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From: "Beck01, Wolfgang" <BeckW@t-systems.com>
To: sipping@ietf.org
Subject:  [Sipping] smartcards and SIP, was:  FYI: RADIUS & SIP
Date: Fri, 1 Aug 2003 08:56:43 +0200 
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Henning Schulzrinne [mailto:hgs@cs.columbia.edu] wrote:
> Also, while SIM cards work ok when kept in the cell phone, the existing 
> card mechanics don't seem to work well if you want to, say, quickly use 
> the phone in the meeting room. Would you need to remove and insert the 
> SIM card each time? "Oops, I left the card in my office phone, I can't 
> make a call from home right now."
"Oops, i forgot to remove my home identity from my office phone. Now my
co-worker will be disturbed by aunt Hillary's phone calls."

Seriously, there are problems with smartcards: you will need to have one
for every phone you want to use in parallel. The usage is different, too.
A cell phone is a personal item and there is no need to change SIM cards
frequently. If people 'personalize' stationary SIP phones, this would be
different. In this case, a credit card sized SIM card would be better
than the thumbnail-sized variety. With WiFi SIP phones, the need to
change SIM cars would go away.

There are problems with soft clients, too. Some companies are using
smartcards instead of login passwords. How many smartcard readers
fit into a laptop?

--
Wolfgang Beck
T-Systems Nova GmbH 

_______________________________________________
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>
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Michael,

you wrote:
> The way GSM authentication works is that the Authentication Center produces 
> "triplets" (tuples of precomputed challenge, matching response and session 
> key) for a given IMSI.
I've been told that getting IMSIs without having a GSM account is not that easy.
There seem to be regulatory issues and political pressure from mobile operators,
at least in some European countries. Do you know details about this?

--
Wolfgang Beck
T-Systems Nova GmbH 

_______________________________________________
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, 01 Aug 2003 10:17:19 -0400
To: "Beck01, Wolfgang" <BeckW@t-systems.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] smartcards and SIP, was:  FYI: RADIUS & SIP
Cc: sipping@ietf.org
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 om.de>
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--=====================_3428219==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

For what it's worth:

A number of U.S. Federal Agencies are issuing smart-cards for 
authentication of users on PCs and similar devices.  For some (i.e. 
military ID-smart-cards) there are serious consequences if they leave that 
lying around.

Smart card readers can connect through USB ports.

GSM triplets: this may be outdated technology.  Try quintuplets.

Mike

At 08:56 AM 8/1/2003 +0200, Beck01, Wolfgang wrote:


>Henning Schulzrinne [mailto:hgs@cs.columbia.edu] wrote:
> > Also, while SIM cards work ok when kept in the cell phone, the existing
> > card mechanics don't seem to work well if you want to, say, quickly use
> > the phone in the meeting room. Would you need to remove and insert the
> > SIM card each time? "Oops, I left the card in my office phone, I can't
> > make a call from home right now."
>"Oops, i forgot to remove my home identity from my office phone. Now my
>co-worker will be disturbed by aunt Hillary's phone calls."
>
>Seriously, there are problems with smartcards: you will need to have one
>for every phone you want to use in parallel. The usage is different, too.
>A cell phone is a personal item and there is no need to change SIM cards
>frequently. If people 'personalize' stationary SIP phones, this would be
>different. In this case, a credit card sized SIM card would be better
>than the thumbnail-sized variety. With WiFi SIP phones, the need to
>change SIM cars would go away.
>
>There are problems with soft clients, too. Some companies are using
>smartcards instead of login passwords. How many smartcard readers
>fit into a laptop?
>
>--
>Wolfgang Beck
>T-Systems Nova GmbH
>
>_______________________________________________
>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

--=====================_3428219==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<font size=3>For what it's worth:<br>
<br>
A number of U.S. Federal Agencies are issuing smart-cards for
authentication of users on PCs and similar devices.&nbsp; For some (i.e.
military ID-smart-cards) there are serious consequences if they leave
that lying around.<br>
<br>
Smart card readers can connect through USB ports.<br>
<br>
GSM triplets: this may be outdated technology.&nbsp; Try
quintuplets.<br>
<br>
Mike<br>
<br>
At 08:56 AM 8/1/2003 +0200, Beck01, Wolfgang wrote:<br>
<br>
<br>
<blockquote type=cite cite>Henning Schulzrinne
[<a href="mailto:hgs@cs.columbia.edu" eudora="autourl">mailto:hgs@cs.columbia.edu</a>]
wrote:<br>
&gt; Also, while SIM cards work ok when kept in the cell phone, the
existing <br>
&gt; card mechanics don't seem to work well if you want to, say, quickly
use <br>
&gt; the phone in the meeting room. Would you need to remove and insert
the <br>
&gt; SIM card each time? &quot;Oops, I left the card in my office phone,
I can't <br>
&gt; make a call from home right now.&quot;<br>
&quot;Oops, i forgot to remove my home identity from my office phone. Now
my<br>
co-worker will be disturbed by aunt Hillary's phone calls.&quot;<br>
<br>
Seriously, there are problems with smartcards: you will need to have
one<br>
for every phone you want to use in parallel. The usage is different,
too.<br>
A cell phone is a personal item and there is no need to change SIM
cards<br>
frequently. If people 'personalize' stationary SIP phones, this would
be<br>
different. In this case, a credit card sized SIM card would be
better<br>
than the thumbnail-sized variety. With WiFi SIP phones, the need to<br>
change SIM cars would go away.<br>
<br>
There are problems with soft clients, too. Some companies are using<br>
smartcards instead of login passwords. How many smartcard readers<br>
fit into a laptop?<br>
<br>
--<br>
Wolfgang Beck<br>
T-Systems Nova GmbH <br>
<br>
_______________________________________________<br>
Sipping mailing list&nbsp;
<a href="https://www1.ietf.org/mailman/listinfo/sipping" eudora="autourl">https://www1.ietf.org/mailman/listinfo/sipping</a><br>
This list is for NEW development of the application of SIP<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sip@ietf.org for new developments of core SIP
</font></blockquote></html>

--=====================_3428219==_.ALT--


_______________________________________________
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: "Christian Stredicke" <stredicke@snom.de>
To: <sipping@ietf.org>
Date: Sat, 2 Aug 2003 14:04:19 +0200
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Subject: [Sipping] Late Media
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Hi,

I prepared a document on "Late Media". It provides a (as I think) cool
way to finish a call like in the James Bond films (video) and allows
tones known from good old PSTN.

Please take a look at 

http://www.snom.com/draft-stredicke-late-media-00.txt

and let me know what you think.

Christian

P.S. Maybe someone can tell me how to publish this document properly?
--
Christian Stredicke
sip:christian@stredicke.de


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From: "g.jayadevan \(Yahoo!\)" <gjayadevan@yahoo.com>
Subject: RE: [Sipping] Re: Update to 3pcc - corrected 
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Here's my take the e2e vs. a centralized model.

When considering best practices, I prefer to think
in term of building blocks. The c-t-d is only one
application of the 3pcc model. If we took the
REFER approach, it becomes a point solution
applicable well only to a single 3pcc call.
The 3pcc model can be generalized to many
call legs allowing a whole class of application
that require these manipulations to be built. Should
we build efficient point solutions or a generalized
service layer that will allow me to reuse
capability?

Imagine from one customer service agent, I will
like to xfer the call to another. And possibly
add a supervisor. Seems to me that all the phones
at the call center will have to have intelligent
applications loaded to perform these nifty 
manipulations.

Reliability of the controller approach can be made
up in other ways. The controller itself can be 
made reliable by a distibuted protocols such as 
persisting call information, and adding protocol
fault tolerance via SCTP or a distributive proxy.
In the 3GPP world, a model borrowed from the
IN world, an application server is a key player.

e2e is a great concept, but I prefer not to 
architect solution in order to adhere to the 
principle when it does not buy me convenience
of deployment. Like others have pointed out
billing issues get confusing.

And the controller must be thought of as being an
end player too.




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Thread-Topic: draft-melanchuk-sipping-msml-00.txt
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From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Garland Sharratt" <gsharratt@convedia.com>, <timm@convedia.com>
Cc: <sipping@ietf.org>
Subject: [Sipping] draft-melanchuk-sipping-msml-00.txt
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Garland,Tim,
=20
I finally had the opportunity to review the draft
'draft-melanchuk-sipping-msml-00.txt' - very impressive. This certainly
seems a flexible and powerful approach to media server control in
conjunction with 'draft-melanchuk-sipping-moml-00.txt'.  Please find
enclosed a couple of questions raised during my review as well as a
couple of minor 'nits'.
=20
Questions
=20
- Would any benefit be gained from including an optional 'order'
attribute on primitive elements so that when 'pipe' is used, the order
is determined explicitly rather than following a logical path?
=20
- Was there a reason that JOIN/UNJOIN were limited to id1 + id2.  Could
it not have just been id, that was forced to appear at least twice?
This would allow you to join >2 media streams in one transaction.
=20
- <result> element should say that the response code corresponds to the
SIP response code received for the transaction.
=20
- P17 - Media objects may be defined using MOML or other suitable media=20
   resource description languages. A media server which supports=20
   <insert> and <remove> must support MOML and may support other=20
   languages. Those languages must support the ability to identify the=20
   left hand and right hand streams of full duplex objects.
Should this be a 'MUST' or is it not possible to only use another
'suitable media resource description language' with MSML?
=20
Nits
=20
From Abstract:-
Clients can use it define how media sessions interact on a media server
and to apply services to individual or groups of users.
Should read:-
Clients can use it to define how media sessions interact on a media
server and to apply services to individual or groups of users.
=20
2.2 reads:-
Future references to that object must use that name in a structured
format as defined below.
Should:-
Make reference to explicit section/table.
=20
2.2 reads:-
Terms may be either an object class or an object class followed by a
colon ':' followed by qualifier which identifies an object within the
class.
Should read:-
Terms may be either an object class or an object class followed by a
colon ':', followed by a qualifier which identifies an object within the
class.
=20
2.2 reads:-
Dialogs contexts, and object instances of application defined classes,
have identifiers relative to a connection or conference.
Should read:-
Dialog contexts, and object instances of application defined classes,
have identifiers relative to a connection or conference.
=20
4.0 reads:-
Transaction results are returned as part of the SIP request response.
Should read:-
Transaction results are returned as part of the SIP request/response
model.
=20
The Document Footer reads:-
Media Sessions Markup Language (MOML)
Should read:-
Media Sessions Markup Language (MSML)
=20
5.8 - The <dialogend> example does not have an id attribute BUT it is
mandatory.
=20
A number of MAY, MUST etc have not been capitalised throughout the
document e.g. section 5.9.=20
=20
=20
HTH,
=20
Best Regards,
=20
Chris.
=20
=20
------------------------------------------------
Chris Boulton
Ubiquity Software
=20
Tel : +44 (0) 1633765600
Fax : +44 (0) 1633765601=20
------------------------------------------------
=20
=20

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<div class=3DSection1>

<p class=3DMsoNormal><span class=3DSpellE><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Garland<span =
class=3DGramE>,Tim</span></span></font></span><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>,<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'>I finally had the opportunity to review the draft =
&#8216;draft-melanchuk-sipping-msml-00.txt&#8217;
&#8211; very impressive. This certainly seems a flexible and powerful =
approach to
media server control in conjunction with =
&#8216;draft-melanchuk-sipping-moml-00.txt&#8217;.<span
style=3D'mso-spacerun:yes'>&nbsp; </span>Please find enclosed a couple =
of
questions raised during my review as well as a couple of minor =
&#8216;nits&#8217;.<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><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Ques=
tions<o:p></o:p></span></font></u></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- Would any benefit be =
gained
from including an optional &#8216;order&#8217; attribute on primitive =
elements
so that when &#8216;pipe&#8217; is used, the order is determined =
explicitly
rather than following a logical path?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- Was there a reason =
that
JOIN/UNJOIN were limited to id1 + id2.<span =
style=3D'mso-spacerun:yes'>&nbsp;
</span>Could it not have just been <span class=3DGramE>id, that</span> =
was forced
to appear at least twice?<span style=3D'mso-spacerun:yes'>&nbsp; =
</span>This
would allow you to join &gt;2 media streams in one =
transaction.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- &lt;result&gt; =
element
should say that the response code corresponds to the SIP response code =
received
for the transaction.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- P17 - Media objects =
may be
defined using MOML or other suitable media <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span><span =
class=3DGramE>resource</span>
description languages. A media server which supports =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>&lt;<span =
class=3DGramE>insert</span>&gt;
and &lt;remove&gt; must support MOML and may support other =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span><span =
class=3DGramE>languages</span>.
Those languages must support the ability to identify the =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><span
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span><span =
class=3DGramE>left</span> hand
and right hand streams of full duplex =
objects.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should this be a
&#8216;MUST&#8217; or is it not possible to only use another =
&#8216;suitable
media resource description language&#8217; with =
MSML?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Nits=
<o:p></o:p></span></font></u></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>From =
Abstract:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Clients can use it =
define how
media sessions interact on a media server and to apply services to =
individual
or groups of users.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Clients can use it to =
define
how media sessions interact on a media server and to apply services to
individual or groups of users.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>2.2 =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Future references to =
that
object must use that name in a structured format as defined =
below.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should:-<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Make reference to =
explicit
section/table.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>2.2 =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Terms may be either an =
object
class or an object class followed by a colon ':' followed by qualifier =
which
identifies an object within the class.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Terms may be either an =
object
class or an object class followed by a colon ':', followed by a =
qualifier which
identifies an object within the class.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>2.2 =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Dialogs contexts, and =
object
instances of application defined classes, have identifiers relative to a
connection or conference.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Dialog contexts, and =
object instances
of application defined classes, have identifiers relative to a =
connection or conference.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>4.0 =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Transaction results =
are
returned as part of the SIP request =
response.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Transaction results =
are
returned as part of the SIP request/response =
model.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>The Document Footer =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Media Sessions <span
class=3DSpellE>Markup</span> Language =
(MOML)<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Media Sessions <span
class=3DSpellE>Markup</span> Language =
(MSML)<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>5.8 &#8211; The =
&lt;<span
class=3DSpellE>dialogend</span>&gt; example does not have an id =
attribute BUT it
is mandatory.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>A number of MAY, MUST =
etc
have not been capitalised throughout the document e.g. section 5.9. =
<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'>HTH,<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'>Best Regards,<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'>Chris.<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 style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue;
mso-no-proof:yes'>------------------------------------------------<o:p></=
o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dred
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:red;
mso-no-proof:yes'>Chris Boulton</span></font><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'><o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Ubiquity =
Software<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'><o:p>&nbsp;=
</o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Tel : +44 =
(0)
1633765600<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Fax : +44 =
(0)
1633765601 <o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue;
mso-no-proof:yes'>------------------------------------------------</span>=
</font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
yes'><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;mso-no-proof:yes'>&nbsp;</span><o:p></o:p></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>

</div>

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Date: Mon, 4 Aug 2003 10:57:25 +0100
Message-ID: <45730E094814E44488F789C1CDED27AE0219B066@gbnewp0758m.eu.ubiquity.net>
Thread-Topic: draft-melanchuk-sipping-moml-00.txt
Thread-Index: AcNabs25uQEysWFISQiYvKC7A7aLwg==
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Garland Sharratt" <gsharratt@convedia.com>, <timm@convedia.com>
Cc: <sipping@ietf.org>
Subject: [Sipping] draft-melanchuk-sipping-moml-00.txt
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Garland, Tim,
=20
Just a follow-up mail regarding some questions/nits from this draft.
Again, seems extremely flexible.  Hope the comments are productive.
=20
Questions
=20
- Is there an alternative method for defining unlimited occurrences
using the 'max' attribute =3D 0 (e.g. pattern element uses max within =
DTMF
primitive).  This seemed a little confusing.
=20
- 4.5.1.1 The description of <tvoice> and <tsilence> needs to be made
more explicit.=20
=20
Nits
=20
Use of lower case for may and must not on p7, p8, p15.
=20
4.6 reads:-
Gain is used to adjust of the gain of a media stream by a specific
amount. =20
Should read:-
The <gain> child element is used to adjust the gain of a media stream by
a specific amount. =20
=20
=20
Best Regards,
=20
Chris.
=20
=20
------------------------------------------------
Chris Boulton
Ubiquity Software
=20
Tel : +44 (0) 1633765600
Fax : +44 (0) 1633765601=20
------------------------------------------------
=20
=20

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<body lang=3DEN-US link=3Dblue vlink=3Dpurple =
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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'>Garland, Tim,<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><span class=3DGramE><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Just a follow-up mail =
regarding some
questions/nits from this draft.</span></font></span><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'><span
style=3D'mso-spacerun:yes'>&nbsp; </span>Again, seems extremely =
flexible.<span
style=3D'mso-spacerun:yes'>&nbsp; </span>Hope the comments are =
productive.<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><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Ques=
tions<o:p></o:p></span></font></u></p>

<p class=3DMsoNormal><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p=
><span
 =
style=3D'text-decoration:none'>&nbsp;</span></o:p></span></font></u></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- Is there an =
alternative
method for defining unlimited occurrences using the &#8216;max&#8217; =
attribute
=3D 0 (e.g. pattern element uses max within DTMF primitive).<span
style=3D'mso-spacerun:yes'>&nbsp; </span>This seemed a little =
confusing.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>- 4.5.1.1 The =
description of
&lt;<span class=3DSpellE>tvoice</span>&gt; and &lt;<span =
class=3DSpellE>tsilence</span>&gt;
needs to be made more explicit. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p=
><span
 =
style=3D'text-decoration:none'>&nbsp;</span></o:p></span></font></u></p>

<p class=3DMsoNormal><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Nits=
<o:p></o:p></span></font></u></p>

<p class=3DMsoNormal><u><font size=3D2 face=3DArial><span lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p=
><span
 =
style=3D'text-decoration:none'>&nbsp;</span></o:p></span></font></u></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Use of lower case for =
may and
must not on p7, p8, p15.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'><o:p>&nbsp;</o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>4.6 =
reads:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Gain is used to adjust =
of the
gain of a media stream by a specific amount.<span
style=3D'mso-spacerun:yes'>&nbsp; </span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>Should =
read:-<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial;mso-ansi-language:EN-GB'>The &lt;gain&gt; child
element is used to adjust the gain of a media stream by a specific =
amount.<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'><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'>Best Regards,<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'>Chris.<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 style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue;
mso-no-proof:yes'>------------------------------------------------<o:p></=
o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dred
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:red;
mso-no-proof:yes'>Chris Boulton</span></font><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'><o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Ubiquity =
Software<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'><o:p>&nbsp;=
</o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Tel : +44 =
(0)
1633765600<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:yes'>Fax : +44 =
(0)
1633765601 <o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'mso-layout-grid-align:none'><font size=3D2 =
color=3Dblue
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:blue;
mso-no-proof:yes'>------------------------------------------------</span>=
</font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;mso-no-proof:
yes'><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;mso-no-proof:yes'>&nbsp;</span><o:p></o:p></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>

</div>

</body>

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------_=_NextPart_001_01C35A6E.CE9BE3CA--

_______________________________________________
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This list is for NEW development of the application of SIP
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From exim@www1.ietf.org  Mon Aug  4 11:32:38 2003
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Date: Mon, 04 Aug 2003 10:22:31 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Re: Update to 3pcc - corrected
In-reply-to: <20030803172034.43720.qmail@web10901.mail.yahoo.com>
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> e2e is a great concept, but I prefer not to
> architect solution in order to adhere to the
> principle when it does not buy me convenience
> of deployment.

Think of the example of one or more ISP's in the path of a call from an
enterprise local SIP outgoing proxy to a call center with complex services
at both ends that the ISP's need not necessarily understand. This is clearly
a case where only the e2e service model will work. Inserting a B2BUA in the
middle would break any usefulness to the caller and basically make the user
avoid the offending ISP.

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> g.jayadevan (Yahoo!)
> Sent: Sunday, August 03, 2003 12:21 PM
> To: sipping@ietf.org
> Subject: RE: [Sipping] Re: Update to 3pcc - corrected
> 
> Here's my take the e2e vs. a centralized model.
> 
> When considering best practices, I prefer to think
> in term of building blocks. The c-t-d is only one
> application of the 3pcc model. If we took the
> REFER approach, it becomes a point solution
> applicable well only to a single 3pcc call.
> The 3pcc model can be generalized to many
> call legs allowing a whole class of application
> that require these manipulations to be built. Should
> we build efficient point solutions or a generalized
> service layer that will allow me to reuse
> capability?
> 
> Imagine from one customer service agent, I will
> like to xfer the call to another. And possibly
> add a supervisor. Seems to me that all the phones
> at the call center will have to have intelligent
> applications loaded to perform these nifty
> manipulations.
> 
> Reliability of the controller approach can be made
> up in other ways. The controller itself can be
> made reliable by a distibuted protocols such as
> persisting call information, and adding protocol
> fault tolerance via SCTP or a distributive proxy.
> In the 3GPP world, a model borrowed from the
> IN world, an application server is a key player.
> 
> e2e is a great concept, but I prefer not to
> architect solution in order to adhere to the
> principle when it does not buy me convenience
> of deployment. Like others have pointed out
> billing issues get confusing.
> 
> And the controller must be thought of as being an
> end player too.
> 
> 
> 
> 
> __________________________________
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> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.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
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Date: Tue, 05 Aug 2003 18:37:32 +0200
To: "g.jayadevan (Yahoo!)" <gjayadevan@yahoo.com>, sipping@ietf.org
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sipping] Re: Update to 3pcc - corrected 
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At 07:20 PM 8/3/2003, g.jayadevan \(Yahoo!\) wrote:
>Here's my take the e2e vs. a centralized model.
>
>When considering best practices, I prefer to think
>in term of building blocks. The c-t-d is only one
>application of the 3pcc model. If we took the
>REFER approach, it becomes a point solution
>applicable well only to a single 3pcc call.
>The 3pcc model can be generalized to many
>call legs allowing a whole class of application
>that require these manipulations to be built. Should
>we build efficient point solutions or a generalized
>service layer that will allow me to reuse
>capability?

If we should propose "BCP" tools for a generalized
service layer, than we need to make a case. I don't think
we make a case with an example which can be better solved
other way.

>Imagine from one customer service agent, I will
>like to xfer the call to another. And possibly
>add a supervisor. Seems to me that all the phones
>at the call center will have to have intelligent
>applications loaded to perform these nifty 
>manipulations.

Well, phones that don't support call transfer/REFER
should be better junked than used in architectural
discussions.

>Reliability of the controller approach can be made
>up in other ways. The controller itself can be 
>made reliable by a distibuted protocols such as 
>persisting call information, and adding protocol
>fault tolerance via SCTP or a distributive proxy.
>In the 3GPP world, a model borrowed from the
>IN world, an application server is a key player.
>
>e2e is a great concept, but I prefer not to 
>architect solution in order to adhere to the 
>principle when it does not buy me convenience
>of deployment. 

As you might have read previously: REFER is more
convenient than b2bua -- so we don't need to open
a religious discussion about convenience versus
principles. To refresh your mind: there is fair
difference in complexity, interoperability and user 
experience.

>Like others have pointed out billing issues get confusing.

I still have not seen a problem description, so it is
hard to comment.

>And the controller must be thought of as being an
>end player too.

I have no opinion on this statement since I don't understand it.

-jiri


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From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: [Sipping] Re: Update to 3pcc
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Hello David,

I don't see how billing servers are related to method of 
call initiation.

Actually I believe that how a call is initiated (by manual
dialing a destination, by REFER-based click-to-dial, by b2bua-based
click-to-dial, CTI, etc.) is completely orthogonal to how the 
call is charged for. Regardless what charging system you use 
(if any), there will be different methods for initiating calls, 
some will be better off than others. I maintain that a click-to-dial 
b2bua is not a BCP.

-jiri

At 10:10 AM 7/30/2003, David Schwartz wrote:
>Hi Jiri
>
>REFER is good solution if you assume that either there are no billing
>issues, or alternatively, that the endpoint/s handle the billing issues. If
>this is not the case, for example, in an application where a centralized
>controller is setting up a call to two PSTN numbers, then you will need a
>centralized billing entity and REFER will not suffice.
>
>David
>
>-----Original Message-----
>From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
>Sent: Mon, July 28, 2003 7:00 PM
>To: sipping@ietf.org
>Subject: Sipping digest, Vol 1 #1144 - 2 msgs
>
>
>Send Sipping mailing list submissions to
>        sipping@ietf.org
>
>To subscribe or unsubscribe via the World Wide Web, visit
>        https://www1.ietf.org/mailman/listinfo/sipping
>or, via email, send a message with subject or body 'help' to
>        sipping-request@ietf.org
>
>You can reach the person managing the list at
>        sipping-admin@ietf.org
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of Sipping digest..."
>
>
>Today's Topics:
>
>   1. Re: Diversion header (Cullen Jennings)
>   2. Re: Update to 3pcc (Jiri Kuthan)
>
>--__--__--
>
>Message: 1
>Date: Sun, 27 Jul 2003 19:05:10 -0700
>Subject: Re: [Sipping] Diversion header
>From: Cullen Jennings <fluffy@cisco.com>
>To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
>
>
>I assume your question is there some way for  for the final destination that
>receives the call to know the who the 1st callee was that had some device in
>the network redirect the call elsewhere.
>
>The answer is YES - there has been a lot of confusion over this so think
>about my answer carefully - it will seem counter intuitive at first. The
>answer is the "To" header but it will take me a second to explain why.
>
>Imagine case 1 where A send a call to proxy B,  that sent to to C. Now C
>sent a 302 back B and B then forked the call to D. The To header in the
>message D receives will still have the To set to C. C is the element that
>sent the first 302.
>
>Now imagine a case (like your question) where there is not a recursive
>proxy. The 302 will go all the way back to A. Now A forms a new invite to
>but it can leave the To header set to the same thing it was in the original
>message (C in this case). When the message arrives at D, the To header will
>be set to C.
>
>If D was a voice mail system, it might be able to know to act on behalf of C
>instead of some other entity.
>
>There are more complex things that relate to your request. Mary Barnes has
>been moving the history work along
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-04.txt
>
>There is some work on QSIG mappings with diversion
>http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2sip-diversion-
>00.txt
>
>There is a draft-levy-sip-diversion which I (and many others) believe has a
>pretty much zero chance of being accepted or becoming a RFC. This work has
>been superceded by the history work. Quite a few people do not support the
>history work and it is moving very slowly. Don't expect any of this to
>resolve quickly.
>
>Give some careful thought to if the "To" header meets your needs and if it
>does use - The To header is widely implemented :-)
>
>
>Cullen
>
>
>
>
>On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
>
>> Is there any SIP header defined in any SIP RFC or Internet Draft that can
>> include the address of the 1st callee, who redirected the call using 302,
>in
>> the 2nd outgoing INVITE?
>> 
>> _______________________________________________
>> 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
>> 
>
>
>
>--__--__--
>
>Message: 2
>Date: Mon, 28 Jul 2003 10:17:51 +0200
>To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping <sipping@ietf.org>
>From: Jiri Kuthan <jiri@iptel.org>
>Subject: Re: [Sipping] Update to 3pcc
>
>I'm actually not convinced that it is proper to show click-to-dial
>as a motivating example for use of b2bua and label it BCP.
>
>Use of REFER seems technically a superior solution worth being called
>BCP. Roots of the advantages are in a better application of e2e principle. 
>REFER takes no middle-box b2bua -- it delegates signaling to end-points.
>Advantages include:
>- All the pitfalls with media are so straigforward, they don't even 
>  need to be described (like now in sessions 8 and 9).  
>- It takes no media server to transmit ringing tone. 
>- There are no SDP compatibility issues on controller's side.
>
>-jiri
>
>At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
>>Folks,
>>
>>You may have noticed that I posted an update to the 3pcc spec:
>>
>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
>>
>>The update is based on comments received from IESG while reviewing the
>draft. In particular, there were problems with the usage of 0.0.0.0 to
>imply that no media nor RTCP should be sent. This IP address is not meant
>for that purpose. Rather, the .invalid TLD is much better suited. So, I
>changed the draft to make use of .invalid in flows where the controller
>doesnt want media to be sent.
>>
>>Now, there are some concerns with that which folks need to be aware of:
>>
>>1. It is at odds with RFC 3264, which actually recommends 0.0.0.0. In fact,
>the problem is really with RFC 3264, NOT 3pcc. However, 3264 is done, and
>this one isnt. I logged a bug against RFC 3264 on this, and we'll need to
>work on a way to transition folks towards using .invalid and not 0.0.0.0.
>>
>>2. One the premises of 3pcc is that it should be compatible with any UA
>that is RFC 3261 compliant. If a controller uses .invalid in the SDP, I am
>not sure what a SIP endpoint would do. I don't believe anything in RFC 3264
>or in SIP talks about handling the case where there is an issue with a
>domain name in the SDP. As such, if you use .invalid, there could be interop
>problems. The document mentions this.
>>
>>Comments or questions welcome.
>>
>>Thanks,
>>Jonathan R.
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>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
>
>--
>Jiri Kuthan            http://iptel.org/~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
>
>End of Sipping Digest
>
>_______________________________________________
>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: Tue, 05 Aug 2003 18:00:08 +0200
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
Cc: "Liess, Laura" <Laura.Liess@t-systems.com>, hgs@cs.columbia.edu,
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Thanks, I very much agree; we have received identical feedback so far.

-jiri

At 08:12 PM 7/29/2003, Jonathan Rosenberg wrote:
>I know of several carriers with RADIUS infrastructures that want to use them to support SIP. They are currently using expired I-Ds and proprietary attributes. As a result, I would strongly advocate work on standardizing RADIUS usage with SIP.
>
>There are two areas of work that I see a need for:
>
>1. Digest authentication. The sterman draft (http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-sip-00.txt) is used quite a bit, it seems. It would be nice to standardize on this.
>
>2. Prepaid calling. Many folks are using vendor proprietary radius extensiosn to support prepaid calling. It would be nice to bring those  forward and standardize on them.
>
>-Jonathan R.
>
>Bernard Aboba wrote:
>
>>Is there some particular set of draft(s) that you are advocating work on?
>>On Fri, 18 Jul 2003, Liess, Laura wrote:
>>
>>>I think most carriers currently use some kind of RADIUS platform to do user authentication and they would like to reuse it in the future to authenticate their SIP customers.
>>>My colleagues who are responsible for the RADIUS Platform of the Deutsche Telekom (CC)are currently on vacation so I could not check with them now about how to answer this mail, but we already talked about this issue a number of times and my strong opinion is that "yes, Deutsche Telekom cares a lot about SIP and RADIUS". Reusing the existing RADIUS platform for SIP authentication is a strong requirement for the development of SIP services within the Deutsche Telekom.
>>>
>>>Laura (T-Systems/Deutsche Telekom Group)
>>
>>_______________________________________________
>>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.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>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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Date: Tue, 05 Aug 2003 19:02:07 +0200
To: Bernard Aboba <aboba@internaut.com>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "Liess, Laura" <Laura.Liess@t-systems.com>, hgs@cs.columbia.edu,
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At 06:34 PM 8/5/2003, Bernard Aboba wrote:
>Everyone seems to be in agreement on the required work (am I dreaming, or
>is this really the IETF?)

:)

>There has been some discussion of potentially forming a WG to
>handle RADIUS-related work (RADIUSEXT?).  The question is whether
>SIP-specific items (such as the Sterman draft and the SIP accounting work)
>would be appropriate for a new RADIUS-focussed WG, or whether it might be
>more appropriate for a SIP-focussed group (such as SIPPING).

My esthetical process preference is to create specific extensions in the
WG which is involved with the specifics; SIPPING in this particular case.

-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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To: Bernard Aboba <aboba@internaut.com>
cc: sipping@ietf.org
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP 
In-Reply-To: Message from Jiri Kuthan <jiri@iptel.org> 
   of "Tue, 05 Aug 2003 19:02:07 +0200." <5.2.0.9.0.20030805190044.00bab8a8@mailhost.fokus.gmd.de> 
Date: Tue, 05 Aug 2003 10:19:13 -0700
From: Allison Mankin <mankin@psg.com>
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>There has been some discussion of potentially forming a WG to
>handle RADIUS-related work (RADIUSEXT?).  The question is whether
>SIP-specific items (such as the Sterman draft and the SIP accounting work)
>would be appropriate for a new RADIUS-focussed WG, or whether it might be
>more appropriate for a SIP-focussed group (such as SIPPING).

I believe  they'd be more appropriate developed in SIPPING,
with cross-review from the RADIUS group.  But I think SIPPING needs some
information on what the RADIUS working group-to-be is expected to do on RADIUS, 
in what timeframe.  From what I've understood, the charter would not primarily
be applications.  Could you give us more understanding?

Allison

_______________________________________________
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From: David Schwartz <davids@deltathree.com>
To: "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
        "'sipping@ietf.org'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] Re: Update to 3pcc
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Hi Jiri.

I was referring specifically to the call teardown issue associated with most
pre-paid billing models. 

David

-----Original Message-----
From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
Sent: Tue, August 05, 2003 7:39 PM
To: David Schwartz; 'sipping@ietf.org'
Subject: Re: [Sipping] Re: Update to 3pcc


Hello David,

I don't see how billing servers are related to method of 
call initiation.

Actually I believe that how a call is initiated (by manual
dialing a destination, by REFER-based click-to-dial, by b2bua-based
click-to-dial, CTI, etc.) is completely orthogonal to how the 
call is charged for. Regardless what charging system you use 
(if any), there will be different methods for initiating calls, 
some will be better off than others. I maintain that a click-to-dial 
b2bua is not a BCP.

-jiri

At 10:10 AM 7/30/2003, David Schwartz wrote:
>Hi Jiri
>
>REFER is good solution if you assume that either there are no billing
>issues, or alternatively, that the endpoint/s handle the billing issues. If
>this is not the case, for example, in an application where a centralized
>controller is setting up a call to two PSTN numbers, then you will need a
>centralized billing entity and REFER will not suffice.
>
>David
>
>-----Original Message-----
>From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
>Sent: Mon, July 28, 2003 7:00 PM
>To: sipping@ietf.org
>Subject: Sipping digest, Vol 1 #1144 - 2 msgs
>
>
>Send Sipping mailing list submissions to
>        sipping@ietf.org
>
>To subscribe or unsubscribe via the World Wide Web, visit
>        https://www1.ietf.org/mailman/listinfo/sipping
>or, via email, send a message with subject or body 'help' to
>        sipping-request@ietf.org
>
>You can reach the person managing the list at
>        sipping-admin@ietf.org
>
>When replying, please edit your Subject line so it is more specific
>than "Re: Contents of Sipping digest..."
>
>
>Today's Topics:
>
>   1. Re: Diversion header (Cullen Jennings)
>   2. Re: Update to 3pcc (Jiri Kuthan)
>
>--__--__--
>
>Message: 1
>Date: Sun, 27 Jul 2003 19:05:10 -0700
>Subject: Re: [Sipping] Diversion header
>From: Cullen Jennings <fluffy@cisco.com>
>To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
>
>
>I assume your question is there some way for  for the final destination
that
>receives the call to know the who the 1st callee was that had some device
in
>the network redirect the call elsewhere.
>
>The answer is YES - there has been a lot of confusion over this so think
>about my answer carefully - it will seem counter intuitive at first. The
>answer is the "To" header but it will take me a second to explain why.
>
>Imagine case 1 where A send a call to proxy B,  that sent to to C. Now C
>sent a 302 back B and B then forked the call to D. The To header in the
>message D receives will still have the To set to C. C is the element that
>sent the first 302.
>
>Now imagine a case (like your question) where there is not a recursive
>proxy. The 302 will go all the way back to A. Now A forms a new invite to
>but it can leave the To header set to the same thing it was in the original
>message (C in this case). When the message arrives at D, the To header will
>be set to C.
>
>If D was a voice mail system, it might be able to know to act on behalf of
C
>instead of some other entity.
>
>There are more complex things that relate to your request. Mary Barnes has
>been moving the history work along
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-04.txt
>
>There is some work on QSIG mappings with diversion
>http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2sip-diversion
-
>00.txt
>
>There is a draft-levy-sip-diversion which I (and many others) believe has a
>pretty much zero chance of being accepted or becoming a RFC. This work has
>been superceded by the history work. Quite a few people do not support the
>history work and it is moving very slowly. Don't expect any of this to
>resolve quickly.
>
>Give some careful thought to if the "To" header meets your needs and if it
>does use - The To header is widely implemented :-)
>
>
>Cullen
>
>
>
>
>On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
>
>> Is there any SIP header defined in any SIP RFC or Internet Draft that can
>> include the address of the 1st callee, who redirected the call using 302,
>in
>> the 2nd outgoing INVITE?
>> 
>> _______________________________________________
>> 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
>> 
>
>
>
>--__--__--
>
>Message: 2
>Date: Mon, 28 Jul 2003 10:17:51 +0200
>To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping
<sipping@ietf.org>
>From: Jiri Kuthan <jiri@iptel.org>
>Subject: Re: [Sipping] Update to 3pcc
>
>I'm actually not convinced that it is proper to show click-to-dial
>as a motivating example for use of b2bua and label it BCP.
>
>Use of REFER seems technically a superior solution worth being called
>BCP. Roots of the advantages are in a better application of e2e principle. 
>REFER takes no middle-box b2bua -- it delegates signaling to end-points.
>Advantages include:
>- All the pitfalls with media are so straigforward, they don't even 
>  need to be described (like now in sessions 8 and 9).  
>- It takes no media server to transmit ringing tone. 
>- There are no SDP compatibility issues on controller's side.
>
>-jiri
>
>At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
>>Folks,
>>
>>You may have noticed that I posted an update to the 3pcc spec:
>>
>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
>>
>>The update is based on comments received from IESG while reviewing the
>draft. In particular, there were problems with the usage of 0.0.0.0 to
>imply that no media nor RTCP should be sent. This IP address is not meant
>for that purpose. Rather, the .invalid TLD is much better suited. So, I
>changed the draft to make use of .invalid in flows where the controller
>doesnt want media to be sent.
>>
>>Now, there are some concerns with that which folks need to be aware of:
>>
>>1. It is at odds with RFC 3264, which actually recommends 0.0.0.0. In
fact,
>the problem is really with RFC 3264, NOT 3pcc. However, 3264 is done, and
>this one isnt. I logged a bug against RFC 3264 on this, and we'll need to
>work on a way to transition folks towards using .invalid and not 0.0.0.0.
>>
>>2. One the premises of 3pcc is that it should be compatible with any UA
>that is RFC 3261 compliant. If a controller uses .invalid in the SDP, I am
>not sure what a SIP endpoint would do. I don't believe anything in RFC 3264
>or in SIP talks about handling the case where there is an issue with a
>domain name in the SDP. As such, if you use .invalid, there could be
interop
>problems. The document mentions this.
>>
>>Comments or questions welcome.
>>
>>Thanks,
>>Jonathan R.
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>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
>
>--
>Jiri Kuthan            http://iptel.org/~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
>
>End of Sipping Digest
>
>_______________________________________________
>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: "Liess, Laura" <Laura.Liess@t-systems.com>
To: jiri.kuthan@fokus.fraunhofer.de
Cc: hgs@cs.columbia.edu, jdrosen@dynamicsoft.com, aboba@internaut.com,
        sipping@ietf.org, "Dumler, Alexander" <Alexander.Dumler@telekom.de>,
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Subject: AW: AW: [Sipping] FYI: RADIUS & SIP
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What about the accounting draft draft-schulzrinne-accounting-sip-01.txt? Any reason for not working on it? 
Laura 

>-----Ursprungliche Nachricht-----
>Von: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
>Gesendet: Dienstag, 5. August 2003 18:00
>An: Jonathan Rosenberg; Bernard Aboba
>Cc: Liess, Laura; hgs@cs.columbia.edu; sipping@ietf.org; Dumler,
>Alexander; Wolff, Christian
>Betreff: Re: AW: [Sipping] FYI: RADIUS & SIP
>
>
>Thanks, I very much agree; we have received identical feedback so far.
>
>-jiri
>
>At 08:12 PM 7/29/2003, Jonathan Rosenberg wrote:
>>I know of several carriers with RADIUS infrastructures that 
>want to use them to support SIP. They are currently using 
>expired I-Ds and proprietary attributes. As a result, I would 
>strongly advocate work on standardizing RADIUS usage with SIP.
>>
>>There are two areas of work that I see a need for:
>>
>>1. Digest authentication. The sterman draft 
>(http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-si
>p-00.txt) is used quite a bit, it seems. It would be nice to 
>standardize on this.
>>
>>2. Prepaid calling. Many folks are using vendor proprietary 
>radius extensiosn to support prepaid calling. It would be nice 
>to bring those  forward and standardize on them.
>>
>>-Jonathan R.
>>
>>Bernard Aboba wrote:
>>
>>>Is there some particular set of draft(s) that you are 
>advocating work on?
>>>On Fri, 18 Jul 2003, Liess, Laura wrote:
>>>
>>>>I think most carriers currently use some kind of RADIUS 
>platform to do user authentication and they would like to 
>reuse it in the future to authenticate their SIP customers.
>>>>My colleagues who are responsible for the RADIUS Platform 
>of the Deutsche Telekom (CC)are currently on vacation so I 
>could not check with them now about how to answer this mail, 
>but we already talked about this issue a number of times and 
>my strong opinion is that "yes, Deutsche Telekom cares a lot 
>about SIP and RADIUS". Reusing the existing RADIUS platform 
>for SIP authentication is a strong requirement for the 
>development of SIP services within the Deutsche Telekom.
>>>>
>>>>Laura (T-Systems/Deutsche Telekom Group)
>>>
>>>_______________________________________________
>>>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.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>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
>

_______________________________________________
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: "'Rey Jean-Francois'" <jean-francois.rey@bst.bsf.alcatel.fr>
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Subject: [Sipping] RE: Comments on draft-ietf-sipping-qsig2sip-01.txt
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Jean-Francois,

Thanks for your comments. I have dealt with most of the editorials. My
responses to your technical comments:

<SNIP>
#1 - Page 4 - Introduction

The document states :
"This document specifies signalling interworking for basic services 
that provide a bi-directional transfer capability for speech, DTMF, 
facsimile and modem media"

There are different drafts and RFCs dealing with DTMF, facsimile and modem
media. It's not really explained further down in the text. What about
clarifying
it or removing it ?
[JRE]Good point, because there are other aspects of DTMF, fax and modem that
are deliberately outside the scope of this specification. On the other hand
the same is true of speech, e.g., the use of RTP is not covered in this
document. I propose to add "Other aspects of interworking, e.g., the use of
RTP and SDP, will differ according to the type of media concerned and are
outside the scope of this specification."

#2 - Page 10 - 7 General requirements

"In addition, the gateway SHALL support SIP TU behaviour for a UA in
accordance
with [10] except where stated otherwise in this specification."

Where are the parts of the document where the behaviour is different ? If
there's no difference, the "except..." part of the sentence should be
removed.
[JRE] I believe all the requirements of sections 8 to 10 apply at the TU
level. Will clarify.

#3 - Page 10 -  7 General requirements
"SIP methods not defined in [10], [11], [13] or [14] are outside the"

Do [13] and [14] define SIP methods ?
[JRE] No - will clarify.

#4 - Page 10 -  7 General requirements
"As a result of DNS look-up by the gateway in order to determine where 
to send a SIP INVITE request, a number of candidate destinations can 
be attempted in sequence. The way in which this is handled by the 
gateway is outside the scope of this specification. However, any 
behaviour specified in this document on receipt of a SIP final 
response SHOULD apply only when there are no more candidate 
destinations to try."

Does it mean that the ingress gateway SHOULD try the next retrieved
addresses if
the first one responded 484 for instance ? I guess that rfc3263 narrows the
scope to "network" or "component" failures.
[JRE] I guess when overlap signalling applies in the SIP network the
statement is not quite true, so I will make an exception. Also I believe the
statement is not applicable to 3xx or 6xx responses, so I will limit it to
4xx and 5xx responses. Does this cover all the situations you had in mind?

#5 - Page 13 - 8.2.1.1 Receipt of QSIG SETUP message
"If information in the QSIG SETUP message is unsuitable for generating 
any of the mandatory fields in a SIP INVITE request (e.g., if a 
Request-URI cannot be derived from the QSIG Called party number 
information element)"

I agree in theory, but what could prevent from deriving a R-URI in the real
world ?
[JRE] It depends what method the gateway uses for mapping a number to a
R-URI. It might do it via look-up (e.g., ENUM), in which case if there is no
entry for the number concerned this might apply. Obviously it would not
apply to a gateway that algorithmically converts any number to a URI.

#6 - Page 14 - 8.2.1.3 Receipt of SIP 18x provisional response
"NOTE 2. Media streams are established as a result of receiving SDP 
answer information in a reliable provisional response and can be "

I agree that reliable provisional response are better, but why use reliable
here
? If a UA that does not support PRACK sends a 183 with a SDP body, the media
stream is established too.
[JRE] You are correct. However, the text concerned talks about SDP answer
but media streams can also be established at this time by means of SDP offer
in a provisional response. The text fails to mention this. According to
RFC3261 section 13.2.1 an unreliable provisional response cannot be used for
SDP offer. I will clarify the statement. Also subsequent modification of the
session by means of UPDATE applies only after a offer has been received in a
reliable message and answer has been sent, so this too would need
qualifying. However, it is not really necessary to mention UPDATE so perhaps
that part can be deleted.

#7 - Page 15 - 8.2.1.4 Receipt of SIP 2xx response
"If the gateway receives a SIP 2xx response other than 200 (OK), the 
gateway SHALL send a SIP ACK request."

What happens on the QSIG side ?
[JRE] The intention was to take no action. Will clarify.

#8 - Page 23 - 8.3.9
"If a gateway receives a SIP INVITE request with the same Call-ID as 
        an existing call for which the QSIG state is overlap sending and
with 
        updated Request-URI and To fields from which a called party number 
        with a superset of digits can be derived, it SHALL"
I think that it should read "same Call-ID and the same From header field
including the tag, and no tag in the To header field". 
BTW, the To header field is not so important anyway, since it's not used
(9.2.1).

Same remark for 7 of figure 5 and for 8 of figure 8.
[JRE]I agree. Will clarify.

#9 - Page 24 - 8.3.9
"If a gateway receives a SIP INVITE request for a new dialog with the 
        same Call-ID as an existing SIP INVITE request for which the gateway

        has not yet sent a final response and failing to meet the other 
        conditions above concerning overlap sending, the gateway SHALL 
        respond to the new request with a SIP 485 (Ambiguous) response."
It's not very important, but could you explain the motivation of this
sentence.
Is it related to the merged requests ?
[JRE] Its purpose is to prescribe behaviour if the gateway receives an
INVITE request that in all respects looks like overlap sending but fails to
meet the criterion of the derived called party number being a superset of
the previously derived called party number. I will change the wording
slightly to complement the changes I made to address comment #8, so see if
that helps. Concerning how it arises, I guess merged requests are one
possibility.

#10 - Page 24 - 8.4.1 
"as specified in table 1."
Why aren't some status codes symetric in table 1 and table 2. For instance 
- 408 Request timeout maps 102 Recovery on timer expiry and 
- 102 Recovery on timer expiry maps 504 Server time-out.
BTW, I like the way table 2 is similar to what RFC3398 describes.
[JRE] 504 also maps to 102, but it is not possible to make 102 map to both
408 and 504 - we had to pick one. This illustrates the point that there is
not a 1:1 correspondence between SIP responses and Q.850 causes and
therefore there will be some asymmetries and some rather questionable
mappings. Cause mapping was discussed long and hard for ISUP interworking
and again for QSIG interworking (resulting in some changes to ISUP
interworking to keep alignment). It is not perfect and it never will be, but
I would be reluctant to open up the discussion again. I hope you can live
with what we have.

#11 - Page 24- 8.4.1
Very minor comment. It's obvious that the 200 responses are related to the
INVITE transaction, but this is not written.
[JRE] Will clarify.

#12 - Page 27 - 8.4.4
What is the meaning of () in table 2 
     403 Forbidden                       21 (Call rejected) 
     404 Not found                       1 (Unallocated number) 
[JRE] Deleted.

--
And some remarks regarding other SIPPING drafts

#1 draft-ietf-sipping-overlap-04.txt
It says :
   "However, overlap signalling adds a requirement to this process. 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."

Is this requirement applicable here ?
[JRE] I remember this was discussed and we thought it best not to try to
give guidance in this tricky situation.

#2  draft-ietf-sipping-pstn-call-flows-02.txt 2.2, 4.2

I know that qsig2sip deals with QSIG only, but don't you think that people
might
get confused by the slight difference in the mapping of the messages of the
"ISDN PBX" examples of the call-flow draft ?
[JRE] There are of course some differences between QSIG and Q.931 that can
have impact on the mapping. Also I am not sure of the status of the
call-flows draft and in particular the two examples showing Q.931 mapping,
because I thought the group had decided not to work on SIP-Q.931 mapping.

</SNIP>

Regards,

John (john.elwell@siemens.com)

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From: "Elwell, John" <john.elwell@siemens.com>
To: "'Francois Audet'" <audet@nortelnetworks.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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Francois,
 
Thanks for your comments. I will address the editorials. Concerning the
technical comments:
 
<SNIP>
- Section 8.3.1, page 19 
  
  5th paragraph: Add ", unless the gateway is aware that in-band information

  will not be used" to the end of the paragraph. The reason for this is that

  there are cases where the gateway may have a priori-knowledge that in-band

  tones and announcements are not used; and out-of-band ALERTING or 
  fast connection (i.e., no PROGRESS and no ALERTING) is always 
  used for example. In that case, there is no reason to reject the call with

  488 if the SDP was not included in the INVITE. 
[JRE] I agree that if there is a priori knowledge that in-band tones and
announcements will not be used, then what you propose is reasonable. I am
rather doubtful that the gateway will have sufficient knowledge to determine
this. The 4th and 5th paragraphs of 8.3.3 mandate the connection of media
streams if relevant QSIG progress indicators are present in a PROGRESS or
ALERTING message. The presence of these progress indicators is outside the
control of the gateway. So how can the gateway have a priori knowledge that
these progress indicators will not be received?




- Section 8.3.5, page 21 

  This section mandates the inclusion of SDP in 18x response. This may not
be 
  possible, or even required. For example, it is very common for PBXs not to

  know at the time of rigning where a call will be terminated (i.e., which 
  user): this applies to features such as "hunt group", as well as "multiple

  line appearances". In those cases, it may be impossible for the gateway 
  to include an SDP in the 18x, because it does not know yet whitch media 
  gateway resource will be used. Furthermore, it is against the spirit of 
  the offer-answer model and SIP to mandate that the SDP must be included 
  in the 18x response (it could be in 200 OK for example). 

  Paragraphs 3 and 4 shall thus be modified as follows: "...SDP answer 
  information SHOULD be included in the SIP 18x provisional response if 
  the SDP information is available." 

[JRE] This clause was primarily intended to cover the case where an in-band
tone or announcement is to be sent. I propose changing it to say the rules
SHALL apply if an in-band  tone or announcement is available and MAY apply
otherwise.

</SNIP> 
 Regards,

John (john.elwell@siemens.com <mailto:john.elwell@siemens.com> )

 


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<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=978410313-05082003>Francois,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=978410313-05082003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=978410313-05082003>Thanks 
for your comments. I will address the editorials. Concerning the technical 
comments:</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=978410313-05082003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN 
class=978410313-05082003>&lt;SNIP&gt;</SPAN></FONT></DIV>
<DIV><FONT size=2>- Section 8.3.1, page 19</FONT> <BR><FONT size=2>&nbsp; 
</FONT><BR><FONT size=2>&nbsp; 5th paragraph: Add ", unless the gateway is aware 
that in-band information</FONT> <BR><FONT size=2>&nbsp; will not be used" to the 
end of the paragraph. The reason for this is that</FONT> <BR><FONT size=2>&nbsp; 
there are cases where the gateway may have a priori-knowledge that 
in-band</FONT> <BR><FONT size=2>&nbsp; tones and announcements are not used; and 
out-of-band ALERTING or </FONT><BR><FONT size=2>&nbsp; fast connection (i.e., no 
PROGRESS and no ALERTING) is always</FONT> <BR><FONT size=2>&nbsp; used for 
example. In that case, there is no reason to reject the call with 
</FONT><BR><FONT size=2>&nbsp; 488 if the SDP was not included in the 
INVITE.</FONT> <BR><SPAN class=978410313-05082003><FONT face=Arial color=#0000ff 
size=2>[JRE]&nbsp;I agree that if there is a priori knowledge that in-band tones 
and announcements will not be used, then&nbsp;what you&nbsp;propose is 
reasonable. I am rather doubtful that the gateway will have sufficient knowledge 
to determine this. The 4th and 5th paragraphs of 8.3.3 mandate&nbsp;the 
connection of media streams&nbsp;if relevant QSIG progress indicators&nbsp;are 
present in a PROGRESS or ALERTING message. The presence of these progress 
indicators is outside the control of the gateway. So how can the gateway have a 
priori knowledge that these progress indicators will not be 
received?</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px"><FONT face=Arial color=#0000ff 
  size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial 
  color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT 
  face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff 
  size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT face=Arial 
  color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff size=2></FONT><FONT 
  face=Arial color=#0000ff size=2></FONT><FONT face=Arial color=#0000ff 
  size=2></FONT><BR>
  <P><FONT size=2>- Section 8.3.5, page 21</FONT> </P>
  <P><FONT size=2>&nbsp; This section mandates the inclusion of SDP in 18x 
  response. This may not be</FONT> <BR><FONT size=2>&nbsp; possible, or even 
  required. For example, it is very common for PBXs not to</FONT> <BR><FONT 
  size=2>&nbsp; know at the time of rigning where a call will be terminated 
  (i.e., which </FONT><BR><FONT size=2>&nbsp; user): this applies to features 
  such as "hunt group", as well as "multiple </FONT><BR><FONT size=2>&nbsp; line 
  appearances". In those cases, it may be impossible for the gateway 
  </FONT><BR><FONT size=2>&nbsp; to include an SDP in the 18x, because it does 
  not know yet whitch media</FONT> <BR><FONT size=2>&nbsp; gateway resource will 
  be used. Furthermore, it is against the spirit of</FONT> <BR><FONT 
  size=2>&nbsp; the offer-answer model and SIP to mandate that the SDP must be 
  included</FONT> <BR><FONT size=2>&nbsp; in the 18x response (it could be in 
  200 OK for example). </FONT></P>
  <P dir=ltr><FONT size=2>&nbsp; Paragraphs 3 and 4 shall thus be modified as 
  follows: "...SDP answer </FONT><BR><FONT size=2>&nbsp; information SHOULD be 
  included in the SIP 18x provisional response if</FONT> <BR><FONT size=2>&nbsp; 
  the SDP information is available."</FONT> </P></BLOCKQUOTE>
<P dir=ltr><SPAN class=978410313-05082003></SPAN><SPAN 
class=978410313-05082003><FONT face=Arial color=#0000ff size=2><SPAN 
class=978410313-05082003><FONT face=Arial color=#0000ff size=2>[JRE]&nbsp;This 
clause was primarily intended to cover the case where an in-band tone or 
announcement is to be sent. I&nbsp;propose changing it to say the rules SHALL 
apply if an in-band &nbsp;tone or announcement is&nbsp;available and MAY apply 
otherwise.</FONT></SPAN></FONT></SPAN></P>
<P dir=ltr><SPAN class=978410313-05082003><FONT face=Arial color=#0000ff 
size=2>&lt;/SNIP&gt;&nbsp;</FONT></SPAN><BR><FONT size=2>&nbsp;<SPAN 
class=978410313-05082003><FONT face=Arial>Regards,</FONT></SPAN></FONT></P>
<P dir=ltr><FONT face=Arial size=2><SPAN class=978410313-05082003>John (<A 
href="mailto:john.elwell@siemens.com">john.elwell@siemens.com</A>)</SPAN></FONT></P>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <P><FONT face=Arial color=#0000ff 
size=2></FONT>&nbsp;</P></BLOCKQUOTE></BODY></HTML>

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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] WGLC for Q.SIP-SIP mapping draft
To: "'Stach Thomas'" <thomas.stach@siemens.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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Thomas,
 
Thanks for your comments. I have taken them all into account except for the
following.
 
<SNIP>
In Section 8.1 4th para is stated: 
"The processing of any valid QSIG message that does not satisfy any of 
these conditions is outside the scope of this specification." 
This seems to be a little bit vague for somebody not familiar with QSIG. 
Since the interworking only comprises QSIG Basic Call, you could be more
specific and mention the unconsidered features provided by ISO 11582 by
adding:

"This includes a QSIG SETUP message intended to establish a call-independent
signalling connection and  connectionless transport as specified in [3]."
[JRE] The statements were not really intended to exclude QSIG features
beyond basic call, but were intended to cover QSIG messages that are not
suitable for mapping to SIP messages because they do not relate to an
existing SIP call or, in the case of a QSIG SETUP message, the call is not
suitable for routing to SIP. I believe the existing text is adequate for
this purpose. There is already a statement in section 1 that QSIG signalling
in support of supplementary services is outside the scope. However, I could
stress this with a separate statement in 8.1 that call-independent
signalling connections and connectionless transport are outside the scope.
Do you think this would be helpful?

The QSIG SETUP message may optionally transport HLC and LLC Information
elements which are not mentioned in the draft. You might want to add
something like into section 10:

"The gateway SHOULD (or SHALL(?)) not include a Low layer compatibility in
the QSIG SETUP message. The gateway SHOULD not include a High layer
compatibility information element in the QSIG SETUP message.

NOTE: Inclusion of a HLC in the QSIG SETUP message might be useful, if the
gateway knows prior to sending the QSIG SETUP message that fax is being
transmitted (e.g., because of a corresponding SDP offer), it may include in
the QSIG SETUP message a High layer compatibility information element
indicating group 3 fax."
[JRE] I will add something. In addition I will add something concerning the
opposite direction, allowing HLC and LLC to be taken into account when
deriving SDP offer information.

</SNIP>
 
Regards,
 
John (john.elwell@siemens.com <mailto:john.elwell@siemens.com> )

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<DIV><SPAN class=406110010-05082003><FONT face=Arial><FONT color=#0000ff><FONT 
size=2>Thomas,</FONT></FONT></FONT></SPAN></DIV>
<DIV><SPAN class=406110010-05082003><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=406110010-05082003><FONT face=Arial color=#0000ff size=2>Thanks 
for your comments. I have taken them all into account except for the 
following.</FONT></SPAN></DIV>
<DIV><FONT face=Arial color=#0000ff size=2></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003>&lt;SNIP&gt;</SPAN></SPAN></FONT></DIV>
<DIV><FONT size=2>In Section 8.1 4th para is stated:</FONT> <BR><FONT 
size=2>"The processing of any valid QSIG message that does not satisfy any of 
</FONT><BR><FONT size=2>these conditions is outside the scope of this 
specification."</FONT> <BR><FONT size=2>This seems to be a little bit vague for 
somebody not familiar with QSIG. </FONT><BR><FONT size=2>Since the interworking 
only comprises QSIG Basic Call, you could be more specific and mention the 
unconsidered features provided by ISO 11582 by adding:</FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <P><FONT size=2>"This includes a QSIG SETUP message intended to establish a 
  call-independent signalling connection and&nbsp; connectionless transport as 
  specified in [3]."<BR><SPAN class=406110010-05082003><FONT face=Arial 
  color=#0000ff>[JRE]&nbsp;The statements were not really intended to exclude 
  QSIG features beyond basic call, but were intended to cover&nbsp;QSIG messages 
  that are not suitable for mapping to SIP messages because they do not relate 
  to an existing SIP call or, in the case of a QSIG SETUP message, the call is 
  not suitable for routing to SIP.&nbsp;I believe the existing text is adequate 
  for this purpose. There is already a statement in section 1 that QSIG 
  signalling in support of supplementary services is outside the scope. However, 
  I could stress this with a separate statement in 8.1 that call-independent 
  signalling connections and connectionless transport are outside the scope. Do 
  you think this would be helpful?</FONT></SPAN></FONT></P>
  <P><FONT size=2>The QSIG SETUP message may optionally transport HLC and LLC 
  Information elements which are not mentioned in the draft. You might want to 
  add something like into section 10:</FONT></P>
  <P><FONT size=2>"The gateway SHOULD (or SHALL(?)) not include a Low layer 
  compatibility in the QSIG SETUP message. The gateway SHOULD not include a High 
  layer compatibility information element in the QSIG SETUP message.</FONT></P>
  <P><FONT size=2>NOTE: Inclusion of a HLC in the QSIG SETUP message might be 
  useful, if the gateway knows prior to sending the QSIG SETUP message that fax 
  is being transmitted (e.g., because of a corresponding SDP offer), it may 
  include in the QSIG SETUP message a High layer compatibility information 
  element indicating group 3 fax."<BR><SPAN class=406110010-05082003><FONT 
  face=Arial color=#0000ff>[JRE]&nbsp;I will add something. In addition 
  I&nbsp;will add something concerning the opposite direction, allowing HLC and 
  LLC to be taken into account&nbsp;when deriving SDP offer 
  information.</FONT></SPAN></FONT></P></BLOCKQUOTE><FONT face=Arial color=#0000ff 
size=2><SPAN class=406110010-05082003>
<DIV><FONT face=Arial size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003>&lt;/SNIP&gt;</SPAN></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003></SPAN></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#000000 size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003>Regards,</SPAN></SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#000000 size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003></SPAN></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial color=#000000 size=2><SPAN class=406110010-05082003><SPAN 
class=497211611-05082003>John (<A 
href="mailto:john.elwell@siemens.com">john.elwell@siemens.com</A>)</SPAN></SPAN></FONT></SPAN></FONT></DIV></BODY></HTML>

--Boundary_(ID_aRr8797UZpm4UUyh/ntmyg)--

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Subject: [Sipping] RE: WGLC for draft-ietf-sipping-qsig2sip-01
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Thanks John, see below:

-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com] 
Sent: Wednesday, August 06, 2003 04:48
To: Audet, Francois [SC100:4K02:EXCH]; 'sipping@ietf.org'
Subject: RE: WGLC for draft-ietf-sipping-qsig2sip-01


Francois,
 
Thanks for your comments. I will address the editorials. Concerning the
technical comments:
 
<SNIP>
- Section 8.3.1, page 19 
  
  5th paragraph: Add ", unless the gateway is aware that in-band information

  will not be used" to the end of the paragraph. The reason for this is that

  there are cases where the gateway may have a priori-knowledge that in-band

  tones and announcements are not used; and out-of-band ALERTING or 
  fast connection (i.e., no PROGRESS and no ALERTING) is always 
  used for example. In that case, there is no reason to reject the call with

  488 if the SDP was not included in the INVITE. 
[JRE] I agree that if there is a priori knowledge that in-band tones and
announcements will not be used, then what you propose is reasonable. I am
rather doubtful that the gateway will have sufficient knowledge to determine
this. The 4th and 5th paragraphs of 8.3.3 mandate the connection of media
streams if relevant QSIG progress indicators are present in a PROGRESS or
ALERTING message. The presence of these progress indicators is outside the
control of the gateway. So how can the gateway have a priori knowledge that
these progress indicators will not be received?
 

I'm talking about the gateway RECEIVING the SETUP, not the one sending it.
The case I am thinking of is then the Gateway is an IP PBX (with SIP on one
side, QSIG on the other). If the call terminates on a local phone, it may
use out-of-band ringing (our PBXs work that way). 
 
If we want to be picky about the meaning of  Gateway, and dissociate the
gateway from the actual QSIG PBX, then it can still happen. The gateway may
decide to map the INVITE to SETUP anyways and forward it to the QSIG PBX. If
the PBX responds with the ALERTING indicating out-of-band, everything is
fine (which is why I would like this change). If on the other hand, the
ALERTING indicates in-band, then the gateway could decide to do as per your
procedure and send the 488. But I wouldn't want to mandate 488 everytime the
SDP is not in the INVITE, especially since it is a very normal occurence in
SIP.
 
Even for the second case, where the call ultimately terminates on a line
with an in-band signalling, I am not convinced at all that the qsig2sip
draft should mandate that the call be terminated. To me this should be a
design option. An implementer may decide that it prefers to go through with
the call, and provide some sort of indication to the user that the call is
"progressing" instead of failing the call (I expect that a proper SIP
implemenation would do something like that).
 
I would prefer that the paragraph does not mandate the 488 as other options
are doable, and probably much less drastic than call failure (as an
out-of-band progress indication would be).
 

- Section 8.3.5, page 21 

  This section mandates the inclusion of SDP in 18x response. This may not
be 
  possible, or even required. For example, it is very common for PBXs not to

  know at the time of rigning where a call will be terminated (i.e., which 
  user): this applies to features such as "hunt group", as well as "multiple

  line appearances". In those cases, it may be impossible for the gateway 
  to include an SDP in the 18x, because it does not know yet whitch media 
  gateway resource will be used. Furthermore, it is against the spirit of 
  the offer-answer model and SIP to mandate that the SDP must be included 
  in the 18x response (it could be in 200 OK for example). 

  Paragraphs 3 and 4 shall thus be modified as follows: "...SDP answer 
  information SHOULD be included in the SIP 18x provisional response if 
  the SDP information is available." 

[JRE] This clause was primarily intended to cover the case where an in-band
tone or announcement is to be sent. I propose changing it to say the rules
SHALL apply if an in-band  tone or announcement is available and MAY apply
otherwise. 

I'm OK with your proposed text. 


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<DIV><SPAN class=915350116-06082003><FONT face=Arial color=#800000 size=2>Thanks 
John, see below:</FONT></SPAN></DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT 
  face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> Elwell, John 
  [mailto:john.elwell@siemens.com] <BR><B>Sent:</B> Wednesday, August 06, 2003 
  04:48<BR><B>To:</B> Audet, Francois [SC100:4K02:EXCH]; 
  'sipping@ietf.org'<BR><B>Subject:</B> RE: WGLC for 
  draft-ietf-sipping-qsig2sip-01<BR><BR></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=978410313-05082003>Francois,</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=978410313-05082003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=978410313-05082003>Thanks for your comments. I will address the 
  editorials. Concerning the technical comments:</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=978410313-05082003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=978410313-05082003>&lt;SNIP&gt;</SPAN></FONT></DIV>
  <DIV><FONT size=2>- Section 8.3.1, page 19</FONT> <BR><FONT size=2>&nbsp; 
  </FONT><BR><FONT size=2>&nbsp; 5th paragraph: Add ", unless the gateway is 
  aware that in-band information</FONT> <BR><FONT size=2>&nbsp; will not be 
  used" to the end of the paragraph. The reason for this is that</FONT> 
  <BR><FONT size=2>&nbsp; there are cases where the gateway may have a 
  priori-knowledge that in-band</FONT> <BR><FONT size=2>&nbsp; tones and 
  announcements are not used; and out-of-band ALERTING or </FONT><BR><FONT 
  size=2>&nbsp; fast connection (i.e., no PROGRESS and no ALERTING) is 
  always</FONT> <BR><FONT size=2>&nbsp; used for example. In that case, there is 
  no reason to reject the call with </FONT><BR><FONT size=2>&nbsp; 488 if the 
  SDP was not included in the INVITE.</FONT> <BR><SPAN 
  class=978410313-05082003><FONT face=Arial color=#0000ff size=2>[JRE]&nbsp;I 
  agree that if there is a priori knowledge that in-band tones and announcements 
  will not be used, then&nbsp;what you&nbsp;propose is reasonable. I am rather 
  doubtful that the gateway will have sufficient knowledge to determine this. 
  The 4th and 5th paragraphs of 8.3.3 mandate&nbsp;the connection of media 
  streams&nbsp;if relevant QSIG progress indicators&nbsp;are present in a 
  PROGRESS or ALERTING message. The presence of these progress indicators is 
  outside the control of the gateway. So how can the gateway have a priori 
  knowledge that these progress indicators will not be 
  received?</FONT></SPAN><BR><SPAN class=915350116-06082003><FONT face=Arial 
  color=#800000 size=2>&nbsp;</FONT></SPAN></DIV></BLOCKQUOTE>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2>I'm talking about the gateway RECEIVING the SETUP, not the one sending 
it. The case I am thinking&nbsp;of is then the Gateway is an IP PBX (with SIP on 
one side, QSIG on the other).&nbsp;If the call terminates on a local phone, it 
may use out-of-band ringing (our PBXs work that way). </FONT></SPAN></DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2>If we want to be picky about the meaning of&nbsp; Gateway, and dissociate 
the gateway from the actual QSIG PBX, then it can still happen. The gateway may 
decide to map the INVITE to SETUP anyways and forward it to the QSIG PBX. If the 
PBX responds with the ALERTING indicating out-of-band, everything is fine (which 
is why&nbsp;I would like this change). If on the other hand, the ALERTING 
indicates in-band, then the gateway could decide to do as per your procedure and 
send the 488. But I wouldn't want to mandate 488 everytime the SDP is not in the 
INVITE, especially since it is a very normal occurence in 
SIP.</FONT></SPAN></DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2>Even for the second&nbsp;case, where the call ultimately terminates on a 
line with an in-band signalling, I am not convinced at all that the qsig2sip 
draft should mandate that the call be terminated. To me this should be a design 
option. An implementer may decide that it prefers to go through with the call, 
and provide some sort of indication to the user that the call is "progressing" 
instead of failing the call (I expect that a proper SIP implemenation would do 
something like that).</FONT></SPAN></DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2>I would prefer that the paragraph does not mandate the 488 as other 
options are doable, and probably much less drastic than call failure (as an 
out-of-band progress indication would be).</FONT></SPAN></DIV>
<DIV dir=ltr><SPAN class=915350116-06082003><FONT face=Arial color=#800000 
size=2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #800000 2px solid; MARGIN-RIGHT: 0px">
  <BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
    <P><FONT size=2>- Section 8.3.5, page 21</FONT> </P>
    <P><FONT size=2>&nbsp; This section mandates the inclusion of SDP in 18x 
    response. This may not be</FONT> <BR><FONT size=2>&nbsp; possible, or even 
    required. For example, it is very common for PBXs not to</FONT> <BR><FONT 
    size=2>&nbsp; know at the time of rigning where a call will be terminated 
    (i.e., which </FONT><BR><FONT size=2>&nbsp; user): this applies to features 
    such as "hunt group", as well as "multiple </FONT><BR><FONT size=2>&nbsp; 
    line appearances". In those cases, it may be impossible for the gateway 
    </FONT><BR><FONT size=2>&nbsp; to include an SDP in the 18x, because it does 
    not know yet whitch media</FONT> <BR><FONT size=2>&nbsp; gateway resource 
    will be used. Furthermore, it is against the spirit of</FONT> <BR><FONT 
    size=2>&nbsp; the offer-answer model and SIP to mandate that the SDP must be 
    included</FONT> <BR><FONT size=2>&nbsp; in the 18x response (it could be in 
    200 OK for example). </FONT></P>
    <P dir=ltr><FONT size=2>&nbsp; Paragraphs 3 and 4 shall thus be modified as 
    follows: "...SDP answer </FONT><BR><FONT size=2>&nbsp; information SHOULD be 
    included in the SIP 18x provisional response if</FONT> <BR><FONT 
    size=2>&nbsp; the SDP information is available."</FONT> </P></BLOCKQUOTE>
  <P dir=ltr><SPAN class=978410313-05082003></SPAN><SPAN 
  class=978410313-05082003><FONT color=#0000ff><SPAN 
  class=978410313-05082003><FONT color=#0000ff><FONT face=Arial><FONT 
  size=2>[JRE]&nbsp;This clause was primarily intended to cover the case where 
  an in-band tone or announcement is to be sent. I&nbsp;propose changing it to 
  say the rules SHALL apply if an in-band &nbsp;tone or announcement 
  is&nbsp;available and MAY apply otherwise.<SPAN class=915350116-06082003><FONT 
  color=#800000>&nbsp;</FONT></SPAN></FONT></FONT></FONT></SPAN></FONT></SPAN></P></BLOCKQUOTE>
<P dir=ltr><SPAN class=978410313-05082003><FONT color=#0000ff><SPAN 
class=978410313-05082003><FONT color=#0000ff><FONT face=Arial><FONT size=2><SPAN 
class=915350116-06082003><FONT color=#800000>I'm OK with your proposed 
text.</FONT>&nbsp;</SPAN></FONT></FONT></FONT></SPAN></FONT></SPAN></P></BODY></HTML>

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thanks for your review and comments! nits will be reflected in the
next revision. responses to questions are inline below.

cheers,
timm

Chris Boulton wrote:

 > - Would any benefit be gained from including an optional =91order=92
 > attribute on primitive elements so that when =91pipe=92 is used, the o=
rder
 > is determined explicitly rather than following a logical path?
 >
this question is related to MOML. the goal was to provide several simple
topology schemas to cover the common use cases. i think that explicit
control of topology (i.e. as in mpcp) may be useful. i'm not sure that
explicit ordering in this case would have benefit and may introduce
additional error cases. did you have any specific benefits in mind?

 > - Was there a reason that JOIN/UNJOIN were limited to id1 + id2.  Coul=
d
 > it not have just been id, that was forced to appear at least twice?
 > This would allow you to join >2 media streams in one transaction.
 >
simplicity. msml already allows multiple <join> elements to be present
in a single transaction. allowing multiple streams to be identified in
a single join introduces more complexity and makes failure handling
more difficult.

 >
 > - <result> element should say that the response code corresponds to th=
e
 > SIP response code received for the transaction.
 >
i'm not quite sure what you mean here. the response attribute in <result>
is for the msml transaction. it just happens to be carried in the respons=
e
to the sip message which conveyed the msml transaction. however msml is
designed to be neutral to the transport mechanism and this correlation
may not exist in all cases.

 > from MSML: Media objects may be defined using MOML or other suitable m=
edia
 >    resource description languages. A media server which supports
 >    <insert> and <remove> must support MOML and may support other
 >    languages.
 >
 > Should this be a =91MUST=92 or is it not possible to only use another
 > =91suitable media resource description language=92 with MSML?
 >
a media resource description language is required. the "must" requires
that there be at least one common language to insure interoperability.
it does not preclude that other languages are supported.





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thanks again for your feedback. answers to questions below.

cheers,
timm

>=20
> - Is there an alternative method for defining unlimited occurrences=20
> using the =91max=92 attribute =3D 0 (e.g. pattern element uses max with=
in DTMF=20
> primitive).  This seemed a little confusing.
>=20
i've had this comment from others. would extending the datatype
to include an explicit "unlimited" value be less confusing and/or
renaming "max" to "maxoccurs"?

>=20
> - 4.5.1.1 The description of <tvoice> and <tsilence> needs to be made=20
> more explicit.
>=20
will do.



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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: AW: [Sipping] FYI: RADIUS & SIP
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Hi All,

I just want to plug a draft we are working on in the AAA WG:

 Diameter Credit-control Application
 draft-ietf-aaa-diameter-cc-01.txt

This is primarily Diameter based, though interworking with
RADIUS is somewhat covered.  This is losely based on=20
some 3GPP solutions, but we are trying to generalize it and
make it more useful for the Internet. I do know that there are also
some individual drafts covering RADIUS prepaid. I don't have the time,=20
nor energy to specify RADIUS prepaid work. =20

One of the largest problems that I currently see about RADIUS
is the under-specified nature of RADIUS.  I understand that
many different organizations and vendors have defined, but
not documented, vendor-specific VSAs.  In trying to do any
meaningful work with RADIUS, I think we need to document
some of this first.

Oh, a note to Henry, I have heard some horror stories from
various folks (and talking to different IT support people)
of how large-scale RADIUS deployments react after a power-
outtage.  Since backoff is not specified in RADIUS, if
multiple NASes comeup at the same time, they can flood
the network, causing all sorts of fun things happening ...
This is one area in which RADIUS would need fixing.

br,
John


> -----Original Message-----
> From: ext Liess, Laura [mailto:Laura.Liess@t-systems.com]
> Sent: 31 July, 2003 17:41
> To: jdrosen@dynamicsoft.com; aboba@internaut.com
> Cc: hgs@cs.columbia.edu; sipping@ietf.org; Dumler, Alexander; Wolff,
> Christian
> Subject: AW: AW: [Sipping] FYI: RADIUS & SIP
>=20
>=20
> I agree with Jonathan. Support for prepaid calling seems to=20
> me a very good idea. =20
> Laura
>=20
> Laura Liess
> T-Systems/Deutsche Telekom Group
> Am Kavalleriesand 3
> 64295 Darmstadt
>=20
> >-----Ursprungliche Nachricht-----
> >Von: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> >Gesendet: Dienstag, 29. Juli 2003 20:13
> >An: Bernard Aboba
> >Cc: Liess, Laura; hgs@cs.columbia.edu; sipping@ietf.org; Dumler,
> >Alexander; Wolff, Christian
> >Betreff: Re: AW: [Sipping] FYI: RADIUS & SIP
> >
> >
> >I know of several carriers with RADIUS infrastructures that want to=20
> >use them to support SIP. They are currently using expired I-Ds and=20
> >proprietary attributes. As a result, I would strongly=20
> advocate work on=20
> >standardizing RADIUS usage with SIP.
> >
> >There are two areas of work that I see a need for:
> >
> >1. Digest authentication. The sterman draft=20
> >(http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-si
> >p-00.txt)=20
> >is used quite a bit, it seems. It would be nice to=20
> standardize on this.
> >
> >2. Prepaid calling. Many folks are using vendor proprietary radius=20
> >extensiosn to support prepaid calling. It would be nice to=20
> bring those=20
> >  forward and standardize on them.
> >
> >-Jonathan R.
> >
> >Bernard Aboba wrote:
> >
> >> Is there some particular set of draft(s) that you are=20
> >advocating work on?
> >>=20
> >> On Fri, 18 Jul 2003, Liess, Laura wrote:
> >>=20
> >>=20
> >>>I think most carriers currently use some kind of RADIUS=20
> >platform to do user authentication and they would like to=20
> >reuse it in the future to authenticate their SIP customers.
> >>>My colleagues who are responsible for the RADIUS Platform of=20
> >the Deutsche Telekom (CC)are currently on vacation so I could=20
> >not check with them now about how to answer this mail, but we=20
> >already talked about this issue a number of times and my=20
> >strong opinion is that "yes, Deutsche Telekom cares a lot=20
> >about SIP and RADIUS". Reusing the existing RADIUS platform=20
> >for SIP authentication is a strong requirement for the=20
> >development of SIP services within the Deutsche Telekom.
> >>>
> >>>Laura (T-Systems/Deutsche Telekom Group)
> >>=20
> >>=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
>
>--=20
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>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

_______________________________________________
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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Date: Thu, 07 Aug 2003 07:08:04 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
In-reply-to: <DADF50F5EC506B41A0F375ABEB32063658F247@esebe023.ntc.nokia.com>
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        jdrosen@dynamicsoft.com, aboba@internaut.com
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> Oh, a note to Henry, I have heard some horror stories from
> various folks (and talking to different IT support people)
> of how large-scale RADIUS deployments react after a power-
> outtage.  

Since we know of no significant DIAMETER deployments at ISPs (except in the
minds of its IETF champions), sure, there are no horror stories possible for
DIAMETER :-)

The SIPPING WG charter is to look out for what is good for SIP applications.
Promoting DIAMATER should be done elsewhere. The fact that SIPPING folks are
too busy to work on _real_ SIP-AAA should is not a good reason to accept the
DIAMETER pitch at face value. 

One option is to postpone SIP-AAA work until we have some I-D's based on
real deployments. Or does anyone come forward?

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> john.loughney@nokia.com
> Sent: Wednesday, August 06, 2003 9:30 PM
> To: Laura.Liess@t-systems.com; jdrosen@dynamicsoft.com;
> aboba@internaut.com
> Cc: hgs@cs.columbia.edu; sipping@ietf.org; Alexander.Dumler@telekom.de;
> Christian.Wolff@telekom.de
> Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
> 
> Hi All,
> 
> I just want to plug a draft we are working on in the AAA WG:
> 
>  Diameter Credit-control Application
>  draft-ietf-aaa-diameter-cc-01.txt
> 
> This is primarily Diameter based, though interworking with
> RADIUS is somewhat covered.  This is losely based on
> some 3GPP solutions, but we are trying to generalize it and
> make it more useful for the Internet. I do know that there are also
> some individual drafts covering RADIUS prepaid. I don't have the time,
> nor energy to specify RADIUS prepaid work.
> 
> One of the largest problems that I currently see about RADIUS
> is the under-specified nature of RADIUS.  I understand that
> many different organizations and vendors have defined, but
> not documented, vendor-specific VSAs.  In trying to do any
> meaningful work with RADIUS, I think we need to document
> some of this first.
> 
> Oh, a note to Henry, I have heard some horror stories from
> various folks (and talking to different IT support people)
> of how large-scale RADIUS deployments react after a power-
> outtage.  Since backoff is not specified in RADIUS, if
> multiple NASes comeup at the same time, they can flood
> the network, causing all sorts of fun things happening ...
> This is one area in which RADIUS would need fixing.
> 
> br,
> John
> 
> 
> > -----Original Message-----
> > From: ext Liess, Laura [mailto:Laura.Liess@t-systems.com]
> > Sent: 31 July, 2003 17:41
> > To: jdrosen@dynamicsoft.com; aboba@internaut.com
> > Cc: hgs@cs.columbia.edu; sipping@ietf.org; Dumler, Alexander; Wolff,
> > Christian
> > Subject: AW: AW: [Sipping] FYI: RADIUS & SIP
> >
> >
> > I agree with Jonathan. Support for prepaid calling seems to
> > me a very good idea.
> > Laura
> >
> > Laura Liess
> > T-Systems/Deutsche Telekom Group
> > Am Kavalleriesand 3
> > 64295 Darmstadt
> >
> > >-----Ursprungliche Nachricht-----
> > >Von: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > >Gesendet: Dienstag, 29. Juli 2003 20:13
> > >An: Bernard Aboba
> > >Cc: Liess, Laura; hgs@cs.columbia.edu; sipping@ietf.org; Dumler,
> > >Alexander; Wolff, Christian
> > >Betreff: Re: AW: [Sipping] FYI: RADIUS & SIP
> > >
> > >
> > >I know of several carriers with RADIUS infrastructures that want to
> > >use them to support SIP. They are currently using expired I-Ds and
> > >proprietary attributes. As a result, I would strongly
> > advocate work on
> > >standardizing RADIUS usage with SIP.
> > >
> > >There are two areas of work that I see a need for:
> > >
> > >1. Digest authentication. The sterman draft
> > >(http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-si
> > >p-00.txt)
> > >is used quite a bit, it seems. It would be nice to
> > standardize on this.
> > >
> > >2. Prepaid calling. Many folks are using vendor proprietary radius
> > >extensiosn to support prepaid calling. It would be nice to
> > bring those
> > >  forward and standardize on them.
> > >
> > >-Jonathan R.
> > >
> > >Bernard Aboba wrote:
> > >
> > >> Is there some particular set of draft(s) that you are
> > >advocating work on?
> > >>
> > >> On Fri, 18 Jul 2003, Liess, Laura wrote:
> > >>
> > >>
> > >>>I think most carriers currently use some kind of RADIUS
> > >platform to do user authentication and they would like to
> > >reuse it in the future to authenticate their SIP customers.
> > >>>My colleagues who are responsible for the RADIUS Platform of
> > >the Deutsche Telekom (CC)are currently on vacation so I could
> > >not check with them now about how to answer this mail, but we
> > >already talked about this issue a number of times and my
> > >strong opinion is that "yes, Deutsche Telekom cares a lot
> > >about SIP and RADIUS". Reusing the existing RADIUS platform
> > >for SIP authentication is a strong requirement for the
> > >development of SIP services within the Deutsche Telekom.
> > >>>
> > >>>Laura (T-Systems/Deutsche Telekom Group)
> > >>
> > >>
> > >> _______________________________________________
> > >> 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.                600 Lanidex Plaza
> >Chief Technology Officer                    Parsippany, NJ 07054-2711
> >dynamicsoft
> >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
> 
> _______________________________________________
> 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
Use sip@ietf.org for new developments of core SIP



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From: Jari Arkko <jarkko@piuha.net>
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Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
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Hello folks,

I'm not really sure if its productive to debate RADIUS vs.
DIAMETER deployment status or deficiences. I don't see
them as competing alternatives as much as I see them
as complementary approaches.

Here's what I'd like to have:

1. Published (RFC) specifications for carrying HTTP
    Digest over RADIUS. Maybe start from draft-sterman.
2. Published specifications for the same thing and
    a few additional features over DIAMETER. Start from
    draft-belinchon, but make it less 3GPP-specific.
3. As a part of 2, interworking specification
    between RADIUS & DIAMETER (possibly limited
    to the scope of features supported under 1).
4. Prepaid support module for DIAMETER that can
    work with any application, such as SIP or network
    access. This is work in progress, as John noted.
5. Optionally, a prepaid module for RADIUS,
    constructed in a manner which makes translation
    to 4 possible.

This will enable folks with either RADIUS or
DIAMETER servers to put them together with their
SIP nodes, and will enable gradual transition from
RADIUS to DIAMETER, which will likely take time. The
idea is that AAA proxies start supporting both
protocols, so you don't have to switch your NASes/
SIP proxies all at the same time.

Please also note that SIP is not the only thing
where RADIUS vs. DIAMETER issues arise. In network
access, for instance, the AAA WG has produced a
specification which works over DIAMETER in a manner
compatible with the current RADIUS network access
usage. Similarly, in EAP authentication over RADIUS
can interwork with EAP authentication over DIAMETER,
albeit with DIAMETER the security issues related to
key transport are easier.

--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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Subject: Re: [Sipping] draft-melanchuk-sipping-msml-00.txt
To: Tim Melanchuk <timm@convedia.com>
Cc: Chris Boulton <cboulton@ubiquity.net>,
        Garland Sharratt <gsharratt@convedia.com>, sipping@ietf.org,
        sipping-admin@ietf.org
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Hi Tim,
I have question regarding msml.
MSML defines three classes of media processing objects, viz., network
connection, conference and dialog execution context. Out of these first=
 two
are physical entities within a media server whereas dialog execution
context is an abstract thing, and a dialog will be typically bound to a=

network connection at the time of creation. The discription of the
<dialogstart> element says that  "A media stream is created between the=

dialog execution environment and the connection or conference".  I
understand the second endpoint but I am not sure what is the definition=
 of
dialog execution environment. As per my understanding, assuming dialog =
is
made up of MOML constructs such as play and record, the typical require=
ment
will be to switch the stream between player and recorder. But I can't s=
ee
how a <join> at dialog level can indicate which resource to connect to.=
 Can
you elaborate more on the dialog execution environment?
-Vishal


                                                                       =
                                   =20
                      Tim Melanchuk                                    =
                                   =20
                      <timm@convedia.co        To:       Chris Boulton =
<cboulton@ubiquity.net>            =20
                      m>                       cc:       Garland Sharra=
tt <gsharratt@convedia.com>,       =20
                      Sent by:                  <sipping@ietf.org>     =
                                   =20
                      <sipping-admin@ie        Subject:  Re: [Sipping] =
draft-melanchuk-sipping-msml-00.txt=20
                      tf.org>                                          =
                                   =20
                                                                       =
                                   =20
                                                                       =
                                   =20
                      08/06/2003 08:43                                 =
                                   =20
                      PM                                               =
                                   =20
                                                                       =
                                   =20
                                                                       =
                                   =20




thanks for your review and comments! nits will be reflected in the
next revision. responses to questions are inline below.

cheers,
timm

Chris Boulton wrote:

 > - Would any benefit be gained from including an optional 'order'
 > attribute on primitive elements so that when 'pipe' is used, the ord=
er
 > is determined explicitly rather than following a logical path?
 >
this question is related to MOML. the goal was to provide several simpl=
e
topology schemas to cover the common use cases. i think that explicit
control of topology (i.e. as in mpcp) may be useful. i'm not sure that
explicit ordering in this case would have benefit and may introduce
additional error cases. did you have any specific benefits in mind?

 > - Was there a reason that JOIN/UNJOIN were limited to id1 + id2.  Co=
uld
 > it not have just been id, that was forced to appear at least twice?
 > This would allow you to join >2 media streams in one transaction.
 >
simplicity. msml already allows multiple <join> elements to be present
in a single transaction. allowing multiple streams to be identified in
a single join introduces more complexity and makes failure handling
more difficult.

 >
 > - <result> element should say that the response code corresponds to =
the
 > SIP response code received for the transaction.
 >
i'm not quite sure what you mean here. the response attribute in <resul=
t>
is for the msml transaction. it just happens to be carried in the respo=
nse
to the sip message which conveyed the msml transaction. however msml is=

designed to be neutral to the transport mechanism and this correlation
may not exist in all cases.

 > from MSML: Media objects may be defined using MOML or other suitable=

media
 >    resource description languages. A media server which supports
 >    <insert> and <remove> must support MOML and may support other
 >    languages.
 >
 > Should this be a 'MUST' or is it not possible to only use another
 > 'suitable media resource description language' with MSML?
 >
a media resource description language is required. the "must" requires
that there be at least one common language to insure interoperability.
it does not preclude that other languages are supported.





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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: Thu, 07 Aug 2003 08:59:39 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
In-reply-to: <3F324F5D.6040903@piuha.net>
To: jarkko@piuha.net, "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        john.loughney@nokia.com
Cc: aboba@internaut.com, sipping@ietf.org
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> I'm not really sure if its productive to debate RADIUS vs.
> DIAMETER deployment status or deficiences.

I agree, and we should add WEB SERVICES to the list - probably the most
important contender. To get away from this discussion, the draft on SIP-AAA
needs an additional chapter with an objective comparison on various options,
and to discuss the pros and cons of such as RADIUS, DIAMETER and WEB
SERVICES. (BTW: Is there a lack of expertise on web services in the SIPPING
WG? Some well know WEB SERVICES vendors come to my mind...Invite them?)

A discussion of options is the only basis on which to proceed IMHO.

Comments?

Thanks, Henry 

> -----Original Message-----
> From: Jari Arkko [mailto:jarkko@piuha.net]
> Sent: Thursday, August 07, 2003 8:09 AM
> To: Henry Sinnreich; john.loughney@nokia.com
> Cc: aboba@internaut.com; sipping@ietf.org
> Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> 
> 
> Hello folks,
> 
> I'm not really sure if its productive to debate RADIUS vs.
> DIAMETER deployment status or deficiences. I don't see
> them as competing alternatives as much as I see them
> as complementary approaches.
> 
> Here's what I'd like to have:
> 
> 1. Published (RFC) specifications for carrying HTTP
>     Digest over RADIUS. Maybe start from draft-sterman.
> 2. Published specifications for the same thing and
>     a few additional features over DIAMETER. Start from
>     draft-belinchon, but make it less 3GPP-specific.
> 3. As a part of 2, interworking specification
>     between RADIUS & DIAMETER (possibly limited
>     to the scope of features supported under 1).
> 4. Prepaid support module for DIAMETER that can
>     work with any application, such as SIP or network
>     access. This is work in progress, as John noted.
> 5. Optionally, a prepaid module for RADIUS,
>     constructed in a manner which makes translation
>     to 4 possible.
> 
> This will enable folks with either RADIUS or
> DIAMETER servers to put them together with their
> SIP nodes, and will enable gradual transition from
> RADIUS to DIAMETER, which will likely take time. The
> idea is that AAA proxies start supporting both
> protocols, so you don't have to switch your NASes/
> SIP proxies all at the same time.
> 
> Please also note that SIP is not the only thing
> where RADIUS vs. DIAMETER issues arise. In network
> access, for instance, the AAA WG has produced a
> specification which works over DIAMETER in a manner
> compatible with the current RADIUS network access
> usage. Similarly, in EAP authentication over RADIUS
> can interwork with EAP authentication over DIAMETER,
> albeit with DIAMETER the security issues related to
> key transport are easier.
> 
> --Jari


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From: De Leo Walter <deleow@TELEFONICA.COM.AR>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] draft-melanchuk-sipping-moml-00.txt
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I am interested in knowing if this drafat an msml draft will be =
proposed as
workgroup items.

I think it should there are some proprietary solutions for ms cotnrol =
over
SIP.

Regards

> -----Mensaje original-----
> De:	Tim Melanchuk [SMTP:timm@convedia.com]
> Enviado el:	Mi=E9rcoles 6 de Agosto de 2003 21:46
> Para:	Chris Boulton
> CC:	Garland Sharratt; sipping@ietf.org
> Asunto:	Re: [Sipping] draft-melanchuk-sipping-moml-00.txt
>=20
> thanks again for your feedback. answers to questions below.
>=20
> cheers,
> timm
>=20
> >=20
> > - Is there an alternative method for defining unlimited occurrences =

> > using the 'max' attribute =3D 0 (e.g. pattern element uses max =
within DTMF
>=20
> > primitive).  This seemed a little confusing.
> >=20
> i've had this comment from others. would extending the datatype
> to include an explicit "unlimited" value be less confusing and/or
> renaming "max" to "maxoccurs"?
>=20
> >=20
> > - 4.5.1.1 The description of <tvoice> and <tsilence> needs to be =
made=20
> > more explicit.
> >=20
> will do.
>=20
>=20
>=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

_______________________________________________
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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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Date: Thu, 07 Aug 2003 16:09:43 +0100
From: "Elwell, John" <john.elwell@siemens.com>
To: "'Francois Audet'" <audet@nortelnetworks.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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Subject: [Sipping] RE: WGLC for draft-ietf-sipping-qsig2sip-01
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Francois,

In-line.

Regards,

John


- Section 8.3.1, page 19 
  
  5th paragraph: Add ", unless the gateway is aware that in-band information

  will not be used" to the end of the paragraph. The reason for this is that

  there are cases where the gateway may have a priori-knowledge that in-band

  tones and announcements are not used; and out-of-band ALERTING or 
  fast connection (i.e., no PROGRESS and no ALERTING) is always 
  used for example. In that case, there is no reason to reject the call with

  488 if the SDP was not included in the INVITE. 
[JRE] I agree that if there is a priori knowledge that in-band tones and
announcements will not be used, then what you propose is reasonable. I am
rather doubtful that the gateway will have sufficient knowledge to determine
this. The 4th and 5th paragraphs of 8.3.3 mandate the connection of media
streams if relevant QSIG progress indicators are present in a PROGRESS or
ALERTING message. The presence of these progress indicators is outside the
control of the gateway. So how can the gateway have a priori knowledge that
these progress indicators will not be received?
 
[FA]I'm talking about the gateway RECEIVING the SETUP, not the one sending
it.

[JRE]You referred originally to 8.3.1, which deals with receiving INVITE /
sending SETUP, not receiving SETUP. From the rest of your remarks below I
assume you really do mean 8.3.1 and the gateway receiving the INVITE /
sending the SETUP.

[FA]The case I am thinking of is then the Gateway is an IP PBX (with SIP on
one side, QSIG on the other). If the call terminates on a local phone, it
may use out-of-band ringing (our PBXs work that way).

If we want to be picky about the meaning of  Gateway, and dissociate the
gateway from the actual QSIG PBX, then it can still happen. The gateway may
decide to map the INVITE to SETUP anyways and forward it to the QSIG PBX. If
the PBX responds with the ALERTING indicating out-of-band, everything is
fine (which is why I would like this change). If on the other hand, the
ALERTING indicates in-band, then the gateway could decide to do as per your
procedure and send the 488. But I wouldn't want to mandate 488 everytime the
SDP is not in the INVITE, especially since it is a very normal occurence in
SIP.

[JRE] I would not want to send 488 after receiving a QSIG ALERTING or
PROGRESS message indicating in-band information, because by then the called
user in the QSIG network might already be being alerted and it would not be
good practice to cancel the call at this stage.

[FA]Even for the second case, where the call ultimately terminates on a line
with an in-band signalling, I am not convinced at all that the qsig2sip
draft should mandate that the call be terminated. To me this should be a
design option. An implementer may decide that it prefers to go through with
the call, and provide some sort of indication to the user that the call is
"progressing" instead of failing the call (I expect that a proper SIP
implemenation would do something like that).

I would prefer that the paragraph does not mandate the 488 as other options
are doable, and probably much less drastic than call failure (as an
out-of-band progress indication would be).

[JRE] On reflection I can see some merit in allowing the call to proceed,
even though we cannot guarantee that in-band information will be passed to
the caller. If the caller is a user of a SIP device of some sort, even if we
pass in-band information back through the SIP network, there is no guarantee
that the SIP device will play it to the user. On the other hand, if there is
another legacy network on the originating side of the SIP network, it will
of course be able to pass in-band information and the SIP network (using
unreliable provisional responses) would be the sole cause of failure to pass
in-band information end-to-end. In the case of another QSIG network we have
mandated support for 100rel (see 8.2.1.1) anyway. So on balance I think I
could agree to relax the rules concerning a 488.
I propose making it a SHOULD and including the text: "A gateway that has a
priori knowledge that essential in-band information will not need to be sent
before answer can choose to proceed with the call in these circumstances."

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Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
Date: Thu, 7 Aug 2003 10:41:24 -0500
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From: "Jill B Gemmill" <JGemmill@uab.edu>
To: "Henry Sinnreich" <Henry.Sinnreich@mci.com>, <jarkko@piuha.net>,
        <john.loughney@nokia.com>
Cc: <aboba@internaut.com>, <sipping@ietf.org>
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Why not include PKI (digital certificates) and API to Peterson's
Authentication Service
http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-01.txt in
the options space?

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> Sent: Thursday, August 07, 2003 9:00 AM
> To: jarkko@piuha.net; 'Henry Sinnreich'; john.loughney@nokia.com
> Cc: aboba@internaut.com; sipping@ietf.org
> Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
>=20
> > I'm not really sure if its productive to debate RADIUS vs.
> > DIAMETER deployment status or deficiences.
>=20
> I agree, and we should add WEB SERVICES to the list - probably the
most
> important contender. To get away from this discussion, the draft on
SIP-
> AAA
> needs an additional chapter with an objective comparison on various
> options,
> and to discuss the pros and cons of such as RADIUS, DIAMETER and WEB
> SERVICES. (BTW: Is there a lack of expertise on web services in the
> SIPPING
> WG? Some well know WEB SERVICES vendors come to my mind...Invite
them?)
>=20
> A discussion of options is the only basis on which to proceed IMHO.
>=20
> Comments?
>=20
> Thanks, Henry
>=20
> > -----Original Message-----
> > From: Jari Arkko [mailto:jarkko@piuha.net]
> > Sent: Thursday, August 07, 2003 8:09 AM
> > To: Henry Sinnreich; john.loughney@nokia.com
> > Cc: aboba@internaut.com; sipping@ietf.org
> > Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> >
> >
> > Hello folks,
> >
> > I'm not really sure if its productive to debate RADIUS vs.
> > DIAMETER deployment status or deficiences. I don't see
> > them as competing alternatives as much as I see them
> > as complementary approaches.
> >
> > Here's what I'd like to have:
> >
> > 1. Published (RFC) specifications for carrying HTTP
> >     Digest over RADIUS. Maybe start from draft-sterman.
> > 2. Published specifications for the same thing and
> >     a few additional features over DIAMETER. Start from
> >     draft-belinchon, but make it less 3GPP-specific.
> > 3. As a part of 2, interworking specification
> >     between RADIUS & DIAMETER (possibly limited
> >     to the scope of features supported under 1).
> > 4. Prepaid support module for DIAMETER that can
> >     work with any application, such as SIP or network
> >     access. This is work in progress, as John noted.
> > 5. Optionally, a prepaid module for RADIUS,
> >     constructed in a manner which makes translation
> >     to 4 possible.
> >
> > This will enable folks with either RADIUS or
> > DIAMETER servers to put them together with their
> > SIP nodes, and will enable gradual transition from
> > RADIUS to DIAMETER, which will likely take time. The
> > idea is that AAA proxies start supporting both
> > protocols, so you don't have to switch your NASes/
> > SIP proxies all at the same time.
> >
> > Please also note that SIP is not the only thing
> > where RADIUS vs. DIAMETER issues arise. In network
> > access, for instance, the AAA WG has produced a
> > specification which works over DIAMETER in a manner
> > compatible with the current RADIUS network access
> > usage. Similarly, in EAP authentication over RADIUS
> > can interwork with EAP authentication over DIAMETER,
> > albeit with DIAMETER the security issues related to
> > key transport are easier.
> >
> > --Jari
>=20
>=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

_______________________________________________
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This list is for NEW development of the application of SIP
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Date: Thu, 07 Aug 2003 18:11:01 +0200
To: "Liess, Laura" <Laura.Liess@t-systems.com>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
Cc: hgs@cs.columbia.edu, jdrosen@dynamicsoft.com, aboba@internaut.com,
        sipping@ietf.org, "Dumler, Alexander" <Alexander.Dumler@telekom.de>,
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There is a believe in the IETF that lack of reliability in RADIUS
determines this work to be dropped. (Authentication is a different
story, though.)

As for us, we took freedom not to wait until a solid answer exists
and implemented this draft.

-jiri


At 10:17 AM 8/6/2003, Liess, Laura wrote:
>What about the accounting draft draft-schulzrinne-accounting-sip-01.txt? Any reason for not working on it? 
>Laura 
>
>>-----Ursprungliche Nachricht-----
>>Von: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
>>Gesendet: Dienstag, 5. August 2003 18:00
>>An: Jonathan Rosenberg; Bernard Aboba
>>Cc: Liess, Laura; hgs@cs.columbia.edu; sipping@ietf.org; Dumler,
>>Alexander; Wolff, Christian
>>Betreff: Re: AW: [Sipping] FYI: RADIUS & SIP
>>
>>
>>Thanks, I very much agree; we have received identical feedback so far.
>>
>>-jiri
>>
>>At 08:12 PM 7/29/2003, Jonathan Rosenberg wrote:
>>>I know of several carriers with RADIUS infrastructures that 
>>want to use them to support SIP. They are currently using 
>>expired I-Ds and proprietary attributes. As a result, I would 
>>strongly advocate work on standardizing RADIUS usage with SIP.
>>>
>>>There are two areas of work that I see a need for:
>>>
>>>1. Digest authentication. The sterman draft 
>>(http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-si
>>p-00.txt) is used quite a bit, it seems. It would be nice to 
>>standardize on this.
>>>
>>>2. Prepaid calling. Many folks are using vendor proprietary 
>>radius extensiosn to support prepaid calling. It would be nice 
>>to bring those  forward and standardize on them.
>>>
>>>-Jonathan R.
>>>
>>>Bernard Aboba wrote:
>>>
>>>>Is there some particular set of draft(s) that you are 
>>advocating work on?
>>>>On Fri, 18 Jul 2003, Liess, Laura wrote:
>>>>
>>>>>I think most carriers currently use some kind of RADIUS 
>>platform to do user authentication and they would like to 
>>reuse it in the future to authenticate their SIP customers.
>>>>>My colleagues who are responsible for the RADIUS Platform 
>>of the Deutsche Telekom (CC)are currently on vacation so I 
>>could not check with them now about how to answer this mail, 
>>but we already talked about this issue a number of times and 
>>my strong opinion is that "yes, Deutsche Telekom cares a lot 
>>about SIP and RADIUS". Reusing the existing RADIUS platform 
>>for SIP authentication is a strong requirement for the 
>>development of SIP services within the Deutsche Telekom.
>>>>>
>>>>>Laura (T-Systems/Deutsche Telekom Group)
>>>>
>>>>_______________________________________________
>>>>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.                600 Lanidex Plaza
>>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>>dynamicsoft
>>>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
>> 

--
Jiri Kuthan            http://iptel.org/~jiri/ 


_______________________________________________
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Use sip@ietf.org for new developments of core SIP



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Date: Thu, 07 Aug 2003 18:33:28 +0200
To: David Schwartz <davids@deltathree.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
From: Jiri Kuthan <jiri@iptel.org>
Subject: RE: [Sipping] Re: Update to 3pcc
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David,

ok, I must thave misunderstood your argument. So you are saying
prepaid providers need a b2bua for teardown anyhow, so why to 
bother about REFER? A quick answer is that in an open-service
model, there may be other parties than the prepaid provider and 
these parties may prefer a simple technology over b2bua. And
since REFER is technically better, I'm no way happy about calling
it BCP in IETF.

More details:
- c-t-d may be implemented somewhere completely else than
  prepaid is. For example, prepaid b2bua is in front of your gateway,
  C-T-D is an application at caller's PC. 
- Since the prepaid and ctd applications live somewhere else, there
  is no reason to dictate use of a b2bua mechanism for ctd.
  Caller simply prefers a simple utility over a B2BUA. 
- And since REFER is imho doing a better job for ctd (think for example
  about the ringing problem), there is no reason to call a B2BUA-based 
  construct BCP.

I'm not opposed to b2bua middlebox in general -- there are certainly
applications which are hard to achieve otherwise, prepaid one of them. 
What I'm saying is that a BCP should show examples which are really 
_Best_ Current Practices. That's simply not the case with b2bua/ctd.

-Jiri

At 08:17 AM 8/6/2003, David Schwartz wrote:
>Hi Jiri.
>
>I was referring specifically to the call teardown issue associated with most
>pre-paid billing models. 
>
>David
>
>-----Original Message-----
>From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
>Sent: Tue, August 05, 2003 7:39 PM
>To: David Schwartz; 'sipping@ietf.org'
>Subject: Re: [Sipping] Re: Update to 3pcc
>
>
>Hello David,
>
>I don't see how billing servers are related to method of 
>call initiation.
>
>Actually I believe that how a call is initiated (by manual
>dialing a destination, by REFER-based click-to-dial, by b2bua-based
>click-to-dial, CTI, etc.) is completely orthogonal to how the 
>call is charged for. Regardless what charging system you use 
>(if any), there will be different methods for initiating calls, 
>some will be better off than others. I maintain that a click-to-dial 
>b2bua is not a BCP.
>
>-jiri
>
>At 10:10 AM 7/30/2003, David Schwartz wrote:
>>Hi Jiri
>>
>>REFER is good solution if you assume that either there are no billing
>>issues, or alternatively, that the endpoint/s handle the billing issues. If
>>this is not the case, for example, in an application where a centralized
>>controller is setting up a call to two PSTN numbers, then you will need a
>>centralized billing entity and REFER will not suffice.
>>
>>David
>>
>>-----Original Message-----
>>From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
>>Sent: Mon, July 28, 2003 7:00 PM
>>To: sipping@ietf.org
>>Subject: Sipping digest, Vol 1 #1144 - 2 msgs
>>
>>
>>Send Sipping mailing list submissions to
>>        sipping@ietf.org
>>
>>To subscribe or unsubscribe via the World Wide Web, visit
>>        https://www1.ietf.org/mailman/listinfo/sipping
>>or, via email, send a message with subject or body 'help' to
>>        sipping-request@ietf.org
>>
>>You can reach the person managing the list at
>>        sipping-admin@ietf.org
>>
>>When replying, please edit your Subject line so it is more specific
>>than "Re: Contents of Sipping digest..."
>>
>>
>>Today's Topics:
>>
>>   1. Re: Diversion header (Cullen Jennings)
>>   2. Re: Update to 3pcc (Jiri Kuthan)
>>
>>--__--__--
>>
>>Message: 1
>>Date: Sun, 27 Jul 2003 19:05:10 -0700
>>Subject: Re: [Sipping] Diversion header
>>From: Cullen Jennings <fluffy@cisco.com>
>>To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
>>
>>
>>I assume your question is there some way for  for the final destination
>that
>>receives the call to know the who the 1st callee was that had some device
>in
>>the network redirect the call elsewhere.
>>
>>The answer is YES - there has been a lot of confusion over this so think
>>about my answer carefully - it will seem counter intuitive at first. The
>>answer is the "To" header but it will take me a second to explain why.
>>
>>Imagine case 1 where A send a call to proxy B,  that sent to to C. Now C
>>sent a 302 back B and B then forked the call to D. The To header in the
>>message D receives will still have the To set to C. C is the element that
>>sent the first 302.
>>
>>Now imagine a case (like your question) where there is not a recursive
>>proxy. The 302 will go all the way back to A. Now A forms a new invite to
>>but it can leave the To header set to the same thing it was in the original
>>message (C in this case). When the message arrives at D, the To header will
>>be set to C.
>>
>>If D was a voice mail system, it might be able to know to act on behalf of
>C
>>instead of some other entity.
>>
>>There are more complex things that relate to your request. Mary Barnes has
>>been moving the history work along
>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-04.txt
>>
>>There is some work on QSIG mappings with diversion
>>http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2sip-diversion
>-
>>00.txt
>>
>>There is a draft-levy-sip-diversion which I (and many others) believe has a
>>pretty much zero chance of being accepted or becoming a RFC. This work has
>>been superceded by the history work. Quite a few people do not support the
>>history work and it is moving very slowly. Don't expect any of this to
>>resolve quickly.
>>
>>Give some careful thought to if the "To" header meets your needs and if it
>>does use - The To header is widely implemented :-)
>>
>>
>>Cullen
>>
>>
>>
>>
>>On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
>>
>>> Is there any SIP header defined in any SIP RFC or Internet Draft that can
>>> include the address of the 1st callee, who redirected the call using 302,
>>in
>>> the 2nd outgoing INVITE?
>>> 
>>> _______________________________________________
>>> 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
>>> 
>>
>>
>>
>>--__--__--
>>
>>Message: 2
>>Date: Mon, 28 Jul 2003 10:17:51 +0200
>>To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping
><sipping@ietf.org>
>>From: Jiri Kuthan <jiri@iptel.org>
>>Subject: Re: [Sipping] Update to 3pcc
>>
>>I'm actually not convinced that it is proper to show click-to-dial
>>as a motivating example for use of b2bua and label it BCP.
>>
>>Use of REFER seems technically a superior solution worth being called
>>BCP. Roots of the advantages are in a better application of e2e principle. 
>>REFER takes no middle-box b2bua -- it delegates signaling to end-points.
>>Advantages include:
>>- All the pitfalls with media are so straigforward, they don't even 
>>  need to be described (like now in sessions 8 and 9).  
>>- It takes no media server to transmit ringing tone. 
>>- There are no SDP compatibility issues on controller's side.
>>
>>-jiri
>>
>>At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
>>>Folks,
>>>
>>>You may have noticed that I posted an update to the 3pcc spec:
>>>
>>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
>>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
>>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
>>>
>>>The update is based on comments received from IESG while reviewing the
>>draft. In particular, there were problems with the usage of 0.0.0.0 to
>>imply that no media nor RTCP should be sent. This IP address is not meant
>>for that purpose. Rather, the .invalid TLD is much better suited. So, I
>>changed the draft to make use of .invalid in flows where the controller
>>doesnt want media to be sent.
>>>
>>>Now, there are some concerns with that which folks need to be aware of:
>>>
>>>1. It is at odds with RFC 3264, which actually recommends 0.0.0.0. In
>fact,
>>the problem is really with RFC 3264, NOT 3pcc. However, 3264 is done, and
>>this one isnt. I logged a bug against RFC 3264 on this, and we'll need to
>>work on a way to transition folks towards using .invalid and not 0.0.0.0.
>>>
>>>2. One the premises of 3pcc is that it should be compatible with any UA
>>that is RFC 3261 compliant. If a controller uses .invalid in the SDP, I am
>>not sure what a SIP endpoint would do. I don't believe anything in RFC 3264
>>or in SIP talks about handling the case where there is an issue with a
>>domain name in the SDP. As such, if you use .invalid, there could be
>interop
>>problems. The document mentions this.
>>>
>>>Comments or questions welcome.
>>>
>>>Thanks,
>>>Jonathan R.
>>>-- 
>>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>>dynamicsoft
>>>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
>>
>>--
>>Jiri Kuthan            http://iptel.org/~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
>>
>>End of Sipping Digest
>>
>>_______________________________________________
>>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 

--
Jiri Kuthan            http://iptel.org/~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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Date: Thu, 07 Aug 2003 11:33:42 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
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> Why not include PKI (digital certificates) and API to Peterson's
> Authentication Service

Excellent suggestion! We have now four options to discuss in the SIP-AAA
I-D.

Thanks, Henry

> -----Original Message-----
> From: Jill B Gemmill [mailto:JGemmill@uab.edu]
> Sent: Thursday, August 07, 2003 10:41 AM
> To: Henry Sinnreich; jarkko@piuha.net; john.loughney@nokia.com
> Cc: aboba@internaut.com; sipping@ietf.org
> Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
> 
> Why not include PKI (digital certificates) and API to Peterson's
> Authentication Service
> http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-01.txt in
> the options space?
> 
> > -----Original Message-----
> > From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> > Sent: Thursday, August 07, 2003 9:00 AM
> > To: jarkko@piuha.net; 'Henry Sinnreich'; john.loughney@nokia.com
> > Cc: aboba@internaut.com; sipping@ietf.org
> > Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
> >
> > > I'm not really sure if its productive to debate RADIUS vs.
> > > DIAMETER deployment status or deficiences.
> >
> > I agree, and we should add WEB SERVICES to the list - probably the
> most
> > important contender. To get away from this discussion, the draft on
> SIP-
> > AAA
> > needs an additional chapter with an objective comparison on various
> > options,
> > and to discuss the pros and cons of such as RADIUS, DIAMETER and WEB
> > SERVICES. (BTW: Is there a lack of expertise on web services in the
> > SIPPING
> > WG? Some well know WEB SERVICES vendors come to my mind...Invite
> them?)
> >
> > A discussion of options is the only basis on which to proceed IMHO.
> >
> > Comments?
> >
> > Thanks, Henry
> >
> > > -----Original Message-----
> > > From: Jari Arkko [mailto:jarkko@piuha.net]
> > > Sent: Thursday, August 07, 2003 8:09 AM
> > > To: Henry Sinnreich; john.loughney@nokia.com
> > > Cc: aboba@internaut.com; sipping@ietf.org
> > > Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> > >
> > >
> > > Hello folks,
> > >
> > > I'm not really sure if its productive to debate RADIUS vs.
> > > DIAMETER deployment status or deficiences. I don't see
> > > them as competing alternatives as much as I see them
> > > as complementary approaches.
> > >
> > > Here's what I'd like to have:
> > >
> > > 1. Published (RFC) specifications for carrying HTTP
> > >     Digest over RADIUS. Maybe start from draft-sterman.
> > > 2. Published specifications for the same thing and
> > >     a few additional features over DIAMETER. Start from
> > >     draft-belinchon, but make it less 3GPP-specific.
> > > 3. As a part of 2, interworking specification
> > >     between RADIUS & DIAMETER (possibly limited
> > >     to the scope of features supported under 1).
> > > 4. Prepaid support module for DIAMETER that can
> > >     work with any application, such as SIP or network
> > >     access. This is work in progress, as John noted.
> > > 5. Optionally, a prepaid module for RADIUS,
> > >     constructed in a manner which makes translation
> > >     to 4 possible.
> > >
> > > This will enable folks with either RADIUS or
> > > DIAMETER servers to put them together with their
> > > SIP nodes, and will enable gradual transition from
> > > RADIUS to DIAMETER, which will likely take time. The
> > > idea is that AAA proxies start supporting both
> > > protocols, so you don't have to switch your NASes/
> > > SIP proxies all at the same time.
> > >
> > > Please also note that SIP is not the only thing
> > > where RADIUS vs. DIAMETER issues arise. In network
> > > access, for instance, the AAA WG has produced a
> > > specification which works over DIAMETER in a manner
> > > compatible with the current RADIUS network access
> > > usage. Similarly, in EAP authentication over RADIUS
> > > can interwork with EAP authentication over DIAMETER,
> > > albeit with DIAMETER the security issues related to
> > > key transport are easier.
> > >
> > > --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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Hi John,

I agree with your replies to my comments. More details in-line.

regards,

Jean-Francois.

"Elwell, John" wrote :
> 
> Jean-Francois,
> 
> Thanks for your comments. I have dealt with most of the editorials. My
> responses to your technical comments:
> 
> <SNIP>
<SNIP>
> #4 - Page 10 -  7 General requirements
> "As a result of DNS look-up by the gateway in order to determine where
> to send a SIP INVITE request, a number of candidate destinations can
> be attempted in sequence. The way in which this is handled by the
> gateway is outside the scope of this specification. However, any
> behaviour specified in this document on receipt of a SIP final
> response SHOULD apply only when there are no more candidate
> destinations to try."
> 
> Does it mean that the ingress gateway SHOULD try the next retrieved
> addresses if
> the first one responded 484 for instance ? I guess that rfc3263 narrows the
> scope to "network" or "component" failures.
> [JRE] I guess when overlap signalling applies in the SIP network the
> statement is not quite true, so I will make an exception. Also I believe the
> statement is not applicable to 3xx or 6xx responses, so I will limit it to
> 4xx and 5xx responses. Does this cover all the situations you had in mind?

I guess that this behaviour is not specific to qsig2sip, so I'm not sure it is a
good idea to mention it. 

<SNIP>
> 
> #6 - Page 14 - 8.2.1.3 Receipt of SIP 18x provisional response
> "NOTE 2. Media streams are established as a result of receiving SDP
> answer information in a reliable provisional response and can be "
> 
> I agree that reliable provisional response are better, but why use reliable
> here
> ? If a UA that does not support PRACK sends a 183 with a SDP body, the media
> stream is established too.
> [JRE] You are correct. However, the text concerned talks about SDP answer
> but media streams can also be established at this time by means of SDP offer
> in a provisional response. The text fails to mention this. According to
> RFC3261 section 13.2.1 an unreliable provisional response cannot be used for
> SDP offer. I will clarify the statement. Also subsequent modification of the
> session by means of UPDATE applies only after a offer has been received in a
> reliable message and answer has been sent, so this too would need
> qualifying. However, it is not really necessary to mention UPDATE so perhaps
> that part can be deleted.

[JFR] I agree that it would be better to remove it.
<SNIP>

> #9 - Page 24 - 8.3.9
> "If a gateway receives a SIP INVITE request for a new dialog with the
>         same Call-ID as an existing SIP INVITE request for which the gateway
> 
>         has not yet sent a final response and failing to meet the other
>         conditions above concerning overlap sending, the gateway SHALL
>         respond to the new request with a SIP 485 (Ambiguous) response."
> It's not very important, but could you explain the motivation of this
> sentence.
> Is it related to the merged requests ?
> [JRE] Its purpose is to prescribe behaviour if the gateway receives an
> INVITE request that in all respects looks like overlap sending but fails to
> meet the criterion of the derived called party number being a superset of
> the previously derived called party number. 
<SNIP>
[JFR] OK. Now, I undertand what you meant. No need to change anything.

> #10 - Page 24 - 8.4.1
> "as specified in table 1."
> Why aren't some status codes symetric in table 1 and table 2. For instance
> - 408 Request timeout maps 102 Recovery on timer expiry and
> - 102 Recovery on timer expiry maps 504 Server time-out.
> BTW, I like the way table 2 is similar to what RFC3398 describes.
> [JRE] 504 also maps to 102, but it is not possible to make 102 map to both
> 408 and 504 - we had to pick one. This illustrates the point that there is
> not a 1:1 correspondence between SIP responses and Q.850 causes and
> therefore there will be some asymmetries and some rather questionable
> mappings. Cause mapping was discussed long and hard for ISUP interworking
> and again for QSIG interworking (resulting in some changes to ISUP
> interworking to keep alignment). It is not perfect and it never will be, but
> I would be reluctant to open up the discussion again. I hope you can live
> with what we have.

[JFR] You are correct. Since it is not a 1:1 mapping, a choice must be made. It
suits me.

<SNIP>
> And some remarks regarding other SIPPING drafts
> 
> #1 draft-ietf-sipping-overlap-04.txt
> It says :
>    "However, overlap signalling adds a requirement to this process. 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."
> 
> Is this requirement applicable here ?
> [JRE] I remember this was discussed and we thought it best not to try to
> give guidance in this tricky situation.
> 
[JFR] OK.

> #2  draft-ietf-sipping-pstn-call-flows-02.txt 2.2, 4.2
> 
> I know that qsig2sip deals with QSIG only, but don't you think that people
> might
> get confused by the slight difference in the mapping of the messages of the
> "ISDN PBX" examples of the call-flow draft ?
> [JRE] There are of course some differences between QSIG and Q.931 that can
> have impact on the mapping. Also I am not sure of the status of the
> call-flows draft and in particular the two examples showing Q.931 mapping,
> because I thought the group had decided not to work on SIP-Q.931 mapping.

[JFR] I'll check with the authors. Anyway a call-flow draft is not normative.

> 
> </SNIP>
> 
> Regards,
> 
> John (john.elwell@siemens.com)

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From: "Francois Audet" <audet@nortelnetworks.com>
To: "'Elwell, John'" <john.elwell@siemens.com>,
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Subject: [Sipping] RE: WGLC for draft-ietf-sipping-qsig2sip-01
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I'm OK with your new wording.

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Thursday, August 07, 2003 08:10
> To: Audet, Francois [SC100:4K02:EXCH]; 'sipping@ietf.org'
> Subject: RE: WGLC for draft-ietf-sipping-qsig2sip-01
> 
> 
> Francois,
> 
> In-line.
> 
> Regards,
> 
> John
> 
> 
> - Section 8.3.1, page 19 
>   
>   5th paragraph: Add ", unless the gateway is aware that 
> in-band information
> 
>   will not be used" to the end of the paragraph. The reason 
> for this is that
> 
>   there are cases where the gateway may have a 
> priori-knowledge that in-band
> 
>   tones and announcements are not used; and out-of-band ALERTING or 
>   fast connection (i.e., no PROGRESS and no ALERTING) is always 
>   used for example. In that case, there is no reason to 
> reject the call with
> 
>   488 if the SDP was not included in the INVITE. 
> [JRE] I agree that if there is a priori knowledge that 
> in-band tones and announcements will not be used, then what 
> you propose is reasonable. I am rather doubtful that the 
> gateway will have sufficient knowledge to determine this. The 
> 4th and 5th paragraphs of 8.3.3 mandate the connection of 
> media streams if relevant QSIG progress indicators are 
> present in a PROGRESS or ALERTING message. The presence of 
> these progress indicators is outside the control of the 
> gateway. So how can the gateway have a priori knowledge that 
> these progress indicators will not be received?
>  
> [FA]I'm talking about the gateway RECEIVING the SETUP, not 
> the one sending it.
> 
> [JRE]You referred originally to 8.3.1, which deals with 
> receiving INVITE / sending SETUP, not receiving SETUP. From 
> the rest of your remarks below I assume you really do mean 
> 8.3.1 and the gateway receiving the INVITE / sending the SETUP.
> 
> [FA]The case I am thinking of is then the Gateway is an IP 
> PBX (with SIP on one side, QSIG on the other). If the call 
> terminates on a local phone, it may use out-of-band ringing 
> (our PBXs work that way).
> 
> If we want to be picky about the meaning of  Gateway, and 
> dissociate the gateway from the actual QSIG PBX, then it can 
> still happen. The gateway may decide to map the INVITE to 
> SETUP anyways and forward it to the QSIG PBX. If the PBX 
> responds with the ALERTING indicating out-of-band, everything 
> is fine (which is why I would like this change). If on the 
> other hand, the ALERTING indicates in-band, then the gateway 
> could decide to do as per your procedure and send the 488. 
> But I wouldn't want to mandate 488 everytime the SDP is not 
> in the INVITE, especially since it is a very normal occurence in SIP.
> 
> [JRE] I would not want to send 488 after receiving a QSIG 
> ALERTING or PROGRESS message indicating in-band information, 
> because by then the called user in the QSIG network might 
> already be being alerted and it would not be good practice to 
> cancel the call at this stage.
> 
> [FA]Even for the second case, where the call ultimately 
> terminates on a line with an in-band signalling, I am not 
> convinced at all that the qsig2sip draft should mandate that 
> the call be terminated. To me this should be a design option. 
> An implementer may decide that it prefers to go through with 
> the call, and provide some sort of indication to the user 
> that the call is "progressing" instead of failing the call (I 
> expect that a proper SIP implemenation would do something like that).
> 
> I would prefer that the paragraph does not mandate the 488 as 
> other options are doable, and probably much less drastic than 
> call failure (as an out-of-band progress indication would be).
> 
> [JRE] On reflection I can see some merit in allowing the call 
> to proceed, even though we cannot guarantee that in-band 
> information will be passed to the caller. If the caller is a 
> user of a SIP device of some sort, even if we pass in-band 
> information back through the SIP network, there is no 
> guarantee that the SIP device will play it to the user. On 
> the other hand, if there is another legacy network on the 
> originating side of the SIP network, it will of course be 
> able to pass in-band information and the SIP network (using 
> unreliable provisional responses) would be the sole cause of 
> failure to pass in-band information end-to-end. In the case 
> of another QSIG network we have mandated support for 100rel 
> (see 8.2.1.1) anyway. So on balance I think I could agree to 
> relax the rules concerning a 488. I propose making it a 
> SHOULD and including the text: "A gateway that has a priori 
> knowledge that essential in-band information will not need to 
> be sent before answer can choose to proceed with the call in 
> these circumstances."
> 

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<P><FONT SIZE=2>I'm OK with your new wording.</FONT>
</P>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: Elwell, John [<A HREF="mailto:john.elwell@siemens.com">mailto:john.elwell@siemens.com</A>] </FONT>
<BR><FONT SIZE=2>&gt; Sent: Thursday, August 07, 2003 08:10</FONT>
<BR><FONT SIZE=2>&gt; To: Audet, Francois [SC100:4K02:EXCH]; 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=2>&gt; Subject: RE: WGLC for draft-ietf-sipping-qsig2sip-01</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Francois,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; In-line.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Regards,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; John</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; - Section 8.3.1, page 19 </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; 5th paragraph: Add &quot;, unless the gateway is aware that </FONT>
<BR><FONT SIZE=2>&gt; in-band information</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; will not be used&quot; to the end of the paragraph. The reason </FONT>
<BR><FONT SIZE=2>&gt; for this is that</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; there are cases where the gateway may have a </FONT>
<BR><FONT SIZE=2>&gt; priori-knowledge that in-band</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; tones and announcements are not used; and out-of-band ALERTING or </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; fast connection (i.e., no PROGRESS and no ALERTING) is always </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; used for example. In that case, there is no reason to </FONT>
<BR><FONT SIZE=2>&gt; reject the call with</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp; 488 if the SDP was not included in the INVITE. </FONT>
<BR><FONT SIZE=2>&gt; [JRE] I agree that if there is a priori knowledge that </FONT>
<BR><FONT SIZE=2>&gt; in-band tones and announcements will not be used, then what </FONT>
<BR><FONT SIZE=2>&gt; you propose is reasonable. I am rather doubtful that the </FONT>
<BR><FONT SIZE=2>&gt; gateway will have sufficient knowledge to determine this. The </FONT>
<BR><FONT SIZE=2>&gt; 4th and 5th paragraphs of 8.3.3 mandate the connection of </FONT>
<BR><FONT SIZE=2>&gt; media streams if relevant QSIG progress indicators are </FONT>
<BR><FONT SIZE=2>&gt; present in a PROGRESS or ALERTING message. The presence of </FONT>
<BR><FONT SIZE=2>&gt; these progress indicators is outside the control of the </FONT>
<BR><FONT SIZE=2>&gt; gateway. So how can the gateway have a priori knowledge that </FONT>
<BR><FONT SIZE=2>&gt; these progress indicators will not be received?</FONT>
<BR><FONT SIZE=2>&gt;&nbsp; </FONT>
<BR><FONT SIZE=2>&gt; [FA]I'm talking about the gateway RECEIVING the SETUP, not </FONT>
<BR><FONT SIZE=2>&gt; the one sending it.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; [JRE]You referred originally to 8.3.1, which deals with </FONT>
<BR><FONT SIZE=2>&gt; receiving INVITE / sending SETUP, not receiving SETUP. From </FONT>
<BR><FONT SIZE=2>&gt; the rest of your remarks below I assume you really do mean </FONT>
<BR><FONT SIZE=2>&gt; 8.3.1 and the gateway receiving the INVITE / sending the SETUP.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; [FA]The case I am thinking of is then the Gateway is an IP </FONT>
<BR><FONT SIZE=2>&gt; PBX (with SIP on one side, QSIG on the other). If the call </FONT>
<BR><FONT SIZE=2>&gt; terminates on a local phone, it may use out-of-band ringing </FONT>
<BR><FONT SIZE=2>&gt; (our PBXs work that way).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; If we want to be picky about the meaning of&nbsp; Gateway, and </FONT>
<BR><FONT SIZE=2>&gt; dissociate the gateway from the actual QSIG PBX, then it can </FONT>
<BR><FONT SIZE=2>&gt; still happen. The gateway may decide to map the INVITE to </FONT>
<BR><FONT SIZE=2>&gt; SETUP anyways and forward it to the QSIG PBX. If the PBX </FONT>
<BR><FONT SIZE=2>&gt; responds with the ALERTING indicating out-of-band, everything </FONT>
<BR><FONT SIZE=2>&gt; is fine (which is why I would like this change). If on the </FONT>
<BR><FONT SIZE=2>&gt; other hand, the ALERTING indicates in-band, then the gateway </FONT>
<BR><FONT SIZE=2>&gt; could decide to do as per your procedure and send the 488. </FONT>
<BR><FONT SIZE=2>&gt; But I wouldn't want to mandate 488 everytime the SDP is not </FONT>
<BR><FONT SIZE=2>&gt; in the INVITE, especially since it is a very normal occurence in SIP.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; [JRE] I would not want to send 488 after receiving a QSIG </FONT>
<BR><FONT SIZE=2>&gt; ALERTING or PROGRESS message indicating in-band information, </FONT>
<BR><FONT SIZE=2>&gt; because by then the called user in the QSIG network might </FONT>
<BR><FONT SIZE=2>&gt; already be being alerted and it would not be good practice to </FONT>
<BR><FONT SIZE=2>&gt; cancel the call at this stage.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; [FA]Even for the second case, where the call ultimately </FONT>
<BR><FONT SIZE=2>&gt; terminates on a line with an in-band signalling, I am not </FONT>
<BR><FONT SIZE=2>&gt; convinced at all that the qsig2sip draft should mandate that </FONT>
<BR><FONT SIZE=2>&gt; the call be terminated. To me this should be a design option. </FONT>
<BR><FONT SIZE=2>&gt; An implementer may decide that it prefers to go through with </FONT>
<BR><FONT SIZE=2>&gt; the call, and provide some sort of indication to the user </FONT>
<BR><FONT SIZE=2>&gt; that the call is &quot;progressing&quot; instead of failing the call (I </FONT>
<BR><FONT SIZE=2>&gt; expect that a proper SIP implemenation would do something like that).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I would prefer that the paragraph does not mandate the 488 as </FONT>
<BR><FONT SIZE=2>&gt; other options are doable, and probably much less drastic than </FONT>
<BR><FONT SIZE=2>&gt; call failure (as an out-of-band progress indication would be).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; [JRE] On reflection I can see some merit in allowing the call </FONT>
<BR><FONT SIZE=2>&gt; to proceed, even though we cannot guarantee that in-band </FONT>
<BR><FONT SIZE=2>&gt; information will be passed to the caller. If the caller is a </FONT>
<BR><FONT SIZE=2>&gt; user of a SIP device of some sort, even if we pass in-band </FONT>
<BR><FONT SIZE=2>&gt; information back through the SIP network, there is no </FONT>
<BR><FONT SIZE=2>&gt; guarantee that the SIP device will play it to the user. On </FONT>
<BR><FONT SIZE=2>&gt; the other hand, if there is another legacy network on the </FONT>
<BR><FONT SIZE=2>&gt; originating side of the SIP network, it will of course be </FONT>
<BR><FONT SIZE=2>&gt; able to pass in-band information and the SIP network (using </FONT>
<BR><FONT SIZE=2>&gt; unreliable provisional responses) would be the sole cause of </FONT>
<BR><FONT SIZE=2>&gt; failure to pass in-band information end-to-end. In the case </FONT>
<BR><FONT SIZE=2>&gt; of another QSIG network we have mandated support for 100rel </FONT>
<BR><FONT SIZE=2>&gt; (see 8.2.1.1) anyway. So on balance I think I could agree to </FONT>
<BR><FONT SIZE=2>&gt; relax the rules concerning a 488. I propose making it a </FONT>
<BR><FONT SIZE=2>&gt; SHOULD and including the text: &quot;A gateway that has a priori </FONT>
<BR><FONT SIZE=2>&gt; knowledge that essential in-band information will not need to </FONT>
<BR><FONT SIZE=2>&gt; be sent before answer can choose to proceed with the call in </FONT>
<BR><FONT SIZE=2>&gt; these circumstances.&quot;</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
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Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Aug  7 13:33:32 2003
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From: "Liess, Laura" <Laura.Liess@t-systems.com>
To: KMcCandless@verisign.com
Cc: sipping@ietf.org
Subject: AW: [Sipping] FYI: RADIUS & SIP
Date: Tue, 29 Jul 2003 22:14:04 +0200
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>As I 
>understood it, the companies that have the embedded base of 
>RADIUS servers could continue to use them for SIP 

Current RADIUS does not support Digest Authentication, so you need a RADIUS extension as proposed in http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-sip-00.txt. 

There was no problem in using RADIUS together with Basic Authentication for SIP. Possibly the discussion you remember happened before the usage of Basic Authentication has been deprecated.

Laura     




>and the 
>entities deploying new DIAMETER based servers must be able to 
>interoperate with the RADIUS servers.  
>
>Kevin....
> 
>
>> -----Original Message-----
>> From: Liess, Laura [mailto:Laura.Liess@t-systems.com]
>> Sent: Friday, July 18, 2003 4:44 AM
>> To: hgs@cs.columbia.edu; sipping@ietf.org; aboba@internaut.com
>> Cc: Dumler, Alexander; Wolff, Christian
>> Subject: AW: [Sipping] FYI: RADIUS & SIP
>> 
>> 
>> I think most carriers currently use some kind of RADIUS 
>> platform to do user authentication and they would like to 
>> reuse it in the future to authenticate their SIP customers. 
>> My colleagues who are responsible for the RADIUS Platform of 
>> the Deutsche Telekom (CC)are currently on vacation so I could 
>> not check with them now about how to answer this mail, but we 
>> already talked about this issue a number of times and my 
>> strong opinion is that "yes, Deutsche Telekom cares a lot 
>> about SIP and RADIUS". Reusing the existing RADIUS platform 
>> for SIP authentication is a strong requirement for the 
>> development of SIP services within the Deutsche Telekom.  
>> 
>> Laura (T-Systems/Deutsche Telekom Group)  
>> 
>> 
>> >-----Ursprungliche Nachricht-----
>> >Von: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>> >Gesendet: Freitag, 18. Juli 2003 09:45
>> >An: sipping@ietf.org
>> >Betreff: [Sipping] FYI: RADIUS & SIP
>> >
>> >
>> >Bernard Aboba <aboba@internaut.com> would like to know about 
>> >people and 
>> >companies that care about SIP and RADIUS (for authentication). 
>> >There is 
>> >an attempt afoot to rationalize RADIUS-related efforts within 
>> >the IETF, 
>> >but there seems to be a lack of representation from SIP 
>> folks, as per 
>> >the note to the radius-authors mailing list:
>> >
>> >"I was not clear about the need for SIP/RADIUS work. 
>> >Apparently there is
>> >ongoing deployment of I-Ds (e.g. the Sterman draft on RADIUS 
>> attributes
>> >for HTTP Digest Authentication), so perhaps standardization would be
>> >valuable in documenting what is in use. On the other hand, I 
>> >also heard 
>> >that "there
>> >isn't much commercial interest." Note sure what to make of 
>> >this, perhaps
>> >someone can clarify."
>> >
>> >Henning
>> >
>> >
>> >_______________________________________________
>> >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



From exim@www1.ietf.org  Thu Aug  7 13:33:41 2003
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From: Amir Zmora <AmirZ@radvision.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'David Schwartz'"
	 <davids@deltathree.com>, sipping@ietf.org
Subject: RE: [Sipping] Re: Update to 3pcc
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I'm not sure the REFER option will solve everything. For example if a call
is connected via a pre paid system that connected the 2 parties with REFER
how will it now disconnect the call or play some warning message to the
caller if it is not actively participating in the call.
Regarding S/MIME - I don't know of too many implementations actually using
it today in real deployment. Many are using TLS and that should work OK with
the B2B.

Amir
-----Original Message-----
From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com] 
Sent: Wednesday, July 30, 2003 6:09 PM
To: 'David Schwartz'; sipping@ietf.org
Subject: RE: [Sipping] Re: Update to 3pcc

This may turn out to be a very productive discussion. David Schwartz writes:


> an application where a centralized controller is setting up a 
> call to two PSTN numbers, then you will need a centralized 
> billing entity and REFER will not suffice.

There may be more attractive billing collecting alternatives: CDRs in the
gateways. If the calls are between two IP endpoints, flat rate is a good
option.

The centralized controller, AKA B2BUA, has however some tremendous drawbacks
compared to REFER:

- The B2BUA breaks the e2e trust model and security, such as S/MIME. How
many B2BUAs would one have to trust in an interdomain call?

- The B2BUA keeps state and customer information. If attacked, loss of
customer privacy can be a huge liability for the ISP. And an unending source
of spam (being criminalized soon by legislators) and telemarketing calls for
the unhappy users.

- Scalability, single point of failure, etc. 

Turns actually the ISP into some sort of large PBX. Is this a good idea on
the Internet? Add Jiri's and Jonathan's comments to the above. I will take
REFER any time instead. 

What do the people on the list think about this? If anyone cares...

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of David Schwartz
> Sent: Wednesday, July 30, 2003 3:11 AM
> To: 'sipping@ietf.org'
> Subject: [Sipping] Re: Update to 3pcc
> 
> Hi Jiri
> 
> REFER is good solution if you assume that either there are no 
> billing issues, or alternatively, that the endpoint/s handle 
> the billing issues. If this is not the case, for example, in 
> an application where a centralized controller is setting up a 
> call to two PSTN numbers, then you will need a centralized 
> billing entity and REFER will not suffice.
> 
> David
> 
> -----Original Message-----
> From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
> Sent: Mon, July 28, 2003 7:00 PM
> To: sipping@ietf.org
> Subject: Sipping digest, Vol 1 #1144 - 2 msgs
> 
> 
> Send Sipping mailing list submissions to
> 	sipping@ietf.org
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	https://www1.ietf.org/mailman/listinfo/sipping
> or, via email, send a message with subject or body 'help' to
> 	sipping-request@ietf.org
> 
> You can reach the person managing the list at
> 	sipping-admin@ietf.org
> 
> When replying, please edit your Subject line so it is more 
> specific than "Re: Contents of Sipping digest..."
> 
> 
> Today's Topics:
> 
>    1. Re: Diversion header (Cullen Jennings)
>    2. Re: Update to 3pcc (Jiri Kuthan)
> 
> --__--__--
> 
> Message: 1
> Date: Sun, 27 Jul 2003 19:05:10 -0700
> Subject: Re: [Sipping] Diversion header
> From: Cullen Jennings <fluffy@cisco.com>
> To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
> 
> 
> I assume your question is there some way for  for the final 
> destination that receives the call to know the who the 1st 
> callee was that had some device in the network redirect the 
> call elsewhere.
> 
> The answer is YES - there has been a lot of confusion over 
> this so think about my answer carefully - it will seem 
> counter intuitive at first. The answer is the "To" header but 
> it will take me a second to explain why.
> 
> Imagine case 1 where A send a call to proxy B,  that sent to 
> to C. Now C sent a 302 back B and B then forked the call to 
> D. The To header in the message D receives will still have 
> the To set to C. C is the element that sent the first 302.
> 
> Now imagine a case (like your question) where there is not a 
> recursive proxy. The 302 will go all the way back to A. Now A 
> forms a new invite to but it can leave the To header set to 
> the same thing it was in the original message (C in this 
> case). When the message arrives at D, the To header will be set to C.
> 
> If D was a voice mail system, it might be able to know to act 
> on behalf of C instead of some other entity.
> 
> There are more complex things that relate to your request. 
> Mary Barnes has been moving the history work along 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-his
> tory-04.txt
> 
> There is some work on QSIG mappings with diversion
> http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2
> sip-diversion-
> 00.txt
> 
> There is a draft-levy-sip-diversion which I (and many others) 
> believe has a pretty much zero chance of being accepted or 
> becoming a RFC. This work has been superceded by the history 
> work. Quite a few people do not support the history work and 
> it is moving very slowly. Don't expect any of this to resolve quickly.
> 
> Give some careful thought to if the "To" header meets your 
> needs and if it does use - The To header is widely implemented :-)
> 
> 
> Cullen
> 
> 
> 
> 
> On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
> 
> > Is there any SIP header defined in any SIP RFC or Internet 
> Draft that 
> > can include the address of the 1st callee, who redirected the call 
> > using 302,
> in
> > the 2nd outgoing INVITE?
> > 
> > _______________________________________________
> > 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
> > 
> 
> 
> 
> --__--__--
> 
> Message: 2
> Date: Mon, 28 Jul 2003 10:17:51 +0200
> To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping 
> <sipping@ietf.org>
> From: Jiri Kuthan <jiri@iptel.org>
> Subject: Re: [Sipping] Update to 3pcc
> 
> I'm actually not convinced that it is proper to show 
> click-to-dial as a motivating example for use of b2bua and 
> label it BCP.
> 
> Use of REFER seems technically a superior solution worth 
> being called BCP. Roots of the advantages are in a better 
> application of e2e principle. 
> REFER takes no middle-box b2bua -- it delegates signaling to 
> end-points.
> Advantages include:
> - All the pitfalls with media are so straigforward, they don't even
>   need to be described (like now in sessions 8 and 9).  
> - It takes no media server to transmit ringing tone. 
> - There are no SDP compatibility issues on controller's side.
> 
> -jiri
> 
> At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
> >Folks,
> >
> >You may have noticed that I posted an update to the 3pcc spec:
> >
> >http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
> >http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
> >http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
> >
> >The update is based on comments received from IESG while 
> reviewing the
> draft. In particular, there were problems with the usage of 
> 0.0.0.0 to imply that no media nor RTCP should be sent. This 
> IP address is not meant for that purpose. Rather, the 
> .invalid TLD is much better suited. So, I changed the draft 
> to make use of .invalid in flows where the controller doesnt 
> want media to be sent.
> >
> >Now, there are some concerns with that which folks need to 
> be aware of:
> >
> >1. It is at odds with RFC 3264, which actually recommends 
> 0.0.0.0. In 
> >fact,
> the problem is really with RFC 3264, NOT 3pcc. However, 3264 
> is done, and this one isnt. I logged a bug against RFC 3264 
> on this, and we'll need to work on a way to transition folks 
> towards using .invalid and not 0.0.0.0.
> >
> >2. One the premises of 3pcc is that it should be compatible 
> with any UA
> that is RFC 3261 compliant. If a controller uses .invalid in 
> the SDP, I am not sure what a SIP endpoint would do. I don't 
> believe anything in RFC 3264 or in SIP talks about handling 
> the case where there is an issue with a domain name in the 
> SDP. As such, if you use .invalid, there could be interop 
> problems. The document mentions this.
> >
> >Comments or questions welcome.
> >
> >Thanks,
> >Jonathan R.
> >-- 
> >Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> >Chief Technology Officer                    Parsippany, NJ 07054-2711
> >dynamicsoft
> >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
> 
> --
> Jiri Kuthan            http://iptel.org/~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
> 
> End of Sipping Digest
> 
> _______________________________________________
> 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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<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">I'm not sure the =
REFER option will solve everything. For example if a call is connected =
via a pre paid system that</FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">connected the 2 parties with REFER how will it now disconnect the =
call or play some warning message</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"> to the caller</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> if it is =
not</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">actively</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">participating in the call.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Regarding S/MIME - =
I don't know of too many implementations actually using it today in =
real deployment. Many are using TLS and that should work OK with the =
B2B</FONT><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Amir</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">-----Original =
Message-----<BR>
From: Henry Sinnreich [<A =
HREF=3D"mailto:Henry.Sinnreich@mci.com">mailto:Henry.Sinnreich@mci.com</=
A>]<BR>
Sent: Wednesday, July 30, 2003 6:09 PM<BR>
To: 'David Schwartz'; sipping@ietf.org<BR>
Subject: RE: [Sipping] Re: Update to 3pcc</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">This may turn out =
to be a very productive discussion. David Schwartz writes:</FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; an application =
where a centralized controller is setting up a </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; call to two =
PSTN numbers, then you will need a centralized </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; billing entity =
and REFER will not suffice.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">There may be more =
attractive billing collecting alternatives: CDRs in the</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">gateways. If the =
calls are between two IP endpoints, flat rate is a good</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">option.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">The centralized =
controller, AKA B2BUA, has however some tremendous drawbacks</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">compared to =
REFER:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- The B2BUA breaks =
the e2e trust model and security, such as S/MIME. How</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">many B2BUAs would =
one have to trust in an interdomain call?</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- The B2BUA keeps =
state and customer information. If attacked, loss of</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">customer privacy =
can be a huge liability for the ISP. And an unending source</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">of spam (being =
criminalized soon by legislators) and telemarketing calls =
for</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">the unhappy =
users.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">- Scalability, =
single point of failure, etc. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Turns actually the =
ISP into some sort of large PBX. Is this a good idea on</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">the Internet? Add =
Jiri's and Jonathan's comments to the above. I will take</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">REFER any time =
instead. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">What do the people =
on the list think about this? If anyone cares...</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Thanks, =
Henry</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; -----Original =
Message-----</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; From: =
sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
] </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; On Behalf Of =
David Schwartz</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Sent: =
Wednesday, July 30, 2003 3:11 AM</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; To: =
'sipping@ietf.org'</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Subject: =
[Sipping] Re: Update to 3pcc</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Hi =
Jiri</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; REFER is good =
solution if you assume that either there are no </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; billing =
issues, or alternatively, that the endpoint/s handle </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the billing =
issues. If this is not the case, for example, in </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; an application =
where a centralized controller is setting up a </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; call to two =
PSTN numbers, then you will need a centralized </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; billing entity =
and REFER will not suffice.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
David</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; -----Original =
Message-----</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; From: =
sipping-request@ietf.org [<A =
HREF=3D"mailto:sipping-request@ietf.org">mailto:sipping-request@ietf.org=
</A>]</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Sent: Mon, =
July 28, 2003 7:00 PM</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; To: =
sipping@ietf.org</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Subject: =
Sipping digest, Vol 1 #1144 - 2 msgs</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Send Sipping =
mailing list submissions to</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sipping@ietf.org</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; To subscribe =
or unsubscribe via the World Wide Web, visit</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; or, via email, =
send a message with subject or body 'help' to</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sipping-request@ietf.org</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; You can reach =
the person managing the list at</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sipping-admin@ietf.org</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; When replying, =
please edit your Subject line so it is more </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; specific than =
&quot;Re: Contents of Sipping digest...&quot;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Today's =
Topics:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp;&nbsp;&nbsp; 1. Re: Diversion header (Cullen =
Jennings)</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp;&nbsp;&nbsp; 2. Re: Update to 3pcc (Jiri =
Kuthan)</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
--__--__--</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Message: =
1</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Date: Sun, 27 =
Jul 2003 19:05:10 -0700</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Subject: Re: =
[Sipping] Diversion header</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; From: Cullen =
Jennings &lt;fluffy@cisco.com&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; To: Romel Khan =
&lt;rkhan@net2phone.com&gt;, &lt;sipping@ietf.org&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; I assume your =
question is there some way for&nbsp; for the final </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; destination =
that receives the call to know the who the 1st </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; callee was =
that had some device in the network redirect the </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; call =
elsewhere.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; The answer is =
YES - there has been a lot of confusion over </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; this so think =
about my answer carefully - it will seem </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; counter =
intuitive at first. The answer is the &quot;To&quot; header but =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; it will take =
me a second to explain why.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Imagine case 1 =
where A send a call to proxy B,&nbsp; that sent to </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; to C. Now C =
sent a 302 back B and B then forked the call to </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; D. The To =
header in the message D receives will still have </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the To set to =
C. C is the element that sent the first 302.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Now imagine a =
case (like your question) where there is not a </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; recursive =
proxy. The 302 will go all the way back to A. Now A </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; forms a new =
invite to but it can leave the To header set to </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the same thing =
it was in the original message (C in this </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; case). When =
the message arrives at D, the To header will be set to C.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; If D was a =
voice mail system, it might be able to know to act </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; on behalf of C =
instead of some other entity.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; There are more =
complex things that relate to your request. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Mary Barnes =
has been moving the history work along </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-his" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sipping=
-req-his</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
tory-04.txt</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; There is some =
work on QSIG mappings with diversion</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-elwell-sippi=
ng-qsig2</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
sip-diversion-</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
00.txt</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; There is a =
draft-levy-sip-diversion which I (and many others) </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; believe has a =
pretty much zero chance of being accepted or </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; becoming a =
RFC. This work has been superceded by the history </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; work. Quite a =
few people do not support the history work and </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; it is moving =
very slowly. Don't expect any of this to resolve quickly.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Give some =
careful thought to if the &quot;To&quot; header meets your </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; needs and if =
it does use - The To header is widely implemented :-)</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
Cullen</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; On 7/21/03 =
8:55, &quot;Romel Khan&quot; &lt;rkhan@net2phone.com&gt; =
wrote:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; Is there =
any SIP header defined in any SIP RFC or Internet </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Draft that =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; can =
include the address of the 1st callee, who redirected the call =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; using =
302,</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; in</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; the 2nd =
outgoing INVITE?</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; =
_______________________________________________</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; 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></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; This list =
is for NEW development of the application of SIP Use </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; =
sip-implementors@cs.columbia.edu for questions on current sip Use =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; =
sip@ietf.org for new developments of core SIP</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt; =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
--__--__--</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Message: =
2</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Date: Mon, 28 =
Jul 2003 10:17:51 +0200</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; To: Jonathan =
Rosenberg &lt;jdrosen@dynamicsoft.com&gt;, sipping </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&lt;sipping@ietf.org&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; From: Jiri =
Kuthan &lt;jiri@iptel.org&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Subject: Re: =
[Sipping] Update to 3pcc</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; I'm actually =
not convinced that it is proper to show </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; click-to-dial =
as a motivating example for use of b2bua and </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; label it =
BCP.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Use of REFER =
seems technically a superior solution worth </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; being called =
BCP. Roots of the advantages are in a better </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; application of =
e2e principle. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; REFER takes no =
middle-box b2bua -- it delegates signaling to </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
end-points.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Advantages =
include:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; - All the =
pitfalls with media are so straigforward, they don't even</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt;&nbsp;&nbsp; =
need to be described (like now in sessions 8 and 9).&nbsp; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; - It takes no =
media server to transmit ringing tone. </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; - There are no =
SDP compatibility issues on controller's side.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
-jiri</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; At 07:51 PM =
7/8/2003, Jonathan Rosenberg wrote:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;Folks,</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;You may =
have noticed that I posted an update to the 3pcc spec:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;<A =
HREF=3D"http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt" =
TARGET=3D"_blank">http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-=
04.txt</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;<A =
HREF=3D"http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html" =
TARGET=3D"_blank">http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-=
04.html</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;<A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.t=
xt" TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-sip=
ping-3pcc-04.txt</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;The update =
is based on comments received from IESG while </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; reviewing =
the</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; draft. In =
particular, there were problems with the usage of </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; 0.0.0.0 to =
imply that no media nor RTCP should be sent. This </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; IP address is =
not meant for that purpose. Rather, the </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; .invalid TLD =
is much better suited. So, I changed the draft </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; to make use of =
.invalid in flows where the controller doesnt </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; want media to =
be sent.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;Now, there =
are some concerns with that which folks need to </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; be aware =
of:</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;1. It is =
at odds with RFC 3264, which actually recommends </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; 0.0.0.0. In =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;fact,</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the problem is =
really with RFC 3264, NOT 3pcc. However, 3264 </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; is done, and =
this one isnt. I logged a bug against RFC 3264 </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; on this, and =
we'll need to work on a way to transition folks </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; towards using =
.invalid and not 0.0.0.0.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;2. One the =
premises of 3pcc is that it should be compatible </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; with any =
UA</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; that is RFC =
3261 compliant. If a controller uses .invalid in </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the SDP, I am =
not sure what a SIP endpoint would do. I don't </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; believe =
anything in RFC 3264 or in SIP talks about handling </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; the case where =
there is an issue with a domain name in the </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; SDP. As such, =
if you use .invalid, there could be interop </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; problems. The =
document mentions this.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;Comments =
or questions welcome.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;Thanks,</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;Jonathan =
R.</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;-- =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;Jonathan =
D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;Chief =
Technology =
Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ =
07054-2711</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;dynamicsoft</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;<A =
HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;<A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;_______________________________________________</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;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></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; &gt;This list =
is for NEW development of the application of SIP Use </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;sip-implementors@cs.columbia.edu for questions on current sip Use =
</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
&gt;sip@ietf.org for new developments of core SIP</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; --</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Jiri =
Kuthan&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 <A HREF=3D"http://iptel.org/~jiri/" =
TARGET=3D"_blank">http://iptel.org/~jiri/</A>&nbsp; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
--__--__--</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; 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></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; This list is =
for NEW development of the application of SIP </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Use =
sip-implementors@cs.columbia.edu for questions on current </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; sip Use =
sip@ietf.org for new developments of core SIP</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; End of Sipping =
Digest</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; =
_______________________________________________</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; 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></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; This list is =
for NEW development of the application of SIP </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; Use =
sip-implementors@cs.columbia.edu for questions on current </FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; sip Use =
sip@ietf.org for new developments of core SIP</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">&gt; </FONT></P>
<BR>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier =
New">_______________________________________________</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">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></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">This list is for =
NEW development of the application of SIP</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Use =
sip-implementors@cs.columbia.edu for questions on current =
sip</FONT></P>

<P ALIGN=3DLEFT><FONT SIZE=3D2 FACE=3D"Courier New">Use sip@ietf.org =
for new developments of core SIP</FONT></P>

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</HTML>
------_=_NextPart_001_01C356B3.583292F0--

_______________________________________________
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 exim@www1.ietf.org  Thu Aug  7 13:33:42 2003
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Date: Wed, 6 Aug 2003 23:18:31 -0700 (PDT)
From: Bernard Aboba <aboba@internaut.com>
To: john.loughney@nokia.com
cc: sipping@ietf.org
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
In-Reply-To: <DADF50F5EC506B41A0F375ABEB32063658F247@esebe023.ntc.nokia.com>
Message-ID: <Pine.LNX.4.53.0308062306320.21030@internaut.com>
References: <DADF50F5EC506B41A0F375ABEB32063658F247@esebe023.ntc.nokia.com>
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> In trying to do any
> meaningful work with RADIUS, I think we need to document
> some of this first.

RFC 2882 does some of this, but it's a moving target, unfortunately.

> Since backoff is not specified in RADIUS, if
> multiple NASes comeup at the same time, they can flood
> the network, causing all sorts of fun things happening.

"Fun" is not the first word that comes to mind to describe several
thousand APs retransmitting once a second with no backoff.  About the only
saving grace was that RADIUS accounting packets are relatively small
so that the result wasn't congestive collapse in the entire network, only
the RADIUS servers.  However, if RTOmin had been 300 ms instead of
1 second, that might have happened too.

> This is one area in which RADIUS would need fixing.

The problem of course is that one cannot "fix" these problems without
upgrading deployed implementations.

And without a heartbeat, it is not clear whether the "fix" can include a
conservative failover algorithm such as the one specified in RFC 3539.

So overall, we need to be realistic about what is achievable.  It's
probably too ambitious a goal to try to truly fix RADIUS
transport behavior -- without major surgery about all we can do is remedy
the most egregious mis-behavior.

_______________________________________________
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 exim@www1.ietf.org  Thu Aug  7 13:33:43 2003
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Date: Tue, 29 Jul 2003 08:29:25 -0700 (PDT)
From: Bernard Aboba <aboba@internaut.com>
To: Henry Sinnreich <Henry.Sinnreich@mci.com>
cc: hgs@cs.columbia.edu, sipping@ietf.org,
        "'Dumler, Alexander'" <Alexander.Dumler@telekom.de>,
        "'Liess, Laura'" <Laura.Liess@t-systems.com>,
        "'Wolff, Christian'" <Christian.Wolff@telekom.de>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
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> Yes, these are good proposals. I was wondering though about the
> unreliability of RADIUS AAA, given that ALL Internet dial-up services are
> using RADIUS. Were there any reports of compliants?

Yes, there are. I've personally seen RADIUS clients that do not implement
exponential backoff cause "congestive collapse" of the RADIUS servers
after a system-wide reboot.  If you've got several thousand NAS devices
and they retransmit once a second with no jittering and no backoff, this
will result in sufficient load that few RADIUS servers can handle it on a
sustained basis.

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From: Bernard Aboba <aboba@internaut.com>
To: jdrosen@dynamicsoft.com
cc: sipping@ietf.org
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Thanks for the feedback.

Here is a short summary of the items that have been requested:

1. Sterman draft on digest authentication w/RADIUS.
2. Draft on RADIUS accounting w/SIP.
3. RADIUS Prepaid accounting support.
4. Document on RADIUS transport behavior, to support (2).

There is also a request for enhanced SIM/AKA support, but I'm not sure
this is specifically RADIUS-related.

Anything else?

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From: Bernard Aboba <aboba@internaut.com>
To: Henry Sinnreich <Henry.Sinnreich@mci.com>
cc: hgs@cs.columbia.edu, sipping@ietf.org,
        "'Dumler, Alexander'" <Alexander.Dumler@telekom.de>,
        "'Liess, Laura'" <Laura.Liess@t-systems.com>,
        "'Wolff, Christian'" <Christian.Wolff@telekom.de>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
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> I have to agree with Laura Liess. In my compnay we have a huge installed
> base of RADIUS and I don't know anybody in the carrier community who would
> prefer to replace RADIUS with DIAMETER.

The question under discussion here is whether new RADIUS work is needed in
the IETF, and if so, what.  So far, there seems to be support for
working on the Sterman draft for digest authentication, as well as
the SIP accounting draft, and perhaps a document on RADIUS transport behavior that
would provide more reliable accounting.

Is this sufficient or is something else needed too?

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From: Bernard Aboba <aboba@internaut.com>
To: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
cc: "McCandless, Kevin" <KMcCandless@verisign.com>,
        "'Liess, Laura'" <Laura.Liess@t-systems.com>, hgs@cs.columbia.edu,
        sipping@ietf.org, "Dumler, Alexander" <Alexander.Dumler@telekom.de>,
        "Wolff, Christian" <Christian.Wolff@telekom.de>
Subject: RE: [Sipping] FYI: RADIUS & SIP
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> I remember a different decision back in San Francisco IETF. That was
> RADIUS for SIP Authentication. There were concerns (unresolved, if
> I remember right) about what to do with accounting due to lack of
> reliability in RADIUS.
>
> Also, my testimony -- quite some iptel.org users (mostly from ISP crowd)
> have been asking us for RADIUS. There was no way around it: they
> are much more uncomfortable with changing largely deployed base than
> with claimed RADIUS shortcomings. None has ever asked for Diameter.
>
> -jiri

There is now an I-D discussing RADIUS transport behavior:

http://www.ietf.org/internet-drafts/draft-lior-radius-reliable-accounting-00.txt

The suggestion is that a specification of this nature could address some
of the concerns about RADIUS accounting reliability.  The debate is how
far such a specification could go -- since RADIUS has no "heartbeat"
functionality, it isn't clear that failover functionality equivalent to
RFC 3539 can be added to it:

http://www.ietf.org/rfc/rfc3539.txt

Obviously, RADIUS clients and servers would need to change to be
compatible with any such specification, if it were to be created.  The
question is whether the issues of "changing largely deployed base" would
represent an obstacle to such an effort.

Any opinions?



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cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Everyone seems to be in agreement on the required work (am I dreaming, or
is this really the IETF?)

There has been some discussion of potentially forming a WG to
handle RADIUS-related work (RADIUSEXT?).  The question is whether
SIP-specific items (such as the Sterman draft and the SIP accounting work)
would be appropriate for a new RADIUS-focussed WG, or whether it might be
more appropriate for a SIP-focussed group (such as SIPPING).

Any opinions?

On Tue, 5 Aug 2003, Jiri Kuthan wrote:

> Thanks, I very much agree; we have received identical feedback so far.
>
> -jiri

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Date: Tue, 05 Aug 2003 17:58:14 +0200
To: "McCandless, Kevin" <KMcCandless@verisign.com>,
        "'Liess, Laura'" <Laura.Liess@t-systems.com>, hgs@cs.columbia.edu,
        sipping@ietf.org, aboba@internaut.com
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: RE: [Sipping] FYI: RADIUS & SIP
Cc: "Dumler, Alexander" <Alexander.Dumler@telekom.de>,
        "Wolff, Christian" <Christian.Wolff@telekom.de>
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I remember a different decision back in San Francisco IETF. That was
RADIUS for SIP Authentication. There were concerns (unresolved, if
I remember right) about what to do with accounting due to lack of 
reliability in RADIUS.

Also, my testimony -- quite some iptel.org users (mostly from ISP crowd) 
have been asking us for RADIUS. There was no way around it: they
are much more uncomfortable with changing largely deployed base than
with claimed RADIUS shortcomings. None has ever asked for Diameter.

-jiri

At 04:39 PM 7/29/2003, McCandless, Kevin wrote:
>This thread seems to rehash an old issue and that was DIAMETER or RADIUS for SIP authentication.  As I remember the discussion and tentative recommendation is that SIP would adopt DIAMETER as the  standard for AAA functionality and that DIAMETER servers would interoperate with RADIUS servers.  As I understood it, the companies that have the embedded base of RADIUS servers could continue to use them for SIP and the entities deploying new DIAMETER based servers must be able to interoperate with the RADIUS servers.  
>
>Kevin....
> 
>
>> -----Original Message-----
>> From: Liess, Laura [mailto:Laura.Liess@t-systems.com]
>> Sent: Friday, July 18, 2003 4:44 AM
>> To: hgs@cs.columbia.edu; sipping@ietf.org; aboba@internaut.com
> Cc: Dumler, Alexander; Wolff, Christian
>> Subject: AW: [Sipping] FYI: RADIUS & SIP
>> 
>> 
>> I think most carriers currently use some kind of RADIUS 
>> platform to do user authentication and they would like to 
>> reuse it in the future to authenticate their SIP customers. 
>> My colleagues who are responsible for the RADIUS Platform of 
>> the Deutsche Telekom (CC)are currently on vacation so I could 
>> not check with them now about how to answer this mail, but we 
>> already talked about this issue a number of times and my 
>> strong opinion is that "yes, Deutsche Telekom cares a lot 
>> about SIP and RADIUS". Reusing the existing RADIUS platform 
>> for SIP authentication is a strong requirement for the 
>> development of SIP services within the Deutsche Telekom.  
>> 
>> Laura (T-Systems/Deutsche Telekom Group)  
>> 
>> 
>> >-----Ursprungliche Nachricht-----
>> >Von: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>> >Gesendet: Freitag, 18. Juli 2003 09:45
>> >An: sipping@ietf.org
>> >Betreff: [Sipping] FYI: RADIUS & SIP
>> >
>> >
>> >Bernard Aboba <aboba@internaut.com> would like to know about 
>> >people and 
>> >companies that care about SIP and RADIUS (for authentication). 
>> >There is 
>> >an attempt afoot to rationalize RADIUS-related efforts within 
>> >the IETF, 
>> >but there seems to be a lack of representation from SIP 
>> folks, as per 
>> >the note to the radius-authors mailing list:
>> >
>> >"I was not clear about the need for SIP/RADIUS work. 
>> >Apparently there is
>> >ongoing deployment of I-Ds (e.g. the Sterman draft on RADIUS 
>> attributes
>> >for HTTP Digest Authentication), so perhaps standardization would be
>> >valuable in documenting what is in use. On the other hand, I 
>> >also heard 
>> >that "there
>> >isn't much commercial interest." Note sure what to make of 
>> >this, perhaps
>> >someone can clarify."
>> >
>> >Henning
>> >
>> >
>> >_______________________________________________
>> >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: Bernard Aboba <aboba@internaut.com>
To: Allison Mankin <mankin@psg.com>
cc: sipping@ietf.org
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP 
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> I believe  they'd be more appropriate developed in SIPPING,
> with cross-review from the RADIUS group.  But I think SIPPING needs some
> information on what the RADIUS working group-to-be is expected to do on RADIUS,
> in what timeframe.  From what I've understood, the charter would not primarily
> be applications.  Could you give us more understanding?

Here is a summary of the discussion so far:

Basic RADIUS work

Since RADIUS is running out of attribute space, it was suggested that a
basic attribute extension mechanism (such as one using the reserved
vendorid of 0) be added.

This item could be handled in a 1-2 page document so it should have a
short timeframe (~6 months).

There is interest in specifying how RADIUS handles prepaid accounting.
Requests for this are coming from the SIP community as well as 3GPP2
and folks interested in WLAN prepaid:

http://www.watersprings.org/pub/id/draft-lior-radius-prepaid-extensions-01.txt

My understanding is that 3GPP2 intends to include this in a forthcoming
standard so that the timeframe needs to be set to meet their needs.

In order to address concerns about RADIUS transport behavior and
accounting reliability, there has been interest expressed on a draft on
RADIUS transport behavior. This would presumably address some of the more
egregious problems encountered with existing implementations (such as
lack of backoff or jittering) as well as describing how RADIUS might be
made to work more reliably:

http://www.watersprings.org/pub/id/draft-lior-radius-reliable-accounting-00.txt

Without some fundamental changes to RADIUS (such as addition of
a heartbeat), it is not clear that it will be possible to specify a
demonstrably correct failover algorithm in addition to handling basic
transport response issues (e.g. backoff, jittering, RTT estimation, etc.).

So there is some question about the scope of a potential transport
document (e.g. just basic retransmission behavior or retransmssion +
failover).  On the face of it, it seems that it would require a lot of
changes to make RADIUS as reliable as Diameter:

http://www.ietf.org/rfc/rfc3539.txt

The timeframe of this work could be short (~9 months) assuming that the
scope is containable.

WLAN-related

A number of different organizations (WFA, IEEE, etc.) have been specifying
WLAN-related attributes.  There appears to be interest in standardizing
the most useful ones.  This and some other WLAN-related work is described
here:

http://www.watersprings.org/pub/id/draft-adrangi-radius-issues-in-pwlan-roaming-00.txt

The timeframe for this is hard to estimate, but I'd guess it is ~18 months
or so.

We also had a request from the IEEE 802.11 Liason, Dorothy Stanley, for
work on fast handoff.  Bill Arbaugh at the University of Maryland is
implementing the draft below:

http://www.watersprings.org/pub/id/draft-irtf-aaaarch-handoff-02.txt

However, Bill has requested that this work not be considered as an IETF WG
work item until his research is complete and the usefulness of the
proposed extension can be evaluated.

_______________________________________________
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This list is for NEW development of the application of SIP
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From: Bernard Aboba <aboba@internaut.com>
To: Jiri Kuthan <jiri@iptel.org>
cc: hgs@cs.columbia.edu, sipping@ietf.org
Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
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> There is a believe in the IETF that lack of reliability in RADIUS
> determines this work to be dropped. (Authentication is a different
> story, though.)

The point of this particular discussion is to understand whether
specification of RADIUS transport behavior might be helpful in that
regard, or whether we've have to go further (such as specifying failover
behavior).

Reading RFC 2865, one of the reasons that RADIUS was not made to run over
TCP in the first place was that it was desired that failover occur in a
timeframe prior to connection failure.  RFC 3539 handles this issue by
defining application-layer timers and heartbeats that allow the AAA
application to re-send an accounting packet over another connection before
tearing down a suspect one.

Via the heartbeat mechanism, the AAA client can determine whether the
issue is due to its immediate connection, or something downstream.
Failover only occurs if the immediate connection is found to be suspect,
so failover occurs on a hop-by-hop basis.

RFC 3539 cannot be applied to the RADIUS protocol as it stands because
RADIUS does not support a heartbeat.  As a result, a RADIUS client can
failover even where its immediate proxy is healthy, because of a problem
on a downstream RADIUS server.  Since RADIUS failover is
typically end-to-end, there may be no failover in proxies, even if a
server is not responding to requests proxied to it.

Even if one were to define appropriate RTT/RTO measurements, and use traffic
from the proxy as a demonstration of "liveness", inappropriate failover
can still happen in cases where the time between proxy traffic is greater
than the failover timer.

The end result is that to be able to apply RFC 3539, one would need to add
a heartbeat command to RADIUS.  This is a fairly major step, since it
would require changing both RADIUS client and server implementations.

So the question is:

"If a major protocol change were to be made to RADIUS to improve
reliability, would such a change be deployed on RADIUS clients and
servers and would it be acceptable for the SIP accounting specification
to depend on such a change?"

If the answer is yes, then a RADIUS failover spec  might be worth
discussing further. If no, then it seems to me that a RADIUS failover spec it might
not be worth doing -- it's not considered important enough to get over the deployment hurdles.
That raises the question of whether a dependency even on transport
behavior improvements is possible, since even this would still require
changes on RADIUS clients (though not servers).

Comments?

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From: Bernard Aboba <aboba@internaut.com>
To: Jari Arkko <jarkko@piuha.net>
cc: Henry Sinnreich <Henry.Sinnreich@mci.com>, john.loughney@nokia.com,
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Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
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> Here's what I'd like to have:
>
> 1. Published (RFC) specifications for carrying HTTP
>     Digest over RADIUS. Maybe start from draft-sterman.
> 2. Published specifications for the same thing and
>     a few additional features over DIAMETER. Start from
>     draft-belinchon, but make it less 3GPP-specific.
> 3. As a part of 2, interworking specification
>     between RADIUS & DIAMETER (possibly limited
>     to the scope of features supported under 1).
> 4. Prepaid support module for DIAMETER that can
>     work with any application, such as SIP or network
>     access. This is work in progress, as John noted.
> 5. Optionally, a prepaid module for RADIUS,
>     constructed in a manner which makes translation
>     to 4 possible.

This is a summary of the work that is being proposed or ongoing (in AAA
WG).


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Date: Thu, 31 Jul 2003 06:58:32 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
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I believe this is an excellent proposal!

Though a firm believer in the supremacy of IETF technology, sometimes it
makes sense to import innovation on to the Internet from other areas. The
SIM card is certainly on top of the list for technologies where its creators
had their act well together. 

Thanks, Henry

Henry Sinnreich
MCI
400 International Parkway
Richardson, Texas 75081
USA

Have you disconnected the PBX?
 

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Michael Haberler
> Sent: Wednesday, July 30, 2003 11:55 AM
> To: Jonathan Rosenberg; Bernard Aboba
> Cc: Liess, Laura; hgs@cs.columbia.edu; sipping@ietf.org; 
> Dumler, Alexander; Wolff, Christian
> Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> 
> authenticating SIP users is very much like authenticating 
> mobile users in cellular networks, and the solution developed 
> for GSM - subscriber identity in a smart card - is IMV 
> somthing we would be well advised to carry over to SIP space.
> 
> The utility of SIM cards has been recognized for 
> Internet-side authentication like EAP-SIM based WLAN roaming, 
> in particular as there is an existing billing an 
> authentication machinery which one can plug into from the 
> Internet side. However, SIM cards could also be used in a 
> strictly Internet-only service provider context as well.  
> Plus, SIM-based authentication could be used as a bootstrap 
> mechanism for X.509 certficate distribution and therefore for 
> TLS mutual authentication.
> 
> There is a draft for SIP authentication with Authentication 
> and Key Agreement (AKA, 
> http://www.ietf.org/internet-drafts/draft-torvinen-http-digest
> -aka-v2-00.txt
> and predecessor) which adresses authentication with UMTS SIM 
> (USIM)cards. 
> These have superior security properties compared to plain SIM cards. 
> However, HTTP AKA cannot negotiate down to plain SIM 
> authentication, and very few operators use USIM cards, so if 
> we wait for USIM cards to be ubiquitious we might wait very long.
> 
> Therefore, there is more pressing need for authentication 
> with plain GSM SIM cards (about 900 million cards out 
> there!), and there is currently no standard way of using 
> those at the SIP level - although at  the WLAN level, this 
> has been adressed by EAP/SIM which gets around some of the 
> weaknessess of plain SIM authentication. However, the HTTP 
> EAP Digest draft apparently fell through, so we cannot 
> combine HTTP EAP and EAP/SIM.
> 
> So the area of work I see a need for is retrofitting EAP/SIM 
> authentication flows into Digest authentication to create a 
> strong interoperable HTTP SIM authentication method, and 
> appropriate RADIUS support to back it up. Also, the RADIUS 
> support behind HTTP AKA needs fleshing out.
> 
> my vision is to have both WLAN/LAN access and SIP 
> authenticated through the same SIM card. I see quite a market 
> potential for such a feature set.
> 
> -Michael Haberler
> 
> 
> 
> At 14:12 29.07.2003 -0400, Jonathan Rosenberg wrote:
> 
> >I know of several carriers with RADIUS infrastructures that 
> want to use 
> >them to support SIP. They are currently using expired I-Ds and 
> >proprietary attributes. As a result, I would strongly 
> advocate work on 
> >standardizing RADIUS usage with SIP.
> >
> >There are two areas of work that I see a need for:
> >
> >1. Digest authentication. The sterman draft
> >(http://www.freeradius.org/radiusd/doc/rfc/draft-sterman-aaa-
> sip-00.txt
> >) is used quite a bit, it seems. It would be nice to standardize on 
> >this.
> >
> >2. Prepaid calling. Many folks are using vendor proprietary radius 
> >extensiosn to support prepaid calling. It would be nice to 
> bring those  
> >forward and standardize on them.
> >
> >-Jonathan R.
> >
> >Bernard Aboba wrote:
> >
> >>Is there some particular set of draft(s) that you are 
> advocating work on?
> >>On Fri, 18 Jul 2003, Liess, Laura wrote:
> >>
> >>>I think most carriers currently use some kind of RADIUS 
> platform to 
> >>>do user authentication and they would like to reuse it in 
> the future 
> >>>to authenticate their SIP customers.
> >>>My colleagues who are responsible for the RADIUS Platform of the 
> >>>Deutsche Telekom (CC)are currently on vacation so I could 
> not check 
> >>>with them now about how to answer this mail, but we already talked 
> >>>about this issue a number of times and my strong opinion is that 
> >>>"yes, Deutsche Telekom cares a lot about SIP and RADIUS". 
> Reusing the 
> >>>existing RADIUS platform for SIP authentication is a strong 
> >>>requirement for the development of SIP services within the 
> Deutsche Telekom.
> >>>
> >>>Laura (T-Systems/Deutsche Telekom Group)
> >>
> >>_______________________________________________
> >>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.                600 Lanidex Plaza
> >Chief Technology Officer                    Parsippany, NJ 07054-2711
> >dynamicsoft
> >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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Arnoud,

During some earlier discussions, I pointed out that ICQ already has
character-at-a-time capability.  It's a very popular instant messaging
client.  I think it's very unfortunate that SIMPLE didn't take that route.
The "all at once" text message is really "old style" in my opinion... not
better than e-mail, really.  OK.. a little.

Character-at-a-time is stressful on the servers, though.  Rather than
processing 100 messages per chat session, for example, it might be 100 * 30
messages.

I think that the best solution would have been to design SIMPLE to use SIP
for session establishment and simply open an RFC 2793-type of RTP stream.  I
understand that there was a desire to be able to pass formatting information
and other content, which was not precisely compatible with RFC 2793.  Even
so, I think that if we were to treat the RFC 2793 stream as a stream of
HTML, that might have satisfied most people.  I don't know what other things
people expected to get from passing thing as blocks of text.

I really liked ICQ's character-at-a-time mode and I think others would have
found it preferable.  It allows for much faster message exchange, as one can
start typing immediately after the other person finishes.. or even start out
right away.  It's possible to have a real interactive argument with ICQ :-)

Paul

----- Original Message ----- 
From: "A vWijk" <A.vWijk@viataal.nl>
To: <fluffy@cisco.com>; <sipping@ietf.org>; <gunnar.hellstrom@omnitor.se>
Cc: <barry.dingle@bigfoot.com.au>; <hwildfeu@cisco.com>;
<mgarakan@cisco.com>; <oran@cisco.com>; <rkumar@cisco.com>;
<rohan@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
<Henry.Sinnreich@mci.com>; <andreas.piirimets@omnitor.se>;
<Erik.Zetterstrom@omnitor.se>; <paulej@packetizer.com>;
<Guido.Gybels@rnid.org.uk>; <James.Hamlin@rnid.org.uk>;
<mike.spanner@rnid.org.uk>; <gv@trace.wisc.edu>; <trefor@trefor.plus.com>
Sent: Monday, July 28, 2003 9:59 AM
Subject: Betr.: Real time streaming text conversations


> Hello Cullen.
>
> Good to "hear" that you agree. :-)
>
> About IM, I agree. character by character is not going to kill IM.
> I already told a few IM people that such real-time streaming text can be
added as a "chat" option.
> So that I can for example fire off a few IM messages about something. And
the other side reacts and we then both enter chat mode to discuss the
proceedings.
> Will this become popular? Oh yes, think of the good old days before HTML.
What was one of the most used functions of the web? IRC and ntalk/talk.
>
> About the sipdev meeting. Well yes, a MUST is very good. But I see it as
the beginning. The first step. We still have a lot of work to do. And a
paper explaining to phone manufactures that implementing this requirement
gives them a competitive advantage because everyone is going to want it. Is
a very good idea.
> Do you mean a kind of Internet draft or white paper?
> But I am quite willing to help and I am positive that others will help
too.
>
> Greetz
>
> Arnoud
>
> PS Currently a BCP is being written, though that is more about
implementation itself.
>
>
>
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
>
> >>> Cullen Jennings <fluffy@cisco.com> 24-07-03 23:47 >>>
> On 7/13/03 5:08, "A vWijk" <A.vWijk@viataal.nl> wrote:
>
> > I just want to give them a very clear message how important real-time
> > streaming text conversation will be for hundred of millions of users.
And
> > while it will be vitally important for the hearing and speech impaired,
other
> > users will also benefit from this capability. (RFC 2793 text in every
SIP
> > telephony device, regardless presence of a keyboard or screen).
> >
> > greetz
> >
> > Arnoud van Wijk
> >
>
> I think this is a very good message to get out. The hearing and speech
> impaired community has significant experience with using text messaging to
> communicate as quickly as possible. This community resoundingly says that
> character by character improves the communication experience. I can not
see
> any reason that this same fact will not apply to anyone using IM.
>
> The hearing and speech impaired community should make sure that the SIMPLE
> and XMPP WG understand that IM systems should include an option of
character
> by character sending and display of messages.
>
> The details of how this is implemented are really less important (yah I
know
> the folks like 3GPP will need a bandwidth efficient way of doing this).
>
> Cullen
>
> PS. IMHO, The IETF putting a MUST in some random document is less likely
to
> make this happen than a bunch of highly qualified people writing a
document
> that explains to phone manufactures that implementing this requirement
gives
> them a competitive advantage because everyone is going to want it.
>
>
>


_______________________________________________
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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: Mon, 28 Jul 2003 21:18:05 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Re: items for vor voting in the BOF
In-reply-to: <002501c34edf$3ff12e30$2a1b550a@cisco.com>
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Paul, finally, we disagree!

We have 3-4 SIP phones in my house at all times and we are using all of them
in various rooms. We would not even dream touching a VoIP phone that does
not have a display and without its own web page where we can see the address
book and dial from the PC or from some another web absed personal call
manager.

I would certainly not attempt to sell any friend or family member a SIP
phone withou display. Depressing thought.

Nor would we ever by a mobile phone without display.

Cheers :-)

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Paul E. Jones
> Sent: Sunday, July 20, 2003 11:52 AM
> To: Rohan Mahy
> Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; Henry 
> Sinnreich; Gunnar Hellstrom; Mike Spanner; 
> andreas.piirimets@omnitor.se; trefor@trefor.plus.com; 'Rajesh 
> Kumar'; marcelo@envilogg.se; jorgen.bjorkner@hotsip.com; 
> James Hamlin; Erik.Zetterstrom@omnitor.se; 
> hwildfeu@cisco.com; barry.dingle@bigfoot.com.au; 
> mgarakan@cisco.com; oran@cisco.com; sipping@ietf.org; Andrea Saks
> Subject: Re: [Sipping] Re: items for vor voting in the BOF
> 
> Rohan,
> 
> I offered up a few questions to folks that I hoped people 
> would have voted on.  The questions I had was whether or not 
> there should be a requirement to provide *some* interface to 
> legacy TTY devices.  Whether it's RJ-11 or 2.5mm makes no 
> difference.  However, I think it should be a requirement that 
> pass-through work at the very least.  If this was agreeable, 
> I could certainly work to craft specific text for such a requirement.
> 
> I do not have stats on how many phones do not have screens.  
> However, note that requirement 1 also requires a keyboard, 
> though that's never mentioned.
> One could say that text could be entered via some web 
> interface, but doing so is a bit hard in a hotel room or at a 
> gas station on the side of the road.  Regardless, I've said 
> before and I'll say it again: I do not expect that most of 
> the IP phones that will go into mass production will have a 
> suitable screen for text communication.  Most of today's IP 
> phones have screens, because it is a way to catch people's 
> eyes.  However, there are also many that do not.  Would you 
> consider Cisco's ATA, for example, an IP phone?  I think it 
> qualifies and there is no screen.  There are a number of 
> service providers deploying that and similar devices from 
> other manufacturers.  None of those have displays.
> 
> I assert that in most enterprise environments, the display is 
> of very little value.  It's interesting at first, but it's 
> not necessary to have a fancy
> screen: just something simple to control hold, transfer, 
> conferencing, etc., is enough.  In fact, I'd be happy with 
> buttons that did that.
> 
> At home, I think it's even less of a requirement to have a 
> nice-size display.  Recognizing that many phones today have a 
> display for caller ID, etc., such displays are not suitable 
> for text communication.  Requirement 1 is asking for a nice 
> display... and I think you'll find a much smaller percentage 
> of those... especially as they become a commodity item.
> 
> Paul
> 
> ----- Original Message -----
> From: "Rohan Mahy" <rohan@cisco.com>
> To: "Paul E. Jones" <paulej@packetizer.com>
> Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "Henry Sinnreich"
> <Henry.Sinnreich@mci.com>; "Gunnar Hellstrom" 
> <gunnar.hellstrom@omnitor.se>; "Mike Spanner" 
> <mike.spanner@rnid.org.uk>; <andreas.piirimets@omnitor.se>; 
> <trefor@trefor.plus.com>; "'Rajesh Kumar'" 
> <rkumar@cisco.com>; <marcelo@envilogg.se>; 
> <jorgen.bjorkner@hotsip.com>; "James Hamlin"
> <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>; 
> <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>; 
> <mgarakan@cisco.com>; <oran@cisco.com>; <sipping@ietf.org>; 
> "Andrea Saks" <asaks@waitrose.com>
> Sent: Saturday, July 19, 2003 11:14 PM
> Subject: Re: [Sipping] Re: items for vor voting in the BOF
> 
> 
> > Hi Paul,
> >
> > So it sounds like you wanted an additional requirement for 
> pass-through
> > via a 2.5 mm jack for phones that do not support RFC2793?  If I
> > understood this correctly, how would you word the requirement?
> >
> > Also, do you have any data on how many phones currently 
> exist (number
> > or percentage) for which requirement 1 would not apply? I 
> suspect that
> > the number of phones not covered by the first requirement 
> is actually
> > quite small.
> >
> > thanks,
> > -rohan
> >
> > On Friday, July 18, 2003, at 05:04 PM, Paul E. Jones wrote:
> >
> > > Guido,
> > >
> > > I wish I could have attended, but I was attending a TDI conference
> > > here in
> > > the US.  I couldn't be in both places at the same time, 
> though I wish I
> > > could.  I was disheartened that the questions were not 
> presented as I
> > > proposed.  You should know that, supposedly, work gets 
> done over the
> > > e-mail
> > > reflector, not only in the meeting, so not being there 
> should not have
> > > lessened my input on the matter.
> > >
> > > Looking at the results of the ad-hoc reported by Rohan, I see:
> > >   1) *If* an IP phone supports a display (and keyboard??), it
> > >      must support RFC 2793
> > >   2) IP phones *should* provide a way to input text
> > >   3) IP phones *may* provide an RJ-11 interface and *must*
> > >      support pass-through.
> > >
> > > Now, my concerns are several.  It appears that you have 
> absolutely no
> > > requirement for the mass of products that do not sport a 
> display to
> > > support
> > > text in either relay or pass-through mode.  Further, there is no
> > > mention of
> > > a 2.5mm jack being an option.  Question 3, as I 
> mentioned, "pressured"
> > > folks
> > > more than it should.  There should have been no reason to require
> > > support of
> > > pass-through, at the very least.
> > >
> > > I told you that my questions were to ensure that we at least get
> > > support for
> > > text on every phone in one way or another.  It looks like 
> you lost.
> > > Company
> > > X is now free to build an IP phone without any text 
> support.  I'm not
> > > really
> > > happy about that and I certainly don't see why you should be.
> > >
> > > Paul
> > >
> > > ----- Original Message -----
> > > From: "Guido Gybels" <ggybels@webclient.eu.blackberry.net>
> > > To: "Paul E. Jones" <paulej@packetizer.com>; <gv@trace.wisc.edu>
> > > Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>; "Gunnar 
> Hellstrom"
> > > <gunnar.hellstrom@omnitor.se>; "Guido Gybels"
> > > <Guido.Gybels@rnid.org.uk>;
> > > "Mike Spanner" <mike.spanner@rnid.org.uk>;
> > > <andreas.piirimets@omnitor.se>;
> > > <gv@trace.wisc.edu>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > <jorgen.bjorkner@hotsip.com>;
> > > "James Hamlin" <James.Hamlin@rnid.org.uk>;
> > > <Erik.Zetterstrom@omnitor.se>;
> > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > <mgarakan@cisco.com>;
> > > <oran@cisco.com>; <rohan@cisco.com>; <sipping@ietf.org>
> > > Sent: Friday, July 18, 2003 2:31 AM
> > > Subject: [Sipping] Re: items for vor voting in the BOF
> > >
> > >
> > >> Paul,
> > >>
> > >> <<While I guess it might be a bit late now>>
> > >>
> > >> The discussion on the sipdev document took place 
> yesterday, and if you
> > > wanted to make a point you were free to attend it. You'll see the
> > > results
> > > reflected in the next version of the document, and I can 
> say that both
> > > Arnoud and myself believe this result to be extremely 
> sensible as well
> > > as in
> > > the best interest of all stakeholders.
> > >> I also suggest this no longer belongs here on the 
> mailing list until
> > >> a new
> > > issue would arise. I have stopped replying to your emails 
> a while ago
> > > as I
> > > see no point in continuing the discussion (and swamping 
> the list) with
> > > someone who, despite being a native English speaker, 
> seems unable to
> > > understand what we are saying and more importantly deliberately
> > > misrepresents my statements or attributes words to my 
> mouth that I did
> > > not
> > > speak.
> > >>
> > >> <<would guess most people will say "yes" to, while I 
> know you'd say
> > >> "no"
> > > to
> > >> it.>>
> > >>
> > >> This being a nice example of that.
> > >>
> > >>
> > >> I believe the IETF community has shown great wisdom and a true
> > > understanding of what is at stake here. We have now got a 
> solid path
> > > to the
> > > future, with a clear strategy to get rid of obsolete and 
> unsatisfactory
> > > solutions allowing a gradual move of deaf, hard of 
> hearing and speech
> > > impaired people into mainstream technology. That will of 
> course take
> > > time
> > > during which there will have to be legacy support and all the
> > > instruments
> > > are put in place to do exactly that.
> > >>
> > >> Guido
> > >>
> > >> Guido Gybels
> > >> Director of New Technologies
> > >>
> > >> RNID, 19-23 Featherstone Street
> > >> London EC1Y 8SL, UK
> > >> Tel +44(0)20-7294 3713
> > >> Fax +44(0)20-7296 8069
> > >>
> > >> _______________________________________________
> > >> 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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Message-ID: <023b01c35642$f2aa1950$0401a8c0@packetizer.org>
From: "Paul E. Jones" <paulej@packetizer.com>
To: "Henry Sinnreich" <Henry.Sinnreich@mci.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Cc: <guido.gybels@rnid.org.uk>, <gv@trace.wisc.edu>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        "'Mike Spanner'" <mike.spanner@rnid.org.uk>,
        <andreas.piirimets@omnitor.se>, <trefor@trefor.plus.com>,
        "'Rajesh Kumar'" <rkumar@cisco.com>, <marcelo@envilogg.se>,
        <jorgen.bjorkner@hotsip.com>,
        "'James Hamlin'" <James.Hamlin@rnid.org.uk>,
        <Erik.Zetterstrom@omnitor.se>, <hwildfeu@cisco.com>,
        <barry.dingle@bigfoot.com.au>, <mgarakan@cisco.com>, <oran@cisco.com>,
        <sipping@ietf.org>, "'Andrea Saks'" <asaks@waitrose.com>
References: <0HIR003E1L2B5Q@dgismtp04.wcomnet.com>
Subject: Re: [Sipping] Re: items for vor voting in the BOF
Date: Tue, 29 Jul 2003 22:33:18 -0400
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Henry,

The same would likely be said by a boss or busy secretary in many companies.
Even so, the bulk of the phones in most enterprises are just little dumb
phones with no real features (perhaps feature keys... not much else).  I
don't expect that to change substantially, even with the introduction of
VoIP.  After all, more and more capability can be moved onto the PC.  Why
add hundreds of dollars in value to an IP phone when all of the "cool"
features that you want can be placed on the PC (and updated much more
easily)?  The address book, instant messaging, etc., can all be located on
the PC.  The PC can interface with the server and with your phone.  The
phone needs little functionality of its own, really.

There are people who buy fancy phones for their homes.  Even so, there will
be a substantial number of people who just don't care and will buy the
cheapest thing they can find.  I'm sure that manufacturers will try to cut
costs at every opportunity and screens are a target.  Should we not plan on
it?

Paul

----- Original Message ----- 
From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Rohan Mahy'"
<rohan@cisco.com>
Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry Sinnreich'"
<Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
<gunnar.hellstrom@omnitor.se>; "'Mike Spanner'" <mike.spanner@rnid.org.uk>;
<andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
<rkumar@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
"'James Hamlin'" <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
<hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>; <mgarakan@cisco.com>;
<oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'" <asaks@waitrose.com>
Sent: Monday, July 28, 2003 10:18 PM
Subject: RE: [Sipping] Re: items for vor voting in the BOF


> Paul, finally, we disagree!
>
> We have 3-4 SIP phones in my house at all times and we are using all of
them
> in various rooms. We would not even dream touching a VoIP phone that does
> not have a display and without its own web page where we can see the
address
> book and dial from the PC or from some another web absed personal call
> manager.
>
> I would certainly not attempt to sell any friend or family member a SIP
> phone withou display. Depressing thought.
>
> Nor would we ever by a mobile phone without display.
>
> Cheers :-)
>
> Henry
>
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of Paul E. Jones
> > Sent: Sunday, July 20, 2003 11:52 AM
> > To: Rohan Mahy
> > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; Henry
> > Sinnreich; Gunnar Hellstrom; Mike Spanner;
> > andreas.piirimets@omnitor.se; trefor@trefor.plus.com; 'Rajesh
> > Kumar'; marcelo@envilogg.se; jorgen.bjorkner@hotsip.com;
> > James Hamlin; Erik.Zetterstrom@omnitor.se;
> > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au;
> > mgarakan@cisco.com; oran@cisco.com; sipping@ietf.org; Andrea Saks
> > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> >
> > Rohan,
> >
> > I offered up a few questions to folks that I hoped people
> > would have voted on.  The questions I had was whether or not
> > there should be a requirement to provide *some* interface to
> > legacy TTY devices.  Whether it's RJ-11 or 2.5mm makes no
> > difference.  However, I think it should be a requirement that
> > pass-through work at the very least.  If this was agreeable,
> > I could certainly work to craft specific text for such a requirement.
> >
> > I do not have stats on how many phones do not have screens.
> > However, note that requirement 1 also requires a keyboard,
> > though that's never mentioned.
> > One could say that text could be entered via some web
> > interface, but doing so is a bit hard in a hotel room or at a
> > gas station on the side of the road.  Regardless, I've said
> > before and I'll say it again: I do not expect that most of
> > the IP phones that will go into mass production will have a
> > suitable screen for text communication.  Most of today's IP
> > phones have screens, because it is a way to catch people's
> > eyes.  However, there are also many that do not.  Would you
> > consider Cisco's ATA, for example, an IP phone?  I think it
> > qualifies and there is no screen.  There are a number of
> > service providers deploying that and similar devices from
> > other manufacturers.  None of those have displays.
> >
> > I assert that in most enterprise environments, the display is
> > of very little value.  It's interesting at first, but it's
> > not necessary to have a fancy
> > screen: just something simple to control hold, transfer,
> > conferencing, etc., is enough.  In fact, I'd be happy with
> > buttons that did that.
> >
> > At home, I think it's even less of a requirement to have a
> > nice-size display.  Recognizing that many phones today have a
> > display for caller ID, etc., such displays are not suitable
> > for text communication.  Requirement 1 is asking for a nice
> > display... and I think you'll find a much smaller percentage
> > of those... especially as they become a commodity item.
> >
> > Paul
> >
> > ----- Original Message -----
> > From: "Rohan Mahy" <rohan@cisco.com>
> > To: "Paul E. Jones" <paulej@packetizer.com>
> > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "Henry Sinnreich"
> > <Henry.Sinnreich@mci.com>; "Gunnar Hellstrom"
> > <gunnar.hellstrom@omnitor.se>; "Mike Spanner"
> > <mike.spanner@rnid.org.uk>; <andreas.piirimets@omnitor.se>;
> > <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > <jorgen.bjorkner@hotsip.com>; "James Hamlin"
> > <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
> > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > <mgarakan@cisco.com>; <oran@cisco.com>; <sipping@ietf.org>;
> > "Andrea Saks" <asaks@waitrose.com>
> > Sent: Saturday, July 19, 2003 11:14 PM
> > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> >
> >
> > > Hi Paul,
> > >
> > > So it sounds like you wanted an additional requirement for
> > pass-through
> > > via a 2.5 mm jack for phones that do not support RFC2793?  If I
> > > understood this correctly, how would you word the requirement?
> > >
> > > Also, do you have any data on how many phones currently
> > exist (number
> > > or percentage) for which requirement 1 would not apply? I
> > suspect that
> > > the number of phones not covered by the first requirement
> > is actually
> > > quite small.
> > >
> > > thanks,
> > > -rohan
> > >
> > > On Friday, July 18, 2003, at 05:04 PM, Paul E. Jones wrote:
> > >
> > > > Guido,
> > > >
> > > > I wish I could have attended, but I was attending a TDI conference
> > > > here in
> > > > the US.  I couldn't be in both places at the same time,
> > though I wish I
> > > > could.  I was disheartened that the questions were not
> > presented as I
> > > > proposed.  You should know that, supposedly, work gets
> > done over the
> > > > e-mail
> > > > reflector, not only in the meeting, so not being there
> > should not have
> > > > lessened my input on the matter.
> > > >
> > > > Looking at the results of the ad-hoc reported by Rohan, I see:
> > > >   1) *If* an IP phone supports a display (and keyboard??), it
> > > >      must support RFC 2793
> > > >   2) IP phones *should* provide a way to input text
> > > >   3) IP phones *may* provide an RJ-11 interface and *must*
> > > >      support pass-through.
> > > >
> > > > Now, my concerns are several.  It appears that you have
> > absolutely no
> > > > requirement for the mass of products that do not sport a
> > display to
> > > > support
> > > > text in either relay or pass-through mode.  Further, there is no
> > > > mention of
> > > > a 2.5mm jack being an option.  Question 3, as I
> > mentioned, "pressured"
> > > > folks
> > > > more than it should.  There should have been no reason to require
> > > > support of
> > > > pass-through, at the very least.
> > > >
> > > > I told you that my questions were to ensure that we at least get
> > > > support for
> > > > text on every phone in one way or another.  It looks like
> > you lost.
> > > > Company
> > > > X is now free to build an IP phone without any text
> > support.  I'm not
> > > > really
> > > > happy about that and I certainly don't see why you should be.
> > > >
> > > > Paul
> > > >
> > > > ----- Original Message -----
> > > > From: "Guido Gybels" <ggybels@webclient.eu.blackberry.net>
> > > > To: "Paul E. Jones" <paulej@packetizer.com>; <gv@trace.wisc.edu>
> > > > Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>; "Gunnar
> > Hellstrom"
> > > > <gunnar.hellstrom@omnitor.se>; "Guido Gybels"
> > > > <Guido.Gybels@rnid.org.uk>;
> > > > "Mike Spanner" <mike.spanner@rnid.org.uk>;
> > > > <andreas.piirimets@omnitor.se>;
> > > > <gv@trace.wisc.edu>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > <jorgen.bjorkner@hotsip.com>;
> > > > "James Hamlin" <James.Hamlin@rnid.org.uk>;
> > > > <Erik.Zetterstrom@omnitor.se>;
> > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > <mgarakan@cisco.com>;
> > > > <oran@cisco.com>; <rohan@cisco.com>; <sipping@ietf.org>
> > > > Sent: Friday, July 18, 2003 2:31 AM
> > > > Subject: [Sipping] Re: items for vor voting in the BOF
> > > >
> > > >
> > > >> Paul,
> > > >>
> > > >> <<While I guess it might be a bit late now>>
> > > >>
> > > >> The discussion on the sipdev document took place
> > yesterday, and if you
> > > > wanted to make a point you were free to attend it. You'll see the
> > > > results
> > > > reflected in the next version of the document, and I can
> > say that both
> > > > Arnoud and myself believe this result to be extremely
> > sensible as well
> > > > as in
> > > > the best interest of all stakeholders.
> > > >> I also suggest this no longer belongs here on the
> > mailing list until
> > > >> a new
> > > > issue would arise. I have stopped replying to your emails
> > a while ago
> > > > as I
> > > > see no point in continuing the discussion (and swamping
> > the list) with
> > > > someone who, despite being a native English speaker,
> > seems unable to
> > > > understand what we are saying and more importantly deliberately
> > > > misrepresents my statements or attributes words to my
> > mouth that I did
> > > > not
> > > > speak.
> > > >>
> > > >> <<would guess most people will say "yes" to, while I
> > know you'd say
> > > >> "no"
> > > > to
> > > >> it.>>
> > > >>
> > > >> This being a nice example of that.
> > > >>
> > > >>
> > > >> I believe the IETF community has shown great wisdom and a true
> > > > understanding of what is at stake here. We have now got a
> > solid path
> > > > to the
> > > > future, with a clear strategy to get rid of obsolete and
> > unsatisfactory
> > > > solutions allowing a gradual move of deaf, hard of
> > hearing and speech
> > > > impaired people into mainstream technology. That will of
> > course take
> > > > time
> > > > during which there will have to be legacy support and all the
> > > > instruments
> > > > are put in place to do exactly that.
> > > >>
> > > >> Guido
> > > >>
> > > >> Guido Gybels
> > > >> Director of New Technologies
> > > >>
> > > >> RNID, 19-23 Featherstone Street
> > > >> London EC1Y 8SL, UK
> > > >> Tel +44(0)20-7294 3713
> > > >> Fax +44(0)20-7296 8069
> > > >>
> > > >> _______________________________________________
> > > >> 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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As noted at IETF57, I'm picking up editorship of the torture-test
document.

One of the things I want to do is disambiguate the printed text
examples. There are cases now where the torture consists of having very
long text fields. For example:

CSeq:
239sdfadfnm34r5u09sdfawetrq3o4t5nodifnaoihesfoiw34foisdfeijjnaosdfj4iosdfnaehnaasoo34ijdfn5kl3neasdfvnazs

which gets broken into a format that is not valid for SIP when you try
to render withing the 72 column limit of internet drafts. (Its probably
also broken badly in this message by many mail readers).

I propose making such breaks explicit as follows:

If I needed to break the following (short for clarity) header field:

Foo: areallybigtoken

I'll render it as

Foo: areally<break>
     </break>bigtoken

or

Foo: areally<break>
     </break>big<break>
     </break>token


Objections? Alternative suggestions?

Additionally, I'm considering including an appendix with an
ascii-armored container (a zip-file or compressed tarball
perhaps) with each message (so that we can include some
torturous whitespace and non-ASCII range UTF8 tests.

RjS




_______________________________________________
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: "Krishnan Padmanabhan, Packet Strategies" <krishnan@packetstrategies.com>
To: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
Cc: <sipping@ietf.org>
References: <0HIR003E1L2B5Q@dgismtp04.wcomnet.com>
Subject: Re: [Sipping] Re: items for vor voting in the BOF
Date: Thu, 7 Aug 2003 16:15:58 -0400
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I wonder how representative the tastes and preferences in your
household are of those of the consumers of SIP service today. We do
know that the 40,000+ lines of Vonage's SIP service are based on the
Cisco ATA, with at most a Caller ID display on their phones. Ditto
for the consumers for VoicePulse and Packet8. Delta3's Iconnecthere
SIP service permits consumers to use any SIP phone they want, but
provides technical support only for the ATA.
OTOH Cisco recently announced their 2 millionth IP Phone shipment.
These are probably more representative of the phones in your
household (assuming they didn't include the ATAs in this milestone),
but it appears Cisco sold the overwhelming majority of these phones
to enterprises.
Also, the only sub-$100 SIP phone I am aware of today is Grandstream
Networks BudgetOne-101, which has only a Caller-ID display.

Krishnan

----- Henry Sinnreich wrote ----- 
From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
To: "'Paul E. Jones'" <paulej@packetizer.com> et al.
Sent: Monday, July 28, 2003 10:18 PM
Subject: RE: [Sipping] Re: items for vor voting in the BOF


> Paul, finally, we disagree!
>
> We have 3-4 SIP phones in my house at all times and we are using
all of them
> in various rooms. We would not even dream touching a VoIP phone
that does
> not have a display and without its own web page where we can see
the address
> book and dial from the PC or from some another web absed personal
call
> manager.
>
> I would certainly not attempt to sell any friend or family member a
SIP
> phone withou display. Depressing thought.
>
> Nor would we ever by a mobile phone without display.




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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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Date: Thu, 07 Aug 2003 17:54:31 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
In-reply-to: <Pine.LNX.4.53.0308051001250.23121@internaut.com>
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Bernard, you have got it right:

1. Sterman draft on digest authentication w/RADIUS.
2. Draft on RADIUS accounting w/SIP.
3. RADIUS Prepaid accounting support.
4. Document on RADIUS transport behavior, to support (2).

Also, his is an excellent overview of RADIUS. Many of us work however in
organizations that either sell/deploy web services and/or actively develop
it. How can we get a handle on web services when comparing the options?
(This means I am unfortunately not volunteering)

BTW: My observations are the same as Jiri's: Nobody has asked for DIAMETER,
and all are asking for RADIUS.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Bernard Aboba
> Sent: Tuesday, August 05, 2003 12:11 PM
> To: Allison Mankin
> Cc: sipping@ietf.org
> Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> 
> > I believe  they'd be more appropriate developed in SIPPING,
> > with cross-review from the RADIUS group.  But I think SIPPING needs some
> > information on what the RADIUS working group-to-be is expected to do on
> RADIUS,
> > in what timeframe.  From what I've understood, the charter would not
> primarily
> > be applications.  Could you give us more understanding?
> 
> Here is a summary of the discussion so far:
> 
> Basic RADIUS work
> 
> Since RADIUS is running out of attribute space, it was suggested that a
> basic attribute extension mechanism (such as one using the reserved
> vendorid of 0) be added.
> 
> This item could be handled in a 1-2 page document so it should have a
> short timeframe (~6 months).
> 
> There is interest in specifying how RADIUS handles prepaid accounting.
> Requests for this are coming from the SIP community as well as 3GPP2
> and folks interested in WLAN prepaid:
> 
> http://www.watersprings.org/pub/id/draft-lior-radius-prepaid-extensions-
> 01.txt
> 
> My understanding is that 3GPP2 intends to include this in a forthcoming
> standard so that the timeframe needs to be set to meet their needs.
> 
> In order to address concerns about RADIUS transport behavior and
> accounting reliability, there has been interest expressed on a draft on
> RADIUS transport behavior. This would presumably address some of the more
> egregious problems encountered with existing implementations (such as
> lack of backoff or jittering) as well as describing how RADIUS might be
> made to work more reliably:
> 
> http://www.watersprings.org/pub/id/draft-lior-radius-reliable-accounting-
> 00.txt
> 
> Without some fundamental changes to RADIUS (such as addition of
> a heartbeat), it is not clear that it will be possible to specify a
> demonstrably correct failover algorithm in addition to handling basic
> transport response issues (e.g. backoff, jittering, RTT estimation, etc.).
> 
> So there is some question about the scope of a potential transport
> document (e.g. just basic retransmission behavior or retransmssion +
> failover).  On the face of it, it seems that it would require a lot of
> changes to make RADIUS as reliable as Diameter:
> 
> http://www.ietf.org/rfc/rfc3539.txt
> 
> The timeframe of this work could be short (~9 months) assuming that the
> scope is containable.
> 
> WLAN-related
> 
> A number of different organizations (WFA, IEEE, etc.) have been specifying
> WLAN-related attributes.  There appears to be interest in standardizing
> the most useful ones.  This and some other WLAN-related work is described
> here:
> 
> http://www.watersprings.org/pub/id/draft-adrangi-radius-issues-in-pwlan-
> roaming-00.txt
> 
> The timeframe for this is hard to estimate, but I'd guess it is ~18 months
> or so.
> 
> We also had a request from the IEEE 802.11 Liason, Dorothy Stanley, for
> work on fast handoff.  Bill Arbaugh at the University of Maryland is
> implementing the draft below:
> 
> http://www.watersprings.org/pub/id/draft-irtf-aaaarch-handoff-02.txt
> 
> However, Bill has requested that this work not be considered as an IETF WG
> work item until his research is complete and the usefulness of the
> proposed extension can be evaluated.
> 
> _______________________________________________
> 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: Thu, 7 Aug 2003 17:57:20 -0600
Subject: Re: [Sipping] Request for editorial input
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On Thursday, Aug 7, 2003, at 12:41 Canada/Mountain, Robert Sparks wrote:

> As noted at IETF57, I'm picking up editorship of the torture-test
> document.
>

Excellent.

> [snip]
> I propose making such breaks explicit as follows:
> [snip]
> Foo: areally<break>
>      </break>bigtoken
>
> or
>
> Foo: areally<break>
>      </break>big<break>
>      </break>token
>
>

I'm tempted to suggest something less intrusive like a single EOL 
marker that indicates that the next line's leading WS is to be elided 
from the example text. Something like '*' or '~'.
eg)
Foo: areally~
      big~
	 token

equivalent to:

Foo: areallybigtoken

Of course this means that the above (legal) SIP would be hard to 
represent in the draft in literal form.

Not that this is better, but is IS more readable (IMHO). Perhaps there 
is a problem that I am overlooking with this, but it is certainly less 
intrusive. :-) Thoughts?

As for the appendix, are you proposing that the published draft would 
contain the (potentially base 64) encoded compressed archive of example 
messages -- completely unreadable by humans, but useful for extracting 
the examples?


Alan Hawrylyshen
Jasomi Networks Inc.


_______________________________________________
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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: Tim Melanchuk <timm@convedia.com>
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CC: Chris Boulton <cboulton@ubiquity.net>,
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Subject: Re: [Sipping] draft-melanchuk-sipping-msml-00.txt
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vishal, i hope the following explanation helps. please let me
know if you still have any questions or comments.

"dialog execution environment" is intended to be a generic term
which represents a black box from the msml perspective. that is,
msml creates media streams to and from the black box, and the
behaviour in the black box is specified by the dialog language.

moml allows an application to specify how (and when) primitives
are wired together. when primitives are defined to execute
sequentially, (i.e. are not in a group), then each primitive
is connected to a media stream in turn. when one completes, the
next is connected etc. when used with msml, the media stream
is the "target" of the <dialogstart>.

when primitives are grouped for concurrent execution, then
the primitives may be bridged to the "target" depending upon
the <group> topology as described in section 2.2 of moml for
a one dimensional group. the dimension of a group defines
the number of media streams which it supports. a dialog has
one media stream.

timm

Vishal Phirke wrote:

> 
> Hi Tim,
> I have question regarding msml.
> MSML defines three classes of media processing objects, viz., network
> connection, conference and dialog execution context. Out of these first two
> are physical entities within a media server whereas dialog execution
> context is an abstract thing, and a dialog will be typically bound to a
> network connection at the time of creation. The discription of the
> <dialogstart> element says that  "A media stream is created between the
> dialog execution environment and the connection or conference".  I
> understand the second endpoint but I am not sure what is the definition of
> dialog execution environment. As per my understanding, assuming dialog is
> made up of MOML constructs such as play and record, the typical requirement
> will be to switch the stream between player and recorder. But I can't see
> how a <join> at dialog level can indicate which resource to connect to. Can
> you elaborate more on the dialog execution environment?
> -Vishal
> 
> 



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This list is for NEW development of the application of SIP
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From: "Garland Sharratt" <gsharratt@convedia.com>
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Subject: RE: [Sipping] draft-melanchuk-sipping-moml-00.txt
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Thanks for the encouragement.  We are certainly favorable to this.  We
designed MSML/MOML to be a powerful and generic system for advanced contr=
ol
of media servers using SIP, and do hope that SIPPING takes on the drafts =
as
WG items at some point.  Since the drafts were only released in late June=
,
though, I think they probably need a bit more time for review and concens=
us
before being formally advanced as WG items.  We would certainly welcome a=
ny
comments you (or others) might have at this point.

Garland

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
De Leo Walter
Sent: Thursday, August 07, 2003 08:04
To: 'sipping@ietf.org'
Subject: RE: [Sipping] draft-melanchuk-sipping-moml-00.txt


I am interested in knowing if this drafat an msml draft will be proposed =
as
workgroup items.

I think it should there are some proprietary solutions for ms cotnrol ove=
r
SIP.

Regards

> -----Mensaje original-----
> De:	Tim Melanchuk [SMTP:timm@convedia.com]
> Enviado el:	Mi=E9rcoles 6 de Agosto de 2003 21:46
> Para:	Chris Boulton
> CC:	Garland Sharratt; sipping@ietf.org
> Asunto:	Re: [Sipping] draft-melanchuk-sipping-moml-00.txt
>
> thanks again for your feedback. answers to questions below.
>
> cheers,
> timm
>
> >
> > - Is there an alternative method for defining unlimited occurrences
> > using the 'max' attribute =3D 0 (e.g. pattern element uses max within=
 DTMF
>
> > primitive).  This seemed a little confusing.
> >
> i've had this comment from others. would extending the datatype
> to include an explicit "unlimited" value be less confusing and/or
> renaming "max" to "maxoccurs"?
>
> >
> > - 4.5.1.1 The description of <tvoice> and <tsilence> needs to be made
> > more explicit.
> >
> will do.
>
>
>
> _______________________________________________
> 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: Bernard Aboba <aboba@internaut.com>
CC: Jiri Kuthan <jiri@iptel.org>, hgs@cs.columbia.edu, sipping@ietf.org
Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
References: <5.2.0.9.0.20030807180926.053e0eb8@mailhost.fokus.gmd.de> <Pine.LNX.4.53.0308070904400.21844@internaut.com>
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Bernard Aboba wrote:

> So the question is:
> 
> "If a major protocol change were to be made to RADIUS to improve
> reliability, would such a change be deployed on RADIUS clients and
> servers and would it be acceptable for the SIP accounting specification
> to depend on such a change?"

Given that RADIUS and DIAMETER can work together for dial-in, I
think we can make them work together in SIP. Folks who need a
reliable & extensible protocol can already deploy DIAMETER.
Therefore, my suggestion would be to provide the basic functionality
in RADIUS, but NOT start any work on fundamental and possibly
non-backwards compatible RADIUS enhancements. That is, lets
publish Digest for RADIUS, SIP Accounting attributes for RADIUS*,
corresponding versions for DIAMETER, allow these to work
together, and let folks decide which protocols their particular
situation needs.

--Jari

*) This one may have to be informational, as is RADIUS accounting
base RFC.



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To: Henry Sinnreich <Henry.Sinnreich@mci.com>
CC: john.loughney@nokia.com, aboba@internaut.com, sipping@ietf.org
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
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Henry Sinnreich wrote:

> We have now four options to discuss in the SIP-AAA
> I-D.

Hmm.... again, I'm worried that we're spending too much
time arguing about the different underlying options.
We could easily end up in a lengthy beauty contest and
leave everyone using expired I-Ds and vendor specific
things in the meanwhile -- which is the current state of
affairs.

 From what I can see, folks have required SIP auth and
acct services for both RADIUS and DIAMETER. Lets give
these to them, and ensure that they can work together
so that folks who want high reliability, for instance,
can upgrade to DIAMETER without breaking the possibility
of talking to someone who only does RADIUS.

--Jari


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From exim@www1.ietf.org  Fri Aug  8 10:02:27 2003
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Subject: Re: [Sipping] Request for editorial input
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Alan Hawrylyshen <alan@jasomi.com>
Cc: sipping@ietf.org
In-Reply-To: <E20E4104-C932-11D7-9237-000A956A5F38@jasomi.com>
References: <E20E4104-C932-11D7-9237-000A956A5F38@jasomi.com>
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inline
On Thu, 2003-08-07 at 18:57, Alan Hawrylyshen wrote:
> On Thursday, Aug 7, 2003, at 12:41 Canada/Mountain, Robert Sparks wrote:
> > [snip]
> > I propose making such breaks explicit as follows:
> > [snip]
> > Foo: areally<break>
> >      </break>bigtoken
> >
> > or
> >
> > Foo: areally<break>
> >      </break>big<break>
> >      </break>token
> >
> >
> 
> I'm tempted to suggest something less intrusive like a single EOL 
> marker that indicates that the next line's leading WS is to be elided 
> from the example text. Something like '*' or '~'.
> eg)
> Foo: areally~
>       big~
> 	 token
> 
> equivalent to:
> 
> Foo: areallybigtoken
> 
> Of course this means that the above (legal) SIP would be hard to 
> represent in the draft in literal form.
> 
> Not that this is better, but is IS more readable (IMHO). Perhaps there 
> is a problem that I am overlooking with this, but it is certainly less 
> intrusive. :-) Thoughts?

Adam suggested something similar (based on MIME conventions). 

My concern (fwiw) is that I don't expect it to be easy to find a single
symbol that wouldn't be a legal character where it was breaking a line
(it's not always a token that needs breaking). We could adopt a
convention that anytime that appeared last on a line in the text of
this document it has special meaning, but it brings us back to
ambiguity. Arguably, this <break></break> can suffer from the same
deficiency, but it uses a high level of intrusiveness as a tool to help
resolve the ambiguity in the intended way.

An alternative would be to put in the examples something along the lines
of:

   INVITE sip:a.example.com SIP/2.0
   To: sip:b.example.com
<allOneLine>
   Call-ID: 23423423039480239840234
            2390498u239759237947293
            39457349539485739485739
            20234802908u09823095480
</allOneLine>
   CSeq: 83347 INVITE

That said, I don't feel too strongly about how we add this extra hint
as long as we add something.

> 
> As for the appendix, are you proposing that the published draft would 
> contain the (potentially base 64) encoded compressed archive of example 
> messages -- completely unreadable by humans, but useful for extracting 
> the examples?

Exactly. A few pages of 4799EA9280F923870CDD3988A... or the like.


> 
> 
> Alan Hawrylyshen
> Jasomi Networks Inc.


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Date: Fri, 08 Aug 2003 09:16:21 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: [Sipping] FYI: RADIUS & SIP
In-reply-to: <3F3371D6.3080700@piuha.net>
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> Hmm.... again, I'm worried that we're spending too much
> time arguing about the different underlying options.

The SIP-AAA WG has to find the best solution for SIP-AAA and is not a venue
to promote DIAMETER. 

Certainly no decision should be made without evaluating all available
options.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Jari Arkko
> Sent: Friday, August 08, 2003 4:48 AM
> To: Henry Sinnreich
> Cc: john.loughney@nokia.com; aboba@internaut.com; sipping@ietf.org
> Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
> 
> Henry Sinnreich wrote:
> 
> > We have now four options to discuss in the SIP-AAA
> > I-D.
> 
> Hmm.... again, I'm worried that we're spending too much
> time arguing about the different underlying options.
> We could easily end up in a lengthy beauty contest and
> leave everyone using expired I-Ds and vendor specific
> things in the meanwhile -- which is the current state of
> affairs.
> 
>  From what I can see, folks have required SIP auth and
> acct services for both RADIUS and DIAMETER. Lets give
> these to them, and ensure that they can work together
> so that folks who want high reliability, for instance,
> can upgrade to DIAMETER without breaking the possibility
> of talking to someone who only does RADIUS.
> 
> --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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From exim@www1.ietf.org  Fri Aug  8 14:54:19 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: Robert Sparks <rsparks@dynamicsoft.com>,
        Alan Hawrylyshen
	 <alan@jasomi.com>
Cc: sipping@ietf.org
Subject: RE: [Sipping] Request for editorial input
Date: Fri, 8 Aug 2003 13:52:10 -0500 
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Actually, my suggestion (although I didn't do a good job
of conveying it) would be to encode *all* the examples
using MIME Quoted-Printable encoding, thus removing all
ambiguity (and providing testers with ready-built decoders).

The downside of having all instances of "=" appear as
"=3D" should, of course, be considered....

/a

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Friday, August 08, 2003 9:00
> To: Alan Hawrylyshen
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] Request for editorial input
> 
> 
> inline
> On Thu, 2003-08-07 at 18:57, Alan Hawrylyshen wrote:
> > On Thursday, Aug 7, 2003, at 12:41 Canada/Mountain, Robert 
> Sparks wrote:
> > > [snip]
> > > I propose making such breaks explicit as follows:
> > > [snip]
> > > Foo: areally<break>
> > >      </break>bigtoken
> > >
> > > or
> > >
> > > Foo: areally<break>
> > >      </break>big<break>
> > >      </break>token
> > >
> > >
> > 
> > I'm tempted to suggest something less intrusive like a single EOL 
> > marker that indicates that the next line's leading WS is to 
> be elided 
> > from the example text. Something like '*' or '~'.
> > eg)
> > Foo: areally~
> >       big~
> > 	 token
> > 
> > equivalent to:
> > 
> > Foo: areallybigtoken
> > 
> > Of course this means that the above (legal) SIP would be hard to 
> > represent in the draft in literal form.
> > 
> > Not that this is better, but is IS more readable (IMHO). 
> Perhaps there 
> > is a problem that I am overlooking with this, but it is 
> certainly less 
> > intrusive. :-) Thoughts?
> 
> Adam suggested something similar (based on MIME conventions). 
> 
> My concern (fwiw) is that I don't expect it to be easy to 
> find a single
> symbol that wouldn't be a legal character where it was breaking a line
> (it's not always a token that needs breaking). We could adopt a
> convention that anytime that appeared last on a line in the text of
> this document it has special meaning, but it brings us back to
> ambiguity. Arguably, this <break></break> can suffer from the same
> deficiency, but it uses a high level of intrusiveness as a 
> tool to help
> resolve the ambiguity in the intended way.
> 
> An alternative would be to put in the examples something 
> along the lines
> of:
> 
>    INVITE sip:a.example.com SIP/2.0
>    To: sip:b.example.com
> <allOneLine>
>    Call-ID: 23423423039480239840234
>             2390498u239759237947293
>             39457349539485739485739
>             20234802908u09823095480
> </allOneLine>
>    CSeq: 83347 INVITE
> 
> That said, I don't feel too strongly about how we add this extra hint
> as long as we add something.
> 
> > 
> > As for the appendix, are you proposing that the published 
> draft would 
> > contain the (potentially base 64) encoded compressed 
> archive of example 
> > messages -- completely unreadable by humans, but useful for 
> extracting 
> > the examples?
> 
> Exactly. A few pages of 4799EA9280F923870CDD3988A... or the like.
> 
> 
> > 
> > 
> > Alan Hawrylyshen
> > Jasomi Networks Inc.
> 
> 
> _______________________________________________
> 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: Sat, 09 Aug 2003 11:51:55 -0700
Subject: Re: [Sipping] DTMF: Why Not A New Method?
From: Cullen Jennings <fluffy@cisco.com>
To: sipping <sipping@ietf.org>
CC: Adam Roach <adam@dynamicsoft.com>
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There has been some discussion on the MESSAGE vs. NEW_METHOD for reporting
the information from KPML. I don't think I have heard any strong argument
for MESSAGE over NEW_METHOD but I have gotten a little lost on the whole
argument.

I think we might have consensus on NEW_METHOD but I'm not sure. I don't care
but I think we need to make a decision on this and get on with it. Do we
have consensus on this?

Cullen
 


On 7/24/03 13:56, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Oh, in that case...
> 
> +------------------------------------------------------------+
> |  You have received an IM of type "application/kpml+xml",   |
> |  which is not a registered media type.                     |
> |                                                            |
> |  [x] Save it as a file                                     |
> |  [ ] Open it with application: ________________            |
> |                                                            |
> |  [x] Always ask when handing type "application/kpml+xml"   |
> |                                                            |
> |                                             [OK]  [Cancel] |
> +------------------------------------------------------------+
> 
> /a
>


_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: Re: [Sipping] Request for editorial input
From: Cullen Jennings <fluffy@cisco.com>
To: Robert Sparks <rsparks@dynamicsoft.com>,
        Alan Hawrylyshen <alan@jasomi.com>
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Some convention like this would be very helpful to me in the S/MIME example
call flows. Just about anything would be fine with me. Does this need to be
a IETF discussion? 


On 8/8/03 7:00, "Robert Sparks" <rsparks@dynamicsoft.com> wrote:

> inline
> On Thu, 2003-08-07 at 18:57, Alan Hawrylyshen wrote:
>> On Thursday, Aug 7, 2003, at 12:41 Canada/Mountain, Robert Sparks wrote:
>>> [snip]
>>> I propose making such breaks explicit as follows:
>>> [snip]
>>> Foo: areally<break>
>>>      </break>bigtoken
>>> 
>>> or
>>> 
>>> Foo: areally<break>
>>>      </break>big<break>
>>>      </break>token
>>> 
>>> 
>> 
>> I'm tempted to suggest something less intrusive like a single EOL
>> marker that indicates that the next line's leading WS is to be elided
>> from the example text. Something like '*' or '~'.
>> eg)
>> Foo: areally~
>>       big~
>> token
>> 
>> equivalent to:
>> 
>> Foo: areallybigtoken
>> 
>> Of course this means that the above (legal) SIP would be hard to
>> represent in the draft in literal form.
>> 
>> Not that this is better, but is IS more readable (IMHO). Perhaps there
>> is a problem that I am overlooking with this, but it is certainly less
>> intrusive. :-) Thoughts?
> 
> Adam suggested something similar (based on MIME conventions).
> 
> My concern (fwiw) is that I don't expect it to be easy to find a single
> symbol that wouldn't be a legal character where it was breaking a line
> (it's not always a token that needs breaking). We could adopt a
> convention that anytime that appeared last on a line in the text of
> this document it has special meaning, but it brings us back to
> ambiguity. Arguably, this <break></break> can suffer from the same
> deficiency, but it uses a high level of intrusiveness as a tool to help
> resolve the ambiguity in the intended way.
> 
> An alternative would be to put in the examples something along the lines
> of:
> 
>  INVITE sip:a.example.com SIP/2.0
>  To: sip:b.example.com
> <allOneLine>
>  Call-ID: 23423423039480239840234
>           2390498u239759237947293
>           39457349539485739485739
>           20234802908u09823095480
> </allOneLine>
>  CSeq: 83347 INVITE
> 
> That said, I don't feel too strongly about how we add this extra hint
> as long as we add something.
> 
>> 
>> As for the appendix, are you proposing that the published draft would
>> contain the (potentially base 64) encoded compressed archive of example
>> messages -- completely unreadable by humans, but useful for extracting
>> the examples?
> 
> Exactly. A few pages of 4799EA9280F923870CDD3988A... or the like.
> 
> 
>> 
>> 
>> Alan Hawrylyshen
>> Jasomi Networks Inc.
> 
> 
> _______________________________________________
> 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: Cullen Jennings <fluffy@cisco.com>
To: <sipping@ietf.org>
CC: Jiri Kuthan <jiri@iptel.org>, Henning Schulzrinne <hgs@cs.columbia.edu>,
        Bernard Aboba <aboba@internaut.com>, <jarkko@piuha.net>
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I agree with Jari that we should be able to define extensions that are
parallel definitions that carry the same semantics in Radius or DIAMETER and
avoid many arguments about what the future will be. It should be the
semantic information we want to transfer that is important to the SIPPING
group, not the transport.

I think we should keep the Authentication work somewhat separate in our
minds from the Accounting work - they have little in common.

I agree with the comment that no one is asking for Diameter - the requests I
have heard have all been for Radius Accounting. I have not had requests for
Diameter or really for Authentication / Authorization outside of 3GPP. I
think it is worth doing, Cisco implemented an Authentication scheme for SIP
over Radius a long time ago but I don't believe it has ever been used in a
production network. A ton of people are using Radius for SIP billing in
production networks - I think Accounting is a higher priority, but less
interesting, issue than Authentication.

I find the arguments of why Radius should not be used for Accounting
somewhat compelling but until service providers request it, it is not likely
to be real high priority. Because of this, I like Jari idea of getting the
semantic information done for both in a way that allows Radius to work today
for providers that use it but minimizes the changes to move to Diameter.

Cullen




On 8/8/03 2:32, "Jari Arkko" <jarkko@piuha.net> wrote:

> Bernard Aboba wrote:
> 
>> So the question is:
>> 
>> "If a major protocol change were to be made to RADIUS to improve
>> reliability, would such a change be deployed on RADIUS clients and
>> servers and would it be acceptable for the SIP accounting specification
>> to depend on such a change?"
> 
> Given that RADIUS and DIAMETER can work together for dial-in, I
> think we can make them work together in SIP. Folks who need a
> reliable & extensible protocol can already deploy DIAMETER.
> Therefore, my suggestion would be to provide the basic functionality
> in RADIUS, but NOT start any work on fundamental and possibly
> non-backwards compatible RADIUS enhancements. That is, lets
> publish Digest for RADIUS, SIP Accounting attributes for RADIUS*,
> corresponding versions for DIAMETER, allow these to work
> together, and let folks decide which protocols their particular
> situation needs.
> 
> --Jari
> 
> *) This one may have to be informational, as is RADIUS accounting
> base RFC.
> 
> 
> 
> _______________________________________________
> 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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Date: Sat, 09 Aug 2003 11:03:16 -0700
Subject: Re: [Sipping] FWD:I-D
	ACTION:draft-johnston-sipping-rtcp-summary-00.txt
From: Cullen Jennings <fluffy@cisco.com>
To: Eric Burger <eburger@snowshore.com>, Alan Johnston <alan.johnston@mci.com>
CC: <sipping@ietf.org>
Message-ID: <BB5A8574.15821%fluffy@cisco.com>
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One of the problems with end of call reporting is that it needs to have
enough time based granularity information to be useful. For example ....
Consider two calls.

Call A. This is a 100 minute call with 1% packet loss distributed evenly.

Call B. This had 0% packet loss for first 99 minutes then 100% packet loss
for 1 minute then user hung up.

Call A was a good call and the service provider is probably happy. Call B is
a bad call and needs attention. The reporting needs to differentiate between
these two cases. 

Cullen


On 7/14/03 8:09, "Eric Burger" <eburger@snowshore.com> wrote:

> Would any reporting system ever be interested in RTCP messages?  I would
> expect network operators (who I presume would be the interested third-parties)
> to be interested in end-of-call summaries only.  Otherwise, a service provider
> of any significant size is signing up for a significant amount of RTCP
> reporting (flooding?).
> 
> Being NOTIFIED of end-of-call statistics is a sensible option.  That way you
> can use the existing SIP security, subscription, and message routing
> infrastructure.  
> 
>> -----Original Message-----
>> From: Alan Johnston [mailto:alan.johnston@mci.com]
>> Sent: Fri, June 27, 2003 2:38 PM
>> To: sipping@ietf.org
>> Subject: [Sipping] FWD:I-D
>> ACTION:draft-johnston-sipping-rtcp-summary-00.txt
>> 
>> 
>> FYI - a new draft on the delivery of RTCP summary reports to
>> third parties.
>> 
>> Comments most welcome.
>> 
>> Thanks,
>> Alan Johnston
>> MCI
>> 
>> 
>> ----------
>> To: IETF-Announce: ;
>> Subject: I-D ACTION:draft-johnston-sipping-rtcp-summary-00.txt
>> From: Internet-Drafts@ietf.org
>> Date: Tue, 24 Jun 2003 06:56:20 -0400
>> Reply-to: Internet-Drafts@ietf.org
>> Sender: owner-ietf-announce@ietf.org
>> 
>> A New Internet-Draft is available from the on-line
>> Internet-Drafts directories.
>> 
>> 
>>          Title           : RTCP Summary Report Delivery to
>> SIP Third Parties
>>          Author(s)       : A. Johnston, H. Sinnreich, A. Clark
>>          Filename        : draft-johnston-sipping-rtcp-summary-00.txt
>>          Pages           : 9
>>          Date            : 2003-6-23
>> 
> 
> 
> _______________________________________________
> 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



From exim@www1.ietf.org  Sun Aug 10 19:48:46 2003
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Date: Sun, 10 Aug 2003 18:43:04 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] FWD:I-DACTION:draft-johnston-sipping-rtcp-summary-00.txt
In-reply-to: <BB5A8574.15821%fluffy@cisco.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
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        "'Alan Johnston'" <alan.johnston@mci.com>
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Cullen, I agree that 

>The reporting needs to differentiate between
> these two cases.

Burst errors are treated very well I believe in 

http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-report-extns-06.txt 

and this benefit extends to

http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-00.t
xt 

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Cullen Jennings
> Sent: Saturday, August 09, 2003 1:03 PM
> To: Eric Burger; Alan Johnston
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] FWD:I-D ACTION:draft-johnston-sipping-rtcp-summary-
> 00.txt
> 
> 
> One of the problems with end of call reporting is that it needs to have
> enough time based granularity information to be useful. For example ....
> Consider two calls.
> 
> Call A. This is a 100 minute call with 1% packet loss distributed evenly.
> 
> Call B. This had 0% packet loss for first 99 minutes then 100% packet loss
> for 1 minute then user hung up.
> 
> Call A was a good call and the service provider is probably happy. Call B
> is
> a bad call and needs attention. The reporting needs to differentiate
> between
> these two cases.
> 
> Cullen
> 
> 
> On 7/14/03 8:09, "Eric Burger" <eburger@snowshore.com> wrote:
> 
> > Would any reporting system ever be interested in RTCP messages?  I would
> > expect network operators (who I presume would be the interested third-
> parties)
> > to be interested in end-of-call summaries only.  Otherwise, a service
> provider
> > of any significant size is signing up for a significant amount of RTCP
> > reporting (flooding?).
> >
> > Being NOTIFIED of end-of-call statistics is a sensible option.  That way
> you
> > can use the existing SIP security, subscription, and message routing
> > infrastructure.
> >
> >> -----Original Message-----
> >> From: Alan Johnston [mailto:alan.johnston@mci.com]
> >> Sent: Fri, June 27, 2003 2:38 PM
> >> To: sipping@ietf.org
> >> Subject: [Sipping] FWD:I-D
> >> ACTION:draft-johnston-sipping-rtcp-summary-00.txt
> >>
> >>
> >> FYI - a new draft on the delivery of RTCP summary reports to
> >> third parties.
> >>
> >> Comments most welcome.
> >>
> >> Thanks,
> >> Alan Johnston
> >> MCI
> >>
> >>
> >> ----------
> >> To: IETF-Announce: ;
> >> Subject: I-D ACTION:draft-johnston-sipping-rtcp-summary-00.txt
> >> From: Internet-Drafts@ietf.org
> >> Date: Tue, 24 Jun 2003 06:56:20 -0400
> >> Reply-to: Internet-Drafts@ietf.org
> >> Sender: owner-ietf-announce@ietf.org
> >>
> >> A New Internet-Draft is available from the on-line
> >> Internet-Drafts directories.
> >>
> >>
> >>          Title           : RTCP Summary Report Delivery to
> >> SIP Third Parties
> >>          Author(s)       : A. Johnston, H. Sinnreich, A. Clark
> >>          Filename        : draft-johnston-sipping-rtcp-summary-00.txt
> >>          Pages           : 9
> >>          Date            : 2003-6-23
> >>
> >
> >
> > _______________________________________________
> > 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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Subject: RE: [Sipping] Request for editorial input
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From: "Chris Boulton" <cboulton@ubiquity.net>
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        "Alan Hawrylyshen" <alan@jasomi.com>
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inline

>-----Original Message-----
>From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
>Sent: 08 August 2003 15:00
>To: Alan Hawrylyshen
>Cc: sipping@ietf.org
>Subject: Re: [Sipping] Request for editorial input
>
>inline
>On Thu, 2003-08-07 at 18:57, Alan Hawrylyshen wrote:
>> On Thursday, Aug 7, 2003, at 12:41 Canada/Mountain, Robert Sparks
wrote:
>> > [snip]
>> > I propose making such breaks explicit as follows:
>> > [snip]
>> > Foo: areally<break>
>> >      </break>bigtoken
>> >
>> > or
>> >
>> > Foo: areally<break>
>> >      </break>big<break>
>> >      </break>token
>> >
>> >
>>
>> I'm tempted to suggest something less intrusive like a single EOL
>> marker that indicates that the next line's leading WS is to be elided
>> from the example text. Something like '*' or '~'.
>> eg)
>> Foo: areally~
>>       big~
>> 	 token
>>
>> equivalent to:
>>
>> Foo: areallybigtoken
>>
>> Of course this means that the above (legal) SIP would be hard to
>> represent in the draft in literal form.
>>
>> Not that this is better, but is IS more readable (IMHO). Perhaps
there
>> is a problem that I am overlooking with this, but it is certainly
less
>> intrusive. :-) Thoughts?
>
>Adam suggested something similar (based on MIME conventions).
>
>My concern (fwiw) is that I don't expect it to be easy to find a single
>symbol that wouldn't be a legal character where it was breaking a line
>(it's not always a token that needs breaking). We could adopt a
>convention that anytime that appeared last on a line in the text of
>this document it has special meaning, but it brings us back to
>ambiguity. Arguably, this <break></break> can suffer from the same
>deficiency, but it uses a high level of intrusiveness as a tool to help
>resolve the ambiguity in the intended way.
>
>An alternative would be to put in the examples something along the
lines
>of:
>
>   INVITE sip:a.example.com SIP/2.0
>   To: sip:b.example.com
><allOneLine>
>   Call-ID: 23423423039480239840234
>            2390498u239759237947293
>            39457349539485739485739
>            20234802908u09823095480
></allOneLine>
>   CSeq: 83347 INVITE
>


[Chris Boulton] Robert, This one looks fine to me.


>That said, I don't feel too strongly about how we add this extra hint
>as long as we add something.
>
>>
>> As for the appendix, are you proposing that the published draft would
>> contain the (potentially base 64) encoded compressed archive of
example
>> messages -- completely unreadable by humans, but useful for
extracting
>> the examples?
>
>Exactly. A few pages of 4799EA9280F923870CDD3988A... or the like.
>
>
>>
>>
>> Alan Hawrylyshen
>> Jasomi Networks Inc.
>
>
>_______________________________________________
>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] DTMF: Why Not A New Method?
Date: Mon, 11 Aug 2003 10:18:46 +0100
Message-ID: <45730E094814E44488F789C1CDED27AE01E22CD7@gbnewp0758m.eu.ubiquity.net>
Thread-Topic: [Sipping] DTMF: Why Not A New Method?
Thread-Index: AcNfadIU6Cu0pYuqQim0nR5s29t0OQAf4F2Q
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
Cc: "Adam Roach" <adam@dynamicsoft.com>
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NEW_MTHOD gets my vote.

Chris.


>-----Original Message-----
>From: Cullen Jennings [mailto:fluffy@cisco.com]
>Sent: 09 August 2003 19:52
>To: sipping
>Cc: Adam Roach
>Subject: Re: [Sipping] DTMF: Why Not A New Method?
>
>
>There has been some discussion on the MESSAGE vs. NEW_METHOD for
reporting
>the information from KPML. I don't think I have heard any strong
argument
>for MESSAGE over NEW_METHOD but I have gotten a little lost on the
whole
>argument.
>
>I think we might have consensus on NEW_METHOD but I'm not sure. I don't
>care
>but I think we need to make a decision on this and get on with it. Do
we
>have consensus on this?
>
>Cullen
>
>
>
>On 7/24/03 13:56, "Adam Roach" <adam@dynamicsoft.com> wrote:
>
>> Oh, in that case...
>>
>> +------------------------------------------------------------+
>> |  You have received an IM of type "application/kpml+xml",   |
>> |  which is not a registered media type.                     |
>> |                                                            |
>> |  [x] Save it as a file                                     |
>> |  [ ] Open it with application: ________________            |
>> |                                                            |
>> |  [x] Always ask when handing type "application/kpml+xml"   |
>> |                                                            |
>> |                                             [OK]  [Cancel] |
>> +------------------------------------------------------------+
>>
>> /a
>>
>
>
>_______________________________________________
>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] DTMF: Why Not A New Method?
Date: Mon, 11 Aug 2003 11:41:58 +0100
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Thread-Topic: [Sipping] DTMF: Why Not A New Method?
Thread-Index: AcNfaf9/+MFjsvF0TJO5d70lQSqXQgAifbqw
From: "Mills, Duncan, D, CND Tech Dev, VF UK" <Duncan.Mills@gb.vodafone.co.uk>
To: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
Cc: "Adam Roach" <adam@dynamicsoft.com>
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Cullen and others,

I haven't been following the discussion too closely, but if we do decide =
to have a new method, could we aim to make it reasonably generic?  I =
don't have a proposal really, but it seems there are a few problems =
floating around already where people end up suggesting that MESSAGE can =
be (ab)used to solve them.  I guess we're talking about a way of passing =
'information' that, unlike INFO, is NOT necessarily related to an =
ongoing session?

Regards,

Duncan



-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com]
Sent: 09 August 2003 19:52
To: sipping
Cc: Adam Roach
Subject: Re: [Sipping] DTMF: Why Not A New Method?



There has been some discussion on the MESSAGE vs. NEW_METHOD for =
reporting
the information from KPML. I don't think I have heard any strong =
argument
for MESSAGE over NEW_METHOD but I have gotten a little lost on the whole
argument.

I think we might have consensus on NEW_METHOD but I'm not sure. I don't =
care
but I think we need to make a decision on this and get on with it. Do we
have consensus on this?

Cullen
=20


On 7/24/03 13:56, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Oh, in that case...
>=20
> +------------------------------------------------------------+
> |  You have received an IM of type "application/kpml+xml",   |
> |  which is not a registered media type.                     |
> |                                                            |
> |  [x] Save it as a file                                     |
> |  [ ] Open it with application: ________________            |
> |                                                            |
> |  [x] Always ask when handing type "application/kpml+xml"   |
> |                                                            |
> |                                             [OK]  [Cancel] |
> +------------------------------------------------------------+
>=20
> /a
>


_______________________________________________
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: Tim Melanchuk <timm@convedia.com>
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To: Cullen Jennings <fluffy@cisco.com>
CC: sipping <sipping@ietf.org>, Adam Roach <adam@dynamicsoft.com>
Subject: Re: [Sipping] DTMF: Why Not A New Method?
References: <BB5A90DB.15828%fluffy@cisco.com>
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will NEW_METHOD also be able to satisfy the requirements for an app
asynchronously pushing messages to a client (e.g. for pre-paid
calling cards) as defined in REQ 4 & 7 of the app-interaction-
framework-01.

alternatively, what SIP message would be used here? the requirements
and context from that draft are below.

timm

from draft-rosenberg-sipping-app-interaction-framework-01:

    In this case, there is no message already in transit that can be used
    as a vehicle for pushing a user interface component. This requires
    that applications can generate their own messages to push a new
    component to a UA:

    REQ 4: A UA application must be able to send a SIP message to the UA
       at the other end of the dialog, asking it to create a new
       interface component.

<snip>

    However, synchronous updates are not sufficient for many
    applications. Frequently, the interface will need to be updated
    asynchronously by the application, without an explicit user action. A
    good example of this is, once again, the pre-paid calling card
    application. The application might like to update the user interface
    when the timer runs out on the call. This introduces several
    requirements:

    REQ 7: It must be possible for an application to asynchronously push
       an update to an existing user interface component, either in a
       message that was already in transit, or by generating a new
       message.


Cullen Jennings wrote:

> There has been some discussion on the MESSAGE vs. NEW_METHOD for reporting
> the information from KPML. I don't think I have heard any strong argument
> for MESSAGE over NEW_METHOD but I have gotten a little lost on the whole
> argument.
> 
> I think we might have consensus on NEW_METHOD but I'm not sure. I don't care
> but I think we need to make a decision on this and get on with it. Do we
> have consensus on this?
> 



_______________________________________________
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: Adam Roach <adam@dynamicsoft.com>
To: "'Tim Melanchuk'" <timm@convedia.com>,
        Cullen Jennings
	 <fluffy@cisco.com>
Cc: sipping <sipping@ietf.org>, Adam Roach <adam@dynamicsoft.com>
Subject: RE: [Sipping] DTMF: Why Not A New Method?
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> will NEW_METHOD also be able to...

No!

This is precisely what I'm trying to avoid happening. Methods
have actual semantics associated with them. They should mean
exactly one thing, not several. INVITE means you're starting
a session. SUBSCRIBE means that you're subscribing to the state
of a resource. MESSAGE means you're sending an instant message.

The new method that we define for KPML should also mean precisely
one thing: "I'm conveying requested stimulus from the user."

If you need to define something else with different semantics
than that, I would argue that there's a fair chance that a new
method is indicated.

Otherwise, the actual method names become meaningless. If that's
the path we are going down, consistency would argue for deprecating
all existing methods in favor of a single method (perhaps, "DO"),
and inferring all semantics from some horribly complex combination
of Content-Type, Content-Disposition, Accept, Supported, and similar
header fields.

/a

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--NextPart

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


	Title		: Late Media in the Session Initiation Protocol
	Author(s)	: C. Stredicke
	Filename	: draft-stredicke-sipping-late-media-00.txt
	Pages		: 7
	Date		: 2003-8-11
	
'Late Media' is presented after a dialog is disconnected or when the 
dialog is temporarily without media. The 'Late-Media' header 
indicates under which circumstances late media should be rendered.

A URL for this Internet-Draft is:
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From exim@www1.ietf.org  Mon Aug 11 18:12:33 2003
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Subject: Re: [Sipping] DTMF: Why Not A New Method?
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fine, i'm all for precisely defined semantics.

the app-interaction-framework seems to me to identify a requirement
for an application to asynchronously push an interface component.
am i reading this correctly? if not, fine. if so, what method should
be used there, and, if we need a new method for that as well, perhaps
we should look at 2 new methods. one to "request" stimulus and one
to "convey" the requested stimulus.

note, i'm simply trying to understand how different proposals/drafts
fit together. i'm not advocating violating sacrosanct principles of
protocol design :-)

timm

Adam Roach wrote:

>>will NEW_METHOD also be able to...
> 
> 
> No!
> 
[snip]
> 
> The new method that we define for KPML should also mean precisely
> one thing: "I'm conveying requested stimulus from the user."
> 
> If you need to define something else with different semantics
> than that, I would argue that there's a fair chance that a new
> method is indicated.
> 
[snip]


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I will not stand in the way of concensus on this. A new method is 
acceptable to me.

	Paul

Cullen Jennings wrote:
> There has been some discussion on the MESSAGE vs. NEW_METHOD for reporting
> the information from KPML. I don't think I have heard any strong argument
> for MESSAGE over NEW_METHOD but I have gotten a little lost on the whole
> argument.
> 
> I think we might have consensus on NEW_METHOD but I'm not sure. I don't care
> but I think we need to make a decision on this and get on with it. Do we
> have consensus on this?
> 
> Cullen
>  
> 
> 
> On 7/24/03 13:56, "Adam Roach" <adam@dynamicsoft.com> wrote:
> 
> 
>>Oh, in that case...
>>
>>+------------------------------------------------------------+
>>|  You have received an IM of type "application/kpml+xml",   |
>>|  which is not a registered media type.                     |
>>|                                                            |
>>|  [x] Save it as a file                                     |
>>|  [ ] Open it with application: ________________            |
>>|                                                            |
>>|  [x] Always ask when handing type "application/kpml+xml"   |
>>|                                                            |
>>|                                             [OK]  [Cancel] |
>>+------------------------------------------------------------+
>>
>>/a
>>
> 
> 
> 
> _______________________________________________
> 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] DTMF: Why Not A New Method?
Date: Tue, 12 Aug 2003 09:03:50 +0100
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Thread-Topic: [Sipping] DTMF: Why Not A New Method?
Thread-Index: AcNgVbJOVCW0L5INRcWHyzXPPDWQAwAUKQ2A
From: "Mills, Duncan, D, CND Tech Dev, VF UK" <Duncan.Mills@gb.vodafone.co.uk>
To: "Tim Melanchuk" <timm@convedia.com>, "Adam Roach" <adam@dynamicsoft.com>
Cc: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
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Tim,

You appear to be coming from the same angle as me.  If we are =
considering having a new method for KPML, because MESSAGE is being =
abused, then we may well find that we need other new methods to cover =
other areas where MESSAGE could be abused.  Pushing something from an =
application server to an end-user is one area where people are already =
wrongly proposing to use MESSAGE.

I never really meant to say that we define a generic method with very =
broad semantics.  (Wrist slapped, mouth washed).  My mail was probably =
worded badly.  I merely meant to say that there are probably a couple of =
areas where a new method is perhaps needed, and maybe if we think about =
it, we only need one method to meet the requirements of these areas?

I guess any new method would be bi-directional anyway, so for instance, =
if the new method was called "PUSH", then it could be used from an =
end-user to an application to push key-presses and it could be used from =
an application server to an end-point to push (e.g.) an interface =
component.

Rather than having a new method to solve the key press issue and a =
second new method to solve the 'push' issue.  Maybe this is obvious to =
everyone?

Cheers,

Duncan


-----Original Message-----
From: Tim Melanchuk [mailto:timm@convedia.com]
Sent: 11 August 2003 23:12
To: Adam Roach
Cc: Cullen Jennings; sipping
Subject: Re: [Sipping] DTMF: Why Not A New Method?


fine, i'm all for precisely defined semantics.

the app-interaction-framework seems to me to identify a requirement
for an application to asynchronously push an interface component.
am i reading this correctly? if not, fine. if so, what method should
be used there, and, if we need a new method for that as well, perhaps
we should look at 2 new methods. one to "request" stimulus and one
to "convey" the requested stimulus.

note, i'm simply trying to understand how different proposals/drafts
fit together. i'm not advocating violating sacrosanct principles of
protocol design :-)

timm

Adam Roach wrote:

>>will NEW_METHOD also be able to...
>=20
>=20
> No!
>=20
[snip]
>=20
> The new method that we define for KPML should also mean precisely
> one thing: "I'm conveying requested stimulus from the user."
>=20
> If you need to define something else with different semantics
> than that, I would argue that there's a fair chance that a new
> method is indicated.
>=20
[snip]


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I've migrated the supplementary web page for SIPPING onto the same
database as SIP.

Follow the "Drafts" or "Morgue" links from:

http://www.softarmor.com/sipping/

Of course, the detail information for the drafts isn't current -- it
went stale just before the last IETF meetng -- but we're working on it.
Right, Rohan and Gonzalo?

--
Dean



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On Tuesday, August 12, 2003, at 01:17 PM, Dean Willis wrote:

> I've migrated the supplementary web page for SIPPING onto the same
> database as SIP.
>
> Follow the "Drafts" or "Morgue" links from:
>
> http://www.softarmor.com/sipping/
>
> Of course, the detail information for the drafts isn't current -- it
> went stale just before the last IETF meetng -- but we're working on it.
> Right, Rohan and Gonzalo?

Correct.

thx,
-r

> --
> Dean
>
>


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On Wednesday, July 30, 2003, at 09:13 AM, Jonathan Rosenberg wrote:

>
>
> Adam Roach wrote:
>
>> Alan Johnston [mailto:alan.johnston@mci.com] writes:
>>  > If we do not use a GRUU in a Contact, then it will not help solve 
>> the
>>  > attended transfer problem or the conferencing application -
>>  > both of these use the Contact URI.
>> Admittedly I haven't given this a whole lot of thought
>> yet, so it might not make perfect sense... but... can't
>> we solve these cases by using a new header and making such
>> applications GRUU aware?
>
> That was what I had in mind, yes.
>
> Attended transfer, it seems, already needs to be aware of how to get a 
> GRUU. The current draft has a whole slew of mechanisms for doing it, 
> the aim would be that the GRUU approach would replace those > mechanisms.

To be fair, the current mechanism is try and pray.

> Similarly in cc-conferencing, it already talks about using the 
> contact, so now it could talk about using the gruu in this other 
> header.
>
> I don't think we have so much of either of these deployed that this 
> cannot change.

I disagree.  Most of the SIP phones I see at SIPit support completing a 
transfer using a REFER with a Replaces header embedded in the Refer-To. 
  The Refer-To URI is always set to the Contact URI of the appropriate 
UA.

thanks,
-rohan


>
> -Jonathan R.
>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> 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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Subject: Re: [Sipping] Re: Update to 3pcc
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Hi David,

I think it is completely reasonable to build a pre-paid calling system  
using a proxy with AAA and session timer and optionally a cooperating  
UA (for IVR features).

If a click-to-dial application sends a REFER to my phone which is using  
a pre-paid PSTN gateway service, there is no reason that the gateway  
service needs to use a B2BUA.  Instead, a proxy can use the session  
timer extension when proxying the call for a portion of the user's  
available time.

However, even if the PSTN gateway service decided to use a B2BUA for  
its prepaid service, this is no reason the click-to-dial application  
(possibly from a completely different organization) needs to use 3pcc.   
The two should be disconnected.

thanks,
-rohan

On Tuesday, August 5, 2003, at 11:17 PM, David Schwartz wrote:

> Hi Jiri.
>
> I was referring specifically to the call teardown issue associated  
> with most
> pre-paid billing models.
>
> David
>
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri.kuthan@fokus.fraunhofer.de]
> Sent: Tue, August 05, 2003 7:39 PM
> To: David Schwartz; 'sipping@ietf.org'
> Subject: Re: [Sipping] Re: Update to 3pcc
>
>
> Hello David,
>
> I don't see how billing servers are related to method of
> call initiation.
>
> Actually I believe that how a call is initiated (by manual
> dialing a destination, by REFER-based click-to-dial, by b2bua-based
> click-to-dial, CTI, etc.) is completely orthogonal to how the
> call is charged for. Regardless what charging system you use
> (if any), there will be different methods for initiating calls,
> some will be better off than others. I maintain that a click-to-dial
> b2bua is not a BCP.
>
> -jiri
>
> At 10:10 AM 7/30/2003, David Schwartz wrote:
>> Hi Jiri
>>
>> REFER is good solution if you assume that either there are no billing
>> issues, or alternatively, that the endpoint/s handle the billing  
>> issues. If
>> this is not the case, for example, in an application where a  
>> centralized
>> controller is setting up a call to two PSTN numbers, then you will  
>> need a
>> centralized billing entity and REFER will not suffice.
>>
>> David
>>
>> -----Original Message-----
>> From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
>> Sent: Mon, July 28, 2003 7:00 PM
>> To: sipping@ietf.org
>> Subject: Sipping digest, Vol 1 #1144 - 2 msgs
>>
>>
>> Send Sipping mailing list submissions to
>>        sipping@ietf.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>        https://www1.ietf.org/mailman/listinfo/sipping
>> or, via email, send a message with subject or body 'help' to
>>        sipping-request@ietf.org
>>
>> You can reach the person managing the list at
>>        sipping-admin@ietf.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of Sipping digest..."
>>
>>
>> Today's Topics:
>>
>>   1. Re: Diversion header (Cullen Jennings)
>>   2. Re: Update to 3pcc (Jiri Kuthan)
>>
>> --__--__--
>>
>> Message: 1
>> Date: Sun, 27 Jul 2003 19:05:10 -0700
>> Subject: Re: [Sipping] Diversion header
>> From: Cullen Jennings <fluffy@cisco.com>
>> To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
>>
>>
>> I assume your question is there some way for  for the final  
>> destination
> that
>> receives the call to know the who the 1st callee was that had some  
>> device
> in
>> the network redirect the call elsewhere.
>>
>> The answer is YES - there has been a lot of confusion over this so  
>> think
>> about my answer carefully - it will seem counter intuitive at first.  
>> The
>> answer is the "To" header but it will take me a second to explain why.
>>
>> Imagine case 1 where A send a call to proxy B,  that sent to to C.  
>> Now C
>> sent a 302 back B and B then forked the call to D. The To header in  
>> the
>> message D receives will still have the To set to C. C is the element  
>> that
>> sent the first 302.
>>
>> Now imagine a case (like your question) where there is not a recursive
>> proxy. The 302 will go all the way back to A. Now A forms a new  
>> invite to
>> but it can leave the To header set to the same thing it was in the  
>> original
>> message (C in this case). When the message arrives at D, the To  
>> header will
>> be set to C.
>>
>> If D was a voice mail system, it might be able to know to act on  
>> behalf of
> C
>> instead of some other entity.
>>
>> There are more complex things that relate to your request. Mary  
>> Barnes has
>> been moving the history work along
>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history- 
>> 04.txt
>>
>> There is some work on QSIG mappings with diversion
>> http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2sip- 
>> diversion
> -
>> 00.txt
>>
>> There is a draft-levy-sip-diversion which I (and many others) believe  
>> has a
>> pretty much zero chance of being accepted or becoming a RFC. This  
>> work has
>> been superceded by the history work. Quite a few people do not  
>> support the
>> history work and it is moving very slowly. Don't expect any of this to
>> resolve quickly.
>>
>> Give some careful thought to if the "To" header meets your needs and  
>> if it
>> does use - The To header is widely implemented :-)
>>
>>
>> Cullen
>>
>>
>>
>>
>> On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
>>
>>> Is there any SIP header defined in any SIP RFC or Internet Draft  
>>> that can
>>> include the address of the 1st callee, who redirected the call using  
>>> 302,
>> in
>>> the 2nd outgoing INVITE?
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>>
>> --__--__--
>>
>> Message: 2
>> Date: Mon, 28 Jul 2003 10:17:51 +0200
>> To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping
> <sipping@ietf.org>
>> From: Jiri Kuthan <jiri@iptel.org>
>> Subject: Re: [Sipping] Update to 3pcc
>>
>> I'm actually not convinced that it is proper to show click-to-dial
>> as a motivating example for use of b2bua and label it BCP.
>>
>> Use of REFER seems technically a superior solution worth being called
>> BCP. Roots of the advantages are in a better application of e2e  
>> principle.
>> REFER takes no middle-box b2bua -- it delegates signaling to  
>> end-points.
>> Advantages include:
>> - All the pitfalls with media are so straigforward, they don't even
>>  need to be described (like now in sessions 8 and 9).
>> - It takes no media server to transmit ringing tone.
>> - There are no SDP compatibility issues on controller's side.
>>
>> -jiri
>>
>> At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
>>> Folks,
>>>
>>> You may have noticed that I posted an update to the 3pcc spec:
>>>
>>> http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
>>> http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
>>> http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
>>>
>>> The update is based on comments received from IESG while reviewing  
>>> the
>> draft. In particular, there were problems with the usage of 0.0.0.0 to
>> imply that no media nor RTCP should be sent. This IP address is not  
>> meant
>> for that purpose. Rather, the .invalid TLD is much better suited. So,  
>> I
>> changed the draft to make use of .invalid in flows where the  
>> controller
>> doesnt want media to be sent.
>>>
>>> Now, there are some concerns with that which folks need to be aware  
>>> of:
>>>
>>> 1. It is at odds with RFC 3264, which actually recommends 0.0.0.0. In
> fact,
>> the problem is really with RFC 3264, NOT 3pcc. However, 3264 is done,  
>> and
>> this one isnt. I logged a bug against RFC 3264 on this, and we'll  
>> need to
>> work on a way to transition folks towards using .invalid and not  
>> 0.0.0.0.
>>>
>>> 2. One the premises of 3pcc is that it should be compatible with any  
>>> UA
>> that is RFC 3261 compliant. If a controller uses .invalid in the SDP,  
>> I am
>> not sure what a SIP endpoint would do. I don't believe anything in  
>> RFC 3264
>> or in SIP talks about handling the case where there is an issue with a
>> domain name in the SDP. As such, if you use .invalid, there could be
> interop
>> problems. The document mentions this.
>>>
>>> Comments or questions welcome.
>>>
>>> Thanks,
>>> Jonathan R.
>>> -- 
>>> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>> Chief Technology Officer                    Parsippany, NJ 07054-2711
>>> dynamicsoft
>>> 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
>>
>> --
>> Jiri Kuthan            http://iptel.org/~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
>>
>> End of Sipping Digest
>>
>> _______________________________________________
>> 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
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Subject: [Sipping] I-D ACTION:draft-ietf-sipping-qsig2sip-02.txt
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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		: Interworking between SIP and QSIG
	Author(s)	: J. Elwell et al.
	Filename	: draft-ietf-sipping-qsig2sip-02.txt
	Pages		: 57
	Date		: 2003-8-13
	
This document specifies interworking between the Session Initiation 
Protocol (SIP) and QSIG within corporate telecommunication networks 
(also known as enterprise networks). SIP is an Internet application-
layer control (signalling) protocol for creating, modifying, and 
terminating sessions with one or more participants. These sessions 
include, in particular, telephone calls. QSIG is a signalling 
protocol for creating, modifying and terminating circuit-switched 
calls, in particular telephone calls, within Private Integrated 
Services Networks (PISNs). QSIG is specified in a number of ECMA 
Standards and published also as ISO/IEC standards. 
As the support of telephony within corporate networks evolves from 
circuit-switched technology to Internet technology, the two 
technologies will co-exist in many networks for a period, perhaps 
several years. Therefore there is a need to be able to establish, 
modify and terminate sessions involving a participant in the SIP 
network and a participant in the QSIG network. Such calls are 
supported by gateways that perform interworking between SIP and QSIG. 
This document is a product of the authors' activities in ECMA 
(www.ecma.ch) on interoperability of QSIG with IP networks.

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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.

--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:	<2003-8-13102145.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-sipping-qsig2sip-02.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-sipping-qsig2sip-02.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-8-13102145.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



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--MIMEStream=_0+219613_9521271356181_57113941163

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		: Interworking between SIP and QSIG
	Author(s)	: J. Elwell et al.
	Filename	: draft-ietf-sipping-qsig2sip-02.txt
	Pages		: 57
	Date		: 2003-8-13
	
This document specifies interworking between the Session Initiation 
Protocol (SIP) and QSIG within corporate telecommunication networks 
(also known as enterprise networks). SIP is an Internet application-
layer control (signalling) protocol for creating, modifying, and 
terminating sessions with one or more participants. These sessions 
include, in particular, telephone calls. QSIG is a signalling 
protocol for creating, modifying and terminating circuit-switched 
calls, in particular telephone calls, within Private Integrated 
Services Networks (PISNs). QSIG is specified in a number of ECMA 
Standards and published also as ISO/IEC standards. 
As the support of telephony within corporate networks evolves from 
circuit-switched technology to Internet technology, the two 
technologies will co-exist in many networks for a period, perhaps 
several years. Therefore there is a need to be able to establish, 
modify and terminate sessions involving a participant in the SIP 
network and a participant in the QSIG network. Such calls are 
supported by gateways that perform interworking between SIP and QSIG. 
This document is a product of the authors' activities in ECMA 
(www.ecma.ch) on interoperability of QSIG with IP networks.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-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-qsig2sip-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-qsig2sip-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.

--MIMEStream=_0+219613_9521271356181_57113941163
Content-Type: Multipart/Alternative; boundary="MIMEStream=_1+144097_8712534892177_4407090204"


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ENCODING mime
FILE /internet-drafts/draft-ietf-sipping-qsig2sip-02.txt

--MIMEStream=_1+144097_8712534892177_4407090204
Content-Type: Message/External-body; name="draft-ietf-sipping-qsig2sip-02.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-8-13102145.I-D@ietf.org>

--MIMEStream=_1+144097_8712534892177_4407090204--
--MIMEStream=_0+219613_9521271356181_57113941163--

_______________________________________________
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 exim@www1.ietf.org  Wed Aug 13 11:08:40 2003
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Subject: Re: [Sipping] DTMF: Why Not A New Method?
From: Dean Willis <dean.willis@softarmor.com>
To: Tim Melanchuk <timm@convedia.com>
Cc: Adam Roach <adam@dynamicsoft.com>, Cullen Jennings <fluffy@cisco.com>,
        sipping <sipping@ietf.org>
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On Mon, 2003-08-11 at 17:12, Tim Melanchuk wrote:
> fine, i'm all for precisely defined semantics.
> 
> the app-interaction-framework seems to me to identify a requirement
> for an application to asynchronously push an interface component.
> am i reading this correctly? if not, fine. if so, what method should
> be used there, and, if we need a new method for that as well, perhaps
> we should look at 2 new methods. one to "request" stimulus and one
> to "convey" the requested stimulus.
> 

We already have an extensible mechanism for asynchronous non-dialog
"push" type operations.

I really don't yet understand why the UA that supports
interface-component notification doesn't just SUBSCRIBE to the
interface-component-notification event package and then get NOTIFYed
with the new interface component.

--
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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Subject: Re: [Sipping] Re: Update to 3pcc
From: Dean Willis <dean.willis@softarmor.com>
To: Rohan Mahy <rohan@cisco.com>
Cc: David Schwartz <davids@deltathree.com>,
        "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
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Rohan said:
> I think it is completely reasonable to build a pre-paid calling system  
> using a proxy with AAA and session timer and optionally a cooperating  
> UA (for IVR features).

Just as billing from session-timer is not a use case, neither is prepaid
control using session timer appropriate. We designed session-timer to
have a resolution on the order of minutes or hours, not seconds. It's
for cleaning up state when stuff goes wrong -- basically, a soft-crash
mechanism. One doesn't drive a car by manipulating its air-bags or
discard the brakes because one's seat belt is buckled.

It sounds to me like you're introducing a time-bounded variant of the
authorization-token header. Something like that COULD be used by a
cooperating UA (say, the PSTN gateway) with reasonable success.

--
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



From exim@www1.ietf.org  Wed Aug 13 11:47:32 2003
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Date: Wed, 13 Aug 2003 11:46:09 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Bernard Aboba <aboba@internaut.com>
CC: john.loughney@nokia.com, sipping@ietf.org
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
References: <DADF50F5EC506B41A0F375ABEB32063658F247@esebe023.ntc.nokia.com> <Pine.LNX.4.53.0308062306320.21030@internaut.com>
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Bernard Aboba wrote:


>>This is one area in which RADIUS would need fixing.
> 
> 
> The problem of course is that one cannot "fix" these problems without
> upgrading deployed implementations.

I suspect that, given the huge problems that have occurred with 
RADIUS, operators would be amenable to upgrading these boxes if 
vendors could provide fixes that were backwards compatible and gave 
them a path towards improving overall performance as other boxes were 
upgraded too. I also suspect RADIUS server vendors would welcome 
standardized ways to achieve those goals.

It seems to me that exponential backoff in retransmits is certainly 
possible in a backwards compatible way. I'm not enough of a RADIUS 
expert to know whether the watchdog mechanism as described in rfc3539 
is possible in a backwards compatible way.

BTW, while on the subject of rfc3539, I am aware of many sip 
implementations which perform watchdog operations very similar to 
those described in section 3.4. I wonder if it would be useful to have 
a sipping specification that describes how to properly do this in sip? 
  There are many ways to do this wrong, and we could leverage the 
thinking Bernard has put into the AAA document to write up a version 
for SIP that works well.

-Jonathan R.




-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Garland Sharratt <gsharratt@convedia.com>
CC: De Leo Walter <deleow@TELEFONICA.COM.AR>, sipping@ietf.org
Subject: Re: [Sipping] draft-melanchuk-sipping-moml-00.txt
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I would think we would need to start with some requiremetns documents=20
first, no?

-Jonathan R.

Garland Sharratt wrote:
> Thanks for the encouragement.  We are certainly favorable to this.  We
> designed MSML/MOML to be a powerful and generic system for advanced con=
trol
> of media servers using SIP, and do hope that SIPPING takes on the draft=
s as
> WG items at some point.  Since the drafts were only released in late Ju=
ne,
> though, I think they probably need a bit more time for review and conce=
nsus
> before being formally advanced as WG items.  We would certainly welcome=
 any
> comments you (or others) might have at this point.
>=20
> Garland
>=20
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf O=
f
> De Leo Walter
> Sent: Thursday, August 07, 2003 08:04
> To: 'sipping@ietf.org'
> Subject: RE: [Sipping] draft-melanchuk-sipping-moml-00.txt
>=20
>=20
> I am interested in knowing if this drafat an msml draft will be propose=
d as
> workgroup items.
>=20
> I think it should there are some proprietary solutions for ms cotnrol o=
ver
> SIP.
>=20
> Regards
>=20
>=20
>>-----Mensaje original-----
>>De:	Tim Melanchuk [SMTP:timm@convedia.com]
>>Enviado el:	Mi=E9rcoles 6 de Agosto de 2003 21:46
>>Para:	Chris Boulton
>>CC:	Garland Sharratt; sipping@ietf.org
>>Asunto:	Re: [Sipping] draft-melanchuk-sipping-moml-00.txt
>>
>>thanks again for your feedback. answers to questions below.
>>
>>cheers,
>>timm
>>
>>
>>>- Is there an alternative method for defining unlimited occurrences
>>>using the 'max' attribute =3D 0 (e.g. pattern element uses max within =
DTMF
>>
>>>primitive).  This seemed a little confusing.
>>>
>>
>>i've had this comment from others. would extending the datatype
>>to include an explicit "unlimited" value be less confusing and/or
>>renaming "max" to "maxoccurs"?
>>
>>
>>>- 4.5.1.1 The description of <tvoice> and <tsilence> needs to be made
>>>more explicit.
>>>
>>
>>will do.
>>
>>
>>
>>_______________________________________________
>>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
>=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
>=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

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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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Subject: RE: [Sipping] DTMF: Why Not A New Method?
Date: Wed, 13 Aug 2003 12:44:04 -0400
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Thread-Topic: [Sipping] DTMF: Why Not A New Method?
Thread-Index: AcNhrTFlNd0vWOl2TF2IXMPyYpBElAADNQTQ
From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "Adam Roach" <adam@dynamicsoft.com>, "sipping" <sipping@ietf.org>
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Folks:

Unless Adam would explain to us why MESSAGE is not good to transfer
DTMF, we would probably not know precisely why we need to a new METHOD
for DTMF reporting.

Similarly, Jonathan also explained why INFO is a problem for DTMF
reporting.

The semantics of all other methods in SIP seem to be pretty clear how to
use them, and no controversies appear how and when to use them to
satisfy many needs.

Can I request Adam and Jonathan to write the BCP drafts for both MESSAGE
and INFO method, respectively, explaining the things how to use these
methods as they have been explaining in the email threads?

Best regards,

Radhika R. Roy
rrroy@att.com

PS: Vendors are implementing many things using INFO and MESSAGE method,
and let us minimize those non-interoperable implementations.

_______________________________________________
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, 13 Aug 2003 12:50:46 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Stupka Jean-Marie <jean-marie.stupka@siemens.com>
CC: "'Rohan Mahy'" <rohan@cisco.com>,
        "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>, alan.johnston@mci.com,
        sipping@ietf.org
Subject: Re: AW: AW: AW: AW: AW: [Sipping] Question on INVITE Initiated Dialog
  Eve nt P ackage and Automatic Callback
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Jean-Marie,

Myself, and many others on this list, appear to be confused as to why
you believe what is currently specified will not work. You had
previously made this comment:

> The transaction being terminated, the data of the call attempt are
> deleted and thus the data (including user B's number) are no more
> available for a subsequent automatic callback request, right? If I
> am wrong, could you please show me where it is said in RFC3261 that
> a SIP user agent has to keep data of a transaction that is
> terminated.

Your assumption is that this data needs to exist for the service to 
work, but I dont see why. The caller has the original URI it used to 
call B. So, it subscribes to B. This gets routed to B's device once 
more. B accepts the subscription. Now, B reports information on 
whether or not it has any active dialogs. If its still in a call, it 
will, and the caller knows this. When B no longer has any active 
calls, the caller gets a notification and can now initiate a new INVITE.

What doesn't work, in your mind?

Perhaps you are concerned about authorization? i.e., how does B know 
whether to authorize the subscription? THe dialog package talks about 
this, and it says to provide summary information, identical to what a 
random caller would know if they simply sent INVITEs.

Perhaps you are concerned about routing? i.e., that the SUBSCRIBE 
would somehow get routed to a different device? Thats possible. The 
GRUU work would resolve that. However, I would argue that an even 
better way to implement this service is to use PRESENCE.

Instead of having the calling device subscribe to the dialog package 
on one device, it would subscribe to presence for the user. One 
problem with the dialog package (which Christian Stredicke has pointed 
out to me) is that if the phone is merely off-hook, and not in an 
actual call, the dialog package would report it as available. However, 
a presence indicator, being an application layer view on state, would 
indicate that the user is not available.

Another benefit of presence is that it works well for multi-device 
situations. It can also cover cases where I'm not actually in a call, 
but busy for some other reason (i.e., in a meeting).

-Jonathan R.


Stupka Jean-Marie wrote:

> The protocols imply procedures, the procedures imply data. If you
> want to offer a network wide service you need the network elements
> and the user equipments to work in a predictable manner with the
> protocols, the procedures and the data.
> 
> What is basically said in the previous mails is that standard
> compliant equipment is nice, but to really work, it needs more. I
> can easily understand that an operator can't accept this view.
> 
> Regards, Jean-Marie
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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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Date: Wed, 13 Aug 2003 13:20:29 -0400
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: AW: AW: AW: AW: AW: [Sipping] Question on INVITE Initiated
  Dialog Eve nt P ackage and Automatic Callback
Cc: Stupka Jean-Marie <jean-marie.stupka@siemens.com>,
        "'Rohan Mahy'" <rohan@cisco.com>,
        "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>, alan.johnston@mci.com,
        sipping@ietf.org
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Jonathan and Jean-Marie,

There are 3 dialogs here:
INVITE resulting in busy
SUBSCRIPTION resulting in notify when free
INVITE resulting in completion of call.

Each of these can be independently interworked to the corresponding SS7 
messages on the PSTN side.

One question is which if any of these dialogs needs to be bound together in 
any way.  I would contend that it is not necessary to relate the first two 
in any way.  The common information element is the called party number 
which is known when the INVITE was sent.  One could conceivably launch the 
subscription without the first INVITE ever having occurred and the 
interworking should be identical.  A Subscription probe should translate to 
a CCBS probe, which is either accepted or rejected.  I don't believe this 
causes any problems for the PSTN EO.  (Someone please correct me if not true!)

A second question is whether the indication of "party free" from CCBS 
interworked to a Notify should somehow provide some information (e.g. a 
token in a Contact address) that could be used in an INVITE interworked 
back to an IAM to distinguish the CCBS IAM from any other IAM.

Question:  Does that busy subscriber's EO upon notification of "subscriber 
free" to a camped-on subscriber block IAMs not containing the appropriate 
CCBS indicator?  Otherwise, I am assuming that IAMs are treated 
first-come-first-served.

Either way, there is no issue with this at the interworking point because 
such a Contact header with a CCBS indicator token is possible in the 200 OK 
SUBSCRIBE and Notify messages.  As I understand it, the nature of the 
user-part of the user@domain address is completely up to the GW to fashion 
if the domain is that GW, and thus does not need standardization.  That 
address could be used in the INVITE in the 3rd dialog and interworked to an 
IAM.  If the SIP UAC doesn't use the contact, but instead just reuses the 
Called Party number, then a normal IAM will result and be treated accordingly.

Mike

At 12:50 PM 8/13/2003 -0400, Jonathan Rosenberg wrote:
>Jean-Marie,
>
>Myself, and many others on this list, appear to be confused as to why
>you believe what is currently specified will not work. You had
>previously made this comment:
>
>>The transaction being terminated, the data of the call attempt are
>>deleted and thus the data (including user B's number) are no more
>>available for a subsequent automatic callback request, right? If I
>>am wrong, could you please show me where it is said in RFC3261 that
>>a SIP user agent has to keep data of a transaction that is
>>terminated.
>
>Your assumption is that this data needs to exist for the service to work, 
>but I dont see why. The caller has the original URI it used to call B. So, 
>it subscribes to B. This gets routed to B's device once more. B accepts 
>the subscription. Now, B reports information on whether or not it has any 
>active dialogs. If its still in a call, it will, and the caller knows 
>this. When B no longer has any active calls, the caller gets a 
>notification and can now initiate a new INVITE.
>
>What doesn't work, in your mind?
>
>Perhaps you are concerned about authorization? i.e., how does B know 
>whether to authorize the subscription? THe dialog package talks about 
>this, and it says to provide summary information, identical to what a 
>random caller would know if they simply sent INVITEs.
>
>Perhaps you are concerned about routing? i.e., that the SUBSCRIBE would 
>somehow get routed to a different device? Thats possible. The GRUU work 
>would resolve that. However, I would argue that an even better way to 
>implement this service is to use PRESENCE.
>
>Instead of having the calling device subscribe to the dialog package on 
>one device, it would subscribe to presence for the user. One problem with 
>the dialog package (which Christian Stredicke has pointed out to me) is 
>that if the phone is merely off-hook, and not in an actual call, the 
>dialog package would report it as available. However, a presence 
>indicator, being an application layer view on state, would indicate that 
>the user is not available.
>
>Another benefit of presence is that it works well for multi-device 
>situations. It can also cover cases where I'm not actually in a call, but 
>busy for some other reason (i.e., in a meeting).
>
>-Jonathan R.
>
>
>Stupka Jean-Marie wrote:
>
>>The protocols imply procedures, the procedures imply data. If you
>>want to offer a network wide service you need the network elements
>>and the user equipments to work in a predictable manner with the
>>protocols, the procedures and the data.
>>What is basically said in the previous mails is that standard
>>compliant equipment is nice, but to really work, it needs more. I
>>can easily understand that an operator can't accept this view.
>>Regards, Jean-Marie
>
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>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

--=====================_14231794==_.ALT
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<font size=3D3>Jonathan and Jean-Marie,<br>
<br>
There are 3 dialogs here:<br>
INVITE resulting in busy<br>
SUBSCRIPTION resulting in notify when free<br>
INVITE resulting in completion of call.<br>
<br>
Each of these can be independently interworked to the corresponding SS7
messages on the PSTN side.<br>
<br>
One question is which if any of these dialogs needs to be bound together
in any way.&nbsp; I would contend that it is not necessary to relate the
first two in any way.&nbsp; The common information element is the called
party number which is known when the INVITE was sent.&nbsp; One could
conceivably launch the subscription without the first INVITE ever having
occurred and the interworking should be identical.&nbsp; A Subscription
probe should translate to a CCBS probe, which is either accepted or
rejected.&nbsp; I don't believe this causes any problems for the PSTN
EO.&nbsp; (Someone please correct me if not true!)<br>
<br>
A second question is whether the indication of &quot;party free&quot;
from CCBS interworked to a Notify should somehow provide some information
(e.g. a token in a Contact address) that could be used in an INVITE
interworked back to an IAM to distinguish the CCBS IAM from any other
IAM.<br>
<br>
Question:&nbsp; Does that busy subscriber's EO upon notification of
&quot;subscriber free&quot; to a camped-on subscriber block IAMs not
containing the appropriate CCBS indicator?&nbsp; Otherwise, I am assuming
that IAMs are treated first-come-first-served.<br>
<br>
Either way, there is no issue with this at the interworking point because
such a Contact header with a CCBS indicator token is possible in the 200
OK SUBSCRIBE and Notify messages.&nbsp; As I understand it, the nature of
the user-part of the user@domain address is completely up to the GW to
fashion if the domain is that GW, and thus does not need
standardization.&nbsp; That address could be used in the INVITE in the
3rd dialog and interworked to an IAM.&nbsp; If the SIP UAC doesn't use
the contact, but instead just reuses the Called Party number, then a
normal IAM will result and be treated accordingly.<br>
<br>
Mike<br>
<br>
At 12:50 PM 8/13/2003 -0400, Jonathan Rosenberg wrote:<br>
<blockquote type=3Dcite cite>Jean-Marie,<br>
<br>
Myself, and many others on this list, appear to be confused as to
why<br>
you believe what is currently specified will not work. You had<br>
previously made this comment:<br>
<br>
<blockquote type=3Dcite cite>The transaction being terminated, the data of
the call attempt are<br>
deleted and thus the data (including user B's number) are no more<br>
available for a subsequent automatic callback request, right? If I<br>
am wrong, could you please show me where it is said in RFC3261 that<br>
a SIP user agent has to keep data of a transaction that is<br>
terminated.</blockquote><br>
Your assumption is that this data needs to exist for the service to work,
but I dont see why. The caller has the original URI it used to call B.
So, it subscribes to B. This gets routed to B's device once more. B
accepts the subscription. Now, B reports information on whether or not it
has any active dialogs. If its still in a call, it will, and the caller
knows this. When B no longer has any active calls, the caller gets a
notification and can now initiate a new INVITE.<br>
<br>
What doesn't work, in your mind?<br>
<br>
Perhaps you are concerned about authorization? i.e., how does B know
whether to authorize the subscription? THe dialog package talks about
this, and it says to provide summary information, identical to what a
random caller would know if they simply sent INVITEs.<br>
<br>
Perhaps you are concerned about routing? i.e., that the SUBSCRIBE would
somehow get routed to a different device? Thats possible. The GRUU work
would resolve that. However, I would argue that an even better way to
implement this service is to use PRESENCE.<br>
<br>
Instead of having the calling device subscribe to the dialog package on
one device, it would subscribe to presence for the user. One problem with
the dialog package (which Christian Stredicke has pointed out to me) is
that if the phone is merely off-hook, and not in an actual call, the
dialog package would report it as available. However, a presence
indicator, being an application layer view on state, would indicate that
the user is not available.<br>
<br>
Another benefit of presence is that it works well for multi-device
situations. It can also cover cases where I'm not actually in a call, but
busy for some other reason (i.e., in a meeting).<br>
<br>
-Jonathan R.<br>
<br>
<br>
Stupka Jean-Marie wrote:<br>
<br>
<blockquote type=3Dcite cite>The protocols imply procedures, the procedures
imply data. If you<br>
want to offer a network wide service you need the network elements<br>
and the user equipments to work in a predictable manner with the<br>
protocols, the procedures and the data.<br>
What is basically said in the previous mails is that standard<br>
compliant equipment is nice, but to really work, it needs more. I<br>
can easily understand that an operator can't accept this view.<br>
Regards, Jean-Marie<br>
</blockquote><br>
-- <br>
Jonathan D. Rosenberg,
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;
600 Lanidex Plaza<br>
Chief Technology
Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Parsippany, NJ 07054-2711<br>
dynamicsoft<br>
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (973) 952-5050<br>
<a href=3D"http://www.jdrosen.net=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0/" eudora=3D"autourl">http://www.jdrosen.net&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</a> PHONE: (973) 952-5000<br>
<a href=3D"http://www.dynamicsoft.com/"=
 eudora=3D"autourl">http://www.dynamicsoft.com</a><br>
<br>
<br>
_______________________________________________<br>
Sipping mailing list&nbsp;
<a href=3D"https://www1.ietf.org/mailman/listinfo/sipping"=
 eudora=3D"autourl">https://www1.ietf.org/mailman/listinfo/sipping</a><br>
This list is for NEW development of the application of SIP<br>
Use sip-implementors@cs.columbia.edu for questions on current sip<br>
Use sip@ietf.org for new developments of core
SIP</font></blockquote></html>

--=====================_14231794==_.ALT--


_______________________________________________
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: Adam Roach <adam@dynamicsoft.com>
To: "'Rohan Mahy'" <rohan@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
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	 <alan.johnston@mci.com>,
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Subject: RE: [Sipping] Re: Comment on GRUU requirements
Date: Wed, 13 Aug 2003 12:55:47 -0500
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Rohan Mahy [mailto:rohan@cisco.com] writes:

> Most of the SIP phones I see at SIPit support 
> completing a transfer using a REFER with a
> Replaces header embedded in the Refer-To. 
> The Refer-To URI is always set to the Contact
> URI of the appropriate UA.

And this will continue to work as well as it does today.
Newer, GRUU aware clients will simply exhibit an additional
robustness.

It seems a bit misguided to do things in a suboptimal way
just to try to fix currently deployed clients.

/a

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Rohan Mahy wrote:

> 

>> I don't think we have so much of either of these deployed that this 
>> cannot change.
> 
> 
> I disagree.  Most of the SIP phones I see at SIPit support completing a 
> transfer using a REFER with a Replaces header embedded in the Refer-To. 
>  The Refer-To URI is always set to the Contact URI of the appropriate UA.

Well, we have here the standard problem here of having implementations 
of long-running drafts, and then deciding down the road that something 
needs to change.

I think we want a non-GRUU in the Contact header (to avoid spirals 
through proxies, and needless load when the proxy doesnt intend to 
record-route), but we want a GRUU for attended-transfer. I would 
propose to use a separate header to identify GRUUs. Existing phones 
will work as they currently do, and when two phones both support GRUU, 
you get more reliable service. I recognize that putting the GRUU in 
the contact would mean only one of the phones needs to support it 
(clearly better), but the cost of this is a long-term situation that 
we don't want.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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] Question on pick-up in service-examples-04
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Old email, and a sort-of-related point.

It seems to me that the INVITE/Replaces really needs to be targeted at 
the GRUU for Alice's phone, no? Otherwise this service won't work in 
places where Carol is located in a different network. For example, if 
Carol takes here SIP phone to a hotel, she won't be able to directly 
reach the contact, and so the pickup will fail. I think we want pickup 
services to work independent of the physical locations of the two devices.

In that case what is more useful in the dialog event package is the 
gruu, not just the contact.

-Jonathan R.

Alan Johnston wrote:

> John,
> 
> You are correct - the NOTIFY F5 does not show the Contact URI.  It 
> should have been shown in a Remote Target tag.  However, this tag is 
> listed as optional in the dialog package - I'm not sure why as it is 
> useful in a number of situations.
> 
> I agree that it is important to use the Contact URI in this flow.
> 
> Thanks,
> Alan Johnston
> MCI
> 
> At 09:41 AM 7/10/2003 +0100, Elwell, John wrote:
> 
>> Can anyone explain something to me in section 2.16 (call pick-up)?
>>
>> In F7, the Request-URI is the contact URI for Alice: 
>> alice@client.atlanta.example.com
>>
>> It is not clear how Carol's UA receives this contact address. F5 from 
>> Bob to Carol provides only the AoR: alice@atlanta.example.com.
>>
>> I believe it is important to use the contact URI in F7 to avoid the 
>> case where the INVITE request reaches the wrong UA at Alice. This is 
>> particularly true when Alice is in a legacy network - the contact 
>> address might help selection of the correct gateway.
>>
>> John (john.elwell@siemens.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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Sorry for the delay. Responses inline.

Paul Kyzivat wrote:

> I'm way behind in my mail and just discovered this discussion. Earlier I 
> had actually discovered the new draft and posted a question on this very 
> point without first finding this conversation. I apologize for that.
> 
> Could this problem not be dealt with in a backward compatible manner by 
> using a domain name that actually exists in the public DNS and resolves 
> to the address 0.0.0.0?

I'm not sure how this would work in practice. Who would own such a 
domain? Why would they agree to manage it in this way? I guess it 
would need to be some kind of public service, managed by IETF or 
something. I don't like the idea of tying the operation of the 
protocol to something like this.


> 
> I don't see how Jonathan's alternative helps in practice. The .invalid 
> address often is needed in an initial offer, so there has as yet been no 
> opportunity to exchange option tags. So in this case it would require 
> starting with a Require of the option, which will result in the request 
> failing, and the need to retry with 0.0.0.0 in that case. So everybody 
> needs to support both ways, and for awhile we get lots of extra 
> messaging incurred when somebody tries to do the right thing. Anybody in 
> their right mind is just going to start out using 0.0.0.0, because it is 
> more likely to work on the first try.

I agree this is a problem.

I don't really have a better solution. All I can say is that the only 
flow which has an initial offer with SDP of 0.0.0.0 is flow II, which 
is not one of the recommended flows. As such, much of the usages of 
0.0.0.0 will be for suspending streams post call setup, and those 
would not have this problem of extra RTT; the initial INVITE would 
indicated that .invalid is supported.

So, I'd propose to still go ahead and revert back to 0.0.0.0 with some 
caveats, and then submit a separate I-D on using .invalid.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Date: Wed, 13 Aug 2003 15:17:52 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
CC: David Schwartz <davids@deltathree.com>,
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Subject: Re: [Sipping] Re: Update to 3pcc
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Sorry for taking so long to weigh in here.

To address the specific billing issue, the problem is that, in fact, 
there ARE cases where the call is charged for differently if it was 
initiated as a result of click-to-dial. The user that makes the 
initial INVITE may not be the one that needs to be charged. There may 
be ways around that for REFER by having the INVITE carry some kind of 
special billing tokens or something (such tokens being pushed to the 
user in the Refer-To), but these then require the c2d provider and the 
caller to utilize the same provider. The 3pcc mechanism does not have 
this constraint.

However, in my mind, the real issue is one of control. 3pcc is useful 
in applications where the server needs to be in the signaling path in 
order to do or modify things. Thats not the case for all applications, 
but its true for many. Prepaid is one we have discussed as well. REFER 
just doesnt work there.

I'd also point out that the 3pcc spec is not saying that the best 
practice for c2d is to use 3pcc; it saying that if you want to do 
third party call control, here is how you do it.

I don't disagree that 3pcc for c2d can be complicated, and REFER does 
have a better story in terms of the media story. However, other work 
going on, namely the app interaction work, does address some of the 
big issues there in a pretty elegant way, I think.

You do raise a valid concern about security. The story there could 
definitely be better. I've been considering, for some time, writing 
down some requirements for a standards-based 3pcc mechanism (as 
opposed to a practice for using current mechanisms) that would do a 
few things just a little differently to achieve much better properties 
overall. In particular, note how flow I has a nice property that the 
b2bua doesnt actually need to touch the SDP - it can, therefore, flow 
e2e even if its encrypted. Flow I has other problems that prevent it 
from being used generally. Some actual SIP extensions could be used to 
rectify that.

However, this is our best current practice for doing 3pcc with what we 
have, and so we are constrained by 3261.

-Jonathan R.



Jiri Kuthan wrote:

> Hello David,
> 
> I don't see how billing servers are related to method of 
> call initiation.
> 
> Actually I believe that how a call is initiated (by manual
> dialing a destination, by REFER-based click-to-dial, by b2bua-based
> click-to-dial, CTI, etc.) is completely orthogonal to how the 
> call is charged for. Regardless what charging system you use 
> (if any), there will be different methods for initiating calls, 
> some will be better off than others. I maintain that a click-to-dial 
> b2bua is not a BCP.
> 
> -jiri
> 
> At 10:10 AM 7/30/2003, David Schwartz wrote:
> 
>>Hi Jiri
>>
>>REFER is good solution if you assume that either there are no billing
>>issues, or alternatively, that the endpoint/s handle the billing issues. If
>>this is not the case, for example, in an application where a centralized
>>controller is setting up a call to two PSTN numbers, then you will need a
>>centralized billing entity and REFER will not suffice.
>>
>>David
>>
>>-----Original Message-----
>>From: sipping-request@ietf.org [mailto:sipping-request@ietf.org]
>>Sent: Mon, July 28, 2003 7:00 PM
>>To: sipping@ietf.org
>>Subject: Sipping digest, Vol 1 #1144 - 2 msgs
>>
>>
>>Send Sipping mailing list submissions to
>>       sipping@ietf.org
>>
>>To subscribe or unsubscribe via the World Wide Web, visit
>>       https://www1.ietf.org/mailman/listinfo/sipping
>>or, via email, send a message with subject or body 'help' to
>>       sipping-request@ietf.org
>>
>>You can reach the person managing the list at
>>       sipping-admin@ietf.org
>>
>>When replying, please edit your Subject line so it is more specific
>>than "Re: Contents of Sipping digest..."
>>
>>
>>Today's Topics:
>>
>>  1. Re: Diversion header (Cullen Jennings)
>>  2. Re: Update to 3pcc (Jiri Kuthan)
>>
>>--__--__--
>>
>>Message: 1
>>Date: Sun, 27 Jul 2003 19:05:10 -0700
>>Subject: Re: [Sipping] Diversion header
>>From: Cullen Jennings <fluffy@cisco.com>
>>To: Romel Khan <rkhan@net2phone.com>, <sipping@ietf.org>
>>
>>
>>I assume your question is there some way for  for the final destination that
>>receives the call to know the who the 1st callee was that had some device in
>>the network redirect the call elsewhere.
>>
>>The answer is YES - there has been a lot of confusion over this so think
>>about my answer carefully - it will seem counter intuitive at first. The
>>answer is the "To" header but it will take me a second to explain why.
>>
>>Imagine case 1 where A send a call to proxy B,  that sent to to C. Now C
>>sent a 302 back B and B then forked the call to D. The To header in the
>>message D receives will still have the To set to C. C is the element that
>>sent the first 302.
>>
>>Now imagine a case (like your question) where there is not a recursive
>>proxy. The 302 will go all the way back to A. Now A forms a new invite to
>>but it can leave the To header set to the same thing it was in the original
>>message (C in this case). When the message arrives at D, the To header will
>>be set to C.
>>
>>If D was a voice mail system, it might be able to know to act on behalf of C
>>instead of some other entity.
>>
>>There are more complex things that relate to your request. Mary Barnes has
>>been moving the history work along
>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-req-history-04.txt
>>
>>There is some work on QSIG mappings with diversion
>>http://www.ietf.org/internet-drafts/draft-elwell-sipping-qsig2sip-diversion-
>>00.txt
>>
>>There is a draft-levy-sip-diversion which I (and many others) believe has a
>>pretty much zero chance of being accepted or becoming a RFC. This work has
>>been superceded by the history work. Quite a few people do not support the
>>history work and it is moving very slowly. Don't expect any of this to
>>resolve quickly.
>>
>>Give some careful thought to if the "To" header meets your needs and if it
>>does use - The To header is widely implemented :-)
>>
>>
>>Cullen
>>
>>
>>
>>
>>On 7/21/03 8:55, "Romel Khan" <rkhan@net2phone.com> wrote:
>>
>>
>>>Is there any SIP header defined in any SIP RFC or Internet Draft that can
>>>include the address of the 1st callee, who redirected the call using 302,
>>
>>in
>>
>>>the 2nd outgoing INVITE?
>>>
>>>_______________________________________________
>>>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
>>>
>>
>>
>>
>>--__--__--
>>
>>Message: 2
>>Date: Mon, 28 Jul 2003 10:17:51 +0200
>>To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping <sipping@ietf.org>
>>From: Jiri Kuthan <jiri@iptel.org>
>>Subject: Re: [Sipping] Update to 3pcc
>>
>>I'm actually not convinced that it is proper to show click-to-dial
>>as a motivating example for use of b2bua and label it BCP.
>>
>>Use of REFER seems technically a superior solution worth being called
>>BCP. Roots of the advantages are in a better application of e2e principle. 
>>REFER takes no middle-box b2bua -- it delegates signaling to end-points.
>>Advantages include:
>>- All the pitfalls with media are so straigforward, they don't even 
>> need to be described (like now in sessions 8 and 9).  
>>- It takes no media server to transmit ringing tone. 
>>- There are no SDP compatibility issues on controller's side.
>>
>>-jiri
>>
>>At 07:51 PM 7/8/2003, Jonathan Rosenberg wrote:
>>
>>>Folks,
>>>
>>>You may have noticed that I posted an update to the 3pcc spec:
>>>
>>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.txt
>>>http://www.jdrosen.net/papers/draft-ietf-sipping-3pcc-04.html
>>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-3pcc-04.txt
>>>
>>>The update is based on comments received from IESG while reviewing the
>>
>>draft. In particular, there were problems with the usage of 0.0.0.0 to
>>imply that no media nor RTCP should be sent. This IP address is not meant
>>for that purpose. Rather, the .invalid TLD is much better suited. So, I
>>changed the draft to make use of .invalid in flows where the controller
>>doesnt want media to be sent.
>>
>>>Now, there are some concerns with that which folks need to be aware of:
>>>
>>>1. It is at odds with RFC 3264, which actually recommends 0.0.0.0. In fact,
>>
>>the problem is really with RFC 3264, NOT 3pcc. However, 3264 is done, and
>>this one isnt. I logged a bug against RFC 3264 on this, and we'll need to
>>work on a way to transition folks towards using .invalid and not 0.0.0.0.
>>
>>>2. One the premises of 3pcc is that it should be compatible with any UA
>>
>>that is RFC 3261 compliant. If a controller uses .invalid in the SDP, I am
>>not sure what a SIP endpoint would do. I don't believe anything in RFC 3264
>>or in SIP talks about handling the case where there is an issue with a
>>domain name in the SDP. As such, if you use .invalid, there could be interop
>>problems. The document mentions this.
>>
>>>Comments or questions welcome.
>>>
>>>Thanks,
>>>Jonathan R.
>>>-- 
>>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>>dynamicsoft
>>>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
>>
>>--
>>Jiri Kuthan            http://iptel.org/~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
>>
>>End of Sipping Digest
>>
>>_______________________________________________
>>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.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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



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Henry Sinnreich wrote:

>  > * The mechanism must provide the ability to inform the UA 
> 
>>about the set of session policies when the device starts up
> 
> 
>>* The mechanism must provide a way to inform the client about 
>>changes in session policies when they occur
> 
> 
> Would these lead to requirements for SIP telephony devices? Plus countelss
> other..

Yes, this is one of many things which could be considered part of a 
device recommendation.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sipping] DTMF: Why Not A New Method?
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Having had a chance to think about all of this some more, here are my 
thoughts.

One - I give up on MESSAGE. You (Adam) are right, the semantics don't 
match.

The method we want, I think, has a semantic of "here is the 
information you requested". In this case, it was requested with the 
deposit of the KPML script onto the device. As such, I believe we have 
such a method, and its called NOTIFY (well, I REALLY think its called 
HTTP POST, but I seem to be the only one in that boat...). Also note 
that this is NOT an implicit subscription. It is quite explicit. Its 
just not done with SUBSCRIBE. Given that, lets examine each of the 
concerns you raised about implicit subscriptions and see if they apply:

Adam wrote:
 > - Duration of subscription. RFC 3265 includes an "expires"
 >    parameter on the "Subscription-State" header field for each
 >    NOTIFY. A major implication of this is that you can't
 >    run a subscription for an indeterminate length of time
 >    (such as the duration of an INVITE dialog).  Subscription
 >    duration has two sub-issues:

The duration is exactly equal to the length of the validity of the 
KPML script. That lifetime is defined in the framework. It terminates 
when the dialog terminates. Otherwise, the application can 
"unsubscribe" by pushing a new App-Info header w/o a KPML script for 
that instance (Tim had asked how this is done - see my answer in a 
reply to his note).

 >
 >    - If anything other than a SUBSCRIBE is used to put
 >      the subscription in place, then the subscriber can't
 >      even provide a suggested duration. The notifier must
 >      provide one unilaterally. This is what REFER does.

The duration is as defined above.

 >
 >    - Once this value reaches 0, the subscriber must
 >      resubscribe or the subscription will disappear.
 >      In many cases, this requires an actual SUBSCRIBE
 >      message, which tends to eliminate whatever slight
 >      gain was expected by not having an explicit SUBSCRIBE
 >      in the first place.

Here, the subscription is bound to the lifetime of the dialog, and 
there is no need or possibility to extend it beyond that.

 >
 >  - Correlation. In the general case, whatever mechanism
 >    creates the implicit subscription must include sufficient
 >    information to create a dialog. In the KPML case, it is
 >    not clear how this would happen. For many calls, the
 >    dialog between the calling endpoint and the server
 >    receiving KPML feedback is not the same dialog as the
 >    dialog between the calling and called parties.

In all cases, its not the same dialog.

Correlation is easy. The KPML script IS the subscription. The place to 
send the notify is implied from the URI in the KPML script. We would 
need to clarify in KPML that once that URI is used, subsequent 
notificatiosn to the same URI ought to go on the dialog established by 
the NOTIFY. But, this is a KPML issue.

 >
 >  - Handling of multiple subscriptions in the same dialog.
 >    It was asserted in the SIPPING meeting that this does
 >    not apply to the KPML case. This assertion is *probably*
 >    true, depending on how the "correlation" problem is solved.

See above.

 >
 >  - Filtering. Unless there is a way to add filtering to
 >    the mechanism that creates the subscription, installation
 >    of filters must be provisioned or performed via some
 >    other out-of-band mechanism. With KMPL, it is arguable that
 >    the script itself provides a filtering mechanism.

Exactly. Because the KPML script is the subscription. The digit maps 
in the KPML are EXACTLY a filter.


My argument against a new method, as a result, is the same as the 
general argument against two methods for doing the same thing. I can 
easily see applications later on where, in fact, a SUBSCRIBE gets used 
to get this information, instead of pushing a KPML script in App-Info. 
As a result of this, a NOTIFY would be sent. I think it would be bad 
if the server had to have support for two different methods even 
though they both contain the same information.

Thanks,
Jonathan R.





Adam Roach wrote:
>>will NEW_METHOD also be able to...
> 
> 
> No!
> 
> This is precisely what I'm trying to avoid happening. Methods
> have actual semantics associated with them. They should mean
> exactly one thing, not several. INVITE means you're starting
> a session. SUBSCRIBE means that you're subscribing to the state
> of a resource. MESSAGE means you're sending an instant message.
> 
> The new method that we define for KPML should also mean precisely
> one thing: "I'm conveying requested stimulus from the user."
> 
> If you need to define something else with different semantics
> than that, I would argue that there's a fair chance that a new
> method is indicated.
> 
> Otherwise, the actual method names become meaningless. If that's
> the path we are going down, consistency would argue for deprecating
> all existing methods in favor of a single method (perhaps, "DO"),
> and inferring all semantics from some horribly complex combination
> of Content-Type, Content-Disposition, Accept, Supported, and similar
> header fields.
> 
> /a
> 
> _______________________________________________
> 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.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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
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Tim Melanchuk wrote:

> fine, i'm all for precisely defined semantics.
> 
> the app-interaction-framework seems to me to identify a requirement
> for an application to asynchronously push an interface component.
> am i reading this correctly? if not, fine. if so, what method should
> be used there, and, if we need a new method for that as well, perhaps
> we should look at 2 new methods. one to "request" stimulus and one
> to "convey" the requested stimulus.

The new scripts are pushed in App-Info headers. If the application is 
a UA application, it would presumably use a re-INVITE or INFO. If its 
a proxy that started the application, it can't initiate new messages, 
and therefore it needs to wait for something to fly by in which it can 
insert an App-Info header.

An alternative, which we considered, was to have the application do a 
separate session just for the application interaction. So, it would 
generate an INVITE towards the UA. In that case, a "proxy" would be 
able to initiate pushes of new KPML scripts at any time by sending a 
re-INVITE on the separate dialog it just created, for which it is the 
UA. However, this quickly becomes complicated, with a lot more 
messages being exchanged. It also has the drawback of makign security 
harder. If you use sips, and you have scripts pushed in headers in 
in-dialog messages, you can get the property that only proxies on the 
call path (and the UAS) can send scripts to the UA. THats an important 
property. By allowing separate dialogs for this, you now need a 
different authorization tactic. You'd also need GRUUs to allow the 
server in the network to target a request to that device instance.

I would go so far as to say that I wouldn't even mind eliminating the 
push requirement for new scripts. We don't have this in the web. I 
suspect most applications don't really need it - we seem to get by on 
the web without it.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Use sip@ietf.org for new developments of core SIP



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Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:

> Also note that this is NOT an implicit subscription.
> It is quite explicit. Its just not done with SUBSCRIBE.

You can't define the problem away by twisting commonly accepted
terms in random directions. "Implicit Subscription" is the term
that we, as a community, have historically used to mean "a
subscription created by any means other than a SIP subscribe
request, as described in RFC 3265." It's just shorter to say
"implicit subscription."

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Tim Melanchuk'" <timm@convedia.com>,
        Cullen Jennings
	 <fluffy@cisco.com>, sipping <sipping@ietf.org>
Subject: RE: [Sipping] DTMF: Why Not A New Method?
Date: Wed, 13 Aug 2003 16:31:13 -0500
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Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:

> The duration is exactly equal to the length of the validity of the 
> KPML script. That lifetime is defined in the framework. It terminates 
> when the dialog terminates. Otherwise, the application can 
> "unsubscribe" by pushing a new App-Info header w/o a KPML script for 
> that instance (Tim had asked how this is done - see my answer in a 
> reply to his note).

You missed my point. I'll try to make it more obvious.
Below, I've placed an example NOTIFY message. Find the
"XXXXX" in it, and change it so that it makes syntactic
and semantic sense.

NOTIFY sip:appserver.example.com SIP/2.0
Via: SIP/2.0/TCP terminal.example.com;branch=z9hG4bK4EPlfSFQK1
Max-Forwards: 70
From: <sip:user@example.com>;tag=zpNctbZq
To: <sip:appserver.example.com>;tag=ie4hbb8t
Call-ID: cdB34qLToC@terminal.example.com
CSeq: 997935769 NOTIFY
Contact: <sip:terminal.example.com>
Event: stimulus
Subscription-State: active;expires=XXXXX
Content-Type: application/kpml+xml
Content-Length: 74

<?xml version="1.0">
<kpml version="1.0">
  <response digits="0113224321234"/>
</kpml>

/a

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Subject: RE: [Sipping] Re: Update to 3pcc
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Thread-Topic: [Sipping] Re: Update to 3pcc
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Inline [RRR]

... I'd also point out that the 3pcc spec is not saying that the best=20
practice for c2d is to use 3pcc; it saying that if you want to do=20
third party call control, here is how you do it.

[RRR] Right. The draft shows how to use the 3pcc, if one wants to do so.

... However, this is our best current practice for doing 3pcc with what
we=20
have, and so we are constrained by 3261.

[RRR] I agree.

-Jonathan R.


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Subject: Re: [Sipping] Re: Comment on GRUU requirements
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To: Adam Roach <adam@dynamicsoft.com>
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On Wednesday, August 13, 2003, at 10:55 AM, Adam Roach wrote:
> Rohan Mahy [mailto:rohan@cisco.com] writes:
>> Most of the SIP phones I see at SIPit support
>> completing a transfer using a REFER with a
>> Replaces header embedded in the Refer-To.
>> The Refer-To URI is always set to the Contact
>> URI of the appropriate UA.
>
> And this will continue to work as well as it does today.
> Newer, GRUU aware clients will simply exhibit an additional
> robustness.
>
> It seems a bit misguided to do things in a suboptimal way
> just to try to fix currently deployed clients.

and Jonathan wrote:
>>> I don't think we have so much of either of these deployed that this 
>>> cannot change.
>> I disagree.  Most of the SIP phones I see at SIPit support completing 
>> a transfer using a REFER with a Replaces header embedded in the 
>> Refer-To.  The Refer-To URI is always set to the Contact URI of the 
>> appropriate UA.
>
> Well, we have here the standard problem here of having implementations 
> of long-running drafts, and then deciding down the road that something 
> needs to change.
>
> I think we want a non-GRUU in the Contact header (to avoid spirals 
> through proxies, and needless load when the proxy doesnt intend to 
> record-route), but we want a GRUU for attended-transfer. I would 
> propose to use a separate header to identify GRUUs. Existing phones 
> will work as they currently do, and when two phones both support GRUU, 
> you get more reliable service. I recognize that putting the GRUU in 
> the contact would mean only one of the phones needs to support it 
> (clearly better), but the cost of this is a long-term situation that 
> we don't want.
>
> -Jonathan R.

Hi,

I think both of you are ignoring the "generality before efficiency" 
principle.  Putting the GRUU in the Contact header always works, and in 
most deployments where it is needed (middlebox traversal), you WILL 
need to travel through the same set of proxies for nearly every request 
anyhow. With the increased deployment of TLS and SIPS we will 
increasingly have dialogs for which all or none of the requests are 
proxied.

We are talking about interop with thousands of UAs which have 
implemented Proposed Standard level RFCs (SIP and Refer) and an 
long-lived, stable I-D that has completed WGLC and was submitted to the 
IESG (Replaces)`.  We are not talking about random drafts that very few 
have implemented. To be convinced that it is worth adding a new GRUU 
header, I would at least like to see some plausible use cases that are 
common enough for us to be worrying about.  Maybe the right way to look 
at this is to put the GRUU in the Contact, UNLESS you can tell that 
your peer understands the special significance of GRUUs.

Also, where this extension is most needed, you need the right 2 out of 
3 to support the extension.  (This isn't just any two of three.  If Bob 
supports the GRUU extension, but Carol does not, Alice still cannot 
cause a transfer to occur in the appropriate direction.)

Starting with the classic Alice, Bob, and Carol actors.

Alice and Bob have an existing dialog.
Alice and Carol have an existing dialog.
Alice wants Bob to talk to Carol, and will REFER Bob to INVITE Carol 
replacing the dialog between Alice and Carol.

Carol needs a GRUU; if the GRUU is conveyed in a separate header Alice 
and Carol both need to support the GRUU extension for the Refer-To 
header to be set properly.

thanks,
-rohan


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Date: Wed, 13 Aug 2003 15:17:11 -0700
Subject: Re: [Sipping] Re: Update to 3pcc
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Cc: David Schwartz <davids@deltathree.com>,
        "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
        "'sipping@ietf.org'" <sipping@ietf.org>
To: Dean Willis <dean.willis@softarmor.com>
From: Rohan Mahy <rohan@cisco.com>
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On Wednesday, August 13, 2003, at 08:13 AM, Dean Willis wrote:

>
> Rohan said:
>> I think it is completely reasonable to build a pre-paid calling system
>> using a proxy with AAA and session timer and optionally a cooperating
>> UA (for IVR features).
>
> Just as billing from session-timer is not a use case, neither is 
> prepaid
> control using session timer appropriate. We designed session-timer to
> have a resolution on the order of minutes or hours, not seconds.

In many prepaid applications that is all that is required. For example:

UA sends an INVITE with apropos credentials.
Proxy does a AAA lookup, finds out there is enough credit for 2 hours, 
sets session timer at 15 minutes and proxies the request to a gateway 
with Require: timer.
After 30 minutes the gateway reINVITEs (a timer refresh request).  The 
Proxy again does a AAA lookup.  At this point there is only enough 
credit for 10 minutes (the caller used the prepaid service to make a 
more expensive call). The proxy forwards the request with session timer 
set to 5 minutes.
On the next timer refresh, the proxy forwards the request with a 3 
minute timer.  Finally the proxy uses a 2 minute timer.  When the last 
timer refresh arrives, the Proxy sends a 403, and the gateway tears 
down the session.

This mechanism works very well and is perfectly appropriate if either 
only one session is allowed at a time, or if you are willing to reserve 
resources for an existing session for a few minutes in the future.

> It's for cleaning up state when stuff goes wrong -- basically, a 
> soft-crash
> mechanism.

I have no problem with that.  But it is a perfectly reasonable solution 
for some prepaid applications (not all of them obviously), and requires 
no changes to the session timer spec.

thanks,
-rohan


> One doesn't drive a car by manipulating its air-bags or
> discard the brakes because one's seat belt is buckled.
>
> It sounds to me like you're introducing a time-bounded variant of the
> authorization-token header. Something like that COULD be used by a
> cooperating UA (say, the PSTN gateway) with reasonable success.
>
> --
> Dean


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From: Adam Roach <adam@dynamicsoft.com>
To: sipping <sipping@ietf.org>
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Subject: [Sipping] App-Interaction: Considering models
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After considering Jonathan's contention that defining
a new method for conveying stimulus is creating something
that is similar in meaning to NOTIFY, I'm inclined to
agree to some point. There are still subtle differences
in meaning, so I'm not completely convinced that a new
method is wrong, but I understand the point he's trying
to make.

After discussing it privately with others, I've concluded
that there are three viable models worth considering. There
are, of course, some minor variations that can be made to
each.

1) Standard HTTP Handling

UAC           App Server          UAS
1|-----INVITE---->|                |   
2|                |-----INVITE---->|   
3|                |<-----200-------|   
4|<-----200-------|                |   AS adds HTTP URL to 200
5|---HTTP GET---->|                |   UAC requests KPML script
6|<-----200-------|                |   AS replies with script
7|---HTTP POST--->|                |   UAC provides digits
8|<-----200-------|                |   AS acknowledges digits

  Of the three options presented here, I see this as the best
  solution. Basically, the app server includes an HTTP URL
  in the 200 response to the INVITE that it has proxied. The
  UAC then does a GET on the KPML script. Just like with
  HTML forms, the UAC does whatever local stuff is necessary
  to "fill out the form" (in this case, collect digits), and
  then posts the result back to the App Server. This is all
  plain-Jane unextended HTTP as it was intended to work.

2) Explicit Subscriptions (with GRUU)

UAC           App Server          UAS
1|-----INVITE---->|                |   
2|                |-----INVITE---->|   AS extracts GRUU from INVITE
3|                |<-----200-------|   
4|<-----200-------|                |   
5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
6|------200------>|                |   
7|-----NOTIFY---->|                |   
8|<-----200-------|                |   
9|-----NOTIFY---->|                |   UAC provides digits
A|<-----200-------|                |   AS acknowledges digits

  My second choice would be an explicit subscription to stimulus
  events. The only sticky part in this solution is the question
  of how the AS can route a SUBSCRIBE to the UAC. Once we have
  a GRUU solution in place, this is a no brainer: the AS can
  extract the GRUU from the INVITE, and use it as the target
  of its SUBSCRIBE. The KPML script is included in the SUBSCRIBE,
  and essentially serves as a filter.

3) Explicit Subscriptions (with REFER)

UAC           App Server          UAS
1|-----INVITE---->|                |   
2|                |-----INVITE---->|   
3|                |<-----200-------|   
4|<-----200-------|                |   AS adds SIP URL to 200
5|-----REFER----->|                |   
6|<-----200-------|                |   
7|<----NOTIFY-----|                |   
8|------481------>|                |   
9|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
A|------200------>|                |   
B|-----NOTIFY---->|                |   
C|<-----200-------|                |   
D|-----NOTIFY---->|                |   UAC provides digits
E|<-----200-------|                |   AS acknowledges digits

  Although somewhat chattier than the previous solution, this gets around
  the GRUU issue by having the AS provide a SIP URI that routes to itself
  in the 200 response. The UAC then uses this URI to send a REFER to the
  AS. This REFER tells the AS to SUBSCRIBE to the UAC. The flow from that
  point forward is the same as for solution 2.

/a

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Subject: RE: [Sipping] FWD:I-DACTION:draft-johnston-sipping-rtcp-summary-00.txt
Date: Wed, 13 Aug 2003 18:33:01 -0600
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Hi Alan & all,

Here are some comments on the requirement section of ID
draft-johnston-sipping-rtcp-summary-00.txt (see #jfm marks):

>   REQ-1: An authorized third party should be able to receive
>   selected RTCP reports on a near real time basis.
#jfm
Based on section 3, it seems that this requirement is for an authorized
3rd party to be also able to *request and/or* receive selected RTCP
reports?

>   REQ-2: The client should not have to store large amounts of
>   information.
#jfm
This comment applies to the entire ID.=20
Is client =3D SIP UA? Or RTP host acting as receiver? Or the management
agent storing managed objects for some kind of rtp mib objects?  See
comment on REQ-5 on "participant". A terminology section could help.

>   REQ-3: The client must be able to authenticate the third party.
#jfm
For e.g., the client here could be an snmpv3 agent or a sip entity or
(why not) a netconf xml http client? I guess my point is that refining
the requirement will help define the solution that meets your needs.

>   REQ-4: The RTCP report information must be able to be transferred
>   securely.
#jfm
Could you define precisely what you mean by "securely"?
authentication, confidentiality?

>   REQ-5: Only one participant in the session need support the
>   mechanism - the participant will send both sent and received
>   summary reports.
#jfm
I find this requirement confusing, mostly because I'm not sure I
understand what participant it. Does participant mean RTP/RTCP host or
SIP UA client here?=20
I also wonder how relevant this requirement is. What if
2 SIP UAs using the same signaling proxy participate in a conferencing
session and their respective RTP/RTCP hosts mix media streams coming=20
from distinct RTP/RTCP hosts? How does this requirement work in=20
conferencing?=20


>   REQ-6: The reports will include or be associated with dialog
>   identifiers for correlation purposes.
#jfm
In this case, I think you mean the RTCPx reports delivered via the SIP
dialog. Depending on how this draft evolves, these may be different
from the RTCP reports. It may help to qualify "reports" then.

To that extent and reading thru the email threads on this draft, there
should be a requirement for a 3rd party to request specific RTCP reports
(and we should not exclude SR, RR or BYE because some applications may
actually be interested in those).

Jean-Francois=20

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Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I think all of these proposals are significantly more complex than using a
MESSAGE/NEW_METHOD/NOTIFY. I will mention a few issues with the proposal
inline but first let me a few other proposals...

All of these have a call flow like

> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|
> 3|                |<-----200-------|
> 4|<-----200-------|                |
> 5|---SIP MESG---->|                |
> 6|<-----200-------|                |

The question for KMPL is, what method to use in msg 5.

Proposal 1) Use MESSAGE. Many people think the semantics of message are
display this IM to a user. Perhaps, but I send IM to applications and they
send me back things like stock quotes all the time. I have no problem with
semantics of MESSAGE being deliver this message to the addressed user or
automaton. That's what we are doing here. Misrouted message will cause
trivial problems, but this is a bogus issue - misrouted INVITE or MESSAGE
will cause problems to - don't misroute things.

Proposal 2) Use some new method like STIMULUS. The semantics of this are
some user input stimulus is being sent - STIMULUS is to input what MESSAGE
is to output. 

Proposal 3) Use NOTIFY. Semantics are, here is some information about some
event that you requested to be informed about. The usual complaint on this
one is that there was no Subscribe, or it was implicit, or it was explicit,
or it violated some law of the SIP world. I think there was a subscribe, it
was in the message that initiated the script. There is also a clear time it
will end - when the script is terminated or when the dialog of the message
that initiated the script ends. We all remember that implicit subscribes
where bad (whatever an implicit subscribe is) but people seem to have a hard
time remembering why this will be bad here.

All 3 of these proposals have something in common - there is no concrete
argument that any of them would not work. They send to a URI the AS provided
so there are no forking issues. There is some grief on the semantics. Option
2 has the advantage of it lets us define the semantics however we want. I
think that 2 is the best way to go but I don't care greatly. I do need
convincing on what's wrong with these to move to something more complex.

More comments inline

On 8/13/03 3:32 PM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> After considering Jonathan's contention that defining
> a new method for conveying stimulus is creating something
> that is similar in meaning to NOTIFY, I'm inclined to
> agree to some point. There are still subtle differences
> in meaning, so I'm not completely convinced that a new
> method is wrong, but I understand the point he's trying
> to make.
> 
> After discussing it privately with others, I've concluded
> that there are three viable models worth considering. There
> are, of course, some minor variations that can be made to
> each.
> 
> 1) Standard HTTP Handling
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|
> 3|                |<-----200-------|
> 4|<-----200-------|                |   AS adds HTTP URL to 200
> 5|---HTTP GET---->|                |   UAC requests KPML script
> 6|<-----200-------|                |   AS replies with script
> 7|---HTTP POST--->|                |   UAC provides digits
> 8|<-----200-------|                |   AS acknowledges digits
> 
> Of the three options presented here, I see this as the best
> solution. Basically, the app server includes an HTTP URL
> in the 200 response to the INVITE that it has proxied. The
> UAC then does a GET on the KPML script. Just like with
> HTML forms, the UAC does whatever local stuff is necessary
> to "fill out the form" (in this case, collect digits), and
> then posts the result back to the App Server. This is all
> plain-Jane unextended HTTP as it was intended to work.
> 

One of the difficult parts of this is security.

This will get more complicated once you provide a way to send in message 4
the credentials that will allow 7. It may also make the App-Info header only
really work with scripts that use HTTP to post instead of other things.
Also, in the case of a financial institutions that have to log electronic
communications, you will need to provide way to insert logging HTTP proxies
and probably proxies for FW traversal too. Of course the AS or the UAC will
need to figure out the network topology to know if the AS is inside or
outside the FW. It's possible to configure all the domain and HTTP proxies
rules into the device but it still hard to secure - particularly on devices
that don't want to implement HTTPS due to not much memory.


> 2) Explicit Subscriptions (with GRUU)
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
> 3|                |<-----200-------|
> 4|<-----200-------|                |
> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> 6|------200------>|                |
> 7|-----NOTIFY---->|                |
> 8|<-----200-------|                |
> 9|-----NOTIFY---->|                |   UAC provides digits
> A|<-----200-------|                |   AS acknowledges digits
> 
> My second choice would be an explicit subscription to stimulus
> events. The only sticky part in this solution is the question
> of how the AS can route a SUBSCRIBE to the UAC. Once we have
> a GRUU solution in place, this is a no brainer: the AS can
> extract the GRUU from the INVITE, and use it as the target
> of its SUBSCRIBE. The KPML script is included in the SUBSCRIBE,
> and essentially serves as a filter.
> 

I sort of see this one, but what if the App Server wanted to run an App on
the UAS instead of the UAC. Would the 180 have the GRU? Would it be
reliable?

I seem to recall someone asking about the expire time on the subscription
:-)

I'd be keener on GRUU solutions if GRUU was a solved problem.

> 3) Explicit Subscriptions (with REFER)
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|
> 3|                |<-----200-------|
> 4|<-----200-------|                |   AS adds SIP URL to 200
> 5|-----REFER----->|                |
> 6|<-----200-------|                |
> 7|<----NOTIFY-----|                |
> 8|------481------>|                |
> 9|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> A|------200------>|                |
> B|-----NOTIFY---->|                |
> C|<-----200-------|                |
> D|-----NOTIFY---->|                |   UAC provides digits
> E|<-----200-------|                |   AS acknowledges digits
> 
> Although somewhat chattier than the previous solution, this gets around
> the GRUU issue by having the AS provide a SIP URI that routes to itself
> in the 200 response. The UAC then uses this URI to send a REFER to the
> AS. This REFER tells the AS to SUBSCRIBE to the UAC. The flow from that
> point forward is the same as for solution 2.
>

I got to admit, I had never considered anything like this. I think it is
going to be hard to convince folks this is really easier and better than
sending DTMF over INFO. Keep in mind that KPML is for little device that
think 14k of memory is a lot. Real devices are just going to do VXML and
HTML and ignore KPML.

Cullen


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Date: Thu, 14 Aug 2003 08:00:02 +0100
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-qsig2sip-02.txt
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This draft takes into account all comments from the reviewers. I believe it
is ready to progress.

John (john.elwell@siemens.com)

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org] 
Sent: 13 August 2003 15:03
Cc: sipping@ietf.org
Subject: [Sipping] I-D ACTION:draft-ietf-sipping-qsig2sip-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		: Interworking between SIP and QSIG
	Author(s)	: J. Elwell et al.
	Filename	: draft-ietf-sipping-qsig2sip-02.txt
	Pages		: 57
	Date		: 2003-8-13
	
This document specifies interworking between the Session Initiation 
Protocol (SIP) and QSIG within corporate telecommunication networks 
(also known as enterprise networks). SIP is an Internet application-
layer control (signalling) protocol for creating, modifying, and 
terminating sessions with one or more participants. These sessions 
include, in particular, telephone calls. QSIG is a signalling 
protocol for creating, modifying and terminating circuit-switched 
calls, in particular telephone calls, within Private Integrated 
Services Networks (PISNs). QSIG is specified in a number of ECMA 
Standards and published also as ISO/IEC standards. 
As the support of telephony within corporate networks evolves from 
circuit-switched technology to Internet technology, the two 
technologies will co-exist in many networks for a period, perhaps 
several years. Therefore there is a need to be able to establish, 
modify and terminate sessions involving a participant in the SIP 
network and a participant in the QSIG network. Such calls are 
supported by gateways that perform interworking between SIP and QSIG. 
This document is a product of the authors' activities in ECMA 
(www.ecma.ch) on interoperability of QSIG with IP networks.

A URL for this Internet-Draft is:
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From: Bernard Aboba <aboba@internaut.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
cc: john.loughney@nokia.com, sipping@ietf.org
Subject: Re: AW: [Sipping] FYI: RADIUS & SIP
In-Reply-To: <3F3A5D41.5040505@dynamicsoft.com>
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References: <DADF50F5EC506B41A0F375ABEB32063658F247@esebe023.ntc.nokia.com>
 <Pine.LNX.4.53.0308062306320.21030@internaut.com> <3F3A5D41.5040505@dynamicsoft.com>
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> It seems to me that exponential backoff in retransmits is certainly
> possible in a backwards compatible way.

Yes, this part of the issue can be handled in a backwards compatible way.
Since only the RADIUS client retransmits, the RADIUS server code does not
have to change.

> I'm not enough of a RADIUS
> expert to know whether the watchdog mechanism as described in rfc3539
> is possible in a backwards compatible way.

For the watchdog, both the RADIUS client and server would have to
change.

The objective of a watchdog would be to get an immediate response from the
next hop RADIUS proxy/server; no forwarding of the watchdog is permitted.
This would imply a new command, since there is no existing command that
cannot be forwarded.

It's probably possible to do this in a backward compatible way.
One possibility would be to run the new command on a new port.
Since RADIUS has no capabilities negotiation, the RADIUS client can send
the new command to a new port and conclude that the command is not
supported if it receives an ICMP port unreachable.  This is
what is done with the dynamic authorization extensions, for example (RFC
3576). Not elegant (wouldn't want to do this with every new command!) but
it would work.

> BTW, while on the subject of rfc3539, I am aware of many sip
> implementations which perform watchdog operations very similar to
> those described in section 3.4. I wonder if it would be useful to have
> a sipping specification that describes how to properly do this in sip?
> There are many ways to do this wrong, and we could leverage the
> thinking Bernard has put into the AAA document to write up a version
> for SIP that works well.

RFC 3539 sends a watchdog after a given elapsed time without evidence of
liveness (no traffic).  Somewhat better scalability can be obtained if you
wait until there is traffic to be sent; establishing liveness without the
need to actually make use of it is somewhat wasteful.

_______________________________________________
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: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
From: Dean Willis <dean.willis@softarmor.com>
To: "Liess, Laura" <Laura.Liess@t-systems.com>
Cc: jiri.kuthan@fokus.fraunhofer.de, hgs@cs.columbia.edu,
        jdrosen@dynamicsoft.com, aboba@internaut.com, sipping@ietf.org,
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On Wed, 2003-08-06 at 03:17, Liess, Laura wrote:
> What about the accounting draft draft-schulzrinne-accounting-sip-01.txt? Any reason for not working on it? 
> Laura 

Accounting over a non-reliable transport protocol is a Very Hard
Problem. That's one of the reaons for DIAMETER.

--
Dean


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To: Dean Willis <dean.willis@softarmor.com>,
        "Liess, Laura" <Laura.Liess@t-systems.com>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
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At 07:56 PM 8/7/2003, Dean Willis wrote:
>On Wed, 2003-08-06 at 03:17, Liess, Laura wrote:
>> What about the accounting draft draft-schulzrinne-accounting-sip-01.txt? Any reason for not working on it? 
>> Laura 
>
>Accounting over a non-reliable transport protocol is a Very Hard
>Problem. That's one of the reaons for DIAMETER.

Still less hard than accounting over a protocol which does not exist.
We actually developed some DIAMETER code but the problem is it is
not running anywhere.

-jiri 


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From exim@www1.ietf.org  Thu Aug 14 06:43:42 2003
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Date: Wed, 13 Aug 2003 11:07:44 +0200
From: Luis Emilio Martin <lemr@tid.es>
Subject: WG: [Sipping] WGLC for Q.SIP-SIP mapping draft
To: thomas.stach@siemens.com, sipping@ietf.org
Cc: jieying.li@clarent.com, jfmule@packetizer.com
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Hello,

> The QSIG SETUP message may optionally transport HLC and LLC Information elements which are not mentioned
> in the draft. You might want to add something like into section 10:
>
>"The gateway SHOULD (or SHALL(?)) not include a Low layer compatibility in the QSIG SETUP message. The 
> gateway SHOULD not include a High layer compatibility information element in the QSIG SETUP message.
>
>NOTE: Inclusion of a HLC in the QSIG SETUP message might be useful, if the gateway knows prior to 
> sending the QSIG SETUP message that fax is being transmitted (e.g., because of a corresponding
> SDP offer), it may include in the QSIG SETUP message a High layer compatibility information element 
> indicating group 3 fax."

It is discussed the possibility to include HLC IE in the QSIG SETUP if
an INVITE
was received with fax transmission information (i.e. T38 only), so that
fax
detection at destination does not depend on processing audio data. It
possibly is a matter of
configuration in the gateway.

Sorry I am not very used to T.38 procedures and Fax Group 2/3 methods
but we are testing some gateways
with T.38 and automatic fax detection capabilities based on DSPs and
experiencing some strange 
behaviours in these gateways, so that SETUP messages including HLC
produce INVITEs offering G.729 or 
G.711 and a bit later new INVITES offering t38.

In reverse direction, can a SIP gateway, upon reception of a SETUP
Message with Bearer Capability
3,1 kHz Audio and an HLC indicating Fax Group 2/3, to map the HLC field
into an INVITE message
with an SDP offering T.38 only (not any other audio codec). The example
is
taken from chapter 5.1.2
http://www.watersprings.org/pub/id/draft-mule-sip-t38callflows-02.txt :


      Content-Type: application/sdp
      Content-Length: 320

      v=0
      o=ifaxgw1 2890846527 2890846527 IN IP4 ifax.here.com
      s=Session SDP
      c=IN IP4 ifaxmg.here.com
      t=0 0
      m=image 15002 udptl t38
      a=T38FaxVersion:0
      a=T38MaxBitRate:14400
      a=T38FaxFillBitRemoval:0
      a=T38FaxTranscodingMMR:0
      a=T38FaxTranscodingJBIG:0
      a=T38FaxRateManagement:transferredTCF


It should be the most straightforward procedure to map fax but is my
interpretation correct?...
or the reception of a setUp with such IE values does not have enough
information to 
include an SDP offering with T.38 only, so it is required to wait for
CNG/V21 detection 
to pick all the information required, and a renegotiation is thus
required after call is
stablished?


Thank you very much in advance

Best Regards

Luis Martin

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To: "Henry Sinnreich" <Henry.Sinnreich@mci.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Cc: <guido.gybels@rnid.org.uk>, <gv@trace.wisc.edu>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        "'Mike Spanner'" <mike.spanner@rnid.org.uk>,
        <andreas.piirimets@omnitor.se>, <trefor@trefor.plus.com>,
        "'Rajesh Kumar'" <rkumar@cisco.com>, <marcelo@envilogg.se>,
        <jorgen.bjorkner@hotsip.com>,
        "'James Hamlin'" <James.Hamlin@rnid.org.uk>,
        <Erik.Zetterstrom@omnitor.se>, <hwildfeu@cisco.com>,
        <barry.dingle@bigfoot.com.au>, <mgarakan@cisco.com>, <oran@cisco.com>,
        <sipping@ietf.org>, "'Andrea Saks'" <asaks@waitrose.com>,
        "'Erik Lagerway'" <erik@xten.com>
References: <0HJ900I4JHRTGL@pmismtp05.wcomnet.com>
Subject: Re: [Sipping] Re: items for vor voting in the BOF
Date: Thu, 7 Aug 2003 23:23:12 -0400
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Henry,

Apparently, the mail server is a bit back-logged... I just received this
today.

Anyway, the xten site seems to have only a softphone.  Perhaps I overlooked
the other stuff.

I saw this site yesterday: www.sipphone.com

That site has 2 physical SIP phones for $130.  Not a bad price and they do
have a small LCD display for Call ID.  It does not support RFC 2793, but it
claims to support SIMPLE.  I'm not sure how useful SIMPLE would prove to be
from the phone and am quite curious about the UI for entering text.  Anybody
know?

Here's the product info:
http://www.ovislink.com/newovislink/Products/IPPhones/BT100.pdf

Paul

----- Original Message ----- 
From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Henry Sinnreich'"
<Henry.Sinnreich@mci.com>; "'Rohan Mahy'" <rohan@cisco.com>
Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry Sinnreich'"
<Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
<gunnar.hellstrom@omnitor.se>; "'Mike Spanner'" <mike.spanner@rnid.org.uk>;
<andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
<rkumar@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
"'James Hamlin'" <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
<hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>; <mgarakan@cisco.com>;
<oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'" <asaks@waitrose.com>;
"'Erik Lagerway'" <erik@xten.com>
Sent: Thursday, August 07, 2003 2:23 PM
Subject: RE: [Sipping] Re: items for vor voting in the BOF


>
> Paul,
>
> Go to http://xten.com and see what a low cost phone can do. It has display
> and keyboard.
>
> Henry
>
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of
> > Paul E. Jones
> > Sent: Tuesday, July 29, 2003 9:33 PM
> > To: Henry Sinnreich; 'Rohan Mahy'
> > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; 'Henry Sinnreich';
> > 'Gunnar Hellstrom'; 'Mike Spanner'; andreas.piirimets@omnitor.se;
> > trefor@trefor.plus.com; 'Rajesh Kumar'; marcelo@envilogg.se;
> > jorgen.bjorkner@hotsip.com; 'James Hamlin'; Erik.Zetterstrom@omnitor.se;
> > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au; mgarakan@cisco.com;
> > oran@cisco.com; sipping@ietf.org; 'Andrea Saks'
> > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> >
> > Henry,
> >
> > The same would likely be said by a boss or busy secretary in many
> > companies.
> > Even so, the bulk of the phones in most enterprises are just little dumb
> > phones with no real features (perhaps feature keys... not much else).  I
> > don't expect that to change substantially, even with the introduction of
> > VoIP.  After all, more and more capability can be moved onto the PC.
Why
> > add hundreds of dollars in value to an IP phone when all of the "cool"
> > features that you want can be placed on the PC (and updated much more
> > easily)?  The address book, instant messaging, etc., can all be located
on
> > the PC.  The PC can interface with the server and with your phone.  The
> > phone needs little functionality of its own, really.
> >
> > There are people who buy fancy phones for their homes.  Even so, there
> > will
> > be a substantial number of people who just don't care and will buy the
> > cheapest thing they can find.  I'm sure that manufacturers will try to
cut
> > costs at every opportunity and screens are a target.  Should we not plan
> > on
> > it?
> >
> > Paul
> >
> > ----- Original Message -----
> > From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
> > To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Rohan Mahy'"
> > <rohan@cisco.com>
> > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry Sinnreich'"
> > <Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
> > <gunnar.hellstrom@omnitor.se>; "'Mike Spanner'"
> > <mike.spanner@rnid.org.uk>;
> > <andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh
Kumar'"
> > <rkumar@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
> > "'James Hamlin'" <James.Hamlin@rnid.org.uk>;
> > <Erik.Zetterstrom@omnitor.se>;
> > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
<mgarakan@cisco.com>;
> > <oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'"
<asaks@waitrose.com>
> > Sent: Monday, July 28, 2003 10:18 PM
> > Subject: RE: [Sipping] Re: items for vor voting in the BOF
> >
> >
> > > Paul, finally, we disagree!
> > >
> > > We have 3-4 SIP phones in my house at all times and we are using all
of
> > them
> > > in various rooms. We would not even dream touching a VoIP phone that
> > does
> > > not have a display and without its own web page where we can see the
> > address
> > > book and dial from the PC or from some another web absed personal call
> > > manager.
> > >
> > > I would certainly not attempt to sell any friend or family member a
SIP
> > > phone withou display. Depressing thought.
> > >
> > > Nor would we ever by a mobile phone without display.
> > >
> > > Cheers :-)
> > >
> > > Henry
> > >
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > On Behalf Of Paul E. Jones
> > > > Sent: Sunday, July 20, 2003 11:52 AM
> > > > To: Rohan Mahy
> > > > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; Henry
> > > > Sinnreich; Gunnar Hellstrom; Mike Spanner;
> > > > andreas.piirimets@omnitor.se; trefor@trefor.plus.com; 'Rajesh
> > > > Kumar'; marcelo@envilogg.se; jorgen.bjorkner@hotsip.com;
> > > > James Hamlin; Erik.Zetterstrom@omnitor.se;
> > > > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au;
> > > > mgarakan@cisco.com; oran@cisco.com; sipping@ietf.org; Andrea Saks
> > > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > > >
> > > > Rohan,
> > > >
> > > > I offered up a few questions to folks that I hoped people
> > > > would have voted on.  The questions I had was whether or not
> > > > there should be a requirement to provide *some* interface to
> > > > legacy TTY devices.  Whether it's RJ-11 or 2.5mm makes no
> > > > difference.  However, I think it should be a requirement that
> > > > pass-through work at the very least.  If this was agreeable,
> > > > I could certainly work to craft specific text for such a
requirement.
> > > >
> > > > I do not have stats on how many phones do not have screens.
> > > > However, note that requirement 1 also requires a keyboard,
> > > > though that's never mentioned.
> > > > One could say that text could be entered via some web
> > > > interface, but doing so is a bit hard in a hotel room or at a
> > > > gas station on the side of the road.  Regardless, I've said
> > > > before and I'll say it again: I do not expect that most of
> > > > the IP phones that will go into mass production will have a
> > > > suitable screen for text communication.  Most of today's IP
> > > > phones have screens, because it is a way to catch people's
> > > > eyes.  However, there are also many that do not.  Would you
> > > > consider Cisco's ATA, for example, an IP phone?  I think it
> > > > qualifies and there is no screen.  There are a number of
> > > > service providers deploying that and similar devices from
> > > > other manufacturers.  None of those have displays.
> > > >
> > > > I assert that in most enterprise environments, the display is
> > > > of very little value.  It's interesting at first, but it's
> > > > not necessary to have a fancy
> > > > screen: just something simple to control hold, transfer,
> > > > conferencing, etc., is enough.  In fact, I'd be happy with
> > > > buttons that did that.
> > > >
> > > > At home, I think it's even less of a requirement to have a
> > > > nice-size display.  Recognizing that many phones today have a
> > > > display for caller ID, etc., such displays are not suitable
> > > > for text communication.  Requirement 1 is asking for a nice
> > > > display... and I think you'll find a much smaller percentage
> > > > of those... especially as they become a commodity item.
> > > >
> > > > Paul
> > > >
> > > > ----- Original Message -----
> > > > From: "Rohan Mahy" <rohan@cisco.com>
> > > > To: "Paul E. Jones" <paulej@packetizer.com>
> > > > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "Henry
Sinnreich"
> > > > <Henry.Sinnreich@mci.com>; "Gunnar Hellstrom"
> > > > <gunnar.hellstrom@omnitor.se>; "Mike Spanner"
> > > > <mike.spanner@rnid.org.uk>; <andreas.piirimets@omnitor.se>;
> > > > <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > <jorgen.bjorkner@hotsip.com>; "James Hamlin"
> > > > <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
> > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > <mgarakan@cisco.com>; <oran@cisco.com>; <sipping@ietf.org>;
> > > > "Andrea Saks" <asaks@waitrose.com>
> > > > Sent: Saturday, July 19, 2003 11:14 PM
> > > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > > >
> > > >
> > > > > Hi Paul,
> > > > >
> > > > > So it sounds like you wanted an additional requirement for
> > > > pass-through
> > > > > via a 2.5 mm jack for phones that do not support RFC2793?  If I
> > > > > understood this correctly, how would you word the requirement?
> > > > >
> > > > > Also, do you have any data on how many phones currently
> > > > exist (number
> > > > > or percentage) for which requirement 1 would not apply? I
> > > > suspect that
> > > > > the number of phones not covered by the first requirement
> > > > is actually
> > > > > quite small.
> > > > >
> > > > > thanks,
> > > > > -rohan
> > > > >
> > > > > On Friday, July 18, 2003, at 05:04 PM, Paul E. Jones wrote:
> > > > >
> > > > > > Guido,
> > > > > >
> > > > > > I wish I could have attended, but I was attending a TDI
conference
> > > > > > here in
> > > > > > the US.  I couldn't be in both places at the same time,
> > > > though I wish I
> > > > > > could.  I was disheartened that the questions were not
> > > > presented as I
> > > > > > proposed.  You should know that, supposedly, work gets
> > > > done over the
> > > > > > e-mail
> > > > > > reflector, not only in the meeting, so not being there
> > > > should not have
> > > > > > lessened my input on the matter.
> > > > > >
> > > > > > Looking at the results of the ad-hoc reported by Rohan, I see:
> > > > > >   1) *If* an IP phone supports a display (and keyboard??), it
> > > > > >      must support RFC 2793
> > > > > >   2) IP phones *should* provide a way to input text
> > > > > >   3) IP phones *may* provide an RJ-11 interface and *must*
> > > > > >      support pass-through.
> > > > > >
> > > > > > Now, my concerns are several.  It appears that you have
> > > > absolutely no
> > > > > > requirement for the mass of products that do not sport a
> > > > display to
> > > > > > support
> > > > > > text in either relay or pass-through mode.  Further, there is no
> > > > > > mention of
> > > > > > a 2.5mm jack being an option.  Question 3, as I
> > > > mentioned, "pressured"
> > > > > > folks
> > > > > > more than it should.  There should have been no reason to
require
> > > > > > support of
> > > > > > pass-through, at the very least.
> > > > > >
> > > > > > I told you that my questions were to ensure that we at least get
> > > > > > support for
> > > > > > text on every phone in one way or another.  It looks like
> > > > you lost.
> > > > > > Company
> > > > > > X is now free to build an IP phone without any text
> > > > support.  I'm not
> > > > > > really
> > > > > > happy about that and I certainly don't see why you should be.
> > > > > >
> > > > > > Paul
> > > > > >
> > > > > > ----- Original Message -----
> > > > > > From: "Guido Gybels" <ggybels@webclient.eu.blackberry.net>
> > > > > > To: "Paul E. Jones" <paulej@packetizer.com>; <gv@trace.wisc.edu>
> > > > > > Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>; "Gunnar
> > > > Hellstrom"
> > > > > > <gunnar.hellstrom@omnitor.se>; "Guido Gybels"
> > > > > > <Guido.Gybels@rnid.org.uk>;
> > > > > > "Mike Spanner" <mike.spanner@rnid.org.uk>;
> > > > > > <andreas.piirimets@omnitor.se>;
> > > > > > <gv@trace.wisc.edu>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > > > <jorgen.bjorkner@hotsip.com>;
> > > > > > "James Hamlin" <James.Hamlin@rnid.org.uk>;
> > > > > > <Erik.Zetterstrom@omnitor.se>;
> > > > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > > > <mgarakan@cisco.com>;
> > > > > > <oran@cisco.com>; <rohan@cisco.com>; <sipping@ietf.org>
> > > > > > Sent: Friday, July 18, 2003 2:31 AM
> > > > > > Subject: [Sipping] Re: items for vor voting in the BOF
> > > > > >
> > > > > >
> > > > > >> Paul,
> > > > > >>
> > > > > >> <<While I guess it might be a bit late now>>
> > > > > >>
> > > > > >> The discussion on the sipdev document took place
> > > > yesterday, and if you
> > > > > > wanted to make a point you were free to attend it. You'll see
the
> > > > > > results
> > > > > > reflected in the next version of the document, and I can
> > > > say that both
> > > > > > Arnoud and myself believe this result to be extremely
> > > > sensible as well
> > > > > > as in
> > > > > > the best interest of all stakeholders.
> > > > > >> I also suggest this no longer belongs here on the
> > > > mailing list until
> > > > > >> a new
> > > > > > issue would arise. I have stopped replying to your emails
> > > > a while ago
> > > > > > as I
> > > > > > see no point in continuing the discussion (and swamping
> > > > the list) with
> > > > > > someone who, despite being a native English speaker,
> > > > seems unable to
> > > > > > understand what we are saying and more importantly deliberately
> > > > > > misrepresents my statements or attributes words to my
> > > > mouth that I did
> > > > > > not
> > > > > > speak.
> > > > > >>
> > > > > >> <<would guess most people will say "yes" to, while I
> > > > know you'd say
> > > > > >> "no"
> > > > > > to
> > > > > >> it.>>
> > > > > >>
> > > > > >> This being a nice example of that.
> > > > > >>
> > > > > >>
> > > > > >> I believe the IETF community has shown great wisdom and a true
> > > > > > understanding of what is at stake here. We have now got a
> > > > solid path
> > > > > > to the
> > > > > > future, with a clear strategy to get rid of obsolete and
> > > > unsatisfactory
> > > > > > solutions allowing a gradual move of deaf, hard of
> > > > hearing and speech
> > > > > > impaired people into mainstream technology. That will of
> > > > course take
> > > > > > time
> > > > > > during which there will have to be legacy support and all the
> > > > > > instruments
> > > > > > are put in place to do exactly that.
> > > > > >>
> > > > > >> Guido
> > > > > >>
> > > > > >> Guido Gybels
> > > > > >> Director of New Technologies
> > > > > >>
> > > > > >> RNID, 19-23 Featherstone Street
> > > > > >> London EC1Y 8SL, UK
> > > > > >> Tel +44(0)20-7294 3713
> > > > > >> Fax +44(0)20-7296 8069
> > > > > >>
> > > > > >> _______________________________________________
> > > > > >> 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
>


_______________________________________________
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: Thu, 07 Aug 2003 13:23:54 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Re: items for vor voting in the BOF
In-reply-to: <023b01c35642$f2aa1950$0401a8c0@packetizer.org>
To: "'Paul E. Jones'" <paulej@packetizer.com>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Cc: guido.gybels@rnid.org.uk, gv@trace.wisc.edu,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        "'Mike Spanner'" <mike.spanner@rnid.org.uk>,
        andreas.piirimets@omnitor.se, trefor@trefor.plus.com,
        "'Rajesh Kumar'" <rkumar@cisco.com>, marcelo@envilogg.se,
        jorgen.bjorkner@hotsip.com,
        "'James Hamlin'" <James.Hamlin@rnid.org.uk>,
        Erik.Zetterstrom@omnitor.se, hwildfeu@cisco.com,
        barry.dingle@bigfoot.com.au, mgarakan@cisco.com, oran@cisco.com,
        sipping@ietf.org, "'Andrea Saks'" <asaks@waitrose.com>,
        "'Erik Lagerway'" <erik@xten.com>
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Paul,

Go to http://xten.com and see what a low cost phone can do. It has display
and keyboard.

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Paul E. Jones
> Sent: Tuesday, July 29, 2003 9:33 PM
> To: Henry Sinnreich; 'Rohan Mahy'
> Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; 'Henry Sinnreich';
> 'Gunnar Hellstrom'; 'Mike Spanner'; andreas.piirimets@omnitor.se;
> trefor@trefor.plus.com; 'Rajesh Kumar'; marcelo@envilogg.se;
> jorgen.bjorkner@hotsip.com; 'James Hamlin'; Erik.Zetterstrom@omnitor.se;
> hwildfeu@cisco.com; barry.dingle@bigfoot.com.au; mgarakan@cisco.com;
> oran@cisco.com; sipping@ietf.org; 'Andrea Saks'
> Subject: Re: [Sipping] Re: items for vor voting in the BOF
> 
> Henry,
> 
> The same would likely be said by a boss or busy secretary in many
> companies.
> Even so, the bulk of the phones in most enterprises are just little dumb
> phones with no real features (perhaps feature keys... not much else).  I
> don't expect that to change substantially, even with the introduction of
> VoIP.  After all, more and more capability can be moved onto the PC.  Why
> add hundreds of dollars in value to an IP phone when all of the "cool"
> features that you want can be placed on the PC (and updated much more
> easily)?  The address book, instant messaging, etc., can all be located on
> the PC.  The PC can interface with the server and with your phone.  The
> phone needs little functionality of its own, really.
> 
> There are people who buy fancy phones for their homes.  Even so, there
> will
> be a substantial number of people who just don't care and will buy the
> cheapest thing they can find.  I'm sure that manufacturers will try to cut
> costs at every opportunity and screens are a target.  Should we not plan
> on
> it?
> 
> Paul
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
> To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Rohan Mahy'"
> <rohan@cisco.com>
> Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry Sinnreich'"
> <Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
> <gunnar.hellstrom@omnitor.se>; "'Mike Spanner'"
> <mike.spanner@rnid.org.uk>;
> <andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> <rkumar@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
> "'James Hamlin'" <James.Hamlin@rnid.org.uk>;
> <Erik.Zetterstrom@omnitor.se>;
> <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>; <mgarakan@cisco.com>;
> <oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'" <asaks@waitrose.com>
> Sent: Monday, July 28, 2003 10:18 PM
> Subject: RE: [Sipping] Re: items for vor voting in the BOF
> 
> 
> > Paul, finally, we disagree!
> >
> > We have 3-4 SIP phones in my house at all times and we are using all of
> them
> > in various rooms. We would not even dream touching a VoIP phone that
> does
> > not have a display and without its own web page where we can see the
> address
> > book and dial from the PC or from some another web absed personal call
> > manager.
> >
> > I would certainly not attempt to sell any friend or family member a SIP
> > phone withou display. Depressing thought.
> >
> > Nor would we ever by a mobile phone without display.
> >
> > Cheers :-)
> >
> > Henry
> >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > On Behalf Of Paul E. Jones
> > > Sent: Sunday, July 20, 2003 11:52 AM
> > > To: Rohan Mahy
> > > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; Henry
> > > Sinnreich; Gunnar Hellstrom; Mike Spanner;
> > > andreas.piirimets@omnitor.se; trefor@trefor.plus.com; 'Rajesh
> > > Kumar'; marcelo@envilogg.se; jorgen.bjorkner@hotsip.com;
> > > James Hamlin; Erik.Zetterstrom@omnitor.se;
> > > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au;
> > > mgarakan@cisco.com; oran@cisco.com; sipping@ietf.org; Andrea Saks
> > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > >
> > > Rohan,
> > >
> > > I offered up a few questions to folks that I hoped people
> > > would have voted on.  The questions I had was whether or not
> > > there should be a requirement to provide *some* interface to
> > > legacy TTY devices.  Whether it's RJ-11 or 2.5mm makes no
> > > difference.  However, I think it should be a requirement that
> > > pass-through work at the very least.  If this was agreeable,
> > > I could certainly work to craft specific text for such a requirement.
> > >
> > > I do not have stats on how many phones do not have screens.
> > > However, note that requirement 1 also requires a keyboard,
> > > though that's never mentioned.
> > > One could say that text could be entered via some web
> > > interface, but doing so is a bit hard in a hotel room or at a
> > > gas station on the side of the road.  Regardless, I've said
> > > before and I'll say it again: I do not expect that most of
> > > the IP phones that will go into mass production will have a
> > > suitable screen for text communication.  Most of today's IP
> > > phones have screens, because it is a way to catch people's
> > > eyes.  However, there are also many that do not.  Would you
> > > consider Cisco's ATA, for example, an IP phone?  I think it
> > > qualifies and there is no screen.  There are a number of
> > > service providers deploying that and similar devices from
> > > other manufacturers.  None of those have displays.
> > >
> > > I assert that in most enterprise environments, the display is
> > > of very little value.  It's interesting at first, but it's
> > > not necessary to have a fancy
> > > screen: just something simple to control hold, transfer,
> > > conferencing, etc., is enough.  In fact, I'd be happy with
> > > buttons that did that.
> > >
> > > At home, I think it's even less of a requirement to have a
> > > nice-size display.  Recognizing that many phones today have a
> > > display for caller ID, etc., such displays are not suitable
> > > for text communication.  Requirement 1 is asking for a nice
> > > display... and I think you'll find a much smaller percentage
> > > of those... especially as they become a commodity item.
> > >
> > > Paul
> > >
> > > ----- Original Message -----
> > > From: "Rohan Mahy" <rohan@cisco.com>
> > > To: "Paul E. Jones" <paulej@packetizer.com>
> > > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "Henry Sinnreich"
> > > <Henry.Sinnreich@mci.com>; "Gunnar Hellstrom"
> > > <gunnar.hellstrom@omnitor.se>; "Mike Spanner"
> > > <mike.spanner@rnid.org.uk>; <andreas.piirimets@omnitor.se>;
> > > <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > <jorgen.bjorkner@hotsip.com>; "James Hamlin"
> > > <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
> > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > <mgarakan@cisco.com>; <oran@cisco.com>; <sipping@ietf.org>;
> > > "Andrea Saks" <asaks@waitrose.com>
> > > Sent: Saturday, July 19, 2003 11:14 PM
> > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > >
> > >
> > > > Hi Paul,
> > > >
> > > > So it sounds like you wanted an additional requirement for
> > > pass-through
> > > > via a 2.5 mm jack for phones that do not support RFC2793?  If I
> > > > understood this correctly, how would you word the requirement?
> > > >
> > > > Also, do you have any data on how many phones currently
> > > exist (number
> > > > or percentage) for which requirement 1 would not apply? I
> > > suspect that
> > > > the number of phones not covered by the first requirement
> > > is actually
> > > > quite small.
> > > >
> > > > thanks,
> > > > -rohan
> > > >
> > > > On Friday, July 18, 2003, at 05:04 PM, Paul E. Jones wrote:
> > > >
> > > > > Guido,
> > > > >
> > > > > I wish I could have attended, but I was attending a TDI conference
> > > > > here in
> > > > > the US.  I couldn't be in both places at the same time,
> > > though I wish I
> > > > > could.  I was disheartened that the questions were not
> > > presented as I
> > > > > proposed.  You should know that, supposedly, work gets
> > > done over the
> > > > > e-mail
> > > > > reflector, not only in the meeting, so not being there
> > > should not have
> > > > > lessened my input on the matter.
> > > > >
> > > > > Looking at the results of the ad-hoc reported by Rohan, I see:
> > > > >   1) *If* an IP phone supports a display (and keyboard??), it
> > > > >      must support RFC 2793
> > > > >   2) IP phones *should* provide a way to input text
> > > > >   3) IP phones *may* provide an RJ-11 interface and *must*
> > > > >      support pass-through.
> > > > >
> > > > > Now, my concerns are several.  It appears that you have
> > > absolutely no
> > > > > requirement for the mass of products that do not sport a
> > > display to
> > > > > support
> > > > > text in either relay or pass-through mode.  Further, there is no
> > > > > mention of
> > > > > a 2.5mm jack being an option.  Question 3, as I
> > > mentioned, "pressured"
> > > > > folks
> > > > > more than it should.  There should have been no reason to require
> > > > > support of
> > > > > pass-through, at the very least.
> > > > >
> > > > > I told you that my questions were to ensure that we at least get
> > > > > support for
> > > > > text on every phone in one way or another.  It looks like
> > > you lost.
> > > > > Company
> > > > > X is now free to build an IP phone without any text
> > > support.  I'm not
> > > > > really
> > > > > happy about that and I certainly don't see why you should be.
> > > > >
> > > > > Paul
> > > > >
> > > > > ----- Original Message -----
> > > > > From: "Guido Gybels" <ggybels@webclient.eu.blackberry.net>
> > > > > To: "Paul E. Jones" <paulej@packetizer.com>; <gv@trace.wisc.edu>
> > > > > Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>; "Gunnar
> > > Hellstrom"
> > > > > <gunnar.hellstrom@omnitor.se>; "Guido Gybels"
> > > > > <Guido.Gybels@rnid.org.uk>;
> > > > > "Mike Spanner" <mike.spanner@rnid.org.uk>;
> > > > > <andreas.piirimets@omnitor.se>;
> > > > > <gv@trace.wisc.edu>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > > <jorgen.bjorkner@hotsip.com>;
> > > > > "James Hamlin" <James.Hamlin@rnid.org.uk>;
> > > > > <Erik.Zetterstrom@omnitor.se>;
> > > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > > <mgarakan@cisco.com>;
> > > > > <oran@cisco.com>; <rohan@cisco.com>; <sipping@ietf.org>
> > > > > Sent: Friday, July 18, 2003 2:31 AM
> > > > > Subject: [Sipping] Re: items for vor voting in the BOF
> > > > >
> > > > >
> > > > >> Paul,
> > > > >>
> > > > >> <<While I guess it might be a bit late now>>
> > > > >>
> > > > >> The discussion on the sipdev document took place
> > > yesterday, and if you
> > > > > wanted to make a point you were free to attend it. You'll see the
> > > > > results
> > > > > reflected in the next version of the document, and I can
> > > say that both
> > > > > Arnoud and myself believe this result to be extremely
> > > sensible as well
> > > > > as in
> > > > > the best interest of all stakeholders.
> > > > >> I also suggest this no longer belongs here on the
> > > mailing list until
> > > > >> a new
> > > > > issue would arise. I have stopped replying to your emails
> > > a while ago
> > > > > as I
> > > > > see no point in continuing the discussion (and swamping
> > > the list) with
> > > > > someone who, despite being a native English speaker,
> > > seems unable to
> > > > > understand what we are saying and more importantly deliberately
> > > > > misrepresents my statements or attributes words to my
> > > mouth that I did
> > > > > not
> > > > > speak.
> > > > >>
> > > > >> <<would guess most people will say "yes" to, while I
> > > know you'd say
> > > > >> "no"
> > > > > to
> > > > >> it.>>
> > > > >>
> > > > >> This being a nice example of that.
> > > > >>
> > > > >>
> > > > >> I believe the IETF community has shown great wisdom and a true
> > > > > understanding of what is at stake here. We have now got a
> > > solid path
> > > > > to the
> > > > > future, with a clear strategy to get rid of obsolete and
> > > unsatisfactory
> > > > > solutions allowing a gradual move of deaf, hard of
> > > hearing and speech
> > > > > impaired people into mainstream technology. That will of
> > > course take
> > > > > time
> > > > > during which there will have to be legacy support and all the
> > > > > instruments
> > > > > are put in place to do exactly that.
> > > > >>
> > > > >> Guido
> > > > >>
> > > > >> Guido Gybels
> > > > >> Director of New Technologies
> > > > >>
> > > > >> RNID, 19-23 Featherstone Street
> > > > >> London EC1Y 8SL, UK
> > > > >> Tel +44(0)20-7294 3713
> > > > >> Fax +44(0)20-7296 8069
> > > > >>
> > > > >> _______________________________________________
> > > > >> 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


_______________________________________________
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, 08 Aug 2003 09:20:26 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Re: items for vor voting in the BOF
In-reply-to: <018f01c35d5c$6780e220$0401a8c0@packetizer.org>
To: "'Paul E. Jones'" <paulej@packetizer.com>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Rohan Mahy'" <rohan@cisco.com>
Cc: guido.gybels@rnid.org.uk, gv@trace.wisc.edu,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        "'Mike Spanner'" <mike.spanner@rnid.org.uk>,
        andreas.piirimets@omnitor.se, trefor@trefor.plus.com,
        "'Rajesh Kumar'" <rkumar@cisco.com>, marcelo@envilogg.se,
        jorgen.bjorkner@hotsip.com,
        "'James Hamlin'" <James.Hamlin@rnid.org.uk>,
        Erik.Zetterstrom@omnitor.se, hwildfeu@cisco.com,
        barry.dingle@bigfoot.com.au, mgarakan@cisco.com, oran@cisco.com,
        sipping@ietf.org, "'Andrea Saks'" <asaks@waitrose.com>,
        "'Erik Lagerway'" <erik@xten.com>
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This is a good opportunity to encourage all on the list to help register SIP
phones and other products at the www.sipforum.org 

Thanks, Henry

> -----Original Message-----
> From: Paul E. Jones [mailto:paulej@packetizer.com]
> Sent: Thursday, August 07, 2003 10:23 PM
> To: Henry Sinnreich; 'Rohan Mahy'
> Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; 'Henry Sinnreich';
> 'Gunnar Hellstrom'; 'Mike Spanner'; andreas.piirimets@omnitor.se;
> trefor@trefor.plus.com; 'Rajesh Kumar'; marcelo@envilogg.se;
> jorgen.bjorkner@hotsip.com; 'James Hamlin'; Erik.Zetterstrom@omnitor.se;
> hwildfeu@cisco.com; barry.dingle@bigfoot.com.au; mgarakan@cisco.com;
> oran@cisco.com; sipping@ietf.org; 'Andrea Saks'; 'Erik Lagerway'
> Subject: Re: [Sipping] Re: items for vor voting in the BOF
> 
> Henry,
> 
> Apparently, the mail server is a bit back-logged... I just received this
> today.
> 
> Anyway, the xten site seems to have only a softphone.  Perhaps I
> overlooked
> the other stuff.
> 
> I saw this site yesterday: www.sipphone.com
> 
> That site has 2 physical SIP phones for $130.  Not a bad price and they do
> have a small LCD display for Call ID.  It does not support RFC 2793, but
> it
> claims to support SIMPLE.  I'm not sure how useful SIMPLE would prove to
> be
> from the phone and am quite curious about the UI for entering text.
> Anybody
> know?
> 
> Here's the product info:
> http://www.ovislink.com/newovislink/Products/IPPhones/BT100.pdf
> 
> Paul
> 
> ----- Original Message -----
> From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
> To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Henry Sinnreich'"
> <Henry.Sinnreich@mci.com>; "'Rohan Mahy'" <rohan@cisco.com>
> Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry Sinnreich'"
> <Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
> <gunnar.hellstrom@omnitor.se>; "'Mike Spanner'"
> <mike.spanner@rnid.org.uk>;
> <andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> <rkumar@cisco.com>; <marcelo@envilogg.se>; <jorgen.bjorkner@hotsip.com>;
> "'James Hamlin'" <James.Hamlin@rnid.org.uk>;
> <Erik.Zetterstrom@omnitor.se>;
> <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>; <mgarakan@cisco.com>;
> <oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'"
> <asaks@waitrose.com>;
> "'Erik Lagerway'" <erik@xten.com>
> Sent: Thursday, August 07, 2003 2:23 PM
> Subject: RE: [Sipping] Re: items for vor voting in the BOF
> 
> 
> >
> > Paul,
> >
> > Go to http://xten.com and see what a low cost phone can do. It has
> display
> > and keyboard.
> >
> > Henry
> >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
> Of
> > > Paul E. Jones
> > > Sent: Tuesday, July 29, 2003 9:33 PM
> > > To: Henry Sinnreich; 'Rohan Mahy'
> > > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; 'Henry Sinnreich';
> > > 'Gunnar Hellstrom'; 'Mike Spanner'; andreas.piirimets@omnitor.se;
> > > trefor@trefor.plus.com; 'Rajesh Kumar'; marcelo@envilogg.se;
> > > jorgen.bjorkner@hotsip.com; 'James Hamlin';
> Erik.Zetterstrom@omnitor.se;
> > > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au; mgarakan@cisco.com;
> > > oran@cisco.com; sipping@ietf.org; 'Andrea Saks'
> > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > >
> > > Henry,
> > >
> > > The same would likely be said by a boss or busy secretary in many
> > > companies.
> > > Even so, the bulk of the phones in most enterprises are just little
> dumb
> > > phones with no real features (perhaps feature keys... not much else).
> I
> > > don't expect that to change substantially, even with the introduction
> of
> > > VoIP.  After all, more and more capability can be moved onto the PC.
> Why
> > > add hundreds of dollars in value to an IP phone when all of the "cool"
> > > features that you want can be placed on the PC (and updated much more
> > > easily)?  The address book, instant messaging, etc., can all be
> located
> on
> > > the PC.  The PC can interface with the server and with your phone.
> The
> > > phone needs little functionality of its own, really.
> > >
> > > There are people who buy fancy phones for their homes.  Even so, there
> > > will
> > > be a substantial number of people who just don't care and will buy the
> > > cheapest thing they can find.  I'm sure that manufacturers will try to
> cut
> > > costs at every opportunity and screens are a target.  Should we not
> plan
> > > on
> > > it?
> > >
> > > Paul
> > >
> > > ----- Original Message -----
> > > From: "Henry Sinnreich" <Henry.Sinnreich@mci.com>
> > > To: "'Paul E. Jones'" <paulej@packetizer.com>; "'Rohan Mahy'"
> > > <rohan@cisco.com>
> > > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "'Henry
> Sinnreich'"
> > > <Henry.Sinnreich@mci.com>; "'Gunnar Hellstrom'"
> > > <gunnar.hellstrom@omnitor.se>; "'Mike Spanner'"
> > > <mike.spanner@rnid.org.uk>;
> > > <andreas.piirimets@omnitor.se>; <trefor@trefor.plus.com>; "'Rajesh
> Kumar'"
> > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> <jorgen.bjorkner@hotsip.com>;
> > > "'James Hamlin'" <James.Hamlin@rnid.org.uk>;
> > > <Erik.Zetterstrom@omnitor.se>;
> > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> <mgarakan@cisco.com>;
> > > <oran@cisco.com>; <sipping@ietf.org>; "'Andrea Saks'"
> <asaks@waitrose.com>
> > > Sent: Monday, July 28, 2003 10:18 PM
> > > Subject: RE: [Sipping] Re: items for vor voting in the BOF
> > >
> > >
> > > > Paul, finally, we disagree!
> > > >
> > > > We have 3-4 SIP phones in my house at all times and we are using all
> of
> > > them
> > > > in various rooms. We would not even dream touching a VoIP phone that
> > > does
> > > > not have a display and without its own web page where we can see the
> > > address
> > > > book and dial from the PC or from some another web absed personal
> call
> > > > manager.
> > > >
> > > > I would certainly not attempt to sell any friend or family member a
> SIP
> > > > phone withou display. Depressing thought.
> > > >
> > > > Nor would we ever by a mobile phone without display.
> > > >
> > > > Cheers :-)
> > > >
> > > > Henry
> > > >
> > > > > -----Original Message-----
> > > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > > On Behalf Of Paul E. Jones
> > > > > Sent: Sunday, July 20, 2003 11:52 AM
> > > > > To: Rohan Mahy
> > > > > Cc: guido.gybels@rnid.org.uk; gv@trace.wisc.edu; Henry
> > > > > Sinnreich; Gunnar Hellstrom; Mike Spanner;
> > > > > andreas.piirimets@omnitor.se; trefor@trefor.plus.com; 'Rajesh
> > > > > Kumar'; marcelo@envilogg.se; jorgen.bjorkner@hotsip.com;
> > > > > James Hamlin; Erik.Zetterstrom@omnitor.se;
> > > > > hwildfeu@cisco.com; barry.dingle@bigfoot.com.au;
> > > > > mgarakan@cisco.com; oran@cisco.com; sipping@ietf.org; Andrea Saks
> > > > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > > > >
> > > > > Rohan,
> > > > >
> > > > > I offered up a few questions to folks that I hoped people
> > > > > would have voted on.  The questions I had was whether or not
> > > > > there should be a requirement to provide *some* interface to
> > > > > legacy TTY devices.  Whether it's RJ-11 or 2.5mm makes no
> > > > > difference.  However, I think it should be a requirement that
> > > > > pass-through work at the very least.  If this was agreeable,
> > > > > I could certainly work to craft specific text for such a
> requirement.
> > > > >
> > > > > I do not have stats on how many phones do not have screens.
> > > > > However, note that requirement 1 also requires a keyboard,
> > > > > though that's never mentioned.
> > > > > One could say that text could be entered via some web
> > > > > interface, but doing so is a bit hard in a hotel room or at a
> > > > > gas station on the side of the road.  Regardless, I've said
> > > > > before and I'll say it again: I do not expect that most of
> > > > > the IP phones that will go into mass production will have a
> > > > > suitable screen for text communication.  Most of today's IP
> > > > > phones have screens, because it is a way to catch people's
> > > > > eyes.  However, there are also many that do not.  Would you
> > > > > consider Cisco's ATA, for example, an IP phone?  I think it
> > > > > qualifies and there is no screen.  There are a number of
> > > > > service providers deploying that and similar devices from
> > > > > other manufacturers.  None of those have displays.
> > > > >
> > > > > I assert that in most enterprise environments, the display is
> > > > > of very little value.  It's interesting at first, but it's
> > > > > not necessary to have a fancy
> > > > > screen: just something simple to control hold, transfer,
> > > > > conferencing, etc., is enough.  In fact, I'd be happy with
> > > > > buttons that did that.
> > > > >
> > > > > At home, I think it's even less of a requirement to have a
> > > > > nice-size display.  Recognizing that many phones today have a
> > > > > display for caller ID, etc., such displays are not suitable
> > > > > for text communication.  Requirement 1 is asking for a nice
> > > > > display... and I think you'll find a much smaller percentage
> > > > > of those... especially as they become a commodity item.
> > > > >
> > > > > Paul
> > > > >
> > > > > ----- Original Message -----
> > > > > From: "Rohan Mahy" <rohan@cisco.com>
> > > > > To: "Paul E. Jones" <paulej@packetizer.com>
> > > > > Cc: <guido.gybels@rnid.org.uk>; <gv@trace.wisc.edu>; "Henry
> Sinnreich"
> > > > > <Henry.Sinnreich@mci.com>; "Gunnar Hellstrom"
> > > > > <gunnar.hellstrom@omnitor.se>; "Mike Spanner"
> > > > > <mike.spanner@rnid.org.uk>; <andreas.piirimets@omnitor.se>;
> > > > > <trefor@trefor.plus.com>; "'Rajesh Kumar'"
> > > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > > <jorgen.bjorkner@hotsip.com>; "James Hamlin"
> > > > > <James.Hamlin@rnid.org.uk>; <Erik.Zetterstrom@omnitor.se>;
> > > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > > <mgarakan@cisco.com>; <oran@cisco.com>; <sipping@ietf.org>;
> > > > > "Andrea Saks" <asaks@waitrose.com>
> > > > > Sent: Saturday, July 19, 2003 11:14 PM
> > > > > Subject: Re: [Sipping] Re: items for vor voting in the BOF
> > > > >
> > > > >
> > > > > > Hi Paul,
> > > > > >
> > > > > > So it sounds like you wanted an additional requirement for
> > > > > pass-through
> > > > > > via a 2.5 mm jack for phones that do not support RFC2793?  If I
> > > > > > understood this correctly, how would you word the requirement?
> > > > > >
> > > > > > Also, do you have any data on how many phones currently
> > > > > exist (number
> > > > > > or percentage) for which requirement 1 would not apply? I
> > > > > suspect that
> > > > > > the number of phones not covered by the first requirement
> > > > > is actually
> > > > > > quite small.
> > > > > >
> > > > > > thanks,
> > > > > > -rohan
> > > > > >
> > > > > > On Friday, July 18, 2003, at 05:04 PM, Paul E. Jones wrote:
> > > > > >
> > > > > > > Guido,
> > > > > > >
> > > > > > > I wish I could have attended, but I was attending a TDI
> conference
> > > > > > > here in
> > > > > > > the US.  I couldn't be in both places at the same time,
> > > > > though I wish I
> > > > > > > could.  I was disheartened that the questions were not
> > > > > presented as I
> > > > > > > proposed.  You should know that, supposedly, work gets
> > > > > done over the
> > > > > > > e-mail
> > > > > > > reflector, not only in the meeting, so not being there
> > > > > should not have
> > > > > > > lessened my input on the matter.
> > > > > > >
> > > > > > > Looking at the results of the ad-hoc reported by Rohan, I see:
> > > > > > >   1) *If* an IP phone supports a display (and keyboard??), it
> > > > > > >      must support RFC 2793
> > > > > > >   2) IP phones *should* provide a way to input text
> > > > > > >   3) IP phones *may* provide an RJ-11 interface and *must*
> > > > > > >      support pass-through.
> > > > > > >
> > > > > > > Now, my concerns are several.  It appears that you have
> > > > > absolutely no
> > > > > > > requirement for the mass of products that do not sport a
> > > > > display to
> > > > > > > support
> > > > > > > text in either relay or pass-through mode.  Further, there is
> no
> > > > > > > mention of
> > > > > > > a 2.5mm jack being an option.  Question 3, as I
> > > > > mentioned, "pressured"
> > > > > > > folks
> > > > > > > more than it should.  There should have been no reason to
> require
> > > > > > > support of
> > > > > > > pass-through, at the very least.
> > > > > > >
> > > > > > > I told you that my questions were to ensure that we at least
> get
> > > > > > > support for
> > > > > > > text on every phone in one way or another.  It looks like
> > > > > you lost.
> > > > > > > Company
> > > > > > > X is now free to build an IP phone without any text
> > > > > support.  I'm not
> > > > > > > really
> > > > > > > happy about that and I certainly don't see why you should be.
> > > > > > >
> > > > > > > Paul
> > > > > > >
> > > > > > > ----- Original Message -----
> > > > > > > From: "Guido Gybels" <ggybels@webclient.eu.blackberry.net>
> > > > > > > To: "Paul E. Jones" <paulej@packetizer.com>;
> <gv@trace.wisc.edu>
> > > > > > > Cc: "Henry Sinnreich" <Henry.Sinnreich@mci.com>; "Gunnar
> > > > > Hellstrom"
> > > > > > > <gunnar.hellstrom@omnitor.se>; "Guido Gybels"
> > > > > > > <Guido.Gybels@rnid.org.uk>;
> > > > > > > "Mike Spanner" <mike.spanner@rnid.org.uk>;
> > > > > > > <andreas.piirimets@omnitor.se>;
> > > > > > > <gv@trace.wisc.edu>; <trefor@trefor.plus.com>; "'Rajesh
> Kumar'"
> > > > > > > <rkumar@cisco.com>; <marcelo@envilogg.se>;
> > > > > > > <jorgen.bjorkner@hotsip.com>;
> > > > > > > "James Hamlin" <James.Hamlin@rnid.org.uk>;
> > > > > > > <Erik.Zetterstrom@omnitor.se>;
> > > > > > > <hwildfeu@cisco.com>; <barry.dingle@bigfoot.com.au>;
> > > > > > > <mgarakan@cisco.com>;
> > > > > > > <oran@cisco.com>; <rohan@cisco.com>; <sipping@ietf.org>
> > > > > > > Sent: Friday, July 18, 2003 2:31 AM
> > > > > > > Subject: [Sipping] Re: items for vor voting in the BOF
> > > > > > >
> > > > > > >
> > > > > > >> Paul,
> > > > > > >>
> > > > > > >> <<While I guess it might be a bit late now>>
> > > > > > >>
> > > > > > >> The discussion on the sipdev document took place
> > > > > yesterday, and if you
> > > > > > > wanted to make a point you were free to attend it. You'll see
> the
> > > > > > > results
> > > > > > > reflected in the next version of the document, and I can
> > > > > say that both
> > > > > > > Arnoud and myself believe this result to be extremely
> > > > > sensible as well
> > > > > > > as in
> > > > > > > the best interest of all stakeholders.
> > > > > > >> I also suggest this no longer belongs here on the
> > > > > mailing list until
> > > > > > >> a new
> > > > > > > issue would arise. I have stopped replying to your emails
> > > > > a while ago
> > > > > > > as I
> > > > > > > see no point in continuing the discussion (and swamping
> > > > > the list) with
> > > > > > > someone who, despite being a native English speaker,
> > > > > seems unable to
> > > > > > > understand what we are saying and more importantly
> deliberately
> > > > > > > misrepresents my statements or attributes words to my
> > > > > mouth that I did
> > > > > > > not
> > > > > > > speak.
> > > > > > >>
> > > > > > >> <<would guess most people will say "yes" to, while I
> > > > > know you'd say
> > > > > > >> "no"
> > > > > > > to
> > > > > > >> it.>>
> > > > > > >>
> > > > > > >> This being a nice example of that.
> > > > > > >>
> > > > > > >>
> > > > > > >> I believe the IETF community has shown great wisdom and a
> true
> > > > > > > understanding of what is at stake here. We have now got a
> > > > > solid path
> > > > > > > to the
> > > > > > > future, with a clear strategy to get rid of obsolete and
> > > > > unsatisfactory
> > > > > > > solutions allowing a gradual move of deaf, hard of
> > > > > hearing and speech
> > > > > > > impaired people into mainstream technology. That will of
> > > > > course take
> > > > > > > time
> > > > > > > during which there will have to be legacy support and all the
> > > > > > > instruments
> > > > > > > are put in place to do exactly that.
> > > > > > >>
> > > > > > >> Guido
> > > > > > >>
> > > > > > >> Guido Gybels
> > > > > > >> Director of New Technologies
> > > > > > >>
> > > > > > >> RNID, 19-23 Featherstone Street
> > > > > > >> London EC1Y 8SL, UK
> > > > > > >> Tel +44(0)20-7294 3713
> > > > > > >> Fax +44(0)20-7296 8069
> > > > > > >>
> > > > > > >> _______________________________________________
> > > > > > >> 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
> >


_______________________________________________
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: AW: AW: [Sipping] FYI: RADIUS & SIP
Date: Thu, 14 Aug 2003 15:31:16 +0300
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Hi Jiri,

> >Accounting over a non-reliable transport protocol is a Very Hard
> >Problem. That's one of the reaons for DIAMETER.
>=20
> Still less hard than accounting over a protocol which does not exist.

This may be a tupid comment,  but as I understand, there is no =
'standard'=20
solution for this with RADIUS, so it seems that we are in the same boat, =
no
matter which protocol we choose.

John

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From: Jari Arkko <jari.arkko@piuha.net>
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Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
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> This may be a tupid comment,  but as I understand, there is no 'standard' 
> solution for this with RADIUS, so it seems that we are in the same boat, no
> matter which protocol we choose.

Some RADIUS improvements can be made without deployment barriers,
e.g. exponential backoff for transport. Some others require more
fundamental changes to the protocols, in which case you'd have
to both develop the protocol and deploy the new servers/features.
In the case of Diameter, the protocol work is done, but the deployment
is mostly ahead. There's one open source project, by the way, look
for it at sourceforge.net if you're interested.

--Jari



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Subject: RE: [Sipping] WGLC for Q.SIP-SIP mapping draft
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From: "Juan Luis Esteban" <jlesteban@capcom.net>
To: "Luis Emilio Martin" <lemr@tid.es>, <thomas.stach@siemens.com>,
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Luis,

Most T.38 GW implementations rely on D.2.2.4/T.38 procedures (Voice and =
facsimile connection), where an audio session is established and =
switched over to T.38 on V.21 preamble fax seq detection by the =
receiving GW. This scenario is described also in =
5.2.1/draft-mule-sip-t38callflows-02. A reINVITE is issued by the =
receiving GW to modify the session SDP.

IMHO 5.1.2 is limited to the case where IAFs are involved in the =
transmission which is also described in D.2.2.3/T.38. Universal GWs =
supporting voice and fax simultaneously will most likely implement voice =
to fax switchover according to D.2.2.4/T.38, which seems to be obvious.

Does anyone know about SIP GWs implementing D.2.2.3 procedures ?.

regards,


Juan Luis Esteban
Capcom-


-----Mensaje original-----
De: Luis Emilio Martin [mailto:lemr@tid.es]
Enviado el: mi=E9rcoles, 13 de agosto de 2003 11:08
Para: thomas.stach@siemens.com; sipping@ietf.org
CC: jieying.li@clarent.com; jfmule@packetizer.com
Asunto: WG: [Sipping] WGLC for Q.SIP-SIP mapping draft


Hello,

> The QSIG SETUP message may optionally transport HLC and LLC =
Information elements which are not mentioned
> in the draft. You might want to add something like into section 10:
>
>"The gateway SHOULD (or SHALL(?)) not include a Low layer compatibility =
in the QSIG SETUP message. The=20
> gateway SHOULD not include a High layer compatibility information =
element in the QSIG SETUP message.
>
>NOTE: Inclusion of a HLC in the QSIG SETUP message might be useful, if =
the gateway knows prior to=20
> sending the QSIG SETUP message that fax is being transmitted (e.g., =
because of a corresponding
> SDP offer), it may include in the QSIG SETUP message a High layer =
compatibility information element=20
> indicating group 3 fax."

It is discussed the possibility to include HLC IE in the QSIG SETUP if
an INVITE
was received with fax transmission information (i.e. T38 only), so that
fax
detection at destination does not depend on processing audio data. It
possibly is a matter of
configuration in the gateway.

Sorry I am not very used to T.38 procedures and Fax Group 2/3 methods
but we are testing some gateways
with T.38 and automatic fax detection capabilities based on DSPs and
experiencing some strange=20
behaviours in these gateways, so that SETUP messages including HLC
produce INVITEs offering G.729 or=20
G.711 and a bit later new INVITES offering t38.

In reverse direction, can a SIP gateway, upon reception of a SETUP
Message with Bearer Capability
3,1 kHz Audio and an HLC indicating Fax Group 2/3, to map the HLC field
into an INVITE message
with an SDP offering T.38 only (not any other audio codec). The example
is
taken from chapter 5.1.2
http://www.watersprings.org/pub/id/draft-mule-sip-t38callflows-02.txt :


      Content-Type: application/sdp
      Content-Length: 320

      v=3D0
      o=3Difaxgw1 2890846527 2890846527 IN IP4 ifax.here.com
      s=3DSession SDP
      c=3DIN IP4 ifaxmg.here.com
      t=3D0 0
      m=3Dimage 15002 udptl t38
      a=3DT38FaxVersion:0
      a=3DT38MaxBitRate:14400
      a=3DT38FaxFillBitRemoval:0
      a=3DT38FaxTranscodingMMR:0
      a=3DT38FaxTranscodingJBIG:0
      a=3DT38FaxRateManagement:transferredTCF


It should be the most straightforward procedure to map fax but is my
interpretation correct?...
or the reception of a setUp with such IE values does not have enough
information to=20
include an SDP offering with T.38 only, so it is required to wait for
CNG/V21 detection=20
to pick all the information required, and a renegotiation is thus
required after call is
stablished?


Thank you very much in advance

Best Regards

Luis Martin

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One short comment below.

At 10:54 PM 8/13/2003 -0700, Cullen Jennings wrote:

>I think all of these proposals are significantly more complex than using a
>MESSAGE/NEW_METHOD/NOTIFY. I will mention a few issues with the proposal
>inline but first let me a few other proposals...
>
>All of these have a call flow like
>
> > UAC           App Server          UAS
> > 1|-----INVITE---->|                |
> > 2|                |-----INVITE---->|
> > 3|                |<-----200-------|
> > 4|<-----200-------|                |
> > 5|---SIP MESG---->|                |
> > 6|<-----200-------|                |
>
>The question for KMPL is, what method to use in msg 5.
>
>Proposal 1) Use MESSAGE. Many people think the semantics of message are
>display this IM to a user. Perhaps, but I send IM to applications and they
>send me back things like stock quotes all the time. I have no problem with
>semantics of MESSAGE being deliver this message to the addressed user or
>automaton. That's what we are doing here. Misrouted message will cause
>trivial problems, but this is a bogus issue - misrouted INVITE or MESSAGE
>will cause problems to - don't misroute things.

Here's the problem I have with this.  Lets say I have a dumb SIP phone that 
has no display but does support KPML
I also have a separate SIP IM client.  My proxy has very simple logic - 
MESSAGE requests directed at my phone get redirected to my
IM client.  Using MESSAGE like this breaks this very simple (and 
potentially useful) logic.  My proxy now has to take apart
the MESSAGE request and determine if it is a IM MESSAGE, or an RPC MESSAGE 
(or whatever we will call this new use
for MESSAGE.

This is the type of MESSAGE routing problems I'm concerned about.

Thanks,
Alan Johnston
MCI

>Proposal 2) Use some new method like STIMULUS. The semantics of this are
>some user input stimulus is being sent - STIMULUS is to input what MESSAGE
>is to output.
>
>Proposal 3) Use NOTIFY. Semantics are, here is some information about some
>event that you requested to be informed about. The usual complaint on this
>one is that there was no Subscribe, or it was implicit, or it was explicit,
>or it violated some law of the SIP world. I think there was a subscribe, it
>was in the message that initiated the script. There is also a clear time it
>will end - when the script is terminated or when the dialog of the message
>that initiated the script ends. We all remember that implicit subscribes
>where bad (whatever an implicit subscribe is) but people seem to have a hard
>time remembering why this will be bad here.
>
>All 3 of these proposals have something in common - there is no concrete
>argument that any of them would not work. They send to a URI the AS provided
>so there are no forking issues. There is some grief on the semantics. Option
>2 has the advantage of it lets us define the semantics however we want. I
>think that 2 is the best way to go but I don't care greatly. I do need
>convincing on what's wrong with these to move to something more complex.
>
>More comments inline
>
>On 8/13/03 3:32 PM, "Adam Roach" <adam@dynamicsoft.com> wrote:
>
> > After considering Jonathan's contention that defining
> > a new method for conveying stimulus is creating something
> > that is similar in meaning to NOTIFY, I'm inclined to
> > agree to some point. There are still subtle differences
> > in meaning, so I'm not completely convinced that a new
> > method is wrong, but I understand the point he's trying
> > to make.
> >
> > After discussing it privately with others, I've concluded
> > that there are three viable models worth considering. There
> > are, of course, some minor variations that can be made to
> > each.
> >
> > 1) Standard HTTP Handling
> >
> > UAC           App Server          UAS
> > 1|-----INVITE---->|                |
> > 2|                |-----INVITE---->|
> > 3|                |<-----200-------|
> > 4|<-----200-------|                |   AS adds HTTP URL to 200
> > 5|---HTTP GET---->|                |   UAC requests KPML script
> > 6|<-----200-------|                |   AS replies with script
> > 7|---HTTP POST--->|                |   UAC provides digits
> > 8|<-----200-------|                |   AS acknowledges digits
> >
> > Of the three options presented here, I see this as the best
> > solution. Basically, the app server includes an HTTP URL
> > in the 200 response to the INVITE that it has proxied. The
> > UAC then does a GET on the KPML script. Just like with
> > HTML forms, the UAC does whatever local stuff is necessary
> > to "fill out the form" (in this case, collect digits), and
> > then posts the result back to the App Server. This is all
> > plain-Jane unextended HTTP as it was intended to work.
> >
>
>One of the difficult parts of this is security.
>
>This will get more complicated once you provide a way to send in message 4
>the credentials that will allow 7. It may also make the App-Info header only
>really work with scripts that use HTTP to post instead of other things.
>Also, in the case of a financial institutions that have to log electronic
>communications, you will need to provide way to insert logging HTTP proxies
>and probably proxies for FW traversal too. Of course the AS or the UAC will
>need to figure out the network topology to know if the AS is inside or
>outside the FW. It's possible to configure all the domain and HTTP proxies
>rules into the device but it still hard to secure - particularly on devices
>that don't want to implement HTTPS due to not much memory.
>
>
> > 2) Explicit Subscriptions (with GRUU)
> >
> > UAC           App Server          UAS
> > 1|-----INVITE---->|                |
> > 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
> > 3|                |<-----200-------|
> > 4|<-----200-------|                |
> > 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> > 6|------200------>|                |
> > 7|-----NOTIFY---->|                |
> > 8|<-----200-------|                |
> > 9|-----NOTIFY---->|                |   UAC provides digits
> > A|<-----200-------|                |   AS acknowledges digits
> >
> > My second choice would be an explicit subscription to stimulus
> > events. The only sticky part in this solution is the question
> > of how the AS can route a SUBSCRIBE to the UAC. Once we have
> > a GRUU solution in place, this is a no brainer: the AS can
> > extract the GRUU from the INVITE, and use it as the target
> > of its SUBSCRIBE. The KPML script is included in the SUBSCRIBE,
> > and essentially serves as a filter.
> >
>
>I sort of see this one, but what if the App Server wanted to run an App on
>the UAS instead of the UAC. Would the 180 have the GRU? Would it be
>reliable?
>
>I seem to recall someone asking about the expire time on the subscription
>:-)
>
>I'd be keener on GRUU solutions if GRUU was a solved problem.
>
> > 3) Explicit Subscriptions (with REFER)
> >
> > UAC           App Server          UAS
> > 1|-----INVITE---->|                |
> > 2|                |-----INVITE---->|
> > 3|                |<-----200-------|
> > 4|<-----200-------|                |   AS adds SIP URL to 200
> > 5|-----REFER----->|                |
> > 6|<-----200-------|                |
> > 7|<----NOTIFY-----|                |
> > 8|------481------>|                |
> > 9|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> > A|------200------>|                |
> > B|-----NOTIFY---->|                |
> > C|<-----200-------|                |
> > D|-----NOTIFY---->|                |   UAC provides digits
> > E|<-----200-------|                |   AS acknowledges digits
> >
> > Although somewhat chattier than the previous solution, this gets around
> > the GRUU issue by having the AS provide a SIP URI that routes to itself
> > in the 200 response. The UAC then uses this URI to send a REFER to the
> > AS. This REFER tells the AS to SUBSCRIBE to the UAC. The flow from that
> > point forward is the same as for solution 2.
> >
>
>I got to admit, I had never considered anything like this. I think it is
>going to be hard to convince folks this is really easier and better than
>sending DTMF over INFO. Keep in mind that KPML is for little device that
>think 14k of memory is a lot. Real devices are just going to do VXML and
>HTML and ignore KPML.
>
>Cullen
>
>
>_______________________________________________
>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: Thu, 14 Aug 2003 08:57:45 -0700
Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Alan Johnston <alan.johnston@mci.com>, sipping <sipping@ietf.org>
CC: Adam Roach <adam@dynamicsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Ah, this is worth discussing - I don't think it is a problem. The proposal
is to use MESSAGE not as the way to initiate the script on the UA but as the
way for the UA to tell the App Server the results of the script.

Lets say the AOR of the UAC was alice@atlanta, UAS was bob@bilboxi. The KPML
script would contain the address of the app server. So say the script had
bobs-apps@server1.example.com, the URI of message with the KPML result from
the UA to the appserver would be sip:bobs-apps@server1.example.com. The
proxy would just route normally to server1 and would not need to know
anything special. A MESSAGE address to bob or alice would go to them.

Cullen


On 8/14/03 6:47 AM, "Alan Johnston" <alan.johnston@mci.com> wrote:

> One short comment below.
> 
> At 10:54 PM 8/13/2003 -0700, Cullen Jennings wrote:
> 
>> I think all of these proposals are significantly more complex than using a
>> MESSAGE/NEW_METHOD/NOTIFY. I will mention a few issues with the proposal
>> inline but first let me a few other proposals...
>> 
>> All of these have a call flow like
>> 
>>> UAC           App Server          UAS
>>> 1|-----INVITE---->|                |
>>> 2|                |-----INVITE---->|
>>> 3|                |<-----200-------|
>>> 4|<-----200-------|                |
>>> 5|---SIP MESG---->|                |
>>> 6|<-----200-------|                |
>> 
>> The question for KMPL is, what method to use in msg 5.
>> 
>> Proposal 1) Use MESSAGE. Many people think the semantics of message are
>> display this IM to a user. Perhaps, but I send IM to applications and they
>> send me back things like stock quotes all the time. I have no problem with
>> semantics of MESSAGE being deliver this message to the addressed user or
>> automaton. That's what we are doing here. Misrouted message will cause
>> trivial problems, but this is a bogus issue - misrouted INVITE or MESSAGE
>> will cause problems to - don't misroute things.
> 
> Here's the problem I have with this.  Lets say I have a dumb SIP phone that
> has no display but does support KPML
> I also have a separate SIP IM client.  My proxy has very simple logic -
> MESSAGE requests directed at my phone get redirected to my
> IM client.  Using MESSAGE like this breaks this very simple (and
> potentially useful) logic.  My proxy now has to take apart
> the MESSAGE request and determine if it is a IM MESSAGE, or an RPC MESSAGE
> (or whatever we will call this new use
> for MESSAGE.
> 
> This is the type of MESSAGE routing problems I'm concerned about.
> 
> Thanks,
> Alan Johnston
> MCI
>


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Cullen Jennings [mailto:fluffy@cisco.com] writes:

> I seem to recall someone asking about the expire time on the 
> subscription

You appear to have misinterpreted what I said in the same
way as Jonathan did.

The issue I was attempting to raise is that subscriptions,
as defined in 3265, always have a well-defined number of
seconds before they expire. This amount of time can be
expressed using an integer. If you start defining this
period of time to be "until some other event happens,
and we don't know when that will occur", you have to
significantly modify RFC 3265 behavior.

So, for the two SUBSCRIBE solutions I describe, the
subscription would initially be for some arbitrary period
of time (say, two hours). If the need for interaction
lasts longer than that, you refresh. If the interaction
is shorter, you unsubscribe.

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Adam Roach
	 <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Thu, 14 Aug 2003 12:30:57 -0500
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Cullen Jennings [mailto:fluffy@cisco.com] writes:

> > 1) Standard HTTP Handling
> > 
> > UAC           App Server          UAS
> > 1|-----INVITE---->|                |
> > 2|                |-----INVITE---->|
> > 3|                |<-----200-------|
> > 4|<-----200-------|                |   AS adds HTTP URL to 200
> > 5|---HTTP GET---->|                |   UAC requests KPML script
> > 6|<-----200-------|                |   AS replies with script
> > 7|---HTTP POST--->|                |   UAC provides digits
> > 8|<-----200-------|                |   AS acknowledges digits
> > 
>
> One of the difficult parts of this is security.
> 
> This will get more complicated once you provide a way to send 
> in message 4
> the credentials that will allow 7.

Red herring. The exact same issue applies to any of the cases you
describe. In other words, for the following call flow, you need
to get the UAC credentials for step 5:

> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|
> 3|                |<-----200-------|
> 4|<-----200-------|                |
> 5|---SIP MESG---->|                |
> 6|<-----200-------|                |

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Adam Roach
	 <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Thu, 14 Aug 2003 12:33:08 -0500
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Cullen Jennings writes:

> UAC           App Server          UAS
> 1|-----INVITE---->|                |
> 2|                |-----INVITE---->|
> 3|                |<-----200-------|
> 4|<-----200-------|                |
> 5|---SIP MESG---->|                |
> 6|<-----200-------|                |

Are you proposing that the App Server would send the KPML script
in the INVITE 200? Where? In the body? In an egregiously large
header?

/a

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Date: Thu, 14 Aug 2003 12:28:53 -0700
Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On 8/14/03 10:26 AM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Cullen Jennings [mailto:fluffy@cisco.com] writes:
> 
>> I seem to recall someone asking about the expire time on the
>> subscription
> 
> You appear to have misinterpreted what I said in the same
> way as Jonathan did.
> 
> The issue I was attempting to raise is that subscriptions,
> as defined in 3265, always have a well-defined number of
> seconds before they expire. This amount of time can be
> expressed using an integer. If you start defining this
> period of time to be "until some other event happens,
> and we don't know when that will occur", you have to
> significantly modify RFC 3265 behavior.
> 
> So, for the two SUBSCRIBE solutions I describe, the
> subscription would initially be for some arbitrary period
> of time (say, two hours). If the need for interaction
> lasts longer than that, you refresh. If the interaction
> is shorter, you unsubscribe.
> 
> /a
> 
> 

I had wondered about how to solve this - I think putting in a really large
number is more or less what people had imagined. If you were running a
session timer on the original dialog that time might be usable. If you sent
a notify with 100 minutes, then 20 minutes later sent a new notify with 100
minutes (and there had been no subscribe in the meantime) would that violate
3265? What does the implicit subscribe in refer do? Anyways, I'm not really
arguing for NOTIFY but this is one of the issues we need to figure out to
use it. 





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To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Adam Roach
	 <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Thu, 14 Aug 2003 15:08:50 -0500
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Cullen Jennings writes:

> What does the implicit subscribe in refer do?

To prevent expiration, it sends explicit SUBSCRIBE
messages to refresh the subscription.

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Adam Roach
	 <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Thu, 14 Aug 2003 18:11:33 -0500
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Cullen Jennings [mailto:fluffy@cisco.com] writes:

> Proposal 3) Use NOTIFY. Semantics are, here is some 
> information about some event that you requested to
> be informed about. The usual complaint on this one
> is that there was no Subscribe, or it was implicit,
> or it was explicit, or it violated some law of the
> SIP world. I think there was a subscribe, it was in
> the message that initiated the script. There is also
> a clear time it will end - when the script is terminated
> or when the dialog of the message that initiated the
> script ends. We all remember that implicit subscribes
> where bad (whatever an implicit subscribe is) but people
> seem to have a hard time remembering why this will be
> bad here.

It's not simple dogma; there are real problems here. I've
cited a few of them (and, as I point out in another message,
Jonathan's rebuttal is somewhat lacking in the area of
duration; also, I still plan to respond on the correlation
issue, because I think his rebuttal glosses over the
details).

Can these problems be fixed? Probably. REFER provides an
existence proof that implicit subscriptions[1] can be made
to work with enough crafting. After it's said and done,
do we regret choosing that path? I do. Robert does. Do we
want to learn from our mistakes?

Your phrasing above knocked loose in my mind another reason
that implicit subscriptions[1] are ill-advised: you are using
a single transaction to perform two operations. Conveying
partial success (e.g. the call completed, but the subscription
failed) is problematic. We have made efforts to avoid this
type of situation in the past.

/a

[1] To be clear, I mean "subscriptions put in place via some
    other mechanism than that described in RFC 3265."

_______________________________________________
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: Adam Roach <adam@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Tim Melanchuk'" <timm@convedia.com>,
        Cullen Jennings
	 <fluffy@cisco.com>, sipping <sipping@ietf.org>
Subject: RE: [Sipping] DTMF: Why Not A New Method?
Date: Thu, 14 Aug 2003 18:19:18 -0500
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Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:

> Correlation is easy. The KPML script IS the subscription. The 
> place to  send the notify is implied from the URI in the KPML
> script. We would need to clarify in KPML that once that URI is
> used, subsequent notificatiosn to the same URI ought to go on
> the dialog established by the NOTIFY. But, this is a KPML issue.

I mean dialog correlation.

Let's say you're an app server. Client A sends an INVITE
through you. You add something to the 200 response that
ultimately results in Client A getting the following KPML
script:

<?xml version="1.0">
<kpml version="1.0">
  <request>
    <pattern>
      <regex value="xx" href="sip:appserver.example.com" />
    </pattern>
  </request>
</kpml>

Client A sends another INVITE through you. You add something to
the 200 response that ultimately results in Client A getting
the following KPML script:

<?xml version="1.0">
<kpml version="1.0">
  <request>
    <pattern>
      <regex value="xx" href="sip:appserver.example.com" />
    </pattern>
  </request>
</kpml>

Finally, you get a NOTIFY request from the user. How do you
know to which session it pertains?

/a

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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] App-Interaction: Considering models
Date: Fri, 15 Aug 2003 11:46:45 +0300
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Thread-Topic: [Sipping] App-Interaction: Considering models
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I tend to strongly agree with Adam; unless we plan to write an RFC that =
contains the top 10 tests for implicit subscriptions, we need to forbid =
it.

I personally don't see any problem with sending a SUBSCRIBE explicitly. =
An extra round trip delay, so what?

My other favourite is a new method similar to NOTIFY, like REPORT. We =
have done this in the past were a method semantics are not quite right =
for one thing and we create a new method. Take for example REGISTER and =
PUBLISH.

Regards,
Hisham

> -----Original Message-----
> From: ext Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Friday, August 15, 2003 2:12 AM
> To: 'Cullen Jennings'; Adam Roach; sipping; Jonathan Rosenberg
> Subject: RE: [Sipping] App-Interaction: Considering models
>=20
>=20
> Cullen Jennings [mailto:fluffy@cisco.com] writes:
>=20
> > Proposal 3) Use NOTIFY. Semantics are, here is some=20
> > information about some event that you requested to
> > be informed about. The usual complaint on this one
> > is that there was no Subscribe, or it was implicit,
> > or it was explicit, or it violated some law of the
> > SIP world. I think there was a subscribe, it was in
> > the message that initiated the script. There is also
> > a clear time it will end - when the script is terminated
> > or when the dialog of the message that initiated the
> > script ends. We all remember that implicit subscribes
> > where bad (whatever an implicit subscribe is) but people
> > seem to have a hard time remembering why this will be
> > bad here.
>=20
> It's not simple dogma; there are real problems here. I've
> cited a few of them (and, as I point out in another message,
> Jonathan's rebuttal is somewhat lacking in the area of
> duration; also, I still plan to respond on the correlation
> issue, because I think his rebuttal glosses over the
> details).
>=20
> Can these problems be fixed? Probably. REFER provides an
> existence proof that implicit subscriptions[1] can be made
> to work with enough crafting. After it's said and done,
> do we regret choosing that path? I do. Robert does. Do we
> want to learn from our mistakes?
>=20
> Your phrasing above knocked loose in my mind another reason
> that implicit subscriptions[1] are ill-advised: you are using
> a single transaction to perform two operations. Conveying
> partial success (e.g. the call completed, but the subscription
> failed) is problematic. We have made efforts to avoid this
> type of situation in the past.
>=20
> /a
>=20
> [1] To be clear, I mean "subscriptions put in place via some
>     other mechanism than that described in RFC 3265."
>=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

_______________________________________________
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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Date: Fri, 15 Aug 2003 08:53:58 -0400
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inline.

Rohan Mahy wrote:

> 
> On Wednesday, August 13, 2003, at 10:55 AM, Adam Roach wrote:
> 
>> Rohan Mahy [mailto:rohan@cisco.com] writes:
>>
>>> Most of the SIP phones I see at SIPit support
>>> completing a transfer using a REFER with a
>>> Replaces header embedded in the Refer-To.
>>> The Refer-To URI is always set to the Contact
>>> URI of the appropriate UA.
>>
>>
>> And this will continue to work as well as it does today.
>> Newer, GRUU aware clients will simply exhibit an additional
>> robustness.
>>
>> It seems a bit misguided to do things in a suboptimal way
>> just to try to fix currently deployed clients.
> 
> 
> and Jonathan wrote:
> 
>>>> I don't think we have so much of either of these deployed that this 
>>>> cannot change.
>>>
>>> I disagree.  Most of the SIP phones I see at SIPit support completing 
>>> a transfer using a REFER with a Replaces header embedded in the 
>>> Refer-To.  The Refer-To URI is always set to the Contact URI of the 
>>> appropriate UA.
>>
>>
>> Well, we have here the standard problem here of having implementations 
>> of long-running drafts, and then deciding down the road that something 
>> needs to change.
>>
>> I think we want a non-GRUU in the Contact header (to avoid spirals 
>> through proxies, and needless load when the proxy doesnt intend to 
>> record-route), but we want a GRUU for attended-transfer. I would 
>> propose to use a separate header to identify GRUUs. Existing phones 
>> will work as they currently do, and when two phones both support GRUU, 
>> you get more reliable service. I recognize that putting the GRUU in 
>> the contact would mean only one of the phones needs to support it 
>> (clearly better), but the cost of this is a long-term situation that 
>> we don't want.
>>
>> -Jonathan R.
> 
> 
> Hi,
> 
> I think both of you are ignoring the "generality before efficiency" 
> principle.

I don't think that is the tradeoff here. Given a clean slate, it is 
clear that a new header would be used. The argument for contact header 
is just to facilitate introduction of GRUU. Thus, its more of the 
"backwards compatible vs. right thing" kind of debate.

> Putting the GRUU in the Contact header always works, and in 
> most deployments where it is needed (middlebox traversal), you WILL need 
> to travel through the same set of proxies for nearly every request 
> anyhow. With the increased deployment of TLS and SIPS we will 
> increasingly have dialogs for which all or none of the requests are 
> proxied.

A few responses here. Firstly, your comments above would seem to imply 
that a UA wouldn't always put the GRUU in the Contact header - that it 
would do so only in situations where it was needed. The problem is, 
the UA can't know when its needed. It would be needed if someone else 
wants to transfer a call to it, or subscribe to its dialog package. 
Since the UA can't know that ahead of time, for it to work, we need to 
specify that all UAs should always use GRUUs all the time. If the GRUU 
is placed in the Contact header, this would mean the end of e2e sip 
signaling, PERIOD.

Even if you assume a proxy needs to be on the path, a proxy then is 
faced with an impossible decision. When it receives a request, does 
the proxy need to record route (if the downstream UA is using a GRUU) 
or not (if it isn't). It doesn't know what the downstream UA can do at 
this point. So, it can't make a correct decision. Existing proxies 
wouldn't even know about this decision. The result would be that, 
where proxies record-route and the downstream UA uses a GRUU, requests 
will spiral or take extra hops on every call. This results in 
decreased overall performance in the network and increased call setup 
times.


> 
> We are talking about interop with thousands of UAs which have 
> implemented Proposed Standard level RFCs (SIP and Refer) and an 
> long-lived, stable I-D that has completed WGLC and was submitted to the 
> IESG (Replaces)`.  We are not talking about random drafts that very few 
> have implemented.

These drafts are standardized, yes, but their usage for assisted call 
transfer (which is what we are talking about here) has clearly been 
changing and is still an I-D. Nothing in the specs you mention tells 
people to extract the Refer-To URI from the Contact header of the peer.


> To be convinced that it is worth adding a new GRUU 
> header, I would at least like to see some plausible use cases that are 
> common enough for us to be worrying about.  Maybe the right way to look 
> at this is to put the GRUU in the Contact, UNLESS you can tell that your 
> peer understands the special significance of GRUUs.

Hopefully my explanation above describes the problems with putting 
GRUU in contacts. Even if you knew that all of your peers supported 
Contact headers, putting one in would have the problems described above.

IMHO, the GRUU is a lot like the Reply-To URI. Its a URI that provides 
an additional means of communication in the future, outside of the 
current dialog. Just as the Reply-To URI is indicated with a header, 
so to is the GRUU indicated with a header.

> 
> Also, where this extension is most needed, you need the right 2 out of 3 
> to support the extension.  (This isn't just any two of three.  If Bob 
> supports the GRUU extension, but Carol does not, Alice still cannot 
> cause a transfer to occur in the appropriate direction.)
> 
> Starting with the classic Alice, Bob, and Carol actors.
> 
> Alice and Bob have an existing dialog.
> Alice and Carol have an existing dialog.
> Alice wants Bob to talk to Carol, and will REFER Bob to INVITE Carol 
> replacing the dialog between Alice and Carol.
> 
> Carol needs a GRUU; if the GRUU is conveyed in a separate header Alice 
> and Carol both need to support the GRUU extension for the Refer-To 
> header to be set properly.

Yes, I agree with this. I agree its not optimal. With what I am 
proposing, things will work as well as they do today, no worse, no 
better, if Alice and Bob don't support GRUUs.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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Date: Fri, 15 Aug 2003 10:10:10 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Re: Comment on GRUU requirements
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At 01:58 PM 8/13/2003 -0400, Jonathan Rosenberg wrote:


>Rohan Mahy wrote:
>
>
>>>I don't think we have so much of either of these deployed that this 
>>>cannot change.
>>
>>I disagree.  Most of the SIP phones I see at SIPit support completing a 
>>transfer using a REFER with a Replaces header embedded in the 
>>Refer-To.  The Refer-To URI is always set to the Contact URI of the 
>>appropriate UA.
>
>Well, we have here the standard problem here of having implementations of 
>long-running drafts, and then deciding down the road that something needs 
>to change.
>
>I think we want a non-GRUU in the Contact header (to avoid spirals through 
>proxies, and needless load when the proxy doesnt intend to record-route),

I think you are making an assumption that every GRUU must route through a 
proxy.  Why is that?  I have argued all along that most UAs already have a 
GRUU - it is their Contact URI.  Only in cases where a NAT/Firewall or some 
proxy screening logic is in the way does a GRUU need to route through a proxy.

Also, I don't think spirals are an issue since when the proxy first 
receives the GRUU, recognizes it as its own and perform the algorithm to 
extract the Contact, then send it there.

As for needless load, if the proxy doesn't record route then it obviously 
doesn't need to be there in the GRUU - the Contact must work just fine in 
reaching the UA or the existing dialog is broken.

Thanks,
Alan Johnston
MCI

>  but we want a GRUU for attended-transfer. I would propose to use a 
> separate header to identify GRUUs. Existing phones will work as they 
> currently do, and when two phones both support GRUU, you get more 
> reliable service. I recognize that putting the GRUU in the contact would 
> mean only one of the phones needs to support it (clearly better), but the 
> cost of this is a long-term situation that we don't want.
>
>-Jonathan R.
>
>
>
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>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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This list is for NEW development of the application of SIP
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Adam Roach wrote:

> Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:
> 
>  > Also note that this is NOT an implicit subscription.
>  > It is quite explicit. Its just not done with SUBSCRIBE.
> 
> You can't define the problem away by twisting commonly accepted
> terms in random directions. "Implicit Subscription" is the term
> that we, as a community, have historically used to mean "a
> subscription created by any means other than a SIP subscribe
> request, as described in RFC 3265." It's just shorter to say
> "implicit subscription."

My point was that the problem with "implicit" was that a lot of the 
information in a normal subscribe was not present, and a lot of the 
things you can do with a normal subscribe (like refresh) were not 
possible. Many of those problems go away when we have a mechanism like 
the app-info header and kpml, where we can get similar functionality.

I'd welcome your thoughts on my responses to the specific issues you 
raised.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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
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Adam Roach wrote:

> Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:
> 
>  > The duration is exactly equal to the length of the validity of the
>  > KPML script. That lifetime is defined in the framework. It terminates
>  > when the dialog terminates. Otherwise, the application can
>  > "unsubscribe" by pushing a new App-Info header w/o a KPML script for
>  > that instance (Tim had asked how this is done - see my answer in a
>  > reply to his note).
> 
> You missed my point. I'll try to make it more obvious.
> Below, I've placed an example NOTIFY message. Find the
> "XXXXX" in it, and change it so that it makes syntactic
> and semantic sense.

The situation is identical to the normal subscription (i.e., explicit 
with a SUBSCRIBE) case where you don't really want to refresh at all, 
since the resource has a natural termination tiem which is relatively 
short. In such a case, the SUBSCRIBE would basically have a very long 
subscription duration, as does the NOTIFY in response, and when the 
subscription termiantes a NOTIFY is sent with a state of termianted 
and a reason of noresource.

So, the same here. The KPML/App-Info imply no refreshes, so a long 
duration. Thus the NOTIFY has something long - say a week - as the 
duration. That would be XXXX=604800. When the dialog expires the UA 
would send a NOTIFY with xxxx=0 and a reason of noresource.

-Jonathan R.

> 
> NOTIFY sip:appserver.example.com SIP/2.0
> Via: SIP/2.0/TCP terminal.example.com;branch=z9hG4bK4EPlfSFQK1
> Max-Forwards: 70
> From: <sip:user@example.com>;tag=zpNctbZq
> To: <sip:appserver.example.com>;tag=ie4hbb8t
> Call-ID: cdB34qLToC@terminal.example.com
> CSeq: 997935769 NOTIFY
> Contact: <sip:terminal.example.com>
> Event: stimulus
> Subscription-State: active;expires=XXXXX
> Content-Type: application/kpml+xml
> Content-Length: 74
> 
> <?xml version="1.0">
> <kpml version="1.0">
>   <response digits="0113224321234"/>
> </kpml>
> 
> /a
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Adam Roach wrote:

> Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:
> 
>  > Correlation is easy. The KPML script IS the subscription. The
>  > place to  send the notify is implied from the URI in the KPML
>  > script. We would need to clarify in KPML that once that URI is
>  > used, subsequent notificatiosn to the same URI ought to go on
>  > the dialog established by the NOTIFY. But, this is a KPML issue.
> 
> I mean dialog correlation.
> 
> Let's say you're an app server. Client A sends an INVITE
> through you. You add something to the 200 response that
> ultimately results in Client A getting the following KPML
> script:
> 
> <?xml version="1.0">
> <kpml version="1.0">
>   <request>
>     <pattern>
>       <regex value="xx" href="sip:appserver.example.com" />
>     </pattern>
>   </request>
> </kpml>
> 
> Client A sends another INVITE through you. You add something to
> the 200 response that ultimately results in Client A getting
> the following KPML script:
> 
> <?xml version="1.0">
> <kpml version="1.0">
>   <request>
>     <pattern>
>       <regex value="xx" href="sip:appserver.example.com" />
>     </pattern>
>   </request>
> </kpml>
> 
> Finally, you get a NOTIFY request from the user. How do you
> know to which session it pertains?

You wouldnt send the KPML you have described. Each would have a 
different URI in the href. The URI provides the correlation.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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I agree with Dean that this is not at all what session timer was meant 
for. The spec is pretty explicit about its usage as a cleanup mechanism.

You'll have serious problems if the refreshing side is not trusted. 
Session timer allows either side to refresh. So, if this is a PC to 
phone case and PC is refreshing this won't work.

Even when its the gateway refreshing, 403 doesnt cause the gateway to 
terminate the call - at least it shouldnt, I dont think. Nothing in 
the spec says to tear down the call if you get a 403 to an INVITE.

-Jonathan R.

Rohan Mahy wrote:

> 
> On Wednesday, August 13, 2003, at 08:13 AM, Dean Willis wrote:
> 
>>
>> Rohan said:
>>
>>> I think it is completely reasonable to build a pre-paid calling system
>>> using a proxy with AAA and session timer and optionally a cooperating
>>> UA (for IVR features).
>>
>>
>> Just as billing from session-timer is not a use case, neither is prepaid
>> control using session timer appropriate. We designed session-timer to
>> have a resolution on the order of minutes or hours, not seconds.
> 
> 
> In many prepaid applications that is all that is required. For example:
> 
> UA sends an INVITE with apropos credentials.
> Proxy does a AAA lookup, finds out there is enough credit for 2 hours, 
> sets session timer at 15 minutes and proxies the request to a gateway 
> with Require: timer.
> After 30 minutes the gateway reINVITEs (a timer refresh request).  The 
> Proxy again does a AAA lookup.  At this point there is only enough 
> credit for 10 minutes (the caller used the prepaid service to make a 
> more expensive call). The proxy forwards the request with session timer 
> set to 5 minutes.
> On the next timer refresh, the proxy forwards the request with a 3 
> minute timer.  Finally the proxy uses a 2 minute timer.  When the last 
> timer refresh arrives, the Proxy sends a 403, and the gateway tears down 
> the session.
> 
> This mechanism works very well and is perfectly appropriate if either 
> only one session is allowed at a time, or if you are willing to reserve 
> resources for an existing session for a few minutes in the future.
> 
>> It's for cleaning up state when stuff goes wrong -- basically, a 
>> soft-crash
>> mechanism.
> 
> 
> I have no problem with that.  But it is a perfectly reasonable solution 
> for some prepaid applications (not all of them obviously), and requires 
> no changes to the session timer spec.
> 
> thanks,
> -rohan
> 
> 
>> One doesn't drive a car by manipulating its air-bags or
>> discard the brakes because one's seat belt is buckled.
>>
>> It sounds to me like you're introducing a time-bounded variant of the
>> authorization-token header. Something like that COULD be used by a
>> cooperating UA (say, the PSTN gateway) with reasonable success.
>>
>> -- 
>> 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
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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Jonathan Rosenberg wrote:
> 
> IMHO, the GRUU is a lot like the Reply-To URI. Its a URI that provides 
> an additional means of communication in the future, outside of the 
> current dialog. Just as the Reply-To URI is indicated with a header, so 
> to is the GRUU indicated with a header.

This is an interesting analogy, and perhaps a good one.

But Reply-To suffers from underspecification - AFAIK the circumstances 
under which a UA should extract the value from a Refer-To and insert 
them in some other request are entirely unspecified. As a result, the 
Refer-To header is currently pretty much unusable.

If we are to go with a separate header for GRUUs, then we need to do a 
much better job of defining when it should be used.

	Paul


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Date: Fri, 15 Aug 2003 13:07:50 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Thoughts inline.

Adam Roach wrote:

> After considering Jonathan's contention that defining
> a new method for conveying stimulus is creating something
> that is similar in meaning to NOTIFY, I'm inclined to
> agree to some point. There are still subtle differences
> in meaning, so I'm not completely convinced that a new
> method is wrong, but I understand the point he's trying
> to make.
> 
> After discussing it privately with others, I've concluded
> that there are three viable models worth considering. There
> are, of course, some minor variations that can be made to
> each.
> 
> 1) Standard HTTP Handling
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |   
> 2|                |-----INVITE---->|   
> 3|                |<-----200-------|   
> 4|<-----200-------|                |   AS adds HTTP URL to 200
> 5|---HTTP GET---->|                |   UAC requests KPML script
> 6|<-----200-------|                |   AS replies with script
> 7|---HTTP POST--->|                |   UAC provides digits
> 8|<-----200-------|                |   AS acknowledges digits
> 
>   Of the three options presented here, I see this as the best
>   solution. Basically, the app server includes an HTTP URL
>   in the 200 response to the INVITE that it has proxied. The
>   UAC then does a GET on the KPML script. Just like with
>   HTML forms, the UAC does whatever local stuff is necessary
>   to "fill out the form" (in this case, collect digits), and
>   then posts the result back to the App Server. This is all
>   plain-Jane unextended HTTP as it was intended to work.

This is also my favorite, and always has been. However, I had believed 
I was the only one who liked this.

Cullen raised some concerns about it:

> One of the difficult parts of this is security.
> 
> This will get more complicated once you provide a way to send in message 4
> the credentials that will allow 7.

This is only a problem when there is no direct trust relationship 
already between the caller and the application server. If there is 
such a trust relationship, then you can use digest with the same 
credentials used to authenticate the INVITE. If there is not, it is 
more complex.

The case when there is not (which we discussed in the design team 
meetings) was when, for example, I call Cullen. This lands in a Cisco 
proxy, which runs an app that asks me to choose which "fluffy" I want, 
as there are miraculously two at Cisco (probably not true but bear 
with me). Now, the Cisco proxy doesn't know me at all. But, it wants 
to make sure that the entity which does the POST is the one that made 
the call.

I would argue that there is an easy solution to this, which is just 
sips + https. If the call is over sips, the proxy can send the UA a 
random URI in the APp-Info header. No one else can see this header but 
the caller and intervening proxies. Then, the caller does https to the 
proxy, and sends the URI it was handed. I believe this provides a 
reasonably secure correlation.

Note that sips is needed in all cases for any security for app-info 
headers.



> It may also make the App-Info header only
> really work with scripts that use HTTP to post instead of other things.

That would be the proposal here, since its HTTP with the other script 
types (HTML, VoiceXML).

> Also, in the case of a financial institutions that have to log electronic
> communications, you will need to provide way to insert logging HTTP proxies
> and probably proxies for FW traversal too.

All of these exist, no? The only real argument for HTTP is that this 
basically IS a web application of sorts, just one on a really dumb device.


> Of course the AS or the UAC will
> need to figure out the network topology to know if the AS is inside or
> outside the FW. 

I don't follow that.

> It's possible to configure all the domain and HTTP proxies
> rules into the device but it still hard to secure - particularly on devices
> that don't want to implement HTTPS due to not much memory.

Well - thats the rub - if https isnt possible, yes, its a problem. But 
then you can't argue that sips would be OK, in which case you have no 
security anyway.



> 
> 2) Explicit Subscriptions (with GRUU)
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |   
> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
> 3|                |<-----200-------|   
> 4|<-----200-------|                |   
> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> 6|------200------>|                |   
> 7|-----NOTIFY---->|                |   
> 8|<-----200-------|                |   
> 9|-----NOTIFY---->|                |   UAC provides digits
> A|<-----200-------|                |   AS acknowledges digits
> 
>   My second choice would be an explicit subscription to stimulus
>   events. The only sticky part in this solution is the question
>   of how the AS can route a SUBSCRIBE to the UAC. Once we have
>   a GRUU solution in place, this is a no brainer: the AS can
>   extract the GRUU from the INVITE, and use it as the target
>   of its SUBSCRIBE. The KPML script is included in the SUBSCRIBE,
>   and essentially serves as a filter.

This was one of the solutions I outlined in my initial email, yes. 
GRUU is not the only problem. There is also the correlation problem - 
how does the UA know that the SUBSCRIBE comes from an entity along the 
call path? AUthorizing the subscribe as above would be really hard.

> 
> 3) Explicit Subscriptions (with REFER)
> 
> UAC           App Server          UAS
> 1|-----INVITE---->|                |   
> 2|                |-----INVITE---->|   
> 3|                |<-----200-------|   
> 4|<-----200-------|                |   AS adds SIP URL to 200
> 5|-----REFER----->|                |   
> 6|<-----200-------|                |   
> 7|<----NOTIFY-----|                |   
> 8|------481------>|                |   
> 9|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
> A|------200------>|                |   
> B|-----NOTIFY---->|                |   
> C|<-----200-------|                |   
> D|-----NOTIFY---->|                |   UAC provides digits
> E|<-----200-------|                |   AS acknowledges digits
> 
>   Although somewhat chattier than the previous solution, this gets around
>   the GRUU issue by having the AS provide a SIP URI that routes to itself
>   in the 200 response.

How does that solve the GRUU problem? It still needs to get a GRUU to 
hand out in the Refer-To. This just adds another layer of indirection.

Adam later wrote:
> Your phrasing above knocked loose in my mind another reason
> that implicit subscriptions[1] are ill-advised: you are using
> a single transaction to perform two operations. Conveying
> partial success (e.g. the call completed, but the subscription
> failed) is problematic. We have made efforts to avoid this
> type of situation in the past. 

What two operations are being performed by which transaction? The NOTIFY?

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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] Re: Comment on GRUU requirements
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On Friday, August 15, 2003, at 05:53 AM, Jonathan Rosenberg wrote:

> inline.
>
> Rohan Mahy wrote:
>
>> On Wednesday, August 13, 2003, at 10:55 AM, Adam Roach wrote:
>>> Rohan Mahy [mailto:rohan@cisco.com] writes:
>>>
>>>> Most of the SIP phones I see at SIPit support
>>>> completing a transfer using a REFER with a
>>>> Replaces header embedded in the Refer-To.
>>>> The Refer-To URI is always set to the Contact
>>>> URI of the appropriate UA.
>>>
>>>
>>> And this will continue to work as well as it does today.
>>> Newer, GRUU aware clients will simply exhibit an additional
>>> robustness.
>>>
>>> It seems a bit misguided to do things in a suboptimal way
>>> just to try to fix currently deployed clients.
>> and Jonathan wrote:
>>>>> I don't think we have so much of either of these deployed that 
>>>>> this cannot change.
>>>>
>>>> I disagree.  Most of the SIP phones I see at SIPit support 
>>>> completing a transfer using a REFER with a Replaces header embedded 
>>>> in the Refer-To.  The Refer-To URI is always set to the Contact URI 
>>>> of the appropriate UA.
>>>
>>>
>>> Well, we have here the standard problem here of having 
>>> implementations of long-running drafts, and then deciding down the 
>>> road that something needs to change.
>>>
>>> I think we want a non-GRUU in the Contact header (to avoid spirals 
>>> through proxies, and needless load when the proxy doesnt intend to 
>>> record-route), but we want a GRUU for attended-transfer. I would 
>>> propose to use a separate header to identify GRUUs. Existing phones 
>>> will work as they currently do, and when two phones both support 
>>> GRUU, you get more reliable service. I recognize that putting the 
>>> GRUU in the contact would mean only one of the phones needs to 
>>> support it (clearly better), but the cost of this is a long-term 
>>> situation that we don't want.
>>>
>>> -Jonathan R.
>> Hi,
>> I think both of you are ignoring the "generality before efficiency" 
>> principle.
>
> I don't think that is the tradeoff here. Given a clean slate, it is 
> clear that a new header would be used. The argument for contact header 
> is just to facilitate introduction of GRUU. Thus, its more of the 
> "backwards compatible vs. right thing" kind of debate.
>
>> Putting the GRUU in the Contact header always works, and in most 
>> deployments where it is needed (middlebox traversal), you WILL need 
>> to travel through the same set of proxies for nearly every request 
>> anyhow. With the increased deployment of TLS and SIPS we will 
>> increasingly have dialogs for which all or none of the requests are 
>> proxied.
>
> A few responses here. Firstly, your comments above would seem to imply 
> that a UA wouldn't always put the GRUU in the Contact header - that it 
> would do so only in situations where it was needed. The problem is, 
> the UA can't know when its needed. It would be needed if someone else 
> wants to transfer a call to it, or subscribe to its dialog package. 
> Since the UA can't know that ahead of time, for it to work, we need to 
> specify that all UAs should always use GRUUs all the time. If the GRUU 
> is placed in the Contact header, this would mean the end of e2e sip 
> signaling, PERIOD.

AFAIK, there are two cases only.  You either a) can self-generate GRUUs 
(no server leasing) in which case you use these as your Contact 
addresses, or b) you need to lease GRUUs and put them somewhere.  
Clearly, e2e signaling still works just fine in case a).   My position 
is that e2e signaling is only rarely used in practice in case b), and 
that with TLS/SIPS that this will become even less of an option.

IMO this is just modeling the real partition that is occurring among IP 
hosts.  There are full-fledged Internet hosts which can make use of the 
e2e principle (case a) and all the other IP hosts which are stuck 
behind middleboxes with "fogged up windows" (case b).  I believe my 
comments merely reflect that (unfortunate) reality.

> Even if you assume a proxy needs to be on the path, a proxy then is 
> faced with an impossible decision. When it receives a request, does 
> the proxy need to record route (if the downstream UA is using a GRUU) 
> or not (if it isn't). It doesn't know what the downstream UA can do at 
> this point. So, it can't make a correct decision. Existing proxies 
> wouldn't even know about this decision. The result would be that, 
> where proxies record-route and the downstream UA uses a GRUU, requests 
> will spiral or take extra hops on every call. This results in 
> decreased overall performance in the network and increased call setup 
> times.

In practice, this is not very complicated.  If the proxy needs to 
record route for application purposes then it will record route 
according to that policy (no change).  If the proxy is configured to 
record route for middlebox traversal purposes then it is trivial for it 
to check if the request-URI "host is me" before adding record route.  
This has always been a good idea, we just have an additional motivation 
for that check now.

>> We are talking about interop with thousands of UAs which have 
>> implemented Proposed Standard level RFCs (SIP and Refer) and an 
>> long-lived, stable I-D that has completed WGLC and was submitted to 
>> the IESG (Replaces)`.  We are not talking about random drafts that 
>> very few have implemented.
>
> These drafts are standardized, yes, but their usage for assisted call 
> transfer (which is what we are talking about here) has clearly been 
> changing and is still an I-D. Nothing in the specs you mention tells 
> people to extract the Refer-To URI from the Contact header of the > peer.

This usage has not significantly changed in almost the last two years. 
(The caller preferences solution was abandoned almost immediately 
because it had much worse backward compatibility problems.) If you want 
any sort of useful distributed call control (not just transfer, but all 
the services you could build conferencing, replaces, join, and the 
dialog package) you need to have a URI that reaches the same SIP UA for 
all dialogs which are related to the target dialog. If you use the 
rigid definition of Contact as only valid in a dialog, then the Contact 
header is effectively useless (it can't stand alone without the route 
set) except where the Contact URI happens to be a GRUU of the style in 
case a).  In fact we already have a precedent (in the REGISTER method) 
that a Contact should be valid for more than one dialog.  So why don't 
we formalize this convention and state that what you place in the 
Contact header for a dialog MUST be a GRUU and how you get one is up to 
you.

>> To be convinced that it is worth adding a new GRUU header, I would at 
>> least like to see some plausible use cases that are common enough for 
>> us to be worrying about.  Maybe the right way to look at this is to 
>> put the GRUU in the Contact, UNLESS you can tell that your peer 
>> understands the special significance of GRUUs.
>
> Hopefully my explanation above describes the problems with putting 
> GRUU in contacts. Even if you knew that all of your peers supported 
> Contact headers, putting one in would have the problems described 
> above.

again, you haven't described a "problem" that is any different from 
best current practice. most implementations *already* use the URI from 
the Contact in the Refer-To header (and in other places outside of a 
single dialog).

> IMHO, the GRUU is a lot like the Reply-To URI. Its a URI that provides 
> an additional means of communication in the future, outside of the 
> current dialog. Just as the Reply-To URI is indicated with a header, 
> so to is the GRUU indicated with a header.

Then what is the use of having a Contact header?  By redefining the 
Contact header clearly we do not cause any harm.  Implementations which 
do not put GRUUs in Contacts will be just as broken as they are today 
but no more so, and a bunch of things will just work nicely that didn't 
before.

>> Also, where this extension is most needed, you need the right 2 out 
>> of 3 to support the extension.  (This isn't just any two of three.  
>> If Bob supports the GRUU extension, but Carol does not, Alice still 
>> cannot cause a transfer to occur in the appropriate direction.)
>> Starting with the classic Alice, Bob, and Carol actors.
>> Alice and Bob have an existing dialog.
>> Alice and Carol have an existing dialog.
>> Alice wants Bob to talk to Carol, and will REFER Bob to INVITE Carol 
>> replacing the dialog between Alice and Carol.
>> Carol needs a GRUU; if the GRUU is conveyed in a separate header 
>> Alice and Carol both need to support the GRUU extension for the 
>> Refer-To header to be set properly.
>
> Yes, I agree with this. I agree its not optimal. With what I am 
> proposing, things will work as well as they do today, no worse, no 
> better, if Alice and Bob don't support GRUUs.

Right, and with what I am proposing they will sometimes work even 
better than that.

thanks,
-rohan

>
> -Jonathan R.
>
>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> 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] Re: Update to 3pcc
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On Friday, August 15, 2003, at 08:50 AM, Jonathan Rosenberg wrote:

> I agree with Dean that this is not at all what session timer was meant 
> for. The spec is pretty explicit about its usage as a cleanup 
> mechanism.

if it works without changes to the existing mechanism, then what is the 
problem?

> You'll have serious problems if the refreshing side is not trusted. 
> Session timer allows either side to refresh. So, if this is a PC to 
> phone case and PC is refreshing this won't work.

the gateway is the refresher, and is trusted.  the example i gave 
nicely fits a business model where the provider accounts for PSTN 
usage, but does not charge/account for arbitrary SIP sessions.   Easy 
things should be easy, contrived things will be hard and will require a 
B2BUA.

> Even when its the gateway refreshing, 403 doesnt cause the gateway to 
> terminate the call - at least it shouldnt, I dont think. Nothing in 
> the spec says to tear down the call if you get a 403 to an INVITE.

you are correct, but the call will be torn down after the session 
expires.  since the session is not refreshed when the proxy responds 
with a 403, the session will expire and the gateway will send a BYE.

thanks,
-rohan

>
> -Jonathan R.
>
> Rohan Mahy wrote:
>
>> On Wednesday, August 13, 2003, at 08:13 AM, Dean Willis wrote:
>>>
>>> Rohan said:
>>>
>>>> I think it is completely reasonable to build a pre-paid calling 
>>>> system
>>>> using a proxy with AAA and session timer and optionally a 
>>>> cooperating
>>>> UA (for IVR features).
>>>
>>>
>>> Just as billing from session-timer is not a use case, neither is 
>>> prepaid
>>> control using session timer appropriate. We designed session-timer to
>>> have a resolution on the order of minutes or hours, not seconds.
>> In many prepaid applications that is all that is required. For 
>> example:
>> UA sends an INVITE with apropos credentials.
>> Proxy does a AAA lookup, finds out there is enough credit for 2 
>> hours, sets session timer at 15 minutes and proxies the request to a 
>> gateway with Require: timer.
>> After 30 minutes the gateway reINVITEs (a timer refresh request).  
>> The Proxy again does a AAA lookup.  At this point there is only 
>> enough credit for 10 minutes (the caller used the prepaid service to 
>> make a more expensive call). The proxy forwards the request with 
>> session timer set to 5 minutes.
>> On the next timer refresh, the proxy forwards the request with a 3 
>> minute timer.  Finally the proxy uses a 2 minute timer.  When the 
>> last timer refresh arrives, the Proxy sends a 403, and the gateway 
>> tears down the session.
>> This mechanism works very well and is perfectly appropriate if either 
>> only one session is allowed at a time, or if you are willing to 
>> reserve resources for an existing session for a few minutes in the 
>> future.
>>> It's for cleaning up state when stuff goes wrong -- basically, a 
>>> soft-crash
>>> mechanism.
>> I have no problem with that.  But it is a perfectly reasonable 
>> solution for some prepaid applications (not all of them obviously), 
>> and requires no changes to the session timer spec.
>> thanks,
>> -rohan
>>> One doesn't drive a car by manipulating its air-bags or
>>> discard the brakes because one's seat belt is buckled.
>>>
>>> It sounds to me like you're introducing a time-bounded variant of the
>>> authorization-token header. Something like that COULD be used by a
>>> cooperating UA (say, the PSTN gateway) with reasonable success.
>>>
>>> -- 
>>> 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
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> 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



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Subject: Re: [Sipping] Request for editorial input
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Hi Robert,

The issue that you raise is also a problem for SMIME examples.  We need  
a way to display binary non-ASCII data.  We can display this using  
textual conventions like you suggest for short sections, but for longer  
sections of non-text formatted data, perhaps we need to use a hexdump  
format like the compression folks use (see the SIP static compression  
dictionary).

thanks,
-rohan

On Thursday, August 7, 2003, at 11:41 AM, Robert Sparks wrote:

> As noted at IETF57, I'm picking up editorship of the torture-test
> document.
>
> One of the things I want to do is disambiguate the printed text
> examples. There are cases now where the torture consists of having very
> long text fields. For example:
>
> CSeq:
> 239sdfadfnm34r5u09sdfawetrq3o4t5nodifnaoihesfoiw34foisdfeijjnaosdfj4ios 
> dfnaehnaasoo34ijdfn5kl3neasdfvnazs
>
> which gets broken into a format that is not valid for SIP when you try
> to render withing the 72 column limit of internet drafts. (Its probably
> also broken badly in this message by many mail readers).
>
> I propose making such breaks explicit as follows:
>
> If I needed to break the following (short for clarity) header field:
>
> Foo: areallybigtoken
>
> I'll render it as
>
> Foo: areally<break>
>      </break>bigtoken
>
> or
>
> Foo: areally<break>
>      </break>big<break>
>      </break>token
>
>
> Objections? Alternative suggestions?
>
> Additionally, I'm considering including an appendix with an
> ascii-armored container (a zip-file or compressed tarball
> perhaps) with each message (so that we can include some
> torturous whitespace and non-ASCII range UTF8 tests.
>
> RjS
>
>
>
>
> _______________________________________________
> 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: Fri, 15 Aug 2003 14:18:08 -0700
Subject: Re: [Sipping] DTMF: Why Not A New Method?
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>
CC: sipping <sipping@ietf.org>
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I think the href in the VXML would provide the correlation information so it
might be something like

Href="sip:application_23_user_55_instance_2345@appserver.example.com"

Cullen


On 8/14/03 4:19 PM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:
> 
>> Correlation is easy. The KPML script IS the subscription. The
>> place to  send the notify is implied from the URI in the KPML
>> script. We would need to clarify in KPML that once that URI is
>> used, subsequent notificatiosn to the same URI ought to go on
>> the dialog established by the NOTIFY. But, this is a KPML issue.
> 
> I mean dialog correlation.
> 
> Let's say you're an app server. Client A sends an INVITE
> through you. You add something to the 200 response that
> ultimately results in Client A getting the following KPML
> script:
> 
> <?xml version="1.0">
> <kpml version="1.0">
> <request>
>   <pattern>
>     <regex value="xx" href="sip:appserver.example.com" />
>   </pattern>
> </request>
> </kpml>
> 
> Client A sends another INVITE through you. You add something to
> the 200 response that ultimately results in Client A getting
> the following KPML script:
> 
> <?xml version="1.0">
> <kpml version="1.0">
> <request>
>   <pattern>
>     <regex value="xx" href="sip:appserver.example.com" />
>   </pattern>
> </request>
> </kpml>
> 
> Finally, you get a NOTIFY request from the user. How do you
> know to which session it pertains?
> 
> /a
> 


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Date: Fri, 15 Aug 2003 14:18:56 -0700
Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On 8/14/03 10:33 AM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Cullen Jennings writes:
> 
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |
>> 5|---SIP MESG---->|                |
>> 6|<-----200-------|                |
> 
> Are you proposing that the App Server would send the KPML script
> in the INVITE 200? Where? In the body? In an egregiously large
> header?
> 
> 

The App-Info allows the script to be fetched with HTTP or be in a body and
use a cid URL to get it out of the body. I probably should have included the
HTTP get between message 4 and 5 to be fair.

Cullen


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Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, sipping <sipping@ietf.org>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Fair enough, I am not really arguing for NOTIFY, I sort of like the
NEW_METHOD approach because it lets us get past all the semantics arguments
by defining the semantics to be exactly what we want.

I forgot on my list, PUBLISH may be an option to use to depending on what
people believe the semantics of it are.

On the topic of reporting that there was some error in dealing with the
script referenced in the App-Info header, we use the (cough) web model of
error reporting, which is to say we don't do it at all. Thought this seems
lame at first glance, after much discussion, we came to the conclusion that
any useful application you wanted to build with this needed to deal with
never receiving any stimulus from the script and that web model was quite
reasonable and the right thing for this. I think this decision makes the
problem of multiple error report on implicit subscribes a non issue here.

Cullen


On 8/14/03 4:11 PM, "Adam Roach" <adam@dynamicsoft.com> wrote:

> Cullen Jennings [mailto:fluffy@cisco.com] writes:
> 
>> Proposal 3) Use NOTIFY. Semantics are, here is some
>> information about some event that you requested to
>> be informed about. The usual complaint on this one
>> is that there was no Subscribe, or it was implicit,
>> or it was explicit, or it violated some law of the
>> SIP world. I think there was a subscribe, it was in
>> the message that initiated the script. There is also
>> a clear time it will end - when the script is terminated
>> or when the dialog of the message that initiated the
>> script ends. We all remember that implicit subscribes
>> where bad (whatever an implicit subscribe is) but people
>> seem to have a hard time remembering why this will be
>> bad here.
> 
> It's not simple dogma; there are real problems here. I've
> cited a few of them (and, as I point out in another message,
> Jonathan's rebuttal is somewhat lacking in the area of
> duration; also, I still plan to respond on the correlation
> issue, because I think his rebuttal glosses over the
> details).
> 
> Can these problems be fixed? Probably. REFER provides an
> existence proof that implicit subscriptions[1] can be made
> to work with enough crafting. After it's said and done,
> do we regret choosing that path? I do. Robert does. Do we
> want to learn from our mistakes?
> 
> Your phrasing above knocked loose in my mind another reason
> that implicit subscriptions[1] are ill-advised: you are using
> a single transaction to perform two operations. Conveying
> partial success (e.g. the call completed, but the subscription
> failed) is problematic. We have made efforts to avoid this
> type of situation in the past.
> 
> /a
> 
> [1] To be clear, I mean "subscriptions put in place via some
>   other mechanism than that described in RFC 3265."
> 


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Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Adam Roach <adam@dynamicsoft.com>
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Let me propose a realistic scenario that is a little more difficult with
http. It is probably possible to solve, just complicated. I will admit that
much of the reason many people don't want to use HTTP is because they think
sending it in SIP will be less work - they may or may not be right but the
"less work" on low end devices is what has driven us to a fairly messy
situation in handling DTMF today. I want to come up this something that is
appealing enough to the people doing evil things today that they stop doing
evil things. That said, let me describe a scenario that does occur and is
difficult with HTTP.

A residential phone user, called A, has a UA that they can use to make PSTN
calls. This UA connects to a voice service provider, called Provider B that
claims to terminate the calls to the PSTN. This provider does not directly
terminate them but actually pass them to provider C. Provider C runs the
call through a H.323 translator and terminates then sends the call to a 323
GW. The 323 GW does not support RFC 2833. The user A UA is likely to be
behind a NAT and the final GW might be behind a NAT depending on the country
of termination. B does not wish A to learn the identify of C because A might
get a better price and cut B out of the loop. C does not wish B to learn too
much about their network topology, since C might actually be passing traffic
to provider D. Provider B and C and both run transparent B2BUA, broken
proxies, and firewalls to make this all possible. The 323 translator wants
to receive DTMF in the signaling so that it can pass it to the GW. In a
variant of the scenario, provider B is a prepaid service and wants to be
able to intercept long #.

If the phone sent DTMF over INFO, the people with this problem would be
happy. The development effort to send DTMF over INFO is very low. It would
be possible to solve this example with 2833 to the 323 translator. The IETF
suggested this was the right way to go - I am unaware of anyone who did
that. Everyone decided some weird variant of sending it over SIP looked like
a much better idea even though it was not interoperable. Perhaps they were
all insane but people buying equipment convinced vendors to build it. If we
want to standardize something that is going to be widely implemented, I
think we need to keep what is deployed today and why it happened in mind.

It is probably possible to have provider B write a HTTP proxy like thing
that takes posts from A and then relays them on to C and change it's call
control software to rewrite the KPML scripts as they pass through the
signaling proxy/b2bua. But this leads to the the NAT ALG like behavior that
we are trying to avoid.

Cullen


Another scenario similar to this happens when the PSTN GW is a SIP GW that
may support 2833 but it is doing overlap dialing and uses KPML to tell the
UA to keep sending it more digits before media is set up. I don't claim to
fully understand this one so don't ask me too many questions on it :-)




On 8/15/03 10:07 AM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:

> Thoughts inline.
> 
> Adam Roach wrote:
> 
>> 1) Standard HTTP Handling
>> 
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |   AS adds HTTP URL to 200
>> 5|---HTTP GET---->|                |   UAC requests KPML script
>> 6|<-----200-------|                |   AS replies with script
>> 7|---HTTP POST--->|                |   UAC provides digits
>> 8|<-----200-------|                |   AS acknowledges digits
>> 
>>   Of the three options presented here, I see this as the best
>>   solution. Basically, the app server includes an HTTP URL
>>   in the 200 response to the INVITE that it has proxied. The
>>   UAC then does a GET on the KPML script. Just like with
>>   HTML forms, the UAC does whatever local stuff is necessary
>>   to "fill out the form" (in this case, collect digits), and
>>   then posts the result back to the App Server. This is all
>>   plain-Jane unextended HTTP as it was intended to work.
> 
> This is also my favorite, and always has been. However, I had believed
> I was the only one who liked this.
> 
> Cullen raised some concerns about it:
>


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Date: Fri, 15 Aug 2003 15:49:36 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sipping] Re: Comment on GRUU requirements
To: Rohan Mahy <rohan@cisco.com>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Alan Johnston'" <alan.johnston@mci.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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I have to agree with Rohan. I don't understand the
purpose of NOT placing the GRUU in the Contact: header
if possible. Whether or not this GRUU ties into a
proxy is up for debate, but in practice I believe it
will be the rule rather than the exception. Similar
(but not identical) to the fact that a registrar is
not absolutely necessary to use SIP but it sure makes
things easier in practice.


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From: Adam Roach <adam@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: sipping <sipping@ietf.org>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Fri, 15 Aug 2003 18:58:25 -0500
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Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:

> > 2) Explicit Subscriptions (with GRUU)
>
> This was one of the solutions I outlined in my initial email, yes. 
> GRUU is not the only problem. There is also the correlation problem - 
> how does the UA know that the SUBSCRIBE comes from an entity 
> along the 
> call path? AUthorizing the subscribe as above would be really hard.

The SUBSCRIBE itself will need to contain information about the
dialog for which it is requesting information anyway, right?
Getting this information requires you to be in a position that
would allow you to intercept the digits themselves.

If you want to keep this information even more securely under
wraps, use sips: to set up the call; the dialog information
is revealed only to people in the call path, and only they would
be able to subscribe to keypad events.

/a

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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: Adam Roach <adam@dynamicsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        Adam Roach <adam@dynamicsoft.com>
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fluffy writes:

> Another scenario similar to this happens when the PSTN GW is 
> a SIP GW that may support 2833 but it is doing overlap dialing
> and uses KPML to tell the UA to keep sending it more digits
> before media is set up.

*cringe*

That reminds me: the overlap dialing specification should be
popping out as an RFC any moment now. We just finished auth48.

/a

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To: "'Cullen Jennings'" <fluffy@cisco.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>,
        "'Adam Roach'" <adam@dynamicsoft.com>
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Cullen Jennings wrote:
> If the phone sent DTMF over INFO, the people with this 
> problem would be happy. The development effort to send DTMF 
> over INFO is very low. It would be possible to solve this 
> example with 2833 to the 323 translator. The IETF suggested 
> this was the right way to go - I am unaware of anyone who did 
> that. Everyone decided some weird variant of sending it over 
> SIP looked like a much better idea even though it was not 
> interoperable. Perhaps they were all insane but people buying 
> equipment convinced vendors to build it. If we want to 
> standardize something that is going to be widely implemented, 
> I think we need to keep what is deployed today and why it 
> happened in mind.

The RFC2833 to H.323 translator idea is hightly dubious to me
and I really don't see where that translater would sit to be able
to perform this task. It would seem to have to sit at the media 
termination, which would imply that the endpoint supports RFC2833
in the first place.

Certainly, support for RFC2833 is included in H.323 starting with 
version 4. So it would probably be much easier to support that than
the translator idea.

> Another scenario similar to this happens when the PSTN GW is 
> a SIP GW that may support 2833 but it is doing overlap 
> dialing and uses KPML to tell the UA to keep sending it more 
> digits before media is set up. I don't claim to fully 
> understand this one so don't ask me too many questions on it :-)

This is kind of a weird scenario. Wouldn't you just do the digit collection
at the UA, then make the call (en-bloc) to the PSTN gateway? And then it's
up
to the PSTN GW to do overlap sending if it pleases.

If the UA can't do the digit collection and insists on doing overlap
(because
it is itself a PSTN gateway for example), then the normal overlap sending 
procedures should apply (with the sequential INVITEs).

I don't think we should use KPLM for dialling.

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<P><FONT SIZE=2>Cullen Jennings wrote:</FONT>
<BR><FONT SIZE=2>&gt; If the phone sent DTMF over INFO, the people with this </FONT>
<BR><FONT SIZE=2>&gt; problem would be happy. The development effort to send DTMF </FONT>
<BR><FONT SIZE=2>&gt; over INFO is very low. It would be possible to solve this </FONT>
<BR><FONT SIZE=2>&gt; example with 2833 to the 323 translator. The IETF suggested </FONT>
<BR><FONT SIZE=2>&gt; this was the right way to go - I am unaware of anyone who did </FONT>
<BR><FONT SIZE=2>&gt; that. Everyone decided some weird variant of sending it over </FONT>
<BR><FONT SIZE=2>&gt; SIP looked like a much better idea even though it was not </FONT>
<BR><FONT SIZE=2>&gt; interoperable. Perhaps they were all insane but people buying </FONT>
<BR><FONT SIZE=2>&gt; equipment convinced vendors to build it. If we want to </FONT>
<BR><FONT SIZE=2>&gt; standardize something that is going to be widely implemented, </FONT>
<BR><FONT SIZE=2>&gt; I think we need to keep what is deployed today and why it </FONT>
<BR><FONT SIZE=2>&gt; happened in mind.</FONT>
</P>

<P><FONT SIZE=2>The RFC2833 to H.323 translator idea is hightly dubious to me</FONT>
<BR><FONT SIZE=2>and I really don't see where that translater would sit to be able</FONT>
<BR><FONT SIZE=2>to perform this task. It would seem to have to sit at the media </FONT>
<BR><FONT SIZE=2>termination, which would imply that the endpoint supports RFC2833</FONT>
<BR><FONT SIZE=2>in the first place.</FONT>
</P>

<P><FONT SIZE=2>Certainly, support for RFC2833 is included in H.323 starting with </FONT>
<BR><FONT SIZE=2>version 4. So it would probably be much easier to support that than</FONT>
<BR><FONT SIZE=2>the translator idea.</FONT>
</P>

<P><FONT SIZE=2>&gt; Another scenario similar to this happens when the PSTN GW is </FONT>
<BR><FONT SIZE=2>&gt; a SIP GW that may support 2833 but it is doing overlap </FONT>
<BR><FONT SIZE=2>&gt; dialing and uses KPML to tell the UA to keep sending it more </FONT>
<BR><FONT SIZE=2>&gt; digits before media is set up. I don't claim to fully </FONT>
<BR><FONT SIZE=2>&gt; understand this one so don't ask me too many questions on it :-)</FONT>
</P>

<P><FONT SIZE=2>This is kind of a weird scenario. Wouldn't you just do the digit collection</FONT>
<BR><FONT SIZE=2>at the UA, then make the call (en-bloc) to the PSTN gateway? And then it's up</FONT>
<BR><FONT SIZE=2>to the PSTN GW to do overlap sending if it pleases.</FONT>
</P>

<P><FONT SIZE=2>If the UA can't do the digit collection and insists on doing overlap (because</FONT>
<BR><FONT SIZE=2>it is itself a PSTN gateway for example), then the normal overlap sending </FONT>
<BR><FONT SIZE=2>procedures should apply (with the sequential INVITEs).</FONT>
</P>

<P><FONT SIZE=2>I don't think we should use KPLM for dialling.</FONT>
</P>

</BODY>
</HTML>
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Sean Olson'" <seancolson@yahoo.com>, Rohan Mahy <rohan@cisco.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Alan Johnston'"
	 <alan.johnston@mci.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] Re: Comment on GRUU requirements
Date: Mon, 18 Aug 2003 14:02:50 +0200
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Hi,

I don't know if this has been discussed, but how about putting the GRUU in the Contact header, AND define a "gruu" header/uri parameter. Now, existing nodes will, as Rohan put it, be as broken as they are today, but they will still work, and nodes supporting GRUU will use the gruu parameter to indicate/detect that the uri in the Contact header is a GRUU. But, no new header needs to be defined (I think we already have too many headers carrying URIs, many of which we don't use - or even don't know how or when to use), and there would be no backward compability issues, and no need to redefine existing headers/functionality.

Regards,

Christer Holmberg
Ericsson Finland

> -----Original Message-----
> From: Sean Olson [mailto:seancolson@yahoo.com]
> Sent: 16. elokuuta 2003 1:50
> To: Rohan Mahy; Jonathan Rosenberg
> Cc: Adam Roach; 'Alan Johnston'; 'sipping@ietf.org'
> Subject: Re: [Sipping] Re: Comment on GRUU requirements
> 
> 
> I have to agree with Rohan. I don't understand the
> purpose of NOT placing the GRUU in the Contact: header
> if possible. Whether or not this GRUU ties into a
> proxy is up for debate, but in practice I believe it
> will be the rule rather than the exception. Similar
> (but not identical) to the fact that a registrar is
> not absolutely necessary to use SIP but it sure makes
> things easier in practice.
> 
> 
> __________________________________
> Do you Yahoo!?
> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.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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Date: Mon, 18 Aug 2003 08:50:06 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: RE: [Sipping] Re: Comment on GRUU requirements
To: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>,
        Rohan Mahy <rohan@cisco.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
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I believe this was Adam's original suggestion. Works
for me.

--- "Christer Holmberg (JO/LMF)"
<christer.holmberg@ericsson.com> wrote:
> 
> Hi,
> 
> I don't know if this has been discussed, but how
> about putting the GRUU in the Contact header, AND
> define a "gruu" header/uri parameter. Now, existing
> nodes will, as Rohan put it, be as broken as they
> are today, but they will still work, and nodes
> supporting GRUU will use the gruu parameter to
> indicate/detect that the uri in the Contact header
> is a GRUU. But, no new header needs to be defined (I
> think we already have too many headers carrying
> URIs, many of which we don't use - or even don't
> know how or when to use), and there would be no
> backward compability issues, and no need to redefine
> existing headers/functionality.
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> > -----Original Message-----
> > From: Sean Olson [mailto:seancolson@yahoo.com]
> > Sent: 16. elokuuta 2003 1:50
> > To: Rohan Mahy; Jonathan Rosenberg
> > Cc: Adam Roach; 'Alan Johnston';
> 'sipping@ietf.org'
> > Subject: Re: [Sipping] Re: Comment on GRUU
> requirements
> > 
> > 
> > I have to agree with Rohan. I don't understand the
> > purpose of NOT placing the GRUU in the Contact:
> header
> > if possible. Whether or not this GRUU ties into a
> > proxy is up for debate, but in practice I believe
> it
> > will be the rule rather than the exception.
> Similar
> > (but not identical) to the fact that a registrar
> is
> > not absolutely necessary to use SIP but it sure
> makes
> > things easier in practice.
> > 
> > 
> > __________________________________
> > Do you Yahoo!?
> > Yahoo! SiteBuilder - Free, easy-to-use web site
> design software
> > http://sitebuilder.yahoo.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 exim@www1.ietf.org  Mon Aug 18 13:24:33 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Rohan Mahy'" <rohan@cisco.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: Adam Roach <adam@dynamicsoft.com>,
        "'Alan Johnston'"
	 <alan.johnston@mci.com>,
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Subject: RE: [Sipping] Re: Comment on GRUU requirements
Date: Mon, 18 Aug 2003 12:22:27 -0500
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Rohan Mahy [mailto:rohan@cisco.com] writes:

> In fact we already have a precedent (in the REGISTER method) 
> that a Contact should be valid for more than one dialog.

This is one of those perfect-hindsight problems, but
the issue here is that "Contact" in REGISTER is wildly
different in meaning than "Contact" in any other context
(except, arguably, 300-class responses).

These uses really should have employed a different header
than "Contact". Citing them as precedent is shaky
at best.

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: RE: [Sipping] Re: Comment on GRUU requirements
Date: Mon, 18 Aug 2003 13:44:42 -0500
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Yes, this was my original proposal; however, I
appreciate the concern that this approach requires
all requests -- even those in a dialog -- to go
through a GRUU server.

After considerable thought, I believe that this
approach coupled with Alan's suggestion that
non-firewalled clients can generate their own
GRUUs seems to allay those concerns.

So, as long as "colocating a GRUU server with a UAS"
is explicitly supported, I'm happy to see this as
a solution.

/a

> -----Original Message-----
> From: Sean Olson [mailto:seancolson@yahoo.com]
> Sent: Monday, August 18, 2003 10:50
> To: Christer Holmberg (JO/LMF); Rohan Mahy; Jonathan Rosenberg
> Cc: Adam Roach; 'Alan Johnston'; 'sipping@ietf.org'
> Subject: RE: [Sipping] Re: Comment on GRUU requirements
> 
> 
> I believe this was Adam's original suggestion. Works
> for me.
> 
> --- "Christer Holmberg (JO/LMF)"
> <christer.holmberg@ericsson.com> wrote:
> > 
> > Hi,
> > 
> > I don't know if this has been discussed, but how
> > about putting the GRUU in the Contact header, AND
> > define a "gruu" header/uri parameter. Now, existing
> > nodes will, as Rohan put it, be as broken as they
> > are today, but they will still work, and nodes
> > supporting GRUU will use the gruu parameter to
> > indicate/detect that the uri in the Contact header
> > is a GRUU. But, no new header needs to be defined (I
> > think we already have too many headers carrying
> > URIs, many of which we don't use - or even don't
> > know how or when to use), and there would be no
> > backward compability issues, and no need to redefine
> > existing headers/functionality.
> > 
> > Regards,
> > 
> > Christer Holmberg
> > Ericsson Finland
> > 
> > > -----Original Message-----
> > > From: Sean Olson [mailto:seancolson@yahoo.com]
> > > Sent: 16. elokuuta 2003 1:50
> > > To: Rohan Mahy; Jonathan Rosenberg
> > > Cc: Adam Roach; 'Alan Johnston';
> > 'sipping@ietf.org'
> > > Subject: Re: [Sipping] Re: Comment on GRUU
> > requirements
> > > 
> > > 
> > > I have to agree with Rohan. I don't understand the
> > > purpose of NOT placing the GRUU in the Contact:
> > header
> > > if possible. Whether or not this GRUU ties into a
> > > proxy is up for debate, but in practice I believe
> > it
> > > will be the rule rather than the exception.
> > Similar
> > > (but not identical) to the fact that a registrar
> > is
> > > not absolutely necessary to use SIP but it sure
> > makes
> > > things easier in practice.
> > > 
> > > 
> > > __________________________________
> > > Do you Yahoo!?
> > > Yahoo! SiteBuilder - Free, easy-to-use web site
> > design software
> > > http://sitebuilder.yahoo.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
> 
> 
> __________________________________
> Do you Yahoo!?
> Yahoo! SiteBuilder - Free, easy-to-use web site design software
> http://sitebuilder.yahoo.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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Subject: [Sipping] RFC 3578 on Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3578

        Title:      Mapping of Integrated Services Digital Network
                    (ISDN) User Part (ISUP) Overlap Signalling
                    to the Session Initiation Protocol (SIP)
        Author(s):  G. Camarillo, A. B. Roach, J. Peterson, L. Ong
        Status:     Standards Track
        Date:       August 2003
        Mailbox:    Gonzalo.Camarillo@ericsson.com,
                    adam@dynamicsoft.com, jon.peterson@neustar.biz,
                    lyong@ciena.com
        Pages:      13
        Characters: 26667
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sipping-overlap-04.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3578.txt


This document describes a way to map Integrated Services Digital
Network User Part (ISUP) overlap signalling to Session Initiation
Protocol (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).

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
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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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Date: Thu, 14 Aug 2003 10:10:01 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: AW: AW: [Sipping] FYI: RADIUS & SIP
In-reply-to: <5.2.0.9.0.20030807200108.05436950@mailhost.fokus.gmd.de>
To: "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
        "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Liess, Laura'" <Laura.Liess@t-systems.com>
Cc: hgs@cs.columbia.edu, jdrosen@dynamicsoft.com, aboba@internaut.com,
        sipping@ietf.org, "'Dumler, Alexander'" <Alexander.Dumler@telekom.de>,
        "'Wolff, Christian'" <Christian.Wolff@telekom.de>
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> Still less hard than accounting over a protocol which does not exist.

I agree. Also, I will keep reminding folks there are web services out there.

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Jiri Kuthan
> Sent: Thursday, August 07, 2003 1:03 PM
> To: Dean Willis; Liess, Laura
> Cc: hgs@cs.columbia.edu; jdrosen@dynamicsoft.com; aboba@internaut.com;
> sipping@ietf.org; Dumler, Alexander; Wolff, Christian
> Subject: Re: AW: AW: [Sipping] FYI: RADIUS & SIP
> 
> At 07:56 PM 8/7/2003, Dean Willis wrote:
> >On Wed, 2003-08-06 at 03:17, Liess, Laura wrote:
> >> What about the accounting draft draft-schulzrinne-accounting-sip-
> 01.txt? Any reason for not working on it?
> >> Laura
> >
> >Accounting over a non-reliable transport protocol is a Very Hard
> >Problem. That's one of the reaons for DIAMETER.
> 
> Still less hard than accounting over a protocol which does not exist.
> We actually developed some DIAMETER code but the problem is it is
> not running anywhere.
> 
> -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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Subject: Re: [Sipping] Re: Comment on GRUU requirements
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Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'Alan Johnston'" <alan.johnston@mci.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
To: Adam Roach <adam@dynamicsoft.com>
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On Monday, August 18, 2003, at 10:22 AM, Adam Roach wrote:
> Rohan Mahy [mailto:rohan@cisco.com] writes:
>
>> In fact we already have a precedent (in the REGISTER method)
>> that a Contact should be valid for more than one dialog.
>
> This is one of those perfect-hindsight problems, but
> the issue here is that "Contact" in REGISTER is wildly
> different in meaning than "Contact" in any other context
> (except, arguably, 300-class responses).

I completely disagree; and I believe that this is the core of the 
semantic problem we are having.  I take "Contact" to mean a URI where a 
particular user or service can be reached.  This makes a Contact in a 
REGISTER no different from a Contact in an INVITE or a <contact> in the 
dialog package.  I believe that any other definition of Contact is 
completely contrived.

thanks,
-rohan


> These uses really should have employed a different header
> than "Contact". Citing them as precedent is shaky
> at best.
>
> /a


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To: "'Rohan Mahy'" <rohan@cisco.com>, Adam Roach <adam@dynamicsoft.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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	 <alan.johnston@mci.com>,
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Subject: RE: [Sipping] Re: Comment on GRUU requirements
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Rohan Mahy [mailto:rohan@cisco.com] writes:

> I take "Contact" to mean a URI where a particular user or
> service can be reached.  This makes a Contact in a REGISTER
> no different from a Contact in an INVITE or a <contact> in
> the dialog package.  I believe that any other definition of
> Contact is completely contrived.

Contrived, perhaps -- but practical and necessary. Without
the context provided by a route set, many of the applications
of contact you describe don't necessarily resolve from the
Internet. In an ideal world, your characterization of Contact
would be spot on; the real world, however, isn't always so
cooperative.

/a

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Date: Tue, 19 Aug 2003 23:09:50 -0700
Subject: Re: [Sipping] Question on pick-up in service-examples-04
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On Wednesday, August 13, 2003, at 11:26 AM, Jonathan Rosenberg wrote:

> Old email, and a sort-of-related point.
>
> It seems to me that the INVITE/Replaces really needs to be targeted at 
> the GRUU for Alice's phone, no? Otherwise this service won't work in 
> places where Carol is located in a different network. For example, if 
> Carol takes here SIP phone to a hotel, she won't be able to directly 
> reach the contact, and so the pickup will fail. I think we want pickup 
> services to work independent of the physical locations of the two 
> devices.
>
> In that case what is more useful in the dialog event package is the 
> gruu, not just the contact.

Or this can be seen as yet another motivation to always put a GRUU in 
the Contact header....

thx,
-r

>
> -Jonathan R.
>
> Alan Johnston wrote:
>
>> John,
>> You are correct - the NOTIFY F5 does not show the Contact URI.  It 
>> should have been shown in a Remote Target tag.  However, this tag is 
>> listed as optional in the dialog package - I'm not sure why as it is 
>> useful in a number of situations.
>> I agree that it is important to use the Contact URI in this flow.
>> Thanks,
>> Alan Johnston
>> MCI
>> At 09:41 AM 7/10/2003 +0100, Elwell, John wrote:
>>> Can anyone explain something to me in section 2.16 (call pick-up)?
>>>
>>> In F7, the Request-URI is the contact URI for Alice: 
>>> alice@client.atlanta.example.com
>>>
>>> It is not clear how Carol's UA receives this contact address. F5 
>>> from Bob to Carol provides only the AoR: alice@atlanta.example.com.
>>>
>>> I believe it is important to use the contact URI in F7 to avoid the 
>>> case where the INVITE request reaches the wrong UA at Alice. This is 
>>> particularly true when Alice is in a legacy network - the contact 
>>> address might help selection of the correct gateway.
>>>
>>> John (john.elwell@siemens.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
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> 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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Subject: Re: [Sipping] App-Interaction: Considering models
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Inline....


On Wednesday, August 13, 2003, at 10:54 PM, Cullen Jennings wrote:
> I think all of these proposals are significantly more complex than 
> using a
> MESSAGE/NEW_METHOD/NOTIFY. I will mention a few issues with the 
> proposal
> inline but first let me a few other proposals...
>
> All of these have a call flow like
>
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |
>> 5|---SIP MESG---->|                |
>> 6|<-----200-------|                |
>
> The question for KMPL is, what method to use in msg 5.
>
> Proposal 1) Use MESSAGE. Many people think the semantics of message are
> display this IM to a user. Perhaps, but I send IM to applications and 
> they
> send me back things like stock quotes all the time. I have no problem 
> with
> semantics of MESSAGE being deliver this message to the addressed user 
> or
> automaton. That's what we are doing here. Misrouted message will cause
> trivial problems, but this is a bogus issue - misrouted INVITE or 
> MESSAGE
> will cause problems to - don't misroute things.

Adam has already given good/likely concrete examples where "bad things" 
can happen.  Unless you can prove that these won't actually happen, 
then I suggest we discard this option and move on.

> Proposal 2) Use some new method like STIMULUS. The semantics of this 
> are
> some user input stimulus is being sent - STIMULUS is to input what 
> MESSAGE
> is to output.

I'm fine with this option.  It will work.  Should this NEWMETHOD be 
used for responses and new markup or just responses?  as long as you 
are sending them to the app server and not the UAS in your picture, 
this should be on a new dialog.

> Proposal 3) Use NOTIFY. Semantics are, here is some information about 
> some
> event that you requested to be informed about. The usual complaint on 
> this
> one is that there was no Subscribe, or it was implicit, or it was 
> explicit,
> or it violated some law of the SIP world. I think there was a 
> subscribe, it
> was in the message that initiated the script. There is also a clear 
> time it
> will end - when the script is terminated or when the dialog of the 
> message
> that initiated the script ends.

How long in seconds exactly is that again?  ;-) Ah, you don't know 
unless you are using session timer you say.  Well, unless you are 
requiring session timer, I object on the grounds that my UA can never 
recover lost state with this "implicit subscription" mechanism.

> We all remember that implicit subscribes
> where bad (whatever an implicit subscribe is) but people seem to have 
> a hard
> time remembering why this will be bad here.

also, if you plan to use implicit subscriptions, this will require an 
initial NOTIFY after the initial INVITE.  this has been a remarkable 
PAI for REFER. also, since this is going to a different UA than the UAS 
(if you sent it anyway you would screw up the CSeqs) it needs to be on 
a different dialog, but then you need a way to initiate this dialog 
which gets pretty ugly, since there is no SUBSCRIBE. You would need to 
send a NOTIFY with one tag only, and then complete the dialog in the 
response to the NOTIFY.

I think implicit subs are bad in this case because:
a) the dialog identifiers for the subscription are in a (currently) 
illegal combination of SIP messages
b) the state of the associated dialog is completely decoupled from the 
state of subscription, which makes the Subscription-State header (which 
is supposed to convey BOTH) fairly useless.
c) experience has shown that implicit subs for REFER where a pain to 
implement and only now folks are starting to get it right; this use is 
even more complicated
d) you need a different GRUU for each "subscription"
e) even if you use a GRUU for each "subscription", you have a dialog 
correlation problem in the example I give below.

     UA1            App          Proxy           UA2            UA3
      |              |             |              |              |
      |INVITE------->|             |              |              |
      |              |INVITE------>|              |              |
      |              |App-Info: *  |INVITE------->|              |
      |              |             |App-Info: *   |              |
      |              |             |<---------180-|              |
      |              |<----------------NOTIFY-----|              |
      |              |-----------------200--------|              |
      |              |             |              |              |
      |              |             |INVITE---------------------->|
      |              |             |App-Info: *   |              |
      |              |             |<------------------------180-|
      |              |<---------------------NOTIFY---------------|
      |              |----------------------200------------------|
      |              |             |              |              |

if the header is App-Info: sip:unique-instance23988413@appserver, then 
the appserver still receives requests from multiple UAs.  The NOTIFYs 
are on completely separate new dialogs.  I don't think there is any way 
to correlate the first notify with the first INVITE fork, and the 
second notify with the second INVITE fork.

f) timers are not explicit
...
g) there is no technical reason given why a new method would not work.  
the only objections have been philosophical

!!!FOR GODS SAKE DONT USE IMPLICIT SUBSCRIPTIONS!!!!


> All 3 of these proposals have something in common - there is no 
> concrete
> argument that any of them would not work.

I think I have provided ample bad reasons why not to do 1 and 3.

> They send to a URI the AS provided
> so there are no forking issues. There is some grief on the semantics. 
> Option
> 2 has the advantage of it lets us define the semantics however we 
> want. I
> think that 2 is the best way to go but I don't care greatly. I do need
> convincing on what's wrong with these to move to something more 
> complex.
>
> More comments inline

likewise...


> On 8/13/03 3:32 PM, "Adam Roach" <adam@dynamicsoft.com> wrote:
>
>> After considering Jonathan's contention that defining
>> a new method for conveying stimulus is creating something
>> that is similar in meaning to NOTIFY, I'm inclined to
>> agree to some point. There are still subtle differences
>> in meaning, so I'm not completely convinced that a new
>> method is wrong, but I understand the point he's trying
>> to make.
>>
>> After discussing it privately with others, I've concluded
>> that there are three viable models worth considering. There
>> are, of course, some minor variations that can be made to
>> each.
>>
>> 1) Standard HTTP Handling
>>
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |   AS adds HTTP URL to 200
>> 5|---HTTP GET---->|                |   UAC requests KPML script
>> 6|<-----200-------|                |   AS replies with script
>> 7|---HTTP POST--->|                |   UAC provides digits
>> 8|<-----200-------|                |   AS acknowledges digits
>>
>> Of the three options presented here, I see this as the best
>> solution. Basically, the app server includes an HTTP URL
>> in the 200 response to the INVITE that it has proxied. The
>> UAC then does a GET on the KPML script. Just like with
>> HTML forms, the UAC does whatever local stuff is necessary
>> to "fill out the form" (in this case, collect digits), and
>> then posts the result back to the App Server. This is all
>> plain-Jane unextended HTTP as it was intended to work.
>>
>
> One of the difficult parts of this is security.

agreed.  this also potentially has the same dialog correlation problem 
that I described above.

> This will get more complicated once you provide a way to send in 
> message 4
> the credentials that will allow 7. It may also make the App-Info 
> header only
> really work with scripts that use HTTP to post instead of other things.
> Also, in the case of a financial institutions that have to log 
> electronic
> communications, you will need to provide way to insert logging HTTP 
> proxies
> and probably proxies for FW traversal too. Of course the AS or the UAC 
> will
> need to figure out the network topology to know if the AS is inside or
> outside the FW. It's possible to configure all the domain and HTTP 
> proxies
> rules into the device but it still hard to secure - particularly on 
> devices
> that don't want to implement HTTPS due to not much memory.
>
>
>> 2) Explicit Subscriptions (with GRUU)
>>
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |
>> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
>> 6|------200------>|                |
>> 7|-----NOTIFY---->|                |
>> 8|<-----200-------|                |
>> 9|-----NOTIFY---->|                |   UAC provides digits
>> A|<-----200-------|                |   AS acknowledges digits
>>
>> My second choice would be an explicit subscription to stimulus
>> events. The only sticky part in this solution is the question
>> of how the AS can route a SUBSCRIBE to the UAC. Once we have
>> a GRUU solution in place, this is a no brainer: the AS can
>> extract the GRUU from the INVITE, and use it as the target
>> of its SUBSCRIBE. The KPML script is included in the SUBSCRIBE,
>> and essentially serves as a filter.

I don't see how Adam plans to correlate the SUBSCRIBE with the 
INVITE-initiated dialog.  You can't send the SUBSCRIBE on the existing 
dialog, because you will screw up the CSeq space.

btw: If an event package is defined for this, will we allow PUBLISH as 
well?

> I sort of see this one, but what if the App Server wanted to run an 
> App on
> the UAS instead of the UAC. Would the 180 have the GRU? Would it be
> reliable?
>
> I seem to recall someone asking about the expire time on the 
> subscription
> :-)
>
> I'd be keener on GRUU solutions if GRUU was a solved problem.

we need to solve this problem anyway if you want transfer to work, so 
I'm not especially concerned.


>> 3) Explicit Subscriptions (with REFER)
>>
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |   AS adds SIP URL to 200
>> 5|-----REFER----->|                |
>> 6|<-----200-------|                |
>> 7|<----NOTIFY-----|                |
>> 8|------481------>|                |
>> 9|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
>> A|------200------>|                |
>> B|-----NOTIFY---->|                |
>> C|<-----200-------|                |
>> D|-----NOTIFY---->|                |   UAC provides digits
>> E|<-----200-------|                |   AS acknowledges digits
>>
>> Although somewhat chattier than the previous solution, this gets 
>> around
>> the GRUU issue by having the AS provide a SIP URI that routes to 
>> itself
>> in the 200 response. The UAC then uses this URI to send a REFER to the
>> AS. This REFER tells the AS to SUBSCRIBE to the UAC. The flow from 
>> that
>> point forward is the same as for solution 2.
>
> I got to admit, I had never considered anything like this. I think it 
> is
> going to be hard to convince folks this is really easier and better 
> than
> sending DTMF over INFO. Keep in mind that KPML is for little device 
> that
> think 14k of memory is a lot. Real devices are just going to do VXML 
> and
> HTML and ignore KPML.

I don't understand how Adam can correlate the REFER (which has to be 
using a different dialog than the original INVITE) with the dialog for 
which you are interested in collecting digits.

thanks,
-rohan


> Cullen
>
>
> _______________________________________________
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Subject: Re: [Sipping] App-Interaction: Considering models
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On Friday, August 15, 2003, at 04:58 PM, Adam Roach wrote:
> Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] writes:
>
>>> 2) Explicit Subscriptions (with GRUU)
>>
>> This was one of the solutions I outlined in my initial email, yes.
>> GRUU is not the only problem. There is also the correlation problem -
>> how does the UA know that the SUBSCRIBE comes from an entity
>> along the
>> call path? AUthorizing the subscribe as above would be really hard.
>
> The SUBSCRIBE itself will need to contain information about the
> dialog for which it is requesting information anyway, right?
> Getting this information requires you to be in a position that
> would allow you to intercept the digits themselves.
>
> If you want to keep this information even more securely under
> wraps, use sips: to set up the call; the dialog information
> is revealed only to people in the call path, and only they would
> be able to subscribe to keypad events.

I wanted to clarify something about dialog usage in this example.

A long running assumption that many seem to be making (not necessarily 
Jonathan or Adam) is that the stimulus from UA1 running the KPML script 
needs to travel over the same dialog that UA1 has with UA2, even though 
nobody wants the stimulus to go to UA2, it needs to go to some 
intermediary, or another UA somewhere. We no longer have a dialog, we 
have a "trialog".   This trialog thing is an abomination which 
fortunately does not exist.  Handling multiple dialogs resulting in the 
same set of dialogs (resulting from a forked request), and reference 
counting invites and subscriptions to determine when a dialog is 
finally terminated is already a huge pain.
Trialogs simply don't work because each side of a dialog needs to 
manage their own CSeq space.

thanks,
-rohan


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Subject: RE: [Sipping] App-Interaction: Considering models
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Rohan Mahy [mailto:rohan@cisco.com] writes:

>> 1) Standard HTTP Handling
>
> this also potentially has the same dialog
> correlation problem that I described above.

You're right; it does. My enthusiasm for this solution
is consequently waning.

I will add that the "just use a new method" approach
suffers from precisely the same relic. In fact, the
only thing proposed to date which does not is...

>> 2) Explicit Subscriptions (with GRUU)
>>
>> UAC           App Server          UAS
>> 1|-----INVITE---->|                |
>> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
>> 3|                |<-----200-------|
>> 4|<-----200-------|                |
>> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
>> 6|------200------>|                |
>> 7|-----NOTIFY---->|                |
>> 8|<-----200-------|                |
>> 9|-----NOTIFY---->|                |   UAC provides digits
>> A|<-----200-------|                |   AS acknowledges digits

> I don't see how Adam plans to correlate the SUBSCRIBE with the 
> INVITE-initiated dialog.  You can't send the SUBSCRIBE on the 
> existing dialog, because you will screw up the CSeq space.

Of *course* you don't re-use the dialog. Steps 1 through 4 are
in one dialog (between the UAC and the UAS); steps 5 through A
are in a second dialog (between the App Server and the UAC).

To answer the implied question about how to perform correlation,
I suggest wholesale copying of section 3.2 from
draft-ietf-sipping-dialog-package-02.

> > I'd be keener on GRUU solutions if GRUU was a solved problem.
> 
> we need to solve this problem anyway if you want transfer to work, so 
> I'm not especially concerned.

Yes, I forgot to address this. Given the relative maturity
of each, I imagine that App-Interaction will hit the finish
line about the same time as the GRUU work. If you additionally
consider complexity, GRUU should finish first.

/a

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Subject: Re: [Sipping] App-Interaction: Considering models
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On Wednesday, August 20, 2003, at 01:14 PM, Adam Roach wrote:

> Rohan Mahy [mailto:rohan@cisco.com] writes:
>
>>> 1) Standard HTTP Handling
>>
>> this also potentially has the same dialog
>> correlation problem that I described above.
>
> You're right; it does. My enthusiasm for this solution
> is consequently waning.
>
> I will add that the "just use a new method" approach
> suffers from precisely the same relic.

using a new method does not actually suffer from the dialog correlation 
problem.  You are sending a feedback or stimulus body in the new 
method, which just needs to contain a session-id or dialog-information 
to describe the dialog for which the feedback/stimulus is associated.  
You can use the same URI (as long as it has GRUU properties) for ALL 
your feedback for the same service/application if you'd like (imagine a 
prepaid application that can use the same URI for a T3 gateway worth of 
sessions).  I think this is a nice property.

> In fact, the
> only thing proposed to date which does not is...
>
>>> 2) Explicit Subscriptions (with GRUU)
>>>
>>> UAC           App Server          UAS
>>> 1|-----INVITE---->|                |
>>> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
>>> 3|                |<-----200-------|
>>> 4|<-----200-------|                |
>>> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
>>> 6|------200------>|                |
>>> 7|-----NOTIFY---->|                |
>>> 8|<-----200-------|                |
>>> 9|-----NOTIFY---->|                |   UAC provides digits
>>> A|<-----200-------|                |   AS acknowledges digits
>
>> I don't see how Adam plans to correlate the SUBSCRIBE with the
>> INVITE-initiated dialog.  You can't send the SUBSCRIBE on the
>> existing dialog, because you will screw up the CSeq space.
>
> Of *course* you don't re-use the dialog. Steps 1 through 4 are
> in one dialog (between the UAC and the UAS); steps 5 through A
> are in a second dialog (between the App Server and the UAC).
>
> To answer the implied question about how to perform correlation,
> I suggest wholesale copying of section 3.2 from
> draft-ietf-sipping-dialog-package-02.

ok, that would work fine.  however, you require a subscription per 
dialog, unless you write a subscription filter body.

>>> I'd be keener on GRUU solutions if GRUU was a solved problem.
>>
>> we need to solve this problem anyway if you want transfer to work, so
>> I'm not especially concerned.
>
> Yes, I forgot to address this. Given the relative maturity
> of each, I imagine that App-Interaction will hit the finish
> line about the same time as the GRUU work. If you additionally
> consider complexity, GRUU should finish first.
>
> /a


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Subject: Re: [Sipping] App-Interaction: Considering models
From: Cullen Jennings <fluffy@cisco.com>
To: Adam Roach <adam@dynamicsoft.com>, "'Rohan Mahy'" <rohan@cisco.com>
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On 8/20/03 13:14, "Adam Roach" <adam@dynamicsoft.com> wrote:


>> 
>> this also potentially has the same dialog
>> correlation problem that I described above.
.. Snip ...
> In fact, the
> only thing proposed to date which does not is...
> 
>>> 2) Explicit Subscriptions (with GRUU)
>>> 
>>> UAC           App Server          UAS
>>> 1|-----INVITE---->|                |
>>> 2|                |-----INVITE---->|   AS extracts GRUU from INVITE
>>> 3|                |<-----200-------|
>>> 4|<-----200-------|                |
>>> 5|<---SUBSCRIBE---|                |   AS SUBSCRIBEs with KPML script
>>> 6|------200------>|                |
>>> 7|-----NOTIFY---->|                |
>>> 8|<-----200-------|                |
>>> 9|-----NOTIFY---->|                |   UAC provides digits
>>> A|<-----200-------|                |   AS acknowledges digits
> 
.. Snip... 
> 
>>> I'd be keener on GRUU solutions if GRUU was a solved problem.
>> 
>> we need to solve this problem anyway if you want transfer to work, so
>> I'm not especially concerned.
> 
> Yes, I forgot to address this. Given the relative maturity
> of each, I imagine that App-Interaction will hit the finish
> line about the same time as the GRUU work. If you additionally
> consider complexity, GRUU should finish first.
> 
> /a
>

Yes, I agree with this logic. So are we coming to the conclusion that the
GRUU/SUB/NOT is the right way to go? I can try and flesh out some details on
this if we are.





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Rohan Mahy [mailto:rohan@cisco.com] writes:

> using a new method does not actually suffer from the dialog 
> correlation problem.  You are sending a feedback or stimulus
> body in the new method, which just needs to contain a session-id
> or dialog-information to describe the dialog for which the
> feedback/stimulus is associated.  

You are correct. Ignore my previous objection.

/a

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From: Adam Roach <adam@dynamicsoft.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Adam Roach
	 <adam@dynamicsoft.com>, "'Rohan Mahy'" <rohan@cisco.com>
Cc: sipping <sipping@ietf.org>, Jonathan Rosenberg
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Subject: RE: [Sipping] App-Interaction: Considering models
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Cullen Jennings [mailto:fluffy@cisco.com] writes:

> So are we coming to the conclusion that the
> GRUU/SUB/NOT is the right way to go? I can try
> and flesh out some details on this if we are.

Personally, I now favor this method, since I consider
the http correlation issue that we've discussed to
be a non-starter. I don't think it's unsolvable; I
just am wary of solutions that obviously need hacks,
even when viewed at the mile-high level.

However, my arguments for preferring explicit
use of SUBSCRIBE over a new method are, at this point,
largely philosophical, not technical. I'm not
certain that discussion has played out completely
yet.

/a

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Date: Fri, 22 Aug 2003 03:21:39 +0200
To: Rohan Mahy <rohan@cisco.com>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: [Sipping] Re: Update to 3pcc
Cc: Dean Willis <dean.willis@softarmor.com>,
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That's it, thanks Rohan. If there is an element in the scenario (gateway), which 
provides a valuable service (P$S$T$N), then it implements session timer to protect 
itself (to cut-off a call). Elements that don't provide a service may not support 
timer, who cares.

-jiri

At 07:53 PM 8/15/2003, Rohan Mahy wrote:

>On Friday, August 15, 2003, at 08:50 AM, Jonathan Rosenberg wrote:
>
>>I agree with Dean that this is not at all what session timer was meant for. The spec is pretty explicit about its usage as a cleanup mechanism.
>
>if it works without changes to the existing mechanism, then what is the problem?
>
>>You'll have serious problems if the refreshing side is not trusted. Session timer allows either side to refresh. So, if this is a PC to phone case and PC is refreshing this won't work.
>
>the gateway is the refresher, and is trusted.  the example i gave nicely fits a business model where the provider accounts for PSTN usage, but does not charge/account for arbitrary SIP sessions.   Easy things should be easy, contrived things will be hard and will require a B2BUA.
>
>>Even when its the gateway refreshing, 403 doesnt cause the gateway to terminate the call - at least it shouldnt, I dont think. Nothing in the spec says to tear down the call if you get a 403 to an INVITE.
>
>you are correct, but the call will be torn down after the session expires.  since the session is not refreshed when the proxy responds with a 403, the session will expire and the gateway will send a BYE.
>
>thanks,

--
Jiri Kuthan            http://iptel.org/~jiri/ 


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Subject: Re: [Sipping] Re: Update to 3pcc
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At 09:17 PM 8/13/2003, Jonathan Rosenberg wrote:
>Sorry for taking so long to weigh in here.
>To address the specific billing issue, the problem is that, in fact, there ARE cases where the call is charged for differently if it was initiated as a result of click-to-dial. The user that makes the initial INVITE may not be the one that needs to be charged. There may be ways around that for REFER by having the INVITE carry some kind of special billing tokens or something (such tokens being pushed to the user in the Refer-To), but these then require the c2d provider and the caller to utilize the same provider. The 3pcc mechanism does not have this constraint.

I would prefer focusing on a common case. I really do not think 3-rd party billing 
for click-to-dial is something very usual. If you think so, then that's the example which
should come in the 3pcc draft.

>However, in my mind, the real issue is one of control. 3pcc is useful in applications where the server needs to be in the signaling path in order to do or modify things. Thats not the case for all applications, but its true for many. Prepaid is one we have discussed as well. REFER just doesnt work there.

I almost agree, but that's a different topic than I raised.

>I'd also point out that the 3pcc spec is not saying that the best practice for c2d is to use 3pcc; it saying that if you want to do third party call control, here is how you do it.

Sometimes, BCP for a technology is not to use it. I don't think IETF should document  
"BCP for 3pcc" on examples which are better implemented with another technology. 
If you are enthusiastic about HTTP, publish "BCP for Use of HTTP" and show examples 
of using HTTP to control media gateways, some people will not be happy either (unless 
it was April 1st). 


>I don't disagree that 3pcc for c2d can be complicated, and REFER does have a better story in terms of the media story. However, other work going on, namely the app interaction work, does address some of the big issues there in a pretty elegant way, I think.

I really prefer keeping simple things simple. One could of course build click-to-dial
even using VoiceXML-driven MGCP-enabled Softswitch, but that's not the goal. REFER
is simple and its purpose perfectly fits the c-t-d app.

-Jiri 


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Date: Fri, 22 Aug 2003 11:28:52 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Re: Update to 3pcc
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Where are the service providers in this discussion on billing?  ;-)

Thanks, Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Jiri Kuthan
> Sent: Thursday, August 21, 2003 9:22 PM
> To: Jonathan Rosenberg
> Cc: David Schwartz; 'sipping@ietf.org'
> Subject: Re: [Sipping] Re: Update to 3pcc
> 
> At 09:17 PM 8/13/2003, Jonathan Rosenberg wrote:
> >Sorry for taking so long to weigh in here.
> >To address the specific billing issue, the problem is that, in fact,
> there ARE cases where the call is charged for differently if it was
> initiated as a result of click-to-dial. The user that makes the initial
> INVITE may not be the one that needs to be charged. There may be ways
> around that for REFER by having the INVITE carry some kind of special
> billing tokens or something (such tokens being pushed to the user in the
> Refer-To), but these then require the c2d provider and the caller to
> utilize the same provider. The 3pcc mechanism does not have this
> constraint.
> 
> I would prefer focusing on a common case. I really do not think 3-rd party
> billing
> for click-to-dial is something very usual. If you think so, then that's
> the example which
> should come in the 3pcc draft.
> 
> >However, in my mind, the real issue is one of control. 3pcc is useful in
> applications where the server needs to be in the signaling path in order
> to do or modify things. Thats not the case for all applications, but its
> true for many. Prepaid is one we have discussed as well. REFER just doesnt
> work there.
> 
> I almost agree, but that's a different topic than I raised.
> 
> >I'd also point out that the 3pcc spec is not saying that the best
> practice for c2d is to use 3pcc; it saying that if you want to do third
> party call control, here is how you do it.
> 
> Sometimes, BCP for a technology is not to use it. I don't think IETF
> should document
> "BCP for 3pcc" on examples which are better implemented with another
> technology.
> If you are enthusiastic about HTTP, publish "BCP for Use of HTTP" and show
> examples
> of using HTTP to control media gateways, some people will not be happy
> either (unless
> it was April 1st).
> 
> 
> >I don't disagree that 3pcc for c2d can be complicated, and REFER does
> have a better story in terms of the media story. However, other work going
> on, namely the app interaction work, does address some of the big issues
> there in a pretty elegant way, I think.
> 
> I really prefer keeping simple things simple. One could of course build
> click-to-dial
> even using VoiceXML-driven MGCP-enabled Softswitch, but that's not the
> goal. REFER
> is simple and its purpose perfectly fits the c-t-d app.
> 
> -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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Date: Tue, 26 Aug 2003 14:39:11 -0400
To: "Christian Stredicke" <stredicke@snom.de>, "'sipping'" <sipping@ietf.org>
From: Richard Shockey <richard@shockey.us>
Subject: Re: [Sipping] Kill Overlap Dialling
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At 11:06 AM 8/26/2003 +0530, Christian Stredicke wrote:

>Hi All,

Christian I think this should end the issue. The "out of box" experience 
with SIP phones does not have to replicate that of the land line service 
..the most compelling argument for a Dial button on SIP phones is D.


>the feedback which I received so far on overlap dialling was pretty
>clear: There will be no overlap dialling support for SIP.
>
>Arguments:
>
>A. ENUM does not support overlap dialling. There are security concerns
>on who will own the nodes of incomplete numbers and it was also reported
>that the overhead for the node lookup is too much.

Right we had this argument at the ENUM WG in Vienna.


>B. Countries with fixed-length numbers (like USA) can have a dial plan
>on the phone. Other countries always need to press the dial button
>except for cases where extension numbers are dialled. People calling
>from the US have to press a dial key (or #) after dialling a non-US
>number. That is accepted by the customers.
>
>C. The lack of support of 484 responses on PSTN gateways makes overlap
>dialling unusable anyway. A SIP extension would be necessary, but does
>not exist at this time and the probability that we reach consensus on
>such an extension in time is minimal.
>
>D. "Grandma" is not in the customer focus any more. "Grandchildren" are
>used to cell phones and press the dial button after making up their
>minds on which person to call.
>
>E. Phones without display will not be sold in large numbers any more. If
>vendors should decide to make SIP phones without display, they must
>include a dial button (timeout of 5 seconds to start dialling is not
>feasible as this makes printed telephone book lookup impossible).

LED's are cheep..



>Am I right? I this is so, we would like to stop efforts on implementing
>this in out endpoints.
>

Please...



 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 703.593.2683,  Fax: +1 815.333.1237
<mailto:richard(at)shockey.us> or <mailto:richard.shockey(at)neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<


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	 <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
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Christian Stredicke writes:

> C. The lack of support of 484 responses on PSTN gateways makes overlap
> dialling unusable anyway. A SIP extension would be necessary, but does
> not exist at this time

So.... RFC 3578 is a figment of my imagination?

/a

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Subject: Re: [Sipping] Kill Overlap Dialling
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Well there is RFC 3578.


On 8/25/03 22:36, "Christian Stredicke" <stredicke@snom.de> wrote:

> Hi All,
> 
> the feedback which I received so far on overlap dialling was pretty
> clear: There will be no overlap dialling support for SIP.
> 
> Arguments:
> 
> A. ENUM does not support overlap dialling. There are security concerns
> on who will own the nodes of incomplete numbers and it was also reported
> that the overhead for the node lookup is too much.
> 
> B. Countries with fixed-length numbers (like USA) can have a dial plan
> on the phone. Other countries always need to press the dial button
> except for cases where extension numbers are dialled. People calling
> from the US have to press a dial key (or #) after dialling a non-US
> number. That is accepted by the customers.
> 
> C. The lack of support of 484 responses on PSTN gateways makes overlap
> dialling unusable anyway. A SIP extension would be necessary, but does
> not exist at this time and the probability that we reach consensus on
> such an extension in time is minimal.
> 
> D. "Grandma" is not in the customer focus any more. "Grandchildren" are
> used to cell phones and press the dial button after making up their
> minds on which person to call.
> 
> E. Phones without display will not be sold in large numbers any more. If
> vendors should decide to make SIP phones without display, they must
> include a dial button (timeout of 5 seconds to start dialling is not
> feasible as this makes printed telephone book lookup impossible).
> 
> 
> Am I right? I this is so, we would like to stop efforts on implementing
> this in out endpoints.
> 
> Best,
> 
> 
> Christian
> 
> 
> _______________________________________________
> 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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From: Adam Roach <adam@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, "'Rohan Mahy'" <rohan@cisco.com>,
        sipping <sipping@ietf.org>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Mon, 25 Aug 2003 16:28:44 -0500
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Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] wrote:

> I would argue that the App-Info header is what is actually creating 
> the state, and the state of this thing exists for the duration of the 
> dialog. The results of the user input are the state, but they aren't 
> normally sent anywhere. Rather, they are monitored through an 
> explicit subscription from the application server.

I thought I had mentioned it earlier (it was always in the back
of my mind), but the intention, in using explicit subscriptions,
was to not have an "App-Info" header. The SUBSCRIBE itself carries
the KPML script (which, upon close examination, is really more of
a filter than a script, so this usage seems appropriate).

So, I would argue that the INVITE exchange is what actually creates
this state. In other words, it makes just as much sense to subscribe
to this state as it does to subscribe to the state of a dialog.

> In this model, I suppose an explicit subscription makes 
> sense, but the whole thing seems convoluted. Why bother to
> explicitly subscribe to something when you created the state
> in the first place, and you are the only one who is interested
> in finding out about it?

Ah, but what if you *aren't* the only one interested in
finding out about it?

What if I go through some proxy that is doing some sort of
prepaid service [1] with two-stage dialing. If I want to end
my current call and start a new one, I send a long pound.
On the other hand, the thing that I've ended up calling has
a proxy that wants my keypresses also, because it's some
sort of call center that needs me to type in my account number.

It sure seems that it would be much, much simpler to consider
them as subscribing to the same state (keypad events) with
different filters, than to consider the two different "App-Info"
headers in the INVITE 200 response as creating two separate pieces
of state associated with (n.b. *not* "sent in") the dialog.

What is *really* nice about this is that one can envision future
systems where parties not on the call path (but still rendered to
the user through some other means) can tap this state without
additional specification; all they need is some mechanism to get
the dialog identifiers.

/a

[1] Don't get hung up on the fact that I said something that
    has billing implications. The point is that there might
    be two proxies wanting to know about keypresses, and this
    is an easily grasped system with which many people are
    already familiar.

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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inline.

Cullen Jennings wrote:

> Let me propose a realistic scenario that is a little more difficult with
> http. It is probably possible to solve, just complicated. I will admit that
> much of the reason many people don't want to use HTTP is because they think
> sending it in SIP will be less work - they may or may not be right but the
> "less work" on low end devices is what has driven us to a fairly messy
> situation in handling DTMF today. I want to come up this something that is
> appealing enough to the people doing evil things today that they stop doing
> evil things. That said, let me describe a scenario that does occur and is
> difficult with HTTP.

Do these people find KPML objectionable too? No doubt the easiest 
thing is to send INFO within the same dialog. However that is frought 
with real problems. Once you've gotten to the point where its 
something out of dialog, and something that requires you to fetch 
(with http presumably) or look at an embedded script, is http for 
pushing the results worse?

> 
> A residential phone user, called A, has a UA that they can use to make PSTN
> calls. This UA connects to a voice service provider, called Provider B that
> claims to terminate the calls to the PSTN. This provider does not directly
> terminate them but actually pass them to provider C. Provider C runs the
> call through a H.323 translator and terminates then sends the call to a 323
> GW. The 323 GW does not support RFC 2833. The user A UA is likely to be
> behind a NAT and the final GW might be behind a NAT depending on the country
> of termination. B does not wish A to learn the identify of C because A might
> get a better price and cut B out of the loop. C does not wish B to learn too
> much about their network topology, since C might actually be passing traffic
> to provider D. Provider B and C and both run transparent B2BUA, broken
> proxies, and firewalls to make this all possible. The 323 translator wants
> to receive DTMF in the signaling so that it can pass it to the GW. In a
> variant of the scenario, provider B is a prepaid service and wants to be
> able to intercept long #.
> 
> If the phone sent DTMF over INFO, the people with this problem would be
> happy. The development effort to send DTMF over INFO is very low. 

Well, hang on. Its "easy" if its any method within the existing 
dialog. Once its outside of the dialog, whether its SIP or HTTP, this 
seems to be more or less equially "problematic".

It would
> be possible to solve this example with 2833 to the 323 translator. The IETF
> suggested this was the right way to go - I am unaware of anyone who did
> that. Everyone decided some weird variant of sending it over SIP looked like
> a much better idea even though it was not interoperable. Perhaps they were
> all insane but people buying equipment convinced vendors to build it. If we
> want to standardize something that is going to be widely implemented, I
> think we need to keep what is deployed today and why it happened in mind.
> 
> It is probably possible to have provider B write a HTTP proxy like thing
> that takes posts from A and then relays them on to C and change it's call
> control software to rewrite the KPML scripts as they pass through the
> signaling proxy/b2bua. But this leads to the the NAT ALG like behavior that
> we are trying to avoid.

And how would doing this with SIP solve that? Seems like you would 
still need to modify the KPML scripts. However, instead of receiving 
POSTS and relaying thme to C, you would take SIP NEW-METHODS and relay 
them on to C. Indeed, if these NEW-METHOD requests have the dialog 
identifiers in the body, those need to be rewritten by the server at B.

So, what is the difference?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Date: Wed, 27 Aug 2003 10:22:31 +0900
To: Henry Sinnreich <Henry.Sinnreich@mci.com>,
        "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
From: Kumiko Ono <Ono.Kumiko@lab.ntt.co.jp>
Subject: RE: [Sipping] Re: Update to 3pcc
Cc: "'David Schwartz'" <davids@deltathree.com>, sipping@ietf.org
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I think that click-to-dial service providers are usually different
from PSTN gateway provider. The settlement between these providers
is usually done monthly.

In this case, click-to-dial provider that also provides pre-paid 
calling service might want to take the initiative in disconnect procedure 
like sending BYE request.

Thanks,
Kumiko Ono

At 11:28 03/08/22 -0500, Henry Sinnreich wrote:
>Where are the service providers in this discussion on billing?  ;-)
>
>Thanks, Henry
>
>> -----Original Message-----
>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
>> Jiri Kuthan
>> Sent: Thursday, August 21, 2003 9:22 PM
>> To: Jonathan Rosenberg
>> Cc: David Schwartz; 'sipping@ietf.org'
>> Subject: Re: [Sipping] Re: Update to 3pcc
>> 
>> At 09:17 PM 8/13/2003, Jonathan Rosenberg wrote:
>> >Sorry for taking so long to weigh in here.
>> >To address the specific billing issue, the problem is that, in fact,
>> there ARE cases where the call is charged for differently if it was
>> initiated as a result of click-to-dial. The user that makes the initial
>> INVITE may not be the one that needs to be charged. There may be ways
>> around that for REFER by having the INVITE carry some kind of special
>> billing tokens or something (such tokens being pushed to the user in the
>> Refer-To), but these then require the c2d provider and the caller to
>> utilize the same provider. The 3pcc mechanism does not have this
>> constraint.
>> 
>> I would prefer focusing on a common case. I really do not think 3-rd party
>> billing
>> for click-to-dial is something very usual. If you think so, then that's
>> the example which
>> should come in the 3pcc draft.
>> 
>> >However, in my mind, the real issue is one of control. 3pcc is useful in
>> applications where the server needs to be in the signaling path in order
>> to do or modify things. Thats not the case for all applications, but its
>> true for many. Prepaid is one we have discussed as well. REFER just doesnt
>> work there.
>> 
>> I almost agree, but that's a different topic than I raised.
>> 
>> >I'd also point out that the 3pcc spec is not saying that the best
>> practice for c2d is to use 3pcc; it saying that if you want to do third
>> party call control, here is how you do it.
>> 
>> Sometimes, BCP for a technology is not to use it. I don't think IETF
>> should document
>> "BCP for 3pcc" on examples which are better implemented with another
>> technology.
>> If you are enthusiastic about HTTP, publish "BCP for Use of HTTP" and show
>> examples
>> of using HTTP to control media gateways, some people will not be happy
>> either (unless
>> it was April 1st).
>> 
>> 
>> >I don't disagree that 3pcc for c2d can be complicated, and REFER does
>> have a better story in terms of the media story. However, other work going
>> on, namely the app interaction work, does address some of the big issues
>> there in a pretty elegant way, I think.
>> 
>> I really prefer keeping simple things simple. One could of course build
>> click-to-dial
>> even using VoiceXML-driven MGCP-enabled Softswitch, but that's not the
>> goal. REFER
>> is simple and its purpose perfectly fits the c-t-d app.
>> 
>> -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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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Christian Stredicke <stredicke@snom.de>
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Subject: Re: [Sipping] Kill Overlap Dialling
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Christian,

what you are saying is in line with the recently published RFC 3578. 
That RFC recommends that gateways tranforms everything to en-bloc, and 
that the overlap procedures defined in the document should be used only 
as a last resort. That is, only if the gateway *must* meet some 
particular requirements that would be impossible to meet without using them.

Gonzalo

Christian Stredicke wrote:

> Hi All,
> 
> the feedback which I received so far on overlap dialling was pretty
> clear: There will be no overlap dialling support for SIP.
> 
> Arguments:
> 
> A. ENUM does not support overlap dialling. There are security concerns
> on who will own the nodes of incomplete numbers and it was also reported
> that the overhead for the node lookup is too much.
> 
> B. Countries with fixed-length numbers (like USA) can have a dial plan
> on the phone. Other countries always need to press the dial button
> except for cases where extension numbers are dialled. People calling
> from the US have to press a dial key (or #) after dialling a non-US
> number. That is accepted by the customers.
> 
> C. The lack of support of 484 responses on PSTN gateways makes overlap
> dialling unusable anyway. A SIP extension would be necessary, but does
> not exist at this time and the probability that we reach consensus on
> such an extension in time is minimal. 
> 
> D. "Grandma" is not in the customer focus any more. "Grandchildren" are
> used to cell phones and press the dial button after making up their
> minds on which person to call.
> 
> E. Phones without display will not be sold in large numbers any more. If
> vendors should decide to make SIP phones without display, they must
> include a dial button (timeout of 5 seconds to start dialling is not
> feasible as this makes printed telephone book lookup impossible).
> 
> 
> Am I right? I this is so, we would like to stop efforts on implementing
> this in out endpoints.
> 
> Best,
> 
> 
> Christian
> 
> 
> _______________________________________________
> 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


_______________________________________________
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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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Adam Roach wrote:

> Cullen Jennings [mailto:fluffy@cisco.com] writes:
> 
>  > So are we coming to the conclusion that the
>  > GRUU/SUB/NOT is the right way to go? I can try
>  > and flesh out some details on this if we are.
> 
> Personally, I now favor this method, since I consider
> the http correlation issue that we've discussed to
> be a non-starter. I don't think it's unsolvable; I
> just am wary of solutions that obviously need hacks,
> even when viewed at the mile-high level.


Well, the correlation would work identically to the SIP case. The HTTP 
request would need to carry some kind of dialog identifier, whether 
its in the body or as part of the R-URI.


> 
> However, my arguments for preferring explicit
> use of SUBSCRIBE over a new method are, at this point,
> largely philosophical, not technical. I'm not
> certain that discussion has played out completely
> yet.

Well, let me tease this out a bit.

The purpose of SUBSCRIBE is to ask about the state of something that 
exists already in the network. The SUBSCRIBE message itself does not 
create an instance of the thing being monitored, nor does the lifetime 
of the subscription guide the lifetime of the state being monitored. 
Indeed, I recall that Eric had objected to an explicit SUBSCRIBE for 
carrying the KPML, since it doesnt make sense to have SUBSCRIBE create 
the state being monitored.

I would argue that the App-Info header is what is actually creating 
the state, and the state of this thing exists for the duration of the 
dialog. The results of the user input are the state, but they aren't 
normally sent anywhere. Rather, they are monitored through an explicit 
subscription from the application server.

In this model, I suppose an explicit subscription makes sense, but the 
whole thing seems convoluted. Why bother to explicitly subscribe to 
something when you created the state in the first place, and you are 
the only one who is interested in finding out about it? Why bother 
managing the state of the subscription AND the user input state 
separately?

The overall system, to me, feels a lot more like form input on the 
web. A server in the network provides a script that gives context for 
the user to provide input. This input is then sent back to the source 
of the script in the first place, which then returns (possibly) 
another script to execute.

As such, I have more or less talked myself out of implicit 
subscriptions, but also explicit subscriptions at the same time. I 
would say that this is either a new method or HTTP, which still 
remains my favorite.. I'll respond to Cullen's note on this separately.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
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
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Christian Stredicke
	 <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
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RFC3578:

   Therefore, the preferred solution for a gateway handling PSTN overlap
   signalling and SIP is to convert the PSTN overlap signalling into SIP
   en-bloc signalling using number analysis and timers.  The gateway
   waits until all the signalling messages carrying parts of the
   callee's number arrive, and only then, it generates a SIP INVITE
   request.  Section 2 describes how to convert ISUP overlap signalling
   into en-bloc SIP this way.

RFC3578 does not endorse the use of overlap dialing for SIP - it endorses
the conversion of overlap to en-bloc as necessary. The mechanism for overlap
mapping described in the document is restricted to quite a limited
applicability, and has a number of drawbacks.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, August 26, 2003 10:47 AM
> To: Christian Stredicke; 'sipping'
> Subject: Re: [Sipping] Kill Overlap Dialling
> 
> 
> 
> Well there is RFC 3578.
> 
> 
> On 8/25/03 22:36, "Christian Stredicke" <stredicke@snom.de> wrote:
> 
> > Hi All,
> > 
> > the feedback which I received so far on overlap dialling was pretty
> > clear: There will be no overlap dialling support for SIP.
> > 
> > Arguments:
> > 
> > A. ENUM does not support overlap dialling. There are 
> security concerns
> > on who will own the nodes of incomplete numbers and it was 
> also reported
> > that the overhead for the node lookup is too much.
> > 
> > B. Countries with fixed-length numbers (like USA) can have 
> a dial plan
> > on the phone. Other countries always need to press the dial button
> > except for cases where extension numbers are dialled. People calling
> > from the US have to press a dial key (or #) after dialling a non-US
> > number. That is accepted by the customers.
> > 
> > C. The lack of support of 484 responses on PSTN gateways 
> makes overlap
> > dialling unusable anyway. A SIP extension would be 
> necessary, but does
> > not exist at this time and the probability that we reach 
> consensus on
> > such an extension in time is minimal.
> > 
> > D. "Grandma" is not in the customer focus any more. 
> "Grandchildren" are
> > used to cell phones and press the dial button after making up their
> > minds on which person to call.
> > 
> > E. Phones without display will not be sold in large numbers 
> any more. If
> > vendors should decide to make SIP phones without display, they must
> > include a dial button (timeout of 5 seconds to start dialling is not
> > feasible as this makes printed telephone book lookup impossible).
> > 
> > 
> > Am I right? I this is so, we would like to stop efforts on 
> implementing
> > this in out endpoints.
> > 
> > Best,
> > 
> > 
> > Christian
> > 
> > 
> > _______________________________________________
> > 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: Adam Roach <adam@dynamicsoft.com>
To: "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Mon, 25 Aug 2003 16:13:12 -0500
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Okay, this thread appears to have stalled.

I think everyone has pretty much laid all the cards they
have on the table. To summarize:

1) Implicit [1] subscription:
   - Has correlation problems in the forking case
   - Has technical problems inherent in implicit [1] subscriptions
   - Is fewer messages than an explicit [2] subscription
   - Requires additional mechanism (e.g. HTTP) to
     retrieve KPML scripts.
   - Changes normal protocol machinery by setting up a virgin
     dialog using NOTIFY message
   
2) MESSAGE:
   - Has correlation problems in the forking case
   - Arguably has significantly different semantics
     than is needed for app-interaction
   - Can have detrimental, user-visible effects
   - Requires additional mechanism (e.g. HTTP) to
     retrieve KPML scripts.

3) HTTP GET/POST:
   - Has correlation problems in the forking case
   - May interact poorly with firewalls and NATs

4) NEW_METHOD:
   - No technical objections raised
   - Semantically, has some overlap with NOTIFY
   - Requires additional mechanism (e.g. HTTP) to
     retrieve KPML scripts.

5) Explicit [2] subscription:
   - No technical objections raised
   - Is more complicated than NEW_METHOD
   - Relies on GRUU solution (which should finish
     before app-interaction)

Have I left anything off?

/a

[1] By "implicit", I mean "subscriptions put in place
    via some other mechanism than those described in
    RFC 3265."

[2] By "explicit", I mean "subscriptions put in
    place via the mechanisms described in RFC 3265."

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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Christian Stredicke'" <stredicke@snom.de>,
        "'sipping'"
	 <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
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Pleeeeeease don't start this again ...

I don't think the reasons for overlap dialling, which we discussed for a
looooooong time, very painfully, have changed significantly.

As for end-of-dialling keys on SIP phones, I'd be surprised if anyone on
this list had not seen sipphone.com. You can try selling this kind of thing
in Europe without overlap, but expect reviewers to complain about connection
times. They will not always realise that most of it is the inter-digit
timer.

You can certainly change user behaviour if you explain to them that it works
differently from an old-style phone & if there is something in it for the
user as well. But if it 'looks and feels' just like a phone there will
always be people who expect it to work just like a phone. Service providers
& the market, not IETF, should decide how important those people are.

...Mark

-----Original Message-----
From: Christian Stredicke [mailto:stredicke@snom.de] 
Sent: 26 August 2003 06:37
To: 'sipping'
Subject: [Sipping] Kill Overlap Dialling


Hi All,

the feedback which I received so far on overlap dialling was pretty
clear: There will be no overlap dialling support for SIP.

Arguments:

A. ENUM does not support overlap dialling. There are security concerns on
who will own the nodes of incomplete numbers and it was also reported that
the overhead for the node lookup is too much.

B. Countries with fixed-length numbers (like USA) can have a dial plan on
the phone. Other countries always need to press the dial button except for
cases where extension numbers are dialled. People calling from the US have
to press a dial key (or #) after dialling a non-US number. That is accepted
by the customers.

C. The lack of support of 484 responses on PSTN gateways makes overlap
dialling unusable anyway. A SIP extension would be necessary, but does not
exist at this time and the probability that we reach consensus on such an
extension in time is minimal. 

D. "Grandma" is not in the customer focus any more. "Grandchildren" are used
to cell phones and press the dial button after making up their minds on
which person to call.

E. Phones without display will not be sold in large numbers any more. If
vendors should decide to make SIP phones without display, they must include
a dial button (timeout of 5 seconds to start dialling is not feasible as
this makes printed telephone book lookup impossible).


Am I right? I this is so, we would like to stop efforts on implementing this
in out endpoints.

Best,


Christian


_______________________________________________
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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------_=_NextPart_001_01C36CB6.048B380A
Content-Type: text/html
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] Kill Overlap Dialling</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Pleeeeeease don't start this again ...</FONT>
</P>

<P><FONT SIZE=3D2>I don't think the reasons for overlap dialling, which =
we discussed for a looooooong time, very painfully, have changed =
significantly.</FONT></P>

<P><FONT SIZE=3D2>As for end-of-dialling keys on SIP phones, I'd be =
surprised if anyone on this list had not seen sipphone.com. You can try =
selling this kind of thing in Europe without overlap, but expect =
reviewers to complain about connection times. They will not always =
realise that most of it is the inter-digit timer.</FONT></P>

<P><FONT SIZE=3D2>You can certainly change user behaviour if you =
explain to them that it works differently from an old-style phone &amp; =
if there is something in it for the user as well. But if it 'looks and =
feels' just like a phone there will always be people who expect it to =
work just like a phone. Service providers &amp; the market, not IETF, =
should decide how important those people are.</FONT></P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Christian Stredicke [<A =
HREF=3D"mailto:stredicke@snom.de">mailto:stredicke@snom.de</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 26 August 2003 06:37</FONT>
<BR><FONT SIZE=3D2>To: 'sipping'</FONT>
<BR><FONT SIZE=3D2>Subject: [Sipping] Kill Overlap Dialling</FONT>
</P>
<BR>

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

<P><FONT SIZE=3D2>the feedback which I received so far on overlap =
dialling was pretty</FONT>
<BR><FONT SIZE=3D2>clear: There will be no overlap dialling support for =
SIP.</FONT>
</P>

<P><FONT SIZE=3D2>Arguments:</FONT>
</P>

<P><FONT SIZE=3D2>A. ENUM does not support overlap dialling. There are =
security concerns on who will own the nodes of incomplete numbers and =
it was also reported that the overhead for the node lookup is too =
much.</FONT></P>

<P><FONT SIZE=3D2>B. Countries with fixed-length numbers (like USA) can =
have a dial plan on the phone. Other countries always need to press the =
dial button except for cases where extension numbers are dialled. =
People calling from the US have to press a dial key (or #) after =
dialling a non-US number. That is accepted by the customers.</FONT></P>

<P><FONT SIZE=3D2>C. The lack of support of 484 responses on PSTN =
gateways makes overlap dialling unusable anyway. A SIP extension would =
be necessary, but does not exist at this time and the probability that =
we reach consensus on such an extension in time is minimal. </FONT></P>

<P><FONT SIZE=3D2>D. &quot;Grandma&quot; is not in the customer focus =
any more. &quot;Grandchildren&quot; are used to cell phones and press =
the dial button after making up their minds on which person to =
call.</FONT></P>

<P><FONT SIZE=3D2>E. Phones without display will not be sold in large =
numbers any more. If vendors should decide to make SIP phones without =
display, they must include a dial button (timeout of 5 seconds to start =
dialling is not feasible as this makes printed telephone book lookup =
impossible).</FONT></P>
<BR>

<P><FONT SIZE=3D2>Am I right? I this is so, we would like to stop =
efforts on implementing this in out endpoints.</FONT>
</P>

<P><FONT SIZE=3D2>Best,</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Christian</FONT>
</P>
<BR>

<P><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>

</BODY>
</HTML>
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From: "Jesske, R" <R.Jesske@telekom.de>
To: stredicke@snom.de, sipping@ietf.org
Subject: AW: [Sipping] Kill Overlap Dialling
Date: Wed, 27 Aug 2003 10:51:32 +0200
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> 
> B. Countries with fixed-length numbers (like USA) can have a dial plan
> on the phone. Other countries always need to press the dial button
> except for cases where extension numbers are dialled. People calling
> from the US have to press a dial key (or #) after dialling a non-US
> number. That is accepted by the customers.
> 

> 

Hi,
As you surly know there are several countries that are not using fixed-length numbers (like Germany) and there is also no chance to change it. Also we have no dial button! Therefore there is the need for Overlap. As Adam and Cullen mentioned there is RFC3578. 

And also ITU-T defined the interworking of overlap dialling of SIP with ISUP/BICC networks (Q.1912.SIP). 


Best Regards

Roland Jesske


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38-12
Section T38; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@telekom.de





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Date: Wed, 27 Aug 2003 07:28:17 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Kill Overlap Dialling
In-reply-to: 
 <9BF66EBF6BEFD942915B4D4D45C051F3E863C7@dyn-tx-exch-001.dynamicsoft.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Christian Stredicke'" <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
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> So.... RFC 3578 is a figment of my imagination?

The fact that people took care of mapping ISDN ISUP overlap signaling to SIP
does not mean ISUP is the right role model. Or we would not develop SIP in
the first place and just keep using ISUP...

Henry

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Adam Roach
> Sent: Tuesday, August 26, 2003 12:30 PM
> To: 'Christian Stredicke'; 'sipping'
> Subject: RE: [Sipping] Kill Overlap Dialling
> 
> Christian Stredicke writes:
> 
> > C. The lack of support of 484 responses on PSTN gateways makes overlap
> > dialling unusable anyway. A SIP extension would be necessary, but does
> > not exist at this time
> 
> So.... RFC 3578 is a figment of my imagination?
> 
> /a
> 
> _______________________________________________
> 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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Subject: RE: [Sipping] Kill Overlap Dialling
From: Dean Willis <dean.willis@softarmor.com>
To: Adam Roach <adam@dynamicsoft.com>
Cc: "'Christian Stredicke'" <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
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On Tue, 2003-08-26 at 12:30, Adam Roach wrote:
> Christian Stredicke writes:
> 
> > C. The lack of support of 484 responses on PSTN gateways makes overlap
> > dialling unusable anyway. A SIP extension would be necessary, but does
> > not exist at this time
> 
> So.... RFC 3578 is a figment of my imagination?

Good point, Adam. Can you tell us how many implementations of RFC 3578
were exercised at the last SIPit?

--
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: Adam Roach <adam@dynamicsoft.com>
To: "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] App-Interaction: Considering models
Date: Wed, 27 Aug 2003 13:53:16 -0500
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I wrote:

> Okay, this thread appears to have stalled.

n.b.: I sent this on Monday, when the thread had, indeed,
appeared to have stalled. I guess this particular message
got caught in a mailspool somewhere...

/a

_______________________________________________
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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From: Adam Roach <adam@dynamicsoft.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: "'Christian Stredicke'" <stredicke@snom.de>,
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	 <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
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Dean Willis [mailto:dean.willis@softarmor.com] writes:

> Good point, Adam. Can you tell us how many implementations of RFC 3578
> were exercised at the last SIPit?

That's more of a Robert question; I wasn't there.

/a

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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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Date: Wed, 27 Aug 2003 20:54:03 -0700
Subject: Re: AW: [Sipping] Kill Overlap Dialing
From: Cullen Jennings <fluffy@cisco.com>
To: "Jesske, R" <R.Jesske@telekom.de>, <stredicke@snom.de>, <sipping@ietf.org>
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I've been trying to restrain from adding to the never ending overlap dialing
conversation but ....

On 8/27/03 1:51, "Jesske, R" <R.Jesske@telekom.de> wrote:
> Also we have 
> no dial button! 
I keep seeing variants of this but I would like to point out

1) I have yet to see a phone without a # button

2) somehow or other people from the US PSTN can call Germany on the PSTN.
I'm sure that's impossible reading all the threads on this but oddly it
seems to works. 

Yay, Yah I know that this is less than an ideal solution - please don't
explain why this is an awful solution - I know. If there was an ideal
solution to overlap dialing IETF would have made it an RFC long ago. 


_______________________________________________
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: Christian Stredicke <stredicke@snom.de>, "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
Date: Thu, 28 Aug 2003 08:59:46 +0100
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The conclusion everyone has always reached on this issue in the past is that you can only eliminate overlap dialling when you have control over whether the terminal can overlap dial or not.

Therefore the GSM people eliminated overlap dialling from the handset, and therefore in the network, by insisting that all GSM phones had the user entering the number and then hitting a SEND key. They have not eliminated overlap dialling in the core network, in so far as incomplete numbers can still be received from other networks, which have to be resolved into terminal identifiers before they can be declared to be complete.

Likewise, you can insist on the SIP phone dialling complete, but if you want to interwork with the PSTN, then the problem will be there.

regards

Keith



> -----Original Message-----
> From: Christian Stredicke [mailto:stredicke@snom.de]
> Sent: 26 August 2003 06:37
> To: 'sipping'
> Subject: [Sipping] Kill Overlap Dialling
> 
> 
> Hi All,
> 
> the feedback which I received so far on overlap dialling was pretty
> clear: There will be no overlap dialling support for SIP.
> 
> Arguments:
> 
> A. ENUM does not support overlap dialling. There are security concerns
> on who will own the nodes of incomplete numbers and it was 
> also reported
> that the overhead for the node lookup is too much.
> 
> B. Countries with fixed-length numbers (like USA) can have a dial plan
> on the phone. Other countries always need to press the dial button
> except for cases where extension numbers are dialled. People calling
> from the US have to press a dial key (or #) after dialling a non-US
> number. That is accepted by the customers.
> 
> C. The lack of support of 484 responses on PSTN gateways makes overlap
> dialling unusable anyway. A SIP extension would be necessary, but does
> not exist at this time and the probability that we reach consensus on
> such an extension in time is minimal. 
> 
> D. "Grandma" is not in the customer focus any more. 
> "Grandchildren" are
> used to cell phones and press the dial button after making up their
> minds on which person to call.
> 
> E. Phones without display will not be sold in large numbers 
> any more. If
> vendors should decide to make SIP phones without display, they must
> include a dial button (timeout of 5 seconds to start dialling is not
> feasible as this makes printed telephone book lookup impossible).
> 
> 
> Am I right? I this is so, we would like to stop efforts on 
> implementing
> this in out endpoints.
> 
> Best,
> 
> 
> Christian
> 
> 
> _______________________________________________
> 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: "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Christian Stredicke'"
	 <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
Subject: RE: [Sipping] Kill Overlap Dialling
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Henry,

Of course, if it looks and feels like a mobile phone, there may indeed be
people who expect it to work like a mobile phone. And with mobile phones
there is a big pay-off for the user (mobility) which mitigates various
negative performance aspects (e.g. connection time, voice quality).

But if it looks and feels like a good old-fashioned phone then (some) peo=
ple
may expect it to behave like a good old fashioned phone.

My point is that it is not for us to decide how important these people ar=
e
but just to provide the technology which allows such potential requiremen=
ts
to be met if there is market demand.

You say these people are not important, or can be re-educated, but there =
are
Service Providers who feel (i) these people are important and (ii) the
technical barrier is lower/less important than the re-education barrier.

...Mark

-----Original Message-----
From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]=20
Sent: 28 August 2003 14:25
To: Watson, Mark [MOP:EP10:EXCH]; 'Christian Stredicke'; 'sipping'
Subject: RE: [Sipping] Kill Overlap Dialling


>You can try selling this kind of thing in Europe
>without overlap, but expect reviewers to complain about connection times.

Are they buying mobile phones nevertheless?

Henry

________________________________________
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Mark Watson
Sent: Wednesday, August 27, 2003 11:13 AM
To: 'Christian Stredicke'; 'sipping'
Subject: RE: [Sipping] Kill Overlap Dialling

Pleeeeeease don't start this again ...=20
I don't think the reasons for overlap dialling, which we discussed for a
looooooong time, very painfully, have changed significantly. As for
end-of-dialling keys on SIP phones, I'd be surprised if anyone on this li=
st
had not seen sipphone.com. You can try selling this kind of thing in Euro=
pe
without overlap, but expect reviewers to complain about connection times.
They will not always realise that most of it is the inter-digit timer. Yo=
u
can certainly change user behaviour if you explain to them that it works
differently from an old-style phone & if there is something in it for the
user as well. But if it 'looks and feels' just like a phone there will
always be people who expect it to work just like a phone. Service provide=
rs
& the market, not IETF, should decide how important those people are.
...Mark=20
-----Original Message-----=20
From: Christian Stredicke [mailto:stredicke@snom.de]=20
Sent: 26 August 2003 06:37=20
To: 'sipping'=20
Subject: [Sipping] Kill Overlap Dialling=20

Hi All,=20
the feedback which I received so far on overlap dialling was pretty=20
clear: There will be no overlap dialling support for SIP.=20
Arguments:=20
A. ENUM does not support overlap dialling. There are security concerns on
who will own the nodes of incomplete numbers and it was also reported tha=
t
the overhead for the node lookup is too much. B. Countries with fixed-len=
gth
numbers (like USA) can have a dial plan on the phone. Other countries alw=
ays
need to press the dial button except for cases where extension numbers ar=
e
dialled. People calling from the US have to press a dial key (or #) after
dialling a non-US number. That is accepted by the customers. C. The lack =
of
support of 484 responses on PSTN gateways makes overlap dialling unusable
anyway. A SIP extension would be necessary, but does not exist at this ti=
me
and the probability that we reach consensus on such an extension in time =
is
minimal.=20
D. "Grandma" is not in the customer focus any more. "Grandchildren" are u=
sed
to cell phones and press the dial button after making up their minds on
which person to call. E. Phones without display will not be sold in large
numbers any more. If vendors should decide to make SIP phones without
display, they must include a dial button (timeout of 5 seconds to start
dialling is not feasible as this makes printed telephone book lookup
impossible).

Am I right? I this is so, we would like to stop efforts on implementing t=
his
in out endpoints.=20
Best,=20

Christian=20

_______________________________________________=20
Sipping mailing list=A0 https://www1.ietf.org/mailman/listinfo/sipping=20
This list is for NEW development of the application of SIP=20
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP=20


------_=_NextPart_001_01C36D71.45A11042
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2656.31">
<TITLE>RE: [Sipping] Kill Overlap Dialling</TITLE>
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<BODY>

<P><FONT SIZE=3D2>Henry,</FONT>
</P>

<P><FONT SIZE=3D2>Of course, if it looks and feels like a mobile phone, =
there may indeed be people who expect it to work like a mobile phone. =
And with mobile phones there is a big pay-off for the user (mobility) =
which mitigates various negative performance aspects (e.g. connection =
time, voice quality).</FONT></P>

<P><FONT SIZE=3D2>But if it looks and feels like a good old-fashioned =
phone then (some) people may expect it to behave like a good old =
fashioned phone.</FONT></P>

<P><FONT SIZE=3D2>My point is that it is not for us to decide how =
important these people are but just to provide the technology which =
allows such potential requirements to be met if there is market =
demand.</FONT></P>

<P><FONT SIZE=3D2>You say these people are not important, or can be =
re-educated, but there are Service Providers who feel (i) these people =
are important and (ii) the technical barrier is lower/less important =
than the re-education barrier.</FONT></P>

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

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Henry Sinnreich [<A =
HREF=3D"mailto:Henry.Sinnreich@mci.com">mailto:Henry.Sinnreich@mci.com</=
A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 28 August 2003 14:25</FONT>
<BR><FONT SIZE=3D2>To: Watson, Mark [MOP:EP10:EXCH]; 'Christian =
Stredicke'; 'sipping'</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sipping] Kill Overlap Dialling</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt;You can try selling this kind of thing in =
Europe</FONT>
<BR><FONT SIZE=3D2>&gt;without overlap, but expect reviewers to =
complain about connection times.</FONT>
</P>

<P><FONT SIZE=3D2>Are they buying mobile phones nevertheless?</FONT>
</P>

<P><FONT SIZE=3D2>Henry</FONT>
</P>

<P><FONT SIZE=3D2>________________________________________</FONT>
<BR><FONT SIZE=3D2>From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>=
] On Behalf Of Mark Watson</FONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, August 27, 2003 11:13 AM</FONT>
<BR><FONT SIZE=3D2>To: 'Christian Stredicke'; 'sipping'</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sipping] Kill Overlap Dialling</FONT>
</P>

<P><FONT SIZE=3D2>Pleeeeeease don't start this again ... </FONT>
<BR><FONT SIZE=3D2>I don't think the reasons for overlap dialling, =
which we discussed for a looooooong time, very painfully, have changed =
significantly. As for end-of-dialling keys on SIP phones, I'd be =
surprised if anyone on this list had not seen sipphone.com. You can try =
selling this kind of thing in Europe without overlap, but expect =
reviewers to complain about connection times. They will not always =
realise that most of it is the inter-digit timer. You can certainly =
change user behaviour if you explain to them that it works differently =
from an old-style phone &amp; if there is something in it for the user =
as well. But if it 'looks and feels' just like a phone there will =
always be people who expect it to work just like a phone. Service =
providers &amp; the market, not IETF, should decide how important those =
people are. ...Mark </FONT></P>

<P><FONT SIZE=3D2>-----Original Message----- </FONT>
<BR><FONT SIZE=3D2>From: Christian Stredicke [<A =
HREF=3D"mailto:stredicke@snom.de">mailto:stredicke@snom.de</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: 26 August 2003 06:37 </FONT>
<BR><FONT SIZE=3D2>To: 'sipping' </FONT>
<BR><FONT SIZE=3D2>Subject: [Sipping] Kill Overlap Dialling </FONT>
</P>

<P><FONT SIZE=3D2>Hi All, </FONT>
<BR><FONT SIZE=3D2>the feedback which I received so far on overlap =
dialling was pretty </FONT>
<BR><FONT SIZE=3D2>clear: There will be no overlap dialling support for =
SIP. </FONT>
<BR><FONT SIZE=3D2>Arguments: </FONT>
<BR><FONT SIZE=3D2>A. ENUM does not support overlap dialling. There are =
security concerns on who will own the nodes of incomplete numbers and =
it was also reported that the overhead for the node lookup is too much. =
B. Countries with fixed-length numbers (like USA) can have a dial plan =
on the phone. Other countries always need to press the dial button =
except for cases where extension numbers are dialled. People calling =
from the US have to press a dial key (or #) after dialling a non-US =
number. That is accepted by the customers. C. The lack of support of =
484 responses on PSTN gateways makes overlap dialling unusable anyway. =
A SIP extension would be necessary, but does not exist at this time and =
the probability that we reach consensus on such an extension in time is =
minimal. </FONT></P>

<P><FONT SIZE=3D2>D. &quot;Grandma&quot; is not in the customer focus =
any more. &quot;Grandchildren&quot; are used to cell phones and press =
the dial button after making up their minds on which person to call. E. =
Phones without display will not be sold in large numbers any more. If =
vendors should decide to make SIP phones without display, they must =
include a dial button (timeout of 5 seconds to start dialling is not =
feasible as this makes printed telephone book lookup =
impossible).</FONT></P>

<P><FONT SIZE=3D2>Am I right? I this is so, we would like to stop =
efforts on implementing this in out endpoints. </FONT>
<BR><FONT SIZE=3D2>Best, </FONT>
</P>

<P><FONT SIZE=3D2>Christian </FONT>
</P>

<P><FONT SIZE=3D2>_______________________________________________ =
</FONT>
<BR><FONT SIZE=3D2>Sipping mailing list=A0 <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A> =
</FONT>
<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>

</BODY>
</HTML>
------_=_NextPart_001_01C36D71.45A11042--

_______________________________________________
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: Thu, 28 Aug 2003 08:24:39 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Kill Overlap Dialling
In-reply-to: 
 <A3C2399B2FACD411A54200508BE39C740B4956AA@zwcwd00r.europe.nortel.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Christian Stredicke'" <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
Message-id: <0HKB00B86ZX5FU@dgismtp04.wcomnet.com>
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>You can try selling this kind of thing in Europe=20
>without overlap, but expect reviewers to complain about connection =
times.

Are they buying mobile phones nevertheless?

Henry

________________________________________
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
Mark Watson
Sent: Wednesday, August 27, 2003 11:13 AM
To: 'Christian Stredicke'; 'sipping'
Subject: RE: [Sipping] Kill Overlap Dialling

Pleeeeeease don't start this again ...=20
I don't think the reasons for overlap dialling, which we discussed for a
looooooong time, very painfully, have changed significantly.
As for end-of-dialling keys on SIP phones, I'd be surprised if anyone on
this list had not seen sipphone.com. You can try selling this kind of =
thing
in Europe without overlap, but expect reviewers to complain about =
connection
times. They will not always realise that most of it is the inter-digit
timer.
You can certainly change user behaviour if you explain to them that it =
works
differently from an old-style phone & if there is something in it for =
the
user as well. But if it 'looks and feels' just like a phone there will
always be people who expect it to work just like a phone. Service =
providers
& the market, not IETF, should decide how important those people are.
...Mark=20
-----Original Message-----=20
From: Christian Stredicke [mailto:stredicke@snom.de]=20
Sent: 26 August 2003 06:37=20
To: 'sipping'=20
Subject: [Sipping] Kill Overlap Dialling=20

Hi All,=20
the feedback which I received so far on overlap dialling was pretty=20
clear: There will be no overlap dialling support for SIP.=20
Arguments:=20
A. ENUM does not support overlap dialling. There are security concerns =
on
who will own the nodes of incomplete numbers and it was also reported =
that
the overhead for the node lookup is too much.
B. Countries with fixed-length numbers (like USA) can have a dial plan =
on
the phone. Other countries always need to press the dial button except =
for
cases where extension numbers are dialled. People calling from the US =
have
to press a dial key (or #) after dialling a non-US number. That is =
accepted
by the customers.
C. The lack of support of 484 responses on PSTN gateways makes overlap
dialling unusable anyway. A SIP extension would be necessary, but does =
not
exist at this time and the probability that we reach consensus on such =
an
extension in time is minimal.=20
D. "Grandma" is not in the customer focus any more. "Grandchildren" are =
used
to cell phones and press the dial button after making up their minds on
which person to call.
E. Phones without display will not be sold in large numbers any more. If
vendors should decide to make SIP phones without display, they must =
include
a dial button (timeout of 5 seconds to start dialling is not feasible as
this makes printed telephone book lookup impossible).

Am I right? I this is so, we would like to stop efforts on implementing =
this
in out endpoints.=20
Best,=20

Christian=20

_______________________________________________=20
Sipping mailing list=A0 https://www1.ietf.org/mailman/listinfo/sipping=20
This list is for NEW development of the application of SIP=20
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



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Date: Thu, 28 Aug 2003 09:57:45 -0500
From: Henry Sinnreich <Henry.Sinnreich@mci.com>
Subject: RE: [Sipping] Kill Overlap Dialling
In-reply-to: 
 <A3C2399B2FACD411A54200508BE39C740B495C48@zwcwd00r.europe.nortel.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Christian Stredicke'" <stredicke@snom.de>,
        "'sipping'" <sipping@ietf.org>
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I am not sure why we have to keep the "good old fashioned phone" as a
business target or technological icon. Instead of driving by looking in =
the
rear view mirror, we should rather address all those users gobbling up
mobile phones and integrated enterprise communications and applications,
such as in Microsoft Office 2003 or Siemens Openscape.

Using overlap dialing from the PC/laptop/PDA/smart-phone/mobile phone? =
Would
you?

Excuse me for mentioning a few commercial products...

Henry

________________________________________
From: Mark Watson [mailto:mwatson@nortelnetworks.com]=20
Sent: Thursday, August 28, 2003 9:33 AM
To: 'Henry Sinnreich'; 'Christian Stredicke'; 'sipping'
Subject: RE: [Sipping] Kill Overlap Dialling

Henry,=20
Of course, if it looks and feels like a mobile phone, there may indeed =
be
people who expect it to work like a mobile phone. And with mobile phones
there is a big pay-off for the user (mobility) which mitigates various
negative performance aspects (e.g. connection time, voice quality).
But if it looks and feels like a good old-fashioned phone then (some) =
people
may expect it to behave like a good old fashioned phone.
My point is that it is not for us to decide how important these people =
are
but just to provide the technology which allows such potential =
requirements
to be met if there is market demand.
You say these people are not important, or can be re-educated, but there =
are
Service Providers who feel (i) these people are important and (ii) the
technical barrier is lower/less important than the re-education barrier.
...Mark=20
-----Original Message-----=20
From: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]=20
Sent: 28 August 2003 14:25=20
To: Watson, Mark [MOP:EP10:EXCH]; 'Christian Stredicke'; 'sipping'=20
Subject: RE: [Sipping] Kill Overlap Dialling=20

>You can try selling this kind of thing in Europe=20
>without overlap, but expect reviewers to complain about connection =
times.=20
Are they buying mobile phones nevertheless?=20
Henry=20
________________________________________=20
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
Mark Watson=20
Sent: Wednesday, August 27, 2003 11:13 AM=20
To: 'Christian Stredicke'; 'sipping'=20
Subject: RE: [Sipping] Kill Overlap Dialling=20
Pleeeeeease don't start this again ...=20
I don't think the reasons for overlap dialling, which we discussed for a
looooooong time, very painfully, have changed significantly. As for
end-of-dialling keys on SIP phones, I'd be surprised if anyone on this =
list
had not seen sipphone.com. You can try selling this kind of thing in =
Europe
without overlap, but expect reviewers to complain about connection =
times.
They will not always realise that most of it is the inter-digit timer. =
You
can certainly change user behaviour if you explain to them that it works
differently from an old-style phone & if there is something in it for =
the
user as well. But if it 'looks and feels' just like a phone there will
always be people who expect it to work just like a phone. Service =
providers
& the market, not IETF, should decide how important those people are.
...Mark=20
-----Original Message-----=20
From: Christian Stredicke [mailto:stredicke@snom.de]=20
Sent: 26 August 2003 06:37=20
To: 'sipping'=20
Subject: [Sipping] Kill Overlap Dialling=20
Hi All,=20
the feedback which I received so far on overlap dialling was pretty=20
clear: There will be no overlap dialling support for SIP.=20
Arguments:=20
A. ENUM does not support overlap dialling. There are security concerns =
on
who will own the nodes of incomplete numbers and it was also reported =
that
the overhead for the node lookup is too much. B. Countries with =
fixed-length
numbers (like USA) can have a dial plan on the phone. Other countries =
always
need to press the dial button except for cases where extension numbers =
are
dialled. People calling from the US have to press a dial key (or #) =
after
dialling a non-US number. That is accepted by the customers. C. The lack =
of
support of 484 responses on PSTN gateways makes overlap dialling =
unusable
anyway. A SIP extension would be necessary, but does not exist at this =
time
and the probability that we reach consensus on such an extension in time =
is
minimal.=20
D. "Grandma" is not in the customer focus any more. "Grandchildren" are =
used
to cell phones and press the dial button after making up their minds on
which person to call. E. Phones without display will not be sold in =
large
numbers any more. If vendors should decide to make SIP phones without
display, they must include a dial button (timeout of 5 seconds to start
dialling is not feasible as this makes printed telephone book lookup
impossible).
Am I right? I this is so, we would like to stop efforts on implementing =
this
in out endpoints.=20
Best,=20
Christian=20
_______________________________________________=20
Sipping mailing list=A0 https://www1.ietf.org/mailman/listinfo/sipping=20
This list is for NEW development of the application of SIP=20
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



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Subject: RE: [Sipping] Kill Overlap Dialling
From: Dean Willis <dean.willis@softarmor.com>
To: "Drage, Keith (Keith)" <drage@lucent.com>
Cc: Christian Stredicke <stredicke@snom.de>, "'sipping'" <sipping@ietf.org>
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On Thu, 2003-08-28 at 02:59, Drage, Keith (Keith) wrote:
> The conclusion everyone has always reached on this issue in the past is that you can only eliminate overlap dialling when you have control over whether the terminal can overlap dial or not.
> 
> Therefore the GSM people eliminated overlap dialling from the handset, and therefore in the network, by insisting that all GSM phones had the user entering the number and then hitting a SEND key. They have not eliminated overlap dialling in the core network, in so far as incomplete numbers can still be received from other networks, which have to be resolved into terminal identifiers before they can be declared to be complete.
> 
> Likewise, you can insist on the SIP phone dialling complete, but if you want to interwork with the PSTN, then the problem will be there.
> 

yes, but like GSM, only as an INBOUND problem. I would expect calls
ORIGINATING from SIP nodes to always be en-bloc.

--
Dean


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        "'Christian Stredicke'" <stredicke@snom.de>,
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From: Richard Shockey <richard@shockey.us>
Subject: RE: [Sipping] Kill Overlap Dialling
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At 03:32 PM 8/28/2003 +0100, Mark Watson wrote:

>Henry,
>
>Of course, if it looks and feels like a mobile phone, there may indeed be 
>people who expect it to work like a mobile phone. And with mobile phones 
>there is a big pay-off for the user (mobility) which mitigates various 
>negative performance aspects (e.g. connection time, voice quality).
>
>But if it looks and feels like a good old-fashioned phone then (some) 
>people may expect it to behave like a good old fashioned phone.

yes but SIP also provides ample benefits that can over some marginal 
problems such as a requirement full dial string.


>My point is that it is not for us to decide how important these people are 
>but just to provide the technology which allows such potential 
>requirements to be met if there is market demand.
>
>You say these people are not important, or can be re-educated, but there 
>are Service Providers who feel (i) these people are important and (ii) the 
>technical barrier is lower/less important than the re-education barrier.


Well I can speak to this based on the North American market where every 
single month some NPA has to go from 7 to 10 digit dialing as a result of 
overlay codes.

Hey suprise .they get over it ..they whine a bit ..but the major suprise 
for many of us in the industry was New York City which went to full 10 
digit dialing and guess what NO ONE CARED ..why ..they have a well known 
user experience in the mobile phone industry.


Who among the Service Providers is whining .. Incumbent Carriers looking 
for an excuse not to support VoIP?? The same people whining about QoS  etc.

There is no rational reason to expend one more TCP/IP packet on this subject.


>...Mark


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 703.593.2683,  Fax: +1 815.333.1237
<mailto:richard(at)shockey.us> or <mailto:richard.shockey(at)neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<


_______________________________________________
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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'sipping'" <sipping@ietf.org>
Subject: AW: [Sipping] Kill Overlap Dialling
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Sorry I was a few days offline...

The mobility feature allowed the call phones to have horrible features
like dropping packets, bad voice quality, high prices, etc...

Additionally, if you try overlap dialling on a cell phone you would be
surprised on the "look and feel": while you are still looking at the
handset (entering the number), the other party might be already
connected. That=92s why all intelligent users put the handset as quick =
as
possible to the ear waiting for some tone!

That is different with a "classical" phone: When you dial, you have the
handset already on the ear. Hands free mode mimics that behaviour.=20

Chance that we get overlap dialling into ENUM: zero. Chance that we get
all PSTN gateways to support it: almost zero.=20

It's (maybe) sad but true: overlapped dialling is dead for SIP.


Christian

> -----Urspr=FCngliche Nachricht-----
> Von: Henry Sinnreich [mailto:Henry.Sinnreich@mci.com]
> Gesendet: Donnerstag, 28. August 2003 15:25
> An: 'Mark Watson'; 'Christian Stredicke'; 'sipping'
> Betreff: RE: [Sipping] Kill Overlap Dialling
>=20
> >You can try selling this kind of thing in Europe
> >without overlap, but expect reviewers to complain about connection
times.
>=20
> Are they buying mobile phones nevertheless?
>=20
> Henry
>=20
> ________________________________________
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of
> Mark Watson
> Sent: Wednesday, August 27, 2003 11:13 AM
> To: 'Christian Stredicke'; 'sipping'
> Subject: RE: [Sipping] Kill Overlap Dialling
>=20
> Pleeeeeease don't start this again ...
> I don't think the reasons for overlap dialling, which we discussed for
a
> looooooong time, very painfully, have changed significantly.
> As for end-of-dialling keys on SIP phones, I'd be surprised if anyone
on
> this list had not seen sipphone.com. You can try selling this kind of
> thing
> in Europe without overlap, but expect reviewers to complain about
> connection
> times. They will not always realise that most of it is the inter-digit
> timer.
> You can certainly change user behaviour if you explain to them that it
> works
> differently from an old-style phone & if there is something in it for
the
> user as well. But if it 'looks and feels' just like a phone there will
> always be people who expect it to work just like a phone. Service
> providers
> & the market, not IETF, should decide how important those people are.
> ...Mark
> -----Original Message-----
> From: Christian Stredicke [mailto:stredicke@snom.de]
> Sent: 26 August 2003 06:37
> To: 'sipping'
> Subject: [Sipping] Kill Overlap Dialling
>=20
> Hi All,
> the feedback which I received so far on overlap dialling was pretty
> clear: There will be no overlap dialling support for SIP.
> Arguments:
> A. ENUM does not support overlap dialling. There are security concerns
on
> who will own the nodes of incomplete numbers and it was also reported
that
> the overhead for the node lookup is too much.
> B. Countries with fixed-length numbers (like USA) can have a dial plan
on
> the phone. Other countries always need to press the dial button except
for
> cases where extension numbers are dialled. People calling from the US
have
> to press a dial key (or #) after dialling a non-US number. That is
> accepted
> by the customers.
> C. The lack of support of 484 responses on PSTN gateways makes overlap
> dialling unusable anyway. A SIP extension would be necessary, but does
not
> exist at this time and the probability that we reach consensus on such
an
> extension in time is minimal.
> D. "Grandma" is not in the customer focus any more. "Grandchildren"
are
> used
> to cell phones and press the dial button after making up their minds
on
> which person to call.
> E. Phones without display will not be sold in large numbers any more.
If
> vendors should decide to make SIP phones without display, they must
> include
> a dial button (timeout of 5 seconds to start dialling is not feasible
as
> this makes printed telephone book lookup impossible).
>=20
> Am I right? I this is so, we would like to stop efforts on
implementing
> this
> in out endpoints.
> Best,
>=20
> Christian
>=20
> _______________________________________________
> Sipping mailing list=A0 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
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--MIMEStream=_0+289397_8474885512240_9484709243

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


	Title		: Using SRTP with SIP
	Author(s)	: K. Jain, J. Albert
	Filename	: draft-jain-sipping-srtp-00.txt
	Pages		: 0
	Date		: 2003-8-29
	
The Secured Real Time Protocol (SRTP) is a lightweight protocol that 
secures voice data between two end points. The Session Initiation Protocol (SIP) is a simple and adaptive session-controlling protocol. Efforts have been underway to get these two protocols working together. Several ideas have been presented in different documents in past. This document describes one more way to accomplish this task. The proposed approach is simple and does not consume too much CPU cycles. However, some assumptions need to be made in lack of the specific SIP and SDP fields.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-jain-sipping-srtp-00.txt

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Send a message to:
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	"multipart" MIME messages (i.e. documents which have been split
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	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
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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 exim@www1.ietf.org  Fri Aug 29 19:33:34 2003
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	Fri, 29 Aug 2003 17:25:55 +0200 (MEST)
From: "Christian Stredicke" <stredicke@snom.de>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Dean Willis'" <dean.willis@softarmor.com>
Cc: "'sipping'" <sipping@ietf.org>
Subject: AW: [Sipping] Kill Overlap Dialling
Date: Fri, 29 Aug 2003 17:26:10 +0200
Message-ID: <029b01c36e41$e1393c20$020ba8c0@sip>
MIME-Version: 1.0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
X-Priority: 3 (Normal)
X-MSMail-Priority: Normal
X-Mailer: Microsoft Outlook, Build 10.0.4024
Importance: Normal
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Content-Transfer-Encoding: quoted-printable
Content-Transfer-Encoding: quoted-printable

I saw zero (in total, including SipIt). Maybe I am blind.

CS

> -----Urspr=FCngliche Nachricht-----
> Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] Im Auftrag
von
> Adam Roach
> Gesendet: Mittwoch, 27. August 2003 19:27
> An: 'Dean Willis'; Adam Roach
> Cc: 'Christian Stredicke'; 'sipping'
> Betreff: RE: [Sipping] Kill Overlap Dialling
>=20
> Dean Willis [mailto:dean.willis@softarmor.com] writes:
>=20
> > Good point, Adam. Can you tell us how many implementations of RFC
3578
> > were exercised at the last SIPit?
>=20
> That's more of a Robert question; I wasn't there.
>=20
> /a
>=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


_______________________________________________
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 exim@www1.ietf.org  Fri Aug 29 19:34:12 2003
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To: IETF-Announce: ;
CC: sipping@ietf.org
From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
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--NextPart

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


	Title		: Using SRTP with SIP
	Author(s)	: K. Jain, J. Albert
	Filename	: draft-jain-sipping-srtp-00.txt
	Pages		: 0
	Date		: 2003-8-29
	
The Secured Real Time Protocol (SRTP) is a lightweight protocol that 
secures voice data between two end points. The Session Initiation Protocol (SIP) is a simple and adaptive session-controlling protocol. Efforts have been underway to get these two protocols working together. Several ideas have been presented in different documents in past. This document describes one more way to accomplish this task. The proposed approach is simple and does not consume too much CPU cycles. However, some assumptions need to be made in lack of the specific SIP and SDP fields.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-jain-sipping-srtp-00.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-jain-sipping-srtp-00.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:
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In the body type:
	"FILE /internet-drafts/draft-jain-sipping-srtp-00.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
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--OtherAccess
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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 exim@www1.ietf.org  Sat Aug 30 03:08:48 2003
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	 Fri, 29 Aug 2003 16:30:28 -0600
X-MimeOLE: Produced By Microsoft Exchange V6.0.6470.0
content-class: urn:content-classes:message
MIME-Version: 1.0
Content-Type: multipart/alternative;
	boundary="----_=_NextPart_001_01C36E7D.25D2789F"
Date: Fri, 29 Aug 2003 16:30:28 -0600
Message-ID: <3F75233A2E57CC468B35F3B1FAF71EC013A9A7@idc1exc0004.corp.global.level3.com>
Thread-Topic: [Sip] Proposal for 'Transfer-From' INVITE header.
Thread-Index: AcNuWVg8q/cDcjUnSsy4hxqpSJhrbAAH6b7w
From: "Hearty, John" <John.Hearty@Level3.com>
To: "Greier, Tom" <tom.greier@intel.com>
Cc: <sipping@ietf.org>, <rsparks@dynamicsoft.com>, <alan.johnston@mci.com>
X-OriginalArrivalTime: 29 Aug 2003 22:30:28.0520 (UTC) FILETIME=[26123280:01C36E7D]
Subject: [Sipping] RE: [Sip] Proposal for 'Transfer-From' INVITE header.
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Errors-To: sipping-admin@ietf.org
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This probably belongs on the sipping list rather than sip.  If I
understand your flow correctly, this attended transfer flow is covered
in the cc-transfer draft section 6.  By the way, this draft recently
expired, what's the status Robert or Alan?

=20

This statement seems to be at the heart of the issue:

=20

On a PBX, the 'intent' needs to be known upfront, since the way that the
call is put onhold depends on the intent of the requester (put the call
onhold so another call can be made, or put the call onhold-pend-transfer
to eventually dial a transfer target).

=20

It is not clear to me why the transferee needs to know the intent while
going on hold. The transferee certainly learns the intent of transfer
upon receiving the Refer.  What different actions does the transferee
need to take regarding going on hold prior to transfer vs. regular hold?

=20

John Hearty

Level3

=20

=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of
Greier, Tom
Sent: Friday, August 29, 2003 12:14 PM
To: sip@ietf.org
Cc: Greier, Tom
Subject: [Sip] Proposal for 'Transfer-From' INVITE header.

=20

Hello All,

=20

I am usually a passive reader of this forum, but there has been an issue
that my development group has been struggling with for a while. We have
been using a 'proprietary' header for some time, for which I am hoping
to legitimize via a draft or get information as to existing solutions.

=20

On most PBX and key system digital stations, in order to perform a
supervised transfer of a call, the transferee is put onhold and a new
call is automatically created in the dialtone state. The transfer target
is then dialed on the new dialtone call. If the transferor decides to
complete the transfer, the transferor usually either hangs up or presses
a transfer key to complete the transfer.

=20

This scenario is a two-step transfer process:

=20

1) Put transferee onhold-pend-transfer, which creates new call to be
dialed on to reach transfer destination.

2) Complete transfer.

=20

The two-step transfer process allows the transferor to possibly abort
the transfer, drop the transfer destination and reconnect to the
transferee call.

=20

In implementing a PBX/SIP gateway to PBX/key systems that operate in
this manner, we have a problem with Re-INVITE (hold), INVITE (new
party), REFER/replaces to perform the transfer. When the gateway
receives the Re-INVITE (hold) method, the gateway does not know the
'intent' of the hold request. Is it a regular 'put the call onhold', or
is the requestor intending to later make another outbound call on the
same line and transfer the original call. On a PBX, the 'intent' needs
to be known upfront, since the way that the call is put onhold depends
on the intent of the requester (put the call onhold so another call can
be made, or put the call onhold-pend-transfer to eventually dial a
transfer target).

=20

In order to support this functionality, we have been using our own
'Transfer-From' header. The 'Transfer-From' header is placed in the
INVITE that the requestor sends to dial the transfer destination. The
'Transfer-From' header specifies the call-ID of the transferee call that
the gateway should put onhold-pend-transfer. The destination specified
by the INVITE in the To header is then dialed as the transfer
destination. If the requestor later sends a REFER/replaces specifying
the transferee and the transfer target, then the gateway completes the
transfer. The requestor can also send a BYE or CANCEL on the transfer
target dialog to drop the transfer target and unhold the original
transferee call. Media is then re-established to the original transferee
call upon the gateway receiving a Re-INVITE (unhold) request.

=20

I know that this is very specific to a PBX-gateway device, but is what
we are doing relevant to any other interfaces/implementations? Is this
header viable in any broader sense, is it worth a draft, or is there an
existing standard for this type of functionality?

=20

Thank you for your help.

=20

Tom Greier

Intel Corporation
Network Building Block Division
55 Dodge Rd.

Getzville, NY 14068

=20

Phone: (716) 639-3211
mailto:tom.greier@Intel.com

=20


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>This probably belongs on the =
sipping list
rather than sip.&nbsp; If I understand your flow correctly, this =
attended
transfer flow is covered in the cc-transfer draft section 6.&nbsp; By =
the way,
this draft recently expired, what&#8217;s the status Robert or =
Alan?</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'>&nbsp;</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'>This statement seems to be at the =
heart of
the issue:</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'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>On a PBX, the 'intent' needs to be known upfront, =
since the
way that the call is put onhold depends on the intent of the requester =
(put the
call onhold so another call can be made, or put the call =
onhold-pend-transfer
to eventually dial a transfer target).</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'>&nbsp;</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'>It is not clear to me why the =
transferee
needs to know the intent while going on hold. The transferee certainly =
learns
the intent of transfer upon receiving the Refer. &nbsp;What different =
actions
does the transferee need to take regarding going on hold prior to =
transfer vs.
regular hold?</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'>&nbsp;</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'>John Hearty</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'>Level3</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'>&nbsp;</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'>&nbsp;</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> sip-admin@ietf.org
[mailto:sip-admin@ietf.org] <b><span style=3D'font-weight:bold'>On =
Behalf Of </span></b>Greier,
Tom<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> </span></font><font =
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Friday, August
 29, 2003</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'> </span></font><font size=3D2 face=3DTahoma><span
 style=3D'font-size:10.0pt;font-family:Tahoma'>12:14 =
PM</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> Greier, Tom<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Proposal =
for
'Transfer-From' INVITE header.</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<div>

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

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I am usually a passive reader of this forum, but =
there has
been an issue that my development group has been struggling with for a =
while.
We have been using a 'proprietary' header for some time, for which I am =
hoping
to legitimize via a draft or get information as to existing =
solutions.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>On most PBX and key system digital stations, in order =
to
perform a supervised transfer of a call, the transferee is put onhold =
and a new
call is automatically created in the dialtone state. The transfer target =
is
then dialed on the new dialtone call. If the transferor decides to =
complete the
transfer, the transferor usually either hangs up or presses a transfer =
key to
complete the transfer.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>This scenario is a two-step transfer =
process:</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>1) Put transferee onhold-pend-transfer, which creates =
new
call to be dialed on to reach transfer destination.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>2) Complete transfer.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The two-step transfer process allows the transferor =
to
possibly abort the transfer, drop the transfer destination and reconnect =
to the
transferee call.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In implementing a PBX/SIP gateway to PBX/key systems =
that
operate in this manner, we have a problem with Re-INVITE (hold), INVITE =
(new
party), REFER/replaces to perform the transfer. When the gateway =
receives the
Re-INVITE (hold) method, the gateway does not know the 'intent' of the =
hold
request. Is it a regular 'put the call onhold', or is the requestor =
intending
to later make another outbound call on the same line and transfer the =
original
call. On a PBX, the 'intent' needs to be known upfront, since the way =
that the
call is put onhold depends on the intent of the requester (put the call =
onhold
so another call can be made, or put the call onhold-pend-transfer to =
eventually
dial a transfer target).</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In order to support this functionality, we have been =
using
our own 'Transfer-From' header. The 'Transfer-From' header is placed in =
the
INVITE that the requestor sends to dial the transfer destination.
The&nbsp;'Transfer-From' header specifies the call-ID of the transferee =
call
that the gateway should put onhold-pend-transfer. The destination =
specified by
the INVITE in the To header is then dialed as the transfer destination. =
If the
requestor later sends a REFER/replaces specifying the transferee and the =
transfer
target, then the gateway completes the transfer. The requestor can also =
send a
BYE or CANCEL on the transfer target dialog to drop the transfer target =
and
unhold the original transferee call. Media is then re-established to the
original transferee call upon the gateway receiving a Re-INVITE (unhold)
request.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I know that this is very specific to a PBX-gateway =
device,
but is what we are doing relevant to any other =
interfaces/implementations? Is
this header viable in any broader sense, is it worth a draft, or is =
there an
existing standard for this type of functionality?</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thank you for your help.</span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Intel Corporation<br>
Network Building Block Division<br>
55 Dodge Rd.</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Getzville, NY 14068</span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Phone: (716) 639-3211<br>
<a =
href=3D"mailto:tom.greier@Intel.com">mailto:tom.greier@Intel.com</a></spa=
n></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;</span></font></p>

</div>

</div>

</div>

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=00
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