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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3880

        Title:      Call Processing Language (CPL):
                    A Language for User Control of Internet Telephony
                    Services
        Author(s):  J. Lennox, X. Wu, H. Schulzrinne
        Status:     Standards Track
        Date:       October 2004
        Mailbox:    lennox@cs.columbia.edu, xiaotaow@cs.columbia.edu,
                    schulzrinne@cs.columbia.edu
        Pages:      74
        Characters: 154991
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-iptel-cpl-09.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3880.txt


This document defines the Call Processing Language (CPL), a language
to describe and control Internet telephony services.  It is designed
to be implementable on either network servers or user agents.  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 Working Goup 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
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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Folks,

Here is our agenda for the iptel meeting in D.C. I believe our time slot 
has changed due to the conflict with simple, though this does not yet 
appear to be reflected in the online schedule. Please stay posted 
regarding the specific time.

Are there any additional items folks would like to see covered?

IP Telephony WG (IPTEL)

TUESDAY, November 9, 2004, 1300-1400
====================================

CHAIRS:  Jonathan Rosenberg <jdrosen@cisco.com>
          Cullen Jennings <fluffy@cisco.com>

AGENDA:

  5 min  Agenda Bashing                              Chairs
         Bluesheet, Scribe

20 min  Enum Indicator                     Richard Stastny
         draft-stastny-iptel-tel-enumdi-00.txt
         DNS URL

30 min  Plan for concluding work              Chairs Et.Al.
         draft-ietf-iptel-tgrep-04.txt
         draft-ietf-iptel-tel-np-02.txt
         draft-ietf-iptel-trunk-group-02.txt


Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From iptel-bounces@ietf.org  Tue Nov  2 11:42:19 2004
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Subject: RE: [Iptel] Agenda for iptel meeting
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Thread-Topic: [Iptel] Agenda for iptel meeting
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To: <jdrosen@cisco.com>, <iptel@ietf.org>
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Whatever happened to =
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt

Should we take this to sipping?

/Hisham

> -----Original Message-----
> From: iptel-bounces@ietf.org=20
> [mailto:iptel-bounces@ietf.org]On Behalf Of
> ext Jonathan Rosenberg
> Sent: 02.November.2004 17:23
> To: list iptel
> Subject: [Iptel] Agenda for iptel meeting
>=20
>=20
> Folks,
>=20
> Here is our agenda for the iptel meeting in D.C. I believe=20
> our time slot=20
> has changed due to the conflict with simple, though this does not yet=20
> appear to be reflected in the online schedule. Please stay posted=20
> regarding the specific time.
>=20
> Are there any additional items folks would like to see covered?
>=20
> IP Telephony WG (IPTEL)
>=20
> TUESDAY, November 9, 2004, 1300-1400
> =
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> CHAIRS:  Jonathan Rosenberg <jdrosen@cisco.com>
>           Cullen Jennings <fluffy@cisco.com>
>=20
> AGENDA:
>=20
>   5 min  Agenda Bashing                              Chairs
>          Bluesheet, Scribe
>=20
> 20 min  Enum Indicator                     Richard Stastny
>          draft-stastny-iptel-tel-enumdi-00.txt
>          DNS URL
>=20
> 30 min  Plan for concluding work              Chairs Et.Al.
>          draft-ietf-iptel-tgrep-04.txt
>          draft-ietf-iptel-tel-np-02.txt
>          draft-ietf-iptel-trunk-group-02.txt
>=20
>=20
> Thanks,
> Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20

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This is a good question. I pinged Brian on this, and his view was that 
the plan was to submit to iesg directly; i.e., it didn't require 
becoming an iptel item. I am fine either way; the draft is sufficiently 
short that it's not clear it really matters. In either case, if you have 
comments, please do make them. It is appropriate for those comments to 
be made to the iptel list.

Thanks,
Jonathan R.

hisham.khartabil@nokia.com wrote:

> Whatever happened to http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt
> 
> Should we take this to sipping?
> 
> /Hisham
> 
> 
>>-----Original Message-----
>>From: iptel-bounces@ietf.org 
>>[mailto:iptel-bounces@ietf.org]On Behalf Of
>>ext Jonathan Rosenberg
>>Sent: 02.November.2004 17:23
>>To: list iptel
>>Subject: [Iptel] Agenda for iptel meeting
>>
>>
>>Folks,
>>
>>Here is our agenda for the iptel meeting in D.C. I believe 
>>our time slot 
>>has changed due to the conflict with simple, though this does not yet 
>>appear to be reflected in the online schedule. Please stay posted 
>>regarding the specific time.
>>
>>Are there any additional items folks would like to see covered?
>>
>>IP Telephony WG (IPTEL)
>>
>>TUESDAY, November 9, 2004, 1300-1400
>>====================================
>>
>>CHAIRS:  Jonathan Rosenberg <jdrosen@cisco.com>
>>          Cullen Jennings <fluffy@cisco.com>
>>
>>AGENDA:
>>
>>  5 min  Agenda Bashing                              Chairs
>>         Bluesheet, Scribe
>>
>>20 min  Enum Indicator                     Richard Stastny
>>         draft-stastny-iptel-tel-enumdi-00.txt
>>         DNS URL
>>
>>30 min  Plan for concluding work              Chairs Et.Al.
>>         draft-ietf-iptel-tgrep-04.txt
>>         draft-ietf-iptel-tel-np-02.txt
>>         draft-ietf-iptel-trunk-group-02.txt
>>
>>
>>Thanks,
>>Jonathan R.
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>>Director, Service Provider VoIP Architecture   Parsippany, NJ 
>>07054-2711
>>Cisco Systems
>>jdrosen@cisco.com                              FAX:   (973) 952-5050
>>http://www.jdrosen.net                         PHONE: (973) 952-5000
>>http://www.cisco.com
>>
>>
>>_______________________________________________
>>Iptel mailing list
>>Iptel@ietf.org
>>https://www1.ietf.org/mailman/listinfo/iptel
>>
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

_______________________________________________
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Subject: RE: [Iptel] Agenda for iptel meeting
Date: Thu, 4 Nov 2004 18:06:49 +0200
Message-ID: <5816828233DEFA41807A6CFDFDF2343C3A8C1C@esebe056.ntc.nokia.com>
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Myself, Brian and Richard Stastny have been discussing some issues in =
the draft but I haven't seen an update reflecting the discussion. When =
do we expect to see that version?

Thanks,
Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: 02.November.2004 18:39
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: iptel@ietf.org
> Subject: Re: [Iptel] Agenda for iptel meeting
>=20
>=20
> This is a good question. I pinged Brian on this, and his view=20
> was that=20
> the plan was to submit to iesg directly; i.e., it didn't require=20
> becoming an iptel item. I am fine either way; the draft is=20
> sufficiently=20
> short that it's not clear it really matters. In either case,=20
> if you have=20
> comments, please do make them. It is appropriate for those=20
> comments to=20
> be made to the iptel list.
>=20
> Thanks,
> Jonathan R.
>=20
> hisham.khartabil@nokia.com wrote:
>=20
> > Whatever happened to=20
> http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstri
> ng-00.txt
> >=20
> > Should we take this to sipping?
> >=20
> > /Hisham
> >=20
> >=20
> >>-----Original Message-----
> >>From: iptel-bounces@ietf.org=20
> >>[mailto:iptel-bounces@ietf.org]On Behalf Of
> >>ext Jonathan Rosenberg
> >>Sent: 02.November.2004 17:23
> >>To: list iptel
> >>Subject: [Iptel] Agenda for iptel meeting
> >>
> >>
> >>Folks,
> >>
> >>Here is our agenda for the iptel meeting in D.C. I believe=20
> >>our time slot=20
> >>has changed due to the conflict with simple, though this=20
> does not yet=20
> >>appear to be reflected in the online schedule. Please stay posted=20
> >>regarding the specific time.
> >>
> >>Are there any additional items folks would like to see covered?
> >>
> >>IP Telephony WG (IPTEL)
> >>
> >>TUESDAY, November 9, 2004, 1300-1400
> =
>>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>
> >>CHAIRS:  Jonathan Rosenberg <jdrosen@cisco.com>
> >>          Cullen Jennings <fluffy@cisco.com>
> >>
> >>AGENDA:
> >>
> >>  5 min  Agenda Bashing                              Chairs
> >>         Bluesheet, Scribe
> >>
> >>20 min  Enum Indicator                     Richard Stastny
> >>         draft-stastny-iptel-tel-enumdi-00.txt
> >>         DNS URL
> >>
> >>30 min  Plan for concluding work              Chairs Et.Al.
> >>         draft-ietf-iptel-tgrep-04.txt
> >>         draft-ietf-iptel-tel-np-02.txt
> >>         draft-ietf-iptel-trunk-group-02.txt
> >>
> >>
> >>Thanks,
> >>Jonathan R.
> >>--=20
> >>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> >>Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> >>07054-2711
> >>Cisco Systems
> >>jdrosen@cisco.com                              FAX:   (973) 952-5050
> >>http://www.jdrosen.net                         PHONE: (973) 952-5000
> >>http://www.cisco.com
> >>
> >>
> >>_______________________________________________
> >>Iptel mailing list
> >>Iptel@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/iptel
> >>
> >=20
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20

_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel


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I like this document and have read it so many times that I have lost any
objective ability to catch mistakes in it. Anyways, here are some NITS that
come to mind ....

(PS. If you want me to help convert this to XML2RFC to deal with a bunch of
this, I would be happy to help after IETF)


Needs a table of contents as it is over 15 pages.

Expand first use of TRIP in abstract

In the Security Section might want to include some of the evil things that
can happen if you don't follow security recommendations in 3219 like DOS of
all phone calls on your network, misrouting of calls, stealing calls meant
for carrier A to carrier B, acceleration of heat death of universe, and so
on. 

Move to new boiler plate.

Update Jonathans contact info.

Warning in the introduction that you have to read 3219 for this to make any
sense. 

I don't see the numbers for the new address families.

Section 3.7.1 had duplicate text abut how CarrierIdCode is formed. Could we
put this in just one place to reduce the chance of having it conflict.

Split references into Normative and Informative.

In the IANA section, put something along the lines of ....

Trash the Changes sections.


Update the table at

http://www.iana.org/assignments/trip-parameters

to have 

TRIP Attributes
Code      Attribute                       Reference
-------   --------------------------      ---------
     14   AvailableCircuits               [RFC-AAAA]


TRIP Address Families
Code         Address Family               Reference
-----------  ---------------------------  ---------
          4  TrunkGroup                   [RFC-AAAA]

[Note to RFC Editor - replace RFC-AAAA with this RFC number]



Some spelling typos (or perhaps these are OK and I just don't know the
various ways to spell the work but, for what they are worth)

57c57
<    turn can propogate that routing information within and between
---
>    turn can propagate that routing information within and between
59c59
<    of similarites with the TRIP Protocol. It has similar procedures and
---
>    of similarities with the TRIP Protocol. It has similar procedures and
158c158
<    and/or propogated by the TRIP Location Server.
---
>    and/or propagated by the TRIP Location Server.
171c171
<    UPDATE message to propogate the routes supported. It uses
---
>    UPDATE message to propagate the routes supported. It uses
202c202
<    speaker sends the reachibility informaton in the UPDATE messages. The
---
>    speaker sends the reachibility information in the UPDATE messages. The
252c252
<    MAY be propogated beyond the LS that receives it.
---
>    MAY be propagated beyond the LS that receives it.
687c687
<    When the scope of a CarrierIdCode is known to be unique apriori, the
---
>    When the scope of a CarrierIdCode is known to be unique a priori, the
844c844
<    When the scope of a CarrierIdCode is known to be unique apriori, the
---
>    When the scope of a CarrierIdCode is known to be unique a priori, the
983c983
<    MUST NOT send UPDATEs of more than one of the following catagories
---
>    MUST NOT send UPDATEs of more than one of the following categories
1076c1076
<     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
|
---
>     |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
1130c1130
<     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
|
---
>     |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
1181c1181
<    refered to as Consolidation. In the above example, it is possible
---
>    referred to as Consolidation. In the above example, it is possible
1328c1328
<        to accomodate representation of Country codes in association with
---
>        to accommodate representation of Country codes in association with



idnits says ...

idnits 1.34 (28 Jul 2004)

/Users/fluffy/id/draft-ietf-iptel-tgrep-04.txt:

  The document seems to lack separate sections for Informative/Normative
References

  * The document claims conformance with section 10 of RFC 2026, but uses
    some RFC 3667/3668 boilerplate.  As RFC 3667/3668 replaces section 10 of
    RFC 2026, you should not claim conformance with it if you have changed
    to using RFC 3667/3668 boilerplate.

  Looks like you're using RFC 2026 boilerplate. Better change to RFC
3667/3668.
  The document seems to lack an 2026 Section 10.4(C) Copyright Notice
  -- however, there's a paragraph with a matching beginning. Boilerplate
error?
  The document seems to lack an RFC 2026 Section 10.4(C) Permission Grants
Notice
  The document seems to lack an RFC 2026 Section 10.4(C) Disclaimer
  -- however, there's a paragraph with a matching beginning. Boilerplate
error?
  The document is more than 15 pages and seems to lack a Table of Contents
  There are 39 instances of too long lines in the document,
  -- the longest one being 22 characters in excess of 72.

  Warnings:
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document. 

    Line 325 has weird spacing: '...s to be  deter...'
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    Line 393 has weird spacing: '...ges may  be re...'
    Line 466 has weird spacing: '...nkgroup  for t...'
    Line 584 has weird spacing: '...anaging  that ...'
      ...




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Few little NIT like comments on the draft.... If you want help moving it to
xml2rfc after IETF to help with formatting, I would be glad to do that.


Section 5.2.3 seems a little vague. Do you mean that the rn can be
overloaded with the switch id when the rn occurs in a SIP From header? What
about P-Asserted-Identity?

Examples should have invalid phone number such as 555-1212. CIC should also
be invalid.

Should discuss when comparing things that all visual-separators are removed.


Looks like some formatting issues and has some special characters in it.

Move to new boiler plate.


ID Nits says 

idnits 1.34 (28 Jul 2004)

draft-ietf-iptel-tel-np-02.txt:

  Looks like you're using RFC 2026 boilerplate. Better change to RFC
3667/3668.
  The document seems to add a numbered reference to RFC 2026 to the
boilerplate 
  There are 114 instances of too long lines in the document,
  -- the longest one being 3 characters in excess of 72.
  There are 13 instances of lines with non-ascii characters in the document.

  Warnings:
    The document seems to lack an RFC 2026 Section 10.4(A) Disclaimer
    The document seems to lack an RFC 2026 Section 10.4(B) IPR Disclosure
Invitation
    There are 5 instances of lines with hyphenated line breaks in the
document. 






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Subject: [Iptel] Few comments on tel-enumdi-00
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Two really trivial and irrelevant comments ....

In 3.1, an elements that does not know about this conventions will still end
up doing the dip. That makes me wonder if it should be a "SHOULD NOT"
instead of "MUST NOT" do a ENUM dip. Just a thought, not sure it makes any
difference one way or another.

In the examples, should use invalid phone number and example.com instead of
provider.net.

idnits says "No nits found" :-) Congratulations.




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--------------090706010807000100000004
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Cullen,

Thanks for reviewing it. We will incorporate the suggested changes.

Regards,
-Manjax

Cullen Jennings wrote:

>I like this document and have read it so many times that I have lost any
>objective ability to catch mistakes in it. Anyways, here are some NITS that
>come to mind ....
>
>(PS. If you want me to help convert this to XML2RFC to deal with a bunch of
>this, I would be happy to help after IETF)
>
>
>Needs a table of contents as it is over 15 pages.
>
>Expand first use of TRIP in abstract
>
>In the Security Section might want to include some of the evil things that
>can happen if you don't follow security recommendations in 3219 like DOS of
>all phone calls on your network, misrouting of calls, stealing calls meant
>for carrier A to carrier B, acceleration of heat death of universe, and so
>on. 
>
>Move to new boiler plate.
>
>Update Jonathans contact info.
>
>Warning in the introduction that you have to read 3219 for this to make any
>sense. 
>
>I don't see the numbers for the new address families.
>
>Section 3.7.1 had duplicate text abut how CarrierIdCode is formed. Could we
>put this in just one place to reduce the chance of having it conflict.
>
>Split references into Normative and Informative.
>
>In the IANA section, put something along the lines of ....
>
>Trash the Changes sections.
>
>
>Update the table at
>
>http://www.iana.org/assignments/trip-parameters
>
>to have 
>
>TRIP Attributes
>Code      Attribute                       Reference
>-------   --------------------------      ---------
>     14   AvailableCircuits               [RFC-AAAA]
>
>
>TRIP Address Families
>Code         Address Family               Reference
>-----------  ---------------------------  ---------
>          4  TrunkGroup                   [RFC-AAAA]
>
>[Note to RFC Editor - replace RFC-AAAA with this RFC number]
>
>
>
>Some spelling typos (or perhaps these are OK and I just don't know the
>various ways to spell the work but, for what they are worth)
>
>57c57
><    turn can propogate that routing information within and between
>---
>  
>
>>   turn can propagate that routing information within and between
>>    
>>
>59c59
><    of similarites with the TRIP Protocol. It has similar procedures and
>---
>  
>
>>   of similarities with the TRIP Protocol. It has similar procedures and
>>    
>>
>158c158
><    and/or propogated by the TRIP Location Server.
>---
>  
>
>>   and/or propagated by the TRIP Location Server.
>>    
>>
>171c171
><    UPDATE message to propogate the routes supported. It uses
>---
>  
>
>>   UPDATE message to propagate the routes supported. It uses
>>    
>>
>202c202
><    speaker sends the reachibility informaton in the UPDATE messages. The
>---
>  
>
>>   speaker sends the reachibility information in the UPDATE messages. The
>>    
>>
>252c252
><    MAY be propogated beyond the LS that receives it.
>---
>  
>
>>   MAY be propagated beyond the LS that receives it.
>>    
>>
>687c687
><    When the scope of a CarrierIdCode is known to be unique apriori, the
>---
>  
>
>>   When the scope of a CarrierIdCode is known to be unique a priori, the
>>    
>>
>844c844
><    When the scope of a CarrierIdCode is known to be unique apriori, the
>---
>  
>
>>   When the scope of a CarrierIdCode is known to be unique a priori, the
>>    
>>
>983c983
><    MUST NOT send UPDATEs of more than one of the following catagories
>---
>  
>
>>   MUST NOT send UPDATEs of more than one of the following categories
>>    
>>
>1076c1076
><     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
>|
>---
>  
>
>>    |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
>>    
>>
>1130c1130
><     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
>|
>---
>  
>
>>    |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
>>    
>>
>1181c1181
><    refered to as Consolidation. In the above example, it is possible
>---
>  
>
>>   referred to as Consolidation. In the above example, it is possible
>>    
>>
>1328c1328
><        to accomodate representation of Country codes in association with
>---
>  
>
>>       to accommodate representation of Country codes in association with
>>    
>>
>
>
>
>idnits says ...
>
>idnits 1.34 (28 Jul 2004)
>
>/Users/fluffy/id/draft-ietf-iptel-tgrep-04.txt:
>
>  The document seems to lack separate sections for Informative/Normative
>References
>
>  * The document claims conformance with section 10 of RFC 2026, but uses
>    some RFC 3667/3668 boilerplate.  As RFC 3667/3668 replaces section 10 of
>    RFC 2026, you should not claim conformance with it if you have changed
>    to using RFC 3667/3668 boilerplate.
>
>  Looks like you're using RFC 2026 boilerplate. Better change to RFC
>3667/3668.
>  The document seems to lack an 2026 Section 10.4(C) Copyright Notice
>  -- however, there's a paragraph with a matching beginning. Boilerplate
>error?
>  The document seems to lack an RFC 2026 Section 10.4(C) Permission Grants
>Notice
>  The document seems to lack an RFC 2026 Section 10.4(C) Disclaimer
>  -- however, there's a paragraph with a matching beginning. Boilerplate
>error?
>  The document is more than 15 pages and seems to lack a Table of Contents
>  There are 39 instances of too long lines in the document,
>  -- the longest one being 22 characters in excess of 72.
>
>  Warnings:
>    There are 1 instances of lines with hyphenated line breaks in the
>document. 
>
>    Line 325 has weird spacing: '...s to be  deter...'
>    Line 388 has weird spacing: '...refixes  that...'
>    Line 393 has weird spacing: '...ges may  be re...'
>    Line 466 has weird spacing: '...nkgroup  for t...'
>    Line 584 has weird spacing: '...anaging  that ...'
>      ...
>
>
>
>  
>


--------------090706010807000100000004
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
  <title></title>
</head>
<body>
Cullen,<br>
<br>
Thanks for reviewing it. We will incorporate the suggested changes.<br>
<br>
Regards,<br>
-Manjax<br>
<br>
Cullen Jennings wrote:<br>
<blockquote type="cite" cite="midBDAFF340.186C0%25fluffy@cisco.com">
  <pre wrap="">I like this document and have read it so many times that I have lost any
objective ability to catch mistakes in it. Anyways, here are some NITS that
come to mind ....

(PS. If you want me to help convert this to XML2RFC to deal with a bunch of
this, I would be happy to help after IETF)


Needs a table of contents as it is over 15 pages.

Expand first use of TRIP in abstract

In the Security Section might want to include some of the evil things that
can happen if you don't follow security recommendations in 3219 like DOS of
all phone calls on your network, misrouting of calls, stealing calls meant
for carrier A to carrier B, acceleration of heat death of universe, and so
on. 

Move to new boiler plate.

Update Jonathans contact info.

Warning in the introduction that you have to read 3219 for this to make any
sense. 

I don't see the numbers for the new address families.

Section 3.7.1 had duplicate text abut how CarrierIdCode is formed. Could we
put this in just one place to reduce the chance of having it conflict.

Split references into Normative and Informative.

In the IANA section, put something along the lines of ....

Trash the Changes sections.


Update the table at

<a class="moz-txt-link-freetext" href="http://www.iana.org/assignments/trip-parameters">http://www.iana.org/assignments/trip-parameters</a>

to have 

TRIP Attributes
Code      Attribute                       Reference
-------   --------------------------      ---------
     14   AvailableCircuits               [RFC-AAAA]


TRIP Address Families
Code         Address Family               Reference
-----------  ---------------------------  ---------
          4  TrunkGroup                   [RFC-AAAA]

[Note to RFC Editor - replace RFC-AAAA with this RFC number]



Some spelling typos (or perhaps these are OK and I just don't know the
various ways to spell the work but, for what they are worth)

57c57
&lt;    turn can propogate that routing information within and between
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   turn can propagate that routing information within and between
    </pre>
  </blockquote>
  <pre wrap=""><!---->59c59
&lt;    of similarites with the TRIP Protocol. It has similar procedures and
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   of similarities with the TRIP Protocol. It has similar procedures and
    </pre>
  </blockquote>
  <pre wrap=""><!---->158c158
&lt;    and/or propogated by the TRIP Location Server.
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   and/or propagated by the TRIP Location Server.
    </pre>
  </blockquote>
  <pre wrap=""><!---->171c171
&lt;    UPDATE message to propogate the routes supported. It uses
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   UPDATE message to propagate the routes supported. It uses
    </pre>
  </blockquote>
  <pre wrap=""><!---->202c202
&lt;    speaker sends the reachibility informaton in the UPDATE messages. The
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   speaker sends the reachibility information in the UPDATE messages. The
    </pre>
  </blockquote>
  <pre wrap=""><!---->252c252
&lt;    MAY be propogated beyond the LS that receives it.
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   MAY be propagated beyond the LS that receives it.
    </pre>
  </blockquote>
  <pre wrap=""><!---->687c687
&lt;    When the scope of a CarrierIdCode is known to be unique apriori, the
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   When the scope of a CarrierIdCode is known to be unique a priori, the
    </pre>
  </blockquote>
  <pre wrap=""><!---->844c844
&lt;    When the scope of a CarrierIdCode is known to be unique apriori, the
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   When the scope of a CarrierIdCode is known to be unique a priori, the
    </pre>
  </blockquote>
  <pre wrap=""><!---->983c983
&lt;    MUST NOT send UPDATEs of more than one of the following catagories
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   MUST NOT send UPDATEs of more than one of the following categories
    </pre>
  </blockquote>
  <pre wrap=""><!---->1076c1076
&lt;     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
|
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">    |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
    </pre>
  </blockquote>
  <pre wrap=""><!---->1130c1130
&lt;     |   |  (I-TRIP/   |  |     |t|  |d|  |  Mangement  |-++-+-------| GW
|
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">    |   |  (I-TRIP/   |  |     |t|  |d|  |  Management  |-++-+-------| GW  |
    </pre>
  </blockquote>
  <pre wrap=""><!---->1181c1181
&lt;    refered to as Consolidation. In the above example, it is possible
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">   referred to as Consolidation. In the above example, it is possible
    </pre>
  </blockquote>
  <pre wrap=""><!---->1328c1328
&lt;        to accomodate representation of Country codes in association with
---
  </pre>
  <blockquote type="cite">
    <pre wrap="">       to accommodate representation of Country codes in association with
    </pre>
  </blockquote>
  <pre wrap=""><!---->


idnits says ...

idnits 1.34 (28 Jul 2004)

/Users/fluffy/id/draft-ietf-iptel-tgrep-04.txt:

  The document seems to lack separate sections for Informative/Normative
References

  * The document claims conformance with section 10 of RFC 2026, but uses
    some RFC 3667/3668 boilerplate.  As RFC 3667/3668 replaces section 10 of
    RFC 2026, you should not claim conformance with it if you have changed
    to using RFC 3667/3668 boilerplate.

  Looks like you're using RFC 2026 boilerplate. Better change to RFC
3667/3668.
  The document seems to lack an 2026 Section 10.4(C) Copyright Notice
  -- however, there's a paragraph with a matching beginning. Boilerplate
error?
  The document seems to lack an RFC 2026 Section 10.4(C) Permission Grants
Notice
  The document seems to lack an RFC 2026 Section 10.4(C) Disclaimer
  -- however, there's a paragraph with a matching beginning. Boilerplate
error?
  The document is more than 15 pages and seems to lack a Table of Contents
  There are 39 instances of too long lines in the document,
  -- the longest one being 22 characters in excess of 72.

  Warnings:
    There are 1 instances of lines with hyphenated line breaks in the
document. 

    Line 325 has weird spacing: '...s to be  deter...'
    Line 388 has weird spacing: '...refixes  that...'
    Line 393 has weird spacing: '...ges may  be re...'
    Line 466 has weird spacing: '...nkgroup  for t...'
    Line 584 has weird spacing: '...anaging  that ...'
      ...



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

--------------090706010807000100000004--



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From iptel-bounces@ietf.org  Sun Nov  7 12:32:19 2004
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Cullen,

Thanks for the comments.

Please see the responses inline.

I'll send you a revised (temp) I-D in a separate e-mail to see if the
additions are OK.   

James

------ Forwarded Message
From: Cullen Jennings <fluffy@cisco.com>
Date: Thu, 04 Nov 2004 15:41:14 -0800
To: "iptel@ietf.org" <iptel@ietf.org>, <james.yu@neustar.biiz>
Subject: tel-np-02 comments


Few little NIT like comments on the draft.... If you want help moving it to
xml2rfc after IETF to help with formatting, I would be glad to do that.

[YU]  Yes, please help.


Section 5.2.3 seems a little vague. Do you mean that the rn can be
overloaded with the switch id when the rn occurs in a SIP From header? What
about P-Asserted-Identity?

[YU] The I-D focuses on the tel URI so it does not discuss how the signaling
protocol such as SIP carries the tel URI.  The "rn" goes with the phone
number.  For a caller, it can be used to indicate the originating
switch/caller location.   "From" header is a place so is any header that can
carry the tel URI.   If a GW is using the JIP to formulate the tel URI, then
the location of the caller is asserted and P-Asserted-Identity can include
the "rn" in the tel URI.  I guess that SIP signaling procedures need to
define how to formulate the tel URI in various headers.

Examples should have invalid phone number such as 555-1212. CIC should also
be invalid.

[YU]  Added cases E, F and G in Section 6.

Should discuss when comparing things that all visual-separators are removed.

[YU] Added two paragraphs before Section 5.1 about visual-separators and
about invalid "rn" and "cic" processing.
Looks like some formatting issues and has some special characters in it.

Move to new boiler plate.

[YU] Please help.


ID Nits says 

idnits 1.34 (28 Jul 2004)

draft-ietf-iptel-tel-np-02.txt:

  Looks like you're using RFC 2026 boilerplate. Better change to RFC
3667/3668.
  The document seems to add a numbered reference to RFC 2026 to the
boilerplate 
  There are 114 instances of too long lines in the document,
  -- the longest one being 3 characters in excess of 72.

[YU] Please indicated which ones must be converted to shorter sentences.

  There are 13 instances of lines with non-ascii characters in the document.

[YU]  Please identify specific locations for non-ascii characters.

  Warnings:
    The document seems to lack an RFC 2026 Section 10.4(A) Disclaimer
    The document seems to lack an RFC 2026 Section 10.4(B) IPR Disclosure
Invitation

[YU] Please help to insert when using the new boiler plate.

    There are 5 instances of lines with hyphenated line breaks in the
document. 

[YU]  Please identify specific locations for non-ascii characters.



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I missed the cutoff.  I'll submit the update when the I-D editor opens
again.

Brian

> -----Original Message-----
> From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] On Behalf Of
> hisham.khartabil@nokia.com
> Sent: Thursday, November 04, 2004 11:07 AM
> To: jdrosen@cisco.com
> Cc: iptel@ietf.org
> Subject: RE: [Iptel] Agenda for iptel meeting
> 
> Myself, Brian and Richard Stastny have been discussing some issues in the
> draft but I haven't seen an update reflecting the discussion. When do we
> expect to see that version?
> 
> Thanks,
> Hisham
> 
> > -----Original Message-----
> > From: ext Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > Sent: 02.November.2004 18:39
> > To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> > Cc: iptel@ietf.org
> > Subject: Re: [Iptel] Agenda for iptel meeting
> >
> >
> > This is a good question. I pinged Brian on this, and his view
> > was that
> > the plan was to submit to iesg directly; i.e., it didn't require
> > becoming an iptel item. I am fine either way; the draft is
> > sufficiently
> > short that it's not clear it really matters. In either case,
> > if you have
> > comments, please do make them. It is appropriate for those
> > comments to
> > be made to the iptel list.
> >
> > Thanks,
> > Jonathan R.
> >
> > hisham.khartabil@nokia.com wrote:
> >
> > > Whatever happened to
> > http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstri
> > ng-00.txt
> > >
> > > Should we take this to sipping?
> > >
> > > /Hisham
> > >
> > >
> > >>-----Original Message-----
> > >>From: iptel-bounces@ietf.org
> > >>[mailto:iptel-bounces@ietf.org]On Behalf Of
> > >>ext Jonathan Rosenberg
> > >>Sent: 02.November.2004 17:23
> > >>To: list iptel
> > >>Subject: [Iptel] Agenda for iptel meeting
> > >>
> > >>
> > >>Folks,
> > >>
> > >>Here is our agenda for the iptel meeting in D.C. I believe
> > >>our time slot
> > >>has changed due to the conflict with simple, though this
> > does not yet
> > >>appear to be reflected in the online schedule. Please stay posted
> > >>regarding the specific time.
> > >>
> > >>Are there any additional items folks would like to see covered?
> > >>
> > >>IP Telephony WG (IPTEL)
> > >>
> > >>TUESDAY, November 9, 2004, 1300-1400
> > >>====================================
> > >>
> > >>CHAIRS:  Jonathan Rosenberg <jdrosen@cisco.com>
> > >>          Cullen Jennings <fluffy@cisco.com>
> > >>
> > >>AGENDA:
> > >>
> > >>  5 min  Agenda Bashing                              Chairs
> > >>         Bluesheet, Scribe
> > >>
> > >>20 min  Enum Indicator                     Richard Stastny
> > >>         draft-stastny-iptel-tel-enumdi-00.txt
> > >>         DNS URL
> > >>
> > >>30 min  Plan for concluding work              Chairs Et.Al.
> > >>         draft-ietf-iptel-tgrep-04.txt
> > >>         draft-ietf-iptel-tel-np-02.txt
> > >>         draft-ietf-iptel-trunk-group-02.txt
> > >>
> > >>
> > >>Thanks,
> > >>Jonathan R.
> > >>--
> > >>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > >>Director, Service Provider VoIP Architecture   Parsippany, NJ
> > >>07054-2711
> > >>Cisco Systems
> > >>jdrosen@cisco.com                              FAX:   (973) 952-5050
> > >>http://www.jdrosen.net                         PHONE: (973) 952-5000
> > >>http://www.cisco.com
> > >>
> > >>
> > >>_______________________________________________
> > >>Iptel mailing list
> > >>Iptel@ietf.org
> > >>https://www1.ietf.org/mailman/listinfo/iptel
> > >>
> > >
> > >
> >
> > --
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Director, Service Provider VoIP Architecture   Parsippany, NJ
> > 07054-2711
> > Cisco Systems
> > jdrosen@cisco.com                              FAX:   (973) 952-5050
> > http://www.jdrosen.net                         PHONE: (973) 952-5000
> > http://www.cisco.com
> >
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 




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Subject: [Iptel] getting group consensus on moving forward with our work
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Folks,

At the iptel meeting today, there was consensus in the room on plans for 
moving forward with various drafts. I'd like to verify that consensus by 
asking the list if folks are OK proceeding with the following:

* Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears in 
the archives

* Proceed to wglc on 
http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt 
immediately following the ietf

* adopt 
http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt 
as a work item of iptel

* do not adopt 
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt 
as a work item. The author will take it directly to IESG as an 
individual draft.

* do not adopt the expired tel cpc draft 
(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a 
work item. If folks have a problem that they believe is addressed by 
this work, please bring that forward in the next few weeks.

* tgrep will go to IESG after a revision incorporating comments from 
Cullen and the pending comments from Jonathan

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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Subject: Re: [Iptel] getting group consensus on moving forward with our work
	items
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To: Jonathan Rosenberg <jdrosen@cisco.com>
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i am fine with all these proposed actions.
Dave.

On Nov 9, 2004, at 4:03 PM, Jonathan Rosenberg wrote:

> Folks,
>
> At the iptel meeting today, there was consensus in the room on plans  
> for moving forward with various drafts. I'd like to verify that  
> consensus by asking the list if folks are OK proceeding with the  
> following:
>
> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears  
> in the archives
>
> * Proceed to wglc on  
> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group 
> -02.txt immediately following the ietf
>
> * adopt  
> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi 
> -00.txt as a work item of iptel
>
> * do not adopt  
> http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring 
> -00.txt as a work item. The author will take it directly to IESG as an  
> individual draft.
>
> * do not adopt the expired tel cpc draft  
> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a  
> work item. If folks have a problem that they believe is addressed by  
> this work, please bring that forward in the next few weeks.
>
> * tgrep will go to IESG after a revision incorporating comments from  
> Cullen and the pending comments from Jonathan
>
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ  
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From iptel-bounces@ietf.org  Tue Nov  9 17:48:06 2004
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Subject: RE: [Iptel] getting group consensus on moving forward with our
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Jonathan,

Based on our implementation experience of the trunk-group I-D, I'd
suggest we proceed w/ the I-D.=20

Thanks,
Rajnish

-----Original Message-----
From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] On Behalf
Of Jonathan Rosenberg
Sent: Tuesday, November 09, 2004 4:03 PM
To: list iptel
Subject: [Iptel] getting group consensus on moving forward with our
workitems

Folks,

At the iptel meeting today, there was consensus in the room on plans for
moving forward with various drafts. I'd like to verify that consensus by
asking the list if folks are OK proceeding with the following:

* Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears in
the archives

* Proceed to wglc on
http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt
immediately following the ietf

* adopt
http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.tx
t
as a work item of iptel

* do not adopt
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt
as a work item. The author will take it directly to IESG as an
individual draft.

* do not adopt the expired tel cpc draft
(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a
work item. If folks have a problem that they believe is addressed by
this work, please bring that forward in the next few weeks.

* tgrep will go to IESG after a revision incorporating comments from
Cullen and the pending comments from Jonathan

Thanks,
Jonathan R.
--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel

_______________________________________________
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From iptel-bounces@ietf.org  Tue Nov  9 20:37:08 2004
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I wasn't able to attend IPTEL since it was scheduled MIDCOM, so I appreciate
this summary to the list.   

I'm fine with this proposal.  

I reviewed tgrep about 18 mos. ago and it had suggested various editorial
changes, which have been incorporated and since there doesn't seem to have
been any major content change (per the change summaries), I think the major
change to that doc would be updating the template. 

Regards,
Mary H. Barnes
IP Multimedia Standards Strategy
ESN 444-5432 (972-684-5432)
sip: mbarnes@techtrial.com 


-----Original Message-----
From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] On Behalf Of
Jonathan Rosenberg
Sent: Tuesday, November 09, 2004 3:03 PM
To: list iptel
Subject: [Iptel] getting group consensus on moving forward with our
workitems


Folks,

At the iptel meeting today, there was consensus in the room on plans for 
moving forward with various drafts. I'd like to verify that consensus by 
asking the list if folks are OK proceeding with the following:

* Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears in 
the archives

* Proceed to wglc on 
http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt 
immediately following the ietf

* adopt 
http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt 
as a work item of iptel

* do not adopt 
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt 
as a work item. The author will take it directly to IESG as an 
individual draft.

* do not adopt the expired tel cpc draft 
(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a 
work item. If folks have a problem that they believe is addressed by 
this work, please bring that forward in the next few weeks.

* tgrep will go to IESG after a revision incorporating comments from 
Cullen and the pending comments from Jonathan

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


_______________________________________________
Iptel mailing list
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------_=_NextPart_001_01C4C6C4.9A74D15A
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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2658.2">
<TITLE>RE: [Iptel] getting group consensus on moving forward with our =
workitems</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>I wasn't able to attend IPTEL since it was scheduled =
MIDCOM, so I appreciate this summary to the list.&nbsp;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>I'm fine with this proposal.&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>I reviewed tgrep about 18 mos. ago and it had =
suggested various editorial changes, which have been incorporated and =
since there doesn't seem to have been any major content change (per the =
change summaries), I think the major change to that doc would be =
updating the template. </FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>IP Multimedia Standards Strategy</FONT>
<BR><FONT SIZE=3D2>ESN 444-5432 (972-684-5432)</FONT>
<BR><FONT SIZE=3D2>sip: mbarnes@techtrial.com </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: iptel-bounces@ietf.org [<A =
HREF=3D"mailto:iptel-bounces@ietf.org">mailto:iptel-bounces@ietf.org</A>=
] On Behalf Of Jonathan Rosenberg</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, November 09, 2004 3:03 PM</FONT>
<BR><FONT SIZE=3D2>To: list iptel</FONT>
<BR><FONT SIZE=3D2>Subject: [Iptel] getting group consensus on moving =
forward with our workitems</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Folks,</FONT>
</P>

<P><FONT SIZE=3D2>At the iptel meeting today, there was consensus in =
the room on plans for </FONT>
<BR><FONT SIZE=3D2>moving forward with various drafts. I'd like to =
verify that consensus by </FONT>
<BR><FONT SIZE=3D2>asking the list if folks are OK proceeding with the =
following:</FONT>
</P>

<P><FONT SIZE=3D2>* Proceed to wglc on draft-ietf-iptel-tel-np once a =
revision appears in </FONT>
<BR><FONT SIZE=3D2>the archives</FONT>
</P>

<P><FONT SIZE=3D2>* Proceed to wglc on </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group=
-02.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-iptel-t=
runk-group-02.txt</A> </FONT>
<BR><FONT SIZE=3D2>immediately following the ietf</FONT>
</P>

<P><FONT SIZE=3D2>* adopt </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enum=
di-00.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-stastny-ipte=
l-tel-enumdi-00.txt</A> </FONT>
<BR><FONT SIZE=3D2>as a work item of iptel</FONT>
</P>

<P><FONT SIZE=3D2>* do not adopt </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring=
-00.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-rosen-iptel-=
dialstring-00.txt</A> </FONT>
<BR><FONT SIZE=3D2>as a work item. The author will take it directly to =
IESG as an </FONT>
<BR><FONT SIZE=3D2>individual draft.</FONT>
</P>

<P><FONT SIZE=3D2>* do not adopt the expired tel cpc draft </FONT>
<BR><FONT SIZE=3D2>(<A =
HREF=3D"http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt" =
TARGET=3D"_blank">http://www.watersprings.org/pub/id/draft-mahy-iptel-cp=
c-01.txt</A>) as a </FONT>
<BR><FONT SIZE=3D2>work item. If folks have a problem that they believe =
is addressed by </FONT>
<BR><FONT SIZE=3D2>this work, please bring that forward in the next few =
weeks.</FONT>
</P>

<P><FONT SIZE=3D2>* tgrep will go to IESG after a revision =
incorporating comments from </FONT>
<BR><FONT SIZE=3D2>Cullen and the pending comments from Jonathan</FONT>
</P>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Jonathan R.</FONT>
<BR><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=3D2>Director, Service Provider VoIP =
Architecture&nbsp;&nbsp; Parsippany, NJ 07054-2711</FONT>
<BR><FONT SIZE=3D2>Cisco Systems</FONT>
<BR><FONT =
SIZE=3D2>jdrosen@cisco.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2><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;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) =
952-5000</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.cisco.com" =
TARGET=3D"_blank">http://www.cisco.com</A></FONT>
</P>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Iptel mailing list</FONT>
<BR><FONT SIZE=3D2>Iptel@ietf.org</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/iptel" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/iptel</A></FONT=
>
</P>

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From iptel-bounces@ietf.org  Fri Nov 12 16:21:02 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'list iptel'" <iptel@ietf.org>
Subject: Comments on Trunk Group ID and RE: [Iptel] getting group consensus on
	moving forward with our workitems
Date: Fri, 12 Nov 2004 15:49:26 -0500
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I support the plan to move forward with the trunk group draft. In fact, =
my
company intends to implement it.=20

However, I do believe there exists a technical hole in the draft, which =
I
raised on the list before. That is, since "tgrp" is defined as a tel-uri
parameter, it can not exist on its own without the actual tel-uri =
number. In
the case of originating trunk group (ie. PSTN-to-IP direction), the =
calling
number may be absent. The question is then how to express originating =
trunk
group.

Thanks,

Shan Lu =20

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Jonathan Rosenberg
>Sent: Tuesday, November 09, 2004 4:03 PM
>To: list iptel
>Subject: [Iptel] getting group consensus on moving forward=20
>with our workitems
>
>
>Folks,
>
>At the iptel meeting today, there was consensus in the room on=20
>plans for=20
>moving forward with various drafts. I'd like to verify that=20
>consensus by=20
>asking the list if folks are OK proceeding with the following:
>
>* Proceed to wglc on draft-ietf-iptel-tel-np once a revision=20
>appears in=20
>the archives
>
>* Proceed to wglc on=20
>http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-grou
p-02.txt=20
immediately following the ietf

* adopt=20
http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt=
=20
as a work item of iptel

* do not adopt=20
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt=20
as a work item. The author will take it directly to IESG as an=20
individual draft.

* do not adopt the expired tel cpc draft=20
(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a=20
work item. If folks have a problem that they believe is addressed by=20
this work, please bring that forward in the next few weeks.

* tgrep will go to IESG after a revision incorporating comments from=20
Cullen and the pending comments from Jonathan

Thanks,
Jonathan R.
--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/iptel


From iptel-bounces@ietf.org  Mon Nov 15 05:26:52 2004
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From: Takuya Sawada <tu-sawada@kddi.com>
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Hi,

I am not so happy to hear that cpc work will not move forward.
At least when interworking with ISUP or emulating the legacy PSTN,
cpc parameter seems to me to be useful.

For example, cpc=test paramter in P-Asserted-Identity header may
indicate that it is a test call so that it uses testing routing table
and is free of charge through the connection.
With cpc=priority, it can express it requires the call handling
with priority. (This should be done with R-P header, though)
With cpc=payphone, it may require to apply special charge rate, etc.

The scope of cpc work is probably limited to the legacy concept but
is useful in a certain situation.
Is it possible to move this forward to make an RFC as a working group
item, or as an individual submission?

Regards,
Takuya

> Folks,
> 
> At the iptel meeting today, there was consensus in the room on plans for 
> moving forward with various drafts. I'd like to verify that consensus by 
> asking the list if folks are OK proceeding with the following:
> 
> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears in 
> the archives
> 
> * Proceed to wglc on 
> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt 
> immediately following the ietf
> 
> * adopt 
> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt 
> as a work item of iptel
> 
> * do not adopt 
> http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt 
> as a work item. The author will take it directly to IESG as an 
> individual draft.
> 
> * do not adopt the expired tel cpc draft 
> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a 
> work item. If folks have a problem that they believe is addressed by 
> this work, please bring that forward in the next few weeks.
> 
> * tgrep will go to IESG after a revision incorporating comments from 
> Cullen and the pending comments from Jonathan
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

_______________________________________________
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From iptel-bounces@ietf.org  Mon Nov 15 08:14:14 2004
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From: David R Oran <oran@cisco.com>
Subject: Re: Comments on Trunk Group ID and RE: [Iptel] getting group
	consensus on moving forward with our workitems
Date: Mon, 15 Nov 2004 08:05:26 -0500
To: "Shan Lu" <shanlu@sentito.com>
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Is tel:;tgrp=foo not a valid tel: url?
If not, it ought to be.

On Nov 12, 2004, at 3:49 PM, Shan Lu wrote:

> I support the plan to move forward with the trunk group draft. In  
> fact, my
> company intends to implement it.
>
> However, I do believe there exists a technical hole in the draft,  
> which I
> raised on the list before. That is, since "tgrp" is defined as a  
> tel-uri
> parameter, it can not exist on its own without the actual tel-uri  
> number. In
> the case of originating trunk group (ie. PSTN-to-IP direction), the  
> calling
> number may be absent. The question is then how to express originating  
> trunk
> group.
>
> Thanks,
>
> Shan Lu
>
>> -----Original Message-----
>> From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]
>> On Behalf Of Jonathan Rosenberg
>> Sent: Tuesday, November 09, 2004 4:03 PM
>> To: list iptel
>> Subject: [Iptel] getting group consensus on moving forward
>> with our workitems
>>
>>
>> Folks,
>>
>> At the iptel meeting today, there was consensus in the room on
>> plans for
>> moving forward with various drafts. I'd like to verify that
>> consensus by
>> asking the list if folks are OK proceeding with the following:
>>
>> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision
>> appears in
>> the archives
>>
>> * Proceed to wglc on
>> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-grou
> p-02.txt
> immediately following the ietf
>
> * adopt
> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi 
> -00.txt
> as a work item of iptel
>
> * do not adopt
> http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt
> as a work item. The author will take it directly to IESG as an
> individual draft.
>
> * do not adopt the expired tel cpc draft
> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a
> work item. If folks have a problem that they believe is addressed by
> this work, please bring that forward in the next few weeks.
>
> * tgrep will go to IESG after a revision incorporating comments from
> Cullen and the pending comments from Jonathan
>
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ  
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From iptel-bounces@ietf.org  Mon Nov 15 09:39:29 2004
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Date: Mon, 15 Nov 2004 08:30:57 -0600
From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Wireless Research and Development/Internet Software and Services
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To: Shan Lu <shanlu@sentito.com>
Subject: Re: Comments on Trunk Group ID and RE: [Iptel] getting group consensus
	on	moving forward with our workitems
References: <01ac01c4c8f9$1939d950$eb00000a@SAJAK>
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Shan Lu wrote:

> I support the plan to move forward with the trunk group 
> draft. In fact, my company intends to implement it. 

Shan Lu:

Thanks.

> However, I do believe there exists a technical hole in the 
> draft, which I raised on the list before. That is, since 
> "tgrp" is defined as a tel-uri parameter, it can not exist on 
> its own without the actual tel-uri number. In the case of 
> originating trunk group (ie. PSTN-to-IP direction), the calling
> number may be absent. The question is then how to express 
> originating trunk group.

I am trying to understand the question better: Is it a
requirement in your network that the trunk group be
preserved even if the calling party number was absent?
Furthermore, how pervasive is this situation in general?

Clearly, if it is not, then we do the most minimal on
this (probably adding some text on what to do if the calling
party number is absent).  If it is a pervasive situation,
we can spend some capital trying to figure out a way to
combat this.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From iptel-bounces@ietf.org  Mon Nov 15 10:13:58 2004
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To: "'list iptel'" <iptel@ietf.org>
Subject: RE: Comments on Trunk Group ID and RE: [Iptel] getting groupconsensus
	on moving forward with our workitems
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Dave,

Based on 2806bis-09, tel:;tgrp=foo is not valid. I am on the fence whether
it ought to be.

Thanks, Shan Lu

>From draft-ietf-iptel-rfc2806bis-09:

   telephone-uri        = "tel:" telephone-subscriber
   telephone-subscriber = global-number / local-number
   global-number        = global-number-digits *par
   local-number         = local-number-digits *par context *par
   <snip>
   global-number-digits = "+" 1*phonedigit
   local-number-digits  = 1*phonedigit-hex

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of David R Oran
>Sent: Monday, November 15, 2004 8:05 AM
>To: Shan Lu
>Cc: 'list iptel'
>Subject: Re: Comments on Trunk Group ID and RE: [Iptel] 
>getting groupconsensus on moving forward with our workitems
>
>
>Is tel:;tgrp=foo not a valid tel: url?
>If not, it ought to be.
>
>On Nov 12, 2004, at 3:49 PM, Shan Lu wrote:
>
>> I support the plan to move forward with the trunk group draft. In  
>> fact, my
>> company intends to implement it.
>>
>> However, I do believe there exists a technical hole in the draft,  
>> which I
>> raised on the list before. That is, since "tgrp" is defined as a  
>> tel-uri
>> parameter, it can not exist on its own without the actual tel-uri  
>> number. In
>> the case of originating trunk group (ie. PSTN-to-IP direction), the  
>> calling
>> number may be absent. The question is then how to express 
>originating  
>> trunk
>> group.
>>
>> Thanks,
>>
>> Shan Lu
>>
>>> -----Original Message-----
>>> From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]
>>> On Behalf Of Jonathan Rosenberg
>>> Sent: Tuesday, November 09, 2004 4:03 PM
>>> To: list iptel
>>> Subject: [Iptel] getting group consensus on moving forward
>>> with our workitems
>>>
>>>
>>> Folks,
>>>
>>> At the iptel meeting today, there was consensus in the room on
>>> plans for
>>> moving forward with various drafts. I'd like to verify that
>>> consensus by
>>> asking the list if folks are OK proceeding with the following:
>>>
>>> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision
>>> appears in
>>> the archives
>>>
>>> * Proceed to wglc on
>>> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-grou
>> p-02.txt
>> immediately following the ietf
>>
>> * adopt
>> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi 
>> -00.txt
>> as a work item of iptel
>>
>> * do not adopt
>> 
>http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt
>> as a work item. The author will take it directly to IESG as an
>> individual draft.
>>
>> * do not adopt the expired tel cpc draft
>> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a
>> work item. If folks have a problem that they believe is addressed by
>> this work, please bring that forward in the next few weeks.
>>
>> * tgrep will go to IESG after a revision incorporating comments from
>> Cullen and the pending comments from Jonathan
>>
>> Thanks,
>> Jonathan R.
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Director, Service Provider VoIP Architecture   Parsippany, NJ  
>> 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.com
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>David R. Oran
>Cisco Fellow
>Cisco Systems
>7 Ladyslipper Lane
>Acton, MA 01720 USA
>Tel: +1 978 264 2048
>Email: oran@cisco.com
>
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel
>


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From iptel-bounces@ietf.org  Mon Nov 15 10:50:24 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Subject: RE: Comments on Trunk Group ID and RE: [Iptel] getting group
	consensuson moving forward with our workitems
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Vijay,

Thanks for your response.

As you are aware, there are two scenarios where the originating number =
may
be absent. PSTN may withhold the number or the PSTN-SIP gateway may not
entrust downstrem endpoints with the number in case of calling party
privacy.

We have seen both cases and the need to know the incoming PSTN trunk =
group
when calling number is missing. In a lot applications, originating PSTN
trunk group info can be very useful for routing and revealing such info =
is
often not deemed to be a problem for privacy.

I am happy with the draft as far as destination TG is concerned. But I =
don't
think it should prescribe a formula for originating TG whose validity is
sometimes questionable.

Thanks, Shan Lu=20

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Vijay K. Gurbani
>Sent: Monday, November 15, 2004 9:31 AM
>To: Shan Lu
>Cc: 'list iptel'
>Subject: Re: Comments on Trunk Group ID and RE: [Iptel]=20
>getting group consensuson moving forward with our workitems
>
>
>Shan Lu wrote:
>
>> I support the plan to move forward with the trunk group=20
>> draft. In fact, my company intends to implement it.=20
>
>Shan Lu:
>
>Thanks.
>
>> However, I do believe there exists a technical hole in the=20
>> draft, which I raised on the list before. That is, since=20
>> "tgrp" is defined as a tel-uri parameter, it can not exist on=20
>> its own without the actual tel-uri number. In the case of=20
>> originating trunk group (ie. PSTN-to-IP direction), the calling
>> number may be absent. The question is then how to express=20
>> originating trunk group.
>
>I am trying to understand the question better: Is it a
>requirement in your network that the trunk group be
>preserved even if the calling party number was absent?
>Furthermore, how pervasive is this situation in general?
>
>Clearly, if it is not, then we do the most minimal on
>this (probably adding some text on what to do if the calling
>party number is absent).  If it is a pervasive situation,
>we can spend some capital trying to figure out a way to
>combat this.
>
>Thanks,
>
>- vijay
>--=20
>Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
>Wireless Networks Group/Internet Software and Services
>Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
>Naperville, Illinois 60566     Voice: +1 630 224 0216
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel
>


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From: "Shan Lu" <shanlu@sentito.com>
To: "'list iptel'" <iptel@ietf.org>
Subject: RE: Comments on Trunk Group ID and RE: [Iptel] getting
	groupconsensuson moving forward with our workitems
Date: Mon, 15 Nov 2004 10:52:20 -0500
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Gone thru 2806bis-09 one more time, global-number-digits and
local-number-digits are defined as:

global-number-digits = "+" 1*phonedigit
local-number-digits  = 1*phonedigit-hex
<snip>
phonedigit           = DIGIT / [ visual-separator ]
phonedigit-hex       = HEXDIG / "*" / "#" / [ visual-separator ]
visual-separator     = "-" / "." / "(" / ")"

So would this be valid: "tel:-;tgrp=foo"?

Thanks, Shan Lu

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of Shan Lu
>Sent: Monday, November 15, 2004 10:09 AM
>To: 'list iptel'
>Subject: RE: Comments on Trunk Group ID and RE: [Iptel] 
>getting groupconsensuson moving forward with our workitems
>
>
>Dave,
>
>Based on 2806bis-09, tel:;tgrp=foo is not valid. I am on the 
>fence whether
>it ought to be.
>
>Thanks, Shan Lu
>
>>From draft-ietf-iptel-rfc2806bis-09:
>
>   telephone-uri        = "tel:" telephone-subscriber
>   telephone-subscriber = global-number / local-number
>   global-number        = global-number-digits *par
>   local-number         = local-number-digits *par context *par
>   <snip>
>   global-number-digits = "+" 1*phonedigit
>   local-number-digits  = 1*phonedigit-hex
>
>>-----Original Message-----
>>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>>On Behalf Of David R Oran
>>Sent: Monday, November 15, 2004 8:05 AM
>>To: Shan Lu
>>Cc: 'list iptel'
>>Subject: Re: Comments on Trunk Group ID and RE: [Iptel] 
>>getting groupconsensus on moving forward with our workitems
>>
>>
>>Is tel:;tgrp=foo not a valid tel: url?
>>If not, it ought to be.
>>
>>On Nov 12, 2004, at 3:49 PM, Shan Lu wrote:
>>
>>> I support the plan to move forward with the trunk group draft. In  
>>> fact, my
>>> company intends to implement it.
>>>
>>> However, I do believe there exists a technical hole in the draft,  
>>> which I
>>> raised on the list before. That is, since "tgrp" is defined as a  
>>> tel-uri
>>> parameter, it can not exist on its own without the actual tel-uri  
>>> number. In
>>> the case of originating trunk group (ie. PSTN-to-IP 
>direction), the  
>>> calling
>>> number may be absent. The question is then how to express 
>>originating  
>>> trunk
>>> group.
>>>
>>> Thanks,
>>>
>>> Shan Lu
>>>
>>>> -----Original Message-----
>>>> From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]
>>>> On Behalf Of Jonathan Rosenberg
>>>> Sent: Tuesday, November 09, 2004 4:03 PM
>>>> To: list iptel
>>>> Subject: [Iptel] getting group consensus on moving forward
>>>> with our workitems
>>>>
>>>>
>>>> Folks,
>>>>
>>>> At the iptel meeting today, there was consensus in the room on
>>>> plans for
>>>> moving forward with various drafts. I'd like to verify that
>>>> consensus by
>>>> asking the list if folks are OK proceeding with the following:
>>>>
>>>> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision
>>>> appears in
>>>> the archives
>>>>
>>>> * Proceed to wglc on
>>>> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-grou
>>> p-02.txt
>>> immediately following the ietf
>>>
>>> * adopt
>>> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi 
>>> -00.txt
>>> as a work item of iptel
>>>
>>> * do not adopt
>>> 
>>http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstri
>ng-00.txt
>>> as a work item. The author will take it directly to IESG as an
>>> individual draft.
>>>
>>> * do not adopt the expired tel cpc draft
>>> 
>(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a
>>> work item. If folks have a problem that they believe is addressed by
>>> this work, please bring that forward in the next few weeks.
>>>
>>> * tgrep will go to IESG after a revision incorporating comments from
>>> Cullen and the pending comments from Jonathan
>>>
>>> Thanks,
>>> Jonathan R.
>>> -- 
>>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>>> Director, Service Provider VoIP Architecture   Parsippany, NJ  
>>> 07054-2711
>>> Cisco Systems
>>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>>> http://www.cisco.com
>>>
>>>
>>> _______________________________________________
>>> Iptel mailing list
>>> Iptel@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>
>>>
>>> _______________________________________________
>>> Iptel mailing list
>>> Iptel@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>
>>David R. Oran
>>Cisco Fellow
>>Cisco Systems
>>7 Ladyslipper Lane
>>Acton, MA 01720 USA
>>Tel: +1 978 264 2048
>>Email: oran@cisco.com
>>
>>
>>_______________________________________________
>>Iptel mailing list
>>Iptel@ietf.org
>>https://www1.ietf.org/mailman/listinfo/iptel
>>
>
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel
>


_______________________________________________
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From iptel-bounces@ietf.org  Mon Nov 15 21:54:28 2004
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To: shanlu@sentito.com, iptel@ietf.org
Subject: Re: Comments on Trunk Group ID and RE: [Iptel]
	gettinggroupconsensuson moving forward with our workitems
From: Takuya Sawada <tu-sawada@kddi.com>
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	<027801c4cb2b$17201fb0$eb00000a@SAJAK>
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Hi,

As per ABNF in 2806bis, local-number format MUST have phone-context
parameter. So, for example,

tel:-;phone-context=gateway.example.com;tgrp=foo

conforms to the rule in 2806bis.
You can reserve any number format for void-number, "-" or "####"
or whatever, under an administrative domain, e.g. "gateway.example.com".

I think this is an acceptable solution because tgrp draft says in 4.2;
"The use of the "phone-context" parameter in conjunction with this
draft is mandatory".
If we still want global void-number format for tel URI to indicate
tel-related properties without number, we need some more
work in 2806bis.
(This can be used with cpc parameter, if adopted.)

Regards,
Takuya Sawada


> Gone thru 2806bis-09 one more time, global-number-digits and
> local-number-digits are defined as:
> 
> global-number-digits = "+" 1*phonedigit
> local-number-digits  = 1*phonedigit-hex
> <snip>
> phonedigit           = DIGIT / [ visual-separator ]
> phonedigit-hex       = HEXDIG / "*" / "#" / [ visual-separator ]
> visual-separator     = "-" / "." / "(" / ")"
> 
> So would this be valid: "tel:-;tgrp=foo"?
> 
> Thanks, Shan Lu
> 
> >-----Original Message-----
> >From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
> >On Behalf Of Shan Lu
> >Sent: Monday, November 15, 2004 10:09 AM
> >To: 'list iptel'
> >Subject: RE: Comments on Trunk Group ID and RE: [Iptel] 
> >getting groupconsensuson moving forward with our workitems
> >
> >
> >Dave,
> >
> >Based on 2806bis-09, tel:;tgrp=foo is not valid. I am on the 
> >fence whether
> >it ought to be.
> >
> >Thanks, Shan Lu
> >
> >>From draft-ietf-iptel-rfc2806bis-09:
> >
> >   telephone-uri        = "tel:" telephone-subscriber
> >   telephone-subscriber = global-number / local-number
> >   global-number        = global-number-digits *par
> >   local-number         = local-number-digits *par context *par
> >   <snip>
> >   global-number-digits = "+" 1*phonedigit
> >   local-number-digits  = 1*phonedigit-hex
> >
> >>-----Original Message-----
> >>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
> >>On Behalf Of David R Oran
> >>Sent: Monday, November 15, 2004 8:05 AM
> >>To: Shan Lu
> >>Cc: 'list iptel'
> >>Subject: Re: Comments on Trunk Group ID and RE: [Iptel] 
> >>getting groupconsensus on moving forward with our workitems
> >>
> >>
> >>Is tel:;tgrp=foo not a valid tel: url?
> >>If not, it ought to be.
> >>
> >>On Nov 12, 2004, at 3:49 PM, Shan Lu wrote:
> >>
> >>> I support the plan to move forward with the trunk group draft. In  
> >>> fact, my
> >>> company intends to implement it.
> >>>
> >>> However, I do believe there exists a technical hole in the draft,  
> >>> which I
> >>> raised on the list before. That is, since "tgrp" is defined as a  
> >>> tel-uri
> >>> parameter, it can not exist on its own without the actual tel-uri  
> >>> number. In
> >>> the case of originating trunk group (ie. PSTN-to-IP 
> >direction), the  
> >>> calling
> >>> number may be absent. The question is then how to express 
> >>originating  
> >>> trunk
> >>> group.
> >>>
> >>> Thanks,
> >>>
> >>> Shan Lu
> >>>
> >>>> -----Original Message-----
> >>>> From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]
> >>>> On Behalf Of Jonathan Rosenberg
> >>>> Sent: Tuesday, November 09, 2004 4:03 PM
> >>>> To: list iptel
> >>>> Subject: [Iptel] getting group consensus on moving forward
> >>>> with our workitems
> >>>>
> >>>>
> >>>> Folks,
> >>>>
> >>>> At the iptel meeting today, there was consensus in the room on
> >>>> plans for
> >>>> moving forward with various drafts. I'd like to verify that
> >>>> consensus by
> >>>> asking the list if folks are OK proceeding with the following:
> >>>>
> >>>> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision
> >>>> appears in
> >>>> the archives
> >>>>
> >>>> * Proceed to wglc on
> >>>> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-grou
> >>> p-02.txt
> >>> immediately following the ietf
> >>>
> >>> * adopt
> >>> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi 
> >>> -00.txt
> >>> as a work item of iptel
> >>>
> >>> * do not adopt
> >>> 
> >>http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstri
> >ng-00.txt
> >>> as a work item. The author will take it directly to IESG as an
> >>> individual draft.
> >>>
> >>> * do not adopt the expired tel cpc draft
> >>> 
> >(http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a
> >>> work item. If folks have a problem that they believe is addressed by
> >>> this work, please bring that forward in the next few weeks.
> >>>
> >>> * tgrep will go to IESG after a revision incorporating comments from
> >>> Cullen and the pending comments from Jonathan
> >>>
> >>> Thanks,
> >>> Jonathan R.
> >>> -- 
> >>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> >>> Director, Service Provider VoIP Architecture   Parsippany, NJ  
> >>> 07054-2711
> >>> Cisco Systems
> >>> jdrosen@cisco.com                              FAX:   (973) 952-5050
> >>> http://www.jdrosen.net                         PHONE: (973) 952-5000
> >>> http://www.cisco.com
> >>>
> >>>
> >>> _______________________________________________
> >>> Iptel mailing list
> >>> Iptel@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/iptel
> >>>
> >>>
> >>> _______________________________________________
> >>> Iptel mailing list
> >>> Iptel@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/iptel
> >>>
> >>David R. Oran
> >>Cisco Fellow
> >>Cisco Systems
> >>7 Ladyslipper Lane
> >>Acton, MA 01720 USA
> >>Tel: +1 978 264 2048
> >>Email: oran@cisco.com
> >>
> >>
> >>_______________________________________________
> >>Iptel mailing list
> >>Iptel@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/iptel
> >>
> >
> >
> >_______________________________________________
> >Iptel mailing list
> >Iptel@ietf.org
> >https://www1.ietf.org/mailman/listinfo/iptel
> >
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

_______________________________________________
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From iptel-bounces@ietf.org  Mon Nov 15 22:01:21 2004
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To: jdrosen@cisco.com, iptel@ietf.org
Subject: Comments on phone-context prameter (Re: [Iptel] getting group
	consensus on moving forward with our workitems)
From: Takuya Sawada <tu-sawada@kddi.com>
References: <4191309C.4050005@cisco.com>
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Hi,

In the examples in trunk-group I-D, tgrp parameter appears before
phone-context patrameter. This violates the rule of 2806bis, section 3;

   Each parameter name ("pname"), the ISDN subaddress, the 'extension'
   and the 'context' MUST NOT appear more than once.  The
   'isdn-subaddress' or 'extension' MUST appear first, if present,
   followed by the 'context' parameter, if present, followed by any
   other parameters in lexicographical order.

      This simplifies comparison when the "tel" URI is compared
      character-by-character, such as in SIP URIs [RFC3261].

So phone-context parameter MUST appear first, assuming that
2806bis will be approved as it is.

Regards,
Takuya Sawada

> Folks,
> 
> At the iptel meeting today, there was consensus in the room on plans for 
> moving forward with various drafts. I'd like to verify that consensus by 
> asking the list if folks are OK proceeding with the following:
> 
> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears in 
> the archives
> 
> * Proceed to wglc on 
> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt 
> immediately following the ietf
> 
> * adopt 
> http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt 
> as a work item of iptel
> 
> * do not adopt 
> http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt 
> as a work item. The author will take it directly to IESG as an 
> individual draft.
> 
> * do not adopt the expired tel cpc draft 
> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a 
> work item. If folks have a problem that they believe is addressed by 
> this work, please bring that forward in the next few weeks.
> 
> * tgrep will go to IESG after a revision incorporating comments from 
> Cullen and the pending comments from Jonathan
> 
> Thanks,
> Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> 
> _______________________________________________
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> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

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inline.

Takuya Sawada wrote:

> Hi,
> 
> I am not so happy to hear that cpc work will not move forward.
> At least when interworking with ISUP or emulating the legacy PSTN,
> cpc parameter seems to me to be useful.

SIP-T exists for cases like this. For this work to be justified, there 
needs to be a reason for it on an IP network.

> 
> For example, cpc=test paramter in P-Asserted-Identity header may
> indicate that it is a test call so that it uses testing routing table
> and is free of charge through the connection.

There is value in doing test calls and diagnostics. However, the problem 
is broader than just copying the ISUP cpc=test parameter, and it needs 
to be looked at more holistically.

> With cpc=priority, it can express it requires the call handling
> with priority. (This should be done with R-P header, though)

Priority call handling is also a more complicated subject. SIP already 
has a Priority header field for signifying priority to the called party, 
and work is underway in the conext of ieprep to prioritize calls during 
national emergencies.

> With cpc=payphone, it may require to apply special charge rate, etc.

We have RFC3840, which allows us to specify characteristics of the user 
agent. In any case, tying the presence of a capability/characteristic to 
a charging plan is a recipe for fraud.

> 
> The scope of cpc work is probably limited to the legacy concept but
> is useful in a certain situation.
> Is it possible to move this forward to make an RFC as a working group
> item, or as an individual submission?

To date, the reasons brought forward for this draft were all in the 
category of "because isup has this". Generally speaking, that has not 
been sufficient justification for undertaking new work. Rather, specific 
problems need to be brought forward and a case made that existing 
solutions do not suffice.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Shan Lu wrote:

> Vijay,
> 
> Thanks for your response.
> 
> As you are aware, there are two scenarios where the originating number may
> be absent. PSTN may withhold the number or the PSTN-SIP gateway may not
> entrust downstrem endpoints with the number in case of calling party
> privacy.
> 
> We have seen both cases and the need to know the incoming PSTN trunk group
> when calling number is missing. In a lot applications, originating PSTN
> trunk group info can be very useful for routing and revealing such info is
> often not deemed to be a problem for privacy.

Can you explain these use cases? Specifically, under what cases is the 
originating trunk group used for routing decisions on the SIP side of 
the gateway?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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Folks,

This is to formally announce a 2 week working group last call on:

http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt

I'm glad to see we already have some discussion going on a comment made 
on this document. Please post any other comments on this draft during 
the last call period.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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On Nov 16, 2004, at 12:41 AM, Jonathan Rosenberg wrote:

>
>
> Shan Lu wrote:
>
>> Vijay,
>> Thanks for your response.
>> As you are aware, there are two scenarios where the originating 
>> number may
>> be absent. PSTN may withhold the number or the PSTN-SIP gateway may 
>> not
>> entrust downstrem endpoints with the number in case of calling party
>> privacy.
>> We have seen both cases and the need to know the incoming PSTN trunk 
>> group
>> when calling number is missing. In a lot applications, originating 
>> PSTN
>> trunk group info can be very useful for routing and revealing such 
>> info is
>> often not deemed to be a problem for privacy.
>
> Can you explain these use cases? Specifically, under what cases is the 
> originating trunk group used for routing decisions on the SIP side of 
> the gateway?
>
Uh, trunk group routing? E.G. all calls coming in on trunk group 234156 
(which happens to be from MCI in Alaska) and going to Tazmania get 
routed to C&W out trunk group 598275 (because I have a deal with C&W on 
calls to Tazmania this week).

> -Jonathan R.
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Wireless Research and Development/Internet Software and Services
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To: Takuya Sawada <tu-sawada@kddi.com>
Subject: Re: Comments on phone-context prameter (Re: [Iptel] getting group
	consensus on moving forward with our workitems)
References: <4191309C.4050005@cisco.com>
	<200411161159.HAC46832.TBE-BUVBX@kddi.com>
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Takuya Sawada wrote:

> Hi,
> 
> In the examples in trunk-group I-D, tgrp parameter appears before
> phone-context patrameter. This violates the rule of 2806bis, section 3;
[...]
> So phone-context parameter MUST appear first, assuming that
> 2806bis will be approved as it is.

Noted.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From iptel-bounces@ietf.org  Tue Nov 16 09:55:08 2004
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To: jdrosen@cisco.com, tu-sawada@kddi.com
Subject: Re: [Iptel] getting group consensus on moving forward with
	ourworkitems
From: Takuya Sawada <tu-sawada@kddi.com>
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Jonathan,

I admit that you are saying right thing, but...

comments inline.

> inline.
> 
> Takuya Sawada wrote:
> 
> > Hi,
> > 
> > I am not so happy to hear that cpc work will not move forward.
> > At least when interworking with ISUP or emulating the legacy PSTN,
> > cpc parameter seems to me to be useful.
> 
> SIP-T exists for cases like this. For this work to be justified, there 
> needs to be a reason for it on an IP network.
> 
For ISUP-SIP-ISUP interworking, I agree that SIP-T fits perfectly.
But there are many cases that SIP-T, ISUP encapslation, can not apply.
For SIP UA originating scenario, normally SIP UA can not build IAM
message in INVITE request.

> > 
> > For example, cpc=test paramter in P-Asserted-Identity header may
> > indicate that it is a test call so that it uses testing routing table
> > and is free of charge through the connection.
> 
> There is value in doing test calls and diagnostics. However, the problem 
> is broader than just copying the ISUP cpc=test parameter, and it needs 
> to be looked at more holistically.
> 
ISUP network can live with cpc=test for a long time :-)
Yes, SIP can be more sophisticated.
But the purpose of cpc=test is not diagnostics.
It is not for checking what is wrong but just for making sure 
it is good before making the resources or configuration available
to the ordinary usage.

> > With cpc=priority, it can express it requires the call handling
> > with priority. (This should be done with R-P header, though)
> 
> Priority call handling is also a more complicated subject. SIP already 
> has a Priority header field for signifying priority to the called party, 
> and work is underway in the conext of ieprep to prioritize calls during 
> national emergencies.
> 
Priority header can be applicable for that purpose. 
In table 3 of RFC 3261, "proxy" column of Priority header indicates
"ar", add and read. I would be happier if it were "adr".
Resource-Priority header will be a good candidate if approved.

> > With cpc=payphone, it may require to apply special charge rate, etc.
> 
> We have RFC3840, which allows us to specify characteristics of the user 
> agent. In any case, tying the presence of a capability/characteristic to 
> a charging plan is a recipe for fraud.
> 
RFC 3840 define Contact header parameters. I assume cpc parameter
to use with P-Asserted-Identity header. In that case cpc value
is network asserted as it is in ISUP.
For that purpose, defining new P- header or new P-Asserted-Identity
header parameter may be appropriate, though...

> > 
> > The scope of cpc work is probably limited to the legacy concept but
> > is useful in a certain situation.
> > Is it possible to move this forward to make an RFC as a working group
> > item, or as an individual submission?
> 
> To date, the reasons brought forward for this draft were all in the 
> category of "because isup has this". Generally speaking, that has not 
> been sufficient justification for undertaking new work. Rather, specific 
> problems need to be brought forward and a case made that existing 
> solutions do not suffice.
> 
I can not disagree "because isup has this" is not sufficient.
However, emulating ISUP with SIP is actually very important application
of SIP, unfortunately.
cpc parameter is the easiest way to emulate cpc parameter in ISUP.
Probably the easiest way is not necessarily the best way.
Umm.. does anyone have any other opinion?

Regards,
Takuya

> Thanks,
> Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

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>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of David R Oran
>Sent: Tuesday, November 16, 2004 2:24 AM
>To: 'list iptel'
>Subject: Re: Comments on Trunk Group ID and RE: [Iptel] 
>gettinggroup consensuson moving forward with our workitems
>
>
<snip>
>>
>> Can you explain these use cases? Specifically, under what 
>cases is the 
>> originating trunk group used for routing decisions on the 
>SIP side of 
>> the gateway?
>>
>Uh, trunk group routing? E.G. all calls coming in on trunk 
>group 234156 
>(which happens to be from MCI in Alaska) and going to Tazmania get 
>routed to C&W out trunk group 598275 (because I have a deal 
>with C&W on 
>calls to Tazmania this week).

That's what I want to way.

Regards, Shan Lu
>
>> -Jonathan R.
>>
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>> Director, Service Provider VoIP Architecture   Parsippany, NJ 
>> 07054-2711
>> Cisco Systems
>> jdrosen@cisco.com                              FAX:   (973) 952-5050
>> http://www.jdrosen.net                         PHONE: (973) 952-5000
>> http://www.cisco.com
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>David R. Oran
>Cisco Fellow
>Cisco Systems
>7 Ladyslipper Lane
>Acton, MA 01720 USA
>Tel: +1 978 264 2048
>Email: oran@cisco.com
>
>
>_______________________________________________
>Iptel mailing list
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>https://www1.ietf.org/mailman/listinfo/iptel
>


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Hi,

I agree that the cpc draft is needed especially if one is not using =
rfc3204.  I would like to see this draft move forward as a WG item.  I =
also have a couple of suggestions for the author.

First, it would be nice to see a clarification in this draft for cases =
where the From header does not have user's identity and  P-Asserted-ID =
or P-Preferred-ID is present containing a tel-uri. =20

Second, include a complete list of cpc values. =20

Thanks,
Fardad
  =20

-----Original Message-----
From: Takuya Sawada [mailto:tu-sawada@kddi.com]
Sent: Monday, November 15, 2004 5:20 AM
To: jdrosen@cisco.com; iptel@ietf.org
Subject: Re: [Iptel] getting group consensus on moving forward with our
workitems


Hi,

I am not so happy to hear that cpc work will not move forward.
At least when interworking with ISUP or emulating the legacy PSTN,
cpc parameter seems to me to be useful.

For example, cpc=3Dtest paramter in P-Asserted-Identity header may
indicate that it is a test call so that it uses testing routing table
and is free of charge through the connection.
With cpc=3Dpriority, it can express it requires the call handling
with priority. (This should be done with R-P header, though)
With cpc=3Dpayphone, it may require to apply special charge rate, etc.

The scope of cpc work is probably limited to the legacy concept but
is useful in a certain situation.
Is it possible to move this forward to make an RFC as a working group
item, or as an individual submission?

Regards,
Takuya

> Folks,
>=20
> At the iptel meeting today, there was consensus in the room on plans =
for=20
> moving forward with various drafts. I'd like to verify that consensus =
by=20
> asking the list if folks are OK proceeding with the following:
>=20
> * Proceed to wglc on draft-ietf-iptel-tel-np once a revision appears =
in=20
> the archives
>=20
> * Proceed to wglc on=20
> =
http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt=20
> immediately following the ietf
>=20
> * adopt=20
> =
http://www.ietf.org/internet-drafts/draft-stastny-iptel-tel-enumdi-00.txt=
=20
> as a work item of iptel
>=20
> * do not adopt=20
> =
http://www.ietf.org/internet-drafts/draft-rosen-iptel-dialstring-00.txt=20
> as a work item. The author will take it directly to IESG as an=20
> individual draft.
>=20
> * do not adopt the expired tel cpc draft=20
> (http://www.watersprings.org/pub/id/draft-mahy-iptel-cpc-01.txt) as a=20
> work item. If folks have a problem that they believe is addressed by=20
> this work, please bring that forward in the next few weeks.
>=20
> * tgrep will go to IESG after a revision incorporating comments from=20
> Cullen and the pending comments from Jonathan
>=20
> Thanks,
> Jonathan R.
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ =
07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi,=20
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

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From iptel-bounces@ietf.org  Tue Nov 16 10:36:06 2004
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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Wireless Research and Development/Internet Software and Services
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Subject: [Iptel] Originating trunk group without calling number (Was [Re:
 Comments on Trunk Group ID])
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Shan Lu wrote:

> Vijay,
[...]
> I am happy with the draft as far as destination TG is concerned. 
> But I don't think it should prescribe a formula for originating 
> TG whose validity is sometimes questionable.

I think you mean above that the validity of the calling party
number is questionable, not the trunk group's?

OK, so given the ensuing discussion on the appropriateness of
carrying a trunk group even if the calling party number is not
available, it would seem that there are three ways to move
forward.

I would like to reach consensus on one of them.  Let me list
them and provide some pros and cons.

(1) As suggested by Shan Lu, using a "tel:-;tgrp=foo" in
     the Contact URI.

(2) In the Contact, use a tel URI of the form:
     "tel:0000;tgrp=foo;phone-context=example.com"

(3) In the Contact, use a SIP URI of the form:
     "sip:anonymous;tgrp=foo;phone-context=example.com@example.com"

For (1), I wonder if the intent of the author of rfc2806-bis
was indeed to sanction such use.  While the ABNF may allow
it implicitly, should we endorse such a usage?  I am sure
with enough ingenuity, production rules of many ABNFs can
yield pretty interesting outcomes.

Maybe the WG can decide if the use of tel URI in this form
is okay.

In (2), the "0000" in the Contact URI serves as a filler.
The proxy receiving a request from its upstream gateway
with such a filler will know that an calling party number
was not provided.

Note that this case is distinct from a "0000" in the
R-URI of a request arriving from the PSTN, which could
signify a valid number in the domain of the proxy handling
the request.

(3) uses a SIP URI in the Contact header, avoiding the
ambiguities associated with (1) and (2).

My preference would be (3).

Any feedback and discussion most welcome.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Subject: RE: [Iptel] getting group consensus on moving forward withourworkitems
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Hi all,

I support progressing the CPC draft as WG item. Comments inline. 

Regards,

Shan Lu

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of Takuya Sawada
>Sent: Tuesday, November 16, 2004 9:39 AM
>To: jdrosen@cisco.com; tu-sawada@kddi.com
>Cc: iptel@ietf.org
>Subject: Re: [Iptel] getting group consensus on moving forward 
>withourworkitems
>
>
>Jonathan,
>
>I admit that you are saying right thing, but...
>
>comments inline.
>
>> inline.
>> 
>> Takuya Sawada wrote:
>> 
>> > Hi,
>> > 
>> > I am not so happy to hear that cpc work will not move forward.
>> > At least when interworking with ISUP or emulating the legacy PSTN,
>> > cpc parameter seems to me to be useful.
>> 
>> SIP-T exists for cases like this. For this work to be 
>justified, there 
>> needs to be a reason for it on an IP network.
>> 
>For ISUP-SIP-ISUP interworking, I agree that SIP-T fits perfectly.
>But there are many cases that SIP-T, ISUP encapslation, can not apply.
>For SIP UA originating scenario, normally SIP UA can not build IAM
>message in INVITE request.
>

One such case would be a feature server (SBC, for example) managing end
users (user agents). But SIP/ISUP interworking is performed by a completely
separated entitiy. This interworking GW is not aware of end user profiles.
It will be useful for the feature server to inform the gateway a calling
UA's CPC so that IAM can be correctly constructed. Obviously this still
falls under "because ISUP has it" category. But at least it provides an
interworking motivation.  

<snip>



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inline.

Shan Lu wrote:


>>For ISUP-SIP-ISUP interworking, I agree that SIP-T fits perfectly.
>>But there are many cases that SIP-T, ISUP encapslation, can not apply.
>>For SIP UA originating scenario, normally SIP UA can not build IAM
>>message in INVITE request.
>>
> 
> 
> One such case would be a feature server (SBC, for example) managing end
> users (user agents). But SIP/ISUP interworking is performed by a completely
> separated entitiy. This interworking GW is not aware of end user profiles.
> It will be useful for the feature server to inform the gateway a calling
> UA's CPC so that IAM can be correctly constructed. Obviously this still
> falls under "because ISUP has it" category. But at least it provides an
> interworking motivation.  

My point was, in all such cases that I have heard, there were other ways 
to indicate the desired information in a manner appropriate for SIP, and 
then these could be converted to their appropriate ISUP versions at a 
gateway.

For example, we have a mechanism for defining the characteristics of a 
UA (rfc3840) which can be used to signal things like the fact that this 
is a prison phone. Seems like such a characteristic does not depend on 
the fact that the endpoint is represented with a tel URI, and might be 
useful for SIP to SIP calls. This means that defining a tel URI 
parameter to map 1-1 with ISUP equivalents is the wrong way to do this.

In SIP, we have not elected to simply redefine every ISUP parameter into 
SIP. Rather, we defined things appropriately for a SIP network, and then 
specified how they get converted at a gateway. Why should CPC be different?

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To: vkg@lucent.com, shanlu@sentito.com
Subject: Re: [Iptel] Originating trunk group without calling number (Was
	[Re:Comments on Trunk Group ID])
From: Takuya Sawada <tu-sawada@kddi.com>
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Hi,

I think only (3) conforms to both 2806bis and RFC 3261.
But (3) is not tel URI at all. It is just the convention of the
SIP URI's user part usage.

Comments inline.

> Shan Lu wrote:
> 
> > Vijay,
> [...]
> > I am happy with the draft as far as destination TG is concerned. 
> > But I don't think it should prescribe a formula for originating 
> > TG whose validity is sometimes questionable.
> 
> I think you mean above that the validity of the calling party
> number is questionable, not the trunk group's?
> 
> OK, so given the ensuing discussion on the appropriateness of
> carrying a trunk group even if the calling party number is not
> available, it would seem that there are three ways to move
> forward.
> 
> I would like to reach consensus on one of them.  Let me list
> them and provide some pros and cons.
> 
> (1) As suggested by Shan Lu, using a "tel:-;tgrp=foo" in
>      the Contact URI.
> 
ABNF in 2806bis-09 explicitly does not allow the form above.

   telephone-uri        = "tel:" telephone-subscriber
   telephone-subscriber = global-number / local-number
   global-number        = global-number-digits *par
   local-number         = local-number-digits *par context *par
   context              = ";phone-context=" descriptor
   global-number-digits = "+" 1*phonedigit

> (2) In the Contact, use a tel URI of the form:
>      "tel:0000;tgrp=foo;phone-context=example.com"
> 
Contact header MUST be a SIP or SIPS URI according to RFC 3261,
section 8.1.18.
TEL URI is not allowed to appear in Contact header in INVITE request.

> (3) In the Contact, use a SIP URI of the form:
>      "sip:anonymous;tgrp=foo;phone-context=example.com@example.com"
> 
This is legal but not related to tel URI...
Probably we can live with this.
But we may have another choice.

(4)
sip:0000;tgrp=foo;phone-context=example.com@example.com"

I do not have any preference between (3) and (4).

Regards,
Takuya

> For (1), I wonder if the intent of the author of rfc2806-bis
> was indeed to sanction such use.  While the ABNF may allow
> it implicitly, should we endorse such a usage?  I am sure
> with enough ingenuity, production rules of many ABNFs can
> yield pretty interesting outcomes.
> 
> Maybe the WG can decide if the use of tel URI in this form
> is okay.
> 
> In (2), the "0000" in the Contact URI serves as a filler.
> The proxy receiving a request from its upstream gateway
> with such a filler will know that an calling party number
> was not provided.
> 
> Note that this case is distinct from a "0000" in the
> R-URI of a request arriving from the PSTN, which could
> signify a valid number in the domain of the proxy handling
> the request.
> 
> (3) uses a SIP URI in the Contact header, avoiding the
> ambiguities associated with (1) and (2).
> 
> My preference would be (3).
> 
> Any feedback and discussion most welcome.
> 
> Thanks,
> 
> - vijay
> -- 
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Wireless Networks Group/Internet Software and Services
> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224 0216
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

_______________________________________________
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From iptel-bounces@ietf.org  Wed Nov 17 10:13:44 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Takuya Sawada'" <tu-sawada@kddi.com>, <vkg@lucent.com>
Subject: RE: [Iptel] Originating trunk group without calling number
	(Was[Re:Comments on Trunk Group ID])
Date: Wed, 17 Nov 2004 10:02:27 -0500
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Takuya and Vijay,

I believe there are two steps here. The first one is to come up with a =
valid
tel-uri format. The second step is to put it in a SIP header and most =
likely
convert it to SIP URI.

WRT step 1, we have two proposals:

A) tel:-;phone-context=3Dlocal;tgrp=3Dfoo
B) tel:0000;phone-context=3Dlocal;tgrp=3Dfoo

Takuya correctly pointed out that "phone-context" must be present in =
both
formats.=20

Assuming Proposal A passes 2806bis validity test, I definitely favor it. =
My
concern w/ B is the potential confusion that "0000" (or any other digit
string) may create in a local numbering plan.

Regards,

Shan Lu

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Takuya Sawada
>Sent: Wednesday, November 17, 2004 2:59 AM
>To: vkg@lucent.com; shanlu@sentito.com
>Cc: iptel@ietf.org
>Subject: Re: [Iptel] Originating trunk group without calling=20
>number (Was[Re:Comments on Trunk Group ID])
>
>
>Hi,
>
>I think only (3) conforms to both 2806bis and RFC 3261.
>But (3) is not tel URI at all. It is just the convention of the
>SIP URI's user part usage.
>
>Comments inline.
>
>> Shan Lu wrote:
>>=20
>> > Vijay,
>> [...]
>> > I am happy with the draft as far as destination TG is concerned.=20
>> > But I don't think it should prescribe a formula for originating=20
>> > TG whose validity is sometimes questionable.
>>=20
>> I think you mean above that the validity of the calling party
>> number is questionable, not the trunk group's?
>>=20
>> OK, so given the ensuing discussion on the appropriateness of
>> carrying a trunk group even if the calling party number is not
>> available, it would seem that there are three ways to move
>> forward.
>>=20
>> I would like to reach consensus on one of them.  Let me list
>> them and provide some pros and cons.
>>=20
>> (1) As suggested by Shan Lu, using a "tel:-;tgrp=3Dfoo" in
>>      the Contact URI.
>>=20
>ABNF in 2806bis-09 explicitly does not allow the form above.
>
>   telephone-uri        =3D "tel:" telephone-subscriber
>   telephone-subscriber =3D global-number / local-number
>   global-number        =3D global-number-digits *par
>   local-number         =3D local-number-digits *par context *par
>   context              =3D ";phone-context=3D" descriptor
>   global-number-digits =3D "+" 1*phonedigit
>
>> (2) In the Contact, use a tel URI of the form:
>>      "tel:0000;tgrp=3Dfoo;phone-context=3Dexample.com"
>>=20
>Contact header MUST be a SIP or SIPS URI according to RFC 3261,
>section 8.1.18.
>TEL URI is not allowed to appear in Contact header in INVITE request.
>
>> (3) In the Contact, use a SIP URI of the form:
>>      =
"sip:anonymous;tgrp=3Dfoo;phone-context=3Dexample.com@example.com"
>>=20
>This is legal but not related to tel URI...
>Probably we can live with this.
>But we may have another choice.
>
>(4)
>sip:0000;tgrp=3Dfoo;phone-context=3Dexample.com@example.com"
>
>I do not have any preference between (3) and (4).
>
>Regards,
>Takuya
>
>> For (1), I wonder if the intent of the author of rfc2806-bis
>> was indeed to sanction such use.  While the ABNF may allow
>> it implicitly, should we endorse such a usage?  I am sure
>> with enough ingenuity, production rules of many ABNFs can
>> yield pretty interesting outcomes.
>>=20
>> Maybe the WG can decide if the use of tel URI in this form
>> is okay.
>>=20
>> In (2), the "0000" in the Contact URI serves as a filler.
>> The proxy receiving a request from its upstream gateway
>> with such a filler will know that an calling party number
>> was not provided.
>>=20
>> Note that this case is distinct from a "0000" in the
>> R-URI of a request arriving from the PSTN, which could
>> signify a valid number in the domain of the proxy handling
>> the request.
>>=20
>> (3) uses a SIP URI in the Contact header, avoiding the
>> ambiguities associated with (1) and (2).
>>=20
>> My preference would be (3).
>>=20
>> Any feedback and discussion most welcome.
>>=20
>> Thanks,
>>=20
>> - vijay
>> --=20
>> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
>> Wireless Networks Group/Internet Software and Services
>> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane,=20
>Rm 6G-440
>> Naperville, Illinois 60566     Voice: +1 630 224 0216
>>=20
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>
>
>--------
>Takuya Sawada
>KDDI Corporation (KDDI)
>Garden Air Tower, 3-10-10, Iidabashi,=20
>Chiyoda-ku, Tokyo 102-8460, Japan
>Tel: +81-3-6678-2997
>Fax: +81-3-6678-0286
>tu-sawada@kddi.com
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel
>


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	(Was[Re:Comments on Trunk Group ID])
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Shan Lu wrote:

> Takuya and Vijay,
> 
> I believe there are two steps here. The first one is to come up 
> with a valid tel-uri format. The second step is to put it in 
> a SIP header and most likely convert it to SIP URI.
> 
> WRT step 1, we have two proposals:
> 
> A) tel:-;phone-context=local;tgrp=foo
> B) tel:0000;phone-context=local;tgrp=foo
> Assuming Proposal A passes 2806bis validity test, I definitely 
> favor it. 

I don't think A is a valid tel URI as per the production
rules of rfc2806bis.  Please double-check, but I do not
think it is.

> My concern w/ B is the potential confusion that "0000" (or 
> any other digit string) may create in a local numbering plan.

Yes, I agree.  I had pointed that out in my original email
as a disadvantage.

So, that leaves the following two choices (one from my original
email and the second one from Takyua's follow-up):

1) sip:anonymous;tgrp=foo;phone-context=example.com@example.com
2) sip:0000;tgrp=foo;phone-context=example.com@example.com

(2) still suffers from the ambiguity problem...so, I still
prefer (1).

The ambiguity in (2) is manageable since it is the proxy that
will be sending this URI in subsequent requests to the
gateway.  The gateway can associate a sip:0000 to mean that
the incoming number was absent when the call was initially
setup.  Of course, this is brittle and assumes some handshaking
between the gateway and the proxy (what if they want to use
sip:9999 instead?).

That is why I favor (1).  However, I do not have religious
feelings about this, so I am happy with whatever the WG
decides.

Comments....?

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Subject: [Iptel] I-D ACTION:draft-ietf-iptel-tel-np-03.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 IP Telephony Working Group of the IETF.

	Title		: New Parameters for the 'tel' URI to Support
                          Number Portability
	Author(s)	: J. Yu
	Filename	: draft-ietf-iptel-tel-np-03.txt
	Pages		: 12
	Date		: 2004-11-18
	
This document defines several new parameters in the 'tel' Uniform 
   Resource Identifier (URI) to support number portability (NP) for 
   geographical telephone numbers and freephone numbers.  The 'rn' 
   parameter carries the routing number for a ported geographical 
   telephone number.  The presence of the 'npdi' parameter indicates 
   that NP database dip has been performed on a geographical telephone 
   number.  The 'cic' parameter identifies the freephone service 
   provider for a freephone number.  The 'rn-context' and 'cic-context' 
   parameters describe the 'rn' and 'cic' parameters respectively when 
   the 'rn' and 'cic' parameters contain information in the 'local' 
   format.

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From iptel-bounces@ietf.org  Fri Nov 19 07:59:29 2004
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	number(Was[Re:Comments on Trunk Group ID])
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Hi,

In 2806bis-09

   local-number         = local-number-digits *par context *par
   local-number-digits  = 1*phonedigit-hex
   phonedigit-hex       = HEXDIG / "*" / "#" / [ visual-separator ]

I think "tel:-;phone-context=local;tgrp=foo" is sytactically OK.
(I don't know why visual-separator is inside square blackets.)

Anyway, I think now we reach the consensus to use the following
format;

sip:local-number-digits;phone-context=domain;tgrp=foo@domain

First, I want to ask whether we should
a) leave this as the matter of the adminstrative domain indicated
   by phone-context, or
b) define global unique format for void-number.

If a) no further question is needed.
If b), then, I want to ask what the unique format should be;
b-1) fixed non-number format which is not compliant to "local-number-digits"
     to avoid conflict (like "anonymous"),
b-2) fixed visual-separator only format to be syntacticaly compliant to
     "local-number-digits" and to avoid conflict (like "-"),
b-3) fixed number format to "fully" compliant to the intended format of
     "local-number-digits", but taking the risk of conflict (like "0000"),
b-4) fixed non-DIGIT format to reduce the risk of conflict comparing
     b-3) (like "####"),
b-5) any visual-separator only string which can be interpreted that any
     valid digits is not included (like "-", ".", "----", and so on),
b-6) void.
     (like sip:;phone-context=domain;tgrp=foo@domain)

b-6) is not compliant to ABNF in 2806bis unless 2806bis is modified
to allow it as follows;

    local-number-digits  = *phonedigit-hex

I don't know if it is good idea to modify, though. (maybe ugly)

I do not have any strong preference. Just pick one of these.
Ummm.. a) or b-2)/b-5)?

Comments?

Regards,
Takuya

> Shan Lu wrote:
> 
> > Takuya and Vijay,
> > 
> > I believe there are two steps here. The first one is to come up 
> > with a valid tel-uri format. The second step is to put it in 
> > a SIP header and most likely convert it to SIP URI.
> > 
> > WRT step 1, we have two proposals:
> > 
> > A) tel:-;phone-context=local;tgrp=foo
> > B) tel:0000;phone-context=local;tgrp=foo
> > Assuming Proposal A passes 2806bis validity test, I definitely 
> > favor it. 
> 
> I don't think A is a valid tel URI as per the production
> rules of rfc2806bis.  Please double-check, but I do not
> think it is.
> 
> > My concern w/ B is the potential confusion that "0000" (or 
> > any other digit string) may create in a local numbering plan.
> 
> Yes, I agree.  I had pointed that out in my original email
> as a disadvantage.
> 
> So, that leaves the following two choices (one from my original
> email and the second one from Takyua's follow-up):
> 
> 1) sip:anonymous;tgrp=foo;phone-context=example.com@example.com
> 2) sip:0000;tgrp=foo;phone-context=example.com@example.com
> 
> (2) still suffers from the ambiguity problem...so, I still
> prefer (1).
> 
> The ambiguity in (2) is manageable since it is the proxy that
> will be sending this URI in subsequent requests to the
> gateway.  The gateway can associate a sip:0000 to mean that
> the incoming number was absent when the call was initially
> setup.  Of course, this is brittle and assumes some handshaking
> between the gateway and the proxy (what if they want to use
> sip:9999 instead?).
> 
> That is why I favor (1).  However, I do not have religious
> feelings about this, so I am happy with whatever the WG
> decides.
> 
> Comments....?
> 
> Thanks,
> 
> - vijay
> -- 
> Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
> Wireless Networks Group/Internet Software and Services
> Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224 0216
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10, Iidabashi, 
Chiyoda-ku, Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0286
tu-sawada@kddi.com

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From iptel-bounces@ietf.org  Mon Nov 22 13:49:52 2004
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Subject: Re: [Iptel] I-D ACTION:draft-ietf-iptel-tel-np-03.txt
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At 03:15 PM 11/18/2004, Internet-Drafts@ietf.org wrote:

I think there was consensus at 61 that this could go to last call.



>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           : New Parameters for the 'tel' URI to Support
>                           Number Portability
>         Author(s)       : J. Yu
>         Filename        : draft-ietf-iptel-tel-np-03.txt
>         Pages           : 12
>         Date            : 2004-11-18
>
>This document defines several new parameters in the 'tel' Uniform
>    Resource Identifier (URI) to support number portability (NP) for
>    geographical telephone numbers and freephone numbers.  The 'rn'
>    parameter carries the routing number for a ported geographical
>    telephone number.  The presence of the 'npdi' parameter indicates
>    that NP database dip has been performed on a geographical telephone
>    number.  The 'cic' parameter identifies the freephone service
>    provider for a freephone number.  The 'rn-context' and 'cic-context'
>    parameters describe the 'rn' and 'cic' parameters respectively when
>    the 'rn' and 'cic' parameters contain information in the 'local'
>    format.


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
sip:rshockey(at)iptel.org   sip:57141@fwd.pulver.com
ENUM +87810-13313-31331
PSTN Office +1 571.434.5651 PSTN Mobile: +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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Subject: [Iptel] Begin WGLC for draft-ietf-iptel-tel-np-03
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The draft can now be found here:
http://www.ietf.org/internet-drafts/draft-ietf-iptel-tel-np-03.txt

The WGLC will last two weeks. Please send comments either to the list or 
to the authors if the comments are minor.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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To: Shan Lu <shanlu@sentito.com>, "'Takuya Sawada'" <tu-sawada@kddi.com>
Subject: Re: [Iptel] Originating trunk group without calling
	number(Was[Re:Comments on Trunk Group ID])
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Continuing our leftover discussion on representing the
trunk group if the calling party was not present in an
incoming session setup request...a question for the IPTEL
co-chairs is embedded in the writeup below.

With the subsequent confirmation from rfc2806bis author
regarding the inappropriateness of the following tel
URI (despite it being syntactically correct):

    tel:-;phone-context=example.com;tgrp=foo

I would like to revisit my earlier suggestion to use
the following URI when the calling party is not known:

    sip:anonymous;phone-context=example.com;tgrp=foo@example.com

If we go this route, Shan Lu raises good questions on the
choice of this URI.  But I believe that these questions have
answers that fit within the current usage of SIP.  Namely,

 > I am a little confused about this. The draft started by defining a
 > tel-uri extension "tgrp" and now we seemed to end up reserving a
 > section of userinfo space in sip uri. My other concern with reserving
 > sip uri userinfo space is that we may need SIP WG blessing, which
 > will further delay the draft, if we can get the nod at all.

I don't think that this is insurmountable.  Note that RFC3261
uses the display name "Anonymous" and a URI of
anonymous@anonymous.invalid for privacy reasons.  While I agree
that the use of such display names and URI is not normative in
RFC3261, the inherent understanding is that this use of 'anonymous'
is okay (at least, that is my read of the specification).

So, I do not believe that we will need to seek the blessing of
the SIP WG to use the username 'anonymous'.  IPTEL co-chairs can
provide further guidance if my understanding is flawed on
this count.

 > Also, a format that stays completely within tel-uri can be used in
 > applications other than SIP.

I believe that using the anonymous SIP URI in this manner is a way
to plug a corner case that you have identified; namely that the
calling party's number may be absent in the ISUP request coming into
a gateway.  This is an exception; the normal case will be to
populate the Contact with the SIP equivalent of the tel URI of
the calling party.

 > While I agree that the above sip uri format solves our
 > issue at hand, questions came to mind. For example, when calling
 > number IS present, do we use "tgrp" as part of tel-uri or do we use
 > sip uri similar to above?

When the calling number is present, the Contact is populated with
the SIP equivalent of the tel URI of the calling party.  The
username of anonymous MUST not be used in such cases.  We will make
this a normative behavior in the I-D.

Questions?  Comments?

Thanks,

- vijay

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Vijay K. Gurbani wrote:
> Continuing our leftover discussion on representing the
> trunk group if the calling party was not present in an
> incoming session setup request...a question for the IPTEL
> co-chairs is embedded in the writeup below.
> 
> With the subsequent confirmation from rfc2806bis author
> regarding the inappropriateness of the following tel
> URI (despite it being syntactically correct):

Actually, this is no longer syntactically correct. The BNF has been 
fixed in 48h to agree with the intent. No other URI scheme allows empty 
URIs, as far as I know.

> 
>    tel:-;phone-context=example.com;tgrp=foo
> 
> I would like to revisit my earlier suggestion to use
> the following URI when the calling party is not known:
> 
>    sip:anonymous;phone-context=example.com;tgrp=foo@example.com
> 
> If we go this route, Shan Lu raises good questions on the
> choice of this URI.  But I believe that these questions have
> answers that fit within the current usage of SIP.  Namely,

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From: "Jesske, R" <R.Jesske@t-com.net>
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Subject: [Iptel] Calling Party's Category
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Dear all,
in the past there was a discussion regarding the specification of a Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt)
What is the actual status of this activity. 
We are still interested in such kind of originating indication if the call/communication is coming from a normal SIP user or a SIP -Payphone or SIP-Hotelphone. 
Is there a interest from other parties in this issue?

Best Regards

Roland

Deutsche Telekom AG
T-Com Zentrale
Roland Jesske, TE332-2
Section TE33; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@t-com.net




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To: "Jesske, R" <R.Jesske@t-com.net>
Subject: Re: [Iptel] Calling Party's Category
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The question was what the requirement is for this sort of indication.  How is it used?

Jesske, R wrote:
> Dear all,
> in the past there was a discussion regarding the specification of a Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt)
> What is the actual status of this activity. 
> We are still interested in such kind of originating indication if the call/communication is coming from a normal SIP user or a SIP -Payphone or SIP-Hotelphone. 
> Is there a interest from other parties in this issue?
> 
> Best Regards
> 
> Roland
> 
> Deutsche Telekom AG
> T-Com Zentrale
> Roland Jesske, TE332-2
> Section TE33; Signalling, Gateways and Switching Systems 
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940 
> Fax:      +49 6151 83-4577 
> email:   r.jesske@t-com.net
> 
> 
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 
> 

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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Subject: RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 16:42:07 +0100
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From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "Tom Taylor" <taylor@nortelnetworks.com>, "Jesske, R" <R.Jesske@t-com.net>
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Hi,

As an fixed line operator we also are interested in this feature and =
would be happy that the IETF finally moves forward with this draft. This =
type of parameter is widely signalled in ISDN/PSTN networks. In SIP (for =
example) it is required in context of both:

- pure SIP networks (e.g. in case of a SIP payphone), and
- interworking with legacy networks=20

I think the uses case for this parameter will depend on the operator =
service portfolio. For example, in France, the calling party's category =
parameter is used for the Call Return service.

Best regards,
sebastien=20

-----Message d'origine-----
De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part =
de Tom Taylor
Envoy=E9 : mardi 30 novembre 2004 16:07
=C0 : Jesske, R
Cc : iptel@ietf.org; sip@ietf.org
Objet : Re: [Iptel] Calling Party's Category

The question was what the requirement is for this sort of indication.  =
How is it used?

Jesske, R wrote:
> Dear all,
> in the past there was a discussion regarding the specification of a=20
> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What =
is the actual status of this activity.
> We are still interested in such kind of originating indication if the =
call/communication is coming from a normal SIP user or a SIP -Payphone =
or SIP-Hotelphone.=20
> Is there a interest from other parties in this issue?
>=20
> Best Regards
>=20
> Roland
>=20
> Deutsche Telekom AG
> T-Com Zentrale
> Roland Jesske, TE332-2
> Section TE33; Signalling, Gateways and Switching Systems Am=20
> Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940=20
> Fax:      +49 6151 83-4577=20
> email:   r.jesske@t-com.net
>=20
>=20
>=20
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20
>=20

--
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel

_______________________________________________
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 11:23:16 -0500
To: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
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On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> Hi,
>
> As an fixed line operator we also are interested in this feature and=20=

> would be happy that the IETF finally moves forward with this draft.=20
> This type of parameter is widely signalled in ISDN/PSTN networks. In=20=

> SIP (for example) it is required in context of both:
>
> - pure SIP networks (e.g. in case of a SIP payphone), and
> - interworking with legacy networks
>
> I think the uses case for this parameter will depend on the operator=20=

> service portfolio. For example, in France, the calling party's=20
> category parameter is used for the Call Return service.
>
Could you explain a bit about how this is used? I, for one would find=20
it helpful and possibly a motivating use case.

Also note that if this information is used for any security or=20
billing-sensitive operations, we have to deal with the issue of how to=20=

secure it against spoofing (SIP UAs are generally not trusted). Would=20
you believe a call made by Robert Mitnick from a prison phone would=20
actually have a prison CPC in it's SIP headers :-) ?

(Mitnick is a widely known phone freak and network hacker who served a=20=

number of years in US prisons).

> Best regards,
> sebastien
>
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part=20=

> de Tom Taylor
> Envoy=E9 : mardi 30 novembre 2004 16:07
> =C0 : Jesske, R
> Cc : iptel@ietf.org; sip@ietf.org
> Objet : Re: [Iptel] Calling Party's Category
>
> The question was what the requirement is for this sort of indication. =20=

> How is it used?
>
> Jesske, R wrote:
>> Dear all,
>> in the past there was a discussion regarding the specification of a
>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What=20=

>> is the actual status of this activity.
>> We are still interested in such kind of originating indication if the=20=

>> call/communication is coming from a normal SIP user or a SIP=20
>> -Payphone or SIP-Hotelphone.
>> Is there a interest from other parties in this issue?
>>
>> Best Regards
>>
>> Roland
>>
>> Deutsche Telekom AG
>> T-Com Zentrale
>> Roland Jesske, TE332-2
>> Section TE33; Signalling, Gateways and Switching Systems Am
>> Kavalleriesand 3, 64295 Darmstadt, Germany
>> Phone:  +49 6151 83-5940
>> Fax:      +49 6151 83-4577
>> email:   r.jesske@t-com.net
>>
>>
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>
> --
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


_______________________________________________
Iptel mailing list
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https://www1.ietf.org/mailman/listinfo/iptel


From iptel-bounces@ietf.org  Tue Nov 30 12:09:24 2004
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Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 17:51:33 +0100
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In the Call Return service, a destination exchange memories an incoming =
call which failed due to non-reply. If the call was orginated by a =
payphone then the destination exchange does not memories the number =
since it makes no sense in calling back a payphone.=20

BR,
sebastien
 =20

-----Message d'origine-----
De : David R Oran [mailto:oran@cisco.com]=20
Envoy=E9 : mardi 30 novembre 2004 17:23
=C0 : GARCIN Sebastien RD-CORE-ISS
Cc : Jesske, R; Tom Taylor; iptel@ietf.org; sip@ietf.org
Objet : Re: [Sip] RE: [Iptel] Calling Party's Category


On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> Hi,
>
> As an fixed line operator we also are interested in this feature and=20
> would be happy that the IETF finally moves forward with this draft.
> This type of parameter is widely signalled in ISDN/PSTN networks. In=20
> SIP (for example) it is required in context of both:
>
> - pure SIP networks (e.g. in case of a SIP payphone), and
> - interworking with legacy networks
>
> I think the uses case for this parameter will depend on the operator=20
> service portfolio. For example, in France, the calling party's=20
> category parameter is used for the Call Return service.
>
Could you explain a bit about how this is used? I, for one would find it =
helpful and possibly a motivating use case.

Also note that if this information is used for any security or =
billing-sensitive operations, we have to deal with the issue of how to =
secure it against spoofing (SIP UAs are generally not trusted). Would =
you believe a call made by Robert Mitnick from a prison phone would =
actually have a prison CPC in it's SIP headers :-) ?

(Mitnick is a widely known phone freak and network hacker who served a =
number of years in US prisons).

> Best regards,
> sebastien
>
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part =

> de Tom Taylor Envoy=E9 : mardi 30 novembre 2004 16:07 =C0 : Jesske, R =
Cc :=20
> iptel@ietf.org; sip@ietf.org Objet : Re: [Iptel] Calling Party's=20
> Category
>
> The question was what the requirement is for this sort of indication.  =

> How is it used?
>
> Jesske, R wrote:
>> Dear all,
>> in the past there was a discussion regarding the specification of a=20
>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What=20
>> is the actual status of this activity.
>> We are still interested in such kind of originating indication if the =

>> call/communication is coming from a normal SIP user or a SIP=20
>> -Payphone or SIP-Hotelphone.
>> Is there a interest from other parties in this issue?
>>
>> Best Regards
>>
>> Roland
>>
>> Deutsche Telekom AG
>> T-Com Zentrale
>> Roland Jesske, TE332-2
>> Section TE33; Signalling, Gateways and Switching Systems Am=20
>> Kavalleriesand 3, 64295 Darmstadt, Germany
>> Phone:  +49 6151 83-5940
>> Fax:      +49 6151 83-4577
>> email:   r.jesske@t-com.net
>>
>>
>>
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>>
>
> --
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel


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Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
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On Nov 30, 2004, at 11:51 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> In the Call Return service, a destination exchange memories an=20
> incoming call which failed due to non-reply. If the call was orginated=20=

> by a payphone then the destination exchange does not memories the=20
> number since it makes no sense in calling back a payphone.
>
In the US it does. Do payphones in France not accept incoming calls?

> BR,
> sebastien
>
>
> -----Message d'origine-----
> De : David R Oran [mailto:oran@cisco.com]
> Envoy=E9 : mardi 30 novembre 2004 17:23
> =C0 : GARCIN Sebastien RD-CORE-ISS
> Cc : Jesske, R; Tom Taylor; iptel@ietf.org; sip@ietf.org
> Objet : Re: [Sip] RE: [Iptel] Calling Party's Category
>
>
> On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>
>> Hi,
>>
>> As an fixed line operator we also are interested in this feature and
>> would be happy that the IETF finally moves forward with this draft.
>> This type of parameter is widely signalled in ISDN/PSTN networks. In
>> SIP (for example) it is required in context of both:
>>
>> - pure SIP networks (e.g. in case of a SIP payphone), and
>> - interworking with legacy networks
>>
>> I think the uses case for this parameter will depend on the operator
>> service portfolio. For example, in France, the calling party's
>> category parameter is used for the Call Return service.
>>
> Could you explain a bit about how this is used? I, for one would find=20=

> it helpful and possibly a motivating use case.
>
> Also note that if this information is used for any security or=20
> billing-sensitive operations, we have to deal with the issue of how to=20=

> secure it against spoofing (SIP UAs are generally not trusted). Would=20=

> you believe a call made by Robert Mitnick from a prison phone would=20
> actually have a prison CPC in it's SIP headers :-) ?
>
> (Mitnick is a widely known phone freak and network hacker who served a=20=

> number of years in US prisons).
>
>> Best regards,
>> sebastien
>>
>> -----Message d'origine-----
>> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la =
part
>> de Tom Taylor Envoy=E9 : mardi 30 novembre 2004 16:07 =C0 : Jesske, R =
Cc :
>> iptel@ietf.org; sip@ietf.org Objet : Re: [Iptel] Calling Party's
>> Category
>>
>> The question was what the requirement is for this sort of indication.
>> How is it used?
>>
>> Jesske, R wrote:
>>> Dear all,
>>> in the past there was a discussion regarding the specification of a
>>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What
>>> is the actual status of this activity.
>>> We are still interested in such kind of originating indication if =
the
>>> call/communication is coming from a normal SIP user or a SIP
>>> -Payphone or SIP-Hotelphone.
>>> Is there a interest from other parties in this issue?
>>>
>>> Best Regards
>>>
>>> Roland
>>>
>>> Deutsche Telekom AG
>>> T-Com Zentrale
>>> Roland Jesske, TE332-2
>>> Section TE33; Signalling, Gateways and Switching Systems Am
>>> Kavalleriesand 3, 64295 Darmstadt, Germany
>>> Phone:  +49 6151 83-5940
>>> Fax:      +49 6151 83-4577
>>> email:   r.jesske@t-com.net
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Iptel mailing list
>>> Iptel@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>
>>>
>>
>> --
>> Tom Taylor
>> Carrier VoIP Standards Development
>> Nortel Networks
>> Phone +1 613 763 1496  (ESN 393-1496)
>> E-mail: taylor@nortelnetworks.com
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip Use
>> sipping@ietf.org for new developments on the application of sip
>>
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From iptel-bounces@ietf.org  Tue Nov 30 12:34:31 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Vijay K. Gurbani'" <vkg@lucent.com>
Subject: RE: [Iptel] Originating trunk group without
	calling	number(Was[Re:Comments on Trunk Group ID])
Date: Tue, 30 Nov 2004 12:12:00 -0500
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>-----Original Message-----
>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu] 
>Sent: Monday, November 29, 2004 3:20 PM
>To: Vijay K. Gurbani
>Cc: Shan Lu; 'Takuya Sawada'; IETF IPTEL WG
>Subject: Re: [Iptel] Originating trunk group without calling 
>number(Was[Re:Comments on Trunk Group ID])
>
>Vijay K. Gurbani wrote:
>> Continuing our leftover discussion on representing the
>> trunk group if the calling party was not present in an
>> incoming session setup request...a question for the IPTEL
>> co-chairs is embedded in the writeup below.
>> 
>> With the subsequent confirmation from rfc2806bis author
>> regarding the inappropriateness of the following tel
>> URI (despite it being syntactically correct):
>
>Actually, this is no longer syntactically correct. The BNF has been 
>fixed in 48h to agree with the intent. No other URI scheme 
>allows empty 
>URIs, as far as I know.
>
>>    tel:-;phone-context=example.com;tgrp=foo
>> 

Fixing the BNF does not make the issue disappear. The issue is, tel uri, as
currently defined, does not allow expression of unavailable phone digits. 

For example, "tel:xxx;phone-context=internal" can mean something even when
"xxx" isn't there. Think of a calling number that is "internal" but has
requested privacy. 

I am not for or against any particular way to indicate empty phone number
(which may or may not lead to empty uri). But I think -A- way will be
desirable.

Regards,

Shan Lu 


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From iptel-bounces@ietf.org  Tue Nov 30 12:40:34 2004
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Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 18:18:09 +0100
Message-ID: <49E7012A614B024B80A7D175CB9A64ECB04A03@ftrdmel1.rd.francetelecom.fr>
Thread-Topic: [Sip] RE: [Iptel] Calling Party's Category
Thread-Index: AcTW/a3LC+z6Z0GQSiCaZTd8uFUnWgAAahyw
From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "David R Oran" <oran@cisco.com>
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????=20

In the example, the payphone is the calling party.=20

1 - a payphone establishes a call,=20
2 - upon receipt of the incoming call request the destination exchange =
alerts the called party
3 - the called party does not reply=20

Given that the calling party is a payphone (and not an ordinary =
subscriber), the destination exchange does not memorise the number which =
it normally does for if the calling party is an ordinary subscriber.
  =20

-----Message d'origine-----
De : David R Oran [mailto:oran@cisco.com]=20
Envoy=E9 : mardi 30 novembre 2004 17:57
=C0 : GARCIN Sebastien RD-CORE-ISS
Cc : 'list iptel'
Objet : Re: [Sip] RE: [Iptel] Calling Party's Category


On Nov 30, 2004, at 11:51 AM, GARCIN Sebastien RD-CORE-ISS wrote:

> In the Call Return service, a destination exchange memories an=20
> incoming call which failed due to non-reply. If the call was orginated =

> by a payphone then the destination exchange does not memories the=20
> number since it makes no sense in calling back a payphone.
>
In the US it does. Do payphones in France not accept incoming calls?

> BR,
> sebastien
>
>
> -----Message d'origine-----
> De : David R Oran [mailto:oran@cisco.com] Envoy=E9 : mardi 30 novembre =

> 2004 17:23 =C0 : GARCIN Sebastien RD-CORE-ISS Cc : Jesske, R; Tom=20
> Taylor; iptel@ietf.org; sip@ietf.org Objet : Re: [Sip] RE: [Iptel]=20
> Calling Party's Category
>
>
> On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>
>> Hi,
>>
>> As an fixed line operator we also are interested in this feature and=20
>> would be happy that the IETF finally moves forward with this draft.
>> This type of parameter is widely signalled in ISDN/PSTN networks. In=20
>> SIP (for example) it is required in context of both:
>>
>> - pure SIP networks (e.g. in case of a SIP payphone), and
>> - interworking with legacy networks
>>
>> I think the uses case for this parameter will depend on the operator=20
>> service portfolio. For example, in France, the calling party's=20
>> category parameter is used for the Call Return service.
>>
> Could you explain a bit about how this is used? I, for one would find=20
> it helpful and possibly a motivating use case.
>
> Also note that if this information is used for any security or=20
> billing-sensitive operations, we have to deal with the issue of how to =

> secure it against spoofing (SIP UAs are generally not trusted). Would=20
> you believe a call made by Robert Mitnick from a prison phone would=20
> actually have a prison CPC in it's SIP headers :-) ?
>
> (Mitnick is a widely known phone freak and network hacker who served a =

> number of years in US prisons).
>
>> Best regards,
>> sebastien
>>
>> -----Message d'origine-----
>> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la=20
>> part de Tom Taylor Envoy=E9 : mardi 30 novembre 2004 16:07 =C0 : =
Jesske, R Cc :
>> iptel@ietf.org; sip@ietf.org Objet : Re: [Iptel] Calling Party's=20
>> Category
>>
>> The question was what the requirement is for this sort of indication.
>> How is it used?
>>
>> Jesske, R wrote:
>>> Dear all,
>>> in the past there was a discussion regarding the specification of a=20
>>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) What =

>>> is the actual status of this activity.
>>> We are still interested in such kind of originating indication if=20
>>> the call/communication is coming from a normal SIP user or a SIP=20
>>> -Payphone or SIP-Hotelphone.
>>> Is there a interest from other parties in this issue?
>>>
>>> Best Regards
>>>
>>> Roland
>>>
>>> Deutsche Telekom AG
>>> T-Com Zentrale
>>> Roland Jesske, TE332-2
>>> Section TE33; Signalling, Gateways and Switching Systems Am=20
>>> Kavalleriesand 3, 64295 Darmstadt, Germany
>>> Phone:  +49 6151 83-5940
>>> Fax:      +49 6151 83-4577
>>> email:   r.jesske@t-com.net
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Iptel mailing list
>>> Iptel@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>
>>>
>>
>> --
>> Tom Taylor
>> Carrier VoIP Standards Development
>> Nortel Networks
>> Phone +1 613 763 1496  (ESN 393-1496)
>> E-mail: taylor@nortelnetworks.com
>>
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>>
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 12:42:18 -0500
To: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
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	c:"Date: Tue, 30 Nov 2004 12:42:18 -0500"
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On Nov 30, 2004, at 12:18 PM, GARCIN Sebastien RD-CORE-ISS wrote:

> ????
>
> In the example, the payphone is the calling party.
>
> 1 - a payphone establishes a call,
> 2 - upon receipt of the incoming call request the destination exchange=20=

> alerts the called party
> 3 - the called party does not reply
>
> Given that the calling party is a payphone (and not an ordinary=20
> subscriber), the destination exchange does not memorise the number=20
> which it normally does for if the calling party is an ordinary=20
> subscriber.
>
Sorry if I was too terse. Take your scenario where the callee doesn't=20
answer. The caller at the payphone is standing around waiting. The=20
callee comes out of the bathroom, sees he has a missed call, and hits=20
call return. Payphone rings. Waiting person picks up.

You see this sort of thing in the movies all the time :-)


>
> -----Message d'origine-----
> De : David R Oran [mailto:oran@cisco.com]
> Envoy=E9 : mardi 30 novembre 2004 17:57
> =C0 : GARCIN Sebastien RD-CORE-ISS
> Cc : 'list iptel'
> Objet : Re: [Sip] RE: [Iptel] Calling Party's Category
>
>
> On Nov 30, 2004, at 11:51 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>
>> In the Call Return service, a destination exchange memories an
>> incoming call which failed due to non-reply. If the call was =
orginated
>> by a payphone then the destination exchange does not memories the
>> number since it makes no sense in calling back a payphone.
>>
> In the US it does. Do payphones in France not accept incoming calls?
>
>> BR,
>> sebastien
>>
>>
>> -----Message d'origine-----
>> De : David R Oran [mailto:oran@cisco.com] Envoy=E9 : mardi 30 =
novembre
>> 2004 17:23 =C0 : GARCIN Sebastien RD-CORE-ISS Cc : Jesske, R; Tom
>> Taylor; iptel@ietf.org; sip@ietf.org Objet : Re: [Sip] RE: [Iptel]
>> Calling Party's Category
>>
>>
>> On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>>
>>> Hi,
>>>
>>> As an fixed line operator we also are interested in this feature and
>>> would be happy that the IETF finally moves forward with this draft.
>>> This type of parameter is widely signalled in ISDN/PSTN networks. In
>>> SIP (for example) it is required in context of both:
>>>
>>> - pure SIP networks (e.g. in case of a SIP payphone), and
>>> - interworking with legacy networks
>>>
>>> I think the uses case for this parameter will depend on the operator
>>> service portfolio. For example, in France, the calling party's
>>> category parameter is used for the Call Return service.
>>>
>> Could you explain a bit about how this is used? I, for one would find
>> it helpful and possibly a motivating use case.
>>
>> Also note that if this information is used for any security or
>> billing-sensitive operations, we have to deal with the issue of how =
to
>> secure it against spoofing (SIP UAs are generally not trusted). Would
>> you believe a call made by Robert Mitnick from a prison phone would
>> actually have a prison CPC in it's SIP headers :-) ?
>>
>> (Mitnick is a widely known phone freak and network hacker who served =
a
>> number of years in US prisons).
>>
>>> Best regards,
>>> sebastien
>>>
>>> -----Message d'origine-----
>>> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la
>>> part de Tom Taylor Envoy=E9 : mardi 30 novembre 2004 16:07 =C0 : =
Jesske,=20
>>> R Cc :
>>> iptel@ietf.org; sip@ietf.org Objet : Re: [Iptel] Calling Party's
>>> Category
>>>
>>> The question was what the requirement is for this sort of =
indication.
>>> How is it used?
>>>
>>> Jesske, R wrote:
>>>> Dear all,
>>>> in the past there was a discussion regarding the specification of a
>>>> Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) =
What
>>>> is the actual status of this activity.
>>>> We are still interested in such kind of originating indication if
>>>> the call/communication is coming from a normal SIP user or a SIP
>>>> -Payphone or SIP-Hotelphone.
>>>> Is there a interest from other parties in this issue?
>>>>
>>>> Best Regards
>>>>
>>>> Roland
>>>>
>>>> Deutsche Telekom AG
>>>> T-Com Zentrale
>>>> Roland Jesske, TE332-2
>>>> Section TE33; Signalling, Gateways and Switching Systems Am
>>>> Kavalleriesand 3, 64295 Darmstadt, Germany
>>>> Phone:  +49 6151 83-5940
>>>> Fax:      +49 6151 83-4577
>>>> email:   r.jesske@t-com.net
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Iptel mailing list
>>>> Iptel@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>>
>>>>
>>>
>>> --
>>> Tom Taylor
>>> Carrier VoIP Standards Development
>>> Nortel Networks
>>> Phone +1 613 763 1496  (ESN 393-1496)
>>> E-mail: taylor@nortelnetworks.com
>>>
>>> _______________________________________________
>>> Iptel mailing list
>>> Iptel@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/iptel
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use
>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>> sipping@ietf.org for new developments on the application of sip
>>>
>> David R. Oran
>> Cisco Fellow
>> Cisco Systems
>> 7 Ladyslipper Lane
>> Acton, MA 01720 USA
>> Tel: +1 978 264 2048
>> Email: oran@cisco.com
>>
>>
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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Jonathan, I hope I am stating this correctly.

The Chair's proposal at the meeting was that this draft be dropped, because with 
SIP-T defined, there is no need to define other means in SIP to carry signalling 
destined specifically to the PSTN.  However, if use cases can be provided which 
demonstrate a requirement to be met in the pure SIP environment, work can be done 
(probably in SIPPING) to meet those requirements.

Nguyen, An wrote:
> Mr. Taylor;
> 
> I am just curious ... What is the status of the Internet Draft?
> 
> An
> 
> 
[snip]

-- 
Tom Taylor
Carrier VoIP Standards Development
Nortel Networks
Phone +1 613 763 1496  (ESN 393-1496)
E-mail: taylor@nortelnetworks.com

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In a message dated 11/30/2004 11:49:18 AM Eastern Standard Time, 
oran@cisco.com writes:


> Also note that if this information is used for any security or 
> billing-sensitive operations, we have to deal with the issue of how to 
> secure it against spoofing (SIP UAs are generally not trusted). Would 
> you believe a call made by Robert Mitnick from a prison phone would 
> actually have a prison CPC in it's SIP headers :-) ?
> 

Of course, this shows a basic difference in the way this capability is 
handled in SIP vs today's telephone network. It's an important thing to deal with, 
since many of these things are mandated by law. For example, I believe the 
requirement to identify calls from a prison as such is mandated.

In today's network, in order to provide assurance that the call is properly 
marked, not just for CPC but calling party ID as well, the phone in the prison 
is rnot esponsible for the marking. The switch outside the prison marks any 
call that comes over that pair of wires. Robert Mitnick had no opportunity to 
change that marking.

The issue for SIP is to provide the same level of assurance for such 
functions. I'm sure anxious to hear a resolution to this problem, since it keeps 
coming back. CPC is not unique in this regard, so a general solution is needed.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/30/2004 11:49:18 AM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Also note that if this info=
rmation is used for any security or=20
<BR>billing-sensitive operations, we have to deal with the issue of how to=20
<BR>secure it against spoofing (SIP UAs are generally not trusted). Would=20
<BR>you believe a call made by Robert Mitnick from a prison phone would=20
<BR>actually have a prison CPC in it's SIP headers :-) ?
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Of course, this shows a basic difference in the way this capability is h=
andled in SIP vs today's telephone network. It's an important thing to deal=20=
with, since many of these things are mandated by law. For example, I believe=
 the requirement to identify calls from a prison as such is mandated.
<BR>
<BR>In today's network, in order to provide assurance that the call is prope=
rly marked, not just for CPC but calling party ID as well, the phone in the=20=
prison is rnot esponsible for the marking. The switch outside the prison mar=
ks any call that comes over that pair of wires. Robert Mitnick had no opport=
unity to change that marking.
<BR>
<BR>The issue for SIP is to provide the same level of assurance for such fun=
ctions. I'm sure anxious to hear a resolution to this problem, since it keep=
s coming back. CPC is not unique in this regard, so a general solution is ne=
eded.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Adam Roach'" <adam@nostrum.com>, David R Oran <oran@cisco.com>
Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
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I view these sorts of security attributes as elements of trait-based
authorization. The calling party's category requires the sort of
pre-arrangement and federation to which something like SAML naturally lends
itself.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, November 30, 2004 9:27 PM
> To: David R Oran
> Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
> Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
> 
> 
> David R Oran wrote:
> 
> >
> > On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
> >
> >> As an fixed line operator we also are interested in this 
> feature and 
> >> would be happy that the IETF finally moves forward with 
> this draft. 
> >> This type of parameter is widely signalled in ISDN/PSTN 
> networks. In 
> >> SIP (for example) it is required in context of both:
> >>
> >> - pure SIP networks (e.g. in case of a SIP payphone), and
> >> - interworking with legacy networks
> >>
> >> I think the uses case for this parameter will depend on 
> the operator 
> >> service portfolio. For example, in France, the calling party's 
> >> category parameter is used for the Call Return service.
> >>
> > Could you explain a bit about how this is used? I, for one 
> would find 
> > it helpful and possibly a motivating use case.
> 
> 
> I'm equally confused. I'm familiar with the calling party category in 
> ISUP, which works only because of the walled-garden nature of the SS7 
> network. Without that, there are really two major factors around this 
> which need to be characterized:
> 
>    1. Who is authorized to assert a particular calling party category?
>       This is much trickier than the identity problem; with 
> identity, we
>       can trust that a domain is the authority for the use of 
> user names
>       in that domain. For calling party category, there is no such
>       relationship. For example, it would probably not be valid for a
>       domain of "adamroach.com" to assert a calling party category of
>       "priority," "operator," or "police."
>    2. Under which circumstances is the calling party category required
>       to be present? Is there some architectural mechanism that can be
>       used to enforce this? (We don't need to define it, but there
>       should at least be some idea about whether it's 
> possible). If not,
>       then there is no purpose in coming up with a protocol mechanism
>       for conveying such information.
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Adam Roach'" <adam@nostrum.com>, David R Oran <oran@cisco.com>
Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 19:39:50 -0500
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I view these sorts of security attributes as elements of trait-based
authorization. The calling party's category requires the sort of
pre-arrangement and federation to which something like SAML naturally lends
itself.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, November 30, 2004 9:27 PM
> To: David R Oran
> Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
> Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
> 
> 
> David R Oran wrote:
> 
> >
> > On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
> >
> >> As an fixed line operator we also are interested in this 
> feature and 
> >> would be happy that the IETF finally moves forward with 
> this draft. 
> >> This type of parameter is widely signalled in ISDN/PSTN 
> networks. In 
> >> SIP (for example) it is required in context of both:
> >>
> >> - pure SIP networks (e.g. in case of a SIP payphone), and
> >> - interworking with legacy networks
> >>
> >> I think the uses case for this parameter will depend on 
> the operator 
> >> service portfolio. For example, in France, the calling party's 
> >> category parameter is used for the Call Return service.
> >>
> > Could you explain a bit about how this is used? I, for one 
> would find 
> > it helpful and possibly a motivating use case.
> 
> 
> I'm equally confused. I'm familiar with the calling party category in 
> ISUP, which works only because of the walled-garden nature of the SS7 
> network. Without that, there are really two major factors around this 
> which need to be characterized:
> 
>    1. Who is authorized to assert a particular calling party category?
>       This is much trickier than the identity problem; with 
> identity, we
>       can trust that a domain is the authority for the use of 
> user names
>       in that domain. For calling party category, there is no such
>       relationship. For example, it would probably not be valid for a
>       domain of "adamroach.com" to assert a calling party category of
>       "priority," "operator," or "police."
>    2. Under which circumstances is the calling party category required
>       to be present? Is there some architectural mechanism that can be
>       used to enforce this? (We don't need to define it, but there
>       should at least be some idea about whether it's 
> possible). If not,
>       then there is no purpose in coming up with a protocol mechanism
>       for conveying such information.
> 
> /a
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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

_______________________________________________
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From: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
Date: Tue, 30 Nov 2004 22:01:14 -0500
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On Nov 30, 2004, at 7:39 PM, Peterson, Jon wrote:

>
> I view these sorts of security attributes as elements of trait-based
> authorization. The calling party's category requires the sort of
> pre-arrangement and federation to which something like SAML naturally 
> lends
> itself.
>
Yes. I agree this is the right approach to any characteristic of an 
endpoint/subscriber that has to be securely asserted, and particularly, 
which should not be easily repudiatable.
Dave.

> Jon Peterson
> NeuStar, Inc.
>
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Tuesday, November 30, 2004 9:27 PM
>> To: David R Oran
>> Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
>> Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
>>
>>
>> David R Oran wrote:
>>
>>>
>>> On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:
>>>
>>>> As an fixed line operator we also are interested in this
>> feature and
>>>> would be happy that the IETF finally moves forward with
>> this draft.
>>>> This type of parameter is widely signalled in ISDN/PSTN
>> networks. In
>>>> SIP (for example) it is required in context of both:
>>>>
>>>> - pure SIP networks (e.g. in case of a SIP payphone), and
>>>> - interworking with legacy networks
>>>>
>>>> I think the uses case for this parameter will depend on
>> the operator
>>>> service portfolio. For example, in France, the calling party's
>>>> category parameter is used for the Call Return service.
>>>>
>>> Could you explain a bit about how this is used? I, for one
>> would find
>>> it helpful and possibly a motivating use case.
>>
>>
>> I'm equally confused. I'm familiar with the calling party category in
>> ISUP, which works only because of the walled-garden nature of the SS7
>> network. Without that, there are really two major factors around this
>> which need to be characterized:
>>
>>    1. Who is authorized to assert a particular calling party category?
>>       This is much trickier than the identity problem; with
>> identity, we
>>       can trust that a domain is the authority for the use of
>> user names
>>       in that domain. For calling party category, there is no such
>>       relationship. For example, it would probably not be valid for a
>>       domain of "adamroach.com" to assert a calling party category of
>>       "priority," "operator," or "police."
>>    2. Under which circumstances is the calling party category required
>>       to be present? Is there some architectural mechanism that can be
>>       used to enforce this? (We don't need to define it, but there
>>       should at least be some idea about whether it's
>> possible). If not,
>>       then there is no purpose in coming up with a protocol mechanism
>>       for conveying such information.
>>
>> /a
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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Date: Tue, 30 Nov 2004 16:06:13 EST
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
To: oran@cisco.com, sebastien.garcin@francetelecom.com
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In a message dated 11/30/2004 11:49:18 AM Eastern Standard Time, 
oran@cisco.com writes:


> Also note that if this information is used for any security or 
> billing-sensitive operations, we have to deal with the issue of how to 
> secure it against spoofing (SIP UAs are generally not trusted). Would 
> you believe a call made by Robert Mitnick from a prison phone would 
> actually have a prison CPC in it's SIP headers :-) ?
> 

Of course, this shows a basic difference in the way this capability is 
handled in SIP vs today's telephone network. It's an important thing to deal with, 
since many of these things are mandated by law. For example, I believe the 
requirement to identify calls from a prison as such is mandated.

In today's network, in order to provide assurance that the call is properly 
marked, not just for CPC but calling party ID as well, the phone in the prison 
is rnot esponsible for the marking. The switch outside the prison marks any 
call that comes over that pair of wires. Robert Mitnick had no opportunity to 
change that marking.

The issue for SIP is to provide the same level of assurance for such 
functions. I'm sure anxious to hear a resolution to this problem, since it keeps 
coming back. CPC is not unique in this regard, so a general solution is needed.

Mike Pierce
Artel


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 11/30/2004 11:49:18 AM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Also note that if this info=
rmation is used for any security or=20
<BR>billing-sensitive operations, we have to deal with the issue of how to=20
<BR>secure it against spoofing (SIP UAs are generally not trusted). Would=20
<BR>you believe a call made by Robert Mitnick from a prison phone would=20
<BR>actually have a prison CPC in it's SIP headers :-) ?
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Of course, this shows a basic difference in the way this capability is h=
andled in SIP vs today's telephone network. It's an important thing to deal=20=
with, since many of these things are mandated by law. For example, I believe=
 the requirement to identify calls from a prison as such is mandated.
<BR>
<BR>In today's network, in order to provide assurance that the call is prope=
rly marked, not just for CPC but calling party ID as well, the phone in the=20=
prison is rnot esponsible for the marking. The switch outside the prison mar=
ks any call that comes over that pair of wires. Robert Mitnick had no opport=
unity to change that marking.
<BR>
<BR>The issue for SIP is to provide the same level of assurance for such fun=
ctions. I'm sure anxious to hear a resolution to this problem, since it keep=
s coming back. CPC is not unique in this regard, so a general solution is ne=
eded.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR></FONT></HTML>

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