulatory 
> environment does 
> not reward them for doing so..
> 
> I agree there are still major issues here ..DTMF,  E911 and other 
> government mandated services..
> 
> and If I hear any more whining about OoS from people who have 
> never made a 
> SIP phone call I'm just going to scream..
> 
> 
>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> Richard Shockey, Senior Manager, Strategic Technology Initiatives
> NeuStar Inc.
> 46000 Center Oak Plaza  -   Sterling, VA  20166
> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
> <mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
>   <http://www.neustar.biz> ; <http://www.enum.org>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 28 Feb 2003 15:06:21 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sipping] Facing Up to Realities in SIP Deployments in Carrie	r
 networks
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I am not sure this is the right place for this discussion and that the
points here are really true or even more, are they relevant?

Not to speak of

> > and If I hear any more whining about OoS from people who have
> > never made a 
> > SIP phone call I'm just going to scream..

Get used to it: IP-IP phone calls over the public Internet are just
better than the PSTN and lab tests have shown the same for SIP based IP
conferencing.

With such concerns, we would not have a booming mobile phone industry
today, and as we know, concerns or not, mobile phones and the Internet
have already started to bite into the "carrier grade" networks where all
these concerns are presumably well taken care of. I will spare this list
of why/who is already using or is going to use SIP, and for very good
reasons :-)>

>It may be that IETF is not the right place to 
> solve all of them.

Right. 

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Rosen, Brian
> Sent: Friday, February 28, 2003 2:16 PM
> To: 'Richard Shockey'; 'sipping@ietf.org'
> Subject: RE: [Sipping] Facing Up to Realities in SIP 
> Deployments in Carrie r networks
> 
> 
> Richard
> 
> I don't think getting into speculation on what motivates 
> carriers to deploy SIP is germane.  What is germane is:
> .	If SIP-related standardization efforts has been slowing 
> deployment.
> .	If there are significant difficulties in interworking different
> vendors' products in laboratory trials of both SIP to SIP and 
> SIP to legacy network element interoperability.  
> .	If Efforts to provide lawful intercept and emergency 
> services will
> be hampered by the nature of SIP-based networks. 
> .	If potential users have difficulty comparing products because
> everyone uses different scalability metrics
> .	If NAT and firewall traversal is a major issue limiting the
> deployment of SIP
> .	If a standard framework for basic authentication and privacy of
> SIP-based communication is required
> .	If there are significant new security issues relating to:
> -	Interception of signaling and media traffic
> -	Introduction of malicious traffic through open interfaces to SIP
> endpoints or other networks
> -	The need to authenticate endpoints through insecure networks
> -	Securing network and service management interfaces based on open
> standards.
> 
> Note that the above list is largely text lifted from the document.
> 
> If I think through these issues, I think most of them are 
> just flat out true. It may be that some service providers 
> don't care about them, but I think most will.  I think that 
> it is well worth our while to sort through this list and 
> figure out if the problems can be solved, and how they should 
> be solved.  It may be that IETF is not the right place to 
> solve all of them.
> 
> Brian
> 
> > -----Original Message-----
> > From: Richard Shockey [mailto:richard@shockey.us]
> > Sent: Friday, February 28, 2003 2:46 PM
> > To: Rosen, Brian; 'sipping@ietf.org'
> > Subject: Re: [Sipping] Facing Up to Realities in SIP Deployments in 
> > Carrier networks
> > 
> > 
> > At 01:45 PM 2/28/2003 -0500, Rosen, Brian wrote:
> > >I urge you to read: 
> > >http://www.vasaforum.org/Docs/SIPRFIAnalysisFinal-Web1202.doc
> > >
> > >It contains a sobering assessment of where we are in SIP 
> deployment 
> > >in carrier networks.  Of course, carrier networks are not the only 
> > >place SIP has a place, but this assessment still rings of 
> truth to me 
> > >beyond this specific market, and spurs me to ask what we 
> in the IETF 
> > >SIPish community can do to address the issues this report cites.
> > >
> > >First, read it.  Then we can talk.
> > >
> > >Brian
> > 
> > 
> > Brian ..why would anything in this report suprise you ..its
> > essentially 
> > written by and for existing carriers who essentially look 
> at VoIP as 
> > cannibalistic to their existing revenue streams. Where SIP is 
> > deploying in 
> > competitive greenfield carrier environments such as 
> > Yahoo/Japan, Vonage, 
> > China etc its working very well and we all have heard of 
> > extremely positive 
> > user experiences from ordinary consumers.
> > 
> > I can almost agree with some the LEC's, to them there is no
> > justification 
> > for SIP. Why should they engineer themselves out of monopoly 
> > profits ( and 
> > jobs ) to provide real and tangeable consumer benefits of 
> > lower costs and 
> > greater choice of services when the existing regulatory 
> > environment does 
> > not reward them for doing so..
> > 
> > I agree there are still major issues here ..DTMF,  E911 and other
> > government mandated services..
> > 
> > and If I hear any more whining about OoS from people who have
> > never made a 
> > SIP phone call I'm just going to scream..
> > 
> > 
> >  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> > Richard Shockey, Senior Manager, Strategic Technology Initiatives 
> > NeuStar Inc.
> > 46000 Center Oak Plaza  -   Sterling, VA  20166
> > Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237 
> > <mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
> >   <http://www.neustar.biz> ; <http://www.enum.org> 
> > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Kevin McCandless <KMcCandless@verisign.com>
To: "'Richard Shockey'" <richard@shockey.us>,
        "Rosen, Brian"
	 <Brian.Rosen@marconi.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] Facing Up to Realities in SIP Deployments in Carrie
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comments inline.....

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Friday, February 28, 2003 1:46 PM
> To: Rosen, Brian; 'sipping@ietf.org'
> Subject: Re: [Sipping] Facing Up to Realities in SIP Deployments in
> Carrier networks
> 
> 
> At 01:45 PM 2/28/2003 -0500, Rosen, Brian wrote:
> >I urge you to read:
> >http://www.vasaforum.org/Docs/SIPRFIAnalysisFinal-Web1202.doc
> >
> >It contains a sobering assessment of where we are in SIP deployment
> >in carrier networks.  Of course, carrier networks are not the
> >only place SIP has a place, but this assessment still rings of
> >truth to me beyond this specific market, and spurs me to ask what
> >we in the IETF SIPish community can do to address the issues this
> >report cites.
> >
> >First, read it.  Then we can talk.
> >
> >Brian
> 
> 
> Brian ..why would anything in this report suprise you ..its 
> essentially 
> written by and for existing carriers who essentially look at VoIP as 
> cannibalistic to their existing revenue streams. Where SIP is 
> deploying in 
> competitive greenfield carrier environments such as 
> Yahoo/Japan, Vonage, 
> China etc its working very well and we all have heard of 
> extremely positive 
> user experiences from ordinary consumers.

That is one person's opinion.  We have a representative on the VASA Forum
and they do not portray the negative motives that are seen here.  The main
point is economics.  Does SIP/VOIP provide financial advantage to carriers
and is it mature enough for full scale deployment?  The resounding answer is
NO!  The carriers that did chase VOIP are either in chapter 11 or dissembled
and the equipment sold in an online auction.
> 
> I can almost agree with some the LEC's, to them there is no 
> justification 
> for SIP. Why should they engineer themselves out of monopoly 
> profits ( and 
> jobs ) to provide real and tangeable consumer benefits of 
> lower costs and 
> greater choice of services when the existing regulatory 
> environment does 
> not reward them for doing so..
> 
> I agree there are still major issues here ..DTMF,  E911 and other 
> government mandated services..
> 
> and If I hear any more whining about OoS from people who have 
> never made a 
> SIP phone call I'm just going to scream..

QOS is a major issue.  Many carriers have not deployed the underlying
transport Qos to support application like SIP.  What carriers offer end to
end QOS?  How many carriers have fully deployed the necessary MPLS and
DIFFSERV technology??

Wake up and smell the coffee.  The VASA report is the coffee.  It may be a
little bitter for those in Latte Land ;-)
> 
> 
>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> Richard Shockey, Senior Manager, Strategic Technology Initiatives
> NeuStar Inc.
> 46000 Center Oak Plaza  -   Sterling, VA  20166
> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
> <mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
>   <http://www.neustar.biz> ; <http://www.enum.org>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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At 03:16 PM 2/28/2003 -0500, Rosen, Brian wrote:
>Richard
>
>I don't think getting into speculation on what motivates carriers
>to deploy SIP is germane.

It is precisely germane... what motivates carriers is money, fear, greed 
and the economic benefits of one technology vs another, as well the 
regulatory environment they must live in.  This report is a classic version 
of whining why they do'n't want to deploy a superior form of technology 
when the real reasons are manifestly self evident to anyone who cares to 
read their balance sheets or understand the regulatory constraints many of 
them live with.

Most carriers are bankrupt, on the verge of bankruptcy, or worry about 
bankruptcy..and that includes US LEC's, major European PTT's and even  NTT 
in Japan as well ..


>What is germane is:
>·       If SIP-related standardization efforts has been slowing deployment.

IMHO no though the IETF does have it own unique ways of doing things the 
process for SIP has been successful and open.. unlike that of the ITU..and 
need I remind you how many years it took the ITU to produce anything in 
H.323 that resembled inter operability? That we have been able to produce 
as much as we have in as short a period of time and reverse the tide of 
technical opinion on H.323 is a near miracle and a testament to the 
Herculean work of SIP, SIPPING and SIMPLE WG's.

>·       If there are significant difficulties in interworking different
>vendors' products in laboratory trials of both SIP to SIP and SIP to legacy
>network element interoperability.

I cannot directly address the results of SIPit trials since my company does 
not directly have SIP products. I trust others can adequately address this 
issue.

>·       If Efforts to provide lawful intercept and emergency services will
>be hampered by the nature of SIP-based networks.

Our community is working on this and I believe we can deliver a better E911 
than is currently deployed.  At to the intercept issue .. geez I'm sorry 
..any ISP with a decent packet sniffer is already reading all of their 
traffic.... to the FBI, NSA , MI5 etal the Internet is the greatest bonanza 
for intercept ever created.. nothing is encrypted find the ports and you 
can get suck in  whatever you want.

NOT one ISP on the planet is not reading or sampling headers right now in 
order for them to optimize the route announcements on their network or make 
fundamental economic decisions about about how to transit or peer their 
traffic.

>·       If potential users have difficulty comparing products because
>everyone uses different scalability metrics

I'M SHOCKED I'M SHOCKED marketing guys devise smoke and mirrors to describe 
the features and  benefits of their products.

>·       If NAT and firewall traversal is a major issue limiting the
>deployment of SIP

Ok ..this is a problem .for consumer markets not for sensible enterprises 
that acquire their own routeable blocks of addreses. This is a larger 
problem of v4 vs v6 addressing we cannot solve here.

>·       If a standard framework for basic authentication and privacy of
>SIP-based communication is required

Privacy ... what privacy ...the entire signalling and media path is in the 
open but that does not seem to bother folks using SMTP.  I'd love 
opportunistic encryption of both the signalling and media paths but I've 
yet to see the problem adequately dealt with in email ..so  why is this 
suddenly a concern?

>·       If there are significant new security issues relating to:
>-       Interception of signaling and media traffic

M&M attacks sure ..

>-       Introduction of malicious traffic through open interfaces to SIP
>endpoints or other networks

Well Brian ..yes you are right on target here ..the last thing we need is 
open SIP "relays"

>-       The need to authenticate endpoints through insecure networks
>-       Securing network and service management interfaces based on open
>standards.
>
>Note that the above list is largely text lifted from the document.

but its exactly what a legacy bunch of  PPT bellheads would say .. thats my 
point..


>If I think through these issues, I think most of them are just flat out
>true.

Brian the simple problem effecting SIP deployment is much much larger..it 
goes to the heart of the current telecom meltdown and the implied threat 
that IP Communications has on carrier revenues. Once upon a time this was 
all a academic exercise to some service providers ...."oh yes this is very 
interesting we need to know more about blah blah" ..but times have changed 
and it is clear that from a purely revenue perspective VoIP is having a 
negative effect on TDM/SP revenues . VoIP dominated by H.323 now has 10% of 
the global market and has ripped 20 BILLION dollars out of the hands of 
International transport SP's.  The effect at the margin of SIP as it 
deploys in greenfield is amazing etc. Its effect on national LD rates is 
pronounced its clear ..at least in the US we are seeing the beginning of 
flat rate monthly distance free pricing, though I will submit that mobile 
is having a effect on these models as well.

I noted with some amusement the line in the document ..that there is no 
economic justification for deploying ENUM.  Well HELLO..... I've had this 
conversation .. the reply I got from a major US LEC I spoke to was  "OK 
what you are telling me enterprises are going to do this eventually but if 
I deploy this I'm going EBIDA negative in 24 months."


>It may be that some service providers don't care about them, but I think
>most will.

It depends on how high you sit on the PPT management food chain

>  I think that it is well worth our while to sort through this
>list and figure out if the problems can be solved, and how they should be
>solved.  It may be that IETF is not the right place to solve all of them.

Well I can tell you where they will be solved ....in the US it will be in 
either the bankruptcy courts or the FCC.


>Brian


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Facing Up to Realities in SIP Deployments in
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I can see this quickly getting out of hand and out of scope ..but one last 
parting shot then I'll retreat back to lurker land....


> > Brian ..why would anything in this report suprise you ..its
> > essentially
> > written by and for existing carriers who essentially look at VoIP as
> > cannibalistic to their existing revenue streams. Where SIP is
> > deploying in
> > competitive greenfield carrier environments such as
> > Yahoo/Japan, Vonage,
> > China etc its working very well and we all have heard of
> > extremely positive
> > user experiences from ordinary consumers.
>
>That is one person's opinion.  We have a representative on the VASA Forum
>and they do not portray the negative motives that are seen here.  The main
>point is economics.  Does SIP/VOIP provide financial advantage to carriers
>and is it mature enough for full scale deployment?

The word you missed was EXISTING carriers....to a greenfield player SIP 
works Japan is teaching us this ...If you have a massive investment in a C5 
infrastructure and a 20 year depreciation schedule .yes there is a big 
problem here :-)

>  The resounding answer is
>NO!  The carriers that did chase VOIP are either in chapter 11 or dissembled
>and the equipment sold in an online auction.

Well a bit too much early "irrational exuberance" is classic market 
behavior for "disruptive technologies" but any competitent business 
historian will be delighted to remind you that it is usually the second 
wave of companies deploying technology that typically reaps the major 
economic gains.

Microsoft being the classic case in point (they bought DOS and Word 
remember) ... and Kevin.. I'd be delighted to hear your views on the market 
effect of the MS Greenwich server on service provider revenues going forward.

> >
> > I can almost agree with some the LEC's, to them there is no
> > justification
> > for SIP. Why should they engineer themselves out of monopoly
> > profits ( and
> > jobs ) to provide real and tangeable consumer benefits of
> > lower costs and
> > greater choice of services when the existing regulatory
> > environment does
> > not reward them for doing so..
>
>
>QOS is a major issue.  Many carriers have not deployed the underlying
>transport Qos to support application like SIP.  What carriers offer end to
>end QOS?  How many carriers have fully deployed the necessary MPLS and
>DIFFSERV technology??

Kevin ..sorry ...this is just Bovine Effluent.. I'm sorry to lash out to 
you like this since I respect your opinion and work in other forums, but I 
believe you are dead wrong. MPLS and DIFSERVE have nothing to do with VoIP 
QoS.. even a modest VoIP carrier like ITXC does not need this kind of 
senseless overhead when they can monitor packet traffic effectively using 
clever techniques such as statistical analysis of traceroute data and 
reroute accordingly.

Any major IP back bone carrier ..when pressed will tell you that capability 
utilization in the back bone is about 20%. If you can get the traffic to 
the back bone you are home free. The bottlenecks to VoIP are typically the 
internal LAN that does not have 802.p priority bitting in their switch 
fabric and the access link.

This is what I hear from VoIP service providers slugging it out with 
customers day to day.

There is not the slightest doubt in my mind based on reports and 
presentations by IP Professionals that ATT, UUnet or Sprint .. individually 
could carry the entire voice traffic of the planet Earth via SIP on their 
networks and not even hiccup.

>Wake up and smell the coffee.  The VASA report is the coffee.  It may be a
>little bitter for those in Latte Land ;-)

This report strikes me a decaf .. fundamentally useless.

The key moving forward is can consumer or enterprises buy IP transport 
independently of SIP service delivery. SIP is fundamentally an application 
on the Internet not a service of the network.

Kevin on this I think we agree ..competition in telecom applications is a 
good thing.



 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

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You make some interesting and realistic points about carrier
deployments but are we trying to solve economical issues here ??

Brian, I understood that the goal of the paper you posted was
purely technological in nature since most of the points you made
addressed technical short-comings or issues the protocol is
trying to deal with.

More inline;

Richard Shockey wrote:

> At 03:16 PM 2/28/2003 -0500, Rosen, Brian wrote:
> >Richard
> >
> >I don't think getting into speculation on what motivates carriers
> >to deploy SIP is germane.
>
> It is precisely germane... what motivates carriers is money, fear, greed
> and the economic benefits of one technology vs another, as well the
> regulatory environment they must live in.  This report is a classic version
> of whining why they do'n't want to deploy a superior form of technology
> when the real reasons are manifestly self evident to anyone who cares to
> read their balance sheets or understand the regulatory constraints many of
> them live with.
>
> Most carriers are bankrupt, on the verge of bankruptcy, or worry about
> bankruptcy..and that includes US LEC's, major European PTT's and even  NTT
> in Japan as well ..
>
> >What is germane is:
> >·       If SIP-related standardization efforts has been slowing deployment.
>
> IMHO no though the IETF does have it own unique ways of doing things the
> process for SIP has been successful and open.. unlike that of the ITU..and
> need I remind you how many years it took the ITU to produce anything in
> H.323 that resembled inter operability? That we have been able to produce
> as much as we have in as short a period of time and reverse the tide of
> technical opinion on H.323 is a near miracle and a testament to the
> Herculean work of SIP, SIPPING and SIMPLE WG's.
>
> >·       If there are significant difficulties in interworking different
> >vendors' products in laboratory trials of both SIP to SIP and SIP to legacy
> >network element interoperability.
>
> I cannot directly address the results of SIPit trials since my company does
> not directly have SIP products. I trust others can adequately address this
> issue.
>
> >·       If Efforts to provide lawful intercept and emergency services will
> >be hampered by the nature of SIP-based networks.
>
> Our community is working on this and I believe we can deliver a better E911
> than is currently deployed.

What community are you referring to ???

>  At to the intercept issue .. geez I'm sorry
> ..any ISP with a decent packet sniffer is already reading all of their
> traffic.... to the FBI, NSA , MI5 etal the Internet is the greatest bonanza
> for intercept ever created.. nothing is encrypted find the ports and you
> can get suck in  whatever you want.
>
> NOT one ISP on the planet is not reading or sampling headers right now in
> order for them to optimize the route announcements on their network or make
> fundamental economic decisions about about how to transit or peer their
> traffic.

I think the point was totally missed. Nobody disagrees to the sniffing that you
are pointing to but the point here is about creating specific holes in the
deployment
where the data can be specifically targetted for tapping and routed under the
covers. This is covered as CALEA and regulatory requirements which to my
knoweldge is still an issue with many SIP vendors out there.

The question if I have to rephrase is: Does not having LI infrastructure a
bottleneck
to SIP deployment thereby discouraging carriers to not go the SIP route ??

>
>
> >·       If potential users have difficulty comparing products because
> >everyone uses different scalability metrics
>
> I'M SHOCKED I'M SHOCKED marketing guys devise smoke and mirrors to describe
> the features and  benefits of their products.
>
> >·       If NAT and firewall traversal is a major issue limiting the
> >deployment of SIP
>
> Ok ..this is a problem .for consumer markets not for sensible enterprises
> that acquire their own routeable blocks of addreses. This is a larger
> problem of v4 vs v6 addressing we cannot solve here.

I disagree on both fronts. Where you say not a problem for enterprise, the fact
of the matter is that NAT/PAT issues boil down to edges and this applies for
all SIP deployments. Enterprises deal this with fronting thier border proxies
that solve the translation problems but it still is an issue. On the other side
of v4
and v6 addressing story, the reality is we are going to have v4 addresses for a
longer period of time than previously thought and for successful SIP
deployments,
we ought to address it right here and I believe the WG is already working to
find a common solution.

>
>
> >·       If a standard framework for basic authentication and privacy of
> >SIP-based communication is required
>
> Privacy ... what privacy ...the entire signalling and media path is in the
> open but that does not seem to bother folks using SMTP.  I'd love
> opportunistic encryption of both the signalling and media paths but I've
> yet to see the problem adequately dealt with in email ..so  why is this
> suddenly a concern?

>
>
> >·       If there are significant new security issues relating to:
> >-       Interception of signaling and media traffic
>
> M&M attacks sure ..
>
> >-       Introduction of malicious traffic through open interfaces to SIP
> >endpoints or other networks
>
> Well Brian ..yes you are right on target here ..the last thing we need is
> open SIP "relays"
>
> >-       The need to authenticate endpoints through insecure networks
> >-       Securing network and service management interfaces based on open
> >standards.
> >
> >Note that the above list is largely text lifted from the document.
>
> but its exactly what a legacy bunch of  PPT bellheads would say .. thats my
> point..
>
> >If I think through these issues, I think most of them are just flat out
> >true.
>
> Brian the simple problem effecting SIP deployment is much much larger..it
> goes to the heart of the current telecom meltdown and the implied threat
> that IP Communications has on carrier revenues. Once upon a time this was
> all a academic exercise to some service providers ...."oh yes this is very
> interesting we need to know more about blah blah" ..but times have changed
> and it is clear that from a purely revenue perspective VoIP is having a
> negative effect on TDM/SP revenues . VoIP dominated by H.323 now has 10% of
> the global market and has ripped 20 BILLION dollars out of the hands of
> International transport SP's.  The effect at the margin of SIP as it
> deploys in greenfield is amazing etc. Its effect on national LD rates is
> pronounced its clear ..at least in the US we are seeing the beginning of
> flat rate monthly distance free pricing, though I will submit that mobile
> is having a effect on these models as well.
>
> I noted with some amusement the line in the document ..that there is no
> economic justification for deploying ENUM.  Well HELLO..... I've had this
> conversation .. the reply I got from a major US LEC I spoke to was  "OK
> what you are telling me enterprises are going to do this eventually but if
> I deploy this I'm going EBIDA negative in 24 months."
>
> >It may be that some service providers don't care about them, but I think
> >most will.
>
> It depends on how high you sit on the PPT management food chain
>
> >  I think that it is well worth our while to sort through this
> >list and figure out if the problems can be solved, and how they should be
> >solved.  It may be that IETF is not the right place to solve all of them.
>
> Well I can tell you where they will be solved ....in the US it will be in
> either the bankruptcy courts or the FCC.
>
> >Brian
>
>  >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
> Richard Shockey, Senior Manager, Strategic Technology Initiatives
> NeuStar Inc.
> 46000 Center Oak Plaza  -   Sterling, VA  20166
> Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
> <mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
>   <http://www.neustar.biz> ; <http://www.enum.org>
> <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
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Subject: RE: [Sipping] Questions regarding the draft-ietf-sipping-realtimefax-00.txt
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Hi Louis,

Sorry for the very late response, better later than never as they say...
Good catch. See inline, this will be reflected in an upcoming update.

Any objections to the understanding below?
Jean-Francois.
> -----Original Message-----
> From: Louis Guindon [mailto:lguindon@mediatrix.com] 
> Sent: Thursday, August 29, 2002 1:59 PM
> To: 'sipping@ietf.org'
> Subject: [Sipping] Questions regarding the 
> draft-ietf-sipping-realtimefax-00.txt
> 
> 
> Hi,
> 
> I have question regarding the format of a att-field in the 
> SDP packet. In all examplesin the 
> draft-ietf-sipping-realtimefax-00.txt, we see
>       a=T38FaxFillBitRemoval:0 
>       a=T38FaxTranscodingMMR:0 
>       a=T38FaxTranscodingJBIG:0
> 
> In the section D.2.3 of ITU-T Rec. T.38 (04/2002) for 
> capabilities negotiation, it's stated that"Note that options 
> without values are boolean - their presence indicates that 
> they are valid for the session". This means that 
> "T38FaxFillBitRemoval", "T38FaxTranscodingMMR" and 
> "T38FaxTranscodingJBIG" are considered boolean.
> 
> Only the present of T38FaxFillBitRemoval attribute without 
> value should be enough to determine if it's valid or not for 
> the session if I consider the RFC2327 (April 1998, page 18). 
> 
> Question: Why are we seeing the ":0" at the end? 
> 
> Can anyone confirm the following syntax for a SDP packet?
>       a=T38FaxFillBitRemoval                 <== Means valid 
> for the session
> or incorrect syntax?
This is the only correct option based on my reading of the latest T.38 recommendation (from 3/2002). Therefore, I have removed the media attribute lines for "T38FaxFillBitRemoval", "T38FaxTranscodingMMR" and "T38FaxTranscodingJBIG".

>       a=T38FaxFillBitRemoval:0              <== Means 
> invalid/valid for the
> session?
Incorrect syntax. But given some implementations, I'd recommend that implementaitons be flexible when receiving this media line attribute (if : is present, then get the boolean value)...

>       a=T38FaxFillBitRemoval:1              <== Means valid 
> for the session
> or incorrect syntax?
Incorrect syntax.

> 
> Regards,
> 
> > Louis Guindon
> > Software Design Engineer
> > Mediatrix Telecom, Inc.
> > lguindon@mediatrix.com
> > Tel :  819-829-8749 ext. 242
> > Fax : 819-829-1973
> > 4229, Garlock Street
> > Sherbrooke, Qc J1L 2C8
> > 
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> > livrer ce message ou son(ses) fichier(s) adjoint(s) à 
> quiconque.  Vous 
> > devriez plutôt effacer de façon permanente ce message et son(ses) 
> > fichier(s)
> > adjoint(s) et aimablement en avertir l'expéditeur par 
> courriel.  Aucune
> > garantie n'est donnée que ce courriel ou son(ses) 
> fichier(s) adjoint(s) ne
> > contiennent pas de virus informatique ou autre défectuosité.
> > 
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