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To: David R Oran <oran@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
References: <49E7012A614B024B80A7D175CB9A64ECB0491A@ftrdmel1.rd.francetelecom.fr>
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Cc: "Jesske, R" <R.Jesske@t-com.net>, iptel@ietf.org,
        Tom Taylor <taylor@nortelnetworks.com>, sip@ietf.org,
        GARCIN Sebastien RD-CORE-ISS <sebastien.garcin@francetelecom.com>
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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

_______________________________________________
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From iptel-bounces@ietf.org  Wed Dec  1 00:36:13 2004
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Subject: RE: [Sip] RE: [Iptel] Calling Party's Category
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From: "Ben Gatewood" <bgatewood@streamdoor.net>
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        "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
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His name is Kevin. Other than that I agree with you.

I'm interested in this area as well as I might be able to make use of thi=
s type of functionality in a hospitality industry scenario.=20

Regards,

B

http://www.freekevin.com/

Ben Gatewood
Streamdoor Ltd
114b Power Rd
London
W4 5PY
+44 20 7422 0439
+44 7835 136 187
bgatewood@streamdoor.net



-----Original Message-----
From: sip-bounces@ietf.org on behalf of David R Oran
Sent: Tue 11/30/2004 4:23 PM
To: GARCIN Sebastien RD-CORE-ISS
Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
=20

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


_______________________________________________
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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<P><FONT SIZE=3D2>His name is Kevin. Other than that I agree with you.<BR=
>
<BR>
I'm interested in this area as well as I might be able to make use of thi=
s type of functionality in a hospitality industry scenario.<BR>
<BR>
Regards,<BR>
<BR>
B<BR>
<BR>
<A HREF=3D"http://www.freekevin.com/">http://www.freekevin.com/</A><BR>
<BR>
Ben Gatewood<BR>
Streamdoor Ltd<BR>
114b Power Rd<BR>
London<BR>
W4 5PY<BR>
+44 20 7422 0439<BR>
+44 7835 136 187<BR>
bgatewood@streamdoor.net<BR>
<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sip-bounces@ietf.org on behalf of David R Oran<BR>
Sent: Tue 11/30/2004 4:23 PM<BR>
To: GARCIN Sebastien RD-CORE-ISS<BR>
Cc: Jesske, R; iptel@ietf.org; Tom Taylor; sip@ietf.org<BR>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category<BR>
<BR>
<BR>
On Nov 30, 2004, at 10:42 AM, GARCIN Sebastien RD-CORE-ISS wrote:<BR>
<BR>
&gt; Hi,<BR>
&gt;<BR>
&gt; As an fixed line operator we also are interested in this feature and=
<BR>
&gt; would be happy that the IETF finally moves forward with this draft.<=
BR>
&gt; This type of parameter is widely signalled in ISDN/PSTN networks. In=
<BR>
&gt; SIP (for example) it is required in context of both:<BR>
&gt;<BR>
&gt; - pure SIP networks (e.g. in case of a SIP payphone), and<BR>
&gt; - interworking with legacy networks<BR>
&gt;<BR>
&gt; I think the uses case for this parameter will depend on the operator=
<BR>
&gt; service portfolio. For example, in France, the calling party's<BR>
&gt; category parameter is used for the Call Return service.<BR>
&gt;<BR>
Could you explain a bit about how this is used? I, for one would find<BR>=

it helpful and possibly a motivating use case.<BR>
<BR>
Also note that if this information is used for any security or<BR>
billing-sensitive operations, we have to deal with the issue of how to<BR=
>
secure it against spoofing (SIP UAs are generally not trusted). Would<BR>=

you believe a call made by Robert Mitnick from a prison phone would<BR>
actually have a prison CPC in it's SIP headers :-) ?<BR>
<BR>
(Mitnick is a widely known phone freak and network hacker who served a<BR=
>
number of years in US prisons).<BR>
<BR>
&gt; Best regards,<BR>
&gt; sebastien<BR>
&gt;<BR>
&gt; -----Message d'origine-----<BR>
&gt; De : iptel-bounces@ietf.org [<A HREF=3D"mailto:iptel-bounces@ietf.or=
g">mailto:iptel-bounces@ietf.org</A>] De la part<BR>
&gt; de Tom Taylor<BR>
&gt; Envoy=E9 : mardi 30 novembre 2004 16:07<BR>
&gt; =C0 : Jesske, R<BR>
&gt; Cc : iptel@ietf.org; sip@ietf.org<BR>
&gt; Objet : Re: [Iptel] Calling Party's Category<BR>
&gt;<BR>
&gt; The question was what the requirement is for this sort of indication=
=2E&nbsp;<BR>
&gt; How is it used?<BR>
&gt;<BR>
&gt; Jesske, R wrote:<BR>
&gt;&gt; Dear all,<BR>
&gt;&gt; in the past there was a discussion regarding the specification o=
f a<BR>
&gt;&gt; Calling Party's Category for SIP. (draft-mahy-iptel-cpc-00.txt) =
What<BR>
&gt;&gt; is the actual status of this activity.<BR>
&gt;&gt; We are still interested in such kind of originating indication i=
f the<BR>
&gt;&gt; call/communication is coming from a normal SIP user or a SIP<BR>=

&gt;&gt; -Payphone or SIP-Hotelphone.<BR>
&gt;&gt; Is there a interest from other parties in this issue?<BR>
&gt;&gt;<BR>
&gt;&gt; Best Regards<BR>
&gt;&gt;<BR>
&gt;&gt; Roland<BR>
&gt;&gt;<BR>
&gt;&gt; Deutsche Telekom AG<BR>
&gt;&gt; T-Com Zentrale<BR>
&gt;&gt; Roland Jesske, TE332-2<BR>
&gt;&gt; Section TE33; Signalling, Gateways and Switching Systems Am<BR>
&gt;&gt; Kavalleriesand 3, 64295 Darmstadt, Germany<BR>
&gt;&gt; Phone:&nbsp; +49 6151 83-5940<BR>
&gt;&gt; Fax:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +49 6151 83-4577<BR>
&gt;&gt; email:&nbsp;&nbsp; r.jesske@t-com.net<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; _______________________________________________<BR>
&gt;&gt; Iptel mailing list<BR>
&gt;&gt; Iptel@ietf.org<BR>
&gt;&gt; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/iptel">https:/=
/www1.ietf.org/mailman/listinfo/iptel</A><BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;<BR>
&gt; --<BR>
&gt; Tom Taylor<BR>
&gt; Carrier VoIP Standards Development<BR>
&gt; Nortel Networks<BR>
&gt; Phone +1 613 763 1496&nbsp; (ESN 393-1496)<BR>
&gt; E-mail: taylor@nortelnetworks.com<BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Iptel mailing list<BR>
&gt; Iptel@ietf.org<BR>
&gt; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/iptel">https://www=
1.ietf.org/mailman/listinfo/iptel</A><BR>
&gt;<BR>
&gt; _______________________________________________<BR>
&gt; Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/mailman/list=
info/sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>
&gt; This list is for NEW development of the core SIP Protocol<BR>
&gt; Use sip-implementors@cs.columbia.edu for questions on current sip<BR=
>
&gt; Use sipping@ietf.org for new developments on the application of sip<=
BR>
&gt;<BR>
David R. Oran<BR>
Cisco Fellow<BR>
Cisco Systems<BR>
7 Ladyslipper Lane<BR>
Acton, MA 01720 USA<BR>
Tel: +1 978 264 2048<BR>
Email: oran@cisco.com<BR>
<BR>
<BR>
_______________________________________________<BR>
Sip mailing list&nbsp; <A HREF=3D"https://www1.ietf.org/mailman/listinfo/=
sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>
This list is for NEW development of the core SIP Protocol<BR>
Use sip-implementors@cs.columbia.edu for questions on current sip<BR>
Use sipping@ietf.org for new developments on the application of sip<BR>
*************************************************************************=
*************<BR>
Disclaimer:<BR>
<BR>
This electronic message may contain privileged or confidential informatio=
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Subject: RE: [Iptel] Calling Party's Category
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Mr. Taylor;

I am just curious ... What is the status of the Internet Draft?

An



-----Original Message-----
From: Tom Taylor [mailto:taylor@nortelnetworks.com]
Sent: Tuesday, November 30, 2004 10:07 AM
To: Jesske, R
Cc: iptel@ietf.org; sip@ietf.org
Subject: 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

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From iptel-bounces@ietf.org  Wed Dec  1 01:57:13 2004
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Subject: [Iptel] AW:  Calling Party's Category
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Dear Tom, all,
I see there are several questions with regard to the CPC. Our approach =
is to use it for billing, charge info and routing purposes.

Payphone:
1. Payphone Calls are routed in an other manner than normal calls
2. Payphone Calls are billed differently compared with normal calls
3. The charge information shown to the user of the payphone is an other =
than for normal once
4. Based on the CPC a blocking of numbers (e.G premium rate or special =
destinations) can be performed
5. Special accounting mechanisms for free phone numbers are possible

A question was how to perform security and trust. Using the CPC within =
the P-Asserted-ID makes this information trusted. The UA can advice =
with the P-Preferred-ID that the SIP Proxy should use the proposed URI =
as P-Asserted-ID. But Proxy verifies this information. If the =
Information in the Hint is not correct the dialog will be candled or =
the stored ID will be provided by the network.

Best Regards

Roland

  =20


-----Urspr=FCngliche Nachricht-----
Von: Tom Taylor [mailto:taylor@nortelnetworks.com]
Gesendet: Dienstag, 30. November 2004 19:51
An: Nguyen, An
Cc: iptel@ietf.org; sipping@ietf.org; Jesske, Roland
Betreff: Re: [Iptel] Calling Party's Category


Jonathan, I hope I am stating this correctly.

The Chair's proposal at the meeting was that this draft be dropped, =
because with=20
SIP-T defined, there is no need to define other means in SIP to carry =
signalling=20
destined specifically to the PSTN.  However, if use cases can be =
provided which=20
demonstrate a requirement to be met in the pure SIP environment, work =
can be done=20
(probably in SIPPING) to meet those requirements.

Nguyen, An wrote:
> Mr. Taylor;
>=20
> I am just curious ... What is the status of the Internet Draft?
>=20
> An
>=20
>=20
[snip]

--=20
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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Just curious - what does it take to get a CPC code point allocated? If=20=

the IETF is to put this into SIP somehow, this issue will undoubtedly=20
come up (as it did in spades for ENUM).

So, let's assume I want to have a CPC called "Bordello", with the=20
following characteristics
- Calls to normal destinations are allowed and billed to the associated=20=

subscriber account
- Calls to the police are blocked
- Incoming calls are diverted to a sexy autoattendant.
- Calls to residences of known customers (provisioned by bordello=20
owner) are anonymized, or alternatively obfuscated with a location=20
header showing the origin is elsewhere.

The following questions obtain:
a) who allocates the "Bordello" codepoint?
b) what is the scope of the allocation?
c) where are the semantics documented (e.g. RFC, part of registration,=20=

etc.)
d) how does it interact with other call handling related functions?

this is beginning to look suspiciously like the "Namespace" discussion=20=

with respect to resource-priority, isn't it?

After thinking about this some more, I feel even more strongly along=20
the lines Jon Peterson suggested - don't put this into SIP; instead=20
deal with it as a role assertion through a SAML document provided as=20
part of the (enhanced) identity work when we get around to it.

This comprises yet another use case for SAML (besides delegation) which=20=

may argue for our being more aggressive about moving this forward right=20=

on the heels of the identity work. Anyone want to tackle this with an=20
i.d.?

Dave.

On Dec 1, 2004, at 1:27 AM, Jesske, R wrote:

> Dear Tom, all,
> I see there are several questions with regard to the CPC. Our approach=20=

> is to use it for billing, charge info and routing purposes.
>
> Payphone:
> 1. Payphone Calls are routed in an other manner than normal calls
> 2. Payphone Calls are billed differently compared with normal calls
> 3. The charge information shown to the user of the payphone is an=20
> other than for normal once
> 4. Based on the CPC a blocking of numbers (e.G premium rate or special=20=

> destinations) can be performed
> 5. Special accounting mechanisms for free phone numbers are possible
>
> A question was how to perform security and trust. Using the CPC within=20=

> the P-Asserted-ID makes this information trusted. The UA can advice=20
> with the P-Preferred-ID that the SIP Proxy should use the proposed URI=20=

> as P-Asserted-ID. But Proxy verifies this information. If the=20
> Information in the Hint is not correct the dialog will be candled or=20=

> the stored ID will be provided by the network.
>
> Best Regards
>
> Roland
>
>
>
>
> -----Urspr=FCngliche Nachricht-----
> Von: Tom Taylor [mailto:taylor@nortelnetworks.com]
> Gesendet: Dienstag, 30. November 2004 19:51
> An: Nguyen, An
> Cc: iptel@ietf.org; sipping@ietf.org; Jesske, Roland
> Betreff: Re: [Iptel] Calling Party's Category
>
>
> Jonathan, I hope I am stating this correctly.
>
> The Chair's proposal at the meeting was that this draft be dropped,=20
> because with
> SIP-T defined, there is no need to define other means in SIP to carry=20=

> signalling
> destined specifically to the PSTN.  However, if use cases can be=20
> provided which
> demonstrate a requirement to be met in the pure SIP environment, work=20=

> 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]
>
> --=20
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
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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Summarizing the discussion to date:

1. It appears that a case has been made for work in the area of transmitting 
characteristics of the calling party in a call, as a basis both for decisions on 
what services to provide and to support charging policies.

2. An architectural framework is required to address the trust issues associated 
with the user's potential control over signalling content.

3. Suggestions have been made about the actual mechanisms that might be applied 
within such a framework.

This seems like a fruitful area of cooperation between the IETF and the ITU-T, 
with the latter defining the framework and requirements and describing the 
application of the mechanisms provided by the IETF.

-- 
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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To: Adam Roach <adam@nostrum.com>, David R Oran <oran@cisco.com>
From: Mike Hammer <mhammer@cisco.com>
Subject: Re: [Sip] RE: [Iptel] Calling Party's Category
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Cc: "Jesske, R" <R.Jesske@t-com.net>, iptel@ietf.org,
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At 01:27 PM 11/30/2004 -0600, Adam Roach wrote:
>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
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel

The underlying assumption of the base peer-peer architectural model is that 
the peers have a mechanism to do end-end security and can establish direct 
trust between the ends.  But, this does not hold much water when one end is 
intentionally betraying trust.  That begs for middleman enforcement.

What happens when:
the calling endpoint can not be trusted,
a middle point proxy can be trusted to know the calling endpoint,
the called endpoint could trust the middle device?

The issue seems to be that we have an end to middle security requirements 
(draft-ietf-sipping-e2m-sec-reqs-04), but not a middle to end security 
requirements, whereby the called end does not trust the caller, but could 
establish trust of the middle with appropriate mechanisms.

Granted this leads back to a walled garden sort of approach to things, but 
walls may serve a purpose.  But, instead of thinking of this as an 
impedance to traffic, better to consider this as a truthful attribution of 
traffic as it is allowed to traverse domain boundaries.  Trust comes from 
authenticity of information and in some cases middlemen are needed to 
enforce that.  Otherwise, expect spam, spit, wormy, virus-sick networks.

Mike


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On Dec 1, 2004, at 3:21 PM, James M. Polk wrote:

> At 09:45 AM 12/1/2004 -0500, David R Oran wrote:
>> ....this is beginning to look suspiciously like the "Namespace"  
>> discussion with respect to resource-priority, isn't it?
>>
>> After thinking about this some more, I feel even more strongly along  
>> the lines Jon Peterson suggested - don't put this into SIP; instead  
>> deal with it as a role assertion through a SAML document provided as  
>> part of the (enhanced) identity work when we get around to it.
>>
>> This comprises yet another use case for SAML (besides delegation)  
>> which may argue for our being more aggressive about moving this  
>> forward right on the heels of the identity work. Anyone want to  
>> tackle this with an i.d.?
>
> How would this new ID idea be different than:
> http://www.ietf.org/internet-drafts/draft-tschofenig-sip-saml-01.txt ?
>
> The above is the answer doc to the requirements ID we authors have  
> been kinda dragging out feet about revving at:
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-trait-authz 
> -00.txt
> but we think is essentially done once revved.
>
I agree.

>
>> Dave.
>
>
> cheers,
> James
>
>                                *******************
>                 Truth is not to be argued... it is to be presented
>
>
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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On Dec 1, 2004, at 3:21 PM, James M. Polk wrote:

> At 09:45 AM 12/1/2004 -0500, David R Oran wrote:
>> ....this is beginning to look suspiciously like the "Namespace"  
>> discussion with respect to resource-priority, isn't it?
>>
>> After thinking about this some more, I feel even more strongly along  
>> the lines Jon Peterson suggested - don't put this into SIP; instead  
>> deal with it as a role assertion through a SAML document provided as  
>> part of the (enhanced) identity work when we get around to it.
>>
>> This comprises yet another use case for SAML (besides delegation)  
>> which may argue for our being more aggressive about moving this  
>> forward right on the heels of the identity work. Anyone want to  
>> tackle this with an i.d.?
>
> How would this new ID idea be different than:
> http://www.ietf.org/internet-drafts/draft-tschofenig-sip-saml-01.txt ?
>
> The above is the answer doc to the requirements ID we authors have  
> been kinda dragging out feet about revving at:
> http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-trait-authz 
> -00.txt
> but we think is essentially done once revved.
>
I agree.

>
>> Dave.
>
>
> cheers,
> James
>
>                                *******************
>                 Truth is not to be argued... it is to be presented
>
>
David R. Oran
Cisco Fellow
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720 USA
Tel: +1 978 264 2048
Email: oran@cisco.com


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

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From iptel-bounces@ietf.org  Thu Dec  2 02:57:02 2004
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Subject: [Iptel] AW: [Sip] AW:  Calling Party's Category
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The CPC in ITU-T/ETSI and the OLI in ANSI (Originating Line =
Identification) are well defined code points. I think IETF is also able =
to restrict definitions of code points used for routing aspects (Or Im =
Wrong?)

more comments in the text

Roland

> -----Urspr=FCngliche Nachricht-----
> Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]Im Auftrag von
> David R Oran
> Gesendet: Mittwoch, 1. Dezember 2004 15:46
> An: Jesske, Roland
> Cc: list iptel; 'sip@ietf.org'
> Betreff: Re: [Sip] AW: Calling Party's Category
>=20
>=20
> Just curious - what does it take to get a CPC code point=20
> allocated? If=20
> the IETF is to put this into SIP somehow, this issue will undoubtedly =

> come up (as it did in spades for ENUM).
>=20
> So, let's assume I want to have a CPC called "Bordello", with the=20
> following characteristics
> - Calls to normal destinations are allowed and billed to the=20
> associated=20
> subscriber account
> - Calls to the police are blocked
> - Incoming calls are diverted to a sexy autoattendant.
> - Calls to residences of known customers (provisioned by bordello=20
> owner) are anonymized, or alternatively obfuscated with a location=20
> header showing the origin is elsewhere.

If I use a P-Asserted-ID (with regard to 3323+3323 a trust model is =
defined) the CPC will be provided by the network and should be secure =
and as I said this element is also used for routing purposes so the =
definition of new code points should be restricted.=20
The concept described within draft-mahy-iptel-cpc-00 is to add the CPC =
as an element within the URI. From my point of view it could be also a =
normal header or P-header.
We see the need of such kind of mechanism and others too.
>=20
> The following questions obtain:
> a) who allocates the "Bordello" codepoint?
> b) what is the scope of the allocation?
> c) where are the semantics documented (e.g. RFC, part of=20
> registration,=20
> etc.)
> d) how does it interact with other call handling related functions?
>=20
> this is beginning to look suspiciously like the "Namespace"=20
> discussion=20
> with respect to resource-priority, isn't it?

No. draft-ietf-sip-resource-priority-05 is addressing priority =
services. CPC is addressing other issues.

>=20
> After thinking about this some more, I feel even more strongly along=20
> the lines Jon Peterson suggested - don't put this into SIP; instead=20
> deal with it as a role assertion through a SAML document provided as=20
> part of the (enhanced) identity work when we get around to it.

This could be used for billing services but not for routing from my =
understanding.

>=20
> This comprises yet another use case for SAML (besides=20
> delegation) which=20
> may argue for our being more aggressive about moving this=20
> forward right=20
> on the heels of the identity work. Anyone want to tackle this with an =

> i.d.?
>=20
> Dave.
>=20
> On Dec 1, 2004, at 1:27 AM, Jesske, R wrote:
>=20
> > Dear Tom, all,
> > I see there are several questions with regard to the CPC.=20
> Our approach=20
> > is to use it for billing, charge info and routing purposes.
> >
> > Payphone:
> > 1. Payphone Calls are routed in an other manner than normal calls
> > 2. Payphone Calls are billed differently compared with normal calls
> > 3. The charge information shown to the user of the payphone is an=20
> > other than for normal once
> > 4. Based on the CPC a blocking of numbers (e.G premium rate=20
> or special=20
> > destinations) can be performed
> > 5. Special accounting mechanisms for free phone numbers are =
possible
> >
> > A question was how to perform security and trust. Using the=20
> CPC within=20
> > the P-Asserted-ID makes this information trusted. The UA can advice =

> > with the P-Preferred-ID that the SIP Proxy should use the=20
> proposed URI=20
> > as P-Asserted-ID. But Proxy verifies this information. If the 
> > Information in the Hint is not correct the dialog will be=20
> candled or=20
> > the stored ID will be provided by the network.
> >
> > Best Regards
> >
> > Roland
> >
> >
> >
> >
> > -----Urspr=FCngliche Nachricht-----
> > Von: Tom Taylor [mailto:taylor@nortelnetworks.com]
> > Gesendet: Dienstag, 30. November 2004 19:51
> > An: Nguyen, An
> > Cc: iptel@ietf.org; sipping@ietf.org; Jesske, Roland
> > Betreff: Re: [Iptel] Calling Party's Category
> >
> >
> > 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=20
> SIP to carry=20
> > signalling
> > destined specifically to the PSTN.  However, if use cases can be=20
> > provided which
> > demonstrate a requirement to be met in the pure SIP=20
> environment, work=20
> > 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]
> >
> > --=20
> > Tom Taylor
> > Carrier VoIP Standards Development
> > Nortel Networks
> > Phone +1 613 763 1496  (ESN 393-1496)
> > E-mail: taylor@nortelnetworks.com
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> David R. Oran
> Cisco Fellow
> Cisco Systems
> 7 Ladyslipper Lane
> Acton, MA 01720 USA
> Tel: +1 978 264 2048
> Email: oran@cisco.com
>=20
>=20
> _______________________________________________
> 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
>=20

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At 09:45 AM 12/1/2004 -0500, David R Oran wrote:
>....this is beginning to look suspiciously like the "Namespace" discussion 
>with respect to resource-priority, isn't it?
>
>After thinking about this some more, I feel even more strongly along the 
>lines Jon Peterson suggested - don't put this into SIP; instead deal with 
>it as a role assertion through a SAML document provided as part of the 
>(enhanced) identity work when we get around to it.
>
>This comprises yet another use case for SAML (besides delegation) which 
>may argue for our being more aggressive about moving this forward right on 
>the heels of the identity work. Anyone want to tackle this with an i.d.?

How would this new ID idea be different than:
http://www.ietf.org/internet-drafts/draft-tschofenig-sip-saml-01.txt ?

The above is the answer doc to the requirements ID we authors have been 
kinda dragging out feet about revving at:
http://www.softarmor.com/wgdb/docs/draft-ietf-sipping-trait-authz-00.txt
but we think is essentially done once revved.


>Dave.


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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--NextPart


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


        RFC 3966

        Title:      The tel URI for Telephone Numbers
        Author(s):  H. Schulzrinne
        Status:     Standards Track
        Date:       December 2004
        Mailbox:    hgs@cs.columbia.edu
        Pages:      17
        Characters: 40783
        Obsoletes:  2806

        I-D Tag:    draft-ietf-iptel-rfc2806bis-09.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3966.txt


This document specifies the URI (Uniform Resource Identifier) scheme
"tel".  The "tel" URI describes resources identified by telephone
numbers.  This document obsoletes RFC 2806.

This document is a product of the IP Telephony Working Group of the
IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
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From iptel-bounces@ietf.org  Sat Dec  4 07:56:36 2004
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Hi Henning, folks,
   Just a quick thanks to all who worked hard on this document.
It's the keystone for much other work, and I speak for many in 
appreciating the effort that went into what appeared to be a sisyphean 
task.

all the best,
   Lawrence

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Xiaotao Wu and I have written two technical reports of possible interest 
to those following CPL. Both can be found at 
http://www1.cs.columbia.edu/~library/2004.html

     The Simplicity and Safety of the Language for End
     System Services (LESS)
     Xiaotao Wu and Henning Schulzrinne

     Service Learning in Internet Telephony
     Xiaotao Wu and Henning Schulzrinne

We would appreciate any comments you might have on these documents.

Henning

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From iptel-bounces@ietf.org  Mon Dec  6 00:37:56 2004
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Shan Lu wrote:

> 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. 

If such a deficiency exists in the tel URI, the trunk-group I-D
is probably not the right forum to argue for putting it in.  I
do note that 2806bis became RFC 3966 over the weekend, so
arguing for a deficiency in the tel URI may be a moot point
at this time.

Instead, I would like to return the focus on what we should
do in the trunk-group I-D when the calling party number
is not present.  The previous options are still on the
table; please, let's pick one.

As an aside, I am curious: does the ISUP ASN represent the
calling party number as an optional parameter to the
IAM message?  If yes, what does an intermediary switch do
when it gets a call setup request and there isn't any
calling party number?

At the very least, it would seem prudent that whatever we
do is no worse than the existing behavior in the PSTN.

Thanks,

- vijay
> 
> 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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I promised to start a thread on this and figure out what people want to do
with it and if we understand the requirements for it then propose next
steps. 

Fortunately, there is a thread going in the sip mailing list. I would
summarize that thread as:

1) the requirements and use case are varied and not clear
2) they involved complex security and authorization issues
3) SAML looks like it might help with many of them
4) there are issues with extension and scope of future categories

This makes me very uncomfortable with moving forward with this work right
now as a WG item in IPTEL. What I suggest is that we shelve this work for
now and let sipping figure out what the requirements are and if the
solutions looks like we need to add something to the tel URL, revive the
draft and figure out what to do with it.

What do people think of this? Folks ok with this? I will take silence as
well, just a fact of life on the iptel list :-)

Cullen



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Any ETA on a new rev of this draft?


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From iptel-bounces@ietf.org  Wed Dec  8 23:29:51 2004
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Jesske, R wrote:

> Dear Tom, all,
> I see there are several questions with regard to the CPC. Our approach is to use it for billing, charge info and routing purposes.
> 
> Payphone:
> 1. Payphone Calls are routed in an other manner than normal calls

In what way are they routed differently?


> 2. Payphone Calls are billed differently compared with normal calls
> 3. The charge information shown to the user of the payphone is an other than for normal once
> 4. Based on the CPC a blocking of numbers (e.G premium rate or special destinations) can be performed
> 5. Special accounting mechanisms for free phone numbers are possible

I think its also worth pointing out that the general topic of 
"payphones" in sip is a complicated one, of which the cpc aspect - 
asserting the trait of being a payphone - is one of the easier ones. 
Figuring out a way to do the payments securely is a much more 
challenging issue.

I also wanted to note that if there are requirements for endpoint 
assertion of characteristics and capabilities, we have a spec for that 
too - namely rfc3840. The usage of SAML for doing the network asserted 
traits also seems right to me.

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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Date: Thu, 09 Dec 2004 12:05:33 +0530
From: Jyoti <jyothij@huawei.com>
Subject: RE: [Iptel] Re: tgrep 04 comments
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Hello All,
For TGrep and TRIP - pls can you let me know if any charging data
framework is available i.e if any charge related information can be
updated , stored and queried.
Thanks
Jyoti 

-----Original Message-----
From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] On Behalf
Of Cullen Jennings
Sent: Thursday, December 09, 2004 7:27 AM
To: Manjunath S Bangalore
Cc: Dhaval N Shah; iptel@ietf.org; Hussein F Salama; rajneesh@cisco.com
Subject: [Iptel] Re: tgrep 04 comments



Any ETA on a new rev of this draft?



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<DIV><SPAN class=236172706-09122004><FONT face=Arial color=#0000ff size=2>Hello 
All,</FONT></SPAN></DIV>
<DIV><SPAN class=236172706-09122004><FONT face=Arial color=#0000ff size=2>For 
TGrep and TRIP - pls can you let me know if any charging data framework is 
available i.e if any charge related information can be updated , stored and 
queried.</FONT></SPAN></DIV>
<DIV><SPAN class=236172706-09122004><FONT face=Arial color=#0000ff 
size=2>Thanks</FONT></SPAN></DIV>
<DIV><SPAN class=236172706-09122004><FONT face=Arial color=#0000ff size=2>Jyoti 
</FONT></SPAN></DIV>
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Hi,

Regarding 1), I think the requirements from those who reqire
cpc parameter are simple and clear; carry the ISUP cpc 
information in SIP messages with tel URI.
Use cases are varied as they are varied in ISUP.

Regaerding 2), IMHO, cpc parameter will be used with 
P-Asserted-Identity header. In this scenario, trust domains 
model described in RFC 3324 is enough solution for security 
and authorization issues. Anyway we will require cpc value to
be asserted by the network as it is in ISUP.

I totally agree that 'because ISUP has it' is not sufficient 
to move it forward as an iptel WG item. During the discussion
I feel it is very difficult to justify cpc work for iptel
work item beyond 'because ISUP has it'.

Now I propose that let's move cpc draft forward as an individual
document and make it an informational RFC. It may be a good idea
to add the limited scope of cpc and some use cases.
If we can find more complete and general solution to cover cpc work
later, we will make it a standard track RFC.
I think we did that way for network asserted identity.

How do people think about the above idea?

Note that according RFC 3966, new tel URI RFC, an informational
RFC is sufficient for optional tel URI parameters.

Regards,
Takuya
> 
> I promised to start a thread on this and figure out what people want to do
> with it and if we understand the requirements for it then propose next
> steps. 
> 
> Fortunately, there is a thread going in the sip mailing list. I would
> summarize that thread as:
> 
> 1) the requirements and use case are varied and not clear
> 2) they involved complex security and authorization issues
> 3) SAML looks like it might help with many of them
> 4) there are issues with extension and scope of future categories
> 
> This makes me very uncomfortable with moving forward with this work right
> now as a WG item in IPTEL. What I suggest is that we shelve this work for
> now and let sipping figure out what the requirements are and if the
> solutions looks like we need to add something to the tel URL, revive the
> draft and figure out what to do with it.
> 
> What do people think of this? Folks ok with this? I will take silence as
> well, just a fact of life on the iptel list :-)
> 
> Cullen
> 
> 
> 
> _______________________________________________
> 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

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


From iptel-bounces@ietf.org  Thu Dec  9 07:52:45 2004
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Subject: RE: [Iptel] Calling Party Category
Date: Thu, 9 Dec 2004 13:41:04 +0100
Message-ID: <49E7012A614B024B80A7D175CB9A64ECD12FAC@ftrdmel1.rd.francetelecom.fr>
Thread-Topic: [Iptel] Calling Party Category
thread-index: AcTd6zCbi4wqbzSHTka1OOwhPL1ubgAAK9/Q
From: "GARCIN Sebastien RD-CORE-ISS" <sebastien.garcin@francetelecom.com>
To: "Takuya Sawada" <tu-sawada@kddi.com>, <fluffy@cisco.com>, <iptel@ietf.org>
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Hi,

I would like to support Takuya' proposal. I'd also like to point out =
that there are clear uses cases for the calling party's category and =
operators experience those requirements every day in their networks.=20

Here is a brief summary of those that relate to the "payphone" value:

Charging
--------
The cpc value "payphone" is used for charging purposes

Services
--------
Some services (e.g. Call return, Freephone services) use the "payphone" =
value.

The fact that we are in a pure SIP or ISUP environment does not change =
those requirements.

Regards,
S=E9bastien
=20

-----Message d'origine-----
De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] De la part =
de Takuya Sawada
Envoy=E9 : jeudi 9 d=E9cembre 2004 13:22
=C0 : fluffy@cisco.com; iptel@ietf.org
Objet : Re: [Iptel] Calling Party Category

Hi,

Regarding 1), I think the requirements from those who reqire cpc =
parameter are simple and clear; carry the ISUP cpc information in SIP =
messages with tel URI.
Use cases are varied as they are varied in ISUP.

Regaerding 2), IMHO, cpc parameter will be used with P-Asserted-Identity =
header. In this scenario, trust domains model described in RFC 3324 is =
enough solution for security and authorization issues. Anyway we will =
require cpc value to be asserted by the network as it is in ISUP.

I totally agree that 'because ISUP has it' is not sufficient to move it =
forward as an iptel WG item. During the discussion I feel it is very =
difficult to justify cpc work for iptel work item beyond 'because ISUP =
has it'.

Now I propose that let's move cpc draft forward as an individual =
document and make it an informational RFC. It may be a good idea to add =
the limited scope of cpc and some use cases.
If we can find more complete and general solution to cover cpc work =
later, we will make it a standard track RFC.
I think we did that way for network asserted identity.

How do people think about the above idea?

Note that according RFC 3966, new tel URI RFC, an informational RFC is =
sufficient for optional tel URI parameters.

Regards,
Takuya
>=20
> I promised to start a thread on this and figure out what people want=20
> to do with it and if we understand the requirements for it then=20
> propose next steps.
>=20
> Fortunately, there is a thread going in the sip mailing list. I would=20
> summarize that thread as:
>=20
> 1) the requirements and use case are varied and not clear
> 2) they involved complex security and authorization issues
> 3) SAML looks like it might help with many of them
> 4) there are issues with extension and scope of future categories
>=20
> This makes me very uncomfortable with moving forward with this work=20
> right now as a WG item in IPTEL. What I suggest is that we shelve this =

> work for now and let sipping figure out what the requirements are and=20
> if the solutions looks like we need to add something to the tel URL,=20
> revive the draft and figure out what to do with it.
>=20
> What do people think of this? Folks ok with this? I will take silence=20
> as well, just a fact of life on the iptel list :-)
>=20
> Cullen
>=20
>=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,
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

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


From iptel-bounces@ietf.org  Thu Dec  9 08:16:55 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: sebastien.garcin@francetelecom.com, tu-sawada@kddi.com, fluffy@cisco.com,
        iptel@ietf.org
Subject: AW: [Iptel] Calling Party Category
Date: Thu, 9 Dec 2004 14:05:45 +0100 
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Hi,
I would also like to support Takuya' and the comments from Sebatien =
proposal.

A in addition it could be used for routing purposes in a SIP Proxy to =
choose the best MGC/MG.=20

EG in PSTN/ISDN the "data-call" can be only transferred over 64kbit/s =
trunks. This can be used for a efficient coos of the correct MGC/MG =
combination with the clearmode codec.

Best Regards

Roland=20

> -----Urspr=FCngliche Nachricht-----
> Von: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]Im Auftrag
> von GARCIN Sebastien RD-CORE-ISS
> Gesendet: Donnerstag, 9. Dezember 2004 13:41
> An: Takuya Sawada; fluffy@cisco.com; iptel@ietf.org
> Betreff: RE: [Iptel] Calling Party Category
>=20
>=20
> Hi,
>=20
> I would like to support Takuya' proposal. I'd also like to=20
> point out that there are clear uses cases for the calling=20
> party's category and operators experience those requirements=20
> every day in their networks.=20
>=20
> Here is a brief summary of those that relate to the "payphone" value:
>=20
> Charging
> --------
> The cpc value "payphone" is used for charging purposes
>=20
> Services
> --------
> Some services (e.g. Call return, Freephone services) use the=20
> "payphone" value.
>=20
> The fact that we are in a pure SIP or ISUP environment does=20
> not change those requirements.
>=20
> Regards,
> S=E9bastien
> =20
>=20
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
> De la part de Takuya Sawada
> Envoy=E9 : jeudi 9 d=E9cembre 2004 13:22
> =C0 : fluffy@cisco.com; iptel@ietf.org
> Objet : Re: [Iptel] Calling Party Category
>=20
> Hi,
>=20
> Regarding 1), I think the requirements from those who reqire=20
> cpc parameter are simple and clear; carry the ISUP cpc=20
> information in SIP messages with tel URI.
> Use cases are varied as they are varied in ISUP.
>=20
> Regaerding 2), IMHO, cpc parameter will be used with=20
> P-Asserted-Identity header. In this scenario, trust domains=20
> model described in RFC 3324 is enough solution for security=20
> and authorization issues. Anyway we will require cpc value to=20
> be asserted by the network as it is in ISUP.
>=20
> I totally agree that 'because ISUP has it' is not sufficient=20
> to move it forward as an iptel WG item. During the discussion=20
> I feel it is very difficult to justify cpc work for iptel=20
> work item beyond 'because ISUP has it'.
>=20
> Now I propose that let's move cpc draft forward as an=20
> individual document and make it an informational RFC. It may=20
> be a good idea to add the limited scope of cpc and some use cases.
> If we can find more complete and general solution to cover=20
> cpc work later, we will make it a standard track RFC.
> I think we did that way for network asserted identity.
>=20
> How do people think about the above idea?
>=20
> Note that according RFC 3966, new tel URI RFC, an=20
> informational RFC is sufficient for optional tel URI parameters.
>=20
> Regards,
> Takuya
> >=20
> > I promised to start a thread on this and figure out what=20
> people want=20
> > to do with it and if we understand the requirements for it then=20
> > propose next steps.
> >=20
> > Fortunately, there is a thread going in the sip mailing=20
> list. I would=20
> > summarize that thread as:
> >=20
> > 1) the requirements and use case are varied and not clear
> > 2) they involved complex security and authorization issues
> > 3) SAML looks like it might help with many of them
> > 4) there are issues with extension and scope of future categories
> >=20
> > This makes me very uncomfortable with moving forward with this work =

> > right now as a WG item in IPTEL. What I suggest is that we=20
> shelve this=20
> > work for now and let sipping figure out what the=20
> requirements are and=20
> > if the solutions looks like we need to add something to the=20
> tel URL,=20
> > revive the draft and figure out what to do with it.
> >=20
> > What do people think of this? Folks ok with this? I will=20
> take silence=20
> > as well, just a fact of life on the iptel list :-)
> >=20
> > Cullen
> >=20
> >=20
> >=20
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
>=20
>=20
> --------
> 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
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20

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


From iptel-bounces@ietf.org  Thu Dec  9 08:35:07 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: jdrosen@cisco.com
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Subject: [Iptel] AW: [Sip] AW:  Calling Party's Category
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> -----Ursprungliche Nachricht-----
> Von: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Gesendet: Donnerstag, 9. Dezember 2004 05:24
> An: Jesske, Roland
> Cc: taylor@nortelnetworks.com; An.Nguyen@ncs.gov; sip@ietf.org;
> iptel@ietf.org; oran@cisco.com; Mpierce1@aol.com
> Betreff: Re: [Sip] AW: Calling Party's Category
> 
> 
> 
> 
> Jesske, R wrote:
> 
> > Dear Tom, all,
> > I see there are several questions with regard to the CPC. 
> Our approach is to use it for billing, charge info and 
> routing purposes.
> > 
> > Payphone:
> > 1. Payphone Calls are routed in an other manner than normal calls
> 
> In what way are they routed differently?

One of our main issues is interoperability between PSTN/ISDN and SIP/NGN. PSTN/ISDN use this indication also for routing.
In PSTN/ISDN the "data-call" can be only transferred over 64kbit/s trunks. This can be used for a efficient coos of the correct MGC/MG combination with the clearmode codec.

> 
> 
> > 2. Payphone Calls are billed differently compared with normal calls
> > 3. The charge information shown to the user of the payphone 
> is an other than for normal once
> > 4. Based on the CPC a blocking of numbers (e.G premium rate 
> or special destinations) can be performed
> > 5. Special accounting mechanisms for free phone numbers are possible
> 
> I think its also worth pointing out that the general topic of 
> "payphones" in sip is a complicated one, of which the cpc aspect - 
> asserting the trait of being a payphone - is one of the easier ones. 
> Figuring out a way to do the payments securely is a much more 
> challenging issue.
> 
> I also wanted to note that if there are requirements for endpoint 
> assertion of characteristics and capabilities, we have a spec 
> for that 
> too - namely rfc3840. The usage of SAML for doing the network 
> asserted 
> traits also seems right to me.

As Takuya said:  cpc parameter will be used with 
P-Asserted-Identity header. In this scenario, trust domains 
model described in RFC 3324 is enough solution for security 
and authorization issues. Anyway we will require cpc value to 
be asserted by the network as it is in ISUP.
> 
> 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  Thu Dec  9 12:38:55 2004
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From: "Chaidez, Fidencio (Fidencio)" <fchaidez@lucent.com>
To: "'iptel@ietf.org'" <iptel@ietf.org>
Subject: RE: [Iptel] Calling Party Category
Date: Thu, 9 Dec 2004 11:08:05 -0600 
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All,
It is clear that a solution is needed
in relation to supporting "cpc" type of information.=20
Many SIP-based service providers will likely be=20
making use of a walled-garden, trust-based, approach,
for which Takuya's proposal will be sufficient
to meet their requirements.  Because of the
non-general nature of this solution (as Takuya=20
pointed out) targetting a revived cpc draft as an=20
Informational RFC would be a good way to go. =20
Regards,
Fidencio

-----Urspr=FCngliche Nachricht-----
Von: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]Im Auftrag
von Jesske, Roland
Gesendet: Donnerstag, 9. Dezember 2004 14:06
An: sebastien.garcin@francetelecom.com; tu-sawada@kddi.com;
fluffy@cisco.com; iptel@ietf.org
Betreff: AW: [Iptel] Calling Party Category


Hi,
I would also like to support Takuya' and the comments from Sebatien =
proposal.

A in addition it could be used for routing purposes in a SIP Proxy to =
choose the best MGC/MG.=20

EG in PSTN/ISDN the "data-call" can be only transferred over 64kbit/s =
trunks. This can be used for a efficient coos of the correct MGC/MG =
combination with the clearmode codec.

Best Regards

Roland=20

> -----Urspr=FCngliche Nachricht-----
> Von: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]Im Auftrag
> von GARCIN Sebastien RD-CORE-ISS
> Gesendet: Donnerstag, 9. Dezember 2004 13:41
> An: Takuya Sawada; fluffy@cisco.com; iptel@ietf.org
> Betreff: RE: [Iptel] Calling Party Category
>=20
>=20
> Hi,
>=20
> I would like to support Takuya' proposal. I'd also like to=20
> point out that there are clear uses cases for the calling=20
> party's category and operators experience those requirements=20
> every day in their networks.=20
>=20
> Here is a brief summary of those that relate to the "payphone" value:
>=20
> Charging
> --------
> The cpc value "payphone" is used for charging purposes
>=20
> Services
> --------
> Some services (e.g. Call return, Freephone services) use the=20
> "payphone" value.
>=20
> The fact that we are in a pure SIP or ISUP environment does=20
> not change those requirements.
>=20
> Regards,
> S=E9bastien
> =20
>=20
> -----Message d'origine-----
> De : iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
> De la part de Takuya Sawada
> Envoy=E9 : jeudi 9 d=E9cembre 2004 13:22
> =C0 : fluffy@cisco.com; iptel@ietf.org
> Objet : Re: [Iptel] Calling Party Category
>=20
> Hi,
>=20
> Regarding 1), I think the requirements from those who reqire=20
> cpc parameter are simple and clear; carry the ISUP cpc=20
> information in SIP messages with tel URI.
> Use cases are varied as they are varied in ISUP.
>=20
> Regaerding 2), IMHO, cpc parameter will be used with=20
> P-Asserted-Identity header. In this scenario, trust domains=20
> model described in RFC 3324 is enough solution for security=20
> and authorization issues. Anyway we will require cpc value to=20
> be asserted by the network as it is in ISUP.
>=20
> I totally agree that 'because ISUP has it' is not sufficient=20
> to move it forward as an iptel WG item. During the discussion=20
> I feel it is very difficult to justify cpc work for iptel=20
> work item beyond 'because ISUP has it'.
>=20
> Now I propose that let's move cpc draft forward as an=20
> individual document and make it an informational RFC. It may=20
> be a good idea to add the limited scope of cpc and some use cases.
> If we can find more complete and general solution to cover=20
> cpc work later, we will make it a standard track RFC.
> I think we did that way for network asserted identity.
>=20
> How do people think about the above idea?
>=20
> Note that according RFC 3966, new tel URI RFC, an=20
> informational RFC is sufficient for optional tel URI parameters.
>=20
> Regards,
> Takuya
> >=20
> > I promised to start a thread on this and figure out what=20
> people want=20
> > to do with it and if we understand the requirements for it then 
> > propose next steps.
> >=20
> > Fortunately, there is a thread going in the sip mailing=20
> list. I would=20
> > summarize that thread as:
> >=20
> > 1) the requirements and use case are varied and not clear
> > 2) they involved complex security and authorization issues
> > 3) SAML looks like it might help with many of them
> > 4) there are issues with extension and scope of future categories
> >=20
> > This makes me very uncomfortable with moving forward with this work =

> > right now as a WG item in IPTEL. What I suggest is that we=20
> shelve this=20
> > work for now and let sipping figure out what the=20
> requirements are and=20
> > if the solutions looks like we need to add something to the=20
> tel URL,=20
> > revive the draft and figure out what to do with it.
> >=20
> > What do people think of this? Folks ok with this? I will=20
> take silence=20
> > as well, just a fact of life on the iptel list :-)
> >=20
> > Cullen
> >=20
> >=20
> >=20
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
>=20
>=20
> --------
> 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
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20

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

_______________________________________________
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Date: Thu, 09 Dec 2004 12:21:21 -0800
Subject: Re: [Iptel] Calling Party Category
From: Cullen Jennings <fluffy@cisco.com>
To: Takuya Sawada <tu-sawada@kddi.com>, "iptel@ietf.org" <iptel@ietf.org>
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Not really disagreeing with but adding a bit more inline ...

On 12/9/04 4:21 AM, "Takuya Sawada" <tu-sawada@kddi.com> wrote:

> Hi,
> 
> Regarding 1), I think the requirements from those who reqire
> cpc parameter are simple and clear; carry the ISUP cpc
> information in SIP messages with tel URI.
> Use cases are varied as they are varied in ISUP.
> 
It seems that this is better addressed by something like SIP-T that can
explicitly deal with caring all the ISUP that is needed.

> Regaerding 2), IMHO, cpc parameter will be used with
> P-Asserted-Identity header. In this scenario, trust domains
> model described in RFC 3324 is enough solution for security
> and authorization issues. Anyway we will require cpc value to
> be asserted by the network as it is in ISUP.
P-Asserted-Identity was an Information document not standards track because
it had a very limited deployment model. It solved a very specific problem of
dealing call trace for anonymous calls in a particular walled garden model.

> 
> I totally agree that 'because ISUP has it' is not sufficient
> to move it forward as an iptel WG item. During the discussion
> I feel it is very difficult to justify cpc work for iptel
> work item beyond 'because ISUP has it'.
> 
> Now I propose that let's move cpc draft forward as an individual
> document and make it an informational RFC. It may be a good idea
> to add the limited scope of cpc and some use cases.
> If we can find more complete and general solution to cover cpc work
> later, we will make it a standard track RFC.
> I think we did that way for network asserted identity.
> 
> How do people think about the above idea?
> 
> Note that according RFC 3966, new tel URI RFC, an informational
> RFC is sufficient for optional tel URI parameters.
> 
> Regards,
> Takuya
>> 
>> I promised to start a thread on this and figure out what people want to do
>> with it and if we understand the requirements for it then propose next
>> steps. 
>> 
>> Fortunately, there is a thread going in the sip mailing list. I would
>> summarize that thread as:
>> 
>> 1) the requirements and use case are varied and not clear
>> 2) they involved complex security and authorization issues
>> 3) SAML looks like it might help with many of them
>> 4) there are issues with extension and scope of future categories
>> 
>> This makes me very uncomfortable with moving forward with this work right
>> now as a WG item in IPTEL. What I suggest is that we shelve this work for
>> now and let sipping figure out what the requirements are and if the
>> solutions looks like we need to add something to the tel URL, revive the
>> draft and figure out what to do with it.
>> 
>> What do people think of this? Folks ok with this? I will take silence as
>> well, just a fact of life on the iptel list :-)
>> 
>> Cullen
>> 
>> 
>> 
>> _______________________________________________
>> 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
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel


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Date: Thu, 09 Dec 2004 12:23:24 -0800
Subject: Re: [Iptel] Calling Party Category
From: Cullen Jennings <fluffy@cisco.com>
To: GARCIN Sebastien RD-CORE-ISS <sebastien.garcin@francetelecom.com>,
        Takuya Sawada <tu-sawada@kddi.com>, "iptel@ietf.org" <iptel@ietf.org>
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Yes, I agree that there is an interest in billing calls from payphone
differently than from say, bordellos (thanks Dave :-). However, I'm sitting
here reading the charter of iptel and sipping and this seem like a great se=
t
of requirements to take to sipping and figure out how to solve the problem.
The connection to a tel URL seem very tenuous other than it's a place this
data could be stuck for transport.


On 12/9/04 4:41 AM, "GARCIN Sebastien RD-CORE-ISS"
<sebastien.garcin@francetelecom.com> wrote:

> Hi,
>=20
> I would like to support Takuya' proposal. I'd also like to point out that
> there are clear uses cases for the calling party's category and operators
> experience those requirements every day in their networks.
>=20
> Here is a brief summary of those that relate to the "payphone" value:
>=20
> Charging
> --------
> The cpc value "payphone" is used for charging purposes
>=20
> Services
> --------
> Some services (e.g. Call return, Freephone services) use the "payphone" v=
alue.
>=20
> The fact that we are in a pure SIP or ISUP environment does not change th=
ose
> requirements.
>=20
> Regards,
> S=E9bastien
> =20
>=20


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From: Cullen Jennings <fluffy@cisco.com>
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------ Forwarded Message
From: Cullen Jennings <fluffy@cisco.com>
Date: Thu, 09 Dec 2004 12:46:42 -0800
To: <proceedings@ietf.org>
Cc: Jonathan Rosenberg <jdrosen@cisco.com>, Cullen Jennings
<fluffy@cisco.com>
Subject: IPTEL Minutes  - IETF 61



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

CHAIRS: Jonathan Rosenberg <jdrosen@cisco.com>
Cullen Jennings <fluffy@cisco.com>
Note-taker - Spencer Dawkins <spencer@mcsr-labs.org>


Status Update: Since last meeting we have published TRIP MIB and CPL.
2806bis was published shortly after the meeting.

Enum dip indicator: Richard Stastny presented
draft-stastny-iptel-tel-enumdi. This would add a flag that indicated that
the ENUM dip had been done and did not need to be repeated. General
consensus in room was to accept this work. Some discussion about if there
should be a way to indicate that dips had been done to different roots.
Eventually decided that there are no current documents specifying ENUM for
multiple roots and we would not do this.

draft-ietf-iptel-tel-np-02.txt Revision is coming and it will be WGLC and
hopefully finished. Volunteers agreed to review. Some interested in
implementing this. 

draft-ietf-iptel-trunk-group-02.txt Volunteers agreed to review. Need to
WGLC. Some interested in implementing this.

Dial string draft: will go forward as individual submission

Calling party category draft: This has expired with very little response to
comments. There is substantial work left to finish this. One or two people
in room thought they might implement it. Not clear what the key uses of it
are. Will discuss on list if we want to move forward with this (as a WG
item) or not. It could always be done as individual.



More detailed notes follow:

* TRIP MIB, CPL published as RFCs since last meeting of  IPTel
* rfc2096bis in RFC editor queue
20 min Enum Indicator Richard Stastny -
draft-stastny-iptel-tel-enumdi-00.txt
* Also heads-uped in ENUM
* Add "enumdi" to E.164 tel URI saying "we've already done an  ENUM lookup,
so you don't have to try again when you're resolving the  URI"
* "MUST NOT attempt e164.arpa resolution" requirement also  discussed in
ENUM 
* Who can add "enumdi?" - if you would do ENUM, you can add it,  and an
endpoint can add it to the original URI before anyone tries to resolve  it
* Cullen points out that this is not a mandatory parameter, so  network
elements that don't understand it will still attempt the  lookup
* <>This is good. Why limit it to e164.arpa? There will  be other
lightweight mechanisms, too.
* <><>2806bis should probably also be a SHOULD -  how consistent do we need
to be?Makes sense, but do we need to say this in the  current draft?
* <><>If we allow this for multiple routes, we need  to say what routes have
been queried. Should we add a translation counter to  help people realize a
URI has been changed? How does transitive trust work?  And please don't
change semantics in a future rev - do it now, or don't do  it.
* Allow other roots in the first version, too - won't this slow  the
document down on the approval process?
* If you know who did the lookup, you can decide whether to  trust them
based on that knowledge.
* Leave it like is, or extend it (loop counter or something  else)? Not a
significant difference in sense of the room. We're trying to shut  the
working group down, so the authors will be guided by whatever the chairs
ask for. 
* We don't actually have a spec that says you can legitimately  query
against another domain. Jonathan doesn't get it... no harm in redoing  the
lookup ("dip"). 
* Punt this question to the ENUM working group chair :-) -  service
providers would like to do something like this. Would like to continue  as a
working group document, would like to rev quickly and WGLC  quickly.
* Ignoring this is like ignoring NATs - don't put our heads in  the sand.
We'll block for six months to a year if we add alternative  roots.
* Accept as WG item with no changes - humm? Sense of room is  "yes". [still
needs to be confirmed on the list, right?]
30 min Plan for concluding work - Chairs Et.Al.

draft-ietf-iptel-tel-np-02.txt
draft-ietf-iptel-trunk-group-02.txt
draft-ietf-iptel-tgrep-04.txt
* Number portability draft revision is coming and addresses  comments to
date. 
* Volunteer reviewers were selected for NP and trunk group  drafts.
* Lite interest in implementing NP extensions and trunk group  drafts. Will
be immediately WGLCed. Reviewers, please report back during that  timeframe.
Dial string draft
* will go forward as individual submission
Calling party category draft
* has expired, no responses to comments received
* some concerns raised (trust boundaries, asserted identity) -  not near
done with this one.
* We discussed making this a WG item many moons ago, and  decided not to do
it for pretty much the same reasons being raised  now.
* This work will take some time to get right. If the working  group takes it
on, we have to expect to come to consensus before we can  close.
* This work is useful, but maybe doesn't matter whether it's a  WG item or
not. 
* Who would implement?  maybe one or two in the  room
* Not sure who needs this - what problem is being solved?  duplicate
parameters, endpoint identification - what? Maybe emergency services  and
coin phones, or something like that. Check with development groups and
discuss this on the list in a couple of weeks after end of IETF.
ENUM Dip? 
* We think we decided to do this at the beginning of the  meeting.


------ End of Forwarded Message


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From iptel-bounces@ietf.org  Fri Dec 10 01:47:17 2004
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From: "Jesske, R" <R.Jesske@t-com.net>
To: fluffy@cisco.com, tu-sawada@kddi.com, iptel@ietf.org
Subject: AW: [Iptel] Calling Party Category
Date: Fri, 10 Dec 2004 07:03:01 +0100
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Some comments to your addings.

Using SIP-T or SIP-I assumes that the applications have to support the full ISUP stack, but we want to use SIP.

The P-Asserted-ID is used within standards from 3GPP, ITU-T, ETSI, T1S1... so I don't think that it is very limited...

I don't think that the use is very limited e.G:

-Billing
-Routing based on originating indications (e.G. special domainnames or numbergroups)
-Services selective Call Forwarding
-incomming screening of Calls based on a trusted number 
...

Best Regards

Roland

> -----Ursprungliche Nachricht-----
> Von: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]Im Auftrag
> von Cullen Jennings
> Gesendet: Donnerstag, 9. Dezember 2004 21:21
> An: Takuya Sawada; iptel@ietf.org
> Betreff: Re: [Iptel] Calling Party Category
> 
> 
> 
> Not really disagreeing with but adding a bit more inline ...
> 
> On 12/9/04 4:21 AM, "Takuya Sawada" <tu-sawada@kddi.com> wrote:
> 
> > Hi,
> > 
> > Regarding 1), I think the requirements from those who reqire
> > cpc parameter are simple and clear; carry the ISUP cpc
> > information in SIP messages with tel URI.
> > Use cases are varied as they are varied in ISUP.
> > 
> It seems that this is better addressed by something like 
> SIP-T that can
> explicitly deal with caring all the ISUP that is needed.
> 
> > Regaerding 2), IMHO, cpc parameter will be used with
> > P-Asserted-Identity header. In this scenario, trust domains
> > model described in RFC 3324 is enough solution for security
> > and authorization issues. Anyway we will require cpc value to
> > be asserted by the network as it is in ISUP.
> P-Asserted-Identity was an Information document not standards 
> track because
> it had a very limited deployment model. It solved a very 
> specific problem of
> dealing call trace for anonymous calls in a particular walled 
> garden model.
> 
> > 
> > I totally agree that 'because ISUP has it' is not sufficient
> > to move it forward as an iptel WG item. During the discussion
> > I feel it is very difficult to justify cpc work for iptel
> > work item beyond 'because ISUP has it'.
> > 
> > Now I propose that let's move cpc draft forward as an individual
> > document and make it an informational RFC. It may be a good idea
> > to add the limited scope of cpc and some use cases.
> > If we can find more complete and general solution to cover cpc work
> > later, we will make it a standard track RFC.
> > I think we did that way for network asserted identity.
> > 
> > How do people think about the above idea?
> > 
> > Note that according RFC 3966, new tel URI RFC, an informational
> > RFC is sufficient for optional tel URI parameters.
> > 
> > Regards,
> > Takuya
> >> 
> >> I promised to start a thread on this and figure out what 
> people want to do
> >> with it and if we understand the requirements for it then 
> propose next
> >> steps. 
> >> 
> >> Fortunately, there is a thread going in the sip mailing 
> list. I would
> >> summarize that thread as:
> >> 
> >> 1) the requirements and use case are varied and not clear
> >> 2) they involved complex security and authorization issues
> >> 3) SAML looks like it might help with many of them
> >> 4) there are issues with extension and scope of future categories
> >> 
> >> This makes me very uncomfortable with moving forward with 
> this work right
> >> now as a WG item in IPTEL. What I suggest is that we 
> shelve this work for
> >> now and let sipping figure out what the requirements are and if the
> >> solutions looks like we need to add something to the tel 
> URL, revive the
> >> draft and figure out what to do with it.
> >> 
> >> What do people think of this? Folks ok with this? I will 
> take silence as
> >> well, just a fact of life on the iptel list :-)
> >> 
> >> Cullen
> >> 
> >> 
> >> 
> >> _______________________________________________
> >> 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
> > 
> > _______________________________________________
> > 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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Subject: Re: AW: [Iptel] Calling Party Category
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On 12/9/04 10:03 PM, "Jesske, R" <R.Jesske@t-com.net> wrote:

> Using SIP-T or SIP-I assumes that the applications have to support the full
> ISUP stack, but we want to use SIP.

Yes - I agree, but this is the grief I am having is an arguments of the
form:

People come and say please add parameter X from ISUP. Then folks say sure,
why what do you want to do with it? Then the answer is well it is in ISUP so
we must have it. 

If this is true for all values of X then you need all of ISUP and just go do
ISUP. 

If their is some feature you want (that ISUP can do), like special billing
from a payphone, then don't even mention ISUP. Say "I would like to be able
to bill payphone users extra" - take that to the sipping WG and I predict
people will come up with a solution to the problem. 


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Cullen,

I was hoping to have the next rev out by early Jan.

-Manjax

Cullen Jennings wrote:

>
> Any ETA on a new rev of this draft?



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Cullen,<br>
<br>
I was hoping to have the next rev out by early Jan.<br>
<br>
-Manjax<br>
<br>
Cullen Jennings wrote:<br>
<blockquote type="cite" cite="midBDDCF108.1DA35%25fluffy@cisco.com">
  <title>Re: tgrep 04 comments</title>
    <font face="Verdana, Helvetica, Arial"><span
 style="font-size: 12px;"><br>
 Any ETA on a new rev of this draft?<br>
 </span></font> </blockquote>
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Jyoti,

There's no attribute currently to express "charge" or "pricing" 
information associated with a route. It was considered at some point for 
TGREP but it was dropped b'coz of the complexity of representing it. For 
instance, the price offered by a provider can vary by time of day,  it 
can potentially be based on an agreement between two service providers 
interconnecting with each other. Least Cost routing choices and Grades 
of service offered by a carrier can affect pricing. There could be other 
factors as well. Expressing this complex interplay between different 
factors that determine pricing is non-trivial to represent.

-Manjax

Jyoti wrote:

> Hello All,
> For TGrep and TRIP - pls can you let me know if any charging data 
> framework is available i.e if any charge related information can be 
> updated , stored and queried.
> Thanks
> Jyoti
>
>     -----Original Message-----
>     *From:* iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] *On
>     Behalf Of *Cullen Jennings
>     *Sent:* Thursday, December 09, 2004 7:27 AM
>     *To:* Manjunath S Bangalore
>     *Cc:* Dhaval N Shah; iptel@ietf.org; Hussein F Salama;
>     rajneesh@cisco.com
>     *Subject:* [Iptel] Re: tgrep 04 comments
>
>
>     Any ETA on a new rev of this draft?
>


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Jyoti,<br>
<br>
There's no attribute currently to express "charge" or "pricing" information
associated with a route. It was considered at some point for TGREP but it
was dropped b'coz of the complexity of representing it. For instance, the
price offered by a provider can vary by time of day,&nbsp; it can potentially
be based on an agreement between two service providers interconnecting with
each other. Least Cost routing choices and Grades of service offered by a
carrier can affect pricing. There could be other factors as well. Expressing
this complex interplay between different factors that determine pricing is
non-trivial to represent.<br>
<br>
-Manjax<br>
<br>
Jyoti wrote:<br>
<blockquote type="cite"
 cite="mid001001c4ddb9$49aefeb0$d204120a@china.huawei.com">  
  <meta http-equiv="Content-Type" content="text/html; ">
  <title>Message</title>
   
  <meta content="MSHTML 6.00.2800.1458" name="GENERATOR">
 
  <div><span class="236172706-09122004"><font face="Arial"
 color="#0000ff" size="2">Hello  All,</font></span></div>
 
  <div><span class="236172706-09122004"><font face="Arial"
 color="#0000ff" size="2">For  TGrep and TRIP - pls can you let me know if
any charging data framework is  available i.e if any charge related information
can be updated , stored and  queried.</font></span></div>
 
  <div><span class="236172706-09122004"><font face="Arial"
 color="#0000ff" size="2">Thanks</font></span></div>
 
  <div><span class="236172706-09122004"><font face="Arial"
 color="#0000ff" size="2">Jyoti  </font></span></div>
 
  <blockquote style="margin-right: 0px;">      
    <div class="OutlookMessageHeader" lang="en-us" dir="ltr"
 align="left"><font face="Tahoma" size="2">-----Original Message-----<br>
    <b>From:</b>    <a class="moz-txt-link-abbreviated" href="mailto:iptel-bounces@ietf.org">iptel-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:iptel-bounces@ietf.org">mailto:iptel-bounces@ietf.org</a>]
    <b>On Behalf Of    </b>Cullen Jennings<br>
    <b>Sent:</b> Thursday, December 09, 2004 7:27    AM<br>
    <b>To:</b> Manjunath S Bangalore<br>
    <b>Cc:</b> Dhaval N Shah;    <a class="moz-txt-link-abbreviated" href="mailto:iptel@ietf.org">iptel@ietf.org</a>; Hussein F Salama; <a class="moz-txt-link-abbreviated" href="mailto:rajneesh@cisco.com">rajneesh@cisco.com</a><br>
    <b>Subject:</b>    [Iptel] Re: tgrep 04 comments<br>
    <br>
    </font></div>
    <font face="Verdana, Helvetica, Arial"><span
 style="font-size: 12px;"><br>
Any ETA on    a new rev of this draft?<br>
    </span></font></blockquote>
</blockquote>
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On 12/11/04 12:04 AM, "Manjunath Bangalore" <manjax@cisco.com> wrote:

> Cullen,
> 
> I was hoping to have the next rev out by early Jan.
> 
> -Manjax
> 
> Cullen Jennings wrote:
>>  Re: tgrep 04 comments
>>  Any ETA on a new rev of this draft?
>>   
> 
> 

Great -  thank you - I would really like to get this done well before
everyone getting busy writing drafts fro next IETF.


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<FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:12.0px'>On 12=
/11/04 12:04 AM, &quot;Manjunath Bangalore&quot; &lt;manjax@cisco.com&gt; wr=
ote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYL=
E=3D'font-size:12.0px'>Cullen,<BR>
<BR>
I was hoping to have the next rev out by early Jan.<BR>
<BR>
-Manjax<BR>
<BR>
Cullen Jennings wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STYL=
E=3D'font-size:12.0px'> Re: tgrep 04 comments <BR>
&nbsp;Any ETA on a new rev of this draft?<BR>
&nbsp;&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:12.0px'><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:12.0px'><BR>
Great - &nbsp;thank you - I would really like to get this done well before =
everyone getting busy writing drafts fro next IETF.<BR>
</SPAN></FONT>
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I would also like to keep SIP clean, really.
I would be happy if we could throw ISUP away and go with pure SIP.
We know it is not possible.

Can we understand cpc work along the following way?

- It is NOT a SIP extension. cpc parameter does not affect any SIP behaviour.
  (Actually, it is a tel URI extension)
- It is just additional information to be added to the resource indicated
  by tel URI in a certain circumstances.
- Billing from a payphone is a use case, not a requirement. Requirement is
  that cpc parameter value range should cover all the characteristics used 
  in ISUP network at minimum. It is not the work for SIPPING WG.

---
For the general problem, adding other parameters X from ISUP, I think
we should be careful in doing that. I hope there are not so much candidates,
though I can imagine ATP, access transport prameter, and UUS, user-to-user
information.....
I noticed that Q.1980.1, which is in progress in ITU-T, is related to that
type of problem. I am not so familiar with that work that I am not sure it
will fit and cover all the problems.

Regards,
Takuya

> On 12/9/04 10:03 PM, "Jesske, R" <R.Jesske@t-com.net> wrote:
> 
> > Using SIP-T or SIP-I assumes that the applications have to support the full
> > ISUP stack, but we want to use SIP.
> 
> Yes - I agree, but this is the grief I am having is an arguments of the
> form:
> 
> People come and say please add parameter X from ISUP. Then folks say sure,
> why what do you want to do with it? Then the answer is well it is in ISUP so
> we must have it. 
> 
> If this is true for all values of X then you need all of ISUP and just go do
> ISUP. 
> 
> If their is some feature you want (that ISUP can do), like special billing
> from a payphone, then don't even mention ISUP. Say "I would like to be able
> to bill payphone users extra" - take that to the sipping WG and I predict
> people will come up with a solution to the problem. 
> 


--------
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: "Gaurav.Kheterpal" <Gaurav.kheterpal@conexant.com>
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Date: Tue, 14 Dec 2004 12:41:06 +0530
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Subject: [Iptel] FW: SIP - TEL URI Implementation Query
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Hello,

I am currently implementing support for <tel:> URI's in our User Agent as
per RFC 3966/ RFC 2806-bis standards. I had the following doubts regarding
the same. Any inputs will be highly appreciated.

1. Contact Header: As per RFC 3261 Section 8.1.1.8, it mentions that this
URI must be in either SIP/ SIPS format. In Section 10.2.1, it is mentioned
that for registrations, the Contact header field can contain any URI Scheme
(e.g the tel URL, RFC 2806 [9]) or email addresses (with a mailto URL, RFC
2368 [32]) as Contacts for an address-of-record.

Does this mean that the 'Contact' field in an INVITE message must always be
a SIP/SIPS URI while in a REGISTER message, it can be a SIP/SIPS/TEL/Mailto
URI ?

2. Besides 'Request URI' and 'To:' header fields, can the TEL URI be present
in any other INVITE header values (e.g. Via, Route Header, Contact etc). In
other words, which fields in the INVITE request does the proxy scan for
possible tel->sip translation using ENUM/ any other mechanism ? In case,
there is no other field which contains this information, how is the TEL URI
information propagated across intermediate proxies when the UA is configured
with a outbound proxy having a defined route set. If the defined route set
is proxy 1-> proxy 2 in the below diagram, the request will be formed using:

INVITE sip:proxy1
Route: <sip:proxy2>
To: <tel:1234>

User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |

Does this mean that in such a case, the only parameter which contains the
TEL URI is the 'To' field. As I understand, this field is never processed by
the proxies unless they implement privacy/ RFC 3324(5) ??

I am current not aware of any proxy which is capable of handling TEL URI's,
any  information/ links regarding the same will be highly useful.

Thanks in advance for your help.

regards,

Gaurav


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From: "Jesske, R" <R.Jesske@t-com.net>
To: An.Nguyen@ncs.gov, taylor@nortelnetworks.com
Subject: AW: [Sip] RE: [Iptel] Calling Party's Category
Date: Wed, 15 Dec 2004 08:18:36 +0100
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Hello An
The below mentioned draft is expiered. There is no work in SIP or IPTEL =
at the moment, but we hope it will.
We brought up the issue, becuse we see the demand on such a element in =
SIP.

Best Regerds

Roland

> -----Urspr=FCngliche Nachricht-----
> Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org]Im Auftrag von
> Nguyen, An
> Gesendet: Dienstag, 30. November 2004 19:00
> An: 'Tom Taylor'
> Cc: iptel@ietf.org; sip@ietf.org; Jesske, Roland
> Betreff: [Sip] RE: [Iptel] Calling Party's Category
>=20
>=20
> Mr. Taylor;
>=20
> I am just curious ... What is the status of the Internet Draft?
>=20
> An
>=20
>=20
>=20
> -----Original Message-----
> From: Tom Taylor [mailto:taylor@nortelnetworks.com]
> Sent: Tuesday, November 30, 2004 10:07 AM
> To: Jesske, R
> Cc: iptel@ietf.org; sip@ietf.org
> Subject: Re: [Iptel] Calling Party's Category
>=20
>=20
> The question was what the requirement is for this sort of=20
> indication.  How
> is it used?
>=20
> 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.=20
> > We are still interested in such kind of originating=20
> indication if the
> call/communication is coming from a normal SIP user or a SIP=20
> -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=20
> > Am 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
>=20
> --=20
> Tom Taylor
> Carrier VoIP Standards Development
> Nortel Networks
> Phone +1 613 763 1496  (ESN 393-1496)
> E-mail: taylor@nortelnetworks.com
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20
>=20
>=20
> _______________________________________________
> 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
>=20

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Cc: James Yu <james.yu@neustar.com>
Subject: [Iptel] Review of draft-ietf-iptel-tel-np-03.txt
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Greetings

Below, please find review comments for draft-ietf-iptel-tel-np-03.txt.=20
In general, I think the document is close to being done, but there are a=20
few minor issues to address as well as some nits.


Non-Editorial
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
- Section 4, ABNF
The production for "global-hex-digits" ends with "*phonedigit-hex". I=20
don't understand what that is for, not least considering the ensuing=20
text for "local-rn"
<quote>
   For a "local-rn", the routing number in the "rn" parameter MUST be
   meaningful in terms of "rn-context".  For example, if a national
   routing number is in the "rn" parameter, the "rn-context" MUST
   contain a valid E.164 country code after "+" if it is in the
   "global-hex-digits" format.
</quote>


Section 5.1, 6th paragraph
"The network node SHALL ignore the "cic" parameter if it identifies a=20
carrier or service provider associated with that node, or if that=20
parameter contains a code for indicating that a geographic number is=20
supplied (e.g., +1-0110 means "local, translated geographical telephone=20
number provided"). ". Given the normative requirement, we either need=20
some more text or references to explain how we determine if the=20
parameter contains a code for indicating that a geographic number is=20
supplied.

Section 5.1, 6th paragraph
"The network node SHALL remove the "cic" parameter and look for the "rn"=20
parameter for making the routing decision." I don't see why the network=20
node SHALL remove the "cic" parameter and would in fact argue for at=20
most a MAY. I agree it shall look for the "rn" parameter for making the=20
routing decision though.

Section 5.1, 9th and 10th paragraph.
Again, I don't see why "the network SHALL remove the "rn" parameter".

Section 5.2.4, 1st paragraph
"A Public Switched Telephone Network (PSTN) gateway needs to convert
   between SS7 ISUP and the VoIP protocol such as SIP or H.323.  This
   type of network node SHALL add the corresponding information from
   the ISUP to the defined parameters to the "tel" URI for routing and
   the "tel" URI associated with the caller and vice versa."
I disagree with the latter part ("and the "tel" URI associated with the=20
caller and vice versa"). We are getting into privacy and security=20
related issues and hence cannot make it mandatory for all.

Section 7
Need to add security considerations for using the extensions for the=20
caller (JIP considerations)


Editorial and/or nits:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
- Global
Three spaces after period in several places.
Not sure if formatting adheres to RFC Editor guidelines (at least I=20
can't get it to print with proper page breaks)

- Abstract
Should not describe the actual extension parameters in the abstract but=20
rather describe the problem and outline the solution.

- Abstract =20
"that NP database dip" =3D> "that an NP database dip"
                                ^^
- Section 2, 1st paragraph
"It has been identified that NP has impacts on several works-in-progress=20
at the IETF." =3D>

 "NP impacts call signaling." or something like that.

- Section 2, 3rd paragraph
"call set up" =3D> "call setup"

- Section 2, 5th paragraph
The sentence
    "Section 5 describes the rules for a network node that deals with=20
some or all of the defined parameters in a "tel" URI"
is not accurate since we are only talking about some specific extensions=20
here. The phrase "defined parameters" is used throughout the document to=20
refer to the extensions defined in this document. I'd suggest using=20
another more accurate phrase throughout.

- Section 4, 1st paragraph
Add some words explaining how this ABNF simply defines new parameters in=20
accordance with the ABNF for "tel" URI provided in RFC 3966.

- Section 4, 5th paragraph
"...the routing number in the "rn" parameter MUST be meaningful in terms=20
of "rn-context". Although the example that follows helps clarify it for=20
E.164 numbers, it is in general not well defined what "meamingful"=20
means. Please rephrase to make it clearer.

- Section 4, last paragraph
"...the CIC value in the "cic" parameter MUST be meaningful in terms of=20
"cic-context". Semantics unclear - please rephrase to make it clearer.

Section 5.1, 4th paragraph
"or routing information" =3D> "or routing number information" (?)

Section 5.1, last paragraph
"making routing decision" =3D> "making routing decisions"
                                                     ^
"set to do so or may route the call to a designated network node that=20
will access the NP database or may route the call" =3D>
              ^^
"set to do so, it may route the call to a designated network node that=20
will access the NP database, or it may route the call"
            =20
^^^                                                                      =
         =20
  ^    ^^

Section 5.2.2, 6th paragraph
"telephone number that is the typical case for the second freephone=20
database access, the" =3D>

"telephone number (which is the typical case for the second freephone=20
database access), the"
                 =20
^^^^^^                                                             ^

Section 5.2.3, 2nd paragraph
"For example, if the network node is a PSTN gateway that receives an=20
ISUP message that contains the JIP, the "rn" parameter in the "tel" URI=20
would normally contain the correct location information." Please=20
rephrase (it is the ISUP message that contains the correct location=20
information which is therefore placed in the "rn" parameter in the "tel"=20
URI).


Section 5.2.4, 2nd paragraph
"put it after the "tel:" in the "tel" URI that is used for routing" =3D>

"put it in the global-number-digits or local-number-digits (see RFC=20
3966) part of the "tel" URI that is used for routing"
       =20
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

Section 5.2.4, 3rd paragraph
"put it after "tel:" in the "tel" URI that is used for routing" =3D>

"put it in the global-number-digits or local-number-digits (see RFC=20
3966) part of the "tel" URI that is used for routing"
       =20
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

Section 5.2.4, 3rd paragraph
"No "rn" SHALL appear in the "tel" URI." =3D>

"An "rn" SHALL NOT appear in the "tel" URI."
 ^^            ^^^

Section 5.2.4, 3rd paragraph
"It is outside the scope of this document how to include the "rn"=20
parameter if the local policies require the network node to do so." I=20
don't follow - the previous sentence just said that you are not allowed=20
to do so.

Section 6, example B
"provider=C6s" =3D> "provider's"

Section 6, example E
"   E. A "tel" URI, tel:+1-202-533-1234;rn=3D+1-202-000-0000;npdi, contai=
ns
     an invalid routing number (e.g., no routing information on "+1-
     202-000"), "
Example seems to assume prefix-based routing - might be good to clarify=20
that.

Section 8 and 9
There are gaps in the reference numbers. In general, it would probably=20
be better to use symbolic references instead of plain numbers.
2806bis-09 is now RFC 3966.


-- Flemming
=20


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From: "Brian Rosen" <br@brianrosen.net>
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Date: Mon, 20 Dec 2004 19:23:31 -0500
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Subject: [Iptel] Review of draft-ietf-iptel-trunk-group-02
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I've reviewed this draft for readiness to advance to the IESG.

I don't find any substantive issues with the draft from a protocol point of
view.

As RFC text, the following changes would make a better document

1. Define trunk group, at least informally, at the beginning of the Problem
Statement.

2. The problem statement could use substantial rewording.  Pointing out the
need to specify a trunk group is sufficient.  At the very least, that point
should be ahead of where you put the parameter.  Think about how this reads
3 years from now as an RFC.

A self reference should use "this document", although I think the RFC editor
fixes things like that.

3. I think REQ1 would be better expressed as "A trunk group parameter MUST
be defined as an extension to the TEL uri".  That makes the second paragraph
make more sense.

4. The last paragraph in 4.2 is misplaced.  It's actually normative text for
the parameter itself.  You can discuss a solution here, but you can't make
it be the MUST language for the extension itself.  This might be the only
thing I think we have to fix.  This mistake is common to several of the
requirements, but this one is the only one I spotted that does not have
normative text elsewhere.

5. The local/global discussion is confusing, because "global" in this
context is not the "global" in a tel uri.  As best I can tell, you mean to
use "phone context" with some guaranteed globally unique name space, like a
domain name.  You can't just say "use phone-context" and get global out of
that.  More text is needed to explain what you are trying to accomplish with
global context.

6. Consider changing the language at the end of 4.2 to mirror that in the
tel uri.  RFC3966 says "If a recipient of
   the URI intends to place a call to the local number, it MUST
   understand the context and be able to place calls within that
   context."

It doesn't say you ignore phone context.  Actually, it's silent on what you
do when you get one that you don't understand.  I'm not really thrilled with
letting the call go through ignoring the parameter.

7. In the security section, it says if you don't want to reveal trunk group,
hide it.  This puzzles me.  How do you hide a parameter to a tel uri?  Did
you mean "not use it"?  

8. You can now update the reference to 2806bis as RFC3966 (YAY!)

9. Nits
Section 3, could use a spacer between the two definitions
	Also, might want to lose the subgroup exception since you didn't
		define subgroup
Section 4.1 lose the reference to RFC2806 (it gets deleted entirely from the
reference section), since 3966 obsoletes it.

Section 5  Missing comma:
GW3 has only one trunk group configured, TG3-1
                                       ^

The following text is awkward:
   On requests arriving to the proxy from GW1, the proxy will have
   access to the trunk group TG1-1 if the request arrived on that
   particular trunk

The proxy always has access to trunk group TG1-1.  If a call arrives on that
group and the proxy wants to let a downstream entity know that, then it puts
the parameter on the Contact header.

Section 8
Extra space in
   security concerns beyond those already enumerated in [3] .
                                                           ^

Section 11 should be dropped sometime

Brian



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From iptel-bounces@ietf.org  Tue Dec 21 10:29:47 2004
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Brian Rosen wrote:
> I've reviewed this draft for readiness to advance to the IESG.
> 
> I don't find any substantive issues with the draft from a protocol 
> point of view.

Brian:

I very much appreciate your time on this.

Your review comments will be folded in the next rev.

Thank you for a thorough review!

Regards,

- 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  Thu Dec 23 22:22:29 2004
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Date: Thu, 23 Dec 2004 05:47:15 -0800
Subject: Re: [Iptel] FW: SIP - TEL URI Implementation Query
From: Cullen Jennings <fluffy@cisco.com>
To: "Gaurav.Kheterpal" <Gaurav.kheterpal@conexant.com>,
        "sip-implementors@cs.columbia.edu" <sip-implementors@cs.columbia.edu>
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This question would probably be better on sip-implementors so if you reply,
please remove iptel. Bit of an answer inline.


On 12/14/04 12:11 AM, "Gaurav.Kheterpal" <Gaurav.kheterpal@conexant.com>
wrote:

> Hello,
> 
> I am currently implementing support for <tel:> URI's in our User Agent as
> per RFC 3966/ RFC 2806-bis standards. I had the following doubts regarding
> the same. Any inputs will be highly appreciated.
> 
> 1. Contact Header: As per RFC 3261 Section 8.1.1.8, it mentions that this
> URI must be in either SIP/ SIPS format. In Section 10.2.1, it is mentioned
> that for registrations, the Contact header field can contain any URI Scheme
> (e.g the tel URL, RFC 2806 [9]) or email addresses (with a mailto URL, RFC
> 2368 [32]) as Contacts for an address-of-record.
> 
> Does this mean that the 'Contact' field in an INVITE message must always be
> a SIP/SIPS URI while in a REGISTER message, it can be a SIP/SIPS/TEL/Mailto
> URI ?

Yes. Pretty weird Eh? The contact header in SIP got overloaded for too many
different uses but that is the way it is. Also, a 302 can have a Contact
header field value that is any URI (such as tel, email).

> 
> 2. Besides 'Request URI' and 'To:' header fields, can the TEL URI be present
> in any other INVITE header values (e.g. Via, Route Header, Contact etc).

It can be in a From header and the RFC 3325 P-Asserted-Identity header. It
should never be in a Via.

> In
> other words, which fields in the INVITE request does the proxy scan for
> possible tel->sip translation using ENUM/ any other mechanism ?

Only the request URI. The others should never be changed by a proxy.

> In case,
> there is no other field which contains this information, how is the TEL URI
> information propagated across intermediate proxies when the UA is configured
> with a outbound proxy having a defined route set. If the defined route set
> is proxy 1-> proxy 2 in the below diagram, the request will be formed using:
> 
> INVITE sip:proxy1
> Route: <sip:proxy2>
> To: <tel:1234>
>

Proxies never route based on the To. Again, weird but true. So to make
something like this work the path below happen, you would have. The routes
are used before the R-URI. (I'm assuming that proxy 2 will route 1234 to
user B)

INVITE tel:1234
Route: <sip:proxy1;lr>, <sip:proxy2;lr>
 
> User A          Proxy 1          Proxy 2          User B
>      |                |                |                |
>      |   INVITE F1    |                |                |
>      |--------------->|                |                |
> 
> Does this mean that in such a case, the only parameter which contains the
> TEL URI is the 'To' field.
No the request URI should have had the tel

> As I understand, this field is never processed by
> the proxies unless they implement privacy/ RFC 3324(5) ??
Yes - even then the the To field is never changed by a proxy.

> 
> I am current not aware of any proxy which is capable of handling TEL URI's,
> any  information/ links regarding the same will be highly useful.

I think there several proxies that can do tel and ENUM - there are also
several GWs capable of tel.

> 
> Thanks in advance for your help.
> 
> regards,
> 
> Gaurav
> 

Hope that helps, Cullen


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Cc: tu-sawada@kddi.com, Shan Lu <shanlu@sentito.com>
Subject: [Iptel] Finalizing trunk-group I-D
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Folks:

As part of WGLC, I have received the nits comments from the WG
reviewers who kindly volunteered their time in reviewing
the trunk-group I-D.  I will be issuing an updated rev
shortly.

There is one pending item that was being discussed on the
list; I will like to reach conclusion on this.  The issue
is summarized in the following email from the WG archives:

http://www1.ietf.org/mail-archive/web/iptel/current/msg00726.html

In the interest of moving forward and concluding this particular
I-D, if I do not hear any counter proposals, I will consider
the issue resolved as outlined in the email.

Thank you for your patience in indulging this discussion, and
happy holidays.

Regards,

- 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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Date: Mon, 27 Dec 2004 12:21:24 -0700
From: Cullen Jennings <fluffy@cisco.com>
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Subject: [Iptel] No meeting at next IETF?
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I am trying to figure out if we need agenda time at the next IETF. My
current assumption is that we do not need agenda time to finish any of our
current work so I am not planning on requesting any. If you think there is
some draft that we can not finish "on the list" without having agenda time,
let me know before the end of January.

Thanks, Cullen



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From iptel-bounces@ietf.org  Wed Dec 29 18:14:11 2004
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>, "'IETF IPTEL WG'" <iptel@ietf.org>
Subject: RE: [Iptel] Finalizing trunk-group I-D
Date: Wed, 29 Dec 2004 17:51:36 -0500
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Vijay,

I like your proposal in spirit, but am weary of the (over)use of
"anonymous". For everybody's benefit, you propose the following format =
when
calling party number is absent:

Contact: =
<sip:anonymous;phone-context=3Dexample.com;tgrp=3Dfoo@example.com>,

where the origination TG is "foo".

RFC3398 (and 3261) hinted that "sip:anonymous <anonymous@...>" SIP URI =
has
privacy implications. To quote RFC3398:

"If the
   presentation indicators are set to 'presentation restricted', then a
   special URI is created by the gateway which communicates to the far
   end that the caller's identity has been omitted.  This URI SHOULD be
   a SIP URI with a display-name and username of 'Anonymous', e.g.:

   From: Anonymous <sip:anonymous@anonymous.invalid>"

There are existing implementations where "anonymous" =
display-name/userinfo
can trigger special processing on both From and Contact headers.

OTOH, what we want is to convey to the recipient of Contact header the
following: the SIP URI userinfo in Contact is a "quasi-" tel URI in that =
any
subsequent fields following the first semi-colon must be treated as tel =
URI
parameters, including ";tgrp=3D".=20

I believe a keyword different than "anonymous" serves this purpose =
better.
For example, we don't have to consider the subtle difference between
"presentation restricted" and "address unavailable". My suggestion is
"telurifmt", i.e.:

Contact: <sip:telurifmt;tgrp=3Dfoo@example.com>.

Note since this is not tel URI, there should not be requirement on =
mandatory
"phone-context" parameter.

Comments?

Regards,

Shan Lu

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Vijay K. Gurbani
>Sent: Monday, December 27, 2004 11:20 AM
>To: IETF IPTEL WG
>Cc: tu-sawada@kddi.com; Shan Lu
>Subject: [Iptel] Finalizing trunk-group I-D
>
>
>Folks:
>
>As part of WGLC, I have received the nits comments from the WG
>reviewers who kindly volunteered their time in reviewing
>the trunk-group I-D.  I will be issuing an updated rev
>shortly.
>
>There is one pending item that was being discussed on the
>list; I will like to reach conclusion on this.  The issue
>is summarized in the following email from the WG archives:
>
>http://www1.ietf.org/mail-archive/web/iptel/current/msg00726.html
>
>In the interest of moving forward and concluding this particular
>I-D, if I do not hear any counter proposals, I will consider
>the issue resolved as outlined in the email.
>
>Thank you for your patience in indulging this discussion, and
>happy holidays.
>
>Regards,
>
>- 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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