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From: "ICON'2000 Secretariat" <icon@comp.nus.edu.sg>
To: <Undisclosed.Recipients@leonis.nus.edu.sg>
Subject: 14 days to countdown! Have you submitted your papers???
Date: Tue, 29 Feb 2000 15:06:17 +0800
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Dear All
=20
* Kindly ignore this email if you have received this before.=20
Thanks you for your kind attention.
=20
The 8th IEEE International Conference On Networks will be held from=20

September 5- 8, 2000 in Singapore. The aim of the conference is to =
provide=20

an international forum for experts to promote, share and discuss various =


issues and developments in the broad field of computer and communication =


networks. We thus seek and solicit your contributions

in the form of original/unpublished papers, tutorials, and topics for=20

special sessions/panel discussions. More information on the scope of the =


conference and the guidelines for the submission of contributions can

be obtained at this web site :http://www.comp.nus.edu.sg/~icon/



We look forward to your participation. Thank you.

Icon 2000 organizing Committee




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</STRONG></FONT></FONT></DIV>
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attention.</STRONG></FONT></FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2><FONT size=3D2><FONT size=3D2><FONT =
size=3D2><FONT=20
size=3D2>
<P>The 8th IEEE International Conference On Networks will be held from =
</P>
<P>September 5- 8, 2000 in Singapore. The aim of the conference is to =
provide=20
</P>
<P>an international forum for experts to promote, share and discuss =
various </P>
<P>issues and developments in the broad field of computer and =
communication </P>
<P>networks. We thus seek and solicit your contributions</P>
<P>in the form of original/unpublished papers, tutorials, and topics for =
</P>
<P>special sessions/panel discussions. More information on the scope of =
the </P>
<P>conference and the guidelines for the submission of contributions =
can</P>
<P>be obtained at this web site :http://www.comp.nus.edu.sg/~icon/</P>
<P>&nbsp;</P>
<P>We look forward to your participation. Thank you.</P>
<P>Icon 2000 organizing Committee</P></FONT>
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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar  3 11:07:08 2000
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Date: Fri, 3 Mar 2000 10:52:08 -0500 (EST)
Message-Id: <200003031552.KAA23011@ind.cs.columbia.edu>
From: Jonathan Lennox <lennox@cs.columbia.edu>
To: sip@lists.research.bell-labs.com, iptel@lists.research.bell-labs.com,
        cgi-wg@golux.com
Cc: hgs@cs.columbia.edu, jdrosen@dynamicsoft.com
Subject: Informal Last Call: Common Gateway Interface for SIP
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With the submission of the most recent version of the SIP Common Gateway
Interface, we'd like to announce an informal Last Call on this document (in
place of a Working Group last call, since this draft has no associated
working group).  We'll solicit comments for four weeks, until March 31,
2000.  At that time we'll submit this draft for Informational RFC status.

Comments on the draft should be sent to the SIP mailing list
sip@lists.research.bell-labs.com, or to the authors.



	Title		: Common Gateway Interface for SIP
	Author(s)	: J. Lennox, J. Rosenberg, H. Schulzrinne
	Filename	: draft-lennox-sip-cgi-03.txt,.ps
	Pages		: 38
	Date		: 02-Mar-00
	
In Internet telephony, there must be a means by which new services
are created and deployed rapidly. In the World Wide Web, the Common
Gateway Interface (CGI) has served as popular means towards
programming web services. Due to the similarities between the Session
Initiation Protocol (SIP) and the Hyper Text Transfer Protocol
(HTTP), CGI seems a good candidate for service creation in a SIP
environment. This draft proposes a SIP-CGI interface for providing
SIP services on a SIP server.

URLs for this Internet-Draft are:
http://www.ietf.org/internet-drafts/draft-lennox-sip-cgi-03.txt
http://www.ietf.org/internet-drafts/draft-lennox-sip-cgi-03.ps
http://www.cs.columbia.edu/~lennox/draft-lennox-sip-cgi-03.pdf

The Postscript and PDF versions contain changebars indicating changes from
the previous version.

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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From owner-iptel-outgoing@lists.research.bell-labs.com  Mon Mar  6 04:52:36 2000
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From: "Peter Lewis" <peter.lewis@upperside.fr>
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To: <Undisclosed-Recipient:@smtp2.cluster.oleane.net;>
Subject: SIP 2000 International Conference 
Date: Mon, 6 Mar 2000 10:33:45 +0100
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Unknown and even rejected by many vendors and operators only a few =
months ago, SIP is on the brink of making a definitive name for itself.=20
Visit the SIP 2000 International Conference programme:
http://www.upperside.fr/basip.htm


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ago, SIP is on the brink of making a definitive name for itself. </DIV>
<DIV>Visit the SIP 2000 International Conference programme:</DIV>
<DIV><A=20
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From owner-iptel-outgoing@lists.research.bell-labs.com  Tue Mar  7 09:58:37 2000
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Message-ID: <38C5176B.90A47AF7@indosat.net.id>
Date: Tue, 07 Mar 2000 22:51:23 +0800
From: "Perry Erick R." <equest@indosat.net.id>
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Hi all,
I don't know whether I asked to the right forum, I hope I could find
some pointers here. To my knowledge, SIP is a lightweight protocol
designed for Internet, so ISP could offer telephone service (cmiiw), CPL
is a programming framework to create Internet Telephony Services, such
as IN. The question is: how to reserve resource (i.e. bandwidth) for a
pair of source-destination in this Internet's TCP/IP mode. The PSTN
enojoys a strict 64-kbps (PCM-30) of bandwidth, eventhough there's no
signal transmitted.

To my knowledge also: Telcordia has developed MGCP which include
resource reservation in its packet-based telephone framework. Other
framework offered is AT&T's DOSA (Distributed Open Signaling
Architecture). Both offer a resource reservation process before
communication between source-destination begins.

How do I reserve resource in the IP Tel framework? I believe without a
sturdy resource method, packet-based communication cannot match the PSTN
(unless someone has another, better arguments). The RSVP, as I know it,
is not sturdy enough to handle millions of call simultaneously to match
PSTN's capability.

Thanks for any answers,
--rick
Telkom Indonesia
Network Division


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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 10 02:40:10 2000
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Message-ID: <38C8A521.DBDCCC08@dynamicsoft.com>
Date: Fri, 10 Mar 2000 02:32:49 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

Every once in a while, the question comes up of why SIP REGISTER is not
the right vehicle for allowing gateways to register phone prefixes with
a proxy. I believe TRIP is far better for this purpose. As it turns out,
a small subset of TRIP is needed to solve this problem, and yet its
still interoperable with full blown TRIP. Hussein Salama and I have
written an I-D that discusses this. Until it appears in the archives,
you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 10 10:10:43 2000
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From: "ICON'2000 Secretariat" <icon@comp.nus.edu.sg>
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Date: Thu, 9 Mar 2000 10:39:57 +0800
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Dear All=20
=20
* Kindly ignore this email if you have received this before.=20
Thanks you for your kind attention.
=20
The 8th IEEE International Conference On Networks will be held from=20

September 5- 8, 2000 in Singapore. The aim of the conference is to =
provide=20

an international forum for experts to promote, share and discuss various =


issues and developments in the broad field of computer and communication =


networks. We thus seek and solicit your contributions

in the form of original/unpublished papers, tutorials, and topics for=20

special sessions/panel discussions. More information on the scope of the =


conference and the guidelines for the submission of contributions can

be obtained at this web site :http://www.comp.nus.edu.sg/~icon/



We look forward to your participation. Thank you.

Icon 2000 organizing Committee


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<DIV><FONT color=3D#000000 size=3D2><FONT color=3D#000000 face=3DArial =
size=3D2><STRONG>*=20
Kindly ignore this email if you have received this before.=20
</STRONG></FONT></FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2><FONT color=3D#000000 face=3DArial=20
size=3D2><STRONG>Thanks you for your kind =
attention.</STRONG></FONT></FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2><FONT size=3D2><FONT size=3D2><FONT =
size=3D2><FONT=20
size=3D2>
<P>The 8th IEEE International Conference On Networks will be held from =
</P>
<P>September 5- 8, 2000 in Singapore. The aim of the conference is to =
provide=20
</P>
<P>an international forum for experts to promote, share and discuss =
various </P>
<P>issues and developments in the broad field of computer and =
communication </P>
<P>networks. We thus seek and solicit your contributions</P>
<P>in the form of original/unpublished papers, tutorials, and topics for =
</P>
<P>special sessions/panel discussions. More information on the scope of =
the </P>
<P>conference and the guidelines for the submission of contributions =
can</P>
<P>be obtained at this web site :http://www.comp.nus.edu.sg/~icon/</P>
<P>&nbsp;</P>
<P>We look forward to your participation. Thank you.</P>
<P>Icon 2000 organizing=20
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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 10 15:41:48 2000
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Date: Fri, 10 Mar 2000 14:28:24 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: Interdomain SIP-QoS-OSP
In-reply-to: <38C8A521.DBDCCC08@dynamicsoft.com>
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Until it appears in the archive, there is a new draft version
available at:
http://www.softarmor.com/sipwg/drafts/draft-sinnreich-sip-qos-os
p-01.txt


Title:
Interdomain IP Communications with QoS, Authorization and Usage
Reporting

Authors:
H. Sinnreich, S. Donovan, D. Rawlins, S. Thomas

Abstract

   Commercial grade IP telephony requires linkage between call
setup,
   end-to-end QoS setup, interdomain authorization and
accounting. This
   draft considers the network model for inter-domain QoS for
access and
   transit networks, the service models and policy
implementation
   options. Also, interdomain authorization and accounting may
require
   the trust services of a clearinghouse, to which the local
policy
   server may outsource authorization for support for
inter-domain
   accounting.

   The draft defines two options for QoS support for telephony:
PSTN-
   style "QoS Assured" or Internet-style "QoS Enabled" service.
   Implementing the local policy or QoS deployment can also have
two
   options: The more usual policy "Pull Model" and the policy
"Push
   Model" that may be advantageous for large IP telephony
gateway
   deployments. The draft illustrates the various combinations
of QoS
   service and policy implementation for interdomain QoS with
   authorization and accounting.

Thanks, Henry

Henry Sinnreich
MCI WorldCom
400 International Parkway
Richardson, Texas 75081
USA


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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 10 16:44:21 2000
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From: "Francois Audet" <audet@nortelnetworks.com>
To: sip@lists.research.bell-labs.com, iptel@lists.research.bell-labs.com
Subject: RE: TRIP for gateway registrations
Date: Fri, 10 Mar 2000 13:09:24 -0500
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I would point out that H.323 RAS messages are not normally 
used to register a range of phone numbers for gateways as it 
is quite impractical, especially for gateways handling a large 
number of phone numbers, and when keep-alive messages are used.

While some implementations uses tricks such as registering 
"dialling prefixes" to map to telephone number, it is not the
intent of prefixes which are supposed to be used for special services.

Normally, a gateway registers an alias with its gatekeeper and the binding
between the alias and the range of phone numbers handled by the 
gateway is "manually" configured in the gatekeeper, as described
in the draft. I would think that TRIP could automate this process.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, March 09, 2000 11:33 PM
> To: sip@lists.research.bell-labs.com; 
> iptel@lists.research.bell-labs.com
> Subject: TRIP for gateway registrations
> 
> 
> Folks,
> 
> Every once in a while, the question comes up of why SIP 
> REGISTER is not
> the right vehicle for allowing gateways to register phone 
> prefixes with
> a proxy. I believe TRIP is far better for this purpose. As it 
> turns out,
> a small subset of TRIP is needed to solve this problem, and yet its
> still interoperable with full blown TRIP. Hussein Salama and I have
> written an I-D that discusses this. Until it appears in the archives,
> you can pick up a copy at:
> 
> http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.txt
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 

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charset=3Diso-8859-1">
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<TITLE>RE: TRIP for gateway registrations</TITLE>
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<BODY>

<P><FONT SIZE=3D2>I would point out that H.323 RAS messages are not =
normally </FONT>
<BR><FONT SIZE=3D2>used to register a range of phone numbers for =
gateways as it </FONT>
<BR><FONT SIZE=3D2>is quite impractical, especially for gateways =
handling a large </FONT>
<BR><FONT SIZE=3D2>number of phone numbers, and when keep-alive =
messages are used.</FONT>
</P>

<P><FONT SIZE=3D2>While some implementations uses tricks such as =
registering </FONT>
<BR><FONT SIZE=3D2>&quot;dialling prefixes&quot; to map to telephone =
number, it is not the</FONT>
<BR><FONT SIZE=3D2>intent of prefixes which are supposed to be used for =
special services.</FONT>
</P>

<P><FONT SIZE=3D2>Normally, a gateway registers an alias with its =
gatekeeper and the binding between the alias and the range of phone =
numbers handled by the </FONT></P>

<P><FONT SIZE=3D2>gateway is &quot;manually&quot; configured in the =
gatekeeper, as described</FONT>
<BR><FONT SIZE=3D2>in the draft. I would think that TRIP could automate =
this process.</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Thursday, March 09, 2000 11:33 PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: sip@lists.research.bell-labs.com; </FONT>
<BR><FONT SIZE=3D2>&gt; iptel@lists.research.bell-labs.com</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: TRIP for gateway registrations</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Every once in a while, the question comes up of =
why SIP </FONT>
<BR><FONT SIZE=3D2>&gt; REGISTER is not</FONT>
<BR><FONT SIZE=3D2>&gt; the right vehicle for allowing gateways to =
register phone </FONT>
<BR><FONT SIZE=3D2>&gt; prefixes with</FONT>
<BR><FONT SIZE=3D2>&gt; a proxy. I believe TRIP is far better for this =
purpose. As it </FONT>
<BR><FONT SIZE=3D2>&gt; turns out,</FONT>
<BR><FONT SIZE=3D2>&gt; a small subset of TRIP is needed to solve this =
problem, and yet its</FONT>
<BR><FONT SIZE=3D2>&gt; still interoperable with full blown TRIP. =
Hussein Salama and I have</FONT>
<BR><FONT SIZE=3D2>&gt; written an I-D that discusses this. Until it =
appears in the archives,</FONT>
<BR><FONT SIZE=3D2>&gt; you can pick up a copy at:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-trip-gw-00.t=
xt" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen/papers/draft-rs-tr=
ip-gw-00.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
200 Executive Drive</FONT>
<BR><FONT SIZE=3D2>&gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Suite 120 </FONT>
<BR><FONT SIZE=3D2>&gt; =
dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; West =
Orange, NJ 07052</FONT>
<BR><FONT SIZE=3D2>&gt; =
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; (732) 741-4778</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (732) 741-7244</FONT>
<BR><FONT SIZE=3D2>&gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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From owner-iptel-outgoing@lists.research.bell-labs.com  Mon Mar 13 18:08:01 2000
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Date: Mon, 13 Mar 2000 17:00:24 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Resource Reservation
In-reply-to: <38C5176B.90A47AF7@indosat.net.id>
To: "Perry Erick R." <equest@indosat.net.id>,
        IP Tel <iptel@lists.research.bell-labs.com>
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There is some reference to this problem in the SIP WG "QoS Interworking"
task discussed at:

http://www.softarmor.com/sipwg/tasks/index.html

Of particular interest are:

http://www.softarmor.com/sipwg/drafts/draft-manyfolks-sip-resource-00.txt
http://www.softarmor.com/sipwg/drafts/draft-sinnreich-sip-qos-osp-01.txt
http://www.softarmor.com/sipwg/drafts/draft-gibson-sip-qos-resv-00.txt

and
http://www.softarmor.com/sipwg/drafts/draft-dcsgroup-sip-resource-00.txt

which I think was obsoleted by the first reference above.

The basic approach is that SIP can use whatever IP level resource management
approach the underlying networks support. Efforts have been made to work
with MPLS, RSVP, DiffServ, DOCSYS, and other mechanisms.

In the carrier arena, this looks likely to converge on a two-level model
with RSVP at the edge and DiffServ in the core.

--
Dean


> -----Original Message-----
> From: owner-iptel@lists.research.bell-labs.com
> [mailto:owner-iptel@lists.research.bell-labs.com]On Behalf Of Perry
> Erick R.
> Sent: Tuesday, March 07, 2000 8:51 AM
> To: IP Tel
> Subject: Resource Reservation
>
>
> Hi all,
> I don't know whether I asked to the right forum, I hope I could find
> some pointers here. To my knowledge, SIP is a lightweight protocol
> designed for Internet, so ISP could offer telephone service (cmiiw), CPL
> is a programming framework to create Internet Telephony Services, such
> as IN. The question is: how to reserve resource (i.e. bandwidth) for a
> pair of source-destination in this Internet's TCP/IP mode. The PSTN
> enojoys a strict 64-kbps (PCM-30) of bandwidth, eventhough there's no
> signal transmitted.
>
> To my knowledge also: Telcordia has developed MGCP which include
> resource reservation in its packet-based telephone framework. Other
> framework offered is AT&T's DOSA (Distributed Open Signaling
> Architecture). Both offer a resource reservation process before
> communication between source-destination begins.
>
> How do I reserve resource in the IP Tel framework? I believe without a
> sturdy resource method, packet-based communication cannot match the PSTN
> (unless someone has another, better arguments). The RSVP, as I know it,
> is not sturdy enough to handle millions of call simultaneously to match
> PSTN's capability.
>
> Thanks for any answers,
> --rick
> Telkom Indonesia
> Network Division
>
>
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.
>


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Dear All

A Gentle Reminder

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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed Mar 15 11:26:16 2000
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To: "Liess, Laura" <Laura.Liess@telekom.de>
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Subject: Re: uploading services from the user's  home network to a sip server of 
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"Liess, Laura" wrote:
> 
> Hi,
> 
> after doing a very good job by deciding to take SIP for call control for multimedia services within 3G.IP >networks, the 3G.IP operators have a lot of open items concerning the architecture, one of them being which >network controls the services for a roaming user: the home network, the visited network or shared control. >The concrete question: how can I upload at the begin or during the call the roamimg user's services >available in the home network into the SIP proxy of the visited network? As far as I know, the CPL and >REGISTER helps me to upload services only from the UA to the proxy.

I'm not an expert in wireless, so I'm not certain of the scope of the
services we are talking about. Mid call services, for example, are not
currently within the scope of CPL. However, there is nothing which
prevents the CPL-upload-in-register mechanism from being used by a
server to upload scripts to another server. Registrations, in general,
can be used for third party controls. The From field identifies the
party initiating the registration. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com

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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu Mar 16 00:42:39 2000
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From: "Liess, Laura" <Laura.Liess@telekom.de>
To: sip@lists.research.bell-labs.com, iptel@lists.research.bell-labs.com
Subject: uploading services from the user's  home network to a sip server of the visited network within a 3G.IP environement   
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Hi,

after doing a very good job by deciding to take SIP for call control for multimedia services within 3G.IP networks, the 3G.IP operators have a lot of open items concerning the architecture, one of them being which network controls the services for a roaming user: the home network, the visited network or shared control. The concrete question: how can I upload at the begin or during the call the roamimg user's services available in the home network into the SIP proxy of the visited network? As far as I know, the CPL and REGISTER helps me to upload services only from the UA to the proxy.    

Is anyone else working on similar issues? I know of the Telcordia mobility efforts, but the 3G.IP requirements and architecture seem to be somehow different. The 3G.IP operators would like to reuse their just implemented GPRS and the GPRS mobility.

Thanks a lot
Laura      

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 17 01:42:26 2000
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Agenda for the iptel Working Group
IETF 47 1930-2200, Monday March 27
==================================

1. Agenda Bashing		[Rosenberg]	5 mins
2. Status of CPL & TRIP
   Frameworks			[Rosenberg]	5 mins
3. Framework and Requirements
   for the Internet Intelligent
   Networks  (IIN)              [Slutsman]      15 mins
4. TRIP				[Rosenberg]	45 mins
5. TRIP for Gateway Route
   Exporting			[Rosenberg]	15 mins
   
6. CPL				[Rosenberg]	45 mins

Reading:

draft-lslutsman-sip-iin-framework-00.txt
draft-rs-trip-gw-00.txt
draft-ietf-iptel-cpl-01.txt
draft-ietf-iptel-trip-01.txt

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com

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--NextPart

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

	Title		: CPL: A Language for User Control of Internet Telephony
                          Services
	Author(s)	: J. Lennox, H. Schulzrinne
	Filename	: draft-ietf-iptel-cpl-01.txt,.ps
	Pages		: 41
	Date		: 16-Mar-00
	
The Call Processing Language (CPL) is a language that can be used to
describe and control Internet telephony services. It is designed to
be implementable on either network servers or user agent servers. It
is meant to be simple, extensible, easily edited by graphical
clients, and independent of operating system or signalling protocol.
It is suitable for running on a server where users may not be allowed
to execute arbitrary programs, as it has no variables, loops, or
ability to run external programs.
This document is a product of the IP Telephony (IPTEL) working group
of the Internet Engineering Task Force. Comments are solicited and
should be addressed to the working group's mailing list at
iptel@lists.research.bell-labs.com and/or the authors.

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

Internet-Drafts are also available by anonymous FTP. Login with the username
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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 17 16:45:36 2000
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To: iptel@lists.research.bell-labs.com
From: Richard Shockey <rshockey@ix.netcom.com>
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For those of you interested in how phone numbers really work in the IN...

>To: IETF-Announce: ;
>From: Internet-Drafts@ietf.org
>Reply-to: Internet-Drafts@ietf.org
>Subject: I-D ACTION:draft-foster-e164-gstn-np-00.txt
>Date: Fri, 17 Mar 2000 08:22:10 -0500
>Sender: nsyracus@cnri.reston.va.us
>
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>
>
>         Title           : Number Portability in the GSTN: An Overview
>         Author(s)       : M. Foster, J. Yu, T. McGarry
>         Filename        : draft-foster-e164-gstn-np-00.txt
>         Pages           : 28
>         Date            : 16-Mar-00
>
>This document provides an overview of E.164 telephone number
>portability (NP) in the Global Switched Telephone Network (GSTN).
>There are three types of number portability: service provider
>portability (SPNP), location portability, and service portability.
>Service provider portability, the focus of the present draft, is a
>regulatory imperative in many countries seeking to liberalize local
>telephony service competition, by enabling end-users to retain pre-
>existing telephone numbers while changing service providers.
>Implementation of NP within national GSTN entails potentially
>significant changes to numbering administration, network element
>signaling, call routing and processing, billing, service management,
>and other functions.  NP changes the fundamental nature of a dialed
>E.164 number from a hierarchical physical routing address to a
>virtual address, thereby requiring the transparent translation of
>the later to the former.  In addition, there are various regulatory
>constraints which establish relevant parameters for NP
>implementation, most of which are not network technology specific.
>Consequently, the implementation of NP behavior consistent with
>applicable regulatory constraints, as well as the need for
>interoperation with the existing GSTN NP implementations, are
>relevant topics for numerous areas of IP telephony work-in-progress
>at IETF.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-foster-e164-gstn-np-00.txt
>
>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-foster-e164-gstn-np-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:
>         mailserv@ietf.org.
>In the body type:
>         "FILE /internet-drafts/draft-foster-e164-gstn-np-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.
>Content-Type: text/plain
>Content-ID:     <20000316144422.I-D@ietf.org>
>
>ENCODING mime
>FILE /internet-drafts/draft-foster-e164-gstn-np-00.txt
>
><ftp://ftp.ietf.org/internet-drafts/draft-foster-e164-gstn-np-00.txt>



 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey
Shockey Consulting LLC
8045 Big Bend Blvd. Suite 110
St. Louis, MO 63119
Voice 314.918.9020
eFAX Fax to EMail 815.333.1237 (Preferred for Fax)
INTERNET Mail & IFAX : rshockey@ix.netcom.com
GSTN Fax 314.918.9015
MediaGate iPost VoiceMail and Fax 800.260.4464
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<


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From owner-iptel-outgoing@lists.research.bell-labs.com  Sun Mar 19 23:36:17 2000
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From: KITAGAWA Takurou <kitagawa@kuis.kyoto-u.ac.jp>
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I'd like to talk about a directory service for IP telephony 
using HTTP, in the IPTEL session in 47th IETF meeting.
Can I have a time (about a few minutes) to speak?

-- 
Kitagawa Takurou
Graduate School of Kyoto University
kitagawa@kuis.kyoto-u.ac.jp

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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu Mar 23 23:54:23 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: iptel@lists.research.bell-labs.com
Subject: phone directory server for IP telephony applications
References: <3g8zzenu26.fsf@jog.kuis.kyoto-u.ac.jp>
 <38D5BB36.1A2B7FDE@dynamicsoft.com> <3gu2i1vsk6.fsf@jog.kuis.kyoto-u.ac.jp>
 <38D7036C.C1EDDDF7@dynamicsoft.com>
From: KITAGAWA Takurou <kitagawa@kuis.kyoto-u.ac.jp>
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If I can, I would like to speak about this server in the IPTEL
session in 47th IETF meeting.

---------------------------------------------------------------------------
Abstract

A telephone directory server, which changes a traditional
telephone number into the information that an IP telephony application uses,
is defined.
This server uses HTTP and CGI to communicate with a client.

1. Introduction

When a user employs IP telephony, he or she must specify 
many types of information such as IP address, port number,
protocol for communication, audio codec, and etc.
This specification bothers users very much.
Thus, when a user can specify all information just by one telephone
number, it will save him or her much trouble and IP telephony will
become easier to use.

In this draft, a telephone directory server, which changes a telephone
number given by a user into the information that IP telephony
applications require, is defined.
This server uses HTTP and CGI as a protocol for communication with
a client.

2. Query from client to server

A client accesses to following URI with HTTP, using a GET method.

http://<server>/[path]?<phone number>

<server> is the host name (or IP address) of a telephone directory
server.
A server MAY require [path], to specify a telephone directory in it.
<phone number> is the telephone number given by a user, and is
composed of digit 0-9, *, and/or #.

<server> and [path] in this URI, which are required information to
specify a telephone directory, are set up in a client in advance.

3. Answer from server to client

A server receives a requested telephone number with the CGI scheme.
Then a sever refers to internal database, whether the information
copes with the requested telephone number.

3.1 The case that server found matched information

A server returns HTTP status code 200 (OK) to a client,
and a body part of the answer contains the information.

The information returned to a client depends on a telephony system,
but usually contains:

 o A hostname or IP address of callee
 o A port number of callee
 o A protocol used in calling
 o An audio codec used in calling

The description format of the information is notified of the client by
the Content-Type field.
For example, SDP can be used as a concrete description format,.

Example: 

( C> indicates the data from a client to a server, and S> indicates
the data from a client to a server )

C>GET /?1234 HTTP/1.0
C>
S>200 OK
S>Content-Type: application/sdp
S>
S>v=0
S>o=user@somedomain
S>s=IP telephone call
S>c=IN IP4 somehost.somedomain
S>t=0 0
S>m=audio 10001 RTP/AVP 0

3.2 The case that the server could not find information

A server returns HTTP status code 404 (Not Found) to a client.

Example:

C>GET /?1234 HTTP/1.0
C>
S>404 Not Found
S>

3.3 Timeout

A client waits for an answer from a server during the specified time
after it sent an query.
When a client receives no answer from a server during the time, 
it judges that a server error occurred.

3.5 Other status codes

A server MAY return other status codes to a client if necessary.
When a client receives an HTTP Status Code not described above,
it SHOULD deal it as an usual HTTP user agents does.

4. Security

A telephone directory server MAY authenticate a client in accordance
with the HTTP framework, if necessary.

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

-- 
Kitagawa Takurou
Graduate School of Kyoto University
kitagawa@kuis.kyoto-u.ac.jp

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 24 04:02:11 2000
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Subject: Re: phone directory server for IP telephony applications
References: <38DB0B83.8AC2D38D@dynamicsoft.com>  <3gbt45ou08.fsf_-_@jog.kuis.kyoto-u.ac.jp>
From: KITAGAWA Takurou <kitagawa@kuis.kyoto-u.ac.jp>
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Jonathan Rosenberg <jdrosen@dynamicsoft.com> writes:

> This is simply too late to be considered for a slot.

In a general IETF WG meeting,
a sudden speaker is usually also welcome.

And...

	IETF 47 1930-2200, Monday March 27
	==================================
	
	1. Agenda Bashing		[Rosenberg]	5 mins
	2. Status of CPL & TRIP
	   Frameworks			[Rosenberg]	5 mins
	3. Framework and Requirements
	   for the Internet Intelligent
	   Networks  (IIN)              [Slutsman]      15 mins
	4. TRIP				[Rosenberg]	45 mins
	5. TRIP for Gateway Route
	   Exporting			[Rosenberg]	15 mins
	   
	6. CPL				[Rosenberg]	45 mins

slots still remain apparently.
5 or 10 ten minutes for this topic are enough.

> Please submit as an
> I-D after the meeting if you wish to pursue this. In any case, I believe
> this to be outside of the charter of iptel. 

I belive this theme interests most people attending 
IPTEL WG. Let's discuss this topic openly.

> 
> -Jonathan R.

-- 
Kitagawa Takurou
Graduate School of Kyoto University
kitagawa@kuis.kyoto-u.ac.jp

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From owner-iptel-outgoing@lists.research.bell-labs.com  Tue Mar 28 09:03:56 2000
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From: Tom_Gray@Mitel.COM
To: iptel@lists.research.bell-labs.com, Tom_Gray@Mitel.COM
Message-ID: <852568B0.004CB3CE.00@kanmta01.software.mitel.com>
Date: Tue, 28 Mar 2000 07:51:28 -0500
Subject: Internet Telephony Research Position Available
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3/27/2000
                     Mitel Corporation
                     Position Description

 Position Title:   Senior Researcher
                   Strategic Technology Group
                   Mitel Corporation
                   www.mitel.com


 Mitel Corporation is a leading participant in the
enterprise communications market. It has recently
announced its DIVA architecture that allows for the
creation of Internet based voice and data
applications. Mitel's Strategic Technology group is
now staffing a research position to act as a
participant in a small dynamic team. This successful
candidate willact as an 'independent contributor'
within our new technology team. This senior person
will be given freedom to set his/her own objectives
and workingconditions.

 Education: A Ph.D. or M.Sc. in Computer Science,
Systems Engineering or equivalent.

Knowledge: Research done recently in the group has
included: agent based applications and architectures,
Bluetooth and other mobility technologies, application
of ontologies to policy-based call routing, use case
maps, patterns and other software engineering topics,
and feature interaction detection and resolution. The
ideal candidate should have expertise in one or more
of the following areas: intelligent software agents,
artificial intelligence, telecommunication and
Internet technologies, mobile and distributed
computing concepts (CORBA, RMI, Mobile Agents), and
software systems design concepts (UML, UCM, OO
design), and application ontologies (XML, policies).

Duties: The successful candidate?s primary duties will
include: identifying and introducing external
technology of relevance to Mitel, initiating and
participating in external university research and
sponsoring internal demonstrations.

Abilities: Ability to relate academic research
objectives to Mitel?s needs and thereby contribute to
advanced communication systems. Develop effective
relationships with other research, and development
teams within Mitel. Develop effective relationships
with Mitel?s collaborating organizations. Communicate
results clearly and effectively through publications
or presentations.

Personal Suitability: Must have demonstrated skills as
a team contributor. Must be enthusiastic and
demonstrate effective communication skills. Must be
highly motivated and a self-starter while
participating in team projects. Must be open to
learning new concepts.

Contact:  Tom Gray
          Strategic Technology Group
          350 Legget Dr. P.O. Box 13089
          Kanata, Ontario, K2K 2W7
          Tel: 613 592-2122 x1742
          Fax: 613 592-4784
          Email: Tom_Gray@mitel.com



__________________________________________________
Do You Yahoo!?
Talk to your friends online with Yahoo! Messenger.
http://im.yahoo.com





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Date: Wed, 29 Mar 2000 08:34:19 -0500
From: IPTS 2000 <ipts2000@research.att.com>
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Subject: Initial CFP: IP Telecom Services 2000 Workshop
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[Please accept our apologies for multiple copies of this call.]

-------------  PRELIMINARY CALL FOR PAPERS  ---------------

       IP TELECOM SERVICES WORKSHOP 2000 (IPTS 2000)

                    co-located with
             Fall 2000 Voice on the Net (VON)

                     organized by
                  AT&T Labs Research
                      pulver.com

         Cobb Galleria, Atlanta, Georgia, U.S.A
                  September 11, 2000

            submission deadline: June 16, 2000
         http://www.research.att.com/conf/ipts2000/

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

IPTS 2000 is a new workshop dedicated to the important, emerging field
of IP Telecom Services research.

In contrast to the services available on the PSTN, a new world of
telecom services is possible on an IP-based infrastructure. This world
presents new challenges for service development, deployment and
management. This one-day workshop will serve as a forum for the
dissemination of research relating to these challenges.

The IPTS 2000 Program Committee invites submission of papers on
substantial, original and previously unpublished research in IP
telecom services, including, but not limited to:

- service architectures
  (e.g. AIN, SoftSwitch, JAIN, Parlay, DFC)
- service creation environments
- protocols, standards and interoperability
  (e.g. H.323, SIP, MGCP, Megaco/H.248, etc.)
- service programming languages
- service management - billing, fault tolerance, monitoring
- novel services

We welcome papers describing original research; position papers
describing research interests in the field, work in progress, or
future directions of research; project or system descriptions.


IMPORTANT DATES

June 16, 2000        Submission deadline
July 21, 2000        Notification of acceptance
August 25, 2000      Final version of papers
September 11, 2000   IPTS 2000


PAPER SUBMISSIONS

Papers must describe original, previously unpublished work that has
not been simultaneously submitted for publication elsewhere. They must
be written in English, not exceeding 6 pages including figures, tables
and references in 10-12 point font on 8.5x11-inch (US letter) paper.
The first page must include an abstract of up to 200 words, keywords,
and email address of the corresponding author.

All papers must be submitted electronically unless specifically
approved by the Workshop Co-Chairs. Submissions in PostScript or PDF
format should be sent to mailto:ipts2000@research.att.com.

Authors of accepted papers are expected to present their contribution
at the workshop.  The proceedings will be published in hardcopy and on
the web.


REVIEW PROCEDURE

All submissions will be subject to academic peer review by the IPTS
2000 Program Committee under the chairmanship of the Workshop
Co-Chairs. Each paper will receive three written reviews, and reviews
will be returned to the authors.  The Workshop Co-Chairs have final
authority over the review process and all decisions relating to
acceptance of papers. Review criteria include originality of ideas,
technical soundness, significance of results, and quality of
presentation.


CONFERENCE VENUE AND RELATED EVENTS

IPTS 2000 will be held on September 11, 2000 in Atlanta, Georgia. The
workshop will be co-located with Fall 2000 Voice on the Net (VON). 
Details about Fall 2000 VON can be found at http://pulver.com.

IPTS 2000 is organized by AT&T Labs Research
(http://www.research.att.com) and pulver.com (http://pulver.com).


WORKSHOP CO-CHAIRS

Greg Bond, AT&T Labs Research
Eric Cheung, AT&T Labs Research

PROGRAM COMMITTEE

Mauricio Arango, Sun Microsystems Labs
Joanne Atlee, University of Waterloo
Ralph Blumenthal, Daewoo Telecom
Scott Hoffpauir, BroadSoft
Evan Magill, University of Strathclyde
Peter Mataga, dynamicsoft
Bernie Pagurek, Carleton University
Jonathan Rosenberg, dynamicsoft
Henning Schulzrinne, Columbia University
Greg Utas, Nortel Networks
Pamela Zave, AT&T Labs Research


FURTHER GENERAL INFORMATION

http://www.research.att.com/conf/ipts2000

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri Mar 31 15:48:00 2000
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From: "Eber Mello" <eber@ss8networks.com>
To: <iptel@lists.research.bell-labs.com>
Subject: "Otherwise" as mandatory output node
Date: Fri, 31 Mar 2000 15:43:04 -0800
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Hi Rosenberg,

Would you comment on this?

The current CPL draft (March 10/2000) states that "Otherwise" is an 
optional node on the output list of the "Switch-" nodes; we understand 
that it makes more sense to always be able to find
a way out of a non signalling action node, meaning that a CPL script
would be able to handle all calls given to it. It seems quite simple
to expect a user to specify either to drop ("Reject") or forward ("Proxy")
all other calls not handled by the other output nodes of a "Switch-" node.
This  probably simplifies the script execution and seems safer as gives 
no room for mistakenly performing an action not meant to be executed
(all actions are defined by the latest signalling action node visited 
vis-a-vis the latest request, response or internal event arrived).

If we could go this way we could have the SIP registration define either
"proxy", "redirect" or "script" policy for a user, meaning that this user 
either has services (policy=script) or not (policy=redirect or proxy).

Thanks,

Eber Mello 
SS8 Networks Canada.
www.ss8networks.com
phone: 613-5921418 



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