Subject: Re: [Sip] GRUU and Anonymity

From: "Nieminen Klaus" <[email-masked]>
Date: Wed, 13 Sep 2006 12:56:20 +0300

Juha,

Juha Heinänen writes:

> there is personal bias, but what comes to regulatory requirements in
> europe, as i said, they currently only apply to calls terminating from
> sip to pstn and wise versa.  in those cases anonymity is easy to
> accomplish with or without gruu without any additions to rfc3261 or the
> current gruu spec.

I must tell that your interpretation is not fully correct. 

The obligation to offer calling line identification barring only concerns telephone services (e.g. PATS subscribtion). However this does not necessarily mean that all calls need to originate from PSTN or terminate to PSTN. So it's applicable also to SIP-to-SIP calls... 

If we think the wording that is used in directive: It's calling line identity. In PSTN this is a E.164 number. However in SIP, at least I wouldn't claim that this would definitely mean only E.164 numbers... (not going to make any guess about future interpretations :)

regards,

- Klaus -

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---

From: Juha Heinanen <[email-masked]>
Date: Wed, 13 Sep 2006 13:24:24 +0300

Nieminen Klaus writes:

 > The obligation to offer calling line identification barring only
 > concerns telephone services (e.g. PATS subscribtion). However this
 > does not necessarily mean that all calls need to originate from PSTN
 > or terminate to PSTN. So it's applicable also to SIP-to-SIP calls... 

klaus,

my message was not clear.  i meant sip-to-sip calls where UAs don't
subscribe to regulated telephony service.  if skype would use SIP, would
it apply to skype-skype calls using skype's proprietary URIs?

 > If we think the wording that is used in directive: It's calling line
 > identity. In PSTN this is a E.164 number. However in SIP, at least I
 > wouldn't claim that this would definitely mean only E.164
 > numbers... (not going to make any guess about future interpretations
 > :)

if calling line identity in sip-sip calls needs to be barred, then you
authorities need to specify how it is done both by SIP UA and proxy
operator.  i don't remember seeing any specifications on SIP UA
requirements when it connects to regulated telephony service nor have i
seen requirements on how two operator connect to each other using SIP.

you cannot simply say that a proxy operator has to do barring of calling
line identification.  in SIP, this definitely also means that the UA
must participate in the barring process somehow.  proxy cannot just go
and change the From header of the request it receives from the UA.

-- juha

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From: "Christer Holmberg \(JO/LMF\)" <[email-masked]>
Date: Wed, 13 Sep 2006 12:33:29 +0200

>you cannot simply say that a proxy operator has to do barring 
>of calling line identification.  in SIP, this definitely also 
>means that the UA must participate in the barring process 
>somehow.  proxy cannot just go and change the From header of 
>the request it receives from the UA.

Why not? You could say it acts like a B2BUA...

Regards,

Christer

---

From: Juha Heinanen <[email-masked]>
Date: Wed, 13 Sep 2006 13:39:27 +0300

> Why not? You could say it acts like a B2BUA...

if regulators require service provider to use b2bua for each request
from subscriber, then i would not call it ietf sip based service.

-- juha

---

From: "Christer Holmberg \(JO/LMF\)" <[email-masked]>
Date: Wed, 13 Sep 2006 13:34:14 +0200

I don't know what IETF SIP is. SIP is SIP, only SIP and nothing but SIP
:) 

> -----Original Message-----
> From: Juha Heinanen [mailto:[email-masked]] 
> Sent: 13. syyskuuta 2006 13:39
> To: Christer Holmberg (JO/LMF)
> Cc: Nieminen Klaus; [email-masked]
> Subject: RE: [Sip] GRUU and Anonymity
> 
> if regulators require service provider to use b2bua for each 
> request from subscriber, then i would not call it ietf sip 
> based service.
> 
> -- juha
>