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Subject: RE: [Sipping] TCAP tunneling
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Hi
   TCAP is used for Transaction Capabilities i.e database related while
SIP with ISUP tunneling is used between MGCs. So I don't think TCAP can
be tunneled in SIP.

Ranjit
Ph: 8520408/16 extn: 2173




-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Kaitki Agarwal
Sent: Monday, February 02, 2004 8:28 AM
To: sipping@ietf.org
Subject: [Sipping] TCAP tunneling


Hi
 Can we tunnel tcap message in SIP as we do  ISUP. If not why.=20
=20
Thanks
Kaitki
Jj~Yzy=3Di~Rz}y

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Banibrata Dutta" <dutta@india.hp.com>
To: <ranjit.avasarala@wipro.com>, <kagarwal@cedarpointcom.com>,
        <sipping@ietf.org>
Subject: RE: [Sipping] TCAP tunneling
Date: Mon, 2 Feb 2004 10:55:31 +0530
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hi,

franly, i'd say that, it depends on your motivation/reason to use SIP
for 
tunneling TCAP. of course you can tunnel TCAP, but why you'd want to, is
a justification you'd have to work out. AFAIK, there's not ietf-draft
that
says, how.

you can obviously tunnel TCAP query in an INVITE or INFO for that matter

(ideally in INVITE s.t. a SIP-dialog is also created which may be used
to
model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
messages may then reside within the body of a multi-part (MIME) enc,
INVITE.

infact 1 reason i see where this may make sense is that you would rather
not have two stacks s.a. SIGTRAN and SIP for signalling transport, and
use
SIP as the *only* signalling solution, i.e. for the signalling
application
as well as for signalling transport, and support legacy/TCAP apps by
tunneling
them within SIP. but then, you generally have the constraints of
not-enough-
space, (for 2 stacks) only on resource-limited devices, s.a. hand-sets,
embedded devices etc., but i am unable see (yet), how TCAP can be
related to
such devices.

my 2 cents,
Banibrata Dutta,
  Hewlett-Packard ISO (P) Ltd,
  29, Cunningham Road,
  Bangalore - 560052
  Ph: +91-80-22051796

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of ranjit.avasarala@wipro.com
> Sent: Monday, February 02, 2004 10:20 AM
> To: kagarwal@cedarpointcom.com; sipping@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
> 
> 
> Hi
>    TCAP is used for Transaction Capabilities i.e database 
> related while SIP with ISUP tunneling is used between MGCs. 
> So I don't think TCAP can be tunneled in SIP.
> 
> Ranjit
> Ph: 8520408/16 extn: 2173
> 
> 
> 
> 
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Kaitki Agarwal
> Sent: Monday, February 02, 2004 8:28 AM
> To: sipping@ietf.org
> Subject: [Sipping] TCAP tunneling
> 
> 
> Hi
>  Can we tunnel tcap message in SIP as we do  ISUP. If not why. 
>  
> Thanks
> Kaitki
> Jj~Yzy=i~Rz}y
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 


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



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To: "Banibrata Dutta" <dutta@india.hp.com>
Cc: kagarwal@cedarpointcom.com, ranjit.avasarala@wipro.com, sipping@ietf.org,
        sipping-admin@ietf.org
Subject: RE: [Sipping] TCAP tunneling
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From: Arjun Roychowdhury <aroychow@hns.com>
Date: Mon, 2 Feb 2004 09:25:27 -0500
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Actually, there _is_ an independant draft on how to carry TCAP in SIP.
http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt

Its another matter whether this is the right thing to do or not (please 
refer to old archives for the debates). 
Kaitaki, you may want to read the above draft as well as the discussions 
on it to conclude for yourself which is the preferred method in your
deployment scenario.

regds
arjun


--
Arjun Roychowdhury @ Hughes Software Systems
11717 Exploration Lane, Germantown MD 20876
(O): 301 212 7860 
VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)




"Banibrata Dutta" <dutta@india.hp.com>
Sent by: sipping-admin@ietf.org
02/02/2004 12:25 AM

 
        To:     <ranjit.avasarala@wipro.com>, <kagarwal@cedarpointcom.com>, 
<sipping@ietf.org>
        cc: 
        Subject:        RE: [Sipping] TCAP tunneling


hi,

franly, i'd say that, it depends on your motivation/reason to use SIP
for 
tunneling TCAP. of course you can tunnel TCAP, but why you'd want to, is
a justification you'd have to work out. AFAIK, there's not ietf-draft
that
says, how.

you can obviously tunnel TCAP query in an INVITE or INFO for that matter

(ideally in INVITE s.t. a SIP-dialog is also created which may be used
to
model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
messages may then reside within the body of a multi-part (MIME) enc,
INVITE.

infact 1 reason i see where this may make sense is that you would rather
not have two stacks s.a. SIGTRAN and SIP for signalling transport, and
use
SIP as the *only* signalling solution, i.e. for the signalling
application
as well as for signalling transport, and support legacy/TCAP apps by
tunneling
them within SIP. but then, you generally have the constraints of
not-enough-
space, (for 2 stacks) only on resource-limited devices, s.a. hand-sets,
embedded devices etc., but i am unable see (yet), how TCAP can be
related to
such devices.

my 2 cents,
Banibrata Dutta,
  Hewlett-Packard ISO (P) Ltd,
  29, Cunningham Road,
  Bangalore - 560052
  Ph: +91-80-22051796

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of ranjit.avasarala@wipro.com
> Sent: Monday, February 02, 2004 10:20 AM
> To: kagarwal@cedarpointcom.com; sipping@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
> 
> 
> Hi
>    TCAP is used for Transaction Capabilities i.e database 
> related while SIP with ISUP tunneling is used between MGCs. 
> So I don't think TCAP can be tunneled in SIP.
> 
> Ranjit
> Ph: 8520408/16 extn: 2173
> 
> 
> 
> 
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Kaitki Agarwal
> Sent: Monday, February 02, 2004 8:28 AM
> To: sipping@ietf.org
> Subject: [Sipping] TCAP tunneling
> 
> 
> Hi
>  Can we tunnel tcap message in SIP as we do  ISUP. If not why. 
> 
> Thanks
> Kaitki
> Jj~Yzy=i~Rz}y
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sip@ietf.org for new developments of core SIP
> 


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




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



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Subject: RE: [Sipping] TCAP tunneling
Date: Mon, 2 Feb 2004 10:59:41 -0500
Message-ID: <66F2ACFAB4E7764EAF85274020DD5342017F9084@exchange.cedarpointcom.com>
Thread-Topic: [Sipping] TCAP tunneling
Thread-Index: AcPpTRE2xG/QMx8BSZuBHZoLYjWmUQATgFxw
From: "Kaitki Agarwal" <kagarwal@cedarpointcom.com>
To: "Banibrata Dutta" <dutta@india.hp.com>, <ranjit.avasarala@wipro.com>,
        <sipping@ietf.org>
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Typically we use ISUP tunneling in SIP when the end exchange is talking
to another switch who is terminating traditional SS7 links to save cost
on SS7 links. The end exchange is the one who has all the subscriber
profiles and knows if it needs CNAM or LNP or AC/AR. If we have a way to
tunnel TCAP in SIP using INVITE (as suggested below) we can access all
these services from the end exchange.=20
TUA is not yet RFC so I would prefer not to use it. With SUA you still
need TCAP ASN.1 BER encoded message. Instead of building Sigtran I can
use SIP if it has a TCAP extension.
It is not just space constraint , it is cost of writing/buying SIGTRAN
stack and building then supporting it in to the system.=20

Thanks
Kaitki  =20

-----Original Message-----
From: Banibrata Dutta [mailto:dutta@india.hp.com]=20
Sent: Monday, February 02, 2004 12:26 AM
To: ranjit.avasarala@wipro.com; Kaitki Agarwal; sipping@ietf.org
Subject: RE: [Sipping] TCAP tunneling


hi,

franly, i'd say that, it depends on your motivation/reason to use SIP
for=20
tunneling TCAP. of course you can tunnel TCAP, but why you'd want to, is
a justification you'd have to work out. AFAIK, there's not ietf-draft
that says, how.

you can obviously tunnel TCAP query in an INVITE or INFO for that matter

(ideally in INVITE s.t. a SIP-dialog is also created which may be used
to model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
messages may then reside within the body of a multi-part (MIME) enc,
INVITE.

infact 1 reason i see where this may make sense is that you would rather
not have two stacks s.a. SIGTRAN and SIP for signalling transport, and
use SIP as the *only* signalling solution, i.e. for the signalling
application as well as for signalling transport, and support legacy/TCAP
apps by tunneling them within SIP. but then, you generally have the
constraints of
not-enough-
space, (for 2 stacks) only on resource-limited devices, s.a. hand-sets,
embedded devices etc., but i am unable see (yet), how TCAP can be
related to such devices.

my 2 cents,
Banibrata Dutta,
  Hewlett-Packard ISO (P) Ltd,
  29, Cunningham Road,
  Bangalore - 560052
  Ph: +91-80-22051796

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> On Behalf Of ranjit.avasarala@wipro.com
> Sent: Monday, February 02, 2004 10:20 AM
> To: kagarwal@cedarpointcom.com; sipping@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
>=20
>=20
> Hi
>    TCAP is used for Transaction Capabilities i.e database
> related while SIP with ISUP tunneling is used between MGCs.=20
> So I don't think TCAP can be tunneled in SIP.
>=20
> Ranjit
> Ph: 8520408/16 extn: 2173
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> On Behalf Of Kaitki Agarwal
> Sent: Monday, February 02, 2004 8:28 AM
> To: sipping@ietf.org
> Subject: [Sipping] TCAP tunneling
>=20
>=20
> Hi
>  Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> =20
> Thanks
> Kaitki
> Jj~Yzy=3Di~Rz}y
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sip@ietf.org for new developments of core SIP
>=20


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



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Subject: RE: [Sipping] TCAP tunneling
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Thread-Topic: [Sipping] TCAP tunneling
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From: "Hearty, John" <John.Hearty@Level3.com>
To: <sipping@ietf.org>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>
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Other than the miller draft below, is there any SIP standard proposed to
carry cnam queries?

John Hearty
Level3

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of
> Arjun Roychowdhury
> Sent: Monday, February 02, 2004 7:25 AM
> To: Banibrata Dutta
> Cc: kagarwal@cedarpointcom.com; ranjit.avasarala@wipro.com;
> sipping@ietf.org; sipping-admin@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
>=20
> Actually, there _is_ an independant draft on how to carry TCAP in SIP.
> http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt
>=20
> Its another matter whether this is the right thing to do or not
(please
> refer to old archives for the debates).
> Kaitaki, you may want to read the above draft as well as the
discussions
> on it to conclude for yourself which is the preferred method in your
> deployment scenario.
>=20
> regds
> arjun
>=20
>=20
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 11717 Exploration Lane, Germantown MD 20876
> (O): 301 212 7860
> VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)
>=20
>=20
>=20
>=20
> "Banibrata Dutta" <dutta@india.hp.com>
> Sent by: sipping-admin@ietf.org
> 02/02/2004 12:25 AM
>=20
>=20
>         To:     <ranjit.avasarala@wipro.com>,
<kagarwal@cedarpointcom.com>,
> <sipping@ietf.org>
>         cc:
>         Subject:        RE: [Sipping] TCAP tunneling
>=20
>=20
> hi,
>=20
> franly, i'd say that, it depends on your motivation/reason to use SIP
> for
> tunneling TCAP. of course you can tunnel TCAP, but why you'd want to,
is
> a justification you'd have to work out. AFAIK, there's not ietf-draft
> that
> says, how.
>=20
> you can obviously tunnel TCAP query in an INVITE or INFO for that
matter
>=20
> (ideally in INVITE s.t. a SIP-dialog is also created which may be used
> to
> model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
> messages may then reside within the body of a multi-part (MIME) enc,
> INVITE.
>=20
> infact 1 reason i see where this may make sense is that you would
rather
> not have two stacks s.a. SIGTRAN and SIP for signalling transport, and
> use
> SIP as the *only* signalling solution, i.e. for the signalling
> application
> as well as for signalling transport, and support legacy/TCAP apps by
> tunneling
> them within SIP. but then, you generally have the constraints of
> not-enough-
> space, (for 2 stacks) only on resource-limited devices, s.a.
hand-sets,
> embedded devices etc., but i am unable see (yet), how TCAP can be
> related to
> such devices.
>=20
> my 2 cents,
> Banibrata Dutta,
>   Hewlett-Packard ISO (P) Ltd,
>   29, Cunningham Road,
>   Bangalore - 560052
>   Ph: +91-80-22051796
>=20
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of ranjit.avasarala@wipro.com
> > Sent: Monday, February 02, 2004 10:20 AM
> > To: kagarwal@cedarpointcom.com; sipping@ietf.org
> > Subject: RE: [Sipping] TCAP tunneling
> >
> >
> > Hi
> >    TCAP is used for Transaction Capabilities i.e database
> > related while SIP with ISUP tunneling is used between MGCs.
> > So I don't think TCAP can be tunneled in SIP.
> >
> > Ranjit
> > Ph: 8520408/16 extn: 2173
> >
> >
> >
> >
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of Kaitki Agarwal
> > Sent: Monday, February 02, 2004 8:28 AM
> > To: sipping@ietf.org
> > Subject: [Sipping] TCAP tunneling
> >
> >
> > Hi
> >  Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> >
> > Thanks
> > Kaitki
> > Jj~Yzy=3Di~Rz}y
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current
> > sip Use sip@ietf.org for new developments of core SIP
> >
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

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



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From: "Banibrata Dutta" <dutta@india.hp.com>
To: "'Kaitki Agarwal'" <kagarwal@cedarpointcom.com>,
        <ranjit.avasarala@wipro.com>, <sipping@ietf.org>
Subject: RE: [Sipping] TCAP tunneling
Date: Tue, 3 Feb 2004 10:35:19 +0530
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Hi Kaitki,

I see your point. Well, the idea of using standards for such
things is:
a) it's a standard, so every node you talk to speaks the same
lingo, and there are no major interworking issues with n/w
elements from other vendors.
b) you don't have to re-invent the wheel, i.e. you get a logic,
that is well understood, mature, robust and stable (i.e. all
the goodies).
c) you can probably buy it off the shelf (so a cost is attached).

The pitfalls of not using a standard, i.e. using a draft, that
*may* not become a standard are just the opposites of a),b) 
and to some extent c). I think the bigger danger is (b) i.e.
you re-invent the wheel and there are chances that it doesn't
go thru a strong and prolonged community review, and you *may*
leave few holes which you keep patching for the rest of the
life of such a product.

coming back to the point of using SIP, well, i've already said,
you definitely can use it. and i didn't realize that your 
intention was to save on the ASN.1 encoding/decoding (that you
have to do in SUA), but then ASN.1 gives you the power (and
at time compactness of "binary encoding") of representation
that is again standard and hence you get a),b) & c), which you
lose if you use something like a customized-XML message-set etc.

for your purpose SIGTRAN (w/SUA) + regular TCAP layer is the
way to go. or even more widely deployed and used - SIGTRAN (w/M3UA)
+ regular TCAP/SCCP. although less popular (i think than SIGTRAN), 
you may like to evaluate TALI (the RFC version).

my 2 cents,
regards,
Banibrata.



> -----Original Message-----
> From: Kaitki Agarwal [mailto:kagarwal@cedarpointcom.com] 
> Sent: Monday, February 02, 2004 9:30 PM
> To: Banibrata Dutta; ranjit.avasarala@wipro.com; sipping@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
> 
> 
> Typically we use ISUP tunneling in SIP when the end exchange 
> is talking to another switch who is terminating traditional 
> SS7 links to save cost on SS7 links. The end exchange is the 
> one who has all the subscriber profiles and knows if it needs 
> CNAM or LNP or AC/AR. If we have a way to tunnel TCAP in SIP 
> using INVITE (as suggested below) we can access all these 
> services from the end exchange. 
> TUA is not yet RFC so I would prefer not to use it. With SUA 
> you still need TCAP ASN.1 BER encoded message. Instead of 
> building Sigtran I can use SIP if it has a TCAP extension. It 
> is not just space constraint , it is cost of writing/buying 
> SIGTRAN stack and building then supporting it in to the system. 
> 
> Thanks
> Kaitki   
> 
> -----Original Message-----
> From: Banibrata Dutta [mailto:dutta@india.hp.com] 
> Sent: Monday, February 02, 2004 12:26 AM
> To: ranjit.avasarala@wipro.com; Kaitki Agarwal; sipping@ietf.org
> Subject: RE: [Sipping] TCAP tunneling
> 
> 
> hi,
> 
> franly, i'd say that, it depends on your motivation/reason to 
> use SIP for 
> tunneling TCAP. of course you can tunnel TCAP, but why you'd 
> want to, is a justification you'd have to work out. AFAIK, 
> there's not ietf-draft that says, how.
> 
> you can obviously tunnel TCAP query in an INVITE or INFO for 
> that matter
> 
> (ideally in INVITE s.t. a SIP-dialog is also created which 
> may be used to model the TCAP dialog as well, perhaps!). the 
> ASN.1 BER encoded TCAP messages may then reside within the 
> body of a multi-part (MIME) enc, INVITE.
> 
> infact 1 reason i see where this may make sense is that you 
> would rather not have two stacks s.a. SIGTRAN and SIP for 
> signalling transport, and use SIP as the *only* signalling 
> solution, i.e. for the signalling application as well as for 
> signalling transport, and support legacy/TCAP apps by 
> tunneling them within SIP. but then, you generally have the 
> constraints of
> not-enough-
> space, (for 2 stacks) only on resource-limited devices, s.a. 
> hand-sets, embedded devices etc., but i am unable see (yet), 
> how TCAP can be related to such devices.
> 
> my 2 cents,
> Banibrata Dutta,
>   Hewlett-Packard ISO (P) Ltd,
>   29, Cunningham Road,
>   Bangalore - 560052
>   Ph: +91-80-22051796
> 
> > -----Original Message-----
> > From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org] On Behalf 
> > Of 
> ranjit.avasarala@wipro.com
> > Sent: Monday, February 02, 2004 10:20 AM
> > To: kagarwal@cedarpointcom.com; sipping@ietf.org
> > Subject: RE: [Sipping] TCAP tunneling
> > 
> > 
> > Hi
> >    TCAP is used for Transaction Capabilities i.e database related 
> > while SIP with ISUP tunneling is used between MGCs. So I 
> don't think 
> > TCAP can be tunneled in SIP.
> > 
> > Ranjit
> > Ph: 8520408/16 extn: 2173
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org] On Behalf 
> > Of Kaitki 
> Agarwal
> > Sent: Monday, February 02, 2004 8:28 AM
> > To: sipping@ietf.org
> > Subject: [Sipping] TCAP tunneling
> > 
> > 
> > Hi
> >  Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> >  
> > Thanks
> > Kaitki
> > Jj~Yzy=i~Rz}y
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sip@ietf.org for new developments of core SIP
> > 
> 
> 


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



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Subject: Re: [Sipping] TCAP tunneling
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SIP isnt a database query protocol, and cnam is just a DB query service. 
As such, I would think that enum would be an appropriate technique for 
this. Any other DB query mechanism will do as well.

Another idea would be to use a redirect service, similar to what has 
been proposed in the past for LNP:

http://www.watersprings.org/pub/id/draft-yu-sip-np-02.txt

except that a request for a number would be redirected to a Contact 
which contains that number plus the display name, representing the 
calling name.

-Jonathan R.

Hearty, John wrote:
> Other than the miller draft below, is there any SIP standard proposed to
> carry cnam queries?
> 
> John Hearty
> Level3
> 
> 
>>-----Original Message-----
>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
> 
> Of
> 
>>Arjun Roychowdhury
>>Sent: Monday, February 02, 2004 7:25 AM
>>To: Banibrata Dutta
>>Cc: kagarwal@cedarpointcom.com; ranjit.avasarala@wipro.com;
>>sipping@ietf.org; sipping-admin@ietf.org
>>Subject: RE: [Sipping] TCAP tunneling
>>
>>Actually, there _is_ an independant draft on how to carry TCAP in SIP.
>>http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt
>>
>>Its another matter whether this is the right thing to do or not
> 
> (please
> 
>>refer to old archives for the debates).
>>Kaitaki, you may want to read the above draft as well as the
> 
> discussions
> 
>>on it to conclude for yourself which is the preferred method in your
>>deployment scenario.
>>
>>regds
>>arjun
>>
>>
>>--
>>Arjun Roychowdhury @ Hughes Software Systems
>>11717 Exploration Lane, Germantown MD 20876
>>(O): 301 212 7860
>>VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)
>>
>>
>>
>>
>>"Banibrata Dutta" <dutta@india.hp.com>
>>Sent by: sipping-admin@ietf.org
>>02/02/2004 12:25 AM
>>
>>
>>        To:     <ranjit.avasarala@wipro.com>,
> 
> <kagarwal@cedarpointcom.com>,
> 
>><sipping@ietf.org>
>>        cc:
>>        Subject:        RE: [Sipping] TCAP tunneling
>>
>>
>>hi,
>>
>>franly, i'd say that, it depends on your motivation/reason to use SIP
>>for
>>tunneling TCAP. of course you can tunnel TCAP, but why you'd want to,
> 
> is
> 
>>a justification you'd have to work out. AFAIK, there's not ietf-draft
>>that
>>says, how.
>>
>>you can obviously tunnel TCAP query in an INVITE or INFO for that
> 
> matter
> 
>>(ideally in INVITE s.t. a SIP-dialog is also created which may be used
>>to
>>model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
>>messages may then reside within the body of a multi-part (MIME) enc,
>>INVITE.
>>
>>infact 1 reason i see where this may make sense is that you would
> 
> rather
> 
>>not have two stacks s.a. SIGTRAN and SIP for signalling transport, and
>>use
>>SIP as the *only* signalling solution, i.e. for the signalling
>>application
>>as well as for signalling transport, and support legacy/TCAP apps by
>>tunneling
>>them within SIP. but then, you generally have the constraints of
>>not-enough-
>>space, (for 2 stacks) only on resource-limited devices, s.a.
> 
> hand-sets,
> 
>>embedded devices etc., but i am unable see (yet), how TCAP can be
>>related to
>>such devices.
>>
>>my 2 cents,
>>Banibrata Dutta,
>>  Hewlett-Packard ISO (P) Ltd,
>>  29, Cunningham Road,
>>  Bangalore - 560052
>>  Ph: +91-80-22051796
>>
>>
>>>-----Original Message-----
>>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
>>>On Behalf Of ranjit.avasarala@wipro.com
>>>Sent: Monday, February 02, 2004 10:20 AM
>>>To: kagarwal@cedarpointcom.com; sipping@ietf.org
>>>Subject: RE: [Sipping] TCAP tunneling
>>>
>>>
>>>Hi
>>>   TCAP is used for Transaction Capabilities i.e database
>>>related while SIP with ISUP tunneling is used between MGCs.
>>>So I don't think TCAP can be tunneled in SIP.
>>>
>>>Ranjit
>>>Ph: 8520408/16 extn: 2173
>>>
>>>
>>>
>>>
>>>-----Original Message-----
>>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
>>>On Behalf Of Kaitki Agarwal
>>>Sent: Monday, February 02, 2004 8:28 AM
>>>To: sipping@ietf.org
>>>Subject: [Sipping] TCAP tunneling
>>>
>>>
>>>Hi
>>> Can we tunnel tcap message in SIP as we do  ISUP. If not why.
>>>
>>>Thanks
>>>Kaitki
>>>Jj~Yzy=i~Rz}y
>>>
>>>_______________________________________________
>>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>This list is for NEW development of the application of SIP
>>>Use sip-implementors@cs.columbia.edu for questions on current
>>>sip Use sip@ietf.org for new developments of core SIP
>>>
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>
>>
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Aarti Iyengar" <aartii@hotmail.com>
To: sipping@ietf.org
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Subject: [Sipping] SIP-T and BICC
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All,
I am trying to understand SIP-T and BICC in detail.
From my initial reading, both are essentially to seamlessly
interface the PSTN to broadband networks. SIP-T is specifically for
SIP based IP networks, whereas BICC is bearer independent and can
interface with IP/ATM networks well. Apart from that, SIP-T deals
with ISUP translation and encapsulation while BICC deals with
extensions with ISUP for additional bearer support.
BICC CS3 will also enable SIP services to co-exist with BICC based
networks.
My questions are -
Are these the only key differences between these protocols which
essentially are intended perform the same function? If there are
others, what are they?
Are there certain situations where one would fit in better than the
other, and if so what are these?
How well would these interoperate with each other if they have to
co-exist?

Also, any pointers to urls/whitepapers on the above (BICC versus
SIP-T) would be appreciated.

thanks
Aarti

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Aarti Iyengar'" <aartii@hotmail.com>, sipping@ietf.org
Subject: RE: [Sipping] SIP-T and BICC
Date: Tue, 3 Feb 2004 08:18:45 -0500 
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Since SIP depends on SDP for describing bearers, and SDP can describe
an ATM bearer (RFC3108), you can't use that as a discriminator.
We have SIP products that are ATM based.

The primary difference between the two is that BICC uses the EXISTING
SS7 network for the signaling, and an IP (or ATM) network for the bearer,
where SIP-T uses IP (or ATM) for both the signaling and the bearer.
Also, the fundamental idea of SIP-T is that SIP is the signaling protocol,
and the -T part allows SIP to carry those parts of ISUP that have no
SIP equivalent.  This allows calls between two ISUP controlled endpoints
to use SIP in the middle of the network.  Contrast that to BICC, where
the assumption is that the whole network is ISUP signaled.
Thus if you believe that SIP will eventually supplant ISUP for voice
signaling, you might want to use SIP-T.  If you expect that multimedia
(video, text, ...) bearers might be interesting, you might want to chose
SIP-T.  Conversely, if you think that separating signaling from bearers is
essential, you might choose BICC.

Interoperating with these two protocols is somewhat problematic, primarily
because they don't anticipate each other; SIP-T does not specifically deal
with the bearer extension introduced by BICC, although it would be pretty
trivial to take it into account.  BICC does not admit any other signaling
for endpoints (CS-3 notwithstanding).  In practice, I expect you COULD
interwork them if you needed to.

Brian

> -----Original Message-----
> From: Aarti Iyengar [mailto:aartii@hotmail.com]
> Sent: Tuesday, February 03, 2004 6:36 AM
> To: sipping@ietf.org
> Subject: [Sipping] SIP-T and BICC
> 
> 
> All,
> I am trying to understand SIP-T and BICC in detail.
> From my initial reading, both are essentially to seamlessly
> interface the PSTN to broadband networks. SIP-T is specifically for
> SIP based IP networks, whereas BICC is bearer independent and can
> interface with IP/ATM networks well. Apart from that, SIP-T deals
> with ISUP translation and encapsulation while BICC deals with
> extensions with ISUP for additional bearer support.
> BICC CS3 will also enable SIP services to co-exist with BICC based
> networks.
> My questions are -
> Are these the only key differences between these protocols which
> essentially are intended perform the same function? If there are
> others, what are they?
> Are there certain situations where one would fit in better than the
> other, and if so what are these?
> How well would these interoperate with each other if they have to
> co-exist?
> 
> Also, any pointers to urls/whitepapers on the above (BICC versus
> SIP-T) would be appreciated.
> 
> thanks
> Aarti
> 
> _________________________________________________________________
> The new MSN 8: advanced junk mail protection and 2 months FREE* 
> http://join.msn.com/?page=features/junkmail
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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inline, and moving to sipping.

Suha Cubukcuoglu wrote:

> Hi,
> 
> During an attended call transfer attempt, the transferred party
> (A-Party) is put on 'hold' by the transferor (B-Party) - i.e. no media
> stream flows from B to A and vice versa. While the A-Party is on hold,
> the B-Party can INVITE a third party to play 'Music on Hold' tune to the
> A-Party until the transfer attempt succeeds. We think that there are
> complications in this approach. Firstly, the SIP Proxy must now where to
> route the new INVITE generated by the B-Party. If the destination URI is
> 'moh@someserver.com', the SIP Proxy must either do a DNS query or use
> its IP dial-plans to know where to route the call.

Sure. The URI needs to exist and be routable in the domain. Why is this 
a problem?

> Secondly, the B-Party
> might eventually end-up receiving the music tune whereas it was
> initially intended for the A-Party - we've had problems in the past
> because of this. 

If the INVITE contains sdp pointing the media to the a-party (which it 
should), this shouldnt happen. Is your on-hold server ignoring the 
address in the SDP?


> We think that third party invitation is a bad idea, and it would rather
> be more suitable to have a dedicated server (IVR unit perhaps) which the
> SIP Proxy Server can transfer calls to/from upon request for a MOH
> service. Could you tell us your opinions about this please?

Proxies dont transfer calls, as they dont maintain call state.

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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From: "Christian Stredicke" <stredicke@snom.de>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <suhac@ednet.co.uk>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] Re: [Sip] Music On Hold
Date: Wed, 4 Feb 2004 08:06:33 +0100
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I am still convinced we need a more lightweight model that puts less =
burden
on the user agent which puts someone on hold. I mean, which user agent =
does
implement the model with a media server?

Why don't we just provide a music-URL like in Alert-Info? The other user
agent then can pick whatever it likes (http/WAV, sip, MIDI or whatever). =
I
think that's ten times easier.

BTW this information could also be inserted by the proxy (without
maintaining call state). This avoids the problem of setting the user =
agents
up for this.

Christian

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
> Jonathan Rosenberg
> Sent: Wednesday, February 04, 2004 6:36 AM
> To: suhac@ednet.co.uk
> Cc: sipping@ietf.org
> Subject: [Sipping] Re: [Sip] Music On Hold
>=20
> inline, and moving to sipping.
>=20
> Suha Cubukcuoglu wrote:
>=20
> > Hi,
> >
> > During an attended call transfer attempt, the transferred party
> > (A-Party) is put on 'hold' by the transferor (B-Party) - i.e. no =
media
> > stream flows from B to A and vice versa. While the A-Party is on =
hold,
> > the B-Party can INVITE a third party to play 'Music on Hold' tune to =
the
> > A-Party until the transfer attempt succeeds. We think that there are
> > complications in this approach. Firstly, the SIP Proxy must now =
where to
> > route the new INVITE generated by the B-Party. If the destination =
URI is
> > 'moh@someserver.com', the SIP Proxy must either do a DNS query or =
use
> > its IP dial-plans to know where to route the call.
>=20
> Sure. The URI needs to exist and be routable in the domain. Why is =
this
> a problem?
>=20
> > Secondly, the B-Party
> > might eventually end-up receiving the music tune whereas it was
> > initially intended for the A-Party - we've had problems in the past
> > because of this.
>=20
> If the INVITE contains sdp pointing the media to the a-party (which it
> should), this shouldnt happen. Is your on-hold server ignoring the
> address in the SDP?
>=20
>=20
> > We think that third party invitation is a bad idea, and it would =
rather
> > be more suitable to have a dedicated server (IVR unit perhaps) which =
the
> > SIP Proxy Server can transfer calls to/from upon request for a MOH
> > service. Could you tell us your opinions about this please?
>=20
> Proxies dont transfer calls, as they dont maintain call state.
>=20
> Thanks,
> Jonathan R.
>=20
>=20
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



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From: "F.S.Salloum" <ssal@intracom.gr>
To: "'Christian Stredicke'" <stredicke@snom.de>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <suhac@ednet.co.uk>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] Re: [Sip] Music On Hold
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Hi, please find my opinions inline

>>-----Original Message-----
>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
>>Christian Stredicke
>>Sent: Wednesday, February 04, 2004 9:07 AM
>>To: 'Jonathan Rosenberg'; suhac@ednet.co.uk
>>Cc: sipping@ietf.org
>>Subject: RE: [Sipping] Re: [Sip] Music On Hold

>>I am still convinced we need a more lightweight model that puts less
>>burden
>>on the user agent which puts someone on hold. I mean, which user agent
>>does
>>implement the model with a media server?


A lot of commercial product also our HW phones(http://netphone.intracom.gr/)
support this approach with no problem



>>Why don't we just provide a music-URL like in Alert-Info? The other user
>>agent then can pick whatever it likes (http/WAV, sip, MIDI or whatever). I
>>think that's ten times easier.


I think we should consider security issues check this link with a
discussions with alert info stuff
(http://www.atm.tut.fi/list-archive/sip/msg01278.html) 


>>BTW this information could also be inserted by the proxy (without
>>maintaining call state). This avoids the problem of setting the user
>>agents
>>up for this.

>>Christian


Enjoy
Sotiris


> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
> Jonathan Rosenberg
> Sent: Wednesday, February 04, 2004 6:36 AM
> To: suhac@ednet.co.uk
> Cc: sipping@ietf.org
> Subject: [Sipping] Re: [Sip] Music On Hold
> 
> inline, and moving to sipping.
> 
> Suha Cubukcuoglu wrote:
> 
> > Hi,
> >
> > During an attended call transfer attempt, the transferred party
> > (A-Party) is put on 'hold' by the transferor (B-Party) - i.e. no media
> > stream flows from B to A and vice versa. While the A-Party is on hold,
> > the B-Party can INVITE a third party to play 'Music on Hold' tune to the
> > A-Party until the transfer attempt succeeds. We think that there are
> > complications in this approach. Firstly, the SIP Proxy must now where to
> > route the new INVITE generated by the B-Party. If the destination URI is
> > 'moh@someserver.com', the SIP Proxy must either do a DNS query or use
> > its IP dial-plans to know where to route the call.
> 
> Sure. The URI needs to exist and be routable in the domain. Why is this
> a problem?
> 
> > Secondly, the B-Party
> > might eventually end-up receiving the music tune whereas it was
> > initially intended for the A-Party - we've had problems in the past
> > because of this.
> 
> If the INVITE contains sdp pointing the media to the a-party (which it
> should), this shouldnt happen. Is your on-hold server ignoring the
> address in the SDP?
> 
> 
> > We think that third party invitation is a bad idea, and it would rather
> > be more suitable to have a dedicated server (IVR unit perhaps) which the
> > SIP Proxy Server can transfer calls to/from upon request for a MOH
> > service. Could you tell us your opinions about this please?
> 
> Proxies dont transfer calls, as they dont maintain call state.
> 
> Thanks,
> Jonathan R.
> 
> 
> 
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



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



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From: "Christian Stredicke" <stredicke@snom.de>
To: "'F.S.Salloum'" <ssal@intracom.gr>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <suhac@ednet.co.uk>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] Re: [Sip] Music On Hold
Date: Wed, 4 Feb 2004 09:37:31 +0100
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Comments inline!

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
> F.S.Salloum
> Sent: Wednesday, February 04, 2004 8:25 AM
> To: 'Christian Stredicke'; 'Jonathan Rosenberg'; suhac@ednet.co.uk
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] Re: [Sip] Music On Hold
>=20
> Hi, please find my opinions inline
>=20
> >>-----Original Message-----
> >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On =
Behalf
> Of
> >>Christian Stredicke
> >>Sent: Wednesday, February 04, 2004 9:07 AM
> >>To: 'Jonathan Rosenberg'; suhac@ednet.co.uk
> >>Cc: sipping@ietf.org
> >>Subject: RE: [Sipping] Re: [Sip] Music On Hold
>=20
> >>I am still convinced we need a more lightweight model that puts less
> >>burden
> >>on the user agent which puts someone on hold. I mean, which user =
agent
> >>does
> >>implement the model with a media server?
>=20
>=20
> A lot of commercial product also our HW
> phones(http://netphone.intracom.gr/)
> support this approach with no problem
>=20
[Christian Stredicke] We also support this, but I can't say "no =
problem".
It's easy to get it working in the lab, but the real world shows us a
different picture. Calls get lost (people just plug the cable), transfer
fails, people are sitting behind NAT, media server running out of =
resources,
media server and negotiated codec don't find a common codec (cause only =
the
answer is available). Maybe is a question of the pain level, but for us =
it's
a "problem".

Of course, anything can be solved, but what I am trying to say is the
Alert-Info model would be much simpler. We could have the same =
discussion in
ring tones!!!

>=20
>=20
> >>Why don't we just provide a music-URL like in Alert-Info? The other =
user
> >>agent then can pick whatever it likes (http/WAV, sip, MIDI or =
whatever).
> I
> >>think that's ten times easier.
>=20
>=20
> I think we should consider security issues check this link with a
> discussions with alert info stuff
> (http://www.atm.tut.fi/list-archive/sip/msg01278.html)
>=20
[Christian Stredicke] There are also (bigger?) security problems with =
the
3PCC model. And the Alert-Info security problem has to be solved anyway =
(or
has been solved already).

>=20
> >>BTW this information could also be inserted by the proxy (without
> >>maintaining call state). This avoids the problem of setting the user
> >>agents
> >>up for this.



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Keypad Stimulus Protocol (KPML)
	Author(s)	: E. Burger
	Filename	: draft-ietf-sipping-kpml-02.txt
	Pages		: 42
	Date		: 2004-2-4
	
The Key Press Stimulus Protocol uses the SIP SUBSCRIBE/NOTIFY
   mechanism and Keypad Markup Language (KPML) to provide instructions
   to SIP User Agents for the reporting of user key presses.

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [Sipping] Re: [Sip] Music On Hold
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inline.

Christian Stredicke wrote:


>>A lot of commercial product also our HW
>>phones(http://netphone.intracom.gr/)
>>support this approach with no problem
>>
> 
> [Christian Stredicke] We also support this, but I can't say "no problem".
> It's easy to get it working in the lab, but the real world shows us a
> different picture. Calls get lost (people just plug the cable), transfer
> fails, people are sitting behind NAT, media server running out of resources,
> media server and negotiated codec don't find a common codec (cause only the
> answer is available). Maybe is a question of the pain level, but for us it's
> a "problem".
> 
> Of course, anything can be solved, but what I am trying to say is the
> Alert-Info model would be much simpler. We could have the same discussion in
> ring tones!!!

Its not clear to me that many of the problems you list above would go 
away with an Alert-Info style approach. Certainly, those calls can get 
lost, there can be nat problems, the server can still run out of 
resources, etc.

You also indicate that there is a problem finding a common codec. Of 
course, if there are simply no codecs in common no signaling in the 
world will help. However, I don't think thats what you mean. In the flow 
in the service examples draft, the user placing the call on hold 
(b-party) issues an invite to the music on hold server with no sdp. That 
triggers the music on hold server to issue an offer with a full codec 
list. That is passed to the A-party, which picks the one(s) it wants. 
Thus, the music on hold can use different codecs than what is negotiated 
in the call between A and B. So, I dont see a specific problem. Can you 
elaborate on the difficulty you see?


>>
>>I think we should consider security issues check this link with a
>>discussions with alert info stuff
>>(http://www.atm.tut.fi/list-archive/sip/msg01278.html)
>>
> 
> [Christian Stredicke] There are also (bigger?) security problems with the
> 3PCC model. And the Alert-Info security problem has to be solved anyway (or
> has been solved already).

3pcc in this specific case is not so hard, since (presumably) there is a 
relationship between the b-party and the music on hold server. If the 
B-party rather tells the A-party to contact the music on hold server 
directly (using an alert-info kind of approach), now there needs to be a 
security relationship between the A-party on the music on hold server, 
which doesnt exist. Relaying signaling through the B-party makes the 
security more tenable, I think.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Eric,

The draft talks about sending the subscribe requests on the same dialog 
as the INVITE. For many reasons that have been discussed ad nauseam on 
the list, dialog sharing is a bad idea. GRUU provides a solution that 
does not require dialog sharing. As I recall, there was discussion on 
this point in the last ietf, and this seemed to be the preferred direction.

Is there any reason we cant change this to use gruu's rather than dialog 
sharing?

Also, you are missing the IANA registration for the kpml event package.

Thanks,
Jonathan R.

Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.
> 
> 	Title		: Keypad Stimulus Protocol (KPML)
> 	Author(s)	: E. Burger
> 	Filename	: draft-ietf-sipping-kpml-02.txt
> 	Pages		: 42
> 	Date		: 2004-2-4
> 	
> The Key Press Stimulus Protocol uses the SIP SUBSCRIBE/NOTIFY
>    mechanism and Keypad Markup Language (KPML) to provide instructions
>    to SIP User Agents for the reporting of user key presses.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-02.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
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> type "cd internet-drafts" and then
> 	"get draft-ietf-sipping-kpml-02.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sipping-kpml-02.txt".
> 	
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
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> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
> 		
> 		
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Gonzalo Camarillo'"
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Subject: [Sipping] General comments and NITs on draft-rosenberg-sipping-nat-scenario
 s-02
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Jonathan and Gonzalo,

Sorry for the delay in providing feedback on this work.  I have a couple of
general points for discussion and then some detailed nits. Following the
thread of this work , I also will post separately a few comments on the
MMUSIC WG ICE draft. 

General: 
1) What are the changes in the -02 from the -01?  The only thing I could
spot was an update to reference [11]. I had done an initial review of the
-01 and the -02 appeared before I had a chance to post those comments, so
perhaps I didn't re-review closely enough. 

2) On the current plan to use this document to replace the existing WG NAT
scenarios document, I have concerns that there will be a loss of background
information that might be useful.  I also think that the current WG document
fairly accurately reflects the plethora of various solution options out
there for this problem (albeit none of them ideal).  And, though I do
generally agree that with the ICE approach in terms of minimizing the
knowledge of topology etc in the client, the reality is that ALGs and some
of the other solutions are likely to remain around for a while and I think
it's useful to keep documentation of where they fit relative to this
solution.  I realize that progressing the existing WG NAT scenarios document
by itself (which would need some work to complete) isn't necessarily
productive. One thought I had was that perhaps parts of the existing
document could be included in the Appendix, with these less desireable
alternatives referenced in the corresponding sections of the ICE based
scenarios.  In a sense, this might add strength to why ICE would be
preferred to these alternatives. 

3) Somewhat related to point 2, I really like the completeness of the flows
in the existing WG NAT scenarios as they do include the SIP messages and the
media flow in many cases. I realize that this draft does detail the
appropriate SDP, but the message flows consist entirely of the STUN and TURN
messages.  I realize that adding the SIP messages will dramatically increase
the size of the flows.  Perhaps adding one or two detailed flows at the
beginning (or as appropriate) and then just referring to Steps x, y, and z,
etc. in those flows would add some value.  


Specific NITs/comments:
1) Relative to General point 2 above, I think one value of ICE beyond what's
in the existing WG NAT scenarios document is the v6 to v4 scenario.  I think
it would be good to highlight that in the abstract. I would propose
extending the sentence:
"The examples cover a range of network topologies and use cases." 
to:
"The examples cover a range of network topologies and use cases, including
IPv6 networks interworking with IPv4." 

2) Page 6, section 2.1, 1st para, next to last sentence:  "Thought not
show,..." should be "Though not shown, ...".

3) page 16, section 2.2, 1st para, last sentence: "A's discovers..." should
be with "A discovers...". 

4) page 18, section 2.2, paragraph between Figure 9 and Figure 2, for
readability I would suggest splitting this into 2 paragraphs starting with
the 4th sentence "B now begins connectivity checks." Also, consistent with
the layout for section 2.2, it might be better to move that second portion
of the paragraph after Figure 10. 

5) Page 21, section 2.2, OPEN ISSUE, last sentence: "doesnt" should be
"doesn't".  Also, shouldn't this OPEN ISSUE be captured in the MMUSIC WG ICE
document, as doesn't it reflect an algorithmic problem with the
prioritization scheme?  

6) Page 28, section 3.1, last para, 2nd sentence: "Thats" should be "That's"


7) Page 29, section 3.2, last para, last sentence would read more clearly by
explicitly mentioning that it will not lockdown as follows:
" Here, it will not lockdown, since the address learned from the TURN server
                    ---------    
  is never advertised to any peers."

7) Page 29, section 3.2, related to my general comment 3 above,  it would be
nice to have a message flow here, showing the continued flow though the TURN
server for this scenario. 

8) Page 33, section 3.3, SDP should be labeled as Figure 21. 

9) Page 33, paragraph immediately following SDP, "... B;s connectivity
test." should be "...B's connectivity test."  

10) Page 47, section 6, 1st paragraph, "...packets addresses to that
PREFIX..." should be "...packets addressed to that PREFIX..."

11) Page 55, I'm assuming Reference [4] needs to be updated as the MMUSIC WG
ICE document.


Regards,
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] Re: General comments and NITs on draft-rosenberg-sipping-nat-scenario
 s-02
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Mary,

a quick answer to your first point:

> 1) What are the changes in the -02 from the -01?  The only thing I could
> spot was an update to reference [11]. I had done an initial review of the
> -01 and the -02 appeared before I had a chance to post those comments, so
> perhaps I didn't re-review closely enough. 

The RFC editor got confused somehow when publishing this draft. So, we 
ended up with two different documents called the same way (version 01): 
an old version, and the new version... to resolve this mess, we 
submitted the 02 version.

Gonzalo



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Transcoding Services Invocation in the Session Initiation Protocol Using Third Party Call Control
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-sipping-transc-3pcc-00.txt
	Pages		: 18
	Date		: 2004-2-5
	
This document describes how to invoke transcoding services using SIP
   and third party call control. This way of invocation meets the
   requirements for SIP regarding transcoding services invocation to
   support deaf, hard of hearing and speech-impaired individuals.

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Subject: RE: [Sipping] Conference policy URI
Date: Fri, 6 Feb 2004 09:51:37 +0200
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Thanks Jonathan. A draft has been submitted to the SIP WG.

http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-i=
nfo-purpose-00.txt

Regards,
Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: 24.January.2004 04:44
> To: Khartabil Hisham (Nokia-TP/Helsinki)
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] Conference policy URI
>=20
>=20
> I think this is a reasonable usage of Call-Info. However, it can only=20
> be done if a "purpose" tag value is registered with IANA, as=20
> indicated=20
> by section 20.9. However, it ocurred to me, when reading=20
> this, that we=20
> never set up an IANA registry for call-info purpose tokens! Oops!=20
> Thats easily rectifiable; when the draft defining the=20
> 'policy-uri' (or=20
> whatever its called) purpose tag is written, it should also=20
> create the=20
> registry and register the other tokens defined in rfc3261.
>=20
> -Jonathan R.
>=20
> hisham.khartabil@nokia.com wrote:
>=20
> > Hi,
> >=20
> > There was consensus on the XCON mailing list that the conference
> > state event package needs to include in the notification the
> > conference policy URI.
> >=20
> > There was also consensus that some participants may not want to
> > subscribe to the conference package but would still like to learn
> > the conference policy URI. In this case, it was suggested that the
> > Call-info header carries such information. Perhaps the
> > cc-conferencing draft can include examples of that.
> >=20
> > Thanks, Hisham
> >=20
> > _______________________________________________ Sipping mailing
> > list  https://www1.ietf.org/mailman/listinfo/sipping This list is
> > for NEW development of the application of SIP Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use
> > sip@ietf.org for new developments of core SIP
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
>=20

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



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From: "Jagdeep Walia" <jagdeepw@hotmail.com>
To: sipping@ietf.org
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Subject: [Sipping] SIP-T and BICC
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Dear Aarti,
Since both have been developed to do the same job, the key to decide which 
one of them, really depends what your network looks like in actual pactice 
but Since the world is not always near to reality, the reason that would 
decide which protocol would be the mighty one standing test of time are, 
industry momemtum and hype generated. Its purely aesthetic than technical 
reasons since there are always ways how you put the point across.
I believe in case it is incumbent operator who has a a fixed network in 
place and would like to migrate to a NGN infrastructure today, then he 
choice of BICC would be more suitable since it is based/extension of SS7 (so 
that a signalling control is there for the call bearer transversing through 
theATM/ IP network). It is possible to transport BICC over TDM/ATM/IP 
networks. The benefit that the incumbent drives from using BICC is that all 
the services and application that are supported via SS7 in his existing 
circuit switching network, can also still be available through BICC. But as 
regards to this SIP is still a evolving protocol which has still to have all 
capabilities e.g regarding support of TCAP services. It is a evolving 
protocol and would take sometime before thinghs settle down.
SIP comes from IETF world where the protocol or standards are defines based 
on RFC which so as to say are bit open and loose when it comes to defining a 
certain procedure as it allows you to have flexibility and there could be 
different RFC which could coexist, so sometime the strenght of being open 
and flexible as we have come to understand is the Buzz word in todays 
telecom industry creates problem when we set out to interoperate. Thats 
quite evident from the fact that eventhough there had been IP router for so 
long, the interoperability for feature support between them took so long.
So these things can be avoided in case BICC is used since it comes from 
ITU-T which truly defines standards and they are rigid to more extent when 
it comes down to implementing certain capabilities and functionalities.
I see SIP-t presently could be good for some new operators who start with 
services that are supported via SIP-T. SIP-t surely is nearer to the concept 
of IP and would eventually be existing in most networks in future but the 
only thing is when does it get stabalised and a more rigid and defined 
protocol.
ITU-T started work in the past on BICC CS3 which finally focussed on the 
defining a frame work for interworking between SIP and BICC/ISUP based 
network. In a recent SG11 meeting held in Geneva from September 1- September 
12, ITU-T has already finalised the receommendation which existed as Q 
1912.SIP and has become a recommendation called Q 1912.5.


Regards
jw

Jagdeep Walia
Telecommunication Engineer
Email: Jagdeepw@hotmail.com

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Framework for Transcoding with the Session Initiation Protocol
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-sipping-transc-framework-00.txt
	Pages		: 10
	Date		: 2004-2-7
	
This document defines a framework for transcoding with SIP. This
   framework includes how to discover the need of transcoding services
   in a session and how to invoke those transcoding services. Two models
   for transcoding services invocation are discussed; the conference
   bridge model and the third party call control model. Both models meet
   the requirements for SIP regarding transcoding services invocation to
   support deaf, hard of hearing and speech-impaired individuals.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-transc-framework-00.txt

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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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--NextPart--



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



From exim@www1.ietf.org  Sun Feb  8 06:41:46 2004
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Date: Sun, 08 Feb 2004 13:37:34 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Folks,

we are trying to maintain our supplemental page up to date; especially 
the draft database. So, I would appreciate if you let me know every time 
you submit an individual contribution to SIPPING (I generally get cc:ed 
in submissions of WG items). Some of them are not announced in the 
mailing list, or I may miss the announcement.

http://www.softarmor.com/wgdb/?wg=SIPPING

We hope that this WG-level ID tracker is useful to the community.

Thanks,

Gonzalo
SIPPING co-chair



This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption, interruption, unauthorized amendment, tampering and viruses, and we only send and receive e-mails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.

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



From exim@www1.ietf.org  Sun Feb  8 11:46:01 2004
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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        sipping
	 <sipping@ietf.org>
Cc: Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>
Subject: RE: [Sipping] Submissions to SIPPING
Date: Sun, 8 Feb 2004 10:41:14 -0600 
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Gonzalo, 

In response to your point about the usefulness of this information; this
information is extremely useful to the community and I really, really
appreciate the time and effort that you chairs put into tracking all of
this. 

Thanks,
Mary H. Barnes
mary.barnes@nortelnetworks.com
972-684-5432

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
Sent: Sunday, February 08, 2004 5:38 AM
To: sipping
Cc: Rohan Mahy; Dean Willis
Subject: [Sipping] Submissions to SIPPING


Folks,

we are trying to maintain our supplemental page up to date; especially 
the draft database. So, I would appreciate if you let me know every time 
you submit an individual contribution to SIPPING (I generally get cc:ed 
in submissions of WG items). Some of them are not announced in the 
mailing list, or I may miss the announcement.

http://www.softarmor.com/wgdb/?wg=SIPPING

We hope that this WG-level ID tracker is useful to the community.

Thanks,

Gonzalo
SIPPING co-chair



This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption,
interruption, unauthorized amendment, tampering and viruses, and we only
send and receive e-mails on the basis that we are not liable for any such
corruption, interception, amendment, tampering or viruses or any
consequences thereof.

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

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



From exim@www1.ietf.org  Sun Feb  8 16:34:01 2004
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Folks,

we would like to announce the start of the WGLC for the following draft.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-reason-header-for-preemption-00.txt

This WGLC will end on Feb 29th.

Send your comments to this mailing list and to the author.

Thanks,

Gonzalo
SIPPING co-chair



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Subject: [Sipping] Agenda Requests
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Folks,

we will be gathering agenda requests for Seoul under the following link:

http://www.softarmor.com/sipping/meets/ietf59/requests.html

As usual, if you use the template provided at the end of this email to 
submit your agenda request , it will make this task easier. Agenda 
requests are submitted in an email addressed to the chairs.

Regarding the requests we will be accepting and rejecting, this time, we 
intend to use a big part of the meeting to discuss issues related to WG 
items. We will only discuss individual contributions that have received 
considerable attention since the last meeting.


     <tr>
       <td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
Gonzalo Camarillo</a></td>
       <td>Exploders</td>
       <td><center>5</center></td>
       <td>
<a 
href="http://www.ietf.org/internet-drafts/draft-camarillo-sipping-exploders-02.txt">
draft-camarillo-sipping-exploders-02.txt</a></td>
     </tr>


Thanks,

Gonzalo
SIPPING co-chair



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



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	Tue, 10 Feb 2004 17:10:31 +0100 (MET)
From: "Christian Stredicke" <stredicke@snom.de>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'F.S.Salloum'" <ssal@intracom.gr>, <suhac@ednet.co.uk>,
        <sipping@ietf.org>
Subject: RE: [Sipping] Music On Hold
Date: Tue, 10 Feb 2004 17:10:31 +0100
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Hi Jonathan,

I wrote something about this topic on

http://softarmor.com/wgdb/docs/draft-stredicke-sipping-late-media-01.txt

Of course, technically it would be possible to initiate a conference =
with
3PCC that mixes billing beeps into the audio stream. Maybe someone can =
prove
that 3PCC is a Turing-machine; then we can stop the discussion if =
something
is possible with this model. But I doubt that this will be a big =
success.

Today I had another discussion with a customer that wanted to know why =
music
on hold is NOT working with a phone (from a very popular vendor, not us)
which is (big surprise) NOT initiating 3PCC MOH. Maybe, again, I am too
stupid to see the configuration option in the product of another vendor.

Or just use a B2BUA...

CS

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
> Jonathan Rosenberg
> Sent: Thursday, February 05, 2004 8:36 AM
> To: Christian Stredicke
> Cc: 'F.S.Salloum'; suhac@ednet.co.uk; sipping@ietf.org
> Subject: Re: [Sipping] Re: [Sip] Music On Hold
>=20
> inline.
>=20
> Christian Stredicke wrote:
>=20
>=20
> >>A lot of commercial product also our HW
> >>phones(http://netphone.intracom.gr/)
> >>support this approach with no problem
> >>
> >
> > [Christian Stredicke] We also support this, but I can't say "no
problem".
> > It's easy to get it working in the lab, but the real world shows us =
a
> > different picture. Calls get lost (people just plug the cable), =
transfer
> > fails, people are sitting behind NAT, media server running out of
> resources,
> > media server and negotiated codec don't find a common codec (cause =
only
> the
> > answer is available). Maybe is a question of the pain level, but for =
us
> it's
> > a "problem".
> >
> > Of course, anything can be solved, but what I am trying to say is =
the
> > Alert-Info model would be much simpler. We could have the same
> discussion in
> > ring tones!!!
>=20
> Its not clear to me that many of the problems you list above would go
> away with an Alert-Info style approach. Certainly, those calls can get
> lost, there can be nat problems, the server can still run out of
> resources, etc.
>=20
> You also indicate that there is a problem finding a common codec. Of
> course, if there are simply no codecs in common no signaling in the
> world will help. However, I don't think thats what you mean. In the =
flow
> in the service examples draft, the user placing the call on hold
> (b-party) issues an invite to the music on hold server with no sdp. =
That
> triggers the music on hold server to issue an offer with a full codec
> list. That is passed to the A-party, which picks the one(s) it wants.
> Thus, the music on hold can use different codecs than what is =
negotiated
> in the call between A and B. So, I dont see a specific problem. Can =
you
> elaborate on the difficulty you see?
>=20
>=20
> >>
> >>I think we should consider security issues check this link with a
> >>discussions with alert info stuff
> >>(http://www.atm.tut.fi/list-archive/sip/msg01278.html)
> >>
> >
> > [Christian Stredicke] There are also (bigger?) security problems =
with
> the
> > 3PCC model. And the Alert-Info security problem has to be solved =
anyway
> (or
> > has been solved already).
>=20
> 3pcc in this specific case is not so hard, since (presumably) there is =
a
> relationship between the b-party and the music on hold server. If the
> B-party rather tells the A-party to contact the music on hold server
> directly (using an alert-info kind of approach), now there needs to be =
a
> security relationship between the A-party on the music on hold server,
> which doesnt exist. Relaying signaling through the B-party makes the
> security more tenable, I think.
>=20
> -Jonathan R.
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



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Cc: sipping <sipping@ietf.org>, Dean Willis <dean.willis@softarmor.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] more on ... Agenda Requests
Date: Tue, 10 Feb 2004 19:07:58 +0100
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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A few more things about Agenda Requests.

This IETF I will provide a printable, pretty, properly-paginated PDF  
file of the reading list on softarmor.  (Options for 1-up double sided  
and 2-up double sided for both US-Letter and A4 paper will be  
available).

If you are requesting agenda time, your draft(s) need to be properly  
formatted (I strongly recommend using xml2rfc).  I will send your draft  
back to you for reformatting if they do not print properly.

Also, if you *did* use xml2rfc, please attach a copy of the HTML  
version in your request.

thanks,
-rohan



On Feb 10, 2004, at 1:28 PM, Gonzalo Camarillo wrote:

> Folks,
>
> we will be gathering agenda requests for Seoul under the following  
> link:
>
> http://www.softarmor.com/sipping/meets/ietf59/requests.html
>
> As usual, if you use the template provided at the end of this email to  
> submit your agenda request , it will make this task easier. Agenda  
> requests are submitted in an email addressed to the chairs.
>
> Regarding the requests we will be accepting and rejecting, this time,  
> we intend to use a big part of the meeting to discuss issues related  
> to WG items. We will only discuss individual contributions that have  
> received considerable attention since the last meeting.
>
>
>     <tr>
>       <td><a href="mailto:Gonzalo.Camarillo@ericsson.com">
> Gonzalo Camarillo</a></td>
>       <td>Exploders</td>
>       <td><center>5</center></td>
>       <td>
> <a  
> href="http://www.ietf.org/internet-drafts/draft-camarillo-sipping- 
> exploders-02.txt">
> draft-camarillo-sipping-exploders-02.txt</a></td>
>     </tr>
>
>
> Thanks,
>
> Gonzalo
> SIPPING co-chair
>
>
>
> This communication is confidential and intended solely for the  
> addressee(s). Any unauthorized review, use, disclosure or distribution  
> is prohibited. If you believe this message has been sent to you in  
> error, please notify the sender by replying to this transmission and  
> delete the message without disclosing it. Thank you.
>
> E-mail including attachments is susceptible to data corruption,  
> interruption, unauthorized amendment, tampering and viruses, and we  
> only send and receive e-mails on the basis that we are not liable for  
> any such corruption, interception, amendment, tampering or viruses or  
> any consequences thereof.
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



From exim@www1.ietf.org  Tue Feb 10 15:23:40 2004
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Date: Tue, 10 Feb 2004 15:17:47 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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CC: sipping@ietf.org
Subject: Re: [Sipping] Conference policy URI
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Thanks for doing this. Some comments.

Firstly, on the registry itself. You need to give clearer guidance to 
IANA on the registry. I would suggest including a table that shows how 
it should look like. You need to tell IANA what information needs to be 
provided in future RFCs that register additional values (value, RFC 
number, registrant, etc.). Please take a look at rfc2434 
(ftp://ftp.rfc-editor.org/in-notes/rfc2434.txt).

Secondly, on the "policy-uri" value - you need to specify, I think, what 
the URI type is, and what the content is at that URI (at least a 
mandatory to implement one) in order to get interoperability. Unlike the 
other stuff, which is just rendered for user information, this resource 
is processed by an automata, and I think you need to say more about it.

Thanks,
Jonathan R.

hisham.khartabil@nokia.com wrote:

> Thanks Jonathan. A draft has been submitted to the SIP WG.
> 
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00.txt
> 
> Regards,
> Hisham
> 
> 
>>-----Original Message-----
>>From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>Sent: 24.January.2004 04:44
>>To: Khartabil Hisham (Nokia-TP/Helsinki)
>>Cc: sipping@ietf.org
>>Subject: Re: [Sipping] Conference policy URI
>>
>>
>>I think this is a reasonable usage of Call-Info. However, it can only 
>>be done if a "purpose" tag value is registered with IANA, as 
>>indicated 
>>by section 20.9. However, it ocurred to me, when reading 
>>this, that we 
>>never set up an IANA registry for call-info purpose tokens! Oops! 
>>Thats easily rectifiable; when the draft defining the 
>>'policy-uri' (or 
>>whatever its called) purpose tag is written, it should also 
>>create the 
>>registry and register the other tokens defined in rfc3261.
>>
>>-Jonathan R.
>>
>>hisham.khartabil@nokia.com wrote:
>>
>>
>>>Hi,
>>>
>>>There was consensus on the XCON mailing list that the conference
>>>state event package needs to include in the notification the
>>>conference policy URI.
>>>
>>>There was also consensus that some participants may not want to
>>>subscribe to the conference package but would still like to learn
>>>the conference policy URI. In this case, it was suggested that the
>>>Call-info header carries such information. Perhaps the
>>>cc-conferencing draft can include examples of that.
>>>
>>>Thanks, Hisham
>>>
>>>_______________________________________________ Sipping mailing
>>>list  https://www1.ietf.org/mailman/listinfo/sipping This list is
>>>for NEW development of the application of SIP Use
>>>sip-implementors@cs.columbia.edu for questions on current sip Use
>>>sip@ietf.org for new developments of core SIP
>>>
>>
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Emergency Services URI for the Session Initiation Protocol
	Author(s)	: H. Schulzrinne
	Filename	: draft-ietf-sipping-sos-00.txt
	Pages		: 17
	Date		: 2004-2-10
	
As part of an overall architecture for supporting emergency calling for the Session Initiation Protocol (SIP), this document defines universal emergency SIP URIs, sip:sos@domain and sips:sos@domain, that allows SIP user agents to contact the local emergency call center.  It also defines conventions that increase the high probability of reaching the
appropriate emergency call center. The document does not define any SIP protocol extensions.

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



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From: "James M. Polk" <jmpolk@cisco.com>
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Subject: [Sipping] Concern within sos ID - user addresses reserved
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I have an concern within the new sos ID, and that concern is in section 4.1 
- which reserves the following additional user addresses:

	sos.fire	fire brigade
	sos.rescue	ambulance (rescue)
	sos.marine	marine guard
	sos.police	police (law enforcement)
	sos.mountain 	mountain rescue

as I have little confidence that Joe Citizen will be generally capable to 
choose the correct one given their emergency experience. If these addresses 
are intended for between two ERCs (for a call transfer, for example) - I 
don't see the text stating that here.

My issue is not really with the reservation of the user addresses, but with 
the lack of semantics and potential drawbacks surrounding using one of 
these when more than one should have been contacted (that a multipurpose 
ERC could determine by asking the right questions) or if the wrong one was 
dispatched because not enough was asked of the person in distress.

I think that if Joe Citizen is educated enough to call the *right* response 
center (say fire vs. mountain rescue) - then I retract this concern

cheers,
James

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


_______________________________________________
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This list is for NEW development of the application of SIP
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'James M. Polk'" <jmpolk@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Tue, 10 Feb 2004 17:12:06 -0500
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Suppose that we assume that Joe Citizen is actually pretty good at figuring
out what he needs, but isn't always thinking about ancilliary issues, like
a big fire requires police to direct traffic, and perhaps ambulances.
A fire is a fire.

Suppose that he calls "sos.fire" for a fire which:
	Causes an automatic dispatch to the closest available fire brigade
	Connects him to the brigade captain
	Bridges him to an emergency response center operator

Joe explains the exact situation.  
It may be that the call is cancelled.
It may be that additional units are needed.
It is unlikely that if he calls "sos.fire" that he doesn't need 
  a fire company but does need a mountain rescue team

In any jurisdiction, it could easily be arranged that all calls to sos.*
are answered at one ERC with a visual indication of what the caller asked
for.

I don't see a downside, and I see a potential upside.

Of course the original idea was that in some jurisdictions, there is no
coordinated emergency response center.  You HAVE to call the specific
service.

Brian

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Tuesday, February 10, 2004 4:55 PM
> To: sipping@ietf.org
> Subject: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> I have an concern within the new sos ID, and that concern is 
> in section 4.1 
> - which reserves the following additional user addresses:
> 
> 	sos.fire	fire brigade
> 	sos.rescue	ambulance (rescue)
> 	sos.marine	marine guard
> 	sos.police	police (law enforcement)
> 	sos.mountain 	mountain rescue
> 
> as I have little confidence that Joe Citizen will be 
> generally capable to 
> choose the correct one given their emergency experience. If 
> these addresses 
> are intended for between two ERCs (for a call transfer, for 
> example) - I 
> don't see the text stating that here.
> 
> My issue is not really with the reservation of the user 
> addresses, but with 
> the lack of semantics and potential drawbacks surrounding 
> using one of 
> these when more than one should have been contacted (that a 
> multipurpose 
> ERC could determine by asking the right questions) or if the 
> wrong one was 
> dispatched because not enough was asked of the person in distress.
> 
> I think that if Joe Citizen is educated enough to call the 
> *right* response 
> center (say fire vs. mountain rescue) - then I retract this concern
> 
> cheers,
> James
> 
>                                 *******************
>                  Truth is not to be argued... it is to be presented
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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Date: Tue, 10 Feb 2004 17:13:52 -0500
From: David Oran <oran@cisco.com>
To: "James M. Polk" <jmpolk@cisco.com>, sipping@ietf.org
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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--On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk" 
<jmpolk@cisco.com> wrote:

> I have an concern within the new sos ID, and that concern is in section
> 4.1 - which reserves the following additional user addresses:
>
> 	sos.fire	fire brigade
> 	sos.rescue	ambulance (rescue)
> 	sos.marine	marine guard
> 	sos.police	police (law enforcement)
> 	sos.mountain 	mountain rescue
>
> as I have little confidence that Joe Citizen will be generally capable to
> choose the correct one given their emergency experience. If these
> addresses are intended for between two ERCs (for a call transfer, for
> example) - I don't see the text stating that here.
>
I thought this was just to reflect current practice in places that already 
have different numbers for the different emergency services. Is your 
concern that the *current* numbers are understood by joe citizen but these 
are not? If so, the UA can just do the mapping as it has to do already for 
911->sos@domain.


> My issue is not really with the reservation of the user addresses, but
> with the lack of semantics and potential drawbacks surrounding using one
> of these when more than one should have been contacted (that a
> multipurpose ERC could determine by asking the right questions) or if the
> wrong one was dispatched because not enough was asked of the person in
> distress.
>
If these were all we had and there were no generic sos@domain I think this 
would be a fairly large concern. Also concerning would be a deployment in a 
geography that only had the generic sos@domain and there was no AoR mapping 
of sos.*@domain -> sos@domain.

> I think that if Joe Citizen is educated enough to call the *right*
> response center (say fire vs. mountain rescue) - then I retract this
> concern
>
I doubt this is a good assumption, but what's the alternative in places 
that have already decided to have separate AoRs (service codes in 
PSTN-speak) for separate emergency service classes?

Dave.
> cheers,
> James
>
>                                 *******************
>                  Truth is not to be argued... it is to be presented
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP





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



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Date: Tue, 10 Feb 2004 16:52:13 -0600
To: "Rosen, Brian" <Brian.Rosen@marconi.com>, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
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 .comms.marconi.com>
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Suppose there is a fire on a boat near a harbor - which do you call?

But seriously, suppose there is a fire and the fire department is called 
only. They show up and a person (who set the fire) is there showing his 
gun. The fire department won't go near that situation until the police show 
up. This happens in big cities.

Should the fire department now place a new call to the police department 
for each of its sos@fire calls?

At 05:12 PM 2/10/2004 -0500, Rosen, Brian wrote:
>Suppose that we assume that Joe Citizen is actually pretty good at figuring
>out what he needs, but isn't always thinking about ancilliary issues, like
>a big fire requires police to direct traffic, and perhaps ambulances.
>A fire is a fire.
>
>Suppose that he calls "sos.fire" for a fire which:
>         Causes an automatic dispatch to the closest available fire brigade
>         Connects him to the brigade captain
>         Bridges him to an emergency response center operator
>
>Joe explains the exact situation.
>It may be that the call is cancelled.
>It may be that additional units are needed.
>It is unlikely that if he calls "sos.fire" that he doesn't need
>   a fire company but does need a mountain rescue team
>
>In any jurisdiction, it could easily be arranged that all calls to sos.*
>are answered at one ERC with a visual indication of what the caller asked
>for.
>
>I don't see a downside, and I see a potential upside.
>
>Of course the original idea was that in some jurisdictions, there is no
>coordinated emergency response center.  You HAVE to call the specific
>service.
>
>Brian
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: Tuesday, February 10, 2004 4:55 PM
> > To: sipping@ietf.org
> > Subject: [Sipping] Concern within sos ID - user addresses reserved
> >
> >
> > I have an concern within the new sos ID, and that concern is
> > in section 4.1
> > - which reserves the following additional user addresses:
> >
> >       sos.fire        fire brigade
> >       sos.rescue      ambulance (rescue)
> >       sos.marine      marine guard
> >       sos.police      police (law enforcement)
> >       sos.mountain    mountain rescue
> >
> > as I have little confidence that Joe Citizen will be
> > generally capable to
> > choose the correct one given their emergency experience. If
> > these addresses
> > are intended for between two ERCs (for a call transfer, for
> > example) - I
> > don't see the text stating that here.
> >
> > My issue is not really with the reservation of the user
> > addresses, but with
> > the lack of semantics and potential drawbacks surrounding
> > using one of
> > these when more than one should have been contacted (that a
> > multipurpose
> > ERC could determine by asking the right questions) or if the
> > wrong one was
> > dispatched because not enough was asked of the person in distress.
> >
> > I think that if Joe Citizen is educated enough to call the
> > *right* response
> > center (say fire vs. mountain rescue) - then I retract this concern
> >
> > cheers,
> > James
> >
> >                                 *******************
> >                  Truth is not to be argued... it is to be presented
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >


cheers,
James

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


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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'James M. Polk'" <jmpolk@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Tue, 10 Feb 2004 18:01:02 -0500
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Of course we are speculating.

In most coastal areas, fireboats are part of the fire department, and not
the coast guard.  If you are far enough out to sea, then your location
will cause the Coast Guard to respond no matter what.

An appropriately smart system would have rules for such things as "do not
enter unguarded", and do the right thing.  I expect the ERC operator will
always answer; the only question is does the responder get going earlier,
with the possibility of increased callout cancellations.  We'll see.
Having it is better than not having it methinks.

Brian

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: Tuesday, February 10, 2004 5:52 PM
> To: Rosen, Brian; sipping@ietf.org
> Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> Suppose there is a fire on a boat near a harbor - which do you call?
> 
> But seriously, suppose there is a fire and the fire 
> department is called 
> only. They show up and a person (who set the fire) is there 
> showing his 
> gun. The fire department won't go near that situation until 
> the police show 
> up. This happens in big cities.
> 
> Should the fire department now place a new call to the police 
> department 
> for each of its sos@fire calls?
> 
> At 05:12 PM 2/10/2004 -0500, Rosen, Brian wrote:
> >Suppose that we assume that Joe Citizen is actually pretty 
> good at figuring
> >out what he needs, but isn't always thinking about 
> ancilliary issues, like
> >a big fire requires police to direct traffic, and perhaps ambulances.
> >A fire is a fire.
> >
> >Suppose that he calls "sos.fire" for a fire which:
> >         Causes an automatic dispatch to the closest 
> available fire brigade
> >         Connects him to the brigade captain
> >         Bridges him to an emergency response center operator
> >
> >Joe explains the exact situation.
> >It may be that the call is cancelled.
> >It may be that additional units are needed.
> >It is unlikely that if he calls "sos.fire" that he doesn't need
> >   a fire company but does need a mountain rescue team
> >
> >In any jurisdiction, it could easily be arranged that all 
> calls to sos.*
> >are answered at one ERC with a visual indication of what the 
> caller asked
> >for.
> >
> >I don't see a downside, and I see a potential upside.
> >
> >Of course the original idea was that in some jurisdictions, 
> there is no
> >coordinated emergency response center.  You HAVE to call the specific
> >service.
> >
> >Brian
> >
> > > -----Original Message-----
> > > From: James M. Polk [mailto:jmpolk@cisco.com]
> > > Sent: Tuesday, February 10, 2004 4:55 PM
> > > To: sipping@ietf.org
> > > Subject: [Sipping] Concern within sos ID - user addresses reserved
> > >
> > >
> > > I have an concern within the new sos ID, and that concern is
> > > in section 4.1
> > > - which reserves the following additional user addresses:
> > >
> > >       sos.fire        fire brigade
> > >       sos.rescue      ambulance (rescue)
> > >       sos.marine      marine guard
> > >       sos.police      police (law enforcement)
> > >       sos.mountain    mountain rescue
> > >
> > > as I have little confidence that Joe Citizen will be
> > > generally capable to
> > > choose the correct one given their emergency experience. If
> > > these addresses
> > > are intended for between two ERCs (for a call transfer, for
> > > example) - I
> > > don't see the text stating that here.
> > >
> > > My issue is not really with the reservation of the user
> > > addresses, but with
> > > the lack of semantics and potential drawbacks surrounding
> > > using one of
> > > these when more than one should have been contacted (that a
> > > multipurpose
> > > ERC could determine by asking the right questions) or if the
> > > wrong one was
> > > dispatched because not enough was asked of the person in distress.
> > >
> > > I think that if Joe Citizen is educated enough to call the
> > > *right* response
> > > center (say fire vs. mountain rescue) - then I retract 
> this concern
> > >
> > > cheers,
> > > James
> > >
> > >                                 *******************
> > >                  Truth is not to be argued... it is to be 
> presented
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> 
> 
> cheers,
> James
> 
>                                 *******************
>                  Truth is not to be argued... it is to be presented
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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Date: Tue, 10 Feb 2004 17:11:15 -0600
To: David Oran <oran@cisco.com>, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
In-Reply-To: <2147483647.1076433232@[10.32.245.153]>
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Comments in-line

At 05:13 PM 2/10/2004 -0500, David Oran wrote:


>--On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk" 
><jmpolk@cisco.com> wrote:
>
>>I have an concern within the new sos ID, and that concern is in section
>>4.1 - which reserves the following additional user addresses:
>>
>>         sos.fire        fire brigade
>>         sos.rescue      ambulance (rescue)
>>         sos.marine      marine guard
>>         sos.police      police (law enforcement)
>>         sos.mountain    mountain rescue
>>
>>as I have little confidence that Joe Citizen will be generally capable to
>>choose the correct one given their emergency experience. If these
>>addresses are intended for between two ERCs (for a call transfer, for
>>example) - I don't see the text stating that here.
>I thought this was just to reflect current practice in places that already 
>have different numbers for the different emergency services.

This was not articulated, if intended

>Is your concern that the *current* numbers are understood by joe citizen 
>but these are not? If so, the UA can just do the mapping as it has to do 
>already for 911->sos@domain.

I'm concerned that, for example in the US, that Joe Citizen needs to learn 
911 as general purpose, 912 is police, 913 is fire, etc...

How is someone to dial into numeric keypad anything other than 911 in the 
US (and I'm picking on the US here)?



>>My issue is not really with the reservation of the user addresses, but
>>with the lack of semantics and potential drawbacks surrounding using one
>>of these when more than one should have been contacted (that a
>>multipurpose ERC could determine by asking the right questions) or if the
>>wrong one was dispatched because not enough was asked of the person in
>>distress.
>If these were all we had and there were no generic sos@domain I think this 
>would be a fairly large concern.

I'm not as much saying that the additional user addresses need necessarily 
be removed, just would like more usage text on how/when/where this could be 
used, with a potentially a "pitfall considerations" section for the 
ramifications of if the wrong organization was contacted and that delayed 
the proper help... that is, unless I'm the lone voice with this opinion.

>Also concerning would be a deployment in a geography that only had the 
>generic sos@domain and there was no AoR mapping of sos.*@domain -> sos@domain.
>
>>I think that if Joe Citizen is educated enough to call the *right*
>>response center (say fire vs. mountain rescue) - then I retract this
>>concern
>I doubt this is a good assumption, but what's the alternative in places 
>that have already decided to have separate AoRs (service codes in 
>PSTN-speak) for separate emergency service classes?

Add the text saying this, and this thread wouldn't have been started....


>Dave.
>>cheers,
>>James
>>
>>                                 *******************
>>                  Truth is not to be argued... it is to be presented
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>
>
>
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


cheers,
James

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


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



From exim@www1.ietf.org  Tue Feb 10 18:39:52 2004
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "'Sipping@Ietf. Org'" <sipping@ietf.org>
Date: Wed, 11 Feb 2004 00:35:04 +0100
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Subject: [Sipping] RE: I-D ACTION:draft-schulzrinne-sipping-emergency-arch-00.txt
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Henning,
I have a couple of small comments to your good emergency access spec.

Section 2
Paragraph 7.
TDD is a local US name on text telephony. So,
replace "TDD text"  with "text telephony"

Section 10.1
Add detail on the real time text medium
Interactive text: RTP according to RFC 2793.

Section "Normative References"
Add:

 [RFC2793]  Hellstrom, G, "RTP Payload for Text Conversation",
            RFC 2793, May 2000

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

Regards
Gunnar


-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Tuesday, February 10, 2004 9:56 PM
To: IETF-Announce:
Subject: I-D ACTION:draft-schulzrinne-sipping-emergency-arch-00.txt


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


	Title		: Emergency Services for Internet Telephony Systems
	Author(s)	: H. Schulzrinne, B. Rosen
	Filename	: draft-schulzrinne-sipping-emergency-arch-00.txt
	Pages		: 22
	Date		: 2004-2-10

Summoning emergency help is a core feature of telephone networks. This
document describes how the Session Initiation Protocol (SIP) can be used =
to
provide advanced emergency services for voice-over-IP (VoIP). The
architecture employs standard SIP features and requires no new protocol
mechanisms. DNS is used to map civil and geospatial locations to the
appropriate emergency call center.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-schulzrinne-sipping-emergency-a=
rch
-00.txt

To remove yourself from the IETF Announcement list, send a message to
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Internet-Drafts are also available by anonymous FTP. Login with the usern=
ame
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A list of Internet-Drafts directories can be found in
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_______________________________________________
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From exim@www1.ietf.org  Tue Feb 10 19:57:49 2004
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "James M. Polk" <jmpolk@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Wed, 11 Feb 2004 00:42:05 -0000
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Well stop trying to think US.

These auxiliary addresses surely only have a direct impact on which to use when there are multiple PSAPs, one for each distinguished entity already.

If there is only one general service, then presumably 911 will translate to the general uri rather than any of the specifics, and USA citizen Joe Public will be told to use the general URI in the USA. 

In countries that use multiple telephone numbers to go to different service PSAPs, then Joe Public will be told the appropriate information, as the same rules will apply for any interservice cooperation where multiple services are needed.

After all, we are not defining the emergency services here, only a means of directing a call to an appropriate PSAP, where an existing emergency service methodology already exists.

regards

Keith

> -----Original Message-----
> From: James M. Polk [mailto:jmpolk@cisco.com]
> Sent: 10 February 2004 23:11
> To: David Oran; sipping@ietf.org
> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> Comments in-line
> 
> At 05:13 PM 2/10/2004 -0500, David Oran wrote:
> 
> 
> >--On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk" 
> ><jmpolk@cisco.com> wrote:
> >
> >>I have an concern within the new sos ID, and that concern 
> is in section
> >>4.1 - which reserves the following additional user addresses:
> >>
> >>         sos.fire        fire brigade
> >>         sos.rescue      ambulance (rescue)
> >>         sos.marine      marine guard
> >>         sos.police      police (law enforcement)
> >>         sos.mountain    mountain rescue
> >>
> >>as I have little confidence that Joe Citizen will be 
> generally capable to
> >>choose the correct one given their emergency experience. If these
> >>addresses are intended for between two ERCs (for a call 
> transfer, for
> >>example) - I don't see the text stating that here.
> >I thought this was just to reflect current practice in 
> places that already 
> >have different numbers for the different emergency services.
> 
> This was not articulated, if intended
> 
> >Is your concern that the *current* numbers are understood by 
> joe citizen 
> >but these are not? If so, the UA can just do the mapping as 
> it has to do 
> >already for 911->sos@domain.
> 
> I'm concerned that, for example in the US, that Joe Citizen 
> needs to learn 
> 911 as general purpose, 912 is police, 913 is fire, etc...
> 
> How is someone to dial into numeric keypad anything other 
> than 911 in the 
> US (and I'm picking on the US here)?
> 
> 
> 
> >>My issue is not really with the reservation of the user 
> addresses, but
> >>with the lack of semantics and potential drawbacks 
> surrounding using one
> >>of these when more than one should have been contacted (that a
> >>multipurpose ERC could determine by asking the right 
> questions) or if the
> >>wrong one was dispatched because not enough was asked of 
> the person in
> >>distress.
> >If these were all we had and there were no generic 
> sos@domain I think this 
> >would be a fairly large concern.
> 
> I'm not as much saying that the additional user addresses 
> need necessarily 
> be removed, just would like more usage text on how/when/where 
> this could be 
> used, with a potentially a "pitfall considerations" section for the 
> ramifications of if the wrong organization was contacted and 
> that delayed 
> the proper help... that is, unless I'm the lone voice with 
> this opinion.
> 
> >Also concerning would be a deployment in a geography that 
> only had the 
> >generic sos@domain and there was no AoR mapping of 
> sos.*@domain -> sos@domain.
> >
> >>I think that if Joe Citizen is educated enough to call the *right*
> >>response center (say fire vs. mountain rescue) - then I retract this
> >>concern
> >I doubt this is a good assumption, but what's the 
> alternative in places 
> >that have already decided to have separate AoRs (service codes in 
> >PSTN-speak) for separate emergency service classes?
> 
> Add the text saying this, and this thread wouldn't have been 
> started....
> 
> 
> >Dave.
> >>cheers,
> >>James
> >>
> >>                                 *******************
> >>                  Truth is not to be argued... it is to be presented
> >>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >
> >
> >
> >
> >
> >_______________________________________________
> >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >This list is for NEW development of the application of SIP
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sip@ietf.org for new developments of core SIP
> 
> 
> cheers,
> James
> 
>                                 *******************
>                  Truth is not to be argued... it is to be presented
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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Date: Tue, 10 Feb 2004 19:15:04 -0600
To: "Drage, Keith (Keith)" <drage@lucent.com>, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
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At 12:42 AM 2/11/2004 +0000, Drage, Keith (Keith) wrote:
>Well stop trying to think US.

sorry  ;-)


>These auxiliary addresses surely only have a direct impact on which to use 
>when there are multiple PSAPs, one for each distinguished entity already.

for sure


>If there is only one general service, then presumably 911 will translate 
>to the general uri rather than any of the specifics, and USA citizen Joe 
>Public will be told to use the general URI in the USA.

I agree


>In countries that use multiple telephone numbers to go to different 
>service PSAPs, then Joe Public will be told the appropriate information, 
>as the same rules will apply for any interservice cooperation where 
>multiple services are needed.

true enough


>After all, we are not defining the emergency services here, only a means 
>of directing a call to an appropriate PSAP, where an existing emergency 
>service methodology already exists.

I guess then I just want some text that gives the above (most of what you 
wrote) as explanation. I am willing to "send text" here.


>regards
>
>Keith
>
> > -----Original Message-----
> > From: James M. Polk [mailto:jmpolk@cisco.com]
> > Sent: 10 February 2004 23:11
> > To: David Oran; sipping@ietf.org
> > Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> >
> >
> > Comments in-line
> >
> > At 05:13 PM 2/10/2004 -0500, David Oran wrote:
> >
> >
> > >--On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk"
> > ><jmpolk@cisco.com> wrote:
> > >
> > >>I have an concern within the new sos ID, and that concern
> > is in section
> > >>4.1 - which reserves the following additional user addresses:
> > >>
> > >>         sos.fire        fire brigade
> > >>         sos.rescue      ambulance (rescue)
> > >>         sos.marine      marine guard
> > >>         sos.police      police (law enforcement)
> > >>         sos.mountain    mountain rescue
> > >>
> > >>as I have little confidence that Joe Citizen will be
> > generally capable to
> > >>choose the correct one given their emergency experience. If these
> > >>addresses are intended for between two ERCs (for a call
> > transfer, for
> > >>example) - I don't see the text stating that here.
> > >I thought this was just to reflect current practice in
> > places that already
> > >have different numbers for the different emergency services.
> >
> > This was not articulated, if intended
> >
> > >Is your concern that the *current* numbers are understood by
> > joe citizen
> > >but these are not? If so, the UA can just do the mapping as
> > it has to do
> > >already for 911->sos@domain.
> >
> > I'm concerned that, for example in the US, that Joe Citizen
> > needs to learn
> > 911 as general purpose, 912 is police, 913 is fire, etc...
> >
> > How is someone to dial into numeric keypad anything other
> > than 911 in the
> > US (and I'm picking on the US here)?
> >
> >
> >
> > >>My issue is not really with the reservation of the user
> > addresses, but
> > >>with the lack of semantics and potential drawbacks
> > surrounding using one
> > >>of these when more than one should have been contacted (that a
> > >>multipurpose ERC could determine by asking the right
> > questions) or if the
> > >>wrong one was dispatched because not enough was asked of
> > the person in
> > >>distress.
> > >If these were all we had and there were no generic
> > sos@domain I think this
> > >would be a fairly large concern.
> >
> > I'm not as much saying that the additional user addresses
> > need necessarily
> > be removed, just would like more usage text on how/when/where
> > this could be
> > used, with a potentially a "pitfall considerations" section for the
> > ramifications of if the wrong organization was contacted and
> > that delayed
> > the proper help... that is, unless I'm the lone voice with
> > this opinion.
> >
> > >Also concerning would be a deployment in a geography that
> > only had the
> > >generic sos@domain and there was no AoR mapping of
> > sos.*@domain -> sos@domain.
> > >
> > >>I think that if Joe Citizen is educated enough to call the *right*
> > >>response center (say fire vs. mountain rescue) - then I retract this
> > >>concern
> > >I doubt this is a good assumption, but what's the
> > alternative in places
> > >that have already decided to have separate AoRs (service codes in
> > >PSTN-speak) for separate emergency service classes?
> >
> > Add the text saying this, and this thread wouldn't have been
> > started....
> >
> >
> > >Dave.
> > >>cheers,
> > >>James
> > >>
> > >>                                 *******************
> > >>                  Truth is not to be argued... it is to be presented
> > >>
> > >>
> > >>_______________________________________________
> > >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > >>This list is for NEW development of the application of SIP
> > >>Use sip-implementors@cs.columbia.edu for questions on current sip
> > >>Use sip@ietf.org for new developments of core SIP
> > >
> > >
> > >
> > >
> > >
> > >_______________________________________________
> > >Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > >This list is for NEW development of the application of SIP
> > >Use sip-implementors@cs.columbia.edu for questions on current sip
> > >Use sip@ietf.org for new developments of core SIP
> >
> >
> > cheers,
> > James
> >
> >                                 *******************
> >                  Truth is not to be argued... it is to be presented
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


cheers,
James

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


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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CC: David Oran <oran@cisco.com>, sipping@ietf.org
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
References: <4.3.2.7.2.20040210153149.02512af8@localhost> <4.3.2.7.2.20040210153149.02512af8@localhost> <4.3.2.7.2.20040210170242.041b63d8@localhost>
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> This was not articulated, if intended

I added

This convention is not meant to
encourage end users to select emergency services where a uniform ECC
address for all such services exist. Rather, these identifiers reflect
current practice in jurisdictions that already have different
numbers for the different emergency services.

Hope that helps.




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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
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This draft is proposed as part of how emergency calls will be routed.
It explains how you find the appropriate Emergency Response Center, and how
you convert geo to civil for dispatch.  Probably, this will be pursued
in DNSEXT, but comments on sipping are appropriate with respect to how
the draft solves the problem.


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


	Title		: Emergency Call Information in the Domain Name
System
	Author(s)	: B. Rosen
	Filename	: draft-rosen-dns-sos-00.txt
	Pages		: 20
	Date		: 2004-2-10
	
Location of a caller is essential to processing an emergency call.
Location is needed to correctly route the call, and to correctly
dispatch help to the right place.  Location can be specified in
geographic (lat/lon/altitude) or civil (country/provin ce/city/
street/floor/room) forms.  Determining which Emergency Response
Center (ERC) should receive the call requires comparing the callers
location to a service area boundary description, which may require
location in geo form.  Dispatching a call generally requires a civil
location.  There is a need for a way to translate a civil to a geo or
vice versa to the resolution of a room.  There is also a need to
publish the service area boundary of the ERC and the emergency
responders.  Further, there is a need for a database of legal civil
addresses (sometimes called a Master Street Address Guide) that civil
locations may be verified against to assure correctness, and
uniformity of naming so that dispatch is accurate.  The memo proposes 
a new DDDS application and a new POLY RR to assist emergency calls.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosen-dns-sos-00.txt

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Date: Wed, 11 Feb 2004 10:45:29 -0500
To: sipping@ietf.org
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
>         sos.fire                fire brigade
>         sos.rescue      ambulance (rescue)
>         sos.marine      marine guard
>         sos.police      police (law enforcement)
>         sos.mountain    mountain rescue

The second sub-address seems vague to me.  Rescue could mean many 
things.  Perhaps, I need to be rescued from a fashion emergency. :)  Would 
something like:

         sos.medic        medical (doctor, nurse, paramedic team)

be more descriptive?

Also, with respect to registrations, I could see commercial interests 
wanting to get into the act, such as an sos.fashion that I hinted at 
above.  Is there some guidance necessary to IANA?  Do we expect that the 
IETF WGs will act as arbiters for this?

Second, wrt location information reporting, is there any need for a 
convention for providing a URL to which a receiving ECC can SUBSCRIBE to 
get location updates from either the UA or a network element that is doing 
triangulation.  I am wondering if something like the following can be 
inserted into the emergency INVITE:

Call-Info: <sips:triangulator@local-proxy.domain.com> ;purpose=location

Mike


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



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Subject: [Sipping] Other concerns within sos ID
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In a message dated 2/10/2004 5:18:03 PM Eastern Standard Time, oran@cisco.com 
writes:


> --On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk" 
> <jmpolk@cisco.com> wrote:
> 
> > I have an concern within the new sos ID, and that concern is in section
> > 4.1 - which reserves the following additional user addresses:
> >
> >     sos.fire    fire brigade
> >     sos.rescue    ambulance (rescue)
> >     sos.marine    marine guard
> >     sos.police    police (law enforcement)
> >     sos.mountain     mountain rescue
> >
> > as I have little confidence that Joe Citizen will be generally capable to
> > choose the correct one given their emergency experience. If these
> > addresses are intended for between two ERCs (for a call transfer, for
> > example) - I don't see the text stating that here.
> >
> I thought this was just to reflect current practice in places that already 
> have different numbers for the different emergency services. Is your 
> concern that the *current* numbers are understood by joe citizen but these 
> are not? If so, the UA can just do the mapping as it has to do already for 
> 911->sos@domain.
> 



So, where in the world are different numbers used to access these different 
emergency services? Not in the US where we use English. If they are used in 
some other country, then why aren't the "reserved" user addresses in the language 
of that country? Joe Citizen may be able to choose the correct one, but I 
don't think Heinrich Staatsangehoeriger or Pierre citoyen should be expected to 
choose the correct English word. We have to stop relying on English for 
everything.

My major concern with this whole plan is the presumption that the user's 
terminal would be able to figure out to use the SOS URI as well as the sub-address 
based on the number the person dials. There are fundamental, unanswered 
questions regarding if and how the terminal would know the local dialing plan.

Further, I have a serious concern with the desciption in 4.2 that a terminal 
must recognize many different digits strings and generate an SOS, independent 
of where it is located, if it "has no knowledge of local emergency numbers". 
This would result in many invalid calls to an emergency center, since many of 
these numbers could be valid (non-emergency) numbers (see note about 118 in 
Finland). Again, this relates to the question of how the terminal knows the local 
dial plan. Unless a method can be defined so that a terminal generates an SOS 
only for the specific emergency number or numbers valid for the place it is 
physically located (or for the place that the user thinks they are in, that is, 
their virtual location), this entire scheme is faulty and can not be used. 
Section 6 covers the issue of Identifying the Local Emergency Number. There 
should be a statement that some method of doing this identification is required, 
or else the terminal can not generate an SOS based on dialed numbers. (Note the 
statement in Section 9 about being able to "clearly and unambiguously 
identity emergency calls".)

The 3rd requirement (in Section 3) is incorrect. When someone is travelling, 
and getting service in a foreign country, there is no requirement that they be 
able to dial their "familiar" home emergency identifier (presuming this 
refers to the emergency "number"). This is contradictory in many places. As noted, 
118 is directory assistance in Finland and a "familiar" emergency number 
elsewhere. This case is proof that this requirement is impossible to meet.

On the other hand, if a person is travelling and getting service from their 
proxy back at the company office, they can't expect to get any emergency 
service from that proxy.

The 4th requirement is misleading. It is not acceptable to have "restricted 
functionality" regarding access to emergency services due to user agents not 
being aware of this document. There is clearly "restricted functionality" 
concerning generation of the SOS URI, but that does not need to be stated. The final 
sentence should state: "User agents conforming to the SIP specification 
[RFC3261], but unaware of this document, must be able to place emergency calls 
without any restriction using entry of normal emergency numbers, although they 
will not generate the SOS URI."

The following requirement is missing: "It MUST be possible for a user of a 
SIP device to dial a local emergency number in the normal manner and properly 
reach the emergency service."

The statement in Section 5 that an "outbound proxy server MUST recognize all 
local emergency numbers ..." is incorrect (and impossible). (Possibly this 
sentence was meant to say " ... all local emergency numbers listed in Section 
4.2...", but this is still impossible). An outbound proxy only needs to recognize 
those emergency numbers applicable to the local areas for which it is 
providing service. An outbound proxy located in and providing service for callers in 
the US certainly does not have to recognize the emergency number used in some 
country in Africa. A visitor from that country must dial 911 for an emergency 
if they are in the US and receiving service from a service provider in the US. 
If the person from Africa is travelling in Asia and uses the proxy in the US 
for service, they can't expect it to correctly handle emergency service. The 
proxy could not possibly distinguish between the emergency numbers for Asia, 
Africa, US, and all the other local dialing plans that the user might know 
about. This is an application that must be limited to entry of E.164 numbers (which 
does not include any emergency numbers or local dialing plans).

Further, in Section 5, it is unclear how to interpret the phrase "as well as 
the tel URIs enumerated in Section Section 4.2" (as a requirement). Section 
4.2 listed two tel URIs (911 and 112) as examples ("such as"), not as an 
exhaustive list. Did this intend to say "as well as tel URIs formed from all the 
emergency numbers enumerated in the first paragraph of Section 4.2"?

The Security Considerations in Section 9 has several problems. Emergency 
calls are not more sensitive than normal calls, so do not have a greater need for 
confidentiality. If "sips" itself may "cause the call to fail", then it can 
not be used.

The requirement for the user agent to "reveal a different amount of 
information to the callee when making an emergency call" also applies to those calls 
dialed with the traditional emergency number when the user agent does not know 
the dial plan and therefore can not "clearly and unambiguously identity 
emergency calls". This function of revealing information must be possible without the 
SOS mechanism. If based on the SOS capability, then this makes it even more 
important for the user agent to correctly identify emergency calls (and not 
mistakenly trigger off of numbers assigned as emergency in other countries).

Conclusion:

The draft needs to make it clear how the SOS uri can be used in specific 3 
cases:

1. User dials the traditional emergency number, the terminal recognizes it, 
and send the SOS URI.
2. User enter the SOS uri directly (types on a keyboard).
3. User presses a button or icon that causes the appropriate SOS URI to be 
sent.

and cannot be used in the following case:

4. User dials the traditional emergency number, the terminal does not 
recognize it, and does not send the SOS URI.

Case 1 can only occur if the terminal knows the "local" dial plan. ("Local" 
is with respect to where the user perceives they are getting service from.)
Case 2 requires support for the local language.
Case 3 is a machine-to-machine communication and does not require 
recognizable words. It can use the reserved English words since no foreign speaker sees 
them.
Case 4 is used when the terminal does not know the local dial plan. It 
depends on a proxy to interpret the number, just as for any other dialed number.

Mike Pierce
Artel



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Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/10/2004 5:18:03 PM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">--On Tuesday, February 10,=20=
2004 3:55 PM -0600 "James M. Polk"=20
<BR>&lt;jmpolk@cisco.com&gt; wrote:
<BR>
<BR>&gt; I have an concern within the new sos ID, and that concern is in sec=
tion
<BR>&gt; 4.1 - which reserves the following additional user addresses:
<BR>&gt;
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.fire &nbsp;&nbsp;&nbsp;fire brigade
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.rescue &nbsp;&nbsp;&nbsp;ambulance (res=
cue)
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.marine &nbsp;&nbsp;&nbsp;marine guard
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.police &nbsp;&nbsp;&nbsp;police (law en=
forcement)
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.mountain &nbsp;&nbsp;&nbsp;&nbsp;mounta=
in rescue
<BR>&gt;
<BR>&gt; as I have little confidence that Joe Citizen will be generally capa=
ble to
<BR>&gt; choose the correct one given their emergency experience. If these
<BR>&gt; addresses are intended for between two ERCs (for a call transfer, f=
or
<BR>&gt; example) - I don't see the text stating that here.
<BR>&gt;
<BR>I thought this was just to reflect current practice in places that alrea=
dy=20
<BR>have different numbers for the different emergency services. Is your=20
<BR>concern that the *current* numbers are understood by joe citizen but the=
se=20
<BR>are not? If so, the UA can just do the mapping as it has to do already f=
or=20
<BR>911-&gt;sos@domain.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>So, where in the world are different numbers used to access these differ=
ent emergency services? Not in the US where we use English. If they are used=
 in some other country, then why aren't the "reserved" user addresses in the=
 language of that country? Joe Citizen may be able to choose the correct one=
, but I don't think Heinrich Staatsangehoeriger or Pierre citoyen should be=20=
expected to choose the correct English word. We have to stop relying on Engl=
ish for everything.
<BR>
<BR>My major concern with this whole plan is the presumption that the user's=
 terminal would be able to figure out to use the SOS URI as well as the sub-=
address based on the number the person dials. There are fundamental, unanswe=
red questions regarding if and how the terminal would know the local dialing=
 plan.
<BR>
<BR>Further, I have a serious concern with the desciption in 4.2 that a term=
inal must recognize many different digits strings and generate an SOS, indep=
endent of where it is located, if it "has no knowledge of local emergency nu=
mbers". This would result in many invalid calls to an emergency center, sinc=
e many of these numbers could be valid (non-emergency) numbers (see note abo=
ut 118 in Finland). Again, this relates to the question of how the terminal=20=
knows the local dial plan. Unless a method can be defined so that a terminal=
 generates an SOS only for the specific emergency number or numbers valid fo=
r the place it is physically located (or for the place that the user thinks=20=
they are in, that is, their virtual location), this entire scheme is faulty=20=
and can not be used. Section 6 covers the issue of Identifying the Local Eme=
rgency Number. There should be a statement that some method of doing this id=
entification is required, or else the terminal can not generate an SOS based=
 on dialed numbers. (Note the statement in Section 9 about being able to "cl=
early and unambiguously identity emergency calls".)
<BR>
<BR>The 3rd requirement (in Section 3) is incorrect. When someone is travell=
ing, and getting service in a foreign country, there is no requirement that=20=
they be able to dial their "familiar" home emergency identifier (presuming t=
his refers to the emergency "number"). This is contradictory in many places.=
 As noted, 118 is directory assistance in Finland and a "familiar" emergency=
 number elsewhere. This case is proof that this requirement is impossible to=
 meet.
<BR>
<BR>On the other hand, if a person is travelling and getting service from th=
eir proxy back at the company office, they can't expect to get any emergency=
 service from that proxy.
<BR>
<BR>The 4th requirement is misleading. It is not acceptable to have "restric=
ted functionality" regarding access to emergency services due to user agents=
 not being aware of this document. There is clearly "restricted functionalit=
y" concerning generation of the SOS URI, but that does not need to be stated=
. The final sentence should state: "User agents conforming to the SIP specif=
ication [RFC3261], but unaware of this document, must be able to place emerg=
ency calls without any restriction using entry of normal emergency numbers,=20=
although they will not generate the SOS URI."
<BR>
<BR>The following requirement is missing: "It MUST be possible for a user of=
 a SIP device to dial a local emergency number in the normal manner and prop=
erly reach the emergency service."
<BR>
<BR>The statement in Section 5 that an "outbound proxy server MUST recognize=
 all local emergency numbers ..." is incorrect (and impossible). (Possibly t=
his sentence was meant to say " ... all local emergency numbers listed in Se=
ction 4.2...", but this is still impossible). An outbound proxy only needs t=
o recognize those emergency numbers applicable to the local areas for which=20=
it is providing service. An outbound proxy located in and providing service=20=
for callers in the US certainly does not have to recognize the emergency num=
ber used in some country in Africa. A visitor from that country must dial 91=
1 for an emergency if they are in the US and receiving service from a servic=
e provider in the US. If the person from Africa is travelling in Asia and us=
es the proxy in the US for service, they can't expect it to correctly handle=
 emergency service. The proxy could not possibly distinguish between the eme=
rgency numbers for Asia, Africa, US, and all the other local dialing plans t=
hat the user might know about. This is an application that must be limited t=
o entry of E.164 numbers (which does not include any emergency numbers or lo=
cal dialing plans).
<BR>
<BR>Further, in Section 5, it is unclear how to interpret the phrase "as wel=
l as the tel URIs enumerated in Section Section 4.2" (as a requirement). Sec=
tion 4.2 listed two tel URIs (911 and 112) as examples ("such as"), not as a=
n exhaustive list. Did this intend to say "as well as tel URIs formed from a=
ll the emergency numbers enumerated in the first paragraph of Section 4.2"?
<BR>
<BR>The Security Considerations in Section 9 has several problems. Emergency=
 calls are not more sensitive than normal calls, so do not have a greater ne=
ed for confidentiality. If "sips" itself may "cause the call to fail", then=20=
it can not be used.
<BR>
<BR>The requirement for the user agent to "reveal a different amount of info=
rmation to the callee when making an emergency call" also applies to those c=
alls dialed with the traditional emergency number when the user agent does n=
ot know the dial plan and therefore can not "clearly and unambiguously ident=
ity emergency calls". This function of revealing information must be possibl=
e without the SOS mechanism. If based on the SOS capability, then this makes=
 it even more important for the user agent to correctly identify emergency c=
alls (and not mistakenly trigger off of numbers assigned as emergency in oth=
er countries).
<BR>
<BR>Conclusion:
<BR>
<BR>The draft needs to make it clear how the SOS uri can be used in specific=
 3 cases:
<BR>
<BR>1. User dials the traditional emergency number, the terminal recognizes=20=
it, and send the SOS URI.
<BR>2. User enter the SOS uri directly (types on a keyboard).
<BR>3. User presses a button or icon that causes the appropriate SOS URI to=20=
be sent.
<BR>
<BR>and cannot be used in the following case:
<BR>
<BR>4. User dials the traditional emergency number, the terminal does not re=
cognize it, and does not send the SOS URI.
<BR>
<BR>Case 1 can only occur if the terminal knows the "local" dial plan. ("Loc=
al" is with respect to where the user perceives they are getting service fro=
m.)
<BR>Case 2 requires support for the local language.
<BR>Case 3 is a machine-to-machine communication and does not require recogn=
izable words. It can use the reserved English words since no foreign speaker=
 sees them.
<BR>Case 4 is used when the terminal does not know the local dial plan. It d=
epends on a proxy to interpret the number, just as for any other dialed numb=
er.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR>
<BR></FONT></HTML>

--part1_1e7.19440e19.2d5baf3c_boundary--

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



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Date: Wed, 11 Feb 2004 11:13:55 EST
Subject: [Sipping] Other concerns within sos ID
To: oran@cisco.com, jmpolk@cisco.com, sipping@ietf.org
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In a message dated 2/10/2004 5:18:03 PM Eastern Standard Time, oran@cisco.com 
writes:


> --On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk" 
> <jmpolk@cisco.com> wrote:
> 
> > I have an concern within the new sos ID, and that concern is in section
> > 4.1 - which reserves the following additional user addresses:
> >
> >     sos.fire    fire brigade
> >     sos.rescue    ambulance (rescue)
> >     sos.marine    marine guard
> >     sos.police    police (law enforcement)
> >     sos.mountain     mountain rescue
> >
> > as I have little confidence that Joe Citizen will be generally capable to
> > choose the correct one given their emergency experience. If these
> > addresses are intended for between two ERCs (for a call transfer, for
> > example) - I don't see the text stating that here.
> >
> I thought this was just to reflect current practice in places that already 
> have different numbers for the different emergency services. Is your 
> concern that the *current* numbers are understood by joe citizen but these 
> are not? If so, the UA can just do the mapping as it has to do already for 
> 911->sos@domain.
> 



So, where in the world are different numbers used to access these different 
emergency services? Not in the US where we use English. If they are used in 
some other country, then why aren't the "reserved" user addresses in the language 
of that country? Joe Citizen may be able to choose the correct one, but I 
don't think Heinrich Staatsangehoeriger or Pierre citoyen should be expected to 
choose the correct English word. We have to stop relying on English for 
everything.

My major concern with this whole plan is the presumption that the user's 
terminal would be able to figure out to use the SOS URI as well as the sub-address 
based on the number the person dials. There are fundamental, unanswered 
questions regarding if and how the terminal would know the local dialing plan.

Further, I have a serious concern with the desciption in 4.2 that a terminal 
must recognize many different digits strings and generate an SOS, independent 
of where it is located, if it "has no knowledge of local emergency numbers". 
This would result in many invalid calls to an emergency center, since many of 
these numbers could be valid (non-emergency) numbers (see note about 118 in 
Finland). Again, this relates to the question of how the terminal knows the local 
dial plan. Unless a method can be defined so that a terminal generates an SOS 
only for the specific emergency number or numbers valid for the place it is 
physically located (or for the place that the user thinks they are in, that is, 
their virtual location), this entire scheme is faulty and can not be used. 
Section 6 covers the issue of Identifying the Local Emergency Number. There 
should be a statement that some method of doing this identification is required, 
or else the terminal can not generate an SOS based on dialed numbers. (Note the 
statement in Section 9 about being able to "clearly and unambiguously 
identity emergency calls".)

The 3rd requirement (in Section 3) is incorrect. When someone is travelling, 
and getting service in a foreign country, there is no requirement that they be 
able to dial their "familiar" home emergency identifier (presuming this 
refers to the emergency "number"). This is contradictory in many places. As noted, 
118 is directory assistance in Finland and a "familiar" emergency number 
elsewhere. This case is proof that this requirement is impossible to meet.

On the other hand, if a person is travelling and getting service from their 
proxy back at the company office, they can't expect to get any emergency 
service from that proxy.

The 4th requirement is misleading. It is not acceptable to have "restricted 
functionality" regarding access to emergency services due to user agents not 
being aware of this document. There is clearly "restricted functionality" 
concerning generation of the SOS URI, but that does not need to be stated. The final 
sentence should state: "User agents conforming to the SIP specification 
[RFC3261], but unaware of this document, must be able to place emergency calls 
without any restriction using entry of normal emergency numbers, although they 
will not generate the SOS URI."

The following requirement is missing: "It MUST be possible for a user of a 
SIP device to dial a local emergency number in the normal manner and properly 
reach the emergency service."

The statement in Section 5 that an "outbound proxy server MUST recognize all 
local emergency numbers ..." is incorrect (and impossible). (Possibly this 
sentence was meant to say " ... all local emergency numbers listed in Section 
4.2...", but this is still impossible). An outbound proxy only needs to recognize 
those emergency numbers applicable to the local areas for which it is 
providing service. An outbound proxy located in and providing service for callers in 
the US certainly does not have to recognize the emergency number used in some 
country in Africa. A visitor from that country must dial 911 for an emergency 
if they are in the US and receiving service from a service provider in the US. 
If the person from Africa is travelling in Asia and uses the proxy in the US 
for service, they can't expect it to correctly handle emergency service. The 
proxy could not possibly distinguish between the emergency numbers for Asia, 
Africa, US, and all the other local dialing plans that the user might know 
about. This is an application that must be limited to entry of E.164 numbers (which 
does not include any emergency numbers or local dialing plans).

Further, in Section 5, it is unclear how to interpret the phrase "as well as 
the tel URIs enumerated in Section Section 4.2" (as a requirement). Section 
4.2 listed two tel URIs (911 and 112) as examples ("such as"), not as an 
exhaustive list. Did this intend to say "as well as tel URIs formed from all the 
emergency numbers enumerated in the first paragraph of Section 4.2"?

The Security Considerations in Section 9 has several problems. Emergency 
calls are not more sensitive than normal calls, so do not have a greater need for 
confidentiality. If "sips" itself may "cause the call to fail", then it can 
not be used.

The requirement for the user agent to "reveal a different amount of 
information to the callee when making an emergency call" also applies to those calls 
dialed with the traditional emergency number when the user agent does not know 
the dial plan and therefore can not "clearly and unambiguously identity 
emergency calls". This function of revealing information must be possible without the 
SOS mechanism. If based on the SOS capability, then this makes it even more 
important for the user agent to correctly identify emergency calls (and not 
mistakenly trigger off of numbers assigned as emergency in other countries).

Conclusion:

The draft needs to make it clear how the SOS uri can be used in specific 3 
cases:

1. User dials the traditional emergency number, the terminal recognizes it, 
and send the SOS URI.
2. User enter the SOS uri directly (types on a keyboard).
3. User presses a button or icon that causes the appropriate SOS URI to be 
sent.

and cannot be used in the following case:

4. User dials the traditional emergency number, the terminal does not 
recognize it, and does not send the SOS URI.

Case 1 can only occur if the terminal knows the "local" dial plan. ("Local" 
is with respect to where the user perceives they are getting service from.)
Case 2 requires support for the local language.
Case 3 is a machine-to-machine communication and does not require 
recognizable words. It can use the reserved English words since no foreign speaker sees 
them.
Case 4 is used when the terminal does not know the local dial plan. It 
depends on a proxy to interpret the number, just as for any other dialed number.

Mike Pierce
Artel



--part1_4f.3a2f7b25.2d5baec3_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/10/2004 5:18:03 PM Eastern Standard Time, oran@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">--On Tuesday, February 10,=20=
2004 3:55 PM -0600 "James M. Polk"=20
<BR>&lt;jmpolk@cisco.com&gt; wrote:
<BR>
<BR>&gt; I have an concern within the new sos ID, and that concern is in sec=
tion
<BR>&gt; 4.1 - which reserves the following additional user addresses:
<BR>&gt;
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.fire &nbsp;&nbsp;&nbsp;fire brigade
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.rescue &nbsp;&nbsp;&nbsp;ambulance (res=
cue)
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.marine &nbsp;&nbsp;&nbsp;marine guard
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.police &nbsp;&nbsp;&nbsp;police (law en=
forcement)
<BR>&gt; &nbsp;&nbsp;&nbsp;&nbsp;sos.mountain &nbsp;&nbsp;&nbsp;&nbsp;mounta=
in rescue
<BR>&gt;
<BR>&gt; as I have little confidence that Joe Citizen will be generally capa=
ble to
<BR>&gt; choose the correct one given their emergency experience. If these
<BR>&gt; addresses are intended for between two ERCs (for a call transfer, f=
or
<BR>&gt; example) - I don't see the text stating that here.
<BR>&gt;
<BR>I thought this was just to reflect current practice in places that alrea=
dy=20
<BR>have different numbers for the different emergency services. Is your=20
<BR>concern that the *current* numbers are understood by joe citizen but the=
se=20
<BR>are not? If so, the UA can just do the mapping as it has to do already f=
or=20
<BR>911-&gt;sos@domain.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>So, where in the world are different numbers used to access these differ=
ent emergency services? Not in the US where we use English. If they are used=
 in some other country, then why aren't the "reserved" user addresses in the=
 language of that country? Joe Citizen may be able to choose the correct one=
, but I don't think Heinrich Staatsangehoeriger or Pierre citoyen should be=20=
expected to choose the correct English word. We have to stop relying on Engl=
ish for everything.
<BR>
<BR>My major concern with this whole plan is the presumption that the user's=
 terminal would be able to figure out to use the SOS URI as well as the sub-=
address based on the number the person dials. There are fundamental, unanswe=
red questions regarding if and how the terminal would know the local dialing=
 plan.
<BR>
<BR>Further, I have a serious concern with the desciption in 4.2 that a term=
inal must recognize many different digits strings and generate an SOS, indep=
endent of where it is located, if it "has no knowledge of local emergency nu=
mbers". This would result in many invalid calls to an emergency center, sinc=
e many of these numbers could be valid (non-emergency) numbers (see note abo=
ut 118 in Finland). Again, this relates to the question of how the terminal=20=
knows the local dial plan. Unless a method can be defined so that a terminal=
 generates an SOS only for the specific emergency number or numbers valid fo=
r the place it is physically located (or for the place that the user thinks=20=
they are in, that is, their virtual location), this entire scheme is faulty=20=
and can not be used. Section 6 covers the issue of Identifying the Local Eme=
rgency Number. There should be a statement that some method of doing this id=
entification is required, or else the terminal can not generate an SOS based=
 on dialed numbers. (Note the statement in Section 9 about being able to "cl=
early and unambiguously identity emergency calls".)
<BR>
<BR>The 3rd requirement (in Section 3) is incorrect. When someone is travell=
ing, and getting service in a foreign country, there is no requirement that=20=
they be able to dial their "familiar" home emergency identifier (presuming t=
his refers to the emergency "number"). This is contradictory in many places.=
 As noted, 118 is directory assistance in Finland and a "familiar" emergency=
 number elsewhere. This case is proof that this requirement is impossible to=
 meet.
<BR>
<BR>On the other hand, if a person is travelling and getting service from th=
eir proxy back at the company office, they can't expect to get any emergency=
 service from that proxy.
<BR>
<BR>The 4th requirement is misleading. It is not acceptable to have "restric=
ted functionality" regarding access to emergency services due to user agents=
 not being aware of this document. There is clearly "restricted functionalit=
y" concerning generation of the SOS URI, but that does not need to be stated=
. The final sentence should state: "User agents conforming to the SIP specif=
ication [RFC3261], but unaware of this document, must be able to place emerg=
ency calls without any restriction using entry of normal emergency numbers,=20=
although they will not generate the SOS URI."
<BR>
<BR>The following requirement is missing: "It MUST be possible for a user of=
 a SIP device to dial a local emergency number in the normal manner and prop=
erly reach the emergency service."
<BR>
<BR>The statement in Section 5 that an "outbound proxy server MUST recognize=
 all local emergency numbers ..." is incorrect (and impossible). (Possibly t=
his sentence was meant to say " ... all local emergency numbers listed in Se=
ction 4.2...", but this is still impossible). An outbound proxy only needs t=
o recognize those emergency numbers applicable to the local areas for which=20=
it is providing service. An outbound proxy located in and providing service=20=
for callers in the US certainly does not have to recognize the emergency num=
ber used in some country in Africa. A visitor from that country must dial 91=
1 for an emergency if they are in the US and receiving service from a servic=
e provider in the US. If the person from Africa is travelling in Asia and us=
es the proxy in the US for service, they can't expect it to correctly handle=
 emergency service. The proxy could not possibly distinguish between the eme=
rgency numbers for Asia, Africa, US, and all the other local dialing plans t=
hat the user might know about. This is an application that must be limited t=
o entry of E.164 numbers (which does not include any emergency numbers or lo=
cal dialing plans).
<BR>
<BR>Further, in Section 5, it is unclear how to interpret the phrase "as wel=
l as the tel URIs enumerated in Section Section 4.2" (as a requirement). Sec=
tion 4.2 listed two tel URIs (911 and 112) as examples ("such as"), not as a=
n exhaustive list. Did this intend to say "as well as tel URIs formed from a=
ll the emergency numbers enumerated in the first paragraph of Section 4.2"?
<BR>
<BR>The Security Considerations in Section 9 has several problems. Emergency=
 calls are not more sensitive than normal calls, so do not have a greater ne=
ed for confidentiality. If "sips" itself may "cause the call to fail", then=20=
it can not be used.
<BR>
<BR>The requirement for the user agent to "reveal a different amount of info=
rmation to the callee when making an emergency call" also applies to those c=
alls dialed with the traditional emergency number when the user agent does n=
ot know the dial plan and therefore can not "clearly and unambiguously ident=
ity emergency calls". This function of revealing information must be possibl=
e without the SOS mechanism. If based on the SOS capability, then this makes=
 it even more important for the user agent to correctly identify emergency c=
alls (and not mistakenly trigger off of numbers assigned as emergency in oth=
er countries).
<BR>
<BR>Conclusion:
<BR>
<BR>The draft needs to make it clear how the SOS uri can be used in specific=
 3 cases:
<BR>
<BR>1. User dials the traditional emergency number, the terminal recognizes=20=
it, and send the SOS URI.
<BR>2. User enter the SOS uri directly (types on a keyboard).
<BR>3. User presses a button or icon that causes the appropriate SOS URI to=20=
be sent.
<BR>
<BR>and cannot be used in the following case:
<BR>
<BR>4. User dials the traditional emergency number, the terminal does not re=
cognize it, and does not send the SOS URI.
<BR>
<BR>Case 1 can only occur if the terminal knows the "local" dial plan. ("Loc=
al" is with respect to where the user perceives they are getting service fro=
m.)
<BR>Case 2 requires support for the local language.
<BR>Case 3 is a machine-to-machine communication and does not require recogn=
izable words. It can use the reserved English words since no foreign speaker=
 sees them.
<BR>Case 4 is used when the terminal does not know the local dial plan. It d=
epends on a proxy to interpret the number, just as for any other dialed numb=
er.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR>
<BR></FONT></HTML>

--part1_4f.3a2f7b25.2d5baec3_boundary--

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



From exim@www1.ietf.org  Wed Feb 11 11:29:51 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Michael Hammer'" <mhammer@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Wed, 11 Feb 2004 11:08:43 -0500
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We can't get into semantic arguments about what the best word choice is;
many of the users won't understand the subtleties, and we expect that in
most cases the phone (or a proxy) will have a locally meaningful way to
invoke the call.

I do understand the concern for explosion of names in the registry.
The "real" criteria is whether an ERC dispatches (directly or indirectly)
such an agency.  More wording needed, for sure.  An RFC to add one is
a pretty high burden.

The doc discusses the issues on how to update location, and subscribe
is one of them.  I think we first have to agree on whether proxies can
insert location, which influences the kind of mechanism you are suggesting.
There are privacy issues associated with that.  I personally think we're
going to have to do it, but at the moment, the agreement with geopriv is
that it's UA only.

Brian

> -----Original Message-----
> From: Michael Hammer [mailto:mhammer@cisco.com]
> Sent: Wednesday, February 11, 2004 10:45 AM
> To: sipping@ietf.org
> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
> >         sos.fire                fire brigade
> >         sos.rescue      ambulance (rescue)
> >         sos.marine      marine guard
> >         sos.police      police (law enforcement)
> >         sos.mountain    mountain rescue
> 
> The second sub-address seems vague to me.  Rescue could mean many 
> things.  Perhaps, I need to be rescued from a fashion 
> emergency. :)  Would 
> something like:
> 
>          sos.medic        medical (doctor, nurse, paramedic team)
> 
> be more descriptive?
> 
> Also, with respect to registrations, I could see commercial interests 
> wanting to get into the act, such as an sos.fashion that I hinted at 
> above.  Is there some guidance necessary to IANA?  Do we 
> expect that the 
> IETF WGs will act as arbiters for this?
> 
> Second, wrt location information reporting, is there any need for a 
> convention for providing a URL to which a receiving ECC can 
> SUBSCRIBE to 
> get location updates from either the UA or a network element 
> that is doing 
> triangulation.  I am wondering if something like the following can be 
> inserted into the emergency INVITE:
> 
> Call-Info: <sips:triangulator@local-proxy.domain.com> 
> ;purpose=location
> 
> Mike
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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Date: Wed, 11 Feb 2004 11:52:03 -0500
To: Mpierce1@aol.com
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] Other concerns within sos ID
Cc: oran@cisco.com, jmpolk@cisco.com, sipping@ietf.org
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One additional question that was touched on in this email:

If an sos URI contains a domain name, I presume that a proxy will forward 
the request based on that domain name and never look at the local part 
containing the "sos.xxx".  Is that correct?

If so, does that mean that the proxy in that domain is expected to look at 
the location information and forward to the appropriate ECC proxy?

As for internationalization, that could get out of hand.  Was the intent 
that these URIs are machine readable only?  I suspect that some 
internationally recognized icons on the user agents that translate into 
these URIs will be needed.  But, that could be tricky as well:

sos.fire     picture of flames
sos.rescue   picture of Red Cross/Crescent?
sos.police   picture of badge/police belt?
sos.marine   picture of boat + cross/crescent?
sos.mountain picture of mountain + cross/crescent?

But those icons are phone implementation details.

Mike


At 11:13 AM 2/11/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 2/10/2004 5:18:03 PM Eastern Standard Time, 
>oran@cisco.com writes:
>
>
>>--On Tuesday, February 10, 2004 3:55 PM -0600 "James M. Polk"
>><jmpolk@cisco.com> wrote:
>>
>> > I have an concern within the new sos ID, and that concern is in section
>> > 4.1 - which reserves the following additional user addresses:
>> >
>> >     sos.fire    fire brigade
>> >     sos.rescue    ambulance (rescue)
>> >     sos.marine    marine guard
>> >     sos.police    police (law enforcement)
>> >     sos.mountain     mountain rescue
>> >
>> > as I have little confidence that Joe Citizen will be generally capable to
>> > choose the correct one given their emergency experience. If these
>> > addresses are intended for between two ERCs (for a call transfer, for
>> > example) - I don't see the text stating that here.
>> >
>>I thought this was just to reflect current practice in places that already
>>have different numbers for the different emergency services. Is your
>>concern that the *current* numbers are understood by joe citizen but these
>>are not? If so, the UA can just do the mapping as it has to do already for
>>911->sos@domain.
>
>
>
>
>So, where in the world are different numbers used to access these 
>different emergency services? Not in the US where we use English. If they 
>are used in some other country, then why aren't the "reserved" user 
>addresses in the language of that country? Joe Citizen may be able to 
>choose the correct one, but I don't think Heinrich Staatsangehoeriger or 
>Pierre citoyen should be expected to choose the correct English word. We 
>have to stop relying on English for everything.
>
>My major concern with this whole plan is the presumption that the user's 
>terminal would be able to figure out to use the SOS URI as well as the 
>sub-address based on the number the person dials. There are fundamental, 
>unanswered questions regarding if and how the terminal would know the 
>local dialing plan.
>
>Further, I have a serious concern with the desciption in 4.2 that a 
>terminal must recognize many different digits strings and generate an SOS, 
>independent of where it is located, if it "has no knowledge of local 
>emergency numbers". This would result in many invalid calls to an 
>emergency center, since many of these numbers could be valid 
>(non-emergency) numbers (see note about 118 in Finland). Again, this 
>relates to the question of how the terminal knows the local dial plan. 
>Unless a method can be defined so that a terminal generates an SOS only 
>for the specific emergency number or numbers valid for the place it is 
>physically located (or for the place that the user thinks they are in, 
>that is, their virtual location), this entire scheme is faulty and can not 
>be used. Section 6 covers the issue of Identifying the Local Emergency 
>Number. There should be a statement that some method of doing this 
>identification is required, or else the terminal can not generate an SOS 
>based on dialed numbers. (Note the statement in Section 9 about being able 
>to "clearly and unambiguously identity emergency calls".)
>
>The 3rd requirement (in Section 3) is incorrect. When someone is 
>travelling, and getting service in a foreign country, there is no 
>requirement that they be able to dial their "familiar" home emergency 
>identifier (presuming this refers to the emergency "number"). This is 
>contradictory in many places. As noted, 118 is directory assistance in 
>Finland and a "familiar" emergency number elsewhere. This case is proof 
>that this requirement is impossible to meet.
>
>On the other hand, if a person is travelling and getting service from 
>their proxy back at the company office, they can't expect to get any 
>emergency service from that proxy.
>
>The 4th requirement is misleading. It is not acceptable to have 
>"restricted functionality" regarding access to emergency services due to 
>user agents not being aware of this document. There is clearly "restricted 
>functionality" concerning generation of the SOS URI, but that does not 
>need to be stated. The final sentence should state: "User agents 
>conforming to the SIP specification [RFC3261], but unaware of this 
>document, must be able to place emergency calls without any restriction 
>using entry of normal emergency numbers, although they will not generate 
>the SOS URI."
>
>The following requirement is missing: "It MUST be possible for a user of a 
>SIP device to dial a local emergency number in the normal manner and 
>properly reach the emergency service."
>
>The statement in Section 5 that an "outbound proxy server MUST recognize 
>all local emergency numbers ..." is incorrect (and impossible). (Possibly 
>this sentence was meant to say " ... all local emergency numbers listed in 
>Section 4.2...", but this is still impossible). An outbound proxy only 
>needs to recognize those emergency numbers applicable to the local areas 
>for which it is providing service. An outbound proxy located in and 
>providing service for callers in the US certainly does not have to 
>recognize the emergency number used in some country in Africa. A visitor 
>from that country must dial 911 for an emergency if they are in the US and 
>receiving service from a service provider in the US. If the person from 
>Africa is travelling in Asia and uses the proxy in the US for service, 
>they can't expect it to correctly handle emergency service. The proxy 
>could not possibly distinguish between the emergency numbers for Asia, 
>Africa, US, and all the other local dialing plans that the user might know 
>about. This is an application that must be limited to entry of E.164 
>numbers (which does not include any emergency numbers or local dialing plans).
>
>Further, in Section 5, it is unclear how to interpret the phrase "as well 
>as the tel URIs enumerated in Section Section 4.2" (as a requirement). 
>Section 4.2 listed two tel URIs (911 and 112) as examples ("such as"), not 
>as an exhaustive list. Did this intend to say "as well as tel URIs formed 
>from all the emergency numbers enumerated in the first paragraph of 
>Section 4.2"?
>
>The Security Considerations in Section 9 has several problems. Emergency 
>calls are not more sensitive than normal calls, so do not have a greater 
>need for confidentiality. If "sips" itself may "cause the call to fail", 
>then it can not be used.
>
>The requirement for the user agent to "reveal a different amount of 
>information to the callee when making an emergency call" also applies to 
>those calls dialed with the traditional emergency number when the user 
>agent does not know the dial plan and therefore can not "clearly and 
>unambiguously identity emergency calls". This function of revealing 
>information must be possible without the SOS mechanism. If based on the 
>SOS capability, then this makes it even more important for the user agent 
>to correctly identify emergency calls (and not mistakenly trigger off of 
>numbers assigned as emergency in other countries).
>
>Conclusion:
>
>The draft needs to make it clear how the SOS uri can be used in specific 3 
>cases:
>
>1. User dials the traditional emergency number, the terminal recognizes 
>it, and send the SOS URI.
>2. User enter the SOS uri directly (types on a keyboard).
>3. User presses a button or icon that causes the appropriate SOS URI to be 
>sent.
>
>and cannot be used in the following case:
>
>4. User dials the traditional emergency number, the terminal does not 
>recognize it, and does not send the SOS URI.
>
>Case 1 can only occur if the terminal knows the "local" dial plan. 
>("Local" is with respect to where the user perceives they are getting 
>service from.)
>Case 2 requires support for the local language.
>Case 3 is a machine-to-machine communication and does not require 
>recognizable words. It can use the reserved English words since no foreign 
>speaker sees them.
>Case 4 is used when the terminal does not know the local dial plan. It 
>depends on a proxy to interpret the number, just as for any other dialed 
>number.
>
>Mike Pierce
>Artel


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Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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Thread-Topic: [Sipping] Concern within sos ID - user addresses reserved
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From: "Andrew Allen" <aallen@rim.net>
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How about sos.medical then since that outlines the type of emergency. I =
agree that rescue may be confused with the others.

Andrew
--------------------------
Andrew Allen
Manager Standards
Research In Motion Ltd
BlackBerry Mobile  +1 847 809 8636
European Mobile    +358 50 467 5870
http://www.rim.com/

Sent from my BlackBerry Wireless Handheld


-----Original Message-----
From: Michael Hammer <mhammer@cisco.com>
To: sipping@ietf.org <sipping@ietf.org>
Sent: Wed Feb 11 10:45:29 2004
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved

At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
>         sos.fire                fire brigade
>         sos.rescue      ambulance (rescue)
>         sos.marine      marine guard
>         sos.police      police (law enforcement)
>         sos.mountain    mountain rescue

The second sub-address seems vague to me.  Rescue could mean many=20
things.  Perhaps, I need to be rescued from a fashion emergency. :)  =
Would=20
something like:

         sos.medic        medical (doctor, nurse, paramedic team)

be more descriptive?

Also, with respect to registrations, I could see commercial interests=20
wanting to get into the act, such as an sos.fashion that I hinted at=20
above.  Is there some guidance necessary to IANA?  Do we expect that the =

IETF WGs will act as arbiters for this?

Second, wrt location information reporting, is there any need for a=20
convention for providing a URL to which a receiving ECC can SUBSCRIBE to =

get location updates from either the UA or a network element that is =
doing=20
triangulation.  I am wondering if something like the following can be=20
inserted into the emergency INVITE:

Call-Info: <sips:triangulator@local-proxy.domain.com> =
;purpose=3Dlocation

Mike


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, oran@cisco.com, jmpolk@cisco.com,
        sipping@ietf.org
Subject: RE: [Sipping] Other concerns within sos ID
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Thanks for your comments.  While I have some replies, in line, you do 
raise a pretty important issue, which deserves attention.

The fundamental idea that this draft proposes is that emergency calls 
get special attention.  The special attention is that location is included 
on the call.  There are also routing implications, but the inclusion of 
location is the key issue.

The draft proposes that, one way or another, the call is determined to 
be an emergency call, and as a consequence, location is included.  
You point out the problem that if you have some address (phone number), 
which neither the UA or the outbound proxy recognizes as an emergency
number, 
then it will not get special attention.

The issue you raise with local emergency numbers is complicated by VPN 
tunnels.  My usual example is the patron of a cafe in Chicago seeing an 
accident while connected via a hotspot through a VPN tunnel to an employer's

VoIP system in Sierra Leone.  In this example, dialing the Sierra Leone 
emergency number would not be an emergency call, but 9-1-1 would be.  

The answer that works is that the PHONE knows its location, and the PHONE 
knows the local emergency number (or the phone has some other way to 
determine an emergency call).  If that is uniformly true, then it works.  
If that's not true, then the outbound proxy has a problem.  It has to know 
the location of the phone to determine the "local" (to the phone) emergency 
number.  So far, we haven't dealt with how we do that, but there are 
proposals that location is added by the outbound proxy when it can determine

location, but the phone cannot.

If the outbound proxy knows the location of the phone, then it could, 
in theory, determine the local (to the phone) emergency number, and then it 
could do
the right thing.  That seems to work also.

The thing that wouldn't work is the cross cases, where the phone knows its 
location, but cannot recognize local numbers (didn't have a dialing plan), 
or where the phone has dialing plans, but know way to be told the local 
emergency number.  The latter is probably fixable.  The former is more 
difficult.  We either have to fix it, or prohibit it somehow.

I also realized that there is a dialing plan conflict.  In my example, 
the employer in Sierra Leone has a local dialing plan.  If it has a conflict

with the local (to the phone) emergency number, there is a problem.  
That problem exists whether the phone or the proxy interprets the dialing 
plan.  The same problem would exist today for someone calling into an access

number for a PBX which allowed dialing within the PBX dial plan.  
If they were in a different continent, the local=to-the-phone)
emergency number would not work, the local-to-the-pbx emergency number 
would get them to an ERC, but not the right one.

There is no real solution that I can think of to this problem.  
You could always force the local-to-the-phone emergency number to override 
the local-to-the-pbx dial plan, but that is not very satisfactory.  
Better to not use a number to reach emergency services (use some other key 
sequence).

More inline.
...snip...

>So, where in the world are different numbers used to access these 
>different emergency services? Not in the US where we use English. 
>If they are used in some other country, then why aren't the "reserved" 
>user addresses in the language of that country? Joe Citizen may be able 
>to choose the correct one, but I don't think Heinrich Staatsangehoeriger 
>or Pierre citoyen should be expected to choose the correct English word. 
>We have to stop relying on English for everything.
The assumption is that the phone presents an appropriate interface for
initiating an emergency call, and the result is a call to one of these
URLs.  You don't normally type them. 

>My major concern with this whole plan is the presumption that the user's 
>terminal would be able to figure out to use the SOS URI as well as the 
>sub-address based on the number the person dials. There are fundamental, 
>unanswered questions regarding if and how the terminal would know the 
>local dialing plan. 
Covered above, noting that it might be a proxy, and not the phone, that has
the dialing plan

>
>Further, I have a serious concern with the description in 4.2 that a
terminal
>must recognize many different digits strings and generate an SOS,
independent 
>of where it is located, if it "has no knowledge of local emergency
numbers". 
>This would result in many invalid calls to an emergency center, since many 
>of these numbers could be valid (non-emergency) numbers (see note about 118

>in Finland). Again, this relates to the question of how the terminal knows 
>the local dial plan. Unless a method can be defined so that a terminal 
>generates an SOS only for the specific emergency number or numbers valid
for 
>the place it is physically located (or for the place that the user thinks 
>they are in, that is, their virtual location), this entire scheme is faulty

>and can not be used. Section 6 covers the issue of Identifying the Local 
>Emergency Number. There should be a statement that some method of doing
this 
>identification is required, or else the terminal can not generate an SOS 
>based on dialed numbers. (Note the statement in Section 9 about being able 
>to "clearly and unambiguously identity emergency calls".) 
Generally, I think you are right; the only emergency numbers are the ones
local-to-the-phone.  Either the phone, or the proxy has to know where the
phone is, both to determine what the local emergency number is, and to
correctly route.

>The 3rd requirement (in Section 3) is incorrect. When someone is traveling,
>and getting service in a foreign country, there is no requirement that they

>be able to dial their "familiar" home emergency identifier (presuming this 
>refers to the emergency "number"). This is contradictory in many places. 
>As noted, 118 is directory assistance in Finland and a "familiar" emergency

>number elsewhere. This case is proof that this requirement is impossible to

>meet.
Again, generally I agree.  It's a very unfortunate situation, because if you
are a visitor, your awareness of the local emergency number is not very
good,
while you clearly know your home emergency number.  

>On the other hand, if a person is traveling and getting service from their 
>proxy back at the company office, they can't expect to get any emergency 
>service from that proxy. 
No, the architecture is designed to make this work.  The local-at-the-pbx
proxy will find the location of the roaming party and route it to the
correct
ERC

>The 4th requirement is misleading. It is not acceptable to have "restricted

>functionality" regarding access to emergency services due to user agents
not 
>being aware of this document. There is clearly "restricted functionality" 
>concerning generation of the SOS URI, but that does not need to be stated. 
There are phones out there now.  What would you have us do with them?
I think this is a reasonable backwards compatibility specification

>The final sentence should state: "User agents conforming to the SIP 
>specification [RFC3261], but unaware of this document, must be able to
place 
>emergency calls without any restriction using entry of normal emergency 
>numbers, although they will not generate the SOS URI." 
I think this is incorrect.  First of all, a proxy can redirect to sos.
Secondly, you can't get "without restriction" because you don't get
location. 

>The following requirement is missing: "It MUST be possible for a user of a 
>SIP device to dial a local emergency number in the normal manner and
properly 
>reach the emergency service." 
I don't understand what this adds to the requirements.  We have to address
some issues, but if we do, then this should be superfluous.

>The statement in Section 5 that an "outbound proxy server MUST recognize 
>all local emergency numbers ..." is incorrect (and impossible). (Possibly 
>this sentence was meant to say " ... all local emergency numbers listed in 
>Section 4.2...", but this is still impossible). An outbound proxy only
needs 
Rewording is required, I agree, but it has to refer to local-to-the-phone.

>to recognize those emergency numbers applicable to the local areas for
which 
>it is providing service. An outbound proxy located in and providing service

>for callers in the US certainly does not have to recognize the emergency
number 
>used in some country in Africa. A visitor from that country must dial 911
for 
>an emergency if they are in the US and receiving service from a service 
>provider in the US. If the person from Africa is traveling in Asia and uses

>the proxy in the US for service, they can't expect it to correctly handle 
>emergency service. The proxy could not possibly distinguish between the 
>emergency numbers for Asia, Africa, US, and all the other local dialing
plans 
>that the user might know about. This is an application that must be limited
to 
>entry of E.164 numbers (which does not include any emergency numbers or
local 
>dialing plans). 
Not correct, as above, the outbound proxy can provide emergency call routing
even if the caller is roaming


>Further, in Section 5, it is unclear how to interpret the phrase "as well
as 
>the tel URIs enumerated in Section Section 4.2" (as a requirement). Section

>4.2 listed two tel URIs (911 and 112) as examples ("such as"), not as an 
>exhaustive list. Did this intend to say "as well as tel URIs formed from
all 
>the emergency numbers enumerated in the first paragraph of Section 4.2"? 
We can improve the wording here

>The Security Considerations in Section 9 has several problems. Emergency
calls 
>are not more sensitive than normal calls, so do not have a greater need for
>confidentiality. If "sips" itself may "cause the call to fail", then it can

>not be used. 
There are those who don't agree with you about confidentiality.  
I agree that calls can't fail if sips fails.  There is text in the location 
requirements for sip document about this.

>The requirement for the user agent to "reveal a different amount of
information 
>to the callee when making an emergency call" also applies to those calls
dialed 
>with the traditional emergency number when the user agent does not know the

>dial plan and therefore can not "clearly and unambiguously identity
emergency 
>calls". This function of revealing information must be possible without the
SOS 
>mechanism. If based on the SOS capability, then this makes it even more 
>important for the user agent to correctly identify emergency calls (and not

>mistakenly trigger off of numbers assigned as emergency in other
countries). 
I think you propose something impossible.  If the call is recognized as an
emergency call, and routed as such, it will be routed as sos@localdomain,
and have location.  If it is not recognized, it will be treated per the 
local-to-the-pbx dial plan, and will not have location.  While we may want
to consider failure cases (you might reach some ERC, without location,
which is better than nothing), the goal is to not ever have this happen.

>Conclusion: 
>
>The draft needs to make it clear how the SOS uri can be used in specific 3 
>cases: 
>
>1. User dials the traditional emergency number, the terminal recognizes it,

>and send the SOS URI. 
>2. User enter the SOS uri directly (types on a keyboard). 
>3. User presses a button or icon that causes the appropriate SOS URI to be
sent. 
>
>and cannot be used in the following case: 
>
>4. User dials the traditional emergency number, the terminal does not 
>recognize it, and does not send the SOS URI. 
The design is that a proxy can recognize the call as an emergency call
and cause the correct treatment for it.

>Case 1 can only occur if the terminal knows the "local" dial plan. ("Local"

>is with respect to where the user perceives they are getting service from.)
Or the proxy
 
>Case 2 requires support for the local language. 
Agreed.  Text needed.

>Case 3 is a machine-to-machine communication and does not require
recognizable 
>words. It can use the reserved English words since no foreign speaker sees
them. 
Agreed

>Case 4 is used when the terminal does not know the local dial plan. It
depends 
>on a proxy to interpret the number, just as for any other dialed number. 
Disagree, as above


Brian 

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CC: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, oran@cisco.com, jmpolk@cisco.com,
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Subject: Re: [Sipping] Other concerns within sos ID
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Rosen, Brian wrote:

> The draft proposes that, one way or another, the call is determined to 
> be an emergency call, and as a consequence, location is included.  
> You point out the problem that if you have some address (phone number), 
> which neither the UA or the outbound proxy recognizes as an emergency
> number, 
> then it will not get special attention.
> 
> The issue you raise with local emergency numbers is complicated by VPN 
> tunnels.  My usual example is the patron of a cafe in Chicago seeing an 
> accident while connected via a hotspot through a VPN tunnel to an employer's
> 
> VoIP system in Sierra Leone.  In this example, dialing the Sierra Leone 
> emergency number would not be an emergency call, but 9-1-1 would be.  
> 
> The answer that works is that the PHONE knows its location, and the PHONE 
> knows the local emergency number (or the phone has some other way to 
> determine an emergency call).  If that is uniformly true, then it works.  
> If that's not true, then the outbound proxy has a problem.  It has to know 
> the location of the phone to determine the "local" (to the phone) emergency 
> number.  So far, we haven't dealt with how we do that, but there are 
> proposals that location is added by the outbound proxy when it can determine
> 
> location, but the phone cannot.
> 
> If the outbound proxy knows the location of the phone, then it could, 
> in theory, determine the local (to the phone) emergency number, and then it 
> could do
> the right thing.  That seems to work also.
> 
> The thing that wouldn't work is the cross cases, where the phone knows its 
> location, but cannot recognize local numbers (didn't have a dialing plan), 
> or where the phone has dialing plans, but know way to be told the local 
> emergency number.  The latter is probably fixable.  The former is more 
> difficult.  We either have to fix it, or prohibit it somehow.
> 
> I also realized that there is a dialing plan conflict.  In my example, 
> the employer in Sierra Leone has a local dialing plan.  If it has a conflict
> 
> with the local (to the phone) emergency number, there is a problem.  
> That problem exists whether the phone or the proxy interprets the dialing 
> plan.  The same problem would exist today for someone calling into an access
> 
> number for a PBX which allowed dialing within the PBX dial plan.  
> If they were in a different continent, the local=to-the-phone)
> emergency number would not work, the local-to-the-pbx emergency number 
> would get them to an ERC, but not the right one.
> 
> There is no real solution that I can think of to this problem.  
> You could always force the local-to-the-phone emergency number to override 
> the local-to-the-pbx dial plan, but that is not very satisfactory.  
> Better to not use a number to reach emergency services (use some other key 
> sequence).
> 

In general, I think the correct long-term user interface solution is to 
have a hotkey or address book entry *with user confirmation* rather than 
having users trying to remember digit strings. The digit string model 
currently works rather badly for the PSTN - do you know, without looking 
at the hotel phone card insert, whether you need to dial 9-911, 0, 911, 
0-911 or some other combination to get emergency help in a US hotel, let 
alone in some hotel in a foreign country?


> Again, generally I agree.  It's a very unfortunate situation, because if you
> are a visitor, your awareness of the local emergency number is not very
> good,
> while you clearly know your home emergency number.  

This is a trade-off with no provably 'right' solution, so we need to 
make choices. GSM cell phone users expect that 112 works everywhere 
(which it apparently does, even in the US), for example. My hunch is 
that the collisions for traveling users are at best a nuisance (and 
outweighed by the safety advantages), but that's a hunch.


> There are those who don't agree with you about confidentiality.  

And I think there's a reasonable case to be made that, on average, 
calling 911 is probably more security-relevant than calling your dry 
cleaner, for a variety of reasons. The document makes no quantitative 
claim, so I just don't see the harm.

> I agree that calls can't fail if sips fails.  There is text in the location 
> requirements for sip document about this.

Among many of the topics discussed here, this has all been discussed 
before, with the current text trying to reflect the earlier consensus. 
We can't make progress if we re-open the same issues again and again.

We should probably split the discussion into separate issues as the mail 
messages are getting far too long.

Henning

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Subject: [Sipping] IPR notice on KPML filed
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-------- Original Message --------
Subject: IPR Notice
Date: Tue, 10 Feb 2004 18:07:08 -0500 (EST)
From: IESG Secretary <iesg-secretary@ietf.org>
To: rohan@cisco.com, <dean.willis@softarmor.com>, 
<gonzalo.camarillo@ericsson.com>
CC: mankin@psg.com, <ietf-secretariat@ietf.org>, <amyk@foretec.com>

Dear Working Group Chairs:

An IPR Notice has been posted that pertains to a document from the SIPPING
Working Group.  The title of the IPR Notice is "SnowShore's Statement
about IPR Cliamed in draft-ietf-sipping-kpml"
(http://www.ietf.org/ietf/IPR/snowshore-ipr-draft-ietf-sipping-kpml.txt).

The IETF Secretariat



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To: Michael Hammer <mhammer@cisco.com>
CC: Mpierce1@aol.com, oran@cisco.com, jmpolk@cisco.com, sipping@ietf.org
Subject: Re: [Sipping] Other concerns within sos ID
References: <4.3.2.7.2.20040211114231.00b29e68@cia.cisco.com>
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The human factors aspect of this stuff has traditionally been the domain of the 
ITU-T.  I have a feeling they should be brought into the discussion.  I would have 
to check, but the relevant experts used to be located in Study Group 2.

Michael Hammer wrote:

> One additional question that was touched on in this email:
> 
> If an sos URI contains a domain name, I presume that a proxy will 
> forward the request based on that domain name and never look at the 
> local part containing the "sos.xxx".  Is that correct?
> 
> If so, does that mean that the proxy in that domain is expected to look 
> at the location information and forward to the appropriate ECC proxy?
> 
> As for internationalization, that could get out of hand.  Was the intent 
> that these URIs are machine readable only?  I suspect that some 
> internationally recognized icons on the user agents that translate into 
> these URIs will be needed.  But, that could be tricky as well:
> 
> sos.fire     picture of flames
> sos.rescue   picture of Red Cross/Crescent?
> sos.police   picture of badge/police belt?
> sos.marine   picture of boat + cross/crescent?
> sos.mountain picture of mountain + cross/crescent?
> 
> But those icons are phone implementation details.
> 
> Mike
> 
> 
> 

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Subject: RE: [Sipping] TCAP tunneling
Date: Wed, 11 Feb 2004 16:50:26 -0700
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Thread-Topic: [Sipping] TCAP tunneling
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From: "Hearty, John" <John.Hearty@Level3.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sipping@ietf.org>, "Peterson, Jon" <jon.peterson@neustar.biz>
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In looking at the ENUM RFC, it looks like NAPTR records are used, and
they convert E.164 numbers to URIs.  The syntax for the URIs does not
appear to allow for Display Names, which is what I believe needs to be
populated when forwarding SIP messages.  I suppose we could still do TXT
type DNS records for this, but can't find any standards specific to
CNAM.  The alternate idea similar to the yu draft seems to contradict
the statement that SIP isn't a database query protocol.  If anyone is
already doing something like this or has plans I'd love to hear about
it.

John

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, February 02, 2004 11:29 PM
> To: Hearty, John
> Cc: sipping@ietf.org; Peterson, Jon
> Subject: Re: [Sipping] TCAP tunneling
>=20
> SIP isnt a database query protocol, and cnam is just a DB query
service.
> As such, I would think that enum would be an appropriate technique for
> this. Any other DB query mechanism will do as well.
>=20
> Another idea would be to use a redirect service, similar to what has
> been proposed in the past for LNP:
>=20
> http://www.watersprings.org/pub/id/draft-yu-sip-np-02.txt
>=20
> except that a request for a number would be redirected to a Contact
> which contains that number plus the display name, representing the
> calling name.
>=20
> -Jonathan R.
>=20
> Hearty, John wrote:
> > Other than the miller draft below, is there any SIP standard
proposed to
> > carry cnam queries?
> >
> > John Hearty
> > Level3
> >
> >
> >>-----Original Message-----
> >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On
Behalf
> >
> > Of
> >
> >>Arjun Roychowdhury
> >>Sent: Monday, February 02, 2004 7:25 AM
> >>To: Banibrata Dutta
> >>Cc: kagarwal@cedarpointcom.com; ranjit.avasarala@wipro.com;
> >>sipping@ietf.org; sipping-admin@ietf.org
> >>Subject: RE: [Sipping] TCAP tunneling
> >>
> >>Actually, there _is_ an independant draft on how to carry TCAP in
SIP.
> >>http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt
> >>
> >>Its another matter whether this is the right thing to do or not
> >
> > (please
> >
> >>refer to old archives for the debates).
> >>Kaitaki, you may want to read the above draft as well as the
> >
> > discussions
> >
> >>on it to conclude for yourself which is the preferred method in your
> >>deployment scenario.
> >>
> >>regds
> >>arjun
> >>
> >>
> >>--
> >>Arjun Roychowdhury @ Hughes Software Systems
> >>11717 Exploration Lane, Germantown MD 20876
> >>(O): 301 212 7860
> >>VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)
> >>
> >>
> >>
> >>
> >>"Banibrata Dutta" <dutta@india.hp.com>
> >>Sent by: sipping-admin@ietf.org
> >>02/02/2004 12:25 AM
> >>
> >>
> >>        To:     <ranjit.avasarala@wipro.com>,
> >
> > <kagarwal@cedarpointcom.com>,
> >
> >><sipping@ietf.org>
> >>        cc:
> >>        Subject:        RE: [Sipping] TCAP tunneling
> >>
> >>
> >>hi,
> >>
> >>franly, i'd say that, it depends on your motivation/reason to use
SIP
> >>for
> >>tunneling TCAP. of course you can tunnel TCAP, but why you'd want
to,
> >
> > is
> >
> >>a justification you'd have to work out. AFAIK, there's not
ietf-draft
> >>that
> >>says, how.
> >>
> >>you can obviously tunnel TCAP query in an INVITE or INFO for that
> >
> > matter
> >
> >>(ideally in INVITE s.t. a SIP-dialog is also created which may be
used
> >>to
> >>model the TCAP dialog as well, perhaps!). the ASN.1 BER encoded TCAP
> >>messages may then reside within the body of a multi-part (MIME) enc,
> >>INVITE.
> >>
> >>infact 1 reason i see where this may make sense is that you would
> >
> > rather
> >
> >>not have two stacks s.a. SIGTRAN and SIP for signalling transport,
and
> >>use
> >>SIP as the *only* signalling solution, i.e. for the signalling
> >>application
> >>as well as for signalling transport, and support legacy/TCAP apps by
> >>tunneling
> >>them within SIP. but then, you generally have the constraints of
> >>not-enough-
> >>space, (for 2 stacks) only on resource-limited devices, s.a.
> >
> > hand-sets,
> >
> >>embedded devices etc., but i am unable see (yet), how TCAP can be
> >>related to
> >>such devices.
> >>
> >>my 2 cents,
> >>Banibrata Dutta,
> >>  Hewlett-Packard ISO (P) Ltd,
> >>  29, Cunningham Road,
> >>  Bangalore - 560052
> >>  Ph: +91-80-22051796
> >>
> >>
> >>>-----Original Message-----
> >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> >>>On Behalf Of ranjit.avasarala@wipro.com
> >>>Sent: Monday, February 02, 2004 10:20 AM
> >>>To: kagarwal@cedarpointcom.com; sipping@ietf.org
> >>>Subject: RE: [Sipping] TCAP tunneling
> >>>
> >>>
> >>>Hi
> >>>   TCAP is used for Transaction Capabilities i.e database
> >>>related while SIP with ISUP tunneling is used between MGCs.
> >>>So I don't think TCAP can be tunneled in SIP.
> >>>
> >>>Ranjit
> >>>Ph: 8520408/16 extn: 2173
> >>>
> >>>
> >>>
> >>>
> >>>-----Original Message-----
> >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> >>>On Behalf Of Kaitki Agarwal
> >>>Sent: Monday, February 02, 2004 8:28 AM
> >>>To: sipping@ietf.org
> >>>Subject: [Sipping] TCAP tunneling
> >>>
> >>>
> >>>Hi
> >>> Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> >>>
> >>>Thanks
> >>>Kaitki
> >>>Jj~Yzy=3Di~Rz}y
> >>>
> >>>_______________________________________________
> >>>Sipping mailing list
https://www1.ietf.org/mailman/listinfo/sipping
> >>>This list is for NEW development of the application of SIP
> >>>Use sip-implementors@cs.columbia.edu for questions on current
> >>>sip Use sip@ietf.org for new developments of core SIP
> >>>
> >>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >>
> >>
> >>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
>=20
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com

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



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This document describes the INFO usage for IP-surveillance=0A=
   application. SIP is used as a means of incorporating terminals of=0A=
   different vendors. In this draft, mid-control commands, for example,=0A=
   retrieving server images or saving images in=0A=
   the server are defined and delivered as session names of INVITE =
messages.=0A=
   IN addition to that, INFO definitions specific to the=0A=
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   Tilt, Zoom) information is defined for the purpose of=0A=
   standardization of controls. =0A=
 =0A=
   A URL for this draft is =0A=
=0A=
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_______________________________________________
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From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] Other concerns within sos ID
Cc: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, oran@cisco.com, sipping@ietf.org
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At 02:15 PM 2/11/2004 -0500, Henning Schulzrinne wrote:

>In general, I think the correct long-term user interface solution is to 
>have a hotkey or address book entry *with user confirmation* rather than 
>having users trying to remember digit strings.

This is something that the wireless industry used for a while with less 
than desirable affects (too many unintentional calls by hitting the one 
button accidentally). The PSAPs in the US (and I am only speaking for them 
in this comment) are quite opposed right now to this hotkey/button idea 
resurfacing.

>The digit string model currently works rather badly for the PSTN - do you 
>know, without looking at the hotel phone card insert, whether you need to 
>dial 9-911, 0, 911, 0-911 or some other combination to get emergency help 
>in a US hotel,

this is a good point - it changes based on the TDM switch configuration for 
that phone

>let alone in some hotel in a foreign country?
>
>
>>Again, generally I agree.  It's a very unfortunate situation, because if you
>>are a visitor, your awareness of the local emergency number is not very
>>good,
>>while you clearly know your home emergency number.
>
>This is a trade-off with no provably 'right' solution, so we need to make 
>choices. GSM cell phone users expect that 112 works everywhere (which it 
>apparently does, even in the US), for example. My hunch is that the 
>collisions for traveling users are at best a nuisance (and outweighed by 
>the safety advantages), but that's a hunch.

curiosity question, if 112 works in the US, does 911 work anywhere in Europe?


>Henning



cheers,
James

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


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> This is something that the wireless industry used for a while with less 
> than desirable affects (too many unintentional calls by hitting the one 
> button accidentally). The PSAPs in the US (and I am only speaking for 
> them in this comment) are quite opposed right now to this hotkey/button 
> idea resurfacing.

I guess my ** were not bold enough. I'm well aware of the hotkey issues 
and that's why I said "with user confirmation". That would eliminate the 
nuisance calls.

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In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:

The fundamental idea that this draft proposes is that emergency calls 
get special attention.  The special attention is that location is included 
on the call.  There are also routing implications, but the inclusion of 
location is the key issue.

[MAP] The point I'm trying to make is that the requirement to provide 
location for emergency calls has nothing to do with this draft (SOS URI). It must be 
accepted as a requirement and solved even if this SOS URI disappeared today. 
It can not be the SOS URI which enables the location to be provided.


In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:

The issue you raise with local emergency numbers is complicated by VPN 
tunnels.  My usual example is the patron of a cafe in Chicago seeing an 
accident while connected via a hotspot through a VPN tunnel to an employer's
VoIP system in Sierra Leone.  In this example, dialing the Sierra Leone 
emergency number would not be an emergency call, but 9-1-1 would be.  

[MAP] If the proxy at the emplyees home office wants to try to handle an 
emergency call and contact the emergency bureau at the location where it thinks 
the person is located, and has all the data to define such things, then let it. 
But I doubt that any regulatory agency would touch this, and any requirement 
they try to place on it would be a mess, since it's so dependent on lots of 
things out of anyone's control. I would say it's not a case we're trying to solve.


In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:

Again, generally I agree.  It's a very unfortunate situation, because if you
are a visitor, your awareness of the local emergency number is not very good,
while you clearly know your home emergency number.  

[MAP] True, but that doesn't mean one should use their  home emergency 
number, just because they know it better. I know how to drive on the right better, 
but if I'm in England, I still have to drive on the left.


In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:

>[MAP] The final sentence should state: "User agents conforming to the SIP 
>specification [RFC3261], but unaware of this document, must be able to place 
>emergency calls without any restriction using entry of normal emergency 
>numbers, although they will not generate the SOS URI." 
I think this is incorrect.  First of all, a proxy can redirect to sos.
Secondly, you can't get "without restriction" because you don't get
location. 

[MAP] I think what I wrote above is a regulatory requirement. The UA must be 
able to place an emergency call "without restriction". Of course the proxy can 
generate an SOS URI. I didn't say it couldn't. As noted above, "location" is 
another issue not dependent on the SOS URI and not made possible by this 
document.


In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:

>[MAP] Case 4 is used when the terminal does not know the local dial plan. It 
depends 
>on a proxy to interpret the number, just as for any other dialed number. 
Disagree, as above

[MAP] I'm not sure why you said "disagreed", since your other comments seem 
to agree with the point I'm making in case 4 (terminal doesn't understand dial 
plan so it depends on a proxy to "recognize the call as an emergcncy call"). 
I'm just trying to be explicit by enumerating the case which this document does 
not cover.


Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/11/2004 1:50:25 PM Eastern Standard Time, Brian.Rosen@marconi.com wr=
ites:
<BR>
<BR>The fundamental idea that this draft proposes is that emergency calls=20
<BR>get special attention. &nbsp;The special attention is that location is i=
ncluded=20
<BR>on the call. &nbsp;There are also routing implications, but the inclusio=
n of=20
<BR>location is the key issue.
<BR>
<BR>[MAP] The point I'm trying to make is that the requirement to provide lo=
cation for emergency calls has nothing to do with this draft (SOS URI). It m=
ust be accepted as a requirement and solved even if this SOS URI disappeared=
 today. It can not be the SOS URI which enables the location to be provided.
<BR>
<BR>
<BR>In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, Brian.Ros=
en@marconi.com writes:
<BR>
<BR>The issue you raise with local emergency numbers is complicated by VPN=20
<BR>tunnels. &nbsp;My usual example is the patron of a cafe in Chicago seein=
g an=20
<BR>accident while connected via a hotspot through a VPN tunnel to an employ=
er's
<BR>VoIP system in Sierra Leone. &nbsp;In this example, dialing the Sierra L=
eone=20
<BR>emergency number would not be an emergency call, but 9-1-1 would be. &nb=
sp;
<BR>
<BR>[MAP] If the proxy at the emplyees home office wants to try to handle an=
 emergency call and contact the emergency bureau at the location where it th=
inks the person is located, and has all the data to define such things, then=
 let it. But I doubt that any regulatory agency would touch this, and any re=
quirement they try to place on it would be a mess, since it's so dependent o=
n lots of things out of anyone's control. I would say it's not a case we're=20=
trying to solve.
<BR>
<BR>
<BR>In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, Brian.Ros=
en@marconi.com writes:
<BR>
<BR>Again, generally I agree. &nbsp;It's a very unfortunate situation, becau=
se if you
<BR>are a visitor, your awareness of the local emergency number is not very=20=
good,
<BR>while you clearly know your home emergency number. &nbsp;
<BR>
<BR>[MAP] True, but that doesn't mean one should use their &nbsp;home emerge=
ncy number, just because they know it better. I know how to drive on the rig=
ht better, but if I'm in England, I still have to drive on the left.
<BR>
<BR>
<BR>In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, Brian.Ros=
en@marconi.com writes:
<BR>
<BR>&gt;[MAP] The final sentence should state: "User agents conforming to th=
e SIP=20
<BR>&gt;specification [RFC3261], but unaware of this document, must be able=20=
to place=20
<BR>&gt;emergency calls without any restriction using entry of normal emerge=
ncy=20
<BR>&gt;numbers, although they will not generate the SOS URI."=20
<BR>I think this is incorrect. &nbsp;First of all, a proxy can redirect to s=
os.
<BR>Secondly, you can't get "without restriction" because you don't get
<BR>location.=20
<BR>
<BR>[MAP] I think what I wrote above is a regulatory requirement. The UA mus=
t be able to place an emergency call "without restriction". Of course the pr=
oxy can generate an SOS URI. I didn't say it couldn't. As noted above, "loca=
tion" is another issue not dependent on the SOS URI and not made possible by=
 this document.
<BR>
<BR>
<BR>In a message dated 2/11/2004 1:50:25 PM Eastern Standard Time, Brian.Ros=
en@marconi.com writes:
<BR>
<BR>&gt;[MAP] Case 4 is used when the terminal does not know the local dial=20=
plan. It depends=20
<BR>&gt;on a proxy to interpret the number, just as for any other dialed num=
ber.=20
<BR>Disagree, as above
<BR>
<BR>[MAP] I'm not sure why you said "disagreed", since your other comments s=
eem to agree with the point I'm making in case 4 (terminal doesn't understan=
d dial plan so it depends on a proxy to "recognize the call as an emergcncy=20=
call"). I'm just trying to be explicit by enumerating the case which this do=
cument does not cover.
<BR>
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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> [MAP] The point I'm trying to make is that the requirement to provide 
> location for emergency calls has nothing to do with this draft (SOS 
> URI). It must be accepted as a requirement and solved even if this SOS 
> URI disappeared today. It can not be the SOS URI which enables the 
> location to be provided.

No, but the sos URI allows a downstream proxy to provide the location to 
the calling user via a redirect response (The caller then re-initiates a 
location-equipped emergency call as if it had the location itself).

Whether the SOS URI is the *only* mechanism for the calling UA to 
request location information is another question.


> [MAP] If the proxy at the emplyees home office wants to try to handle an 
> emergency call and contact the emergency bureau at the location where it 
> thinks the person is located, and has all the data to define such 
> things, then let it. But I doubt that any regulatory agency would touch 
> this, and any requirement they try to place on it would be a mess, since 
> it's so dependent on lots of things out of anyone's control. I would say 
> it's not a case we're trying to solve.

In our architecture, this just works and is no more difficult than 
solving any other problem. If you have a concrete concern, please share 
it, but general regulation fear mongering isn't going to help much.



> 
> [MAP] True, but that doesn't mean one should use their  home emergency 
> number, just because they know it better. I know how to drive on the 
> right better, but if I'm in England, I still have to drive on the left.

The problem is that driving on the wrong side gets you killed, while 
being able to dial a number that you remember while in panic quickly 
saves your life. Thus, the analogy is a rather poor one, in my opinion.

The only drawback is that, in rather rare circumstances, your back-home 
emergency number is a valid service number at your current location. 
This seems, at worst, a minor nuisance. Since you're familiar with your 
home emergency number, if you find that the Easter Island use 911 for 
their directory service, there could be a user interface override 
('escape').

One solution is to make this behavior configurable. If you'd rather use 
a device that does not remember your home dial plan, be my guest. Given 
that this is new and improved behavior compared to today's system, 
making this optional is not a big deal.


> [MAP] I think what I wrote above is a regulatory requirement. The UA 
> must be able to place an emergency call "without restriction". Of course 
> the proxy can generate an SOS URI. I didn't say it couldn't. As noted 
> above, "location" is another issue not dependent on the SOS URI and not 
> made possible by this document.

There are two cases:

(1) Your UA recognizes the digits you dial as an emergency number and 
translates the digits into sos@something.

(2) It doesn't.

In the former case, the call will be placed correctly, assuming that 
(crude) location information is available. We should make sure that case 
(1) is the high-probability event. With the mechanisms described in the 
emergency-arch document, finding the local emergency number is not hard. 
That document suggests two different approaches (DNS and XCAP).

In case (2), two things can happen: the call gets placed as a tel:XXX 
call and your local outbound proxy routes it to a gateway that does the 
right thing. This should be the common scenario, leaving odd VPN cases 
aside. (A well-maintained VPN will always get you to case (1).)

Thus, I don't see the problem.



> an emergcncy call"). I'm just trying to be explicit by enumerating the 
> case which this document does not cover.

Which I still don't understand. Getting the geographically local dial 
plan is beyond the scope of this document, but is part of the 
emergency-arch document.

> 
> 
> Mike Pierce

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Date: Wed, 11 Feb 2004 22:48:27 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [Sipping] Other concerns within sos ID
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> Further, I have a serious concern with the desciption in 4.2 that a 
> terminal must recognize many different digits strings and generate an 
> SOS, independent of where it is located, if it "has no knowledge of 
> local emergency numbers". This would result in many invalid calls to an 
> emergency center, since many of these numbers could be valid 
> (non-emergency) numbers (see note about 118 in Finland). Again, this 

 From my vague recollection from the last iteration of this same 
discussion, some 3GPP folks claimed that this was the list of default 
emergency numbers embedded in 3G (or maybe even 2G) phones. That was the 
reason I added this list; it was specifically requested by somebody else 
(it was not in the earlier versions of -sos-, for example).

If this is not true, I'll gladly remove the default list and rely on 
explicit configuration only, particularly since we now have, I believe, 
a better story on configuration than we did when this document was last 
discussed. (See emergency-arch for some of the details.)



> The statement in Section 5 that an "outbound proxy server MUST recognize 
> all local emergency numbers ..." is incorrect (and impossible). 
> (Possibly this sentence was meant to say " ... all local emergency 
> numbers listed in Section 4.2...", but this is still impossible). An 
> outbound proxy only needs to recognize those emergency numbers 
> applicable to the local areas for which it is providing service. An 
> outbound proxy located in and providing service for callers in the US 
> certainly does not have to recognize the emergency number used in some 
> country in Africa. A visitor from that country must dial 911 for an 

If you looked into the emergency-arch document, you'd find that this 
problem is actually fairly easy to solve.


> emergency if they are in the US and receiving service from a service 
> provider in the US. If the person from Africa is travelling in Asia and 

I'm not assuming the existence of a 'service provider'.


> uses the proxy in the US for service, they can't expect it to correctly 
> handle emergency service. The proxy could not possibly distinguish 
> between the emergency numbers for Asia, Africa, US, and all the other 
> local dialing plans that the user might know about. This is an 
> application that must be limited to entry of E.164 numbers (which does 
> not include any emergency numbers or local dialing plans).
> 
> Further, in Section 5, it is unclear how to interpret the phrase "as 
> well as the tel URIs enumerated in Section Section 4.2" (as a 
> requirement). Section 4.2 listed two tel URIs (911 and 112) as examples 
> ("such as"), not as an exhaustive list. Did this intend to say "as well 
> as tel URIs formed from all the emergency numbers enumerated in the 
> first paragraph of Section 4.2"?


> The requirement for the user agent to "reveal a different amount of 
> information to the callee when making an emergency call" also applies to 
> those calls dialed with the traditional emergency number when the user 
> agent does not know the dial plan and therefore can not "clearly and 
> unambiguously identity emergency calls". This function of revealing 
> information must be possible without the SOS mechanism. If based on the 

Good luck convincing the GEOPRIV guys. This is a discussion best taken 
to a different venue.



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> If an sos URI contains a domain name, I presume that a proxy will 
> forward the request based on that domain name and never look at the 
> local part containing the "sos.xxx".  Is that correct?

We have discussed two options in the past. One is the one you describe 
(and is normal SIP behavior); on the other hand, an outbound proxy could 
indeed inspect the URI and treat the call as an emergency call. I'd like 
to have a separate discussion as to whether such 'early intercept' is 
desirable.

This behavior would need to be specified in the emergency-arch draft.

> 
> If so, does that mean that the proxy in that domain is expected to look 
> at the location information and forward to the appropriate ECC proxy?

Yes.

> 
> As for internationalization, that could get out of hand.  Was the intent 

I don't see how multiple languages could work in practice. Since 
emergency calls can be routed to strange places, outside the local 
calling area, every proxy would have to understand every possible 
variation of 'police' in hundreds of languages.

> that these URIs are machine readable only?  I suspect that some 

Mostly machine-generated. But that's a user interface issue that I 
thought IETF specs generally stayed out of.

> internationally recognized icons on the user agents that translate into 
> these URIs will be needed.  But, that could be tricky as well:
> 
> sos.fire     picture of flames
> sos.rescue   picture of Red Cross/Crescent?
> sos.police   picture of badge/police belt?
> sos.marine   picture of boat + cross/crescent?
> sos.mountain picture of mountain + cross/crescent?
> 
> But those icons are phone implementation details.

I would also consider them somewhat unlikely to be visible on most 
devices, certainly hard phones with plastic buttons.

For soft phones, if there is local dial plan discovery, the phone can 
display the appropriate icons or, easily, a word that means something in 
the user's native language.

In general, I suspect that the use of different numbers is declining. 
For example, in Germany, due to 112 as the GSM emergency number and the 
EU common-emergency-number plans, dialing 112 will get you all emergency 
services, even though 110 is the designated police number.

Henning


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From exim@www1.ietf.org  Wed Feb 11 23:10:31 2004
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Subject: [Sipping] SOS: 'rescue'
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I checked and the delineation of 'fire', 'rescue' and 'police' is quite 
common, says Google, as in

http://www.hinckleytwp.org/fire-rescue-police.html
http://www.tunstallfd.org/freqs.htm

and thousands more. I'll ask the NENA folks whether something like 'ems' 
would be preferable.

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Subject: Re: [Sipping] Other concerns within sos ID
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The Chicago/Sierra Leone scenario reinforces the suggestion of taking =
this issue up with the ITU. Regardless of how much effort goes into the =
correct routing of emergency calls, it will only be a matter of time =
before jurisdictions will find them selves fielding emergency calls from =
placed from other jurisdictions. This more than a technology problem, =
jurisdictions will need procedures/protocols for handing on emergency =
calls to each other.=20

Regards
Evan Keats



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Picked up one relevant clarification item.

Rosen, Brian wrote:


>>The statement in Section 5 that an "outbound proxy server MUST recognize 
>>all local emergency numbers ..." is incorrect (and impossible). (Possibly 
>>this sentence was meant to say " ... all local emergency numbers listed in 
>>Section 4.2...", but this is still impossible). An outbound proxy only
> 
> needs 
> Rewording is required, I agree, but it has to refer to local-to-the-phone.

'Local' has too many meanings - I guess it should always be qualified. I 
reworded this as referring to a the emergency numbers *local to the 
outbound proxy*. This is primarily meant to deal with phones that don't 
understand the conventions in the 'sos' document.


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Please take a look at the emergency-arch draft (and Brian Rosen's DNS 
draft). The whole point of this exercise is to make sure that a PSAP in 
Sierra Leone never has to field an emergency call except those 
pertaining to its own service area, even if an outbound proxy in Sierra 
Leone gets an emergency call from a user currently visiting Chicago. The 
only coordination is the appropriate population of the DNS records.

This is not fundamentally different from the action that the Sierra 
Leonen SMTP MTA has to do if I submit an email addressed to 
uchicago.edu. This also does not require that the ISP in Sierra Leone 
contact the ITU to ask for proper diplomatic protocol.

Tracking the responsible entity for each country may well be an ITU 
issue, as it is somewhat related to the ENUM top-level assignment problem.

Keats, Evan wrote:

> The Chicago/Sierra Leone scenario reinforces the suggestion of taking
> this issue up with the ITU. Regardless of how much effort goes into
> the correct routing of emergency calls, it will only be a matter of
> time before jurisdictions will find them selves fielding emergency
> calls from placed from other jurisdictions. This more than a
> technology problem, jurisdictions will need procedures/protocols for
> handing on emergency calls to each other.
> 
> Regards Evan Keats
> 
> 
> 
> _______________________________________________ Sipping mailing list
> https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW
> development of the application of SIP Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sip@ietf.org for new developments of core SIP

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Thu, 12 Feb 2004 13:54:56 -0000
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With a view of trying to borrow decisions from elsewhere.

In 3GPP when the phone recognises that it is an emergency call, and needs to identify the type of emergency call, then it has access to signal categories with the following names (ref 3GPP TS 22.101):

------------------------------
-	Police
-	Ambulance
-	Fire Brigade
-	Marine Guard
-	Mountain Rescue
-	Spare, at least [three] different types
-----------------------------

which results in a coding accompanying the call establishment of (ref 3GPP TS 24.008 table 10.5.135d)

--------------------------
Emergency Service Category Value (octet 3)
The meaning of the Emergency Category Value is derived from the following settings (Please see 3GPP TS 22.101 clause 8):
Bit 1	Police
Bit 2	Ambulance
Bit 3	Fire Brigade
Bit 4	Marine Guard
Bit 5	Mountain Rescue
Bits 6,7,8 are spare and set to "0"

Mobile station may set one or more bits to "1"
If more than one bit is set to "1", routing to a combined Emergency centre (e.g. ambulance and fire brigade in Japan) is required. If the MSC can not match the received service category to any of the emergency centres, it shall route the call to an operator defined default emergency centre.

If no bit is set to "1", the MSC shall route the Emergency call to an operator defined default emergency centre
-------------------------

These terms are at least accepted by multiple SDOs in the international community for transposition into their national standards.

I also note that 3GPP does have the capability of signalling multiple preferences - this left me wondering whether we would be better off trying to follow that, possibly with a single emergency uri, but with the possibility of signalling category as a URI parameter with a list of values. Any thoughts on this?

regards

Keith


> -----Original Message-----
> From: Andrew Allen [mailto:aallen@rim.net]
> Sent: 11 February 2004 16:57
> To: mhammer@cisco.com; sipping@ietf.org
> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> 
> 
> How about sos.medical then since that outlines the type of 
> emergency. I agree that rescue may be confused with the others.
> 
> Andrew
> --------------------------
> Andrew Allen
> Manager Standards
> Research In Motion Ltd
> BlackBerry Mobile  +1 847 809 8636
> European Mobile    +358 50 467 5870
> http://www.rim.com/
> 
> Sent from my BlackBerry Wireless Handheld
> 
> 
> -----Original Message-----
> From: Michael Hammer <mhammer@cisco.com>
> To: sipping@ietf.org <sipping@ietf.org>
> Sent: Wed Feb 11 10:45:29 2004
> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> 
> At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
> >         sos.fire                fire brigade
> >         sos.rescue      ambulance (rescue)
> >         sos.marine      marine guard
> >         sos.police      police (law enforcement)
> >         sos.mountain    mountain rescue
> 
> The second sub-address seems vague to me.  Rescue could mean many 
> things.  Perhaps, I need to be rescued from a fashion 
> emergency. :)  Would 
> something like:
> 
>          sos.medic        medical (doctor, nurse, paramedic team)
> 
> be more descriptive?
> 
> Also, with respect to registrations, I could see commercial interests 
> wanting to get into the act, such as an sos.fashion that I hinted at 
> above.  Is there some guidance necessary to IANA?  Do we 
> expect that the 
> IETF WGs will act as arbiters for this?
> 
> Second, wrt location information reporting, is there any need for a 
> convention for providing a URL to which a receiving ECC can 
> SUBSCRIBE to 
> get location updates from either the UA or a network element 
> that is doing 
> triangulation.  I am wondering if something like the following can be 
> inserted into the emergency INVITE:
> 
> Call-Info: <sips:triangulator@local-proxy.domain.com> 
> ;purpose=location
> 
> Mike
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sipping] SOS: emergency numbers
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[Let me try to tease out separate issues in the discussion and see if we 
can make progress that way. In some cases, we're revisiting old ground, 
so the proponents of the 'old' solution will have to chime in.]

One of the core problems, spanning the architecture and sos drafts, is 
the question: what is an emergency number, how do systems find out and 
who needs to know?

The current proposal is a bit of hybrid, in that there are pre-defined 
numbers (112, 911, 000, etc.) and acquired numbers. Numbers are acquired 
using three different mechanisms: XCAP to the home domain, XCAP to the 
visited domain (following the configuration draft methodology) and DNS 
based on knowledge of the current country code. None of these mechanisms 
existed in the first iterations of the -sos- draft, so it may be 
appropriate to revisit the issue.

Thus, a streamlined proposal would be:

(1) No protocol pre-defined numeric identifiers, not even 112 and 911. 
(Devices with personalization devices, such as SIM cards, may pre-load a 
dialplan, see (3).)

(2) Any digit string NNN (not just emergency numbers) dialed and 
unrecognized gets translated into sip:NNN@aord;user=dialstring (as 
discussed at the last IETF meeting). AORD is then responsible for 
recognizing this as an emergency number, based on the current and 
possibly home country of the caller.

An alternative is to translate these dialstrings into 
sip:NNN@outbound-proxy;user=dialstring, e.g., using the DHCP-provided 
outbound proxy address. I think this might be better, but that's a more 
general discussion.

(2a) In general, digit strings that match a dialplan MAY become tel 
URIs, after they have been suitably normalized by the dialplan 
transformation rules. This does not apply to emergency numbers, where 
(3) applies.

(3) Any digit dial string recognized as an emergency number gets 
translated into sos*@AORD.

AORD = address of record domain, i.e., the domain part of my address of 
record.

The device owner decides whether to acquire only the geographically 
local dialplan (MUST) or also his 'home' dialplan (MAY).

Is this acceptable?

Henning

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Interesting. The selection sounds like caller preferences to me. The 
nice thing about that is that you might be able to use existing 
techniques to match up services and destinations.

For example, this could be considered a sip.class value, extending 
'home' and 'business', or a new media feature tag, either specific to 
emergency services or a more generic service identifier. (The latter 
does invite the fashion emergency and roadside repair services into the 
same namespace.)

Drage, Keith (Keith) wrote:

> With a view of trying to borrow decisions from elsewhere.
> 
> In 3GPP when the phone recognises that it is an emergency call, and needs to identify the type of emergency call, then it has access to signal categories with the following names (ref 3GPP TS 22.101):
> 
> ------------------------------
> -	Police
> -	Ambulance
> -	Fire Brigade
> -	Marine Guard
> -	Mountain Rescue
> -	Spare, at least [three] different types
> -----------------------------
> 
> which results in a coding accompanying the call establishment of (ref 3GPP TS 24.008 table 10.5.135d)
> 
> --------------------------
> Emergency Service Category Value (octet 3)
> The meaning of the Emergency Category Value is derived from the following settings (Please see 3GPP TS 22.101 clause 8):
> Bit 1	Police
> Bit 2	Ambulance
> Bit 3	Fire Brigade
> Bit 4	Marine Guard
> Bit 5	Mountain Rescue
> Bits 6,7,8 are spare and set to "0"
> 
> Mobile station may set one or more bits to "1"
> If more than one bit is set to "1", routing to a combined Emergency centre (e.g. ambulance and fire brigade in Japan) is required. If the MSC can not match the received service category to any of the emergency centres, it shall route the call to an operator defined default emergency centre.
> 
> If no bit is set to "1", the MSC shall route the Emergency call to an operator defined default emergency centre
> -------------------------
> 
> These terms are at least accepted by multiple SDOs in the international community for transposition into their national standards.
> 
> I also note that 3GPP does have the capability of signalling multiple preferences - this left me wondering whether we would be better off trying to follow that, possibly with a single emergency uri, but with the possibility of signalling category as a URI parameter with a list of values. Any thoughts on this?
> 
> regards
> 
> Keith
> 
> 
> 
>>-----Original Message-----
>>From: Andrew Allen [mailto:aallen@rim.net]
>>Sent: 11 February 2004 16:57
>>To: mhammer@cisco.com; sipping@ietf.org
>>Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
>>
>>
>>
>>
>>How about sos.medical then since that outlines the type of 
>>emergency. I agree that rescue may be confused with the others.
>>
>>Andrew
>>--------------------------
>>Andrew Allen
>>Manager Standards
>>Research In Motion Ltd
>>BlackBerry Mobile  +1 847 809 8636
>>European Mobile    +358 50 467 5870
>>http://www.rim.com/
>>
>>Sent from my BlackBerry Wireless Handheld
>>
>>
>>-----Original Message-----
>>From: Michael Hammer <mhammer@cisco.com>
>>To: sipping@ietf.org <sipping@ietf.org>
>>Sent: Wed Feb 11 10:45:29 2004
>>Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
>>
>>At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
>>
>>>        sos.fire                fire brigade
>>>        sos.rescue      ambulance (rescue)
>>>        sos.marine      marine guard
>>>        sos.police      police (law enforcement)
>>>        sos.mountain    mountain rescue
>>
>>The second sub-address seems vague to me.  Rescue could mean many 
>>things.  Perhaps, I need to be rescued from a fashion 
>>emergency. :)  Would 
>>something like:
>>
>>         sos.medic        medical (doctor, nurse, paramedic team)
>>
>>be more descriptive?
>>
>>Also, with respect to registrations, I could see commercial interests 
>>wanting to get into the act, such as an sos.fashion that I hinted at 
>>above.  Is there some guidance necessary to IANA?  Do we 
>>expect that the 
>>IETF WGs will act as arbiters for this?
>>
>>Second, wrt location information reporting, is there any need for a 
>>convention for providing a URL to which a receiving ECC can 
>>SUBSCRIBE to 
>>get location updates from either the UA or a network element 
>>that is doing 
>>triangulation.  I am wondering if something like the following can be 
>>inserted into the emergency INVITE:
>>
>>Call-Info: <sips:triangulator@local-proxy.domain.com> 
>>;purpose=location
>>
>>Mike
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] Other concerns within sos ID
Cc: sipping@ietf.org
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At 11:00 PM 2/11/2004 -0500, Henning Schulzrinne wrote:
>>If an sos URI contains a domain name, I presume that a proxy will forward 
>>the request based on that domain name and never look at the local part 
>>containing the "sos.xxx".  Is that correct?
>
>We have discussed two options in the past. One is the one you describe 
>(and is normal SIP behavior); on the other hand, an outbound proxy could 
>indeed inspect the URI and treat the call as an emergency call. I'd like 
>to have a separate discussion as to whether such 'early intercept' is 
>desirable.
>
>This behavior would need to be specified in the emergency-arch draft.
>
>>If so, does that mean that the proxy in that domain is expected to look 
>>at the location information and forward to the appropriate ECC proxy?
>
>Yes.
>
>>As for internationalization, that could get out of hand.  Was the intent
>
>I don't see how multiple languages could work in practice. Since emergency 
>calls can be routed to strange places, outside the local calling area, 
>every proxy would have to understand every possible variation of 'police' 
>in hundreds of languages.
>
>>that these URIs are machine readable only?  I suspect that some
>
>Mostly machine-generated. But that's a user interface issue that I thought 
>IETF specs generally stayed out of.

Agreed.  I was only suggesting that the dial-plan+local-language 
combination could be rather difficult.  The below stuff was just suggesting 
that a GUI to machine translation could help solve that.


>>internationally recognized icons on the user agents that translate into 
>>these URIs will be needed.  But, that could be tricky as well:
>>sos.fire     picture of flames
>>sos.rescue   picture of Red Cross/Crescent?
>>sos.police   picture of badge/police belt?
>>sos.marine   picture of boat + cross/crescent?
>>sos.mountain picture of mountain + cross/crescent?
>>But those icons are phone implementation details.
>
>I would also consider them somewhat unlikely to be visible on most 
>devices, certainly hard phones with plastic buttons.

Old POTS black phones yes, but many hard phones and cell phones today all 
come with a display and control software.  So, it does not have to be a 
softphone on a PC.


>For soft phones, if there is local dial plan discovery, the phone can 
>display the appropriate icons or, easily, a word that means something in 
>the user's native language.
>
>In general, I suspect that the use of different numbers is declining. For 
>example, in Germany, due to 112 as the GSM emergency number and the EU 
>common-emergency-number plans, dialing 112 will get you all emergency 
>services, even though 110 is the designated police number.
>
>Henning

It would be nice if Asia, Africa, and South America had standard regional 
emergency numbers.  (perhaps someone will tell us there is)

Australia is apparently "000".

Mike


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From: "Farahmand, Fardad" <FFarahmand@sonusnet.com>
To: "'Hearty, John'" <John.Hearty@Level3.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: sipping@ietf.org, "Peterson, Jon" <jon.peterson@neustar.biz>
Subject: RE: [Sipping] TCAP tunneling
Date: Thu, 12 Feb 2004 11:31:31 -0500
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The Yu draft describes a redirector or proxy which would perform the LNP for
the called number. The redirector would add a Contact in 3XX with the
information about the ported number. The proxy would add the information to
the R-URI.

CNAM is a bit different since we are dealing with the calling party
information. I am a bit confused about Jonathan's comment regarding adding
the CNAM information to the Contact header. It seems that the right approach
would be for Proxies to add a PAI with display name.  Looking at RFC3325, it
seems that PAI can also be present in 3XX responses.  So potentially
redirectors can also add CNAM information by adding a PAI.  

So it seems like there are standard methods of introducing LNP and CNAM
services in SIP networks.  But what about all the other TCAP based services?
This is yet to be resolved.

Thanks,
Fardad

> -----Original Message-----
> From: Hearty, John [mailto:John.Hearty@Level3.com]
> Sent: Wednesday, February 11, 2004 6:50 PM
> To: Jonathan Rosenberg
> Cc: sipping@ietf.org; Peterson, Jon
> Subject: RE: [Sipping] TCAP tunneling
> 
> 
> In looking at the ENUM RFC, it looks like NAPTR records are used, and
> they convert E.164 numbers to URIs.  The syntax for the URIs does not
> appear to allow for Display Names, which is what I believe needs to be
> populated when forwarding SIP messages.  I suppose we could 
> still do TXT
> type DNS records for this, but can't find any standards specific to
> CNAM.  The alternate idea similar to the yu draft seems to contradict
> the statement that SIP isn't a database query protocol.  If anyone is
> already doing something like this or has plans I'd love to hear about
> it.
> 
> John
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Monday, February 02, 2004 11:29 PM
> > To: Hearty, John
> > Cc: sipping@ietf.org; Peterson, Jon
> > Subject: Re: [Sipping] TCAP tunneling
> > 
> > SIP isnt a database query protocol, and cnam is just a DB query
> service.
> > As such, I would think that enum would be an appropriate 
> technique for
> > this. Any other DB query mechanism will do as well.
> > 
> > Another idea would be to use a redirect service, similar to what has
> > been proposed in the past for LNP:
> > 
> > http://www.watersprings.org/pub/id/draft-yu-sip-np-02.txt
> > 
> > except that a request for a number would be redirected to a Contact
> > which contains that number plus the display name, representing the
> > calling name.
> > 
> > -Jonathan R.
> > 
> > Hearty, John wrote:
> > > Other than the miller draft below, is there any SIP standard
> proposed to
> > > carry cnam queries?
> > >
> > > John Hearty
> > > Level3
> > >
> > >
> > >>-----Original Message-----
> > >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On
> Behalf
> > >
> > > Of
> > >
> > >>Arjun Roychowdhury
> > >>Sent: Monday, February 02, 2004 7:25 AM
> > >>To: Banibrata Dutta
> > >>Cc: kagarwal@cedarpointcom.com; ranjit.avasarala@wipro.com;
> > >>sipping@ietf.org; sipping-admin@ietf.org
> > >>Subject: RE: [Sipping] TCAP tunneling
> > >>
> > >>Actually, there _is_ an independant draft on how to carry TCAP in
> SIP.
> > >>http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt
> > >>
> > >>Its another matter whether this is the right thing to do or not
> > >
> > > (please
> > >
> > >>refer to old archives for the debates).
> > >>Kaitaki, you may want to read the above draft as well as the
> > >
> > > discussions
> > >
> > >>on it to conclude for yourself which is the preferred 
> method in your
> > >>deployment scenario.
> > >>
> > >>regds
> > >>arjun
> > >>
> > >>
> > >>--
> > >>Arjun Roychowdhury @ Hughes Software Systems
> > >>11717 Exploration Lane, Germantown MD 20876
> > >>(O): 301 212 7860
> > >>VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)
> > >>
> > >>
> > >>
> > >>
> > >>"Banibrata Dutta" <dutta@india.hp.com>
> > >>Sent by: sipping-admin@ietf.org
> > >>02/02/2004 12:25 AM
> > >>
> > >>
> > >>        To:     <ranjit.avasarala@wipro.com>,
> > >
> > > <kagarwal@cedarpointcom.com>,
> > >
> > >><sipping@ietf.org>
> > >>        cc:
> > >>        Subject:        RE: [Sipping] TCAP tunneling
> > >>
> > >>
> > >>hi,
> > >>
> > >>franly, i'd say that, it depends on your motivation/reason to use
> SIP
> > >>for
> > >>tunneling TCAP. of course you can tunnel TCAP, but why you'd want
> to,
> > >
> > > is
> > >
> > >>a justification you'd have to work out. AFAIK, there's not
> ietf-draft
> > >>that
> > >>says, how.
> > >>
> > >>you can obviously tunnel TCAP query in an INVITE or INFO for that
> > >
> > > matter
> > >
> > >>(ideally in INVITE s.t. a SIP-dialog is also created which may be
> used
> > >>to
> > >>model the TCAP dialog as well, perhaps!). the ASN.1 BER 
> encoded TCAP
> > >>messages may then reside within the body of a multi-part 
> (MIME) enc,
> > >>INVITE.
> > >>
> > >>infact 1 reason i see where this may make sense is that you would
> > >
> > > rather
> > >
> > >>not have two stacks s.a. SIGTRAN and SIP for signalling transport,
> and
> > >>use
> > >>SIP as the *only* signalling solution, i.e. for the signalling
> > >>application
> > >>as well as for signalling transport, and support 
> legacy/TCAP apps by
> > >>tunneling
> > >>them within SIP. but then, you generally have the constraints of
> > >>not-enough-
> > >>space, (for 2 stacks) only on resource-limited devices, s.a.
> > >
> > > hand-sets,
> > >
> > >>embedded devices etc., but i am unable see (yet), how TCAP can be
> > >>related to
> > >>such devices.
> > >>
> > >>my 2 cents,
> > >>Banibrata Dutta,
> > >>  Hewlett-Packard ISO (P) Ltd,
> > >>  29, Cunningham Road,
> > >>  Bangalore - 560052
> > >>  Ph: +91-80-22051796
> > >>
> > >>
> > >>>-----Original Message-----
> > >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > >>>On Behalf Of ranjit.avasarala@wipro.com
> > >>>Sent: Monday, February 02, 2004 10:20 AM
> > >>>To: kagarwal@cedarpointcom.com; sipping@ietf.org
> > >>>Subject: RE: [Sipping] TCAP tunneling
> > >>>
> > >>>
> > >>>Hi
> > >>>   TCAP is used for Transaction Capabilities i.e database
> > >>>related while SIP with ISUP tunneling is used between MGCs.
> > >>>So I don't think TCAP can be tunneled in SIP.
> > >>>
> > >>>Ranjit
> > >>>Ph: 8520408/16 extn: 2173
> > >>>
> > >>>
> > >>>
> > >>>
> > >>>-----Original Message-----
> > >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > >>>On Behalf Of Kaitki Agarwal
> > >>>Sent: Monday, February 02, 2004 8:28 AM
> > >>>To: sipping@ietf.org
> > >>>Subject: [Sipping] TCAP tunneling
> > >>>
> > >>>
> > >>>Hi
> > >>> Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> > >>>
> > >>>Thanks
> > >>>Kaitki
> > >>>Jj~Yzy=i~Rz}y
> > >>>
> > >>>_______________________________________________
> > >>>Sipping mailing list
> https://www1.ietf.org/mailman/listinfo/sipping
> > >>>This list is for NEW development of the application of SIP
> > >>>Use sip-implementors@cs.columbia.edu for questions on current
> > >>>sip Use sip@ietf.org for new developments of core SIP
> > >>>
> > >>
> > >>
> > >>_______________________________________________
> > >>Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > >>This list is for NEW development of the application of SIP
> > >>Use sip-implementors@cs.columbia.edu for questions on current sip
> > >>Use sip@ietf.org for new developments of core SIP
> > >>
> > >>
> > >>
> > >>
> > >>_______________________________________________
> > >>Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > >>This list is for NEW development of the application of SIP
> > >>Use sip-implementors@cs.columbia.edu for questions on current sip
> > >>Use sip@ietf.org for new developments of core SIP
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > 
> > --
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Technology Officer                    Parsippany, NJ 
> 07054-2711
> > dynamicsoft
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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References: <4.3.2.7.2.20040211114231.00b29e68@cia.cisco.com> <4.3.2.7.2.20040211114231.00b29e68@cia.cisco.com> <4.3.2.7.2.20040212103534.00b54878@cia.cisco.com>
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> 
> Old POTS black phones yes, but many hard phones and cell phones today 
> all come with a display and control software.  So, it does not have to 
> be a softphone on a PC.

Certainly.

> It would be nice if Asia, Africa, and South America had standard 
> regional emergency numbers.  (perhaps someone will tell us there is)
> 
> Australia is apparently "000".

http://www.911dispatch.com/web_graphic/graphic1.html has a graphic and 
lists of numbers. As you can see, very little coherence, but lots of 
countries with a single NNN.

> 
> Mike

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



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From: "Tolga Asveren" <asveren@ulticom.com>
To: <sipping@ietf.org>
Subject: RE: [Sipping] TCAP tunneling
Date: Thu, 12 Feb 2004 11:53:44 -0500
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A few thoughts below....

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
> Farahmand, Fardad
> Sent: Thursday, February 12, 2004 11:32 AM
> To: 'Hearty, John'; Jonathan Rosenberg
> Cc: sipping@ietf.org; Peterson, Jon
> Subject: RE: [Sipping] TCAP tunneling
>
>
>
> The Yu draft describes a redirector or proxy which would perform
> the LNP for
> the called number. The redirector would add a Contact in 3XX with the
> information about the ported number. The proxy would add the
> information to
> the R-URI.
>
> CNAM is a bit different since we are dealing with the calling party
> information. I am a bit confused about Jonathan's comment regarding adding
> the CNAM information to the Contact header. It seems that the
> right approach
> would be for Proxies to add a PAI with display name.  Looking at
> RFC3325, it
> seems that PAI can also be present in 3XX responses.  So potentially
> redirectors can also add CNAM information by adding a PAI.
>
> So it seems like there are standard methods of introducing LNP and CNAM
> services in SIP networks.  But what about all the other TCAP
> based services?
> This is yet to be resolved.
[TOLGA]Maybe the key point is that the services are *not* based on TCAP,
TCAP is just a way to do synchronous remote procedure calls, so it does not
really have service related semantics. Important is, what is transferred on
top of TCAP (e.g. INAP), and I would think in such a case, the best way is
thinking on a per service bases rather than trying to encapsulate such a
protocol (afterall also for INAP just the protocol is standardized by ITU,
there are different ways to implement a certain service from message flow
point of view)
>
> Thanks,
> Fardad
>
> > -----Original Message-----
> > From: Hearty, John [mailto:John.Hearty@Level3.com]
> > Sent: Wednesday, February 11, 2004 6:50 PM
> > To: Jonathan Rosenberg
> > Cc: sipping@ietf.org; Peterson, Jon
> > Subject: RE: [Sipping] TCAP tunneling
> >
> >
> > In looking at the ENUM RFC, it looks like NAPTR records are used, and
> > they convert E.164 numbers to URIs.  The syntax for the URIs does not
> > appear to allow for Display Names, which is what I believe needs to be
> > populated when forwarding SIP messages.  I suppose we could
> > still do TXT
> > type DNS records for this, but can't find any standards specific to
> > CNAM.  The alternate idea similar to the yu draft seems to contradict
> > the statement that SIP isn't a database query protocol.  If anyone is
> > already doing something like this or has plans I'd love to hear about
> > it.
> >
> > John
> >
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Monday, February 02, 2004 11:29 PM
> > > To: Hearty, John
> > > Cc: sipping@ietf.org; Peterson, Jon
> > > Subject: Re: [Sipping] TCAP tunneling
> > >
> > > SIP isnt a database query protocol, and cnam is just a DB query
> > service.
> > > As such, I would think that enum would be an appropriate
> > technique for
> > > this. Any other DB query mechanism will do as well.
> > >
> > > Another idea would be to use a redirect service, similar to what has
> > > been proposed in the past for LNP:
> > >
> > > http://www.watersprings.org/pub/id/draft-yu-sip-np-02.txt
> > >
> > > except that a request for a number would be redirected to a Contact
> > > which contains that number plus the display name, representing the
> > > calling name.
> > >
> > > -Jonathan R.
> > >
> > > Hearty, John wrote:
> > > > Other than the miller draft below, is there any SIP standard
> > proposed to
> > > > carry cnam queries?
> > > >
> > > > John Hearty
> > > > Level3
> > > >
> > > >
> > > >>-----Original Message-----
> > > >>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On
> > Behalf
> > > >
> > > > Of
> > > >
> > > >>Arjun Roychowdhury
> > > >>Sent: Monday, February 02, 2004 7:25 AM
> > > >>To: Banibrata Dutta
> > > >>Cc: kagarwal@cedarpointcom.com; ranjit.avasarala@wipro.com;
> > > >>sipping@ietf.org; sipping-admin@ietf.org
> > > >>Subject: RE: [Sipping] TCAP tunneling
> > > >>
> > > >>Actually, there _is_ an independant draft on how to carry TCAP in
> > SIP.
> > > >>http://www.ietf.org/internet-drafts/draft-miller-sip-tcap-00.txt
> > > >>
> > > >>Its another matter whether this is the right thing to do or not
> > > >
> > > > (please
> > > >
> > > >>refer to old archives for the debates).
> > > >>Kaitaki, you may want to read the above draft as well as the
> > > >
> > > > discussions
> > > >
> > > >>on it to conclude for yourself which is the preferred
> > method in your
> > > >>deployment scenario.
> > > >>
> > > >>regds
> > > >>arjun
> > > >>
> > > >>
> > > >>--
> > > >>Arjun Roychowdhury @ Hughes Software Systems
> > > >>11717 Exploration Lane, Germantown MD 20876
> > > >>(O): 301 212 7860
> > > >>VOIP: (SIP) 10303@fwd.pulver.com (PSTN alias:360-207-0467)
> > > >>
> > > >>
> > > >>
> > > >>
> > > >>"Banibrata Dutta" <dutta@india.hp.com>
> > > >>Sent by: sipping-admin@ietf.org
> > > >>02/02/2004 12:25 AM
> > > >>
> > > >>
> > > >>        To:     <ranjit.avasarala@wipro.com>,
> > > >
> > > > <kagarwal@cedarpointcom.com>,
> > > >
> > > >><sipping@ietf.org>
> > > >>        cc:
> > > >>        Subject:        RE: [Sipping] TCAP tunneling
> > > >>
> > > >>
> > > >>hi,
> > > >>
> > > >>franly, i'd say that, it depends on your motivation/reason to use
> > SIP
> > > >>for
> > > >>tunneling TCAP. of course you can tunnel TCAP, but why you'd want
> > to,
> > > >
> > > > is
> > > >
> > > >>a justification you'd have to work out. AFAIK, there's not
> > ietf-draft
> > > >>that
> > > >>says, how.
> > > >>
> > > >>you can obviously tunnel TCAP query in an INVITE or INFO for that
> > > >
> > > > matter
> > > >
> > > >>(ideally in INVITE s.t. a SIP-dialog is also created which may be
> > used
> > > >>to
> > > >>model the TCAP dialog as well, perhaps!). the ASN.1 BER
> > encoded TCAP
> > > >>messages may then reside within the body of a multi-part
> > (MIME) enc,
> > > >>INVITE.
> > > >>
> > > >>infact 1 reason i see where this may make sense is that you would
> > > >
> > > > rather
> > > >
> > > >>not have two stacks s.a. SIGTRAN and SIP for signalling transport,
> > and
> > > >>use
> > > >>SIP as the *only* signalling solution, i.e. for the signalling
> > > >>application
> > > >>as well as for signalling transport, and support
> > legacy/TCAP apps by
> > > >>tunneling
> > > >>them within SIP. but then, you generally have the constraints of
> > > >>not-enough-
> > > >>space, (for 2 stacks) only on resource-limited devices, s.a.
> > > >
> > > > hand-sets,
> > > >
> > > >>embedded devices etc., but i am unable see (yet), how TCAP can be
> > > >>related to
> > > >>such devices.
> > > >>
> > > >>my 2 cents,
> > > >>Banibrata Dutta,
> > > >>  Hewlett-Packard ISO (P) Ltd,
> > > >>  29, Cunningham Road,
> > > >>  Bangalore - 560052
> > > >>  Ph: +91-80-22051796
> > > >>
> > > >>
> > > >>>-----Original Message-----
> > > >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > >>>On Behalf Of ranjit.avasarala@wipro.com
> > > >>>Sent: Monday, February 02, 2004 10:20 AM
> > > >>>To: kagarwal@cedarpointcom.com; sipping@ietf.org
> > > >>>Subject: RE: [Sipping] TCAP tunneling
> > > >>>
> > > >>>
> > > >>>Hi
> > > >>>   TCAP is used for Transaction Capabilities i.e database
> > > >>>related while SIP with ISUP tunneling is used between MGCs.
> > > >>>So I don't think TCAP can be tunneled in SIP.
> > > >>>
> > > >>>Ranjit
> > > >>>Ph: 8520408/16 extn: 2173
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>>-----Original Message-----
> > > >>>From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > >>>On Behalf Of Kaitki Agarwal
> > > >>>Sent: Monday, February 02, 2004 8:28 AM
> > > >>>To: sipping@ietf.org
> > > >>>Subject: [Sipping] TCAP tunneling
> > > >>>
> > > >>>
> > > >>>Hi
> > > >>> Can we tunnel tcap message in SIP as we do  ISUP. If not why.
> > > >>>
> > > >>>Thanks
> > > >>>Kaitki
> > > >>>Jj~Yzy=i~Rz}y
> > > >>>
> > > >>>_______________________________________________
> > > >>>Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > >>>This list is for NEW development of the application of SIP
> > > >>>Use sip-implementors@cs.columbia.edu for questions on current
> > > >>>sip Use sip@ietf.org for new developments of core SIP
> > > >>>
> > > >>
> > > >>
> > > >>_______________________________________________
> > > >>Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > >>This list is for NEW development of the application of SIP
> > > >>Use sip-implementors@cs.columbia.edu for questions on current sip
> > > >>Use sip@ietf.org for new developments of core SIP
> > > >>
> > > >>
> > > >>
> > > >>
> > > >>_______________________________________________
> > > >>Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > >>This list is for NEW development of the application of SIP
> > > >>Use sip-implementors@cs.columbia.edu for questions on current sip
> > > >>Use sip@ietf.org for new developments of core SIP
> > > >
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > > >
> > >
> > > --
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



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Date: Thu, 12 Feb 2004 12:07:54 -0500
From: David Oran <oran@cisco.com>
To: "Drage, Keith (Keith)" <drage@lucent.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Message-ID: <2147483647.1076587674@[10.32.245.153]>
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--On Thursday, February 12, 2004 1:54 PM +0000 "Drage, Keith (Keith)" 
<drage@lucent.com> wrote:

> With a view of trying to borrow decisions from elsewhere.
>
> In 3GPP when the phone recognises that it is an emergency call, and needs
> to identify the type of emergency call, then it has access to signal
> categories with the following names (ref 3GPP TS 22.101):
>
> ------------------------------
> -	Police
> -	Ambulance
> -	Fire Brigade
> -	Marine Guard
> -	Mountain Rescue
> -	Spare, at least [three] different types
> -----------------------------
>
> which results in a coding accompanying the call establishment of (ref
> 3GPP TS 24.008 table 10.5.135d)
>
> --------------------------
> Emergency Service Category Value (octet 3)
> The meaning of the Emergency Category Value is derived from the following
> settings (Please see 3GPP TS 22.101 clause 8): Bit 1	Police
> Bit 2	Ambulance
> Bit 3	Fire Brigade
> Bit 4	Marine Guard
> Bit 5	Mountain Rescue
> Bits 6,7,8 are spare and set to "0"
>
> Mobile station may set one or more bits to "1"
> If more than one bit is set to "1", routing to a combined Emergency
> centre (e.g. ambulance and fire brigade in Japan) is required. If the MSC
> can not match the received service category to any of the emergency
> centres, it shall route the call to an operator defined default emergency
> centre.
>
> If no bit is set to "1", the MSC shall route the Emergency call to an
> operator defined default emergency centre -------------------------
>
> These terms are at least accepted by multiple SDOs in the international
> community for transposition into their national standards.
>
> I also note that 3GPP does have the capability of signalling multiple
> preferences - this left me wondering whether we would be better off
> trying to follow that, possibly with a single emergency uri, but with the
> possibility of signalling category as a URI parameter with a list of
> values. Any thoughts on this?
>
Actually, this seems like ideal fodder for caller prefs, no?

> regards
>
> Keith
>
>
>> -----Original Message-----
>> From: Andrew Allen [mailto:aallen@rim.net]
>> Sent: 11 February 2004 16:57
>> To: mhammer@cisco.com; sipping@ietf.org
>> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
>>
>>
>>
>>
>> How about sos.medical then since that outlines the type of
>> emergency. I agree that rescue may be confused with the others.
>>
>> Andrew
>> --------------------------
>> Andrew Allen
>> Manager Standards
>> Research In Motion Ltd
>> BlackBerry Mobile  +1 847 809 8636
>> European Mobile    +358 50 467 5870
>> http://www.rim.com/
>>
>> Sent from my BlackBerry Wireless Handheld
>>
>>
>> -----Original Message-----
>> From: Michael Hammer <mhammer@cisco.com>
>> To: sipping@ietf.org <sipping@ietf.org>
>> Sent: Wed Feb 11 10:45:29 2004
>> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
>>
>> At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
>> >         sos.fire                fire brigade
>> >         sos.rescue      ambulance (rescue)
>> >         sos.marine      marine guard
>> >         sos.police      police (law enforcement)
>> >         sos.mountain    mountain rescue
>>
>> The second sub-address seems vague to me.  Rescue could mean many
>> things.  Perhaps, I need to be rescued from a fashion
>> emergency. :)  Would
>> something like:
>>
>>          sos.medic        medical (doctor, nurse, paramedic team)
>>
>> be more descriptive?
>>
>> Also, with respect to registrations, I could see commercial interests
>> wanting to get into the act, such as an sos.fashion that I hinted at
>> above.  Is there some guidance necessary to IANA?  Do we
>> expect that the
>> IETF WGs will act as arbiters for this?
>>
>> Second, wrt location information reporting, is there any need for a
>> convention for providing a URL to which a receiving ECC can
>> SUBSCRIBE to
>> get location updates from either the UA or a network element
>> that is doing
>> triangulation.  I am wondering if something like the following can be
>> inserted into the emergency INVITE:
>>
>> Call-Info: <sips:triangulator@local-proxy.domain.com>
>> ;purpose=location
>>
>> Mike
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP





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



From exim@www1.ietf.org  Thu Feb 12 12:51:00 2004
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Thu, 12 Feb 2004 17:45:45 -0000
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I guess if we did things the way 3GPP treats circuited switched emergency calls, it would be along the lines of caller preferences, with a Request-Disposition header parameter indicating route-emergency or some such like (and a Proxy-Require header as specified in callerprefs). Any Request-URI would then be totally ignored, and we would not need an sos* uri to be defined at all.

My wondering below was as to whether we could define a single sos* uri, and deal with different types of emergency service providers by a URI-parameter attached to the Request-URI.

SIPPING, you are welcome to pull either of these to bits.

regards

Keith



> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: 12 February 2004 14:16
> To: Drage, Keith (Keith)
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
> 
> 
> Interesting. The selection sounds like caller preferences to me. The 
> nice thing about that is that you might be able to use existing 
> techniques to match up services and destinations.
> 
> For example, this could be considered a sip.class value, extending 
> 'home' and 'business', or a new media feature tag, either specific to 
> emergency services or a more generic service identifier. (The latter 
> does invite the fashion emergency and roadside repair 
> services into the 
> same namespace.)
> 
> Drage, Keith (Keith) wrote:
> 
> > With a view of trying to borrow decisions from elsewhere.
> > 
> > In 3GPP when the phone recognises that it is an emergency 
> call, and needs to identify the type of emergency call, then 
> it has access to signal categories with the following names 
> (ref 3GPP TS 22.101):
> > 
> > ------------------------------
> > -	Police
> > -	Ambulance
> > -	Fire Brigade
> > -	Marine Guard
> > -	Mountain Rescue
> > -	Spare, at least [three] different types
> > -----------------------------
> > 
> > which results in a coding accompanying the call 
> establishment of (ref 3GPP TS 24.008 table 10.5.135d)
> > 
> > --------------------------
> > Emergency Service Category Value (octet 3)
> > The meaning of the Emergency Category Value is derived from 
> the following settings (Please see 3GPP TS 22.101 clause 8):
> > Bit 1	Police
> > Bit 2	Ambulance
> > Bit 3	Fire Brigade
> > Bit 4	Marine Guard
> > Bit 5	Mountain Rescue
> > Bits 6,7,8 are spare and set to "0"
> > 
> > Mobile station may set one or more bits to "1"
> > If more than one bit is set to "1", routing to a combined 
> Emergency centre (e.g. ambulance and fire brigade in Japan) 
> is required. If the MSC can not match the received service 
> category to any of the emergency centres, it shall route the 
> call to an operator defined default emergency centre.
> > 
> > If no bit is set to "1", the MSC shall route the Emergency 
> call to an operator defined default emergency centre
> > -------------------------
> > 
> > These terms are at least accepted by multiple SDOs in the 
> international community for transposition into their national 
> standards.
> > 
> > I also note that 3GPP does have the capability of 
> signalling multiple preferences - this left me wondering 
> whether we would be better off trying to follow that, 
> possibly with a single emergency uri, but with the 
> possibility of signalling category as a URI parameter with a 
> list of values. Any thoughts on this?
> > 
> > regards
> > 
> > Keith
> > 
> > 
> > 
> >>-----Original Message-----
> >>From: Andrew Allen [mailto:aallen@rim.net]
> >>Sent: 11 February 2004 16:57
> >>To: mhammer@cisco.com; sipping@ietf.org
> >>Subject: Re: [Sipping] Concern within sos ID - user 
> addresses reserved
> >>
> >>
> >>
> >>
> >>How about sos.medical then since that outlines the type of 
> >>emergency. I agree that rescue may be confused with the others.
> >>
> >>Andrew
> >>--------------------------
> >>Andrew Allen
> >>Manager Standards
> >>Research In Motion Ltd
> >>BlackBerry Mobile  +1 847 809 8636
> >>European Mobile    +358 50 467 5870
> >>http://www.rim.com/
> >>
> >>Sent from my BlackBerry Wireless Handheld
> >>
> >>
> >>-----Original Message-----
> >>From: Michael Hammer <mhammer@cisco.com>
> >>To: sipping@ietf.org <sipping@ietf.org>
> >>Sent: Wed Feb 11 10:45:29 2004
> >>Subject: Re: [Sipping] Concern within sos ID - user 
> addresses reserved
> >>
> >>At 03:55 PM 2/10/2004 -0600, James M. Polk wrote:
> >>
> >>>        sos.fire                fire brigade
> >>>        sos.rescue      ambulance (rescue)
> >>>        sos.marine      marine guard
> >>>        sos.police      police (law enforcement)
> >>>        sos.mountain    mountain rescue
> >>
> >>The second sub-address seems vague to me.  Rescue could mean many 
> >>things.  Perhaps, I need to be rescued from a fashion 
> >>emergency. :)  Would 
> >>something like:
> >>
> >>         sos.medic        medical (doctor, nurse, paramedic team)
> >>
> >>be more descriptive?
> >>
> >>Also, with respect to registrations, I could see commercial 
> interests 
> >>wanting to get into the act, such as an sos.fashion that I 
> hinted at 
> >>above.  Is there some guidance necessary to IANA?  Do we 
> >>expect that the 
> >>IETF WGs will act as arbiters for this?
> >>
> >>Second, wrt location information reporting, is there any need for a 
> >>convention for providing a URL to which a receiving ECC can 
> >>SUBSCRIBE to 
> >>get location updates from either the UA or a network element 
> >>that is doing 
> >>triangulation.  I am wondering if something like the 
> following can be 
> >>inserted into the emergency INVITE:
> >>
> >>Call-Info: <sips:triangulator@local-proxy.domain.com> 
> >>;purpose=location
> >>
> >>Mike
> >>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >>
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> 

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



From exim@www1.ietf.org  Thu Feb 12 12:51:01 2004
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sipping@ietf.org
Subject: RE: [Sipping] Other concerns within sos ID
Date: Thu, 12 Feb 2004 17:45:45 -0000
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Not sure we want, or need, to go there.

We are meant to be defining the interface at the SIP level, and whether the buttons are little blue square ones or big red round ones does not make a lot of difference to that.

There are around 1 billion GSM phones out there, all of which are designed to do as much as possible to identify emergency calls in the terminal and make a dedicated emergency call. They all have the capability of implementing options of recognising that from dialled keys, menu buttons or special keys - but the user interface functionality is all not specified, and works perfectly well without being specified.

regards

Keith

> -----Original Message-----
> From: Tom Taylor [mailto:taylor@nortelnetworks.com]
> Sent: 11 February 2004 20:58
> To: Michael Hammer
> Cc: Mpierce1@aol.com; oran@cisco.com; jmpolk@cisco.com; 
> sipping@ietf.org
> Subject: Re: [Sipping] Other concerns within sos ID
> 
> 
> The human factors aspect of this stuff has traditionally been 
> the domain of the 
> ITU-T.  I have a feeling they should be brought into the 
> discussion.  I would have 
> to check, but the relevant experts used to be located in 
> Study Group 2.
> 
> Michael Hammer wrote:
> 
> > One additional question that was touched on in this email:
> > 
> > If an sos URI contains a domain name, I presume that a proxy will 
> > forward the request based on that domain name and never look at the 
> > local part containing the "sos.xxx".  Is that correct?
> > 
> > If so, does that mean that the proxy in that domain is 
> expected to look 
> > at the location information and forward to the appropriate 
> ECC proxy?
> > 
> > As for internationalization, that could get out of hand.  
> Was the intent 
> > that these URIs are machine readable only?  I suspect that some 
> > internationally recognized icons on the user agents that 
> translate into 
> > these URIs will be needed.  But, that could be tricky as well:
> > 
> > sos.fire     picture of flames
> > sos.rescue   picture of Red Cross/Crescent?
> > sos.police   picture of badge/police belt?
> > sos.marine   picture of boat + cross/crescent?
> > sos.mountain picture of mountain + cross/crescent?
> > 
> > But those icons are phone implementation details.
> > 
> > Mike
> > 
> > 
> > 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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Hearty, John wrote:

> In looking at the ENUM RFC, it looks like NAPTR records are used, and
> they convert E.164 numbers to URIs.  The syntax for the URIs does not
> appear to allow for Display Names, which is what I believe needs to be
> populated when forwarding SIP messages.

A URI can represent just about anything. I didnt find a URN namespace 
defined for containing people names; I suspect this is because of 
difficulty in uniqueness. But, in principle one could do:

urn:people:Jonathan.D.Rosenberg

In this case you would need to define an enumservices that maps phone 
numbers to people URNs.


>  I suppose we could still do TXT
> type DNS records for this, but can't find any standards specific to
> CNAM.  The alternate idea similar to the yu draft seems to contradict
> the statement that SIP isn't a database query protocol. 

Well, let me clarify. SIP is not itself a database protocol. However, 
when a DB query is involved in the process of proxying a request, you 
can access a SIP service that effectively performs a proxy operation to 
the result of the DB lookup.

Fardad writes:
> CNAM is a bit different since we are dealing with the calling party
> information. I am a bit confused about Jonathan's comment regarding adding
> the CNAM information to the Contact header. It seems that the right approach
> would be for Proxies to add a PAI with display name.  Looking at RFC3325, it
> seems that PAI can also be present in 3XX responses.  So potentially
> redirectors can also add CNAM information by adding a PAI.  

Redirectors would not add a PAI; it says the wrong thing. It would 
assert the identity of the element generating the response, which is not 
what you want here. The redirect would look like this:

SIP/2.0 301 Name Added
Contact: 
sip:john@example.edu?P-Asserted-ID=John%20Example%20<john@example.com>

In other words, the redirected-to URI contains a header parameter, which 
the "requesting" proxy would automatically populate into the outgoing 
request. Of course it would need to trust the source of the header, as a 
proxy should not normally accept a redirect with a p-asserted-id.

Thanks,
Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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In a message dated 2/12/2004 8:58:35 AM Eastern Standard Time, 
drage@lucent.com writes:


> ... I also note that 3GPP does have the capability of signalling multiple 
preferences - this left me wondering whether we would be better off trying to 
follow that, 
> possibly with a single emergency uri, but with the possibility of 
> signalling category as a URI parameter with a list of values. Any thoughts on this?
> 
It's always good to use other standards as the base. This list of 
"subadresses" is good. But I still don't see how a 3GPP phone can translate number dialed 
into one of these, unless it knows the local dial plan, or knows all dial 
plans and knows its location.

Does the 3GPP spec mean that, if someone dials 911 in the US, that all bits 
must be set?

Mike Pierce



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/12/2004 8:58:35 AM Eastern Standard Time, drage@lucent.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">...</FONT><FONT  COLOR=3D"#=
000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COLOR: #ffffff" SIZE=3D3 PTSIZE=
=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0"></BLOCKQUOTE> </FONT><F=
ONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COLOR: #ffffff"=20=
SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0">I also n=
ote that 3GPP does have the capability of signalling multiple preferences -=20=
this left me wondering whether we would be better off trying to follow that,=
 </FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COLOR:=
 #ffffff" SIZE=3D3 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=3D"0=
">
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARG=
IN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">possibly with a single e=
mergency uri, but with the possibility of signalling category as a URI param=
eter with a list of values. Any thoughts on this?
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">It's always good to use other standards as the base. This list of "su=
badresses" is good. But I still don't see how a 3GPP phone can translate num=
ber dialed into one of these, unless it knows the local dial plan, or knows=20=
all dial plans and knows its location.
<BR>
<BR>Does the 3GPP spec mean that, if someone dials 911 in the US, that all b=
its must be set?
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

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Date: Thu, 12 Feb 2004 15:52:01 EST
Subject: Re: [Sipping] Other concerns within sos ID
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In a message dated 2/11/2004 2:18:51 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


In general, I think the correct long-term user interface solution is to 
have a hotkey or address book entry *with user confirmation* rather than 
having users trying to remember digit strings. The digit string model 
currently works rather badly for the PSTN - do you know, without looking 
at the hotel phone card insert, whether you need to dial 9-911, 0, 911, 
0-911 or some other combination to get emergency help in a US hotel, let 
alone in some hotel in a foreign country?


[MAP] The digit string model works badly only when there is a lack of 
standardization. The solution is to push for standards, which has been happening in 
the US for many years. I suspect that "9" is prettty much "standard" in the US 
to get an outside line from a PBX, and "911" is pretty much "standard" in the 
US, so 9-911 almost always works. (I believe some jurisdictions have mandated 
this operation.) It's easy for a PBX to also interpret 911 itself as meaning 
the same thing, since no outside number begins with "11" in the US as far as I 
know, and the PBX can't have an internal number of "911" if "9" gets an 
external line. I don't know if any PBXs do this. (I'm not going to try!)

The solution to lack of full standardization is not to create yet another 
system, and I think the emergency industry (and US regulators) would have a 
problem with this.

If the user's terminal has a hotkey for emergency, what possibilities are 
there for a "user confirmation"? A voice message that says: "If you really meant 
it, push the button again harder"? Or if you click on an icon, a confirmation 
window pops up asking if you really meant it? I really can't see any 
"confirmation" procedure as being better than simply dialing the 3-digit emergency code 
(or "9" plus the 3-digits) that the government has spent millions 
publicizing. Any such "confirmation" procedure would have serious accessibility issues 
(for blind, deaf, illiterate, etc.)

I don't think that a hotkey or address book entry, even with a "user 
confirmation", is the correct long-term user interface solution. "Remembering" digit 
strings for an emergency is not a problem that needs solving.

On the other hand, if foreign travellers in the US not knowing our emergency 
number were really a problem, then it should be solved (in the US) by pushing 
for acceptance of 112, 118, 999 and maybe a few others as emergency numbers in 
our dial plan, even without SIP phones. This is not solely an IP phone or SIP 
issue.

Mike Pierce



--part1_1df.195c1e24.2d5d4171_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/11/2004 2:18:51 PM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR>In general, I think the correct long-term user interface solution is to=20
<BR>have a hotkey or address book entry *with user confirmation* rather than=
=20
<BR>having users trying to remember digit strings. The digit string model=20
<BR>currently works rather badly for the PSTN - do you know, without looking=
=20
<BR>at the hotel phone card insert, whether you need to dial 9-911, 0, 911,=20
<BR>0-911 or some other combination to get emergency help in a US hotel, let=
=20
<BR>alone in some hotel in a foreign country?
<BR>
<BR>
<BR>[MAP] The digit string model works badly only when there is a lack of st=
andardization. The solution is to push for standards, which has been happeni=
ng in the US for many years. I suspect that "9" is prettty much "standard" i=
n the US to get an outside line from a PBX, and "911" is pretty much "standa=
rd" in the US, so 9-911 almost always works. (I believe some jurisdictions h=
ave mandated this operation.) It's easy for a PBX to also interpret 911 itse=
lf as meaning the same thing, since no outside number begins with "11" in th=
e US as far as I know, and the PBX can't have an internal number of "911" if=
 "9" gets an external line. I don't know if any PBXs do this. (I'm not going=
 to try!)
<BR>
<BR>The solution to lack of full standardization is not to create yet anothe=
r system, and I think the emergency industry (and US regulators) would have=20=
a problem with this.
<BR>
<BR>If the user's terminal has a hotkey for emergency, what possibilities ar=
e there for a "user confirmation"? A voice message that says: "If you really=
 meant it, push the button again harder"? Or if you click on an icon, a conf=
irmation window pops up asking if you really meant it? I really can't see an=
y "confirmation" procedure as being better than simply dialing the 3-digit e=
mergency code (or "9" plus the 3-digits) that the government has spent milli=
ons publicizing. Any such "confirmation" procedure would have serious access=
ibility issues (for blind, deaf, illiterate, etc.)
<BR>
<BR>I don't think that a hotkey or address book entry, even with a "user con=
firmation", is the correct long-term user interface solution. "Remembering"=20=
digit strings for an emergency is not a problem that needs solving.
<BR>
<BR>On the other hand, if foreign travellers in the US not knowing our emerg=
ency number were really a problem, then it should be solved (in the US) by p=
ushing for acceptance of 112, 118, 999 and maybe a few others as emergency n=
umbers in our dial plan, even without SIP phones. This is not solely an IP p=
hone or SIP issue.
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

--part1_1df.195c1e24.2d5d4171_boundary--

_______________________________________________
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Subject: Re: [Sipping] Other concerns within sos ID
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> If the user's terminal has a hotkey for emergency, what possibilities 
> are there for a "user confirmation"? A voice message that says: "If you 
> really meant it, push the button again harder"? Or if you click on an 
> icon, a confirmation window pops up asking if you really meant it? I 
> really can't see any "confirmation" procedure as being better than 
> simply dialing the 3-digit emergency code (or "9" plus the 3-digits) 
> that the government has spent millions publicizing. Any such 
> "confirmation" procedure would have serious accessibility issues (for 
> blind, deaf, illiterate, etc.)

This is a UI issue, not for the IETF to solve. Confirmation dialogs, in 
many forms, are common. Indeed, there are many existing phones that have 
such buttons: Google randomly found 
http://www.allphones.us/shop/shopexd.asp?id=180
and
http://www.gogotech.com/flypage/1086/Ameriphone_Emergency_Response_Telephones
http://www.gogotech.com/flypage/1574/none/Ameriphone_JB-20

The confirmation problem seems to be primarily one for cell phones stuck 
in back pockets.


> 
> I don't think that a hotkey or address book entry, even with a "user 
> confirmation", is the correct long-term user interface solution. 
> "Remembering" digit strings for an emergency is not a problem that needs 
> solving.

The discussion was not about replacing 911 in your home domain. There is 
no disagreement that this should be available. Quick, what's the 
emergency number for Korea?

> 
> On the other hand, if foreign travellers in the US not knowing our 
> emergency number were really a problem, then it should be solved (in the 
> US) by pushing for acceptance of 112, 118, 999 and maybe a few others as 
> emergency numbers in our dial plan, even without SIP phones. This is not 
> solely an IP phone or SIP issue.

The current technical solution seems much better and much more likely.


> 
> Mike Pierce
> 

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



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Hi folks,
  ... and, from my explorations, people commonly use TXT records for 
names.
It's a hack but it works for human displayable items.
Anything more complex (like URN stuff) maybe cleaner but requires more 
work.

BTW, often the TXT records I see are 'cos folk haven't set their "split 
horizon"
correctly in BIND.

Number -> is a major privacy issue for many, and this will be an issue 
whatever
scheme one uses; NAPTRs use/are DNS so a 'people' URN NAPTR doesn't 
help.

Just my 14 centimes.

atb,
   Lawrence

On 12 Feb 2004, at 8:39 pm, Jonathan Rosenberg wrote:

>
>
> Hearty, John wrote:
>
>> In looking at the ENUM RFC, it looks like NAPTR records are used, and
>> they convert E.164 numbers to URIs.  The syntax for the URIs does not
>> appear to allow for Display Names, which is what I believe needs to be
>> populated when forwarding SIP messages.
>
> A URI can represent just about anything. I didnt find a URN namespace 
> defined for containing people names; I suspect this is because of 
> difficulty in uniqueness. But, in principle one could do:
>
> urn:people:Jonathan.D.Rosenberg
>
> In this case you would need to define an enumservices that maps phone 
> numbers to people URNs.
>
>
>>  I suppose we could still do TXT
>> type DNS records for this, but can't find any standards specific to
>> CNAM.  The alternate idea similar to the yu draft seems to contradict
>> the statement that SIP isn't a database query protocol.
>
> Well, let me clarify. SIP is not itself a database protocol. However, 
> when a DB query is involved in the process of proxying a request, you 
> can access a SIP service that effectively performs a proxy operation 
> to the result of the DB lookup.
>
> Fardad writes:
>> CNAM is a bit different since we are dealing with the calling party
>> information. I am a bit confused about Jonathan's comment regarding 
>> adding
>> the CNAM information to the Contact header. It seems that the right 
>> approach
>> would be for Proxies to add a PAI with display name.  Looking at 
>> RFC3325, it
>> seems that PAI can also be present in 3XX responses.  So potentially
>> redirectors can also add CNAM information by adding a PAI.
>
> Redirectors would not add a PAI; it says the wrong thing. It would 
> assert the identity of the element generating the response, which is 
> not what you want here. The redirect would look like this:
>
> SIP/2.0 301 Name Added
> Contact: 
> sip:john@example.edu?P-Asserted-ID=John%20Example%20<john@example.com>
>
> In other words, the redirected-to URI contains a header parameter, 
> which the "requesting" proxy would automatically populate into the 
> outgoing request. Of course it would need to trust the source of the 
> header, as a proxy should not normally accept a redirect with a 
> p-asserted-id.
>
> Thanks,
> Jonathan R.
>
>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>

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Subject: [Sipping] Translating  ISDN IE or SS7 optional fields to SIP
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Is there a document that defines (or proposes) a standard way to map 
protocol specific information into SIP to be consumed by a UA?   SIP-T 
provides  the entire SS7 packet embedded in the SIP message rather than a 
translation of the fields.   I am hoping for a mapping like

redirection-number <==> redir-num: SIP header field

Is there such a document?

                                        Thanks for any info,
                                        Joe H.

--
Joe Herman
IBM Annapolis RM Telephony Solutions
410-571-7117
dzoey@us.ibm.com

Everything is wonderful until you know something about it.

--=_alternative 007ECB9A85256E38_=
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<br><font size=2 face="sans-serif">Is there a document that defines (or
proposes) a standard way to map protocol specific information into SIP
to be consumed by a UA? &nbsp; SIP-T provides &nbsp;the entire SS7 packet
embedded in the SIP message rather than a translation of the fields. &nbsp;
I am hoping for a mapping like</font>
<br>
<br><font size=2 face="sans-serif">redirection-number &lt;==&gt; redir-num:
SIP header field</font>
<br>
<br><font size=2 face="sans-serif">Is there such a document?</font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Thanks
for any info,</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Joe H.</font>
<br>
<br><font size=2 face="sans-serif">--<br>
Joe Herman<br>
IBM Annapolis RM Telephony Solutions<br>
410-571-7117<br>
dzoey@us.ibm.com<br>
<br>
Everything is wonderful until you know something about it.<br>
</font>
--=_alternative 007ECB9A85256E38_=--

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Drage, Keith (Keith) wrote:

> I guess if we did things the way 3GPP treats circuited switched
> emergency calls, it would be along the lines of caller preferences,
> with a Request-Disposition header parameter indicating
> route-emergency or some such like (and a Proxy-Require header as
> specified in callerprefs). 

Request disposition is purposefully non-extensible, so that is not the 
place. This sounds like a routing problem to me, in which case the 
traditional Accept-Contact is the right place. The r-uri would remain 
plain old sos.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Thu, 12 Feb 2004 22:57:52 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: "Drage, Keith (Keith)" <drage@lucent.com>, sipping@ietf.org
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
References: <475FF955A05DD411980D00508B6D5FB00B1B1F94@en0033exch001u.uk.lucent.com> <402C468B.5060709@dynamicsoft.com>
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> Request disposition is purposefully non-extensible, so that is not the 
> place. This sounds like a routing problem to me, in which case the 
> traditional Accept-Contact is the right place. The r-uri would remain 
> plain old sos.

That's what I had in mind. Question is whether there should be a generic 
new 'sip.service' parameter, a more specific emergency-service parameter 
or whether this should be folded as additional values into 'class'. My 
inclination would be the first, with a hiearchical scheme. I'm not sure 
whether there are other services, beyond emergency services, that would 
or should use this.

_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Banibrata Dutta" <dutta@india.hp.com>
To: "'Joseph Herman'" <jiherman@us.ibm.com>, <sipping@ietf.org>
Subject: RE: [Sipping] Translating  ISDN IE or SS7 optional fields to SIP
Date: Fri, 13 Feb 2004 10:06:04 +0530
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One quick way would be to check that out by searching the ID-list hosted
at the
very useful http://softarmor.com/sipping/ site.
AFAIK, such a draft doesn't exist. SIP-T is what comes closest (i
think).
 
-BDutta.

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Joseph Herman
Sent: Friday, February 13, 2004 4:31 AM
To: sipping@ietf.org
Subject: [Sipping] Translating ISDN IE or SS7 optional fields to SIP



Is there a document that defines (or proposes) a standard way to map
protocol specific information into SIP to be consumed by a UA?   SIP-T
provides  the entire SS7 packet embedded in the SIP message rather than
a translation of the fields.   I am hoping for a mapping like 

redirection-number <==> redir-num: SIP header field 

Is there such a document? 

                                        Thanks for any info, 
                                        Joe H. 

--
Joe Herman
IBM Annapolis RM Telephony Solutions
410-571-7117
dzoey@us.ibm.com

Everything is wonderful until you know something about it.



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	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D203073204-13022004><FONT face=3DArial color=3D#0000ff =
size=3D2>One=20
quick way would be to check that out&nbsp;by searching&nbsp;the ID-list =
hosted=20
at the</FONT></SPAN></DIV>
<DIV><SPAN class=3D203073204-13022004><FONT face=3DArial color=3D#0000ff =
size=3D2>very=20
useful&nbsp;<A=20
href=3D"http://softarmor.com/sipping/">http://softarmor.com/sipping/</A>&=
nbsp;site.</FONT></SPAN></DIV>
<DIV><SPAN class=3D203073204-13022004><FONT face=3DArial color=3D#0000ff =
size=3D2>AFAIK,=20
such a draft doesn't exist. SIP-T is what comes closest (i=20
think).</FONT></SPAN></DIV>
<DIV><SPAN class=3D203073204-13022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D203073204-13022004><FONT face=3DArial color=3D#0000ff =

size=3D2>-BDutta.</FONT></SPAN></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] <B>On Behalf Of =

  </B>Joseph Herman<BR><B>Sent:</B> Friday, February 13, 2004 4:31=20
  AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] =
Translating=20
  ISDN IE or SS7 optional fields to SIP<BR><BR></FONT></DIV><BR><FONT=20
  face=3Dsans-serif size=3D2>Is there a document that defines (or =
proposes) a=20
  standard way to map protocol specific information into SIP to be =
consumed by a=20
  UA? &nbsp; SIP-T provides &nbsp;the entire SS7 packet embedded in the =
SIP=20
  message rather than a translation of the fields. &nbsp; I am hoping =
for a=20
  mapping like</FONT> <BR><BR><FONT face=3Dsans-serif =
size=3D2>redirection-number=20
  &lt;=3D=3D&gt; redir-num: SIP header field</FONT> <BR><BR><FONT =
face=3Dsans-serif=20
  size=3D2>Is there such a document?</FONT> <BR><BR><FONT =
face=3Dsans-serif=20
  size=3D2>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Thanks=20
  for any info,</FONT> <BR><FONT face=3Dsans-serif size=3D2>&nbsp; =
&nbsp; &nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;=20
  &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Joe H.</FONT> <BR><BR><FONT=20
  face=3Dsans-serif size=3D2>--<BR>Joe Herman<BR>IBM Annapolis RM =
Telephony=20
  Solutions<BR>410-571-7117<BR>dzoey@us.ibm.com<BR><BR>Everything is =
wonderful=20
  until you know something about =
it.<BR></BLOCKQUOTE></FONT></BODY></HTML>

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: Re: [Sipping] Translating  ISDN IE or SS7 optional fields to SIP
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RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to
Session Initiation Protocol (SIP) Mapping.
This RFC describes a way to perform the mapping between SIP and ISUP
(message and parameter level).

-
Elias
Hughes Software Systems
http://www.hssworld.com
-



                                                                           
             Joseph Herman                                                 
             <jiherman@us.ibm.                                             
             com>                                                       To 
             Sent by:                  sipping@ietf.org                    
             sipping-admin@iet                                          cc 
             f.org                                                         
                                                                   Subject 
                                       [Sipping] Translating  ISDN IE or   
             02/13/04 04:31 AM         SS7 optional fields to SIP          
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           





Is there a document that defines (or proposes) a standard way to map
protocol specific information into SIP to be consumed by a UA?   SIP-T
provides  the entire SS7 packet embedded in the SIP message rather than a
translation of the fields.   I am hoping for a mapping like

redirection-number <==> redir-num: SIP header field

Is there such a document?

                                        Thanks for any info,
                                        Joe H.

--
Joe Herman
IBM Annapolis RM Telephony Solutions
410-571-7117
dzoey@us.ibm.com

Everything is wonderful until you know something about it.


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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Folks,

We have submitted a set of drafts on exploders. The following one 
provides the requirements we are using and a framework to understand the 
rest of the drafts.

http://users.piuha.net/gonzalo/draft-camarillo-sipping-exploders-02.txt

The rest of them can be fetched (until they appear in the archives) from:

http://users.piuha.net/gonzalo/draft-camarillo-sipping-uri-list-01.txt
http://users.piuha.net/gonzalo/draft-camarillo-sipping-adhoc-management-00.txt
http://users.piuha.net/gonzalo/draft-camarillo-sipping-transac-package-00.txt
http://users.piuha.net/gonzalo/draft-camarillo-sipping-multiple-refer-00.txt

Commments are appreciated,

Thanks,

Gonzalo

 

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Hello,

> Why do you need a heavy-weight light B2BUA for "exploding". 

Do you mean light *or* heavy-weight?

In any case, you do not want to use a proxy because the services these 
exploders provide (e.g., conferencing) are provided by B2BUAs, not by 
proxies.

Gonzalo

 

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Subject: RE: [Sipping] Drafts on exploders
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Hi
    I have one doubt here: so u mean to say that B2BUA behaves as a
conference mixer?

Thanks
Ranjit
Ph: 8520408/16 extn: 2173




-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Gonzalo Camarillo
Sent: Friday, February 13, 2004 3:09 PM
To: Banibrata Dutta
Cc: 'sipping'
Subject: Re: [Sipping] Drafts on exploders


Hello,

> Why do you need a heavy-weight light B2BUA for "exploding".

Do you mean light *or* heavy-weight?

In any case, you do not want to use a proxy because the services these=20
exploders provide (e.g., conferencing) are provided by B2BUAs, not by=20
proxies.

Gonzalo

=20

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To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
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Subject: RE: [Sipping] Drafts on exploders
Date: Fri, 13 Feb 2004 15:33:46 +0530
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Hi Gonzalo,

  I am sorry, I meant "heavy-weight" only. Okay, isn't then exploder
itself a service, but yes it would become a feature they may be used
by several services. Confercing is then just another service that uses
the exploder feature, right?

  Won't it be better to take that responsibility out of a B2BUA and
put it in the core network (in the proxies) s.t. the explosion happens
where it is needed? Let me explain my view as follows:-

there are 2 ways:-

1) A single To: header which explodes to multiple INVITES (or other
   method messages).
2) A single INVITE with multiple To: headers which is understood by
   a router and split off into multiple INVITE's with single To:
   headers.

in the first case the exploding proxy (or entity) needs access to a
explosion database (i.e. group-name -> list of members). this "may be"
achieved by DNS mechanisms (i am not even close to being a DNS
expert, but i think that it can be done, the same way SOS URI's
are transformed into actual ERS/PCAP #s).

in the second case the exploding proxy doesn't need a explosion database
and the original INVITE anyway had all the info needed to split it off
into multiple INVITEs with single To: headers. 

with these approaches the benefit of "localization" of explosion can be 
used, i.e. the explosion happens as late as possible, leading to the
minimum # of messages in the network, i.e. multiple exploder proxies
would exist in the network and it may so happen that the orginal
invite with 10 To: headers go split into 2 INVITES with 6 and 4 To:
headers respectively at an exploder proxy, and then the INVITE with
6 To:s got split into 2 with 3 To:s each and so on. this way the
explosion is "Just-In-Time" and minimal.

regards,
banibrata dutta.

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of Gonzalo Camarillo
> Sent: Friday, February 13, 2004 3:09 PM
> To: Banibrata Dutta
> Cc: 'sipping'
> Subject: Re: [Sipping] Drafts on exploders
> 
> 
> Hello,
> 
> > Why do you need a heavy-weight light B2BUA for "exploding".
> 
> Do you mean light *or* heavy-weight?
> 
> In any case, you do not want to use a proxy because the 
> services these 
> exploders provide (e.g., conferencing) are provided by B2BUAs, not by 
> proxies.
> 
> Gonzalo
> 
>  
> 
> This communication is confidential and intended solely for 
> the addressee(s). Any unauthorized review, use, disclosure or 
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> 


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Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
Date: Fri, 13 Feb 2004 10:15:23 -0000
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Thread-Topic: [Sipping] Concern within sos ID - user addresses reserved
Thread-Index: AcPx5eILNuEsXWGZTw2+GqaXvQWE/gANB/1A
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Drage, Keith (Keith)" <drage@lucent.com>, <sipping@ietf.org>
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My inclination would be to have a specific emergency service parameter
to avoid any confusion in usage.  That way we explicitly define
emergency routing without confusion.

Chris.


>-----Original Message-----
>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>Sent: 13 February 2004 03:58
>To: Jonathan Rosenberg
>Cc: Drage, Keith (Keith); sipping@ietf.org
>Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
>
>
>> Request disposition is purposefully non-extensible, so that is not
the
>> place. This sounds like a routing problem to me, in which case the
>> traditional Accept-Contact is the right place. The r-uri would remain
>> plain old sos.
>
>That's what I had in mind. Question is whether there should be a
generic
>new 'sip.service' parameter, a more specific emergency-service
parameter
>or whether this should be folded as additional values into 'class'. My
>inclination would be the first, with a hiearchical scheme. I'm not sure
>whether there are other services, beyond emergency services, that would
>or should use this.
>
>_______________________________________________
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Banibrata Dutta <dutta@india.hp.com>
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Subject: Re: [Sipping] Drafts on exploders
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Banibrata,

The services we intend to provide using exploders are implemented using 
B2BUAs. If the services you have in mind can be implemented using only 
proxies, you do not need to use the mechanisms described in these drafts.

Thanks,

Gonzalo



Banibrata Dutta wrote:

> Hi Gonzalo,
> 
>   I am sorry, I meant "heavy-weight" only. Okay, isn't then exploder
> itself a service, but yes it would become a feature they may be used
> by several services. Confercing is then just another service that uses
> the exploder feature, right?
> 
>   Won't it be better to take that responsibility out of a B2BUA and
> put it in the core network (in the proxies) s.t. the explosion happens
> where it is needed? Let me explain my view as follows:-
> 
> there are 2 ways:-
> 
> 1) A single To: header which explodes to multiple INVITES (or other
>    method messages).
> 2) A single INVITE with multiple To: headers which is understood by
>    a router and split off into multiple INVITE's with single To:
>    headers.
> 
> in the first case the exploding proxy (or entity) needs access to a
> explosion database (i.e. group-name -> list of members). this "may be"
> achieved by DNS mechanisms (i am not even close to being a DNS
> expert, but i think that it can be done, the same way SOS URI's
> are transformed into actual ERS/PCAP #s).
> 
> in the second case the exploding proxy doesn't need a explosion database
> and the original INVITE anyway had all the info needed to split it off
> into multiple INVITEs with single To: headers. 
> 
> with these approaches the benefit of "localization" of explosion can be 
> used, i.e. the explosion happens as late as possible, leading to the
> minimum # of messages in the network, i.e. multiple exploder proxies
> would exist in the network and it may so happen that the orginal
> invite with 10 To: headers go split into 2 INVITES with 6 and 4 To:
> headers respectively at an exploder proxy, and then the INVITE with
> 6 To:s got split into 2 with 3 To:s each and so on. this way the
> explosion is "Just-In-Time" and minimal.
> 
> regards,
> banibrata dutta.

 

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, sipping@ietf.org
Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
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Depending on which 3GPP release you get there are a number of mechanisms.
 
1)    Certain numbers are built into the phone and recognised there.
 
2)    Certain numbers are preloaded onto the SIM/UICC card by the operator at subscription time.
 
3)    Numbers can be downloaded to the phone at registration time from the local providing access network.
 
These are identified in the various versions of 22.101 (see http://www.3gpp.org/ftp/Specs/html-info/22101.htm <http://www.3gpp.org/ftp/Specs/html-info/22101.htm> ) clause 10.
 
I am not sure what terminal manufacturers do in the case of setting the bits for phones supplied for use in the US, however inspection of the 3GPP TS 24.008 text says that in the US, whatever the bits are set to, or not set at all, the call will be routed to an emergency centre. Note that this category element does not determine an emegency call, only the category of an emergency call. A call identified by the terminal as an emergency call will be sent with an EMERGENCY SETUP message rather than a SETUP message. Special characteristics of the EMERGENCY SETUP message include ignoring any called party number sent in that message.
 
Note that none of the above mechanisms can guarantee that an emergency call is always identified as such at the phone, and therefore there is also a responsibility on the local serving network to do its own analysis as well. In the 3GPP SIP environment, that responsibility may well extend to other proxies, with the need to reroute back to the country of origin (all calls carry the mobile country code and mobile network code as well as potentially geographic information).
 
regards
 
Keith

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: 12 February 2004 20:52
To: drage@lucent.com; sipping@ietf.org
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved


In a message dated 2/12/2004 8:58:35 AM Eastern Standard Time, drage@lucent.com writes: 




...

I also note that 3GPP does have the capability of signalling multiple preferences - this left me wondering whether we would be better off trying to follow that, 


possibly with a single emergency uri, but with the possibility of signalling category as a URI parameter with a list of values. Any thoughts on this? 



It's always good to use other standards as the base. This list of "subadresses" is good. But I still don't see how a 3GPP phone can translate number dialed into one of these, unless it knows the local dial plan, or knows all dial plans and knows its location. 

Does the 3GPP spec mean that, if someone dials 911 in the US, that all bits must be set? 

Mike Pierce 




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<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>Depending on which 3GPP release you get there are a number of 
mechanisms.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>1)&nbsp;&nbsp;&nbsp; Certain numbers are built into the phone and 
recognised there.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>2)&nbsp;&nbsp;&nbsp; Certain numbers are preloaded onto the SIM/UICC card 
by the operator at subscription time.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>3)&nbsp;&nbsp;&nbsp; Numbers can be downloaded to the phone at 
registration time from the local providing access network.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff size=2>These 
are identified in the various versions of 22.101 (see <A 
href="http://www.3gpp.org/ftp/Specs/html-info/22101.htm">http://www.3gpp.org/ftp/Specs/html-info/22101.htm</A>) 
clause 10.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff size=2>I am 
not sure what terminal manufacturers do in the case of setting the bits for 
phones supplied for use in the US, however inspection of the 3GPP TS 24.008 text 
says that in the US, whatever the bits are set to, or not set at all, the call 
will be routed to an emergency centre. Note that this category element does not 
determine an emegency call, only the category of an emergency call. A call 
identified by the terminal as an emergency call will be sent with an EMERGENCY 
SETUP message rather than a SETUP message. Special characteristics of the 
EMERGENCY SETUP message include ignoring any called party number sent in that 
message.</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff size=2>Note 
that none of the above mechanisms can guarantee that an emergency call is always 
identified as such at the phone, and therefore there is also a responsibility on 
the local serving network to do its own analysis as well. In the 3GPP SIP 
environment, that responsibility may well extend to other proxies, with the need 
to reroute back to the country of origin (all calls carry the mobile country 
code and mobile network code as well as potentially geographic 
information).</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>regards</FONT></SPAN></DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=026073711-13022004><FONT face=Arial color=#0000ff 
size=2>Keith</FONT></SPAN></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Mpierce1@aol.com 
  [mailto:Mpierce1@aol.com]<BR><B>Sent:</B> 12 February 2004 20:52<BR><B>To:</B> 
  drage@lucent.com; sipping@ietf.org<BR><B>Subject:</B> Re: [Sipping] Concern 
  within sos ID - user addresses reserved<BR><BR></FONT></DIV><FONT 
  face=arial,helvetica><FONT size=2 PTSIZE="10">In a message dated 2/12/2004 
  8:58:35 AM Eastern Standard Time, drage@lucent.com writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">...</FONT><FONT lang=0 style="BACKGROUND-COLOR: #ffffff" 
    face=Arial color=#000000 size=3 PTSIZE="10" FAMILY="SANSSERIF" 
    BACK="#ffffff"></BLOCKQUOTE></FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial color=#000000 size=2 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff">I also note that 3GPP does have the 
  capability of signalling multiple preferences - this left me wondering whether 
  we would be better off trying to follow that, </FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial color=#000000 size=3 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff"><BR></FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial color=#000000 size=2 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff">
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">possibly with a single emergency uri, but with the possibility 
    of signalling category as a URI parameter with a list of values. Any 
    thoughts on this? <BR></FONT><FONT lang=0 style="BACKGROUND-COLOR: #ffffff" 
    face=Arial color=#000000 size=3 PTSIZE="12" FAMILY="SANSSERIF" 
    BACK="#ffffff"></BLOCKQUOTE><BR></FONT><FONT lang=0 
  style="BACKGROUND-COLOR: #ffffff" face=Arial color=#000000 size=2 PTSIZE="10" 
  FAMILY="SANSSERIF" BACK="#ffffff">It's always good to use other standards as 
  the base. This list of "subadresses" is good. But I still don't see how a 3GPP 
  phone can translate number dialed into one of these, unless it knows the local 
  dial plan, or knows all dial plans and knows its location. <BR><BR>Does the 
  3GPP spec mean that, if someone dials 911 in the US, that all bits must be 
  set? <BR><BR>Mike Pierce <BR><BR></BLOCKQUOTE></FONT></FONT></BODY></HTML>

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



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Since SIP generally favors generality, I wonder if the more general 
service description wouldn't be better. I don't see a confusion problem 
if services are defined hierarchically, as in

sos.fire
automobile.towing
restaurant.italian

However, if this causes major delays in discussion, we should stick to 
the simple list of half a dozen sos.* items, I think.

Chris Boulton wrote:

> My inclination would be to have a specific emergency service parameter
> to avoid any confusion in usage.  That way we explicitly define
> emergency routing without confusion.
> 


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Thanks for the pointer. From the description, Mike should talk to 3GPP 
about his concerns :-)

http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip 
(SP-22) has pretty much the mechanisms described in the SOS draft.

Note, in particular, in Section 10.1:

"The user friendly MMI that specifies the type of emergency directly 
(e.g. menu) should be supported for use in any (i.e. home or visited) 
PLMN to avoid the mis-connection in roaming case.  This shall be allowed 
both with and without SIM/USIM being present."

and 10.1.1

"a)	112 and 911 shall always be available. These numbers shall be stored 
on the ME.
b)	Any emergency number stored on a SIM/USIM when the SIM/USIM is present.
c)	000, 08, 110, 999, 118 and 119 when a SIM/USIM is not present. These 
numbers shall be stored on the ME.
d)	Additional emergency numbers that may have been downloaded by the 
serving network when the SIM/USIM is present."

Note that bullet (b) specifically means that a traveling user will have 
his or her home emergency numbers available even when roaming.

Unless we identify an equivalent set of personalization mechanisms for 
item (c), these would always have to be recognized, given that most 
non-3G SIP phones don't seem to carry SIMs or equivalent devices. 
Suggestions are most welcome here.

My general inclination is to follow the spirit of this document, as it 
seems strange that if a user is used to be able to dial 112 on a 
SIM-less 3GPP SIP device that this wouldn't work on a non-3GPP SIP phone.

Henning

Drage, Keith (Keith) wrote:

> Depending on which 3GPP release you get there are a number of mechanisms.
>  
> 1)    Certain numbers are built into the phone and recognised there.
>  
> 2)    Certain numbers are preloaded onto the SIM/UICC card by the 
> operator at subscription time.
>  
> 3)    Numbers can be downloaded to the phone at registration time from 
> the local providing access network.
>  
> These are identified in the various versions of 22.101 (see 
> http://www.3gpp.org/ftp/Specs/html-info/22101.htm) clause 10.
>  
> I am not sure what terminal manufacturers do in the case of setting the 
> bits for phones supplied for use in the US, however inspection of the 
> 3GPP TS 24.008 text says that in the US, whatever the bits are set to, 
> or not set at all, the call will be routed to an emergency centre. Note 
> that this category element does not determine an emegency call, only the 
> category of an emergency call. A call identified by the terminal as an 
> emergency call will be sent with an EMERGENCY SETUP message rather than 
> a SETUP message. Special characteristics of the EMERGENCY SETUP message 
> include ignoring any called party number sent in that message.
>  
> Note that none of the above mechanisms can guarantee that an emergency 
> call is always identified as such at the phone, and therefore there is 
> also a responsibility on the local serving network to do its own 
> analysis as well. In the 3GPP SIP environment, that responsibility may 
> well extend to other proxies, with the need to reroute back to the 
> country of origin (all calls carry the mobile country code and mobile 
> network code as well as potentially geographic information).
>  
> regards
>  
> Keith
> 
>     -----Original Message-----
>     *From:* Mpierce1@aol.com [mailto:Mpierce1@aol.com]
>     *Sent:* 12 February 2004 20:52
>     *To:* drage@lucent.com; sipping@ietf.org
>     *Subject:* Re: [Sipping] Concern within sos ID - user addresses reserved
> 
>     In a message dated 2/12/2004 8:58:35 AM Eastern Standard Time,
>     drage@lucent.com writes:
> 
> 
>>     ...
> 
>     I also note that 3GPP does have the capability of signalling
>     multiple preferences - this left me wondering whether we would be
>     better off trying to follow that,
> 
>>     possibly with a single emergency uri, but with the possibility of
>>     signalling category as a URI parameter with a list of values. Any
>>     thoughts on this?
> 
> 
>     It's always good to use other standards as the base. This list of
>     "subadresses" is good. But I still don't see how a 3GPP phone can
>     translate number dialed into one of these, unless it knows the local
>     dial plan, or knows all dial plans and knows its location.
> 
>     Does the 3GPP spec mean that, if someone dials 911 in the US, that
>     all bits must be set?
> 
>     Mike Pierce
> 

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Subject: RE: [Sipping] Drafts on exploders
Date: Fri, 13 Feb 2004 14:02:35 -0000
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From: "Chris Boulton" <cboulton@ubiquity.net>
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Gonzalo,
 	Really interesting stuff ;-).  Would it be worth mentioning the
use of GRUU when an 'Associated-List-Manipulation' header is equal to a
SIP or SIPS URI in the adhoc management draft?

Chris.


>-----Original Message-----
>From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>Sent: 13 February 2004 07:50
>To: sipping
>Subject: [Sipping] Drafts on exploders
>
>Folks,
>
>We have submitted a set of drafts on exploders. The following one
>provides the requirements we are using and a framework to understand
the
>rest of the drafts.
>
>http://users.piuha.net/gonzalo/draft-camarillo-sipping-exploders-02.txt
>
>The rest of them can be fetched (until they appear in the archives)
from:
>
>http://users.piuha.net/gonzalo/draft-camarillo-sipping-uri-list-01.txt
>http://users.piuha.net/gonzalo/draft-camarillo-sipping-adhoc-management
-
>00.txt
>http://users.piuha.net/gonzalo/draft-camarillo-sipping-transac-package-
>00.txt
>http://users.piuha.net/gonzalo/draft-camarillo-sipping-multiple-refer-
>00.txt
>
>Commments are appreciated,
>
>Thanks,
>
>Gonzalo
>
>
>
>This communication is confidential and intended solely for the
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Chris,

 >  Would it be worth mentioning the
> use of GRUU when an 'Associated-List-Manipulation' header is equal to a
> SIP or SIPS URI in the adhoc management draft?

Yes, definitively. That is the idea. I will clarify that in the next 
revision of the drafts.

Thanks,

Gonzalo


 

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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Chris Boulton" <cboulton@ubiquity.net>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sipping@ietf.org>
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Henning wrote:

>=20
> However, if this causes major delays in discussion, we should=20
> stick to=20
> the simple list of half a dozen sos.* items, I think.
>=20

Agree, we should stick in a first run to genuine _emergency_ calls,
and what this is, is in most countries defined, and only these
are handled special.

This is the list proposed in the -sipping-sos- draft by Henning

Just for information: In Austria we have in addition to this list
112
sos:doctor
sos:child (e.g. for abused children with special trained personal)

but we do not have
sos:marine ;-)

Richard

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Date: Fri, 13 Feb 2004 10:20:31 -0500
To: "Banibrata Dutta" <dutta@india.hp.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: RE: [Sipping] Drafts on exploders
Cc: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        "'sipping'" <sipping@ietf.org>
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I am a little leery of a proxy-based solution, based on the experiences 
with early media.  In essence, the middleman initiates the INVITE at the 
front end of a dialog then expects the UAC to handle who knows how many 
responses that return to it.  For connectionless operations it might work, 
but for connection-oriented operations, it can easily scale beyond the 
capability of the UAC to handle it.  Hopefully, this will be more elegant 
than early media.

Mike

At 03:33 PM 2/13/2004 +0530, Banibrata Dutta wrote:
>Hi Gonzalo,
>
>   I am sorry, I meant "heavy-weight" only. Okay, isn't then exploder
>itself a service, but yes it would become a feature they may be used
>by several services. Confercing is then just another service that uses
>the exploder feature, right?
>
>   Won't it be better to take that responsibility out of a B2BUA and
>put it in the core network (in the proxies) s.t. the explosion happens
>where it is needed? Let me explain my view as follows:-
>
>there are 2 ways:-
>
>1) A single To: header which explodes to multiple INVITES (or other
>    method messages).
>2) A single INVITE with multiple To: headers which is understood by
>    a router and split off into multiple INVITE's with single To:
>    headers.
>
>in the first case the exploding proxy (or entity) needs access to a
>explosion database (i.e. group-name -> list of members). this "may be"
>achieved by DNS mechanisms (i am not even close to being a DNS
>expert, but i think that it can be done, the same way SOS URI's
>are transformed into actual ERS/PCAP #s).
>
>in the second case the exploding proxy doesn't need a explosion database
>and the original INVITE anyway had all the info needed to split it off
>into multiple INVITEs with single To: headers.
>
>with these approaches the benefit of "localization" of explosion can be
>used, i.e. the explosion happens as late as possible, leading to the
>minimum # of messages in the network, i.e. multiple exploder proxies
>would exist in the network and it may so happen that the orginal
>invite with 10 To: headers go split into 2 INVITES with 6 and 4 To:
>headers respectively at an exploder proxy, and then the INVITE with
>6 To:s got split into 2 with 3 To:s each and so on. this way the
>explosion is "Just-In-Time" and minimal.
>
>regards,
>banibrata dutta.
>
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of Gonzalo Camarillo
> > Sent: Friday, February 13, 2004 3:09 PM
> > To: Banibrata Dutta
> > Cc: 'sipping'
> > Subject: Re: [Sipping] Drafts on exploders
> >
> >
> > Hello,
> >
> > > Why do you need a heavy-weight light B2BUA for "exploding".
> >
> > Do you mean light *or* heavy-weight?
> >
> > In any case, you do not want to use a proxy because the
> > services these
> > exploders provide (e.g., conferencing) are provided by B2BUAs, not by
> > proxies.
> >
> > Gonzalo
> >
> >
> >
> > This communication is confidential and intended solely for
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> >
> > _______________________________________________
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> > This list is for NEW development of the application of SIP
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> > sip Use sip@ietf.org for new developments of core SIP
> >
>
>
>_______________________________________________
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>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: mielias@hss.hns.com
CC: Joseph Herman <jiherman@us.ibm.com>, sipping@ietf.org
Subject: Re: [Sipping] Translating  ISDN IE or SS7 optional fields to SIP
References: <OF79E0B96F.75DA117C-ON65256E39.001BBCFD-65256E39.001C6EDD@hss.hns.com>
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Also, the ITU-T and regional bodies (ANSI T1S1 for sure) are defining their own 
rules.  The ITU-T reference is Q.1912.5.

mielias@hss.hns.com wrote:

> 
> 
> 
> RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to
> Session Initiation Protocol (SIP) Mapping.
> This RFC describes a way to perform the mapping between SIP and ISUP
> (message and parameter level).
> 
> -
> Elias
> Hughes Software Systems
> http://www.hssworld.com
> -
> 
> 
> 
>                                                                            
>              Joseph Herman                                                 
>              <jiherman@us.ibm.                                             
>              com>                                                       To 
>              Sent by:                  sipping@ietf.org                    
>              sipping-admin@iet                                          cc 
>              f.org                                                         
>                                                                    Subject 
>                                        [Sipping] Translating  ISDN IE or   
>              02/13/04 04:31 AM         SS7 optional fields to SIP          
>                                                                            
>                                                                            
>                                                                            
>                                                                            
>                                                                            
>                                                                            
> 
> 
> 
> 
> 
> Is there a document that defines (or proposes) a standard way to map
> protocol specific information into SIP to be consumed by a UA?   SIP-T
> provides  the entire SS7 packet embedded in the SIP message rather than a
> translation of the fields.   I am hoping for a mapping like
> 
> redirection-number <==> redir-num: SIP header field
> 
> Is there such a document?
> 
>                                         Thanks for any info,
>                                         Joe H.
> 
> --
> Joe Herman
> IBM Annapolis RM Telephony Solutions
> 410-571-7117
> dzoey@us.ibm.com
> 
> Everything is wonderful until you know something about it.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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Date: Fri, 13 Feb 2004 11:34:29 -0500
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From: Tom Taylor <taylor@nortelnetworks.com>
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Subject: [Sipping] INFO before 200 OK INVITE?
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Q.1912.5 currently has some ISUP messages which don't interwork to call processing 
being encapsulated by INFO except when no dialog-establishing provisional response 
has been sent.  We have had an intervention suggesting that what is required is a 
confirmed dialog, not just an early dialog.  As a result, according to the 
intervention, you can't send an INFO from the UAS before it sends 200 OK INVITE.

Any views on this?  The RFC 2976 text quoted in the intervention is really based 
on RFC 2543, and was written before the concept of a dialog was elaborated.  If it 
seems reasonable to send INFO in an early dialog, the language of RFC 2976 (as 
presented in Q.1912.5 Annex C.3) would need to be tweaked a bit.

Tom Taylor

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CC: Chris Boulton <cboulton@ubiquity.net>,
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Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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I like the callerprefs direction for this. I don't have a particular 
preference for class or a new service feature tag. (But that word 
service continues to cause problems.)

Whatever it is, I think it should work with non-emergency calls, rather 
than just with sos. For instance, often there is a separate number for 
non-emergency calls to the police etc. department. It would be nice to 
be able to use the same "service" callerpref parameter with a generic 
non-emergency public service number, together with the appropriate 
preference setting, to talk in a non-emergency context to the police, 
fire, ems, etc. organization.

	Paul

Henning Schulzrinne wrote:
> Since SIP generally favors generality, I wonder if the more general 
> service description wouldn't be better. I don't see a confusion problem 
> if services are defined hierarchically, as in
> 
> sos.fire
> automobile.towing
> restaurant.italian
> 
> However, if this causes major delays in discussion, we should stick to 
> the simple list of half a dozen sos.* items, I think.
> 
> Chris Boulton wrote:
> 
>> My inclination would be to have a specific emergency service parameter
>> to avoid any confusion in usage.  That way we explicitly define
>> emergency routing without confusion.
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] Drafts on exploders
Date: Fri, 13 Feb 2004 09:32:45 -0800
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Thread-Topic: [Sipping] Drafts on exploders
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From: "Sunil Veluvali" <Sunil.Veluvali@sylantro.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
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Gonzolo,

I believe this idea has merits. If I could somehow instruct my
transaction stateful outbound proxy to preload a set of contacts in it's
request target list (section 16.5 of RFC 3261), I have a generic way to
fan out the request to a set of addresses-of-record. Combine this with
Caller Preferences and now I also have control on how redirections are
handled, etc.=20

Since multiple To: headers will break existing implementations, maybe a
better way to go about it is to define a new header called
Request-Target. Say, I have three buddies in my list a, b, c whose
registrations are for a1,a2,b1,b2,c1,c2 respectively.

I could simply a request to my outbound-proxy as:

REQUEST sip:outbound-proxy SIP/2.0
f: me@domain;tag=3Dxyz
t: all@domain
Call-ID: xxxx
M: me@m1
Request-Target: a@domain,b@domain,c@domain
Request-Disposition: proxy,recurse,parallel,no-queue

Almost all of what I need is available in a transaction stateful proxy
server.

Thanks,
Sunil

> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf

> Of Gonzalo Camarillo
> Sent: Friday, February 13, 2004 3:09 PM
> To: Banibrata Dutta
> Cc: 'sipping'
> Subject: Re: [Sipping] Drafts on exploders
>=20
>=20
> Hello,
>=20
> > Why do you need a heavy-weight light B2BUA for "exploding".
>=20
> Do you mean light *or* heavy-weight?
>=20
> In any case, you do not want to use a proxy because the services these

> exploders provide (e.g., conferencing) are provided by B2BUAs, not by=20
> proxies.
>=20
> Gonzalo
>=20
> =20
>=20
> This communication is confidential and intended solely for the=20
> addressee(s). Any unauthorized review, use, disclosure or distribution

> is prohibited. If you believe this message has been sent to you in=20
> error, please notify the sender by replying to this transmission and=20
> delete the message without disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data corruption,=20
> interruption, unauthorized amendment, tampering and viruses, and we=20
> only send and receive e-mails on the basis that we are not liable for=20
> any such corruption, interception, amendment, tampering or viruses or=20
> any consequences thereof.
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP Use=20
> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> sip@ietf.org for new developments of core SIP
>=20


_______________________________________________
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This list is for NEW development of the application of SIP Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sip@ietf.org for new developments of core SIP



_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: mielias@hss.hns.com
CC: Joseph Herman <jiherman@us.ibm.com>, sipping@ietf.org
Subject: Re: [Sipping] Translating  ISDN IE or SS7 optional fields to SIP
References: <OF79E0B96F.75DA117C-ON65256E39.001BBCFD-65256E39.001C6EDD@hss.hns.com>
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And for ISUP Overlap Signaling to SIP mapping, see rfc3578

Alex.
 

mielias@hss.hns.com wrote:

>
>
>RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to
>Session Initiation Protocol (SIP) Mapping.
>This RFC describes a way to perform the mapping between SIP and ISUP
>(message and parameter level).
>
>-
>Elias
>Hughes Software Systems
>http://www.hssworld.com
>-
>
>
>
>                                                                           
>             Joseph Herman                                                 
>             <jiherman@us.ibm.                                             
>             com>                                                       To 
>             Sent by:                  sipping@ietf.org                    
>             sipping-admin@iet                                          cc 
>             f.org                                                         
>                                                                   Subject 
>                                       [Sipping] Translating  ISDN IE or   
>             02/13/04 04:31 AM         SS7 optional fields to SIP          
>                                                                           
>                                                                           
>                                                                           
>                                                                           
>                                                                           
>                                                                           
>
>
>
>
>
>Is there a document that defines (or proposes) a standard way to map
>protocol specific information into SIP to be consumed by a UA?   SIP-T
>provides  the entire SS7 packet embedded in the SIP message rather than a
>translation of the fields.   I am hoping for a mapping like
>
>redirection-number <==> redir-num: SIP header field
>
>Is there such a document?
>
>                                        Thanks for any info,
>                                        Joe H.
>
>--
>Joe Herman
>IBM Annapolis RM Telephony Solutions
>410-571-7117
>dzoey@us.ibm.com
>
>Everything is wonderful until you know something about it.
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP
>  
>


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Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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In a message dated 2/13/2004 8:52:05 AM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> Thanks for the pointer. From the description, Mike should talk to 3GPP 
> about his concerns :-)
> 
> http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip 
> 


[MAP] The main different being the lack of a SIM, or personalization device, 
in the SOS draft. That leaves it with following the GSM case "without SIM", 
which, as noted below, means that many different numbers, including 118, will be 
reported as emergency. If it is true that 118 is Directory Assistance in 
Finland, as the SOS draft states, then this operation (using GSM, 3GPP, SIP, or a 
rotary dial phone) can not be allowed in Finland). (Anyone in Finland know 
what is done there?)

Guess if I was involved in 3GPP I would take this up with them, but I'm not. 
Certainly looks like a mistake that the actual dialled digits are not included 
in the Emergency Setup message, which would allow the switch in the network 
to sort it out!

> 
> Note, in particular, in Section 10.1:
> 
> "The user friendly MMI that specifies the type of emergency directly 
> (e.g. menu) should be supported for use in any (i.e. home or visited) 
> PLMN to avoid the mis-connection in roaming case.  This shall be allowed 
> both with and without SIM/USIM being present."
> 
> and 10.1.1
> 
> "a)    112 and 911 shall always be available. These numbers shall be stored 
> on the ME.
> b)    Any emergency number stored on a SIM/USIM when the SIM/USIM is present.
> c)    000, 08, 110, 999, 118 and 119 when a SIM/USIM is not present. These 
> numbers shall be stored on the ME.
> d)    Additional emergency numbers that may have been downloaded by the 
> serving network when the SIM/USIM is present."
> 


Mike Pierce



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/13/2004 8:52:05 AM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Thanks for the pointer. Fro=
m the description, Mike should talk to 3GPP=20
<BR>about his concerns :-)
<BR>
<BR>http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip=20
<BR>(SP-22) has pretty much the mechanisms described in the SOS draft.</BLOC=
KQUOTE>
<BR>
<BR>
<BR>[MAP] The main different being the lack of a SIM, or personalization dev=
ice, in the SOS draft. That leaves it with following the GSM case "without S=
IM", which, as noted below, means that many different numbers, including 118=
, will be reported as emergency. If it is true that 118 is Directory Assista=
nce in Finland, as the SOS draft states, then this operation (using GSM, 3GP=
P, SIP, or a rotary dial phone) can not be allowed in Finland). (Anyone in F=
inland know what is done there?)
<BR>
<BR>Guess if I was involved in 3GPP I would take this up with them, but I'm=20=
not. Certainly looks like a mistake that the actual dialled digits are not i=
ncluded in the Emergency Setup message, which would allow the switch in the=20=
network to sort it out!
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
<BR>Note, in particular, in Section 10.1:
<BR>
<BR>"The user friendly MMI that specifies the type of emergency directly=20
<BR>(e.g. menu) should be supported for use in any (i.e. home or visited)=20
<BR>PLMN to avoid the mis-connection in roaming case. &nbsp;This shall be al=
lowed=20
<BR>both with and without SIM/USIM being present."
<BR>
<BR>and 10.1.1
<BR>
<BR>"a) &nbsp;&nbsp;&nbsp;112 and 911 shall always be available. These numbe=
rs shall be stored=20
<BR>on the ME.
<BR>b) &nbsp;&nbsp;&nbsp;Any emergency number stored on a SIM/USIM when the=20=
SIM/USIM is present.
<BR>c) &nbsp;&nbsp;&nbsp;000, 08, 110, 999, 118 and 119 when a SIM/USIM is n=
ot present. These=20
<BR>numbers shall be stored on the ME.
<BR>d) &nbsp;&nbsp;&nbsp;Additional emergency numbers that may have been dow=
nloaded by the=20
<BR>serving network when the SIM/USIM is present."
<BR></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>Mike Pierce
<BR>
<BR></FONT></HTML>

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Subject: Re: [Sipping] Concern within sos ID - user addresses reserved
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I suspect that the number of SIM-less phones is fairly small, as a 
fraction. For general SIP phones, one reasonable option is to say that 
the 'no configuration' case only applies if the device was unable to 
obtain information about geographically-local emergency numbers, using 
one of the several means available (XCAP, DNS).

> [MAP] The main different being the lack of a SIM, or personalization 
> device, in the SOS draft. That leaves it with following the GSM case 
> "without SIM", which, as noted below, means that many different numbers, 
> including 118, will be reported as emergency. If it is true that 118 is 
> Directory Assistance in Finland, as the SOS draft states, then this 
> operation (using GSM, 3GPP, SIP, or a rotary dial phone) can not be 
> allowed in Finland). (Anyone in Finland know what is done there?)
> 


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



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To: "'Joseph Herman'" <jiherman@us.ibm.com>, sipping@ietf.org,
        "Banibrata Dutta" <dutta@india.hp.com>
Date: Fri, 13 Feb 2004 16:02:02 -0500
From: "Jitender Arora" <jitender_arora@lycos.com>
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Refer to RFC 3398 ( ISUP to SIP Mapping ) .....
It contains the Mapping between the ISUP and SIP messages ...
Jitender

--------- Original Message ---------
DATE: Fri, 13 Feb 2004 10:06:04
From: "Banibrata Dutta" <dutta@india.hp.com>
To: "'Joseph Herman'" <jiherman@us.ibm.com>, <sipping@ietf.org>
Cc: 


One quick way would be to check that out by searching the ID-list hosted at the
very useful http://softarmor.com/sipping/ site.
AFAIK, such a draft doesn't exist. SIP-T is what comes closest (i think).

-BDutta.
-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of Joseph Herman
Sent: Friday, February 13, 2004 4:31 AM
To: sipping@ietf.org
Subject: [Sipping] Translating ISDN IE or SS7 optional fields to SIP



Is there a document that defines (or proposes) a standard way to map protocol specific information into SIP to be consumed by a UA?   SIP-T provides  the entire SS7 packet embedded in the SIP message rather than a translation of the fields.   I am hoping for a mapping like 

redirection-number <==> redir-num: SIP header field 

Is there such a document? 

                                        Thanks for any info, 
                                        Joe H. 

--
Joe Herman
IBM Annapolis RM Telephony Solutions
410-571-7117
dzoey@us.ibm.com

Everything is wonderful until you know something about it.


____________________________________________________________
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<PRE>Refer to RFC 3398 ( ISUP to SIP Mapping ) .....</PRE><PRE>It contains the Mapping between the ISUP and SIP messages ...</PRE><PRE>Jitender
</PRE>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 4px; BORDER-LEFT: #1010ff 3px solid"><FONT face="Times New Roman" size=2>--------- Original Message ---------<BR>
<TABLE cellSpacing=0 cellPadding=0 width=800 border=0>
<TBODY>
<TR>
<TD align=left><B>DATE:</B> Fri, 13 Feb 2004 10:06:04</TD></TR>
<TR>
<TD align=left><B>From:</B> "Banibrata Dutta" &lt;dutta@india.hp.com&gt;</TD></TR>
<TR>
<TD align=left><B>To:</B> "'Joseph Herman'" &lt;jiherman@us.ibm.com&gt;, &lt;sipping@ietf.org&gt;</TD></TR>
<TR>
<TD align=left><B>Cc:</B> </TD></TR></TBODY></TABLE></FONT><BR>
<DIV><SPAN class=203073204-13022004><FONT face=Arial color=#0000ff size=2>One quick way would be to check that out&nbsp;by searching&nbsp;the ID-list hosted at the</FONT></SPAN></DIV>
<DIV><SPAN class=203073204-13022004><FONT face=Arial color=#0000ff size=2>very useful&nbsp;<A href="http://softarmor.com/sipping/">http://softarmor.com/sipping/</A>&nbsp;site.</FONT></SPAN></DIV>
<DIV><SPAN class=203073204-13022004><FONT face=Arial color=#0000ff size=2>AFAIK, such a draft doesn't exist. SIP-T is what comes closest (i think).</FONT></SPAN></DIV>
<DIV><SPAN class=203073204-13022004><FONT face=Arial color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=203073204-13022004><FONT face=Arial color=#0000ff size=2>-BDutta.</FONT></SPAN></DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] <B>On Behalf Of </B>Joseph Herman<BR><B>Sent:</B> Friday, February 13, 2004 4:31 AM<BR><B>To:</B> sipping@ietf.org<BR><B>Subject:</B> [Sipping] Translating ISDN IE or SS7 optional fields to SIP<BR><BR></FONT></DIV><BR><FONT face=sans-serif size=2>Is there a document that defines (or proposes) a standard way to map protocol specific information into SIP to be consumed by a UA? &nbsp; SIP-T provides &nbsp;the entire SS7 packet embedded in the SIP message rather than a translation of the fields. &nbsp; I am hoping for a mapping like</FONT> <BR><BR><FONT face=sans-serif size=2>redirection-number &lt;==&gt; redir-num: SIP header field</FONT> <BR><BR><FONT face=sans-serif size=2>Is there such a document?</FONT> <BR><BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nb!
 sp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Thanks for any info,</FONT> <BR><FONT face=sans-serif size=2>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Joe H.</FONT> <BR><BR><FONT face=sans-serif size=2>--<BR>Joe Herman<BR>IBM Annapolis RM Telephony Solutions<BR>410-571-7117<BR>dzoey@us.ibm.com<BR><BR>Everything is wonderful until you know something about it.<BR></BLOCKQUOTE></FONT></BLOCKQUOTE><br><br>____________________________________________________________<br><font face="verdana" size="2" color="000000">Find what you are looking for with the <a href="http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp?SRC=lycos10"> Lycos Yellow Pages</a>.</font>
--=_-=_-KMNBJJFBNOGJFCAA--


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Trait-based Authorization Requirements for the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-ietf-sipping-trait-authz-00.txt
	Pages		: 16
	Date		: 2004-2-12
	
This document lays out a set of requirements related to trait-based
   authorization for the Session Initiation Protocol.  While some
   authentication mechanisms are described in the base SIP
   specification, trait-based authorization provides information used to
   make policy decisions based on the attributes of a participant in a

   session.  This approach provides a richer framework for
   authorization, as well as allow greater privacy for users of an
   identity system.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Requirements for Session Initiation Protocol Location Conveyance
	Author(s)	: J. Polk
	Filename	: draft-ietf-sipping-location-requirements-00.txt
	Pages		: 11
	Date		: 2004-2-12
	
This document presents the framework and requirements for an 
   extension to the Session Initiation Protocol (SIP) [1] for 
   conveyance of user location information from a Session Initiation 
   Protocol (SIP) user agent to another SIP entity.  We consider cases 
   where location information is conveyed from end to end, as well as 
   cases where message routing by intermediaries is influenced by the 
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From: "Chaidez, Fidencio (Fidencio)" <fchaidez@lucent.com>
To: "'Tom Taylor'" <taylor@nortelnetworks.com>, sipping@ietf.org
Subject: RE: [Sipping] INFO before 200 OK INVITE?
Date: Fri, 13 Feb 2004 15:17:39 -0600
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Tom,
The following text from RFC 3398 appears to 
clarify that INFO messages may indeed be sent for early
dialogs:

  From the callee to the caller, if a message is received by a gateway
   before the call has been answered (i.e., ANM is received) it SHOULD
   be encapsulated in an INFO, provided that this will not be the first
   SIP message sent in the backwards direction (in which case it SHOULD
   be encapsulated in a provisional 1xx response).  Similarly a message
   which is received on the originating side (probably in response to an
   INR) before a 200 OK has been received by the gateway should be
   carried within an INFO.  

Fidencio Chaidez

-----Original Message-----
From: Tom Taylor [mailto:taylor@nortelnetworks.com]
Sent: Friday, February 13, 2004 10:34 AM
To: sipping@ietf.org
Subject: [Sipping] INFO before 200 OK INVITE?


Q.1912.5 currently has some ISUP messages which don't interwork to call processing 
being encapsulated by INFO except when no dialog-establishing provisional response 
has been sent.  We have had an intervention suggesting that what is required is a 
confirmed dialog, not just an early dialog.  As a result, according to the 
intervention, you can't send an INFO from the UAS before it sends 200 OK INVITE.

Any views on this?  The RFC 2976 text quoted in the intervention is really based 
on RFC 2543, and was written before the concept of a dialog was elaborated.  If it 
seems reasonable to send INFO in an early dialog, the language of RFC 2976 (as 
presented in Q.1912.5 Annex C.3) would need to be tweaked a bit.

Tom Taylor

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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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



From exim@www1.ietf.org  Fri Feb 13 18:10:07 2004
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Date: Fri, 13 Feb 2004 18:05:48 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sipping] draft-ietf-sipping-location-requirements-00
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I think it would be helpful to do a back-of-the-envelope calculation for 
a typical message exchange and geo value as to what delay a reasonable 
(GPRS, say) wireless channel would impose for the case that the proxy 
provides location information.

My back-of-a-grocery-bag calculation (even rougher than 
back-of-the-envelope) is:

100 kb/s data rate (optimistic)
10000 bit per message with location information

= 100 ms per transit of a location object across the air interface.

In the proxy-provides-location case, we'd need roughly 3 messages 
(INVITE-redirect-INVITE), two of which have location information, for a 
total of 300 ms, not counting the TLS setup time (which may or may not 
be required for each emergency call). This is close enough to the E-1 
requirement limit that somewhat more precise calculations are called for.

I also found it peculiar that the draft does not mention the 
signature-only (unencrypted) mode for S/MIME, which would not interfere 
with proxy inspection. (The chain of reasoning should probably be: need 
privacy + can't use S/MIME for privacy since proxies need to peek -> 
TLS, rather than the current non-sequitur.)

Henning

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From: Cullen Jennings <fluffy@cisco.com>
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Why not just use MGCP instead of SIP if you want to do this. It will work
much better. 



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From: Cullen Jennings <fluffy@cisco.com>
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I think this topic of far end camera control is important but I don't think
this is the right way to do it. It seems like it would work better to use
SIP to set up a session and use the data in this session to do the control.
Sort of like MRCPv2 or session mode IM does. Have you considered something
like using session mode IM to send mime bodies controlled the camera. One
could define a mime type like application/fecc and send IM with a body like

<pan-relative angle="20" \>

To pan clockwise by 20 degrees from the present position. Anyway, I don't
know the best way to do this and we should probably figure out requirements
before jumping into solutions but I would be glad to see some work like this
get done. Not sure that sipping is the best place for it.

Cullen


 


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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Sat, 14 Feb 2004 12:36:59 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [Sipping] draft-cgkim-sipping-info-00
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IM works quite well for this and similar applications. We use it to 
control other devices beyond cameras, too, as sending an IM saying "look 
at me" and "move camera 20 degrees" are semantically rather similar.

Should this be in MMUSIC?

Cullen Jennings wrote:

> I think this topic of far end camera control is important but I don't think
> this is the right way to do it. It seems like it would work better to use
> SIP to set up a session and use the data in this session to do the control.
> Sort of like MRCPv2 or session mode IM does. Have you considered something
> like using session mode IM to send mime bodies controlled the camera. One
> could define a mime type like application/fecc and send IM with a body like
> 
> <pan-relative angle="20" \>
> 
> To pan clockwise by 20 degrees from the present position. Anyway, I don't
> know the best way to do this and we should probably figure out requirements
> before jumping into solutions but I would be glad to see some work like this
> get done. Not sure that sipping is the best place for it.
> 
> Cullen
> 
> 
>  
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
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In a message dated 2/13/2004 3:16:16 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> I suspect that the number of SIM-less phones is fairly small, as a 
> fraction. For general SIP phones, one reasonable option is to say that 
> the 'no configuration' case only applies if the device was unable to 
> obtain information about geographically-local emergency numbers, using 
> one of the several means available (XCAP, DNS).
> 



In a message dated 2/14/2004 4:57:16 AM Eastern Standard Time, 
atle.monrad@ericsson.com writes:

> What would be the first thought of Joe Citizen on vacation somewhere abroad 
> driving off the road desparately needing assistance, calling 911 or the 
> local xyz ??
> Due to the mobility, to me it makes sense to do the analysis whether it is 
> an emergency call or not in the handset in 3gpp. With the current mechanisms 
> the network can download local emergency numbers to the handset, and depending 
> on country, availability and configurastion of (U)SIM, the handset decides 
> whether a normal call setup is placed (that may be analyzed in the switch and 
> routed to the emergency centre anyway) or an emergency call setup is placed. 
> This is in the end about end-user perception and tradeoffs between different 
> numbering plans, analyzing the number based on the origin of the caller might 
> be some administrative work.... 
> Be also aware that local policy may decide whether (U)SIM-less emergency 
> calls are allowed or not, and that the support of 118 in 3gpp only apply in case 
> the (U)SIM is not present, thus other calls than emergency will always fail.
> 
> 


[MAP]
I agree that allowing an American to dial 911 for emergency while in Europe, 
or allowing anyone to dial their well-known emergency number while in the US 
is a good thing, as long as it is based on smarts in the phone (SIM or download 
locally) that says that this particular phone should work that way. I don't 
believe it is acceptable for the other 100 million people in the US to 
unknowingly get routed to emergency if they dial any one of those "foreign" numbers 
that they know nothing about. I suspect that 3GPP has covered this by allowing 
"analysis in the handset", although it isn't clear to me from the spec that 
they have covered all the cases. Anyway, this still doesn't answer the question 
of what is done in countries where this results in a conflict (like Finland 
which apparently uses 118 for Directory Service). I suspose the easy answer is 
that, if my SIM says that 118 is my desired emergency number and I'm in Finland, 
then I can't dial Directory Assistance. The other possible answer, implied 
someplace, is that the phone knows there is a confict, and if I dial 118, it 
asks me which one I want. Would this be a menu on the little screen that I'm 
unlikely to see if I was in an accident?

It seems that the discussion for a SIP device (SOS URI) is lacking the 
presumption of the use of a SIM and the definition of a local download of local 
emergency numbers. If you want to follow the 3GPP model, then I think it's 
important to use all of it. As Henning noted (correctly I believe), the number of 
SIM-less phones is small. It is not the normal, since I understand that you must 
have a SIM to do lots of other things.

Therefore, it is okay to interpret 118 (and just about every other assigned 
number) as emergency when there is no SIM. But, as I read 3GPP TS 22.101, it 
says that 112 and 911 are always interpreted as emergency, without or without a 
SIM. This is the point I don't understand.

Of course, the other difference is that every mobile phone (as far as I know) 
operates on the basis that you enter a complete digit string and then hit 
"send" (or some equivalent). Therefore, there is no conflict between any of the 
emergency numbers and the starting digits of some other local number. So far, I 
don't believe there has been a general assumption that a SIP phone will work 
that way, therefore we also have to be concerned with the case of 911 or 112 
or any other emergency number being the start of some other legitimate dialing 
string. Draft-siinnreich-sipdev-req-03 touches on this issue but seems to 
allow either operation.

Mike Pierce





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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/13/2004 3:16:16 PM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I suspect that the number o=
f SIM-less phones is fairly small, as a=20
<BR>fraction. For general SIP phones, one reasonable option is to say that=20
<BR>the 'no configuration' case only applies if the device was unable to=20
<BR>obtain information about geographically-local emergency numbers, using=20
<BR>one of the several means available (XCAP, DNS).
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>In a message dated 2/14/2004 4:57:16 AM Eastern Standard Time, atle.monr=
ad@ericsson.com writes:
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">What would be the first tho=
ught of Joe Citizen on vacation somewhere abroad driving off the road despar=
ately needing assistance, calling 911 or the local xyz ??
<BR>Due to the mobility, to me it makes sense to do the analysis whether it=20=
is an emergency call or not in the handset in 3gpp. With the current mechani=
sms the network can download local emergency numbers to the handset, and dep=
ending on country, availability and configurastion of (U)SIM, the handset de=
cides whether a normal call setup is placed (that may be analyzed in the swi=
tch and routed to the emergency centre anyway) or an emergency call setup is=
 placed. This is in the end about end-user perception and tradeoffs between=20=
different numbering plans, analyzing the number based on the origin of the c=
aller might be some administrative work....=20
<BR>Be also aware that local policy may decide whether (U)SIM-less emergency=
 calls are allowed or not, and that the support of 118 in 3gpp only apply in=
 case the (U)SIM is not present, thus other calls than emergency will always=
 fail.
<BR>
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>[MAP]
<BR>I agree that allowing an American to dial 911 for emergency while in Eur=
ope, or allowing anyone to dial their well-known emergency number while in t=
he US is a good thing, as long as it is based on smarts in the phone (SIM or=
 download locally) that says that this particular phone should work that way=
. I don't believe it is acceptable for the other 100 million people in the U=
S to unknowingly get routed to emergency if they dial any one of those "fore=
ign" numbers that they know nothing about. I suspect that 3GPP has covered t=
his by allowing "analysis in the handset", although it isn't clear to me fro=
m the spec that they have covered all the cases. Anyway, this still doesn't=20=
answer the question of what is done in countries where this results in a con=
flict (like Finland which apparently uses 118 for Directory Service). I susp=
ose the easy answer is that, if my SIM says that 118 is my desired emergency=
 number and I'm in Finland, then I can't dial Directory Assistance. The othe=
r possible answer, implied someplace, is that the phone knows there is a con=
fict, and if I dial 118, it asks me which one I want. Would this be a menu o=
n the little screen that I'm unlikely to see if I was in an accident?
<BR>
<BR>It seems that the discussion for a SIP device (SOS URI) is lacking the p=
resumption of the use of a SIM and the definition of a local download of loc=
al emergency numbers. If you want to follow the 3GPP model, then I think it'=
s important to use all of it. As Henning noted (correctly I believe), the nu=
mber of SIM-less phones is small. It is not the normal, since I understand t=
hat you must have a SIM to do lots of other things.
<BR>
<BR>Therefore, it is okay to interpret 118 (and just about every other assig=
ned number) as emergency when there is no SIM. But, as I read 3GPP TS 22.101=
, it says that 112 and 911 are always interpreted as emergency, without or w=
ithout a SIM. This is the point I don't understand.
<BR>
<BR>Of course, the other difference is that every mobile phone (as far as I=20=
know) operates on the basis that you enter a complete digit string and then=20=
hit "send" (or some equivalent). Therefore, there is no conflict between any=
 of the emergency numbers and the starting digits of some other local number=
. So far, I don't believe there has been a general assumption that a SIP pho=
ne will work that way, therefore we also have to be concerned with the case=20=
of 911 or 112 or any other emergency number being the start of some other le=
gitimate dialing string. Draft-siinnreich-sipdev-req-03 touches on this issu=
e but seems to allow either operation.
<BR>
<BR>Mike Pierce
<BR>
<BR>
<BR>
<BR></FONT></HTML>

--part1_1a2.2041874c.2d5fbea6_boundary--

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



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Date: Sat, 14 Feb 2004 20:13:25 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sipping] user=dialstring?
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In Minneapolis (http://www.ietf.org/proceedings/03nov/228.htm) we had a 
lively discussion on how to capture dialstrings in SIP URI, with 
something like

sip:*69@localhost;user=dialstring

emerging as the favorite solution. Has this been written down anywhere? 
(This happens to be relevant for the -sos- discussion, for example.)

Henning

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



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Subject: [Sipping] Updated draft draft-ietf-sipping-sos-01
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I have submitted draft-ietf-sipping-sos-01, reflecting some of the 
discussion of the last week. Until it appears in the archives, you can 
find it at 
http://www.cs.columbia.edu/sip/draft/sos/draft-ietf-sipping-sos-01.html

The major semantic change is that the sos.* subaddresses were replaced 
by a callee capabilities media feature tag.

Also, the discovery of emergency numbers was aligned, to the extent 
possible, with the 3GPP behavior: 112 and 911 always recognized, a short 
list of common numbers recognized if and only if the device has no other 
information about local emergency numbers.

Since discovery of local emergency numbers could be relatively simple if 
the 'EN' resource record proposal is adopted (know the country the 
device is in, make one DNS query), I'm hoping that the use of default 
numbers will be rare in practice. 3GPP devices have the advantage that 
you can't call the Finnish directory service if your SIM card has been 
removed, so the collision problem is less pronounced.

Comments are appreciated. I hope to get some session time in Seoul to 
discuss some of the open issues and reach closure on the discussion 
results so far.

The draft notes one open issue:

OPEN ISSUE: Alternatively, if the user dialed a digit string, the "To" 
header could contain the digit string, as in 
"tel:911;phone-context=sos.arpa". (The calling device may not know its 
current location, so it cannot insert a country-specific identifier.)


This relates to my query to the list as to whether the 'dialstring' 
discussion ever yielded a draft.

Henning

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



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Subject: [Sipping] SIP support for hold and ringback for operator / 911 services
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Has anyone looked into how the orig-hold and ringback function should be
supported in SIP?  I realize that the wording may not be fully inline with
the SIP model, but this is a real issue to address. 

 

Note that some of the terms below are in quotes since they are TDM telephony
terms but are used in this context to represent logical equivalent. For
example, "offhook" represents the action to initiate a call and "onhook"
represents the action to end the call.

 

Orig-hold refers to the ability to keep the call even when the originator
goes "onhook" before the operator releases the call. The 911 operator is
notified of such event (i.e., the caller has gone "onhook") so that the
operator can initiate ringback (a special physical ringing is applied.) When
the caller goes "offhook", no "dial tone" is provided. Rather the caller is
reconnected to the original call to the 911 operator.  

 

When the caller is served by a softswitch which is connected to the 911
tandem via a SIP interface, SIP support for such functionality is required.

 

Let's assume UA_a represents the caller, and UA_b represents the 911
operator.

 

CMSS has a DCS-OSPS header (tag value = RING) which provides the mechanism
for UA_b to request the "physical ringback".  However the following pieces
are missing.

1.    Some way to communicate the caller's capabilities, e.g., supports
orig-hold or not

2.    Some mechanism to allow UA_b to request UA_a to apply orig-hold for
this call.

3.    Some mechanism for UA_b to subscribe to UA_a's "onhook" or "offhook"
state changes.  One can argue whether or not these are session states...

4.    Some mechanism for UA_a to notify UA_b the state changes of UA_a

 

For item 1, will the ALLOW header be appropriate?

For item 2, the DCS-OSPS header (tagvalue=RING) can be used

For item 3 and 4, will the SUBSCRIBE / NOTIFY mechanism be appropriate? 

 

Any thoughts, comment?


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<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>Has anyone looked into how the orig-hold and ringback
function should be supported in SIP?&nbsp; I realize that the wording may not be
fully inline with the SIP model, but this is a real issue to address. </span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>Note that some of the terms below are in quotes since
they are TDM telephony terms but are used in this context to represent logical
equivalent. For example, "offhook" represents the action to
initiate a call and "onhook" represents the action to end the call.</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>Orig-hold refers to the ability to keep the call even
when the originator goes &quot;onhook&quot; before the operator releases the
call. The 911 operator is notified of such event (i.e., the caller has gone
&quot;onhook&quot;) so that the operator can initiate ringback (a special
physical ringing is applied.) When the caller goes &quot;offhook&quot;, no
&quot;dial tone&quot; is provided. Rather the caller is reconnected to the
original call to the 911 operator. &nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>When the caller is served by a softswitch which is
connected to the 911 tandem via a SIP interface, SIP support for such
functionality is required.</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>Let's assume UA_a represents the caller, and UA_b
represents the 911 operator.</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>CMSS has a DCS-OSPS header (tag value = RING) which
provides the mechanism for UA_b to request the "physical ringback".&nbsp;
However the following pieces are missing.</span></font></p>

<p class=MsoPlainText style='margin-left:.5in;text-indent:-.25in'><font size=2
color="#993366" face="Courier New"><span style='font-size:10.0pt'>1.<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;
</span></font></span></font><span dir=LTR>Some way to communicate the caller's
capabilities, e.g., supports orig-hold or not</span></p>

<p class=MsoPlainText style='margin-left:.5in;text-indent:-.25in'><font size=2
color="#993366" face="Courier New"><span style='font-size:10.0pt'>2.<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;
</span></font></span></font><span dir=LTR>Some mechanism to allow UA_b to
request UA_a to apply orig-hold for this call.</span></p>

<p class=MsoPlainText style='margin-left:.5in;text-indent:-.25in'><font size=2
color="#993366" face="Courier New"><span style='font-size:10.0pt'>3.<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;
</span></font></span></font><span dir=LTR>Some mechanism for UA_b to subscribe
to UA_a's "onhook" or "offhook" state
changes.&nbsp; One can argue whether or not these are session states...</span></p>

<p class=MsoPlainText style='margin-left:.5in;text-indent:-.25in'><font size=2
color="#993366" face="Courier New"><span style='font-size:10.0pt'>4.<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;
</span></font></span></font><span dir=LTR>Some mechanism for UA_a to notify UA_b
the state changes of UA_a</span></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>For item 1, will the ALLOW header be appropriate?</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>For item 2, the DCS-OSPS header (tagvalue=RING) can be
used</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>For item 3 and 4, will the SUBSCRIBE / NOTIFY
mechanism be appropriate? </span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>&nbsp;</span></font></p>

<p class=MsoPlainText><font size=2 color="#993366" face="Courier New"><span
style='font-size:10.0pt'>Any thoughts, comment?</span></font></p>

</div>

</body>

</html>

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From: =?utf-8?B?7Zmp7KeE6rK9?= <jkhwang@kt.co.kr>
To: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
References: <BC523AB5.312C6%fluffy@cisco.com>
Subject: Re: [Sipping] draft-hwang-sipping-infomidcall-00.txt
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Hi, some further 3gpp-comments

What would be the first thought of Joe Citizen on vacation somewhere abroad driving off the road desparately needing assistance, calling 911 or the local xyz ??
Due to the mobility, to me it makes sense to do the analysis whether it is an emergency call or not in the handset in 3gpp. With the current mechanisms the network can download local emergency numbers to the handset, and depending on country, availability and configurastion of (U)SIM, the handset decides whether a normal call setup is placed (that may be analyzed in the switch and routed to the emergency centre anyway) or an emergency call setup is placed. This is in the end about end-user perception and tradeoffs between different numbering plans, analyzing the number based on the origin of the caller might be some administrative work.... 

Be also aware that local policy may decide whether (U)SIM-less emergency calls are allowed or not, and that the support of 118 in 3gpp only apply in case the (U)SIM is not present, thus other calls than emergency will always fail.


Another reason for having an EMERGENCY SETUP is that this particular message is easier to recognise by lower layers in the handset as well as the network as it has to recognise the message to perform particular characteristics and priority treatment. 

Anyway, as already stated by Keith, if the dialled number is analyzed in the handset and found to be an emergency call, is set up without any digits. The optional indication of fire brigade, marine, etc, is to my knowlede not commonly used (if used at all). It is difficult to predict end-user actions for how an accident is perceived. May such indicators may be more useful for e.g. telemetry-style of handsets where the handset report fire or intruders??

To me the most attractive solution for the emergency service is to use ONE number/URI for routing and have indicators that might help the PSAP to act as professional and coordinated as possible. The 911 approac is attractive to me, compared to countries where e.g. 110, 112 and 113 are used for different purposes but end up in the same PSAP.

For sure a dedicated URI for targeting e.g. the local police, doctors, vererinary, help for abused childeren or the nearest hard rock cafe may be useful, but outside the scope ;-) 
To me the emergency service involves two parties; the end user reporting an accident and the PSAP providing appropriate help and coordinating the assistance.

/atle

Best Regards

Atle Monrad

ERICSSON

Wireless Data Terminals & Solutions

Office: + 47 372 93 665
Mobile: + 47 454 10 665
Fax: + 47 372 94 058
E-mail: atle.monrad@ericsson.com
Web-site: www.ericsson.com/mobileplatforms

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of Mpierce1@aol.com
Sent: Friday, February 13, 2004 8:55 PM
To: hgs@cs.columbia.edu; drage@lucent.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Concern within sos ID - user addresses reserved


In a message dated 2/13/2004 8:52:05 AM Eastern Standard Time, hgs@cs.columbia.edu writes: 




Thanks for the pointer. From the description, Mike should talk to 3GPP 
about his concerns :-) 

http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip 
(SP-22) has pretty much the mechanisms described in the SOS draft.




[MAP] The main different being the lack of a SIM, or personalization device, in the SOS draft. That leaves it with following the GSM case "without SIM", which, as noted below, means that many different numbers, including 118, will be reported as emergency. If it is true that 118 is Directory Assistance in Finland, as the SOS draft states, then this operation (using GSM, 3GPP, SIP, or a rotary dial phone) can not be allowed in Finland). (Anyone in Finland know what is done there?) 

Guess if I was involved in 3GPP I would take this up with them, but I'm not. Certainly looks like a mistake that the actual dialled digits are not included in the Emergency Setup message, which would allow the switch in the network to sort it out! 




Note, in particular, in Section 10.1: 

"The user friendly MMI that specifies the type of emergency directly 
(e.g. menu) should be supported for use in any (i.e. home or visited) 
PLMN to avoid the mis-connection in roaming case.  This shall be allowed 
both with and without SIM/USIM being present." 

and 10.1.1 

"a)    112 and 911 shall always be available. These numbers shall be stored 
on the ME. 
b)    Any emergency number stored on a SIM/USIM when the SIM/USIM is present. 
c)    000, 08, 110, 999, 118 and 119 when a SIM/USIM is not present. These 
numbers shall be stored on the ME. 
d)    Additional emergency numbers that may have been downloaded by the 
serving network when the SIM/USIM is present." 





Mike Pierce 




This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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<DIV>
<P><FONT size=2>Hi<SPAN class=037495109-14022004>, some further 
3gpp-comments</SPAN></FONT></P>
<P><FONT size=2>What would be the first thought of Joe Citizen on vacation 
somewhere abroad driving off the road desparately needing assistance, calling 
911 or the local xyz ??<BR></FONT><FONT size=2>Due to the mobility, to me it 
makes sense to do the analysis whether it is an emergency call or not in the 
handset in 3gpp. With the current mechanisms the network can download local 
emergency numbers to the handset, and depending on country, availability and 
configurastion of (U)SIM, the handset decides whether a normal call setup is 
placed (that may be analyzed in the switch and routed to the emergency centre 
anyway) or an emergency call setup is placed. This is in the end about end-user 
perception and tradeoffs between different numbering plans, analyzing the number 
based on the origin of the caller might be some administrative work.... 
</FONT></P>
<P><FONT size=2>Be also aware that local policy may decide whether (U)SIM-less 
emergency calls are allowed or not, and that the support of 118 in 3gpp only 
apply in case the (U)SIM is not present, thus other calls than emergency will 
always fail.<BR></FONT></P>
<P><FONT size=2>Another reason for having an EMERGENCY SETUP is that this 
particular message is easier to recognise by lower layers in the handset a<SPAN 
class=037495109-14022004>s well as the</SPAN>&nbsp;network as it has to 
recognise the message to perform particular characteristics and priority 
treatment. </FONT></P>
<P><FONT size=2>Anyway, as already stated by Keith, if the dialled number is 
analyzed in the handset and found to be an emergency call, is set up without any 
digits. The optional indication of fire brigade, marine, etc, is to my knowlede 
not commonly used (if used at all). It is difficult to&nbsp;<SPAN 
class=037495109-14022004>predict </SPAN>end-user actions&nbsp;<SPAN 
class=037495109-14022004>for </SPAN><SPAN class=037495109-14022004>how an 
accident is perceived. May s</SPAN>uch indicators may be mo<SPAN 
class=037495109-14022004>re</SPAN> useful&nbsp;<SPAN 
class=037495109-14022004>for </SPAN>e.g. telemetry-style of handsets where the 
handset report fire or intruders<SPAN 
class=037495109-14022004>??</SPAN></FONT></P>
<P><FONT size=2>To me the most attractive solution for the emergency 
service<SPAN class=037495109-14022004> </SPAN>is to use ONE number/URI for 
routing and have indicators that might help the PSAP to act as professional and 
coordinated as possible. The 911 approac is attractive to me, compared to 
countries where e.g. 110, 112 and 113 are used for different purposes but end up 
in the same PSAP.</FONT></P>
<P><FONT size=2>For sure a dedicated URI for targeting e.g. the local police, 
doctors, vererinary, help for abused childeren or the nearest hard rock cafe may 
be useful, but&nbsp;<SPAN class=037495109-14022004>outside the scope ;-) 
<BR>T</SPAN></FONT><FONT size=2>o me the emergency service involves two parties; 
the end user reporting an accident and the PSAP providing appropriate help and 
coordinating the assistance.</FONT></P>
<P><FONT size=2>/atle</FONT></P>
<P><FONT size=2>Best Regards</FONT></P>
<P><FONT size=2>Atle Monrad</FONT></P><FONT color=#0000ff size=3>
<P>ERICSSON</P></FONT><FONT size=2>
<P>Wireless Data Terminals &amp; Solutions<BR><BR>Office: + 47 372 93 
665<BR>Mobile: + 47 454 10 665<BR>Fax: + 47 372 94 058<BR>E-mail: 
atle.monrad@ericsson.com<BR>Web-site: 
www.ericsson.com/mobileplatforms</P></FONT></DIV>
<BLOCKQUOTE>
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> sipping-admin@ietf.org 
  [mailto:sipping-admin@ietf.org]<B>On Behalf Of 
  </B>Mpierce1@aol.com<BR><B>Sent:</B> Friday, February 13, 2004 8:55 
  PM<BR><B>To:</B> hgs@cs.columbia.edu; drage@lucent.com<BR><B>Cc:</B> 
  sipping@ietf.org<BR><B>Subject:</B> Re: [Sipping] Concern within sos ID - user 
  addresses reserved<BR><BR></FONT></DIV><FONT face=arial,helvetica><FONT size=2 
  PTSIZE="10">In a message dated 2/13/2004 8:52:05 AM Eastern Standard Time, 
  hgs@cs.columbia.edu writes: <BR><BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE">Thanks for the pointer. From the description, Mike should talk 
    to 3GPP <BR>about his concerns :-) 
    <BR><BR>http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip 
    <BR>(SP-22) has pretty much the mechanisms described in the SOS 
  draft.</BLOCKQUOTE><BR><BR><BR>[MAP] The main different being the lack of a 
  SIM, or personalization device, in the SOS draft. That leaves it with 
  following the GSM case "without SIM", which, as noted below, means that many 
  different numbers, including 118, will be reported as emergency. If it is true 
  that 118 is Directory Assistance in Finland, as the SOS draft states, then 
  this operation (using GSM, 3GPP, SIP, or a rotary dial phone) can not be 
  allowed in Finland). (Anyone in Finland know what is done there?) 
  <BR><BR>Guess if I was involved in 3GPP I would take this up with them, but 
  I'm not. Certainly looks like a mistake that the actual dialled digits are not 
  included in the Emergency Setup message, which would allow the switch in the 
  network to sort it out! <BR><BR>
  <BLOCKQUOTE 
  style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px" 
  TYPE="CITE"><BR>Note, in particular, in Section 10.1: <BR><BR>"The user 
    friendly MMI that specifies the type of emergency directly <BR>(e.g. menu) 
    should be supported for use in any (i.e. home or visited) <BR>PLMN to avoid 
    the mis-connection in roaming case. &nbsp;This shall be allowed <BR>both 
    with and without SIM/USIM being present." <BR><BR>and 10.1.1 <BR><BR>"a) 
    &nbsp;&nbsp;&nbsp;112 and 911 shall always be available. These numbers shall 
    be stored <BR>on the ME. <BR>b) &nbsp;&nbsp;&nbsp;Any emergency number 
    stored on a SIM/USIM when the SIM/USIM is present. <BR>c) 
    &nbsp;&nbsp;&nbsp;000, 08, 110, 999, 118 and 119 when a SIM/USIM is not 
    present. These <BR>numbers shall be stored on the ME. <BR>d) 
    &nbsp;&nbsp;&nbsp;Additional emergency numbers that may have been downloaded 
    by the <BR>serving network when the SIM/USIM is present." 
  <BR></BLOCKQUOTE><BR></FONT><FONT lang=0 style="BACKGROUND-COLOR: #ffffff" 
  face=Arial color=#000000 size=3 PTSIZE="10" FAMILY="SANSSERIF" 
  BACK="#ffffff"><BR><BR>Mike Pierce 
<BR><BR></BLOCKQUOTE></FONT></FONT><br>This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.<br><br>E-mail including attachments is susceptible to data corruption, interruption, unauthorized amendment, tampering and viruses, and we only send and receive e-mails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.<br></BODY></HTML>

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Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
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Atle Monrad wrote:
=20
>Be also aware that local policy may decide whether (U)SIM-less
emergency calls are allowed or not, and that the support of 118 >in 3gpp
only apply in case the (U)SIM is not present, thus other calls than
emergency will always fail.

As far as I know it is required in GSM (3gpp) that the handset SHALL be
able to make emergency calls to=20
112 if no SIM is available. Of course you also SHALL be able to use 112
in case the SIM in present, but then
you also are able to use other local numbers.
=20
Since we are primarily not talking only 3gpp here, but SIP in general,
especially about VoIP and=20
other IP communications, the problem of _local_ data is still there,
even if you use SIM cards. We=20
are currently investigating the usage of SIM cards for
SIP-authentication and WiFi roaming and in=20
this case the SIP server and _HLR_ responsible may be anywhere in the
world, so you may not be able to download _local_ information being
nomadic.
=20
So we are back to square one regarding location information.
=20
regards
Richard
=20

	-----Original Message-----
	From: Atle Monrad (GR/ETO) [mailto:atle.monrad@ericsson.com]=20
	Sent: Saturday, February 14, 2004 10:57 AM
	To: Mpierce1@aol.com; hgs@cs.columbia.edu; drage@lucent.com
	Cc: sipping@ietf.org
	Subject: RE: [Sipping] Concern within sos ID - user addresses
reserved
=09
=09

	Hi, some further 3gpp-comments

	What would be the first thought of Joe Citizen on vacation
somewhere abroad driving off the road desparately needing assistance,
calling 911 or the local xyz ??
	Due to the mobility, to me it makes sense to do the analysis
whether it is an emergency call or not in the handset in 3gpp. With the
current mechanisms the network can download local emergency numbers to
the handset, and depending on country, availability and configurastion
of (U)SIM, the handset decides whether a normal call setup is placed
(that may be analyzed in the switch and routed to the emergency centre
anyway) or an emergency call setup is placed. This is in the end about
end-user perception and tradeoffs between different numbering plans,
analyzing the number based on the origin of the caller might be some
administrative work....=20

	Be also aware that local policy may decide whether (U)SIM-less
emergency calls are allowed or not, and that the support of 118 in 3gpp
only apply in case the (U)SIM is not present, thus other calls than
emergency will always fail.
=09

	Another reason for having an EMERGENCY SETUP is that this
particular message is easier to recognise by lower layers in the handset
as well as the network as it has to recognise the message to perform
particular characteristics and priority treatment.=20

	Anyway, as already stated by Keith, if the dialled number is
analyzed in the handset and found to be an emergency call, is set up
without any digits. The optional indication of fire brigade, marine,
etc, is to my knowlede not commonly used (if used at all). It is
difficult to predict end-user actions for how an accident is perceived.
May such indicators may be more useful for e.g. telemetry-style of
handsets where the handset report fire or intruders??

	To me the most attractive solution for the emergency service is
to use ONE number/URI for routing and have indicators that might help
the PSAP to act as professional and coordinated as possible. The 911
approac is attractive to me, compared to countries where e.g. 110, 112
and 113 are used for different purposes but end up in the same PSAP.

	For sure a dedicated URI for targeting e.g. the local police,
doctors, vererinary, help for abused childeren or the nearest hard rock
cafe may be useful, but outside the scope ;-)=20
	To me the emergency service involves two parties; the end user
reporting an accident and the PSAP providing appropriate help and
coordinating the assistance.

	/atle

	Best Regards

	Atle Monrad

	ERICSSON

	Wireless Data Terminals & Solutions
=09
	Office: + 47 372 93 665
	Mobile: + 47 454 10 665
	Fax: + 47 372 94 058
	E-mail: atle.monrad@ericsson.com
	Web-site: www.ericsson.com/mobileplatforms

		-----Original Message-----
		From: sipping-admin@ietf.org
[mailto:sipping-admin@ietf.org]On Behalf Of Mpierce1@aol.com
		Sent: Friday, February 13, 2004 8:55 PM
		To: hgs@cs.columbia.edu; drage@lucent.com
		Cc: sipping@ietf.org
		Subject: Re: [Sipping] Concern within sos ID - user
addresses reserved
	=09
	=09
		In a message dated 2/13/2004 8:52:05 AM Eastern Standard
Time, hgs@cs.columbia.edu writes:=20
	=09
	=09
	=09

			Thanks for the pointer. From the description,
Mike should talk to 3GPP=20
			about his concerns :-)=20
		=09
=09
http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.zip=20
			(SP-22) has pretty much the mechanisms described
in the SOS draft.




		[MAP] The main different being the lack of a SIM, or
personalization device, in the SOS draft. That leaves it with following
the GSM case "without SIM", which, as noted below, means that many
different numbers, including 118, will be reported as emergency. If it
is true that 118 is Directory Assistance in Finland, as the SOS draft
states, then this operation (using GSM, 3GPP, SIP, or a rotary dial
phone) can not be allowed in Finland). (Anyone in Finland know what is
done there?)=20
	=09
		Guess if I was involved in 3GPP I would take this up
with them, but I'm not. Certainly looks like a mistake that the actual
dialled digits are not included in the Emergency Setup message, which
would allow the switch in the network to sort it out!=20
	=09
	=09


			Note, in particular, in Section 10.1:=20
		=09
			"The user friendly MMI that specifies the type
of emergency directly=20
			(e.g. menu) should be supported for use in any
(i.e. home or visited)=20
			PLMN to avoid the mis-connection in roaming
case.  This shall be allowed=20
			both with and without SIM/USIM being present."=20
		=09
			and 10.1.1=20
		=09
			"a)    112 and 911 shall always be available.
These numbers shall be stored=20
			on the ME.=20
			b)    Any emergency number stored on a SIM/USIM
when the SIM/USIM is present.=20
			c)    000, 08, 110, 999, 118 and 119 when a
SIM/USIM is not present. These=20
			numbers shall be stored on the ME.=20
			d)    Additional emergency numbers that may have
been downloaded by the=20
			serving network when the SIM/USIM is present."=20
		=09


	=09
	=09
		Mike Pierce=20
	=09
	=09


	This communication is confidential and intended solely for the
addressee(s). Any unauthorized review, use, disclosure or distribution
is prohibited. If you believe this message has been sent to you in
error, please notify the sender by replying to this transmission and
delete the message without disclosing it. Thank you.
=09
	E-mail including attachments is susceptible to data corruption,
interruption, unauthorized amendment, tampering and viruses, and we only
send and receive e-mails on the basis that we are not liable for any
such corruption, interception, amendment, tampering or viruses or any
consequences thereof.
=09


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<BODY>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>Atle=20
Monrad wrote:</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2><FONT=20
face=3D"Times New Roman" color=3D#000000>&gt;Be also aware that local =
policy may=20
decide whether (U)SIM-less emergency calls are allowed or not, and that =
the=20
support of 118 &gt;in 3gpp only apply in case the (U)SIM is not present, =
thus=20
other calls than emergency will always =
fail.</FONT><BR></FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>As far=20
as I know it is required in GSM (3gpp) that the handset SHALL be able to =
make=20
emergency calls to </FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>112 if=20
no SIM is available. Of course you also&nbsp;SHALL be able to&nbsp;use =
112 in=20
case the SIM in present, but then</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>you=20
also are able to use other local numbers.</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>Since=20
we are primarily not talking&nbsp;only&nbsp;3gpp here, but SIP in =
general,=20
especially about VoIP and </FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>other=20
IP communications, </FONT></SPAN><SPAN class=3D608095808-16022004><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>the problem of _local_ data is still there, =
even if you use=20
SIM cards. We </FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>are=20
currently investigating the usage </FONT></SPAN><SPAN=20
class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>of SIM cards for=20
SIP-authentication and WiFi roaming and in </FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>this=20
case the SIP server and _HLR_ responsible </FONT></SPAN><SPAN=20
class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>may be anywhere=20
in the world, so you may not be able to download _local_ information =
being=20
nomadic.</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =
size=3D2>So we=20
are back to square one regarding location =
information.</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2>regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2>Richard</FONT></SPAN></DIV>
<DIV><SPAN class=3D608095808-16022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> Atle =
Monrad=20
  (GR/ETO) [mailto:atle.monrad@ericsson.com] <BR><B>Sent:</B> Saturday, =
February=20
  14, 2004 10:57 AM<BR><B>To:</B> Mpierce1@aol.com; hgs@cs.columbia.edu; =

  drage@lucent.com<BR><B>Cc:</B> sipping@ietf.org<BR><B>Subject:</B> RE: =

  [Sipping] Concern within sos ID - user addresses =
reserved<BR><BR></FONT></DIV>
  <DIV>
  <P><FONT size=3D2>Hi<SPAN class=3D037495109-14022004>, some further=20
  3gpp-comments</SPAN></FONT></P>
  <P><FONT size=3D2>What would be the first thought of Joe Citizen on =
vacation=20
  somewhere abroad driving off the road desparately needing assistance, =
calling=20
  911 or the local xyz ??<BR></FONT><FONT size=3D2>Due to the mobility, =
to me it=20
  makes sense to do the analysis whether it is an emergency call or not =
in the=20
  handset in 3gpp. With the current mechanisms the network can download =
local=20
  emergency numbers to the handset, and depending on country, =
availability and=20
  configurastion of (U)SIM, the handset decides whether a normal call =
setup is=20
  placed (that may be analyzed in the switch and routed to the emergency =
centre=20
  anyway) or an emergency call setup is placed. This is in the end about =

  end-user perception and tradeoffs between different numbering plans, =
analyzing=20
  the number based on the origin of the caller might be some =
administrative=20
  work.... </FONT></P>
  <P><FONT size=3D2>Be also aware that local policy may decide whether =
(U)SIM-less=20
  emergency calls are allowed or not, and that the support of 118 in =
3gpp only=20
  apply in case the (U)SIM is not present, thus other calls than =
emergency will=20
  always fail.<BR></FONT></P>
  <P><FONT size=3D2>Another reason for having an EMERGENCY SETUP is that =
this=20
  particular message is easier to recognise by lower layers in the =
handset=20
  a<SPAN class=3D037495109-14022004>s well as the</SPAN>&nbsp;network as =
it has to=20
  recognise the message to perform particular characteristics and =
priority=20
  treatment. </FONT></P>
  <P><FONT size=3D2>Anyway, as already stated by Keith, if the dialled =
number is=20
  analyzed in the handset and found to be an emergency call, is set up =
without=20
  any digits. The optional indication of fire brigade, marine, etc, is =
to my=20
  knowlede not commonly used (if used at all). It is difficult =
to&nbsp;<SPAN=20
  class=3D037495109-14022004>predict </SPAN>end-user actions&nbsp;<SPAN=20
  class=3D037495109-14022004>for </SPAN><SPAN =
class=3D037495109-14022004>how an=20
  accident is perceived. May s</SPAN>uch indicators may be mo<SPAN=20
  class=3D037495109-14022004>re</SPAN> useful&nbsp;<SPAN=20
  class=3D037495109-14022004>for </SPAN>e.g. telemetry-style of handsets =
where the=20
  handset report fire or intruders<SPAN=20
  class=3D037495109-14022004>??</SPAN></FONT></P>
  <P><FONT size=3D2>To me the most attractive solution for the emergency =

  service<SPAN class=3D037495109-14022004> </SPAN>is to use ONE =
number/URI for=20
  routing and have indicators that might help the PSAP to act as =
professional=20
  and coordinated as possible. The 911 approac is attractive to me, =
compared to=20
  countries where e.g. 110, 112 and 113 are used for different purposes =
but end=20
  up in the same PSAP.</FONT></P>
  <P><FONT size=3D2>For sure a dedicated URI for targeting e.g. the =
local police,=20
  doctors, vererinary, help for abused childeren or the nearest hard =
rock cafe=20
  may be useful, but&nbsp;<SPAN class=3D037495109-14022004>outside the =
scope ;-)=20
  <BR>T</SPAN></FONT><FONT size=3D2>o me the emergency service involves =
two=20
  parties; the end user reporting an accident and the PSAP providing =
appropriate=20
  help and coordinating the assistance.</FONT></P>
  <P><FONT size=3D2>/atle</FONT></P>
  <P><FONT size=3D2>Best Regards</FONT></P>
  <P><FONT size=3D2>Atle Monrad</FONT></P><FONT color=3D#0000ff =
size=3D3>
  <P>ERICSSON</P></FONT><FONT size=3D2>
  <P>Wireless Data Terminals &amp; Solutions<BR><BR>Office: + 47 372 93=20
  665<BR>Mobile: + 47 454 10 665<BR>Fax: + 47 372 94 058<BR>E-mail:=20
  atle.monrad@ericsson.com<BR>Web-site:=20
  www.ericsson.com/mobileplatforms</P></FONT></DIV>
  <BLOCKQUOTE>
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> =
sipping-admin@ietf.org=20
    [mailto:sipping-admin@ietf.org]<B>On Behalf Of=20
    </B>Mpierce1@aol.com<BR><B>Sent:</B> Friday, February 13, 2004 8:55=20
    PM<BR><B>To:</B> hgs@cs.columbia.edu; drage@lucent.com<BR><B>Cc:</B> =

    sipping@ietf.org<BR><B>Subject:</B> Re: [Sipping] Concern within sos =
ID -=20
    user addresses reserved<BR><BR></FONT></DIV><FONT =
face=3Darial,helvetica><FONT=20
    size=3D2 PTSIZE=3D"10">In a message dated 2/13/2004 8:52:05 AM =
Eastern Standard=20
    Time, hgs@cs.columbia.edu writes: <BR><BR><BR>
    <BLOCKQUOTE=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px"=20
    TYPE=3D"CITE">Thanks for the pointer. From the description, Mike =
should talk=20
      to 3GPP <BR>about his concerns :-)=20
      =
<BR><BR>http://www.3gpp.org/ftp/Specs/archive/22_series/22.101/22101-660.=
zip=20
      <BR>(SP-22) has pretty much the mechanisms described in the SOS=20
    draft.</BLOCKQUOTE><BR><BR><BR>[MAP] The main different being the =
lack of a=20
    SIM, or personalization device, in the SOS draft. That leaves it =
with=20
    following the GSM case "without SIM", which, as noted below, means =
that many=20
    different numbers, including 118, will be reported as emergency. If =
it is=20
    true that 118 is Directory Assistance in Finland, as the SOS draft =
states,=20
    then this operation (using GSM, 3GPP, SIP, or a rotary dial phone) =
can not=20
    be allowed in Finland). (Anyone in Finland know what is done there?) =

    <BR><BR>Guess if I was involved in 3GPP I would take this up with =
them, but=20
    I'm not. Certainly looks like a mistake that the actual dialled =
digits are=20
    not included in the Emergency Setup message, which would allow the =
switch in=20
    the network to sort it out! <BR><BR>
    <BLOCKQUOTE=20
    style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff =
2px solid; MARGIN-RIGHT: 0px"=20
    TYPE=3D"CITE"><BR>Note, in particular, in Section 10.1: <BR><BR>"The =
user=20
      friendly MMI that specifies the type of emergency directly =
<BR>(e.g. menu)=20
      should be supported for use in any (i.e. home or visited) <BR>PLMN =
to=20
      avoid the mis-connection in roaming case. &nbsp;This shall be =
allowed=20
      <BR>both with and without SIM/USIM being present." <BR><BR>and =
10.1.1=20
      <BR><BR>"a) &nbsp;&nbsp;&nbsp;112 and 911 shall always be =
available. These=20
      numbers shall be stored <BR>on the ME. <BR>b) =
&nbsp;&nbsp;&nbsp;Any=20
      emergency number stored on a SIM/USIM when the SIM/USIM is =
present. <BR>c)=20
      &nbsp;&nbsp;&nbsp;000, 08, 110, 999, 118 and 119 when a SIM/USIM =
is not=20
      present. These <BR>numbers shall be stored on the ME. <BR>d)=20
      &nbsp;&nbsp;&nbsp;Additional emergency numbers that may have been=20
      downloaded by the <BR>serving network when the SIM/USIM is =
present."=20
    <BR></BLOCKQUOTE><BR></FONT><FONT lang=3D0 =
style=3D"BACKGROUND-COLOR: #ffffff"=20
    face=3DArial color=3D#000000 size=3D3 PTSIZE=3D"10" BACK=3D"#ffffff" =

    FAMILY=3D"SANSSERIF"><BR><BR>Mike Pierce=20
  <BR><BR></BLOCKQUOTE></FONT></FONT><BR>This communication is =
confidential and=20
  intended solely for the addressee(s). Any unauthorized review, use, =
disclosure=20
  or distribution is prohibited. If you believe this message has been =
sent to=20
  you in error, please notify the sender by replying to this =
transmission and=20
  delete the message without disclosing it. Thank you.<BR><BR>E-mail =
including=20
  attachments is susceptible to data corruption, interruption, =
unauthorized=20
  amendment, tampering and viruses, and we only send and receive e-mails =
on the=20
  basis that we are not liable for any such corruption, interception, =
amendment,=20
  tampering or viruses or any consequences =
thereof.<BR></BLOCKQUOTE></BODY></HTML>
=00
------_=_NextPart_001_01C3F46D.BC67F338--

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Subject: RE: [Sipping] Concern within sos ID - user addresses reserved
To: "James M. Polk" <jmpolk@cisco.com>
Cc: "Rosen, Brian" <Brian.Rosen@marconi.com>, sipping@ietf.org
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I would assume that, in the first case, it would be the same as the circuit
switched version.  If the coastal jurisdiction currently has separate
numbers for "fire house" and "coast guard" (for instance), they must
already have some sort of coordination process in place.  That process
wouldn't need to be modified much.

Similarly, in a big city, I assume that the jurisdiction would arrange that
all the addresses (fire, police, rescue, etc) would go to the same place,
where the human attendant would make the dispatch decision based on the
information given, not just the destination address.

Janet


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                      "James M. Polk"                                                                                            
                      <jmpolk                  To:      "Rosen, Brian" <Brian.Rosen@marconi.com>, <sipping@ietf.org>             
                      @cisco.com>              cc:                                                                               
                                               Subject: RE: [Sipping] Concern within sos ID - user addresses reserved            
                      02/10/2004 05:52                                                                                           
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 




Suppose there is a fire on a boat near a harbor - which do you call?

But seriously, suppose there is a fire and the fire department is called
only. They show up and a person (who set the fire) is there showing his
gun. The fire department won't go near that situation until the police show

up. This happens in big cities.

Should the fire department now place a new call to the police department
for each of its sos@fire calls?







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Date: Mon, 16 Feb 2004 11:47:12 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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CC: Cullen Jennings <fluffy@cisco.com>, sipping <sipping@ietf.org>
Subject: Re: [Sipping] draft-hwang-sipping-infomidcall-00.txt
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I agree that these services are needed. However, mechanisms are in place=20
today for performing these services. Call transfer is done with REFER,=20
call waiting is a local feature on the device, and three way calling can=20
be done locally or through invocation of a centralized conferencing serve=
r.

In the SIp services model, all of these features do not live in a=20
centralized server in the network. Rather, they are distributed, and=20
implemented in the place most appropriate for that service. In an IP=20
network, the endpoints have greater intelligence than in the PSTN.=20
Having smarter endpoints means that you can have a better overall=20
service experience and support a broader range of devices.

MGCP takes the position you are advocating, that the services all live=20
in a centralized place. Cullen's point, I believe, is that, if the aim=20
is to develop protocols to support a model of centralized service=20
control, we already have that - mgcp.

Thanks,
Jonathan R.

=ED=99=A9=EC=A7=84=EA=B2=BD wrote:

> Thank you for the question.
>=20
> Your question is why network-based midcall is needed with SIP.
>=20
> My answer is that
> Since service components in the next gen IP network are based on SIP,
> the delivery of midcall information should be through SIP method.
>=20
> Simple INFO (from the SIP phone or from the MGC(by translation)) works =
very well,
> as we implemented trial services with this INFO midcall function both i=
n SIP phone and Softswitch.
>=20
> My proposal is that in standard SIP phone (UA) have midcall button simp=
ly initiating INFO=20
> when the button is pressed.
>=20
> -------------------------------------------
> Title: INFO Usage Examples for Network-based Mid-Call Service=20
> Author: Jinkyung Hwang (KT)
>=20
> Abstract:
>=20
>   This document describes the INFO usage examples for network based=20
>   mid-call service. Mid-call based services are for example, Call=20
>   Transfer, Call Waiting, and Three-Way Calling. Other possible mid-
>   call services are useful and important, even more in IP multimedia=20
>   network. The cases described in SIP service examples (draft-ietf-
>   sipping-service-examples-05) are Terminal-basis. Since the service=20
>   procedures are fixed in the Terminal, the features are not extensible=
.=20
>   This document proposes network-based mid-call control mechanism with=20
>   INFO and shows examples that simple mid-call request (INFO) from the=20
>   Terminal, then the network application server provide various mid-cal=
l=20
>   services by prompting user through the Media Server, according to the=
=20
>   subscription information of the user.=20
>=20
> A URL for this draft is=20
>=20
>  http://www.ietf.org/internet-drafts/draft-hwang-sipping-infomidcall-00=
.txt
>=20
>=20
> BR
>=20
> Jinkyung Hwang
> KT
>=20
> ----- Original Message -----=20
> From: "Cullen Jennings" <fluffy@cisco.com>
> To: "sipping" <sipping@ietf.org>
> Sent: Saturday, February 14, 2004 1:26 AM
> Subject: [Sipping] draft-hwang-sipping-infomidcall-00.txt
>=20
>=20
>=20
>>   =20
>>Why not just use MGCP instead of SIP if you want to do this. It will wo=
rk
>>much better.=20
>>
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>J*ix&j)b	bm?=0C0'~ffX)=DF=A3"8bX+=1FDY=D7=AFzZ)ayi'=1A*'=3DK=1E*b^?=1C%=
fin~-w.=C8=A9R=C7=AC++=1Dz(}+y"=0F

--=20
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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Date: Mon, 16 Feb 2004 13:05:15 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] URGENT: query on another reg-event change
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Folks,

Authors 48 hours have started for reg-event. As you may recall, I had 
found a fatal problem in the schema design, and there was agreement on 
the list on a proposal to fix it. While making the changes in the 
schema, another question came up.

The question is regarding the "params" attribute of the "contact" 
element. Currently, this parameter is meant to contain any of the 
contact parameters not otherwise called out explicitly in other 
attributes (for example, the q-value is contained in the "q" attribute. 
So, if I had a contact like this:

REGISTER
To: sip:user@example.com
Contact: sip:user@pc.example.com;q=1.0;newparam=foo;audio

then the resulting XML would look like (using the new schema I proposed 
on the list):

  <?xml version="1.0"?>
    <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
                 version="0" state="full">
      <registration aor="sip:user@example.com" id="as9"
                    state="active">
        <contact id="76" state="active" event="registered"
                 duration-registered="7322"
                 q="0.8" params="newparam=foo;audio">
           <uri>sip:user@pc887.example.com</uri>
        </contact>
      </registration>
    </reginfo>

Notice how the contact parameters are basically passed untouched in the 
"params" attribute. This makes sense if that is the only point of 
exensibility in the schema. But, now with the structured contact, one 
could represent these "properly" with formal schema definitions. i.e., 
if we did a reg-event extension to support caller-prefs, it could look like:

  <?xml version="1.0"?>
    <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
             xmlns:callerprefs="urn:ietf:params:xml:ns:callerprefs"
                 version="0" state="full">
      <registration aor="sip:user@example.com" id="as9"
                    state="active">
        <contact id="76" state="active" event="registered"
                 duration-registered="7322"
                 q="0.8" params="newparam=foo;audio">
           <uri>sip:user@pc887.example.com</uri>
           <callerprefs:audio/>
           <unknown-param name="newparam>foo</unknown-param>
        </contact>
      </registration>
    </reginfo>

That is, I could instead define a new element called "unknown-param" in 
reg-event that is used for unknown params, and for those that are known 
to the server, and for which more structured xml specifications exist, 
you could have things like <callerprefs:audio>.

The technical drawback here is that whether a parameter appears in 
unknown-param or whether its in something like callerprefs:audio depends 
on whether that particular extension (in this case, callerprefs) is 
supported by the server. Another choice is to not even include unknown 
parameters in the reg-event notifications, however I think thats a bad idea.

The benefit, however, is that it allows us to provide very structured 
notifications for contact header field extensions, and it avoids the 
need for implementing, for example, a caller-prefs parser just to 
receive notifications about caller preferences parameters - all the data 
is represented using XML constructs.

Technically, my preference would be to define the unknown-param in 
reg-event right now, and have extensions (like callerprefs and gruu) 
specify xml schema extensions to reg-event for their data.

Please provide comments on this proposal within the next few days.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: A Framework for Application Interaction in the Session Initiation Protocol (SIP)
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-sipping-app-interaction-framework-01.txt
	Pages		: 38
	Date		: 2004-2-16
	
This document describes a framework and requirements for the
interaction between users and Session Initiation Protocol (SIP) based
applications. By interacting with applications, users can guide the
way in which they operate. The focus of this framework is stimulus
signaling, which allows a user agent to interact with an application
without knowledge of the semantics of that application. Stimulus
signaling can occur to a user interface running locally with the
client, or to a remote user interface, through media streams.
Stimulus signaling encompasses a wide range of mechanisms, ranging
from clicking on hyperlinks, to pressing buttons, to traditional Dual
Tone Multi Frequency (DTMF) input. In all cases, stimulus signaling
is supported through the use of markup languages, which play a key
role in this framework.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction-framework-01.txt

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



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: A Framework for SIP User Agent Configuration
	Author(s)	: D. Petrie
	Filename	: draft-ietf-sipping-config-framework-02.txt
	Pages		: 22
	Date		: 2004-2-16
	
This document defines the application of a set of protocols for 
configuring a SIP user agent.  The SIP user agent must discover how 
and from where to retrieve its initial configuration and be notified 
of changes and updates which impact its configuration.  The 
objective is to define a means for automatically configuring a user 
agent such that it can be functional without user or administrative 
intervention.  The framework for discovery, delivery, notification 
and updates of user agent configuration is defined here.  This 
framework is also intended to ease ongoing administration, 
configuration and upgrading of large scale deployments of SIP user 
agents.  The contents and format of the configuration data to be 
defined is outside the scope of this document.

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Mon Feb 16 16:40:34 2004
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Xuewen Li <mli@nortelnetworks.com>
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Subject: Re: [Sipping] SIP support for hold and ringback for operator / 911
 services
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Here is my opinion FWIW.

You need to be careful how you define these services. You want to make 
sure they work and are sensible for a broad range of devices, including, 
for example, a PC application or a wireless phone, where there is no 
hook at all.

Firstly, I think it makes sense that, when the caller reaches 911, there 
is information in the 200 OK that indicates that the caller has reached 
an emergency service center. The callee capabilities framework would 
allow the 200 OK to INVITE to contain a parameter indicating this, for 
example, actor="emergency-center". When this is conveyed to the caller, 
the caller can take whatever UI action is necesary to indicate to them 
that they shouldnt try to hangup the call. Of course authentication is 
of the utmost importance here. Henning's emergency architecture draft 
alludes to such an indication but I didnt see any specific text on it.

Should the caller anyway send a BYE, the UAS (the emergency center) can 
reject that BYE with a new response code, to indicate that the call 
cannot be disconnected because its an emergency operator. This way, an 
automata would explicitly know that this was not some normal error, and 
that the UI should not be dismissed.

There is a temptation to try and add a state to the dialog event package 
so that the emergency operator can know that the caller is still trying 
to disconnect the call (ie, "on-hook"). This way, if the caller stops 
trying to disconnect, i.e., goes back "off-hook", the emergency service 
center would cease any attempts at interacting with the user to get them 
to pick up.

The danger here is that we are effectively using the dialog package to 
model states of the *user interface*. Thats a dangerous game and a 
slippery slope. The last thing I want to do is get into a mode where, in 
order to deploy a new service, I need to add states to the dialog event 
package to represent some new state in the UI that was added to support 
the feature.... this is one of the problems with the IN model, IMHO.

Indeed, this would lead me to conclude that the local ringback to alert 
the user to pick up is actually a LOCAL feature on the phone; if the 
user hangs up, and a BYE is sent, and a response code is received which 
indicates that the call is NOT terminated, the device itself would 
generate local rings to indicate to the user that they need to pick up 
as the call still exists. Such ringback would persist until the operator 
disconnects the call by sending a BYE. My argument here, and this 
argument is key - is that **this ringback is needed because of the 
physical nature of the UI, and the UI should not be of concern to the 
emergency service center**.

In the case of a PC application, if a user clicked "hangup" and received 
this error response code, it would tell the user about the problem and 
not terminate the call. In this case, the UI is different, and the 
"ringback" is not necessary to alert the user of the problem.

-Jonathan R.



Xuewen Li wrote:

> Has anyone looked into how the orig-hold and ringback function should be 
> supported in SIP?  I realize that the wording may not be fully inline 
> with the SIP model, but this is a real issue to address.
> 
>  
> 
> Note that some of the terms below are in quotes since they are TDM 
> telephony terms but are used in this context to represent logical 
> equivalent. For example, "offhook" represents the action to initiate a 
> call and "onhook" represents the action to end the call.
> 
>  
> 
> Orig-hold refers to the ability to keep the call even when the 
> originator goes "onhook" before the operator releases the call. The 911 
> operator is notified of such event (i.e., the caller has gone "onhook") 
> so that the operator can initiate ringback (a special physical ringing 
> is applied.) When the caller goes "offhook", no "dial tone" is provided. 
> Rather the caller is reconnected to the original call to the 911 operator.  
> 
>  
> 
> When the caller is served by a softswitch which is connected to the 911 
> tandem via a SIP interface, SIP support for such functionality is required.
> 
>  
> 
> Let's assume UA_a represents the caller, and UA_b represents the 911 
> operator.
> 
>  
> 
> CMSS has a DCS-OSPS header (tag value = RING) which provides the 
> mechanism for UA_b to request the "physical ringback".  However the 
> following pieces are missing.
> 
> 1.    Some way to communicate the caller's capabilities, e.g., supports 
> orig-hold or not
> 
> 2.    Some mechanism to allow UA_b to request UA_a to apply orig-hold 
> for this call.
> 
> 3.    Some mechanism for UA_b to subscribe to UA_a's "onhook" or 
> "offhook" state changes.  One can argue whether or not these are session 
> states...
> 
> 4.    Some mechanism for UA_a to notify UA_b the state changes of UA_a
> 
>  
> 
> For item 1, will the ALLOW header be appropriate?
> 
> For item 2, the DCS-OSPS header (tagvalue=RING) can be used
> 
> For item 3 and 4, will the SUBSCRIBE / NOTIFY mechanism be appropriate?
> 
>  
> 
> Any thoughts, comment?
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



From exim@www1.ietf.org  Mon Feb 16 16:47:03 2004
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] Update to session policy requirements
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Folks,

I've submitted an update to the session policy requirements. Until it 
appears in the archives, you can pick it up at:

http://www.jdrosen.net/papers/draft-ietf-sipping-session-policy-req-01.txt

The changes from the previous version are:

* separation of the requirements for policy-specific data (i.e., codec 
trimming) from policy primitives that are independent of the specific 
policy.

* added requirements for policies that are independent of the session, 
and learned by the device when it starts

No doubt the requirements will evolve as we work the specific protocol 
proposals. Updates to those drafts are pending as well.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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



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Message-ID: <403139B2.4080509@dynamicsoft.com>
Date: Mon, 16 Feb 2004 16:44:18 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sipping] Update to caller preferences use cases
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Folks,

I've submitted a minor update to the caller preferences use case 
specification. Until it appears in the archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-sipping-callerprefs-usecases-01.txt

The only changes were some bug fixes reported by Paul. There are still a 
few major todo's, and in particular, documentation of an example of how 
to implement the matching algorithm. I just didnt get around to it this 
time.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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



From exim@www1.ietf.org  Mon Feb 16 18:51:58 2004
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Date: Mon, 16 Feb 2004 18:43:24 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sipping] draft-ietf-sipping-config-framework: discovery
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I wonder why the UA doesn't (also) contact its home AOR, rather than 
just a local server discovered by DHCP. In many cases, this is what I 
want when I'm in a remote network. Clearly, this requires that the 
device knows its AOR and thus it can't start from a blank slate, but 
that information is available in many circumstances.

Henning

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



From exim@www1.ietf.org  Mon Feb 16 20:51:13 2004
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From: =?UTF-8?B?7Zmp7KeE6rK9?= <jkhwang@kt.co.kr>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
References: <BC523AB5.312C6%fluffy@cisco.com> <004701c3f443$c74d6470$71090693@ROMEO> <4030F410.1050801@dynamicsoft.com>
Subject: Re: [Sipping] draft-hwang-sipping-infomidcall-00.txt
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Date: Mon, 16 Feb 2004 22:38:17 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Xuewen Li <mli@nortelnetworks.com>, sipping@ietf.org
Subject: Re: [Sipping] SIP support for hold and ringback for operator / 911
 services
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I agree with Jonathan that end system intelligence is the answer here. 
After all, end systems have an interest in making emergency services work.

The -sos- draft (the version -01 that was refused since the -00 version 
was submitted just before the deadline...) defines the 'service' media 
feature tag for that, which includes plain 'sos' and the various 
sub-services (sos.fire, etc.). However, this is only necessary if the 
caller didn't know already that this was an emergency call. By the 
mechanism defined in the -sos- draft, it should, in almost all cases 
know this and thus be able to adjust UI behavior accordingly.

A simple 403 (Forbidden), in conjunction with the callee capabilities 
tag, should be sufficient as response to a premature BYE.

The problem with the approach of having the callee identify itself as a 
emergency call center and thereby modify caller behavior is that lends 
itself to pranks. Call my number and you can't hang up the phone until I 
let you. Telemarketers will love that feature :-)

(If the calling UA knows it's an emergency call, there is no need for 
the feature tag indication to control services.)

Jonathan Rosenberg wrote:


> Firstly, I think it makes sense that, when the caller reaches 911, there 
> is information in the 200 OK that indicates that the caller has reached 
> an emergency service center. The callee capabilities framework would 
> allow the 200 OK to INVITE to contain a parameter indicating this, for 
> example, actor="emergency-center". When this is conveyed to the caller, 
> the caller can take whatever UI action is necesary to indicate to them 
> that they shouldnt try to hangup the call. Of course authentication is 
> of the utmost importance here. Henning's emergency architecture draft 
> alludes to such an indication but I didnt see any specific text on it.
> 
> Should the caller anyway send a BYE, the UAS (the emergency center) can 
> reject that BYE with a new response code, to indicate that the call 
> cannot be disconnected because its an emergency operator. This way, an 
> automata would explicitly know that this was not some normal error, and 
> that the UI should not be dismissed.
> 
> There is a temptation to try and add a state to the dialog event package 
> so that the emergency operator can know that the caller is still trying 
> to disconnect the call (ie, "on-hook"). This way, if the caller stops 
> trying to disconnect, i.e., goes back "off-hook", the emergency service 
> center would cease any attempts at interacting with the user to get them 
> to pick up.
> 
> The danger here is that we are effectively using the dialog package to 
> model states of the *user interface*. Thats a dangerous game and a 
> slippery slope. The last thing I want to do is get into a mode where, in 
> order to deploy a new service, I need to add states to the dialog event 
> package to represent some new state in the UI that was added to support 
> the feature.... this is one of the problems with the IN model, IMHO.
> 
> Indeed, this would lead me to conclude that the local ringback to alert 
> the user to pick up is actually a LOCAL feature on the phone; if the 
> user hangs up, and a BYE is sent, and a response code is received which 
> indicates that the call is NOT terminated, the device itself would 
> generate local rings to indicate to the user that they need to pick up 
> as the call still exists. Such ringback would persist until the operator 
> disconnects the call by sending a BYE. My argument here, and this 
> argument is key - is that **this ringback is needed because of the 
> physical nature of the UI, and the UI should not be of concern to the 
> emergency service center**.
> 
> In the case of a PC application, if a user clicked "hangup" and received 
> this error response code, it would tell the user about the problem and 
> not terminate the call. In this case, the UI is different, and the 
> "ringback" is not necessary to alert the user of the problem.
> 


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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: Xuewen Li <mli@nortelnetworks.com>, sipping@ietf.org
Subject: Re: [Sipping] SIP support for hold and ringback for operator / 911
 services
References: <ABA227A15B80D511BD1A00508BF93A1C0EA62530@zrtpd0jq.us.nortel.com> <40313789.3010409@dynamicsoft.com> <40318CA9.9090201@cs.columbia.edu>
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Henning Schulzrinne wrote:

> I agree with Jonathan that end system intelligence is the answer here. 
> After all, end systems have an interest in making emergency services work.
> 
> The -sos- draft (the version -01 that was refused since the -00 version 
> was submitted just before the deadline...) defines the 'service' media 
> feature tag for that, which includes plain 'sos' and the various 
> sub-services (sos.fire, etc.). However, this is only necessary if the 
> caller didn't know already that this was an emergency call. By the 
> mechanism defined in the -sos- draft, it should, in almost all cases 
> know this and thus be able to adjust UI behavior accordingly.
> 
> A simple 403 (Forbidden), in conjunction with the callee capabilities 
> tag, should be sufficient as response to a premature BYE.

I really think you will want a new response code here. My concern is 
that an application will effectively ignore the 403 and terminate the 
call anyway. I'd rather have a very explicit response code, so that its 
clear that the call cannot be terminated because its an emergency call.

> 
> The problem with the approach of having the callee identify itself as a 
> emergency call center and thereby modify caller behavior is that lends 
> itself to pranks. Call my number and you can't hang up the phone until I 
> let you. Telemarketers will love that feature :-)

This seems like an S/MIME application that might actually even be 
deployable.

In principal, I would think that someone (NENA??) could hand out certs 
that do, in fact, certify that a user is an operator in an emergency 
service center. If a call I make completes to an user that has a cert 
declaring them as an emergency operation, my UA wouldn't allow me to 
terminate the call.


> 
> (If the calling UA knows it's an emergency call, there is no need for 
> the feature tag indication to control services.)

True. However, I can imagine cases where the call doesnt start as an 
emergency call, but turns into one. For example, what if I use some kind 
of call routing app where the destination number can be entered via 
stimulus (i.e., prepaid calling?). Maybe this is unsolvable, but the 
case does exist.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: [Sipping] Question regarding resource-priority header
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Hi,

In draft-ietf-sip-resource-priority-01.txt, it looks like there is a 
printing mistake in the grammar defined for Resource-Priority header . Its 
given as 
       Resource-Priority  =  "Resource-Priority" HCOLON
                              (*COMMA Resource-value)
 
But I guess, no header can start with a COMMA . It should actually be 
       Resource-Priority  =  "Resource-Priority" HCOLON
                              Resource-value(*COMMA Resource-value)

This is the way the grammar is given for Accept-Resource-Priority. 
Can anyone confirm on this pls? 

thanks in advance... 

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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="Courier New">In draft-ietf-sip-resource-priority-01.txt,
it looks like there is a printing mistake in the grammar defined for Resource-Priority
header . Its given as </font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp;Resource-Priority
&nbsp;= &nbsp;&quot;Resource-Priority&quot; HCOLON</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; (*COMMA
Resource-value)</font>
<br><font size=2 face="Courier New">&nbsp;</font>
<br><font size=2 face="Courier New">But I guess, no header can start with
a COMMA . It should actually be </font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp;Resource-Priority
&nbsp;= &nbsp;&quot;Resource-Priority&quot; HCOLON</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; </font><font size=2 color=blue face="Courier New">Resource-value</font><font size=2 face="Courier New">(*COMMA
Resource-value)</font>
<br>
<br><font size=2 face="Courier New">This is the way the grammar is given
for Accept-Resource-Priority. </font>
<br><font size=2 face="Courier New">Can anyone confirm on this pls? </font>
<br>
<br><font size=2 face="Courier New">thanks in advance... </font>
<br>
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Date: Mon, 16 Feb 2004 16:31:15 -0500
From: "David R. Oran" <oran@cisco.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sipping <sipping@ietf.org>
Subject: Re: [Sipping] URGENT: query on another reg-event change
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--On Monday, February 16, 2004 1:05 PM -0500 Jonathan Rosenberg=20
<jdrosen@dynamicsoft.com> wrote:

> Folks,
>
[snip]

> Technically, my preference would be to define the unknown-param in
> reg-event right now, and have extensions (like callerprefs and gruu)
> specify xml schema extensions to reg-event for their data.
>
I agree pretty strongly. The more precision and granularity we can give to=20
the data model, along with the ability to naturally extend this schema=20
without an ever-increasing "garbage can" of opaque data, the better off we=20
will be going forward. I don't think it's such a burden for a client to=20
have to look in two places to deal with servers that haven't caught up with =

the schema changes.

Dave.

> Please provide comments on this proposal within the next few days.
>
> Thanks,
> Jonathan R.
>

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> 
> This seems like an S/MIME application that might actually even be 
> deployable.
> 
> In principal, I would think that someone (NENA??) could hand out certs 
> that do, in fact, certify that a user is an operator in an emergency 
> service center. If a call I make completes to an user that has a cert 
> declaring them as an emergency operation, my UA wouldn't allow me to 
> terminate the call.

We've talked about this in the various emergency calling discussions and 
nobody who knows anything about the current 911 system, which is highly 
localized and decentralized, had any hope that this can work in the 
foreseeable future. We won't even talk about the difficulties of making 
this work internationally.

This gets more complicated in practice since you have the various 
'telematics' providers such as OnStar.

This is even more difficult than asserting an identity; Jon's 
trait-based auth draft, if I read it right, essentially says that trait 
(or role-based) authentication is only going to work in either a single 
administration or a federated one. The 911 system in the US is the 
opposite of both :-)

I've suggested to various NENA folks that they should look into this as 
a long-term goal, as it would simplify other things, but given that the 
solution is already specified for the callee caps case (in sos-01), 
there's probably not much you can do.

The 'cannot hang up during emergency calls' is mostly a canard - many 
existing analog systems don't have that property and nobody has been 
able to find a rule that says that this is a required feature. Given the 
many other ways you can lose contact with the ECC/PSAP, it is unclear 
that this adds much value in practice. The emergency-arch draft (like 
most existing systems) does specify call-back capability, which seems 
more useful and practical.

> 
> 
>>
>> (If the calling UA knows it's an emergency call, there is no need for 
>> the feature tag indication to control services.)
> 
> 
> True. However, I can imagine cases where the call doesnt start as an 
> emergency call, but turns into one. For example, what if I use some kind 
> of call routing app where the destination number can be entered via 
> stimulus (i.e., prepaid calling?). Maybe this is unsolvable, but the 
> case does exist.

I'm not disagreeing with the desirability of indicating that somebody 
answering is an emergency call taker. I'm concerned that we simply do 
"and then magic happens and everybody agrees on a CA for emergency 
calls" since the problem is then somebody else's.

Henning

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From: Cullen Jennings <fluffy@cisco.com>
To: Brian Rosen <Brian.Rosen@marconi.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
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One small nit on polygons. Need to specify that the vertexes are in
clockwise order. Failing to do this caused endless arguments in air traffic
control specifications of airspaces. Probably needs a way to deal with what
geographic coordinate grid is being used or fix it. Should warn implementers
that polygon will include the poles and the vertex may even be a pole. Polar
flights have caused more than one geographic information system to crash.
Many systems translate the 3 tupple coordinates into 4 tupple coordinates
for all calculations to avoid the singularities that a spherical coordinate
system introduces. Probably need a header format that allows an arbitrary
number of vertexes -  the top level polygons are going to be big. I don't
like the implied additive subtractive form of the poly. I would prefer poly
explicitly identify as additive or subtractive with the region covered being
formed by taking the union of all the additive then subtracting each of the
subtractive polygons. Points on a vertex or edge would be considered in the
region for additive polygons and out for subtractive polygons. None of the
polygons forming a region are allowed to share an edge. The altitudes need
to be altitude ranges to specify a volume that can be clipped. Seems like
you need this to figure out which floor in a building I am on. Recommend
that min alt is include in the volume but max alt is outside the volume. The
specification of points on the surface of the volume being in or out is
important or users can, excuse the pun, fall thought the cracks.

I was not sold on DNS to start with but your arguments about the pleasures
of forming a new global root convinced me DNS is right if it will work. More
text explaining it would work would make me feel warm and fluffy.

Cullen







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From: Cullen Jennings <fluffy@cisco.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>, <sipping@ietf.org>
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I think that a phone informing a user that SOS was broken is very important
for this to work when it is needed. I would like to see a bit more text
around the current text of:

   It is RECOMMENDED that user agents
   periodically automatically check for the availability of the "sos"
   identifier and alert the user if the check fails.  The period of such
   automated checks SHOULDNOT be less than once per day and MUST be
   randomly placed over the testing interval.

Some random not well thought out ideas off the top of my head...

I would prefer MUST to RECOMMENDED since a recent informal poll at sipit
made it quiet clear that most implementers view SHOULD level requirements as
something to consider in the next lifetime. I'm not arguing that this poll
means we should not use SHOULD in general, but I think due to the critical
nature of this specification, we should be a little stronger than normal. I
don't want vendors saying they support this RFC fully when they can't detect
that a safety critical system is broken.

I would prefer specific test times - something like the test should be
performed at a random time between 12 and 36 hours. Phones must prominently
display the information that 911 is broken using a display or by changing
the dial tone on all calls. Would be nice to consider the impact of every
phone doing this. Phones SHOULD not do a test immediately after booting.
Phones MAY provide a UI to allow a user to manual imitate a test - this is
very useful for someone that is trying to check a 911 system but does not
want to wait 36 hours.

Cullen





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From: Cullen Jennings <fluffy@cisco.com>
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One major concern, Section 9.1.3 and 9.2.3 - PAI won't work for this as
described in draft-jennings-sipping-connected-00. I highly recommend we talk
about this on a phone call or in Korea and get it sorted out. Suggesting a
solution here will just make solving this problem more complex. Even if you
think that it does work, it is definitely a new semantic meaning of PAI that
is not covered in any the current documents. This would require the qsig2sip
document to be dealt with in the sip WG instead of sipping it it is going to
extend the meaning of PAI. It is recognized that SIP is lacking in this
area, there have been multiple drafts proposing solutions (including the
work in draft-elwell-sip-state-update). I think it would be better to get
one of theses solutions moving and just use it here.

Few minor things:

If the GW signs a sipfrag, what should be in the SubjectAltName of the
certificate?

Section 9.1 (and other places) The terminology "the gw trusts the next hop"
is wrong and should be something more like "the GW is in the same trust
domain (as defined in RFC 3325) as the next hop"

Section 9.2.2 If the From is Anonymous, and there is no privacy value, then
it should be treated as if there was a privacy value of id.

NIT:

I find reference style of "as described in [10]" much harder to read than "
as described in RFC 3261 [10]"

Cullen


On 1/20/04 7:05 AM, "Internet-Drafts@ietf.org" <Internet-Drafts@ietf.org>
wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Session Initiation Proposal Investigation
> Working Group of the IETF.
> 
> Title        : Interworking between SIP and QSIG
> Author(s)    : J. Elwell
> Filename    : draft-ietf-sipping-qsig2sip-04.txt
> Pages        : 61
> Date        : 2004-1-19
> 
> This document specifies interworking between the Session Initiation
>  Protocol (SIP) and QSIG within corporate telecommunication networks
>  (also known as enterprise networks). SIP is an Internet application-
>  layer control (signalling) protocol for creating, modifying, and
>  terminating sessions with one or more participants. These sessions
>  include, in particular, telephone calls. QSIG is a signalling
>  protocol for creating, modifying and terminating circuit-switched
>  calls, in particular telephone calls, within Private Integrated
>  Services Networks (PISNs). QSIG is specified in a number of ECMA
>  Standards and published also as ISO/IEC standards.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-04.txt
> 
> To remove yourself from the IETF Announcement list, send a message to
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> 
> A list of Internet-Drafts directories can be found in
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> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
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> 
> NOTE:    The mail server at ietf.org can return the document in
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> 


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Date: Tue, 17 Feb 2004 08:00:43 -0800
From: Cullen Jennings <fluffy@cisco.com>
To: sipping <sipping@ietf.org>
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SDP is a scary place so using SDP to separate the SDP bodies for A and B is
scary. Consider the following use case:

I have a transcoder that takes a audio from a UA connected to a microphone
and another camera UA that sends video. I am going to feed both of these to
a transcoder and produce a single mpeg output stream that contains both
audio video. Here I need so specify that the input SDP has two IP addresses
and the output has one. This would be even more complicated if one UA did
the offer/ans in INV/200 while the other used the 200/ACK.

Also related to all this, would be nice to be confident that pre conditions
stuff worked while using all this.

Anyways, I like the overall solution to all this stuff. I wish there was a
better solution to separating the A+B, TA+TB information to the transcoder
than adding more semantics to SDP.

Cullen


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> I would prefer MUST to RECOMMENDED since a recent informal poll at sipit
> made it quiet clear that most implementers view SHOULD level requirements as
> something to consider in the next lifetime. I'm not arguing that this poll
> means we should not use SHOULD in general, but I think due to the critical
> nature of this specification, we should be a little stronger than normal. I
> don't want vendors saying they support this RFC fully when they can't detect
> that a safety critical system is broken.

Gladly obliged.

> 
> I would prefer specific test times - something like the test should be
> performed at a random time between 12 and 36 hours. Phones must prominently

Currently, there are about 200 million real 911 calls a year. Testing 
once per day would essentially impose this load once per day. While I 
don't think building proxies that can handle this is a bad idea (among 
other advantages, it provides ample margin of safety and load-resistance 
since real calls are then a tiny fraction of the test traffic), it does 
need to be considered.

 From local usage, I know that fire alarm bells at Columbia are tested 
weekly, so that may be a more reasonable schedule.

> display the information that 911 is broken using a display or by changing
> the dial tone on all calls. Would be nice to consider the impact of every
> phone doing this. Phones SHOULD not do a test immediately after booting.
> Phones MAY provide a UI to allow a user to manual imitate a test - this is
> very useful for someone that is trying to check a 911 system but does not
> want to wait 36 hours.

I like the gist of this and have added appropriate text.

> 
> Cullen
> 
> 
> 

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In a message dated 2/16/2004 10:42:05 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> I agree with Jonathan that end system intelligence is the answer here. 
> After all, end systems have an interest in making emergency services work.
> 
> <snip>
> The problem with the approach of having the callee identify itself as a 
> emergency call center and thereby modify caller behavior is that lends 
> itself to pranks. Call my number and you can't hang up the phone until I 
> let you. Telemarketers will love that feature :-)
> 
> (If the calling UA knows it's an emergency call, there is no need for 
> the feature tag indication to control services.)
> 


I agree, if one is only considering the requirement for calls to an emergency 
center. It would be better if the operation (holding of the call until the 
callee says to release it) occurs without instructions coming back from the 
callee. In the case of the call to an energency center, I believe the desire is 
that this same operation will occur if someone calls 911 and hangs up before the 
call even gets to the emergency center. In today's network, the originating 
side equipment (central office) can know to do this when someone dials an 
emergency number. For the SIP case, if the originating device can do this 
(recognize the emergency number) or an outbound proxy does it, fine.

But, looking at another part of the issue: When the emergency center gets a 
call, it should be able to assume that the operation will comply with this 
requirement and it has control of releasing the call. Of course, the problem with 
SIP (as with any evolving protocol) is that older devices would not support 
this capability, so I suppose we need to add a function so the originating 
device can tell the emergency center that it can do with (called party control of 
release). Perhaps the existance of the SOS URI could imply this.

Another part of the issue regards the "ringback" function, which allows the 
emergency center to re-ring the phone of someone who has gone "on-hook" (or 
whatever similar function is used on the SIP phone to indicate end-of-call). One 
of the earlier e-mails suggested that, if the phone recognized 911, it could 
immediately re-ring the phone if the person "hangs-up". I don't believe this 
meets the requirement. The re-ring needs to be under the control of the 
emergency center, so they can decide whether a re-ring is necessary or not.

Also, someone needs to look into the possible requirement to allow the 
emergency center to alert the user even while they are "off-hook" while the 
connection is being held. The American National Standard (for ISDN) included this 
function.

Also, note that the subject of these e-mails includes mention of "operator". 
The function of hold/ringback may not just be limited to calls to emergency.

There is another use of the holding function in today's network: tracing of 
malicious calls. In this case, the request to hold the call must come from the 
called end (but not the called party). I guess one of the things about SIP and 
the environment it is used in is that it should not be neccessary to hold a 
"connection" in order to provide malicious call trace. However, we need to 
consider what happens for various packet/Circuit switched interworking cases.

What this points out is the need to agree on the functional requirements for 
Support for Hold and Ringback before doing detailed protocol work.

Mike Pierce
Artel



--part1_122.2b7916de.2d639bd1_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/16/2004 10:42:05 PM Eastern Standard Time, hgs@cs.columbia.edu write=
s:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I agree with Jonathan that=20=
end system intelligence is the answer here.=20
<BR>After all, end systems have an interest in making emergency services wor=
k.
<BR>
<BR>&lt;snip&gt;
<BR>The problem with the approach of having the callee identify itself as a=20
<BR>emergency call center and thereby modify caller behavior is that lends=20
<BR>itself to pranks. Call my number and you can't hang up the phone until I=
=20
<BR>let you. Telemarketers will love that feature :-)
<BR>
<BR>(If the calling UA knows it's an emergency call, there is no need for=20
<BR>the feature tag indication to control services.)
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>I agree, if one is only considering the requirement for calls to an emer=
gency center. It would be better if the operation (holding of the call until=
 the callee says to release it) occurs without instructions coming back from=
 the callee. In the case of the call to an energency center, I believe the d=
esire is that this same operation will occur if someone calls 911 and hangs=20=
up before the call even gets to the emergency center. In today's network, th=
e originating side equipment (central office) can know to do this when someo=
ne dials an emergency number. For the SIP case, if the originating device ca=
n do this (recognize the emergency number) or an outbound proxy does it, fin=
e.
<BR>
<BR>But, looking at another part of the issue: When the emergency center get=
s a call, it should be able to assume that the operation will comply with th=
is requirement and it has control of releasing the call. Of course, the prob=
lem with SIP (as with any evolving protocol) is that older devices would not=
 support this capability, so I suppose we need to add a function so the orig=
inating device can tell the emergency center that it can do with (called par=
ty control of release). Perhaps the existance of the SOS URI could imply thi=
s.
<BR>
<BR>Another part of the issue regards the "ringback" function, which allows=20=
the emergency center to re-ring the phone of someone who has gone "on-hook"=20=
(or whatever similar function is used on the SIP phone to indicate end-of-ca=
ll). One of the earlier e-mails suggested that, if the phone recognized 911,=
 it could immediately re-ring the phone if the person "hangs-up". I don't be=
lieve this meets the requirement. The re-ring needs to be under the control=20=
of the emergency center, so they can decide whether a re-ring is necessary o=
r not.
<BR>
<BR>Also, someone needs to look into the possible requirement to allow the e=
mergency center to alert the user even while they are "off-hook" while the c=
onnection is being held. The American National Standard (for ISDN) included=20=
this function.
<BR>
<BR>Also, note that the subject of these e-mails includes mention of "operat=
or". The function of hold/ringback may not just be limited to calls to emerg=
ency.
<BR>
<BR>There is another use of the holding function in today's network: tracing=
 of malicious calls. In this case, the request to hold the call must come fr=
om the called end (but not the called party). I guess one of the things abou=
t SIP and the environment it is used in is that it should not be neccessary=20=
to hold a "connection" in order to provide malicious call trace. However, we=
 need to consider what happens for various packet/Circuit switched interwork=
ing cases.
<BR>
<BR>What this points out is the need to agree on the functional requirements=
 for Support for Hold and Ringback before doing detailed protocol work.
<BR>
<BR>Mike Pierce
<BR>Artel
<BR>
<BR></FONT></HTML>

--part1_122.2b7916de.2d639bd1_boundary--

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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CC: Cullen Jennings <fluffy@cisco.com>, sipping@ietf.org
Subject: Re: [Sipping] Re: sos - dead or alive
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The OPTIONS mechanism does exactly that - it's mechanical only.

Janet P Gunn wrote:

> Bear in mind that the "most scarce resource" in the emergency response
> system is usually the human operators.
> 
> So any kind of regular, scheduled test should be designed so it does  NOT
> tie up the operators.


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Henning Schulzrinne
	 <hgs@cs.columbia.edu>
Cc: Xuewen Li <mli@nortelnetworks.com>, sipping@ietf.org
Subject: RE: [Sipping] SIP support for hold and ringback for operator / 91
	1 services
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> In principal, I would think that someone (NENA??) could hand 
> out certs 
> that do, in fact, certify that a user is an operator in an emergency 
> service center. If a call I make completes to an user that has a cert 
> declaring them as an emergency operation, my UA wouldn't allow me to 
> terminate the call.
> 
I looked into this when the privacy folks wanted to make sure that
emergency calls in fact reached an emergency service provider.

I was told this would take a treaty.

Basically, the nations won't recognize any specific agency as the
root CA.  I keep hoping we can get ITU to do it as part of the
delegation of sos.arpa.  I really do want a cryptographic way to
authenticate stuff in my sos.arpa DNS tree, and I really do want to
authenticate a PSAP.

It guaranteed to be a problem if any organization that looks vaguely
US in origin was doing it.  NENA could be the CA for the U.S. of course,
but to work really well, you want its key to be signed by the international
CA.

Brian

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Thanks for the comments.  I'll include most of them as you suggest.
The proposal copies the "datum" code from the DHCP draft.  I'd welcome
any suggestions for what to advise clients to do when two entries use
different datums and they are trying to determine inclusion.  I wondered
if just saying "its WGS84, convert on entry" isn't the right way, but
so many other folks have flamed out trying to fix the datum that I
gave up on it.

The Floor problem is solved I think.  The DNS entry would have a floor as
one of the levels, which would presumably have the correct geos with
altitude in meters.  There is also the floor count format of the data,
also copied from the DHCP draft.  Agree I need text on how to describe the
polygons for floors.

Brian



> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, February 17, 2004 10:57 AM
> To: Brian Rosen; 'sipping@ietf.org'
> Subject: comments on rosen-dns-sos
> 
> 
> 
> One small nit on polygons. Need to specify that the vertexes are in
> clockwise order. Failing to do this caused endless arguments 
> in air traffic
> control specifications of airspaces. Probably needs a way to 
> deal with what
> geographic coordinate grid is being used or fix it. Should 
> warn implementers
> that polygon will include the poles and the vertex may even 
> be a pole. Polar
> flights have caused more than one geographic information 
> system to crash.
> Many systems translate the 3 tupple coordinates into 4 tupple 
> coordinates
> for all calculations to avoid the singularities that a 
> spherical coordinate
> system introduces. Probably need a header format that allows 
> an arbitrary
> number of vertexes -  the top level polygons are going to be 
> big. I don't
> like the implied additive subtractive form of the poly. I 
> would prefer poly
> explicitly identify as additive or subtractive with the 
> region covered being
> formed by taking the union of all the additive then 
> subtracting each of the
> subtractive polygons. Points on a vertex or edge would be 
> considered in the
> region for additive polygons and out for subtractive 
> polygons. None of the
> polygons forming a region are allowed to share an edge. The 
> altitudes need
> to be altitude ranges to specify a volume that can be 
> clipped. Seems like
> you need this to figure out which floor in a building I am 
> on. Recommend
> that min alt is include in the volume but max alt is outside 
> the volume. The
> specification of points on the surface of the volume being in 
> or out is
> important or users can, excuse the pun, fall thought the cracks.
> 
> I was not sold on DNS to start with but your arguments about 
> the pleasures
> of forming a new global root convinced me DNS is right if it 
> will work. More
> text explaining it would work would make me feel warm and fluffy.
> 
> Cullen
> 
> 
> 
> 
> 
> 

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
        Henning Schulzrinne
	 <hgs@cs.columbia.edu>, sipping@ietf.org
Subject: RE: [Sipping] sos - dead or alive
Date: Tue, 17 Feb 2004 15:11:39 -0500
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Probably need a separate URI for this; don't want to have even a 
possibility for overwhelming the real proxy server.  It would be easy
for the test one to heartbeat the real one to keep them in sync.

We could define a more precise algorithm based on lsbs of IP address to
randomize test times.

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, February 17, 2004 10:58 AM
> To: Henning Schulzrinne; sipping@ietf.org
> Subject: [Sipping] sos - dead or alive
> 
> 
> 
> I think that a phone informing a user that SOS was broken is 
> very important
> for this to work when it is needed. I would like to see a bit 
> more text
> around the current text of:
> 
>    It is RECOMMENDED that user agents
>    periodically automatically check for the availability of the "sos"
>    identifier and alert the user if the check fails.  The 
> period of such
>    automated checks SHOULDNOT be less than once per day and MUST be
>    randomly placed over the testing interval.
> 
> Some random not well thought out ideas off the top of my head...
> 
> I would prefer MUST to RECOMMENDED since a recent informal 
> poll at sipit
> made it quiet clear that most implementers view SHOULD level 
> requirements as
> something to consider in the next lifetime. I'm not arguing 
> that this poll
> means we should not use SHOULD in general, but I think due to 
> the critical
> nature of this specification, we should be a little stronger 
> than normal. I
> don't want vendors saying they support this RFC fully when 
> they can't detect
> that a safety critical system is broken.
> 
> I would prefer specific test times - something like the test should be
> performed at a random time between 12 and 36 hours. Phones 
> must prominently
> display the information that 911 is broken using a display or 
> by changing
> the dial tone on all calls. Would be nice to consider the 
> impact of every
> phone doing this. Phones SHOULD not do a test immediately 
> after booting.
> Phones MAY provide a UI to allow a user to manual imitate a 
> test - this is
> very useful for someone that is trying to check a 911 system 
> but does not
> want to wait 36 hours.
> 
> Cullen
> 
> 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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Sections 6.2 and 6.3 have been changed, adding "Supported: gruu". But if 
these examples are gruus, they at presumably aren't ones that have been 
dynamically constructed by the registrar, which would presumably use 
some other form of contact. This seems to raise the expectation that a 
UA just knows if its natural contact address is globally routable and 
uses it as a gruu. Is that what we want to show in an example?

	Paul

Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.
> 
> 	Title		: Session Initiation Protocol Call Control - Transfer
> 	Author(s)	: R. Sparks, A. Johnston
> 	Filename	: draft-ietf-sipping-cc-transfer-02.txt
> 	Pages		: 37
> 	Date		: 2004-2-17
> 	
> This document describes providing Call Transfer capabilities in the
> Session Initiation Protocol (SIP).  This work is part of the SIP
> Multiparty Call Control Framework.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-transfer-02.txt
> 
> To remove yourself from the IETF Announcement list, send a message to 
> ietf-announce-request with the word unsubscribe in the body of the message.
> 
> Internet-Drafts are also available by anonymous FTP. Login with the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-sipping-cc-transfer-02.txt".
> 
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html 
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
> 
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
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> 	
> NOTE:	The mail server at ietf.org can return the document in
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> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.


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Rosen, Brian wrote:

> Probably need a separate URI for this; don't want to have even a 
> possibility for overwhelming the real proxy server.  It would be easy
> for the test one to heartbeat the real one to keep them in sync.

While you can obviously do method-based routing, I would worry about 
building any such knowledge into the UA. After all, you want to test as 
much of the whole chain as you can.

To improve scaling, an outbound proxy could answer all such OPTIONS 
requests to keep testing load local, depending on our perception of 
where the likely failures are. (The emergency-call-aware outbound proxy 
could then in turn exercise the path from itself to the ECCs of 
interest, but you'd then have one per company or provider, instead of 
10,000 tests per campus.)

> 
> We could define a more precise algorithm based on lsbs of IP address to
> randomize test times.

I don't think that helps much beyond simple randomization. We have 
reasonably good experience with randomization in RTCP after all.

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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] sos - dead or alive
Date: Tue, 17 Feb 2004 15:37:35 -0500
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If the SIP URI for a PSAP was sip:anyone@emergency.chicago.il.us, and the
test
URI was sip:test@test.chicago.il.us, I don't think we have any routing
problems.  The router for chicago.il.us would get exercised for test code,
but not the proxy.

I like the scaling idea.

RTCP has never been tested with a billion phones randomly trying anything.
I think we need to be careful. Very careful.

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Tuesday, February 17, 2004 3:28 PM
> To: Rosen, Brian
> Cc: 'Cullen Jennings'; sipping@ietf.org
> Subject: Re: [Sipping] sos - dead or alive
> 
> 
> 
> 
> Rosen, Brian wrote:
> 
> > Probably need a separate URI for this; don't want to have even a 
> > possibility for overwhelming the real proxy server.  It 
> would be easy
> > for the test one to heartbeat the real one to keep them in sync.
> 
> While you can obviously do method-based routing, I would worry about 
> building any such knowledge into the UA. After all, you want 
> to test as 
> much of the whole chain as you can.
> 
> To improve scaling, an outbound proxy could answer all such OPTIONS 
> requests to keep testing load local, depending on our perception of 
> where the likely failures are. (The emergency-call-aware 
> outbound proxy 
> could then in turn exercise the path from itself to the ECCs of 
> interest, but you'd then have one per company or provider, instead of 
> 10,000 tests per campus.)
> 
> > 
> > We could define a more precise algorithm based on lsbs of 
> IP address to
> > randomize test times.
> 
> I don't think that helps much beyond simple randomization. We have 
> reasonably good experience with randomization in RTCP after all.
> 

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



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As with the transfer examples, the changes here give the impression that 
things will be wonderful if you add "Supported: gruu" and continue to 
use your old contact address.

If examples are to be given that depend upon gruus, then I think we need 
to show that the gruu is something special - not the same as the UA's 
normal contact address.

	Paul

Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.
> 
> 	Title		: Session Initiation Protocol Service Examples
> 	Author(s)	: A. Johnston, R. Sparks
> 	Filename	: draft-ietf-sipping-service-examples-06.txt
> 	Pages		: 169
> 	Date		: 2004-2-17
> 	
> This document gives examples of Session Initiation Protocol (SIP)
> services.  This covers most features offered in so-called IP Centrex
> offerings from local exchange carriers and PBX (Private Branch
> Exchange) features.  Most of the services shown in this document are
> implemented in the SIP User Agents, although some require the
> assistance of a SIP Proxy.  Some require some extensions to SIP
> including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces
> and Join headers. These features are not intended to be an exhaustive
> set, but rather show implementations of common features likely to be
> implemented on SIP IP telephones in a business environment.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-06.txt
> 
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> 
> A list of Internet-Drafts directories can be found in
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> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> 
> 
> Internet-Drafts can also be obtained by e-mail.
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Unless this is identified in the mapping algorithm (location -> ECC), I 
don't see how a device can find the test destination.

Rosen, Brian wrote:

> If the SIP URI for a PSAP was sip:anyone@emergency.chicago.il.us, and the
> test
> URI was sip:test@test.chicago.il.us, I don't think we have any routing
> problems.  The router for chicago.il.us would get exercised for test code,
> but not the proxy.
> 
> I like the scaling idea.
> 
> RTCP has never been tested with a billion phones randomly trying anything.
> I think we need to be careful. Very careful.


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



From exim@www1.ietf.org  Tue Feb 17 15:55:03 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] sos - dead or alive
Date: Tue, 17 Feb 2004 15:49:59 -0500
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You put another NAPTR in the ECC entry in the DNS :)

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Tuesday, February 17, 2004 3:45 PM
> To: Rosen, Brian
> Cc: 'Cullen Jennings'; sipping@ietf.org
> Subject: Re: [Sipping] sos - dead or alive
> 
> 
> Unless this is identified in the mapping algorithm (location 
> -> ECC), I 
> don't see how a device can find the test destination.
> 
> Rosen, Brian wrote:
> 
> > If the SIP URI for a PSAP was 
> sip:anyone@emergency.chicago.il.us, and the
> > test
> > URI was sip:test@test.chicago.il.us, I don't think we have 
> any routing
> > problems.  The router for chicago.il.us would get exercised 
> for test code,
> > but not the proxy.
> > 
> > I like the scaling idea.
> > 
> > RTCP has never been tested with a billion phones randomly 
> trying anything.
> > I think we need to be careful. Very careful.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Session Initiation Protocol Call Control - Conferencing for User Agents
	Author(s)	: A. Johnston, O. Levin
	Filename	: draft-ietf-sipping-cc-conferencing-03.txt
	Pages		: 34
	Date		: 2004-2-17
	
This document defines conferencing call control features for the
Session Initiation Protocol (SIP).  This document builds on the
Conferencing Requirements and Framework documents to define how a
tightly coupled SIP conference works.  The approach is explored from
different user agent (UA) types perspective: conference-unaware,
conference-aware and focus UAs.  The use of URIs in conferencing,
OPTIONS for capabilities discovery, and call control using REFER are
covered in detail with example call flow diagrams.

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_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: Re: [Sipping] SIP support for hold and ringback for operator / 91 1 services
To: Brian.Rosen@marconi.com, jdrosen@dynamicsoft.com, hgs@cs.columbia.edu
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Content-Type: text/plain; charset="US-ASCII"
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In a message dated 2/17/2004 3:02:49 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:


> Basically, the nations won't recognize any specific agency as the
> root CA.  I keep hoping we can get ITU to do it as part of the
> delegation of sos.arpa.  I really do want a cryptographic way to
> authenticate stuff in my sos.arpa DNS tree, and I really do want to
> authenticate a PSAP.
> 
> It guaranteed to be a problem if any organization that looks vaguely
> US in origin was doing it.  NENA could be the CA for the U.S. of course,
> but to work really well, you want its key to be signed by the international
> CA.
> 


Well, the answer to all this is quite simple - just recognize that emergency 
calling, connection holding, ringback, etc. are national matters and do not 
have to cross boundaries (except maybe where adjacent countries set up specific 
agreements, including CAs if necessary). Anything more than that seems to be 
beyond our ability to deal with.

Mike Pierce


--part1_1db.1a3a9a99.2d63ed45_boundary
Content-Type: text/html; charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/17/2004 3:02:49 PM Eastern Standard Time, Brian.Rosen@marconi.com wr=
ites:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Basically, the nations won'=
t recognize any specific agency as the
<BR>root CA. &nbsp;I keep hoping we can get ITU to do it as part of the
<BR>delegation of sos.arpa. &nbsp;I really do want a cryptographic way to
<BR>authenticate stuff in my sos.arpa DNS tree, and I really do want to
<BR>authenticate a PSAP.
<BR>
<BR>It guaranteed to be a problem if any organization that looks vaguely
<BR>US in origin was doing it. &nbsp;NENA could be the CA for the U.S. of co=
urse,
<BR>but to work really well, you want its key to be signed by the internatio=
nal
<BR>CA.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>Well, the answer to all this is quite simple - just recognize that emerg=
ency calling, connection holding, ringback, etc. are national matters and do=
 not have to cross boundaries (except maybe where adjacent countries set up=20=
specific agreements, including CAs if necessary). Anything more than that se=
ems to be beyond our ability to deal with.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_1db.1a3a9a99.2d63ed45_boundary--

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



From exim@www1.ietf.org  Tue Feb 17 17:47:51 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, jdrosen@dynamicsoft.com,
        hgs@cs.columbia.edu
Cc: mli@nortelnetworks.com, sipping@ietf.org
Subject: RE: [Sipping] SIP support for hold and ringback for operator / 91
	 1 services
Date: Tue, 17 Feb 2004 17:43:18 -0500
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We don't do national variants in the IETF.  It does matter.
If you can spoof, you can create your own imitation sos.arpa tree,
with your own us.sos.arpa, with your own public key.  No one will be
the wiser.  

We can broadly disseminate the us.sos.arpa public key within the U.S.,
but in general, it's not possible to securely disseminate every nation's
sos.arpa public key.  Having a root CA to certify them would make a
basically insecure system a secure system.

Brian


-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Tuesday, February 17, 2004 5:19 PM
To: Brian.Rosen@marconi.com; jdrosen@dynamicsoft.com; hgs@cs.columbia.edu
Cc: mli@nortelnetworks.com; sipping@ietf.org
Subject: Re: [Sipping] SIP support for hold and ringback for operator / 91 1
services


In a message dated 2/17/2004 3:02:49 PM Eastern Standard Time,
Brian.Rosen@marconi.com writes: 



Basically, the nations won't recognize any specific agency as the 
root CA.  I keep hoping we can get ITU to do it as part of the 
delegation of sos.arpa.  I really do want a cryptographic way to 
authenticate stuff in my sos.arpa DNS tree, and I really do want to 
authenticate a PSAP. 

It guaranteed to be a problem if any organization that looks vaguely 
US in origin was doing it.  NENA could be the CA for the U.S. of course, 
but to work really well, you want its key to be signed by the international 
CA. 




Well, the answer to all this is quite simple - just recognize that emergency
calling, connection holding, ringback, etc. are national matters and do not
have to cross boundaries (except maybe where adjacent countries set up
specific agreements, including CAs if necessary). Anything more than that
seems to be beyond our ability to deal with. 

Mike Pierce 

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



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Date: Tue, 17 Feb 2004 16:54:25 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] I-D  ACTION:draft-ietf-sipping-service-examples-06.txt
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At 03:41 PM 2/17/2004 -0500, Paul Kyzivat wrote:
>As with the transfer examples, the changes here give the impression that 
>things will be wonderful if you add "Supported: gruu" and continue to use 
>your old contact address.
>
>If examples are to be given that depend upon gruus, then I think we need 
>to show that the gruu is something special - not the same as the UA's 
>normal contact address.

Paul,

What do you have in mind?  A GRUU is identical to any other URI - it just 
meets the requirements in the draft.  A GRUU doesn't need to look different 
from any other URI - in fact, no one can tell by inspection if a URI is a 
GRUU.

Do you just want the example GRUUs to *look* different??

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

>         Paul
>
>Internet-Drafts@ietf.org wrote:
>>A New Internet-Draft is available from the on-line Internet-Drafts 
>>directories.
>>This draft is a work item of the Session Initiation Proposal 
>>Investigation Working Group of the IETF.
>>         Title           : Session Initiation Protocol Service Examples
>>         Author(s)       : A. Johnston, R. Sparks
>>         Filename        : draft-ietf-sipping-service-examples-06.txt
>>         Pages           : 169
>>         Date            : 2004-2-17
>>
>>This document gives examples of Session Initiation Protocol (SIP)
>>services.  This covers most features offered in so-called IP Centrex
>>offerings from local exchange carriers and PBX (Private Branch
>>Exchange) features.  Most of the services shown in this document are
>>implemented in the SIP User Agents, although some require the
>>assistance of a SIP Proxy.  Some require some extensions to SIP
>>including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces
>>and Join headers. These features are not intended to be an exhaustive
>>set, but rather show implementations of common features likely to be
>>implemented on SIP IP telephones in a business environment.
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-06.txt
>>To remove yourself from the IETF Announcement list, send a message to 
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>>Internet-Drafts are also available by anonymous FTP. Login with the username
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>>A list of Internet-Drafts directories can be found in
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>>
>>Internet-Drafts can also be obtained by e-mail.
>>Send a message to:
>>         mailserv@ietf.org.
>>In the body type:
>>         "FILE /internet-drafts/draft-ietf-sipping-service-examples-06.txt".
>>
>>NOTE:   The mail server at ietf.org can return the document in
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>>         command.  To decode the response(s), you will need "munpack" or
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>>
>>Below is the data which will enable a MIME compliant mail reader
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>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sipping] SIP support for hold and ringback for operator / 91 1 services
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In a message dated 2/17/2004 5:43:57 PM Eastern Standard Time, 
Brian.Rosen@marconi.com writes:


> We don't do national variants in the IETF.


I didn't suggest doing a national variant. I said "recognize that emergency 
calling, connection holding, ringback, etc. are national matters". Then, all 
that the IETF needs to do is make sure it has generic protocols and security 
procedures in place so each country can define ways to use them so that people in 
that country can call emergency centers in that country according to the 
rules and laws applicable in that country.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/17/2004 5:43:57 PM Eastern Standard Time, Brian.Rosen@marconi.com wr=
ites:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">We don't do national varian=
ts in the IETF.</FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BAC=
KGROUND-COLOR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>I didn't suggest doing a national variant. I said "recognize that emerge=
ncy calling, connection holding, ringback, etc. are national matters". Then,=
 all that the IETF needs to do is make sure it has generic protocols and sec=
urity procedures in place so each country can define ways to use them so tha=
t people in that country can call emergency centers in that country accordin=
g to the rules and laws applicable in that country.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_103.3f6fb7fa.2d63fc0f_boundary--

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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CC: jdrosen@dynamicsoft.com, mli@nortelnetworks.com, sipping@ietf.org
Subject: Re: [Sipping] SIP support for hold and ringback for operator / 911
 services
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> someone calls 911 and hangs up before the call even gets to the 
> emergency center. In today's network, the originating side equipment 
> (central office) can know to do this when someone dials an emergency 
> number. For the SIP case, if the originating device can do this 
> (recognize the emergency number) or an outbound proxy does it, fine.

Repeating my earlier comments: even in today's modern networks (E911, 
ISUP, etc.), the callee-control is neither mandated nor universally 
supported. At least, I'm still waiting for a spec that says so. That 
said, the whole process of recognizing digits and sending 'sos' is also 
meant to trigger whatever caller-side behavior is appropriate.


> Another part of the issue regards the "ringback" function, which allows 
> the emergency center to re-ring the phone of someone who has gone 
> "on-hook" (or whatever similar function is used on the SIP phone to 
> indicate end-of-call). One of the earlier e-mails suggested that, if the 
> phone recognized 911, it could immediately re-ring the phone if the 
> person "hangs-up". I don't believe this meets the requirement. The 
> re-ring needs to be under the control of the emergency center, so they 
> can decide whether a re-ring is necessary or not.

Callback is described in emergency-arc.

> 
> Also, someone needs to look into the possible requirement to allow the 
> emergency center to alert the user even while they are "off-hook" while 
> the connection is being held. The American National Standard (for ISDN) 
> included this function.

Can you cite chapter and verse?



> There is another use of the holding function in today's network: tracing 
> of malicious calls. In this case, the request to hold the call must come 
> from the called end (but not the called party). I guess one of the 
> things about SIP and the environment it is used in is that it should not 
> be neccessary to hold a "connection" in order to provide malicious call 
> trace. However, we need to consider what happens for various 
> packet/Circuit switched interworking cases.

I'm sure a malicious caller will faithfully follow all SIP standards :-)

> 
> What this points out is the need to agree on the functional requirements 
> for Support for Hold and Ringback before doing detailed protocol work.
> 
> Mike Pierce
> Artel
> 

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Alan Johnston wrote:
> At 03:41 PM 2/17/2004 -0500, Paul Kyzivat wrote:
> 
>> As with the transfer examples, the changes here give the impression 
>> that things will be wonderful if you add "Supported: gruu" and 
>> continue to use your old contact address.
>>
>> If examples are to be given that depend upon gruus, then I think we 
>> need to show that the gruu is something special - not the same as the 
>> UA's normal contact address.
> 
> 
> Paul,
> 
> What do you have in mind?  A GRUU is identical to any other URI - it 
> just meets the requirements in the draft.  A GRUU doesn't need to look 
> different from any other URI - in fact, no one can tell by inspection if 
> a URI is a GRUU.
> 
> Do you just want the example GRUUs to *look* different??

There are two possibilities here:

- include in the call flows the REGISTER message that gets the gruu.
   Then it is entirely clear.

- just make the GRUUs look different, in a suggestive way.
   It is especially deceptive if one example that doesn't use
   gruu uses the same contact address as another example that does
   use gruu. And using what is obviously intended to be the dns
   address of the endpoint itself, or worse the same address as
   is used for media addresses, is especially misleading.
   Something on the order of GRUUnnn@proxy.com would go a long way.

Including a REGISTER in every example that uses gruus would probably be 
overkill. But it might be worthwhile to at least include examples that 
do obtain gruus appropriate for the examples that use them.

	Paul


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Date: Tue, 17 Feb 2004 20:26:15 EST
Subject: Re: [Sipping] SIP support for hold and ringback for operator / 911 services
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In a message dated 2/17/2004 6:28:23 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> > 
> > Also, someone needs to look into the possible requirement to allow the 
> > emergency center to alert the user even while they are "off-hook" while 
> > the connection is being held. The American National Standard (for ISDN) 
> > included this function.
> 
> Can you cite chapter and verse?
> 

This capability was included in American National Standard T1.628a, Paragraph 
6.2.8.1. I have to admit that I don't know of a document that actually 
describes the requirement to do this. It's unfortunate that many standards bodies, 
T1 included, often write protocol for things for which there is no hard 
requirement. I don't know about this case. I only mentioned it hoping that someone 
would know of a specific requirement.

Mike


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/17/2004 6:28:23 PM Eastern Standard Time, hgs@cs.columbia.edu writes=
:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">&gt;=20
<BR>&gt; Also, someone needs to look into the possible requirement to allow=20=
the=20
<BR>&gt; emergency center to alert the user even while they are "off-hook" w=
hile=20
<BR>&gt; the connection is being held. The American National Standard (for I=
SDN)=20
<BR>&gt; included this function.
<BR>
<BR>Can you cite chapter and verse?
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>This capability was included in American National Standard T1.628a, Para=
graph 6.2.8.1. I have to admit that I don't know of a document that actually=
 describes the requirement to do this. It's unfortunate that many standards=20=
bodies, T1 included, often write protocol for things for which there is no h=
ard requirement. I don't know about this case. I only mentioned it hoping th=
at someone would know of a specific requirement.
<BR>
<BR>Mike
<BR></FONT></HTML>

--part1_199.25f993cb.2d641937_boundary--

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In-Reply-To: <40293C6B.4060404@dynamicsoft.com>
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Cc: hisham.khartabil@nokia.com, sipping@ietf.org
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] Conference policy URI
Date: Wed, 18 Feb 2004 06:17:15 -0800
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Hi,

I believe the SIPPING working group has come down with a bad case of  
Ianaus Registritus.  This rare, but preventable disease is caused by  
trying to register every possible field or parameter mentioned in a set  
of standards track documents.   It is highly contagious to other  
related working groups.  Treatment consists of deleting IANA  
Registration sections until the symptoms subside.  The patient should  
begin feeling better immediately.

Seriously, I like the new Call-Info purpose, but I think the IANA  
Registry goes way too far overboard.  Remember that there is an  
administrative and political cost to setting up new IANA Registries.

thansk,
-rohan


On Feb 10, 2004, at 12:17 PM, Jonathan Rosenberg wrote:

> Thanks for doing this. Some comments.
>
> Firstly, on the registry itself. You need to give clearer guidance to  
> IANA on the registry. I would suggest including a table that shows how  
> it should look like. You need to tell IANA what information needs to  
> be provided in future RFCs that register additional values (value, RFC  
> number, registrant, etc.). Please take a look at rfc2434  
> (ftp://ftp.rfc-editor.org/in-notes/rfc2434.txt).
>
> Secondly, on the "policy-uri" value - you need to specify, I think,  
> what the URI type is, and what the content is at that URI (at least a  
> mandatory to implement one) in order to get interoperability. Unlike  
> the other stuff, which is just rendered for user information, this  
> resource is processed by an automata, and I think you need to say more  
> about it.
>
> Thanks,
> Jonathan R.
>
> hisham.khartabil@nokia.com wrote:
>
>> Thanks Jonathan. A draft has been submitted to the SIP WG.
>> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri- 
>> call-info-purpose-00.txt
>> Regards,
>> Hisham
>>> -----Original Message-----
>>> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>> Sent: 24.January.2004 04:44
>>> To: Khartabil Hisham (Nokia-TP/Helsinki)
>>> Cc: sipping@ietf.org
>>> Subject: Re: [Sipping] Conference policy URI
>>>
>>>
>>> I think this is a reasonable usage of Call-Info. However, it can  
>>> only be done if a "purpose" tag value is registered with IANA, as  
>>> indicated by section 20.9. However, it ocurred to me, when reading  
>>> this, that we never set up an IANA registry for call-info purpose  
>>> tokens! Oops! Thats easily rectifiable; when the draft defining the  
>>> 'policy-uri' (or whatever its called) purpose tag is written, it  
>>> should also create the registry and register the other tokens  
>>> defined in rfc3261.
>>>
>>> -Jonathan R.
>>>
>>> hisham.khartabil@nokia.com wrote:
>>>
>>>
>>>> Hi,
>>>>
>>>> There was consensus on the XCON mailing list that the conference
>>>> state event package needs to include in the notification the
>>>> conference policy URI.
>>>>
>>>> There was also consensus that some participants may not want to
>>>> subscribe to the conference package but would still like to learn
>>>> the conference policy URI. In this case, it was suggested that the
>>>> Call-info header carries such information. Perhaps the
>>>> cc-conferencing draft can include examples of that.
>>>>
>>>> Thanks, Hisham
>>>>
>>>> _______________________________________________ Sipping mailing
>>>> list  https://www1.ietf.org/mailman/listinfo/sipping This list is
>>>> for NEW development of the application of SIP Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>> sip@ietf.org for new developments of core SIP
>>>>
>>>
>>> -- 
>>> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>> Chief Technology Officer                    Parsippany, NJ 07054-2711
>>> dynamicsoft
>>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>>> http://www.dynamicsoft.com
>>>
>>>
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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



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Date: Wed, 18 Feb 2004 12:27:45 -0600
From: Alan Johnston <alan.johnston@mci.com>
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FYI - an update to the RTCP Summary Report Delivery to SIP Third Parties.

The document now makes a first attempt at defining the event package and 
has some example call flows and messages.

Comments/suggestions are most welcome.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

>Date: Tue, 17 Feb 2004 16:37:22 -0500
>From: Internet-Drafts@ietf.org
>Subject: I-D ACTION:draft-johnston-sipping-rtcp-summary-02.txt
>Sender: owner-ietf-announce@ietf.org
>To: IETF-Announce: ;
>Reply-to: Internet-Drafts@ietf.org
>
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>
>
>         Title           : RTCP Summary Report Delivery to SIP Third Parties
>         Author(s)       : A. Johnston
>         Filename        : draft-johnston-sipping-rtcp-summary-02.txt
>         Pages           : 17
>         Date            : 2004-2-17
>
>This document discusses the motivation and requirements for the
>delivery of RTCP extended reports and other summary reports to
>non-participants in the session.  Several solution mechanisms are
>also discussed and compared.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary-02.txt
>
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_______________________________________________
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From: "Veda Hegde" <vhegde@wave3software.com>
To: <sipping@ietf.org>
Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-conference-package-03.txt
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Hi,
I don't see anything that would indicate the state of individual media
streams offered by user. If a user is offering audio and video, focus might
choose to mix just the audio and not the video. In that case, is there a way
to indicate that just the video stream was muted by the focus?
Thanks,
Veda


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Wednesday, February 18, 2004 7:18 AM
To: IETF-Announce:
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Subject: [Sipping] I-D
ACTION:draft-ietf-sipping-conference-package-03.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Proposal Investigation
Working Group of the IETF.

	Title		: A Session Initiation Protocol (SIP) Event Package for Conference
State
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sipping-conference-package-03.txt
	Pages		: 27
	Date		: 2004-2-18

This document defines a conference event package for the Session
   Initiation Protocol (SIP) Events framework, along with a data format
   used in notifications for this package. The conference package allows
   users to subscribe to a conference URI. Notifications are sent about
   changes in the membership of this conference, the status of users'
   participation in the conference, and the sidebars in the conference.

A URL for this Internet-Draft is:
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Date: Wed, 18 Feb 2004 13:48:39 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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Wow! You made a lot of changes in this version.

I do have a number of comments about this version, below. I think I was 
happier with it before. This one seems to have run amuck.

	Paul

First, a nit: The introduction says:

    Finally, there are
    situations where the party issuing the REFER request does not need
    the NOTIFY associated with the REFER, perhaps because that agent is
    already subscribed to the appropriate package outside of the REFER
    request.

I agree there are situations where the referring party doesn't need the 
NOTIFY, but not because it is already subscribed for it. There is no way 
to independently subscribe to the refer event package.

----------

Section 7 says:

    The REFER-Issuer MAY specify the request for subscription to a
    specific package by including the MIME type in an Accept header.

Yuk! All kinds of problems with this:

- event packages are not defined by mime-types, they are defined by 
event package names. It is possible for several event packages to share 
a mime-type for their notifications, and for multiple mime-types to be 
permissible for a single event package.

- subscribing to an event package requires both the event package name 
and the event id. (In the case of the special refer event package, the 
event id is defined by the cseq of the REFER message itself.)

- If bundling REFER with an implicit subscription to the refer event 
package was a bad idea (as most seem to think) then bundling it with the 
establishment of arbitrary subscriptions is an even worse idea.

- Based on the examples, this isn't really a matter of subscribing to a 
different event package. The event package named in the NOTIFY is still 
"refer", but the content-type is changed. So the effect is of redefining 
the refer event package so that it may return content equivalent to that 
of any other event package. This is at best a kludge. It also probably 
violates the rules for defining of event packages. In this model it 
isn't at all clear what defines how the content of the notifies is 
obtained, or when it is generated, since that is generally defined as 
part of an unnamed event package.

If you really want to do this, then I think you must ammend the 
definition of the refer event package to include (presumably by 
reference) most of the definition of any of the other event packages 
that you might want to request this way. But I still think this is a bad 
idea. Even if it requires extra messages, I think subscriptions to 
anthing but the refer event package should be established first, before 
sending the refer.

---------

Later in that section, another nit:

    In the INVITE example, two things that are generally lacking from the
    message/sipfrag content are the dialog identifier (Call-ID plus local
    and remote tags) and the state of the dialog. Not coincidentally,
    this is the same information available in the application/
    dialog-info+xml MIME type [7] used for the dialog event package.

The dialog-id can be available using the sipfrag content if 
refer-recipient chooses to provide them. They are available if the 
fragment contains the to, from, and call-id from the invite response.

(I agree that certain dialog status may not be determined via the refer 
event package.)

----------

norefersub: While I think this is a good feature to support, I think it 
is a stretch to encode it as a required option. That mechanism is 
normally used for more general functionality that transcends a 
particular request. It would be good to have a way to specify this in a 
way that is more specific to refer. One alternative might be to add a 
norefersub parameter to the Refer-To header.

(Also note that returning a 481 in response to the first NOTIFY may not 
always be a cheap way to cancel the subscription without norefersub. If 
the refer was sent within a preexisting dialog, returning a 481 to the 
notify might be construed as absence of the entire dialog. If so, all 
the things sharing the dialog might also be terminated. So in that case 
it may be necessary to cancel by sending a subscribe with expires:0, 
requiring 5 messages rather than one.)

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

In general there seem to be far more option tags than are needed. I 
think it would be sufficient to define one new option tag indicating 
support for all the functionality in this draft. Requesting of specific 
features can be via other parameters or headers.

	Paul


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



From exim@www1.ietf.org  Wed Feb 18 15:41:35 2004
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From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] URGENT: query on another reg-event change
Date: Wed, 18 Feb 2004 12:36:51 -0800
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Jonathan,

The current rev of the dialog package takes a middle approach.  Rather 
than stick everything in a single string or apply semantics to every 
tag, the target tag (the equivalent of the contact tag in the 
registration package) can have sub-elements for contact parameters

<contact id="123" uri="sip:alice@example.com" state="active" etc...>
   <param pname="actor" pvalue="msg-taker"/>
</contact>

I think this would work just fine for the registration package.

thanks,
-rohan



On Feb 16, 2004, at 10:05 AM, Jonathan Rosenberg wrote:

> Folks,
>
> Authors 48 hours have started for reg-event. As you may recall, I had 
> found a fatal problem in the schema design, and there was agreement on 
> the list on a proposal to fix it. While making the changes in the 
> schema, another question came up.
>
> The question is regarding the "params" attribute of the "contact" 
> element. Currently, this parameter is meant to contain any of the 
> contact parameters not otherwise called out explicitly in other 
> attributes (for example, the q-value is contained in the "q" 
> attribute. So, if I had a contact like this:
>
> REGISTER
> To: sip:user@example.com
> Contact: sip:user@pc.example.com;q=1.0;newparam=foo;audio
>
> then the resulting XML would look like (using the new schema I 
> proposed on the list):
>
>  <?xml version="1.0"?>
>    <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
>                 version="0" state="full">
>      <registration aor="sip:user@example.com" id="as9"
>                    state="active">
>        <contact id="76" state="active" event="registered"
>                 duration-registered="7322"
>                 q="0.8" params="newparam=foo;audio">
>           <uri>sip:user@pc887.example.com</uri>
>        </contact>
>      </registration>
>    </reginfo>
>
> Notice how the contact parameters are basically passed untouched in 
> the "params" attribute. This makes sense if that is the only point of 
> exensibility in the schema. But, now with the structured contact, one 
> could represent these "properly" with formal schema definitions. i.e., 
> if we did a reg-event extension to support caller-prefs, it could look 
> like:
>
>  <?xml version="1.0"?>
>    <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
>             xmlns:callerprefs="urn:ietf:params:xml:ns:callerprefs"
>                 version="0" state="full">
>      <registration aor="sip:user@example.com" id="as9"
>                    state="active">
>        <contact id="76" state="active" event="registered"
>                 duration-registered="7322"
>                 q="0.8" params="newparam=foo;audio">
>           <uri>sip:user@pc887.example.com</uri>
>           <callerprefs:audio/>
>           <unknown-param name="newparam>foo</unknown-param>
>        </contact>
>      </registration>
>    </reginfo>
>
> That is, I could instead define a new element called "unknown-param" 
> in reg-event that is used for unknown params, and for those that are 
> known to the server, and for which more structured xml specifications 
> exist, you could have things like <callerprefs:audio>.
>
> The technical drawback here is that whether a parameter appears in 
> unknown-param or whether its in something like callerprefs:audio 
> depends on whether that particular extension (in this case, 
> callerprefs) is supported by the server. Another choice is to not even 
> include unknown parameters in the reg-event notifications, however I 
> think thats a bad idea.
>
> The benefit, however, is that it allows us to provide very structured 
> notifications for contact header field extensions, and it avoids the 
> need for implementing, for example, a caller-prefs parser just to 
> receive notifications about caller preferences parameters - all the 
> data is represented using XML constructs.
>
> Technically, my preference would be to define the unknown-param in 
> reg-event right now, and have extensions (like callerprefs and gruu) 
> specify xml schema extensions to reg-event for their data.
>
> Please provide comments on this proposal within the next few days.
>
> Thanks,
> Jonathan R.
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


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Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-conference-package-03.txt
Date: Wed, 18 Feb 2004 13:23:30 -0800
Message-ID: <DD07841287D0AD428833021705E0D14E01718CDD@RED-MSG-52.redmond.corp.microsoft.com>
Thread-Topic: [Sipping] I-D ACTION:draft-ietf-sipping-conference-package-03.txt
Thread-Index: AcP2TsXbvduGCvUtS/WUy5hNu8iExQAFFpqw
From: "Orit Levin" <oritl@microsoft.com>
To: "Veda Hegde" <vhegde@wave3software.com>, <sipping@ietf.org>
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No, there is none ... By design.

This draft defines a basic conference package. There are may useful
things that we could add, but decided to leave out in order to agree on
minimal set of useful and stable definitions. "Mute" status has been a
topic of controversy in the past: what side does the mute? What streams
are affected? Etc.

The current version proposed to resolve the controversy when still
keeping this useful status alive.
Only "mute by focus" is reflected in the package. Also, as a general
concept, it indicates that the user does not contribute to the
conference an any way. I would say that if voice and/or IM is not muted,
but just a noisy video is - the user is NOT muted-by-focus.

The alternative to this approach would be NOT to have the mute status at
all for the basic conference package...
I obviously voted to have it in the format proposed in the latest
version.

I hope that it will still be useful to your applications,
Orit.

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Veda Hegde
Sent: Wednesday, February 18, 2004 10:38 AM
To: sipping@ietf.org
Subject: RE: [Sipping] I-D
ACTION:draft-ietf-sipping-conference-package-03.txt

Hi,
I don't see anything that would indicate the state of individual media
streams offered by user. If a user is offering audio and video, focus
might choose to mix just the audio and not the video. In that case, is
there a way to indicate that just the video stream was muted by the
focus?
Thanks,
Veda


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Wednesday, February 18, 2004 7:18 AM
To: IETF-Announce:
Cc: sipping@ietf.org
Subject: [Sipping] I-D
ACTION:draft-ietf-sipping-conference-package-03.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Proposal
Investigation Working Group of the IETF.

	Title		: A Session Initiation Protocol (SIP) Event
Package for Conference
State
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sipping-conference-package-03.txt
	Pages		: 27
	Date		: 2004-2-18

This document defines a conference event package for the Session
   Initiation Protocol (SIP) Events framework, along with a data format
   used in notifications for this package. The conference package allows
   users to subscribe to a conference URI. Notifications are sent about
   changes in the membership of this conference, the status of users'
   participation in the conference, and the sidebars in the conference.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-conference-packag
e-03
.txt

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Thread-Topic: draft-olson-sipping-refer-extensions-01.txt
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From: "Orit Levin" <oritl@microsoft.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
        "Sean Olson" <seanol@windows.microsoft.com>
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Thanks a lot for your comments!

Answers are inline,
Orit.=20

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Wednesday, February 18, 2004 10:49 AM
To: Sean Olson; Orit Levin
Cc: sipping@ietf.org
Subject: Re: draft-olson-sipping-refer-extensions-01.txt

Wow! You made a lot of changes in this version.

I do have a number of comments about this version, below. I think I was
happier with it before. This one seems to have run amuck.

OL: At least it draws attention and generates useful comments:-)=20

	Paul

First, a nit: The introduction says:

    Finally, there are
    situations where the party issuing the REFER request does not need
    the NOTIFY associated with the REFER, perhaps because that agent is
    already subscribed to the appropriate package outside of the REFER
    request.

I agree there are situations where the referring party doesn't need the
NOTIFY, but not because it is already subscribed for it. There is no way
to independently subscribe to the refer event package.

OL: Agreed.

----------

Section 7 says:

    The REFER-Issuer MAY specify the request for subscription to a
    specific package by including the MIME type in an Accept header.

Yuk! All kinds of problems with this:

OL: I agree that it is much simpler to just suppress the implicit
subscription (and subscribe to whatever package you need independently).
From authors' perspective, it will meet our requirements. My inclination
is that we stick to this feature only if we solve all the problems
associated with it. We definitely need to work more on this part.

- event packages are not defined by mime-types, they are defined by
event package names. It is possible for several event packages to share
a mime-type for their notifications, and for multiple mime-types to be
permissible for a single event package.

- subscribing to an event package requires both the event package name
and the event id. (In the case of the special refer event package, the
event id is defined by the cseq of the REFER message itself.)

- If bundling REFER with an implicit subscription to the refer event
package was a bad idea (as most seem to think) then bundling it with the
establishment of arbitrary subscriptions is an even worse idea.

- Based on the examples, this isn't really a matter of subscribing to a
different event package. The event package named in the NOTIFY is still
"refer", but the content-type is changed. So the effect is of redefining
the refer event package so that it may return content equivalent to that
of any other event package. This is at best a kludge. It also probably
violates the rules for defining of event packages. In this model it
isn't at all clear what defines how the content of the notifies is
obtained, or when it is generated, since that is generally defined as
part of an unnamed event package.

If you really want to do this, then I think you must ammend the
definition of the refer event package to include (presumably by
reference) most of the definition of any of the other event packages
that you might want to request this way. But I still think this is a bad
idea. Even if it requires extra messages, I think subscriptions to
anthing but the refer event package should be established first, before
sending the refer.

---------

Later in that section, another nit:

    In the INVITE example, two things that are generally lacking from
the
    message/sipfrag content are the dialog identifier (Call-ID plus
local
    and remote tags) and the state of the dialog. Not coincidentally,
    this is the same information available in the application/
    dialog-info+xml MIME type [7] used for the dialog event package.

The dialog-id can be available using the sipfrag content if
refer-recipient chooses to provide them. They are available if the
fragment contains the to, from, and call-id from the invite response.

OL: Will refine the text.

(I agree that certain dialog status may not be determined via the refer
event package.)

----------

norefersub: While I think this is a good feature to support, I think it
is a stretch to encode it as a required option. That mechanism is
normally used for more general functionality that transcends a
particular request. It would be good to have a way to specify this in a
way that is more specific to refer. One alternative might be to add a
norefersub parameter to the Refer-To header.

OL: I agree. What level of support to request should be left up to the
REFER-Issuer decision.

(Also note that returning a 481 in response to the first NOTIFY may not
always be a cheap way to cancel the subscription without norefersub. If
the refer was sent within a preexisting dialog, returning a 481 to the
notify might be construed as absence of the entire dialog. If so, all
the things sharing the dialog might also be terminated. So in that case
it may be necessary to cancel by sending a subscribe with expires:0,
requiring 5 messages rather than one.)

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

In general there seem to be far more option tags than are needed. I
think it would be sufficient to define one new option tag indicating
support for all the functionality in this draft. Requesting of specific
features can be via other parameters or headers.

OL: Yes, it is one of the issues I am going to raise in the meeting:
What level of granularity we need for these extensions.

	Paul


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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: [Sipping] Re: draft-camarillo-sipping-uri-list-01
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In section 3, the following is a sleeper:

    Network administrators should note that they need to configure
    proxies to route correctly Request-URIs that contain a "list"
    parameter and are addressed to their domain.

AFAIK there is no definition of what "route correctly" means, nor is 
there anything in 3261 that would suggest that proxies be configurable 
to do this.

I presume what you mean is that when a a proxy is routing a request that 
was addressed to an AOR it controls, and that address contains a url 
parameter named "list", a copy of the list parameter should be inserted 
into each selected target sip or sips uri.

If that is what you mean, then this needs to be called out as a 3261 
extension. I suppose it doesn't need an option name, because the 
exploder won't be referenced unless it has first been deployed, and it 
can't be deployed unless there is a proxy that supports this option.

	Paul


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From: "Banibrata Dutta" <dutta@india.hp.com>
To: "'sipping'" <sipping@ietf.org>
Date: Thu, 19 Feb 2004 10:29:23 +0530
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Subject: [Sipping] OFF-TOPIC : stripping headers from TXT I-D's
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hi,
 
many of the I-D's submitted have header/footers cutting accross a
diagram in many text files.
this is not desirable many a times, for e.g. when i want to take a
hard-copy for future reference,
it's always easy reading if the full diagram or table fits one page and
not split of 2 (unless it's
unavoidable because of it's size).
 
so, won't it be better to add this as a guideline to I-D submitters i.e.
the pagination takes
care of this (although i've not idea how to do it, as the pagination
program is not that smart
unless it's modified to identify pre-existing ^L's and take that as
page-breaks.
 
finally, i doubt that the above wish would be fulfilled anytime soon, so
wanted to ask if there's
a way to remove the headers/footers smartly, i.e only the 2/3/4 lines
added by a header/
footer are removed, and all other blank-lines / white-space containing
lines are not touched.
a multi-line search-replace is what is required.
 

thanks & regards,
Banibrata Dutta
  Hewlett-Packard ISO Pvt. Ltd,
  29, Cunningham Road,
  Bangalore - 560052, India.
  Tel: +91-80-22051796
  


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D187165404-19022004>hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D187165404-19022004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>many =
of the I-D's=20
submitted have header/footers cutting accross a diagram in many text=20
files.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>this =
is not=20
desirable many a times, for e.g. when i want to take a hard-copy for =
future=20
reference,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>it's =
always easy=20
reading if the full diagram or table fits one page and not split of 2 =
(unless=20
it's</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D187165404-19022004>unavoidable because=20
of it's size).</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D187165404-19022004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>so, =
won't it be=20
better to add this as a guideline to I-D submitters i.e. the pagination=20
takes</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>care =
of this=20
(although i've not idea how to do it, as the pagination program is not =
that=20
smart</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>unless =
it's modified=20
to identify pre-existing ^L's and take that as =
page-breaks.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D187165404-19022004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D187165404-19022004>finally, i doubt=20
that the above wish would be fulfilled anytime soon, so wanted to ask if =

there's</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>a way =
to remove the=20
headers/footers smartly, i.e only the 2/3/4 lines added by a=20
header/</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>footer =
are removed,=20
and all other blank-lines / white-space containing lines are not=20
touched.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D187165404-19022004>a =
multi-line=20
search-replace is what is required.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV><!-- Converted from =
text/plain format -->
<P><FONT size=3D2>thanks &amp; regards,<BR>Banibrata Dutta<BR>&nbsp;=20
Hewlett-Packard ISO Pvt. Ltd,<BR>&nbsp; 29, Cunningham Road,<BR>&nbsp; =
Bangalore=20
- 560052, India.<BR>&nbsp; Tel: +91-80-22051796<BR>&nbsp;</FONT>=20
</P></BODY></HTML>

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Subject: Re: [Sipping] Re: sos - dead or alive
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>
Cc: "Cullen Jennings" <fluffy@cisco.com>, sipping@ietf.org
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Bear in mind that the "most scarce resource" in the emergency response
system is usually the human operators.

So any kind of regular, scheduled test should be designed so it does  NOT
tie up the operators.

Janet

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                      "Henning                                                                                                  
                      Schulzrinne"             To:      "Cullen Jennings" <fluffy@cisco.com>                                    
                      <hgs                     cc:      <sipping@ietf.org>                                                      
                      @cs.columbia.edu         Subject: [Sipping] Re: sos - dead or alive                                       
                      >                                                                                                         
                                                                                                                                
                      02/17/2004 11:18                                                                                          
                      AM                                                                                                        
                                                                                                                                
                                                                                                                                





> I would prefer MUST to RECOMMENDED since a recent informal poll at sipit
> made it quiet clear that most implementers view SHOULD level requirements
as
> something to consider in the next lifetime. I'm not arguing that this
poll
> means we should not use SHOULD in general, but I think due to the
critical
> nature of this specification, we should be a little stronger than normal.
I
> don't want vendors saying they support this RFC fully when they can't
detect
> that a safety critical system is broken.

Gladly obliged.

>
> I would prefer specific test times - something like the test should be
> performed at a random time between 12 and 36 hours. Phones must
prominently

Currently, there are about 200 million real 911 calls a year. Testing
once per day would essentially impose this load once per day. While I
don't think building proxies that can handle this is a bad idea (among
other advantages, it provides ample margin of safety and load-resistance
since real calls are then a tiny fraction of the test traffic), it does
need to be considered.

 From local usage, I know that fire alarm bells at Columbia are tested
weekly, so that may be a more reasonable schedule.

> display the information that 911 is broken using a display or by changing
> the dial tone on all calls. Would be nice to consider the impact of every
> phone doing this. Phones SHOULD not do a test immediately after booting.
> Phones MAY provide a UI to allow a user to manual imitate a test - this
is
> very useful for someone that is trying to check a 911 system but does not
> want to wait 36 hours.

I like the gist of this and have added appropriate text.

>
> Cullen
>
>
>

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Subject: Re: [Sipping] Re: sos - dead or alive
To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: Cullen Jennings <fluffy@cisco.com>, sipping@ietf.org
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Sounds good then.


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                      Henning                                                                                                   
                      Schulzrinne <hgs         To:      Janet P Gunn <jgunn6@csc.com>                                           
                      @cs.columbia.edu         cc:      Cullen Jennings <fluffy@cisco.com>, sipping@ietf.org                    
                      >                        Subject: Re: [Sipping] Re: sos - dead or alive                                   
                                                                                                                                
                      02/17/2004 02:14                                                                                          
                      PM                                                                                                        
                                                                                                                                
                                                                                                                                




The OPTIONS mechanism does exactly that - it's mechanical only.

Janet P Gunn wrote:

> Bear in mind that the "most scarce resource" in the emergency response
> system is usually the human operators.
>
> So any kind of regular, scheduled test should be designed so it does  NOT
> tie up the operators.






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Thread-Topic: Re: [Sipping] draft-cgkim-sipping-info-00
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Subject: RE: Re: [Sipping] draft-cgkim-sipping-info-00
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IM seems to be appropriate for PTZ control and MRCP for exchanging =
sesison commands under RTP circumstances if I define some applications =
specific to this project.=0A=
Camera control, however, needs to be implemented quite simplely to =
reduce the cost of stacking SIP in the control unit.=0A=
It seems to me that this suggestions are heavy a little bit for this =
purpose, though. =0A=
Aren't they ?=0A=
=0A=
=0A=
-----=BF=F8=BA=BB =B8=DE=BD=C3=C1=F6-----=0A=
=BA=B8=B3=BD =BB=E7=B6=F7: "Henning Schulzrinne" <hgs@cs.columbia.edu>=0A=
=BA=B8=B3=BD =B3=AF=C2=A5: 2004-02-15 =BF=C0=C0=FC 2:36:59=0A=
=B9=DE=B4=C2 =BB=E7=B6=F7: "Cullen Jennings" <fluffy@cisco.com>=0A=
=C2=FC=C1=B6: "sipping" <>=0A=
=C1=A6=B8=F1: Re: [Sipping] draft-cgkim-sipping-info-00=0A=
=0A=
=0A=
IM works quite well for this and similar applications. We use it to=0A=
control other devices beyond cameras, too, as sending an IM saying "look=0A=
at me" and "move camera 20 degrees" are semantically rather similar.=0A=
=0A=
Should this be in MMUSIC?=0A=
=0A=
Cullen Jennings wrote:=0A=
=0A=
> I think this topic of far end camera control is important but I don't =
think=0A=
> this is the right way to do it. It seems like it would work better to =
use=0A=
> SIP to set up a session and use the data in this session to do the =
control.=0A=
> Sort of like MRCPv2 or session mode IM does. Have you considered =
something=0A=
> like using session mode IM to send mime bodies controlled the camera. =
One=0A=
> could define a mime type like application/fecc and send IM with a body =
like=0A=
>=0A=
> <pan-relative angle=3D"20" \>=0A=
>=0A=
> To pan clockwise by 20 degrees from the present position. Anyway, I =
don't=0A=
> know the best way to do this and we should probably figure out =
requirements=0A=
> before jumping into solutions but I would be glad to see some work =
like this=0A=
> get done. Not sure that sipping is the best place for it.=0A=
>=0A=
> Cullen=0A=
>=0A=
>=0A=
> =0A=
>=0A=
>=0A=
> _______________________________________________=0A=
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping=0A=
> This list is for NEW development of the application of SIP=0A=
> Use sip-implementors@cs.columbia.edu for questions on current sip=0A=
> Use sip@ietf.org for new developments of core SIP=0A=
=0A=
_______________________________________________=0A=
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping=0A=
This list is for NEW development of the application of SIP=0A=
Use sip-implementors@cs.columbia.edu for questions on current sip=0A=
Use sip@ietf.org for new developments of core SIP
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Date: Thu, 19 Feb 2004 09:03:29 EST
Subject: Re: [Sipping] Re: sos - dead or alive
To: jgunn6@csc.com, hgs@cs.columbia.edu
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In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, jgunn6@csc.com 
writes:


> So any kind of regular, scheduled test should be designed so it does  NOT
> tie up the operators.
> 
I believe the test must be devised so that it does not involve the emergency 
center or any of the equipment directly supporting the emergency center at all.

Mike



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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/19/2004 1:58:18 AM Eastern Standard Time, jgunn6@csc.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">So any kind of regular, sch=
eduled test should be designed so it does &nbsp;NOT
<BR>tie up the operators.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">I believe the test must be devised so that it does not involve the em=
ergency center or any of the equipment directly supporting the emergency cen=
ter at all.
<BR>
<BR>Mike
<BR>
<BR></FONT></HTML>

--part1_3d.3ae4ae03.2d661c31_boundary--

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Mpierce1@aol.com'" <Mpierce1@aol.com>, jgunn6@csc.com,
        hgs@cs.columbia.edu
Cc: sipping@ietf.org
Subject: RE: [Sipping] Re: sos - dead or alive
Date: Thu, 19 Feb 2004 09:29:08 -0500
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I don't agree.  To be effective, you have to know you can get to a system 
that itself has certain knowledge that the call will reach an emergency 
call taker.  Since there is not necessarily any "carrier" involved, you 
have to get to a system that is run by the emergency response system in 
the jurisdiction that you are located.  It has to route (in the IP sense) 
the same as a real emergency call would route.  It doesn't have to get all 
the way to an operator, but it does have to get close.

If the result of the process (such as my suggestion in 
draft-rosen-dns-sos-00) to obtain the URL of the ECC serving the location 
of the caller is sip:anyone@emergency.northwest.allegheny.pa.us, then test 
calls might go to sip:verify@test.northwest.allegheny.pa.us.  That may be 
a different host, but it would share the same router and route that a real 
emergency call would.

It could be that a jurisdiction above the ERC runs a network that handles
all emergency calls within its jurisdictions, and test calls could terminate
there.  For example, the State of Vermont could have all calls within
Vermont have a test service at sip:verify@test.vt.us, as long as it also
had all the emergency calls going to, for example,
sip:anyone@emergency.vt.us.
That might be a proxy server that did more fine grained routing based on
location to the actual ERC that served the caller.  For example, it might
rewrite the Request-URI to sip:anyone@burlington.emergency.vt.us and route
the call over its network to the ERC in the city of Burlington.

Brian
-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Thursday, February 19, 2004 9:03 AM
To: jgunn6@csc.com; hgs@cs.columbia.edu
Cc: sipping@ietf.org
Subject: Re: [Sipping] Re: sos - dead or alive


In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time,
jgunn6@csc.com writes: 



So any kind of regular, scheduled test should be designed so it does  NOT 
tie up the operators. 


I believe the test must be devised so that it does not involve the emergency
center or any of the equipment directly supporting the emergency center at
all. 

Mike 

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



From exim@www1.ietf.org  Thu Feb 19 10:27:41 2004
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] Send your agenda requests
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Folks,

we will be elaborating the agenda for SIPPING shortly (the deadline for 
WG agendas is approaching). So, send me your agenda requests *now*. So 
far, this is what we have:

http://www.softarmor.com/sipping/meets/ietf59/requests.html

Regards,

Gonzalo
SIPPING co-chair

 

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Hi Everyone,

I've updated the dialog packages with the changes we agreed to in 
Minneapolis (removing CSeq and the Route-Set).

I've also provided a detailed example of using the dialog package for a 
"shared-line" application.  I apologize for my tardiness here.  I was 
supposed to post something shortly after the meeting.  I any case, I 
believe this demonstrates that the current syntax is at least 
sufficient for the use cases we've discussed.  I would like to hear 
from all of you if this does the trick.

many thanks,
-rohan


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



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Subject: [Sipping] RFC 3702 on Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP)
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--NextPart


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


        RFC 3702

        Title:      Authentication, Authorization, and Accounting
                    Requirements for the Session Initiation Protocol
                    (SIP)
        Author(s):  J. Loughney, G. Camarillo
        Status:     Informational
        Date:       February 2004
        Mailbox:    John.Loughney@nokia.com,
                    Gonzalo.Camarillo@ericsson.com
        Pages:      15
        Characters: 31243
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sipping-aaa-req-04.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3702.txt


As Session Initiation Protocol (SIP) services are deployed on the
Internet, there is a need for authentication, authorization, and
accounting of SIP sessions.  This document sets out the basic
requirements for this work.

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This memo provides information for the Internet community.  It does
not specify an Internet standard of any kind.  Distribution of this
memo is unlimited.

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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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



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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: =?UTF-8?B?7Zmp7KeE6rK9?= <jkhwang@kt.co.kr>,
        Cullen Jennings <fluffy@cisco.com>, sipping <sipping@ietf.org>
Subject: Re: [Sipping] draft-hwang-sipping-infomidcall-00.txt
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Mr. Wang is saying that they make money doing the services their way, whi=
ch is=20
also possible using SIP.  Your "SIP services model" is one business model=
, but=20
others are possible and more in line with existing businesses.

Jonathan Rosenberg wrote:
> I agree that these services are needed. However, mechanisms are in plac=
e=20
> today for performing these services. Call transfer is done with REFER,=20
> call waiting is a local feature on the device, and three way calling ca=
n=20
> be done locally or through invocation of a centralized conferencing ser=
ver.
>=20
> In the SIp services model, all of these features do not live in a=20
> centralized server in the network. Rather, they are distributed, and=20
> implemented in the place most appropriate for that service. In an IP=20
> network, the endpoints have greater intelligence than in the PSTN.=20
> Having smarter endpoints means that you can have a better overall=20
> service experience and support a broader range of devices.
>=20
> MGCP takes the position you are advocating, that the services all live=20
> in a centralized place. Cullen's point, I believe, is that, if the aim=20
> is to develop protocols to support a model of centralized service=20
> control, we already have that - mgcp.
>=20
> Thanks,
> Jonathan R.
>=20
> =ED=99=A9=EC=A7=84=EA=B2=BD wrote:
>=20
>> Thank you for the question.
>>
>> Your question is why network-based midcall is needed with SIP.
>>
>> My answer is that
>> Since service components in the next gen IP network are based on SIP,
>> the delivery of midcall information should be through SIP method.
>>
>> Simple INFO (from the SIP phone or from the MGC(by translation)) works=
=20
>> very well,
>> as we implemented trial services with this INFO midcall function both=20
>> in SIP phone and Softswitch.
>>
>> My proposal is that in standard SIP phone (UA) have midcall button=20
>> simply initiating INFO when the button is pressed.
>>
>> -------------------------------------------
>> Title: INFO Usage Examples for Network-based Mid-Call Service Author:=20
>> Jinkyung Hwang (KT)
>>
>> Abstract:
>>
>>   This document describes the INFO usage examples for network based  =20
>> mid-call service. Mid-call based services are for example, Call  =20
>> Transfer, Call Waiting, and Three-Way Calling. Other possible mid-
>>   call services are useful and important, even more in IP multimedia  =
=20
>> network. The cases described in SIP service examples (draft-ietf-
>>   sipping-service-examples-05) are Terminal-basis. Since the service  =
=20
>> procedures are fixed in the Terminal, the features are not extensible.=
=20
>>   This document proposes network-based mid-call control mechanism with=
=20
>>   INFO and shows examples that simple mid-call request (INFO) from the=
=20
>>   Terminal, then the network application server provide various=20
>> mid-call   services by prompting user through the Media Server,=20
>> according to the   subscription information of the user.
>> A URL for this draft is
>>  http://www.ietf.org/internet-drafts/draft-hwang-sipping-infomidcall-0=
0.txt=20
>>
>>
>>
>> BR
>>
>> Jinkyung Hwang
>> KT
>>
>> ----- Original Message ----- From: "Cullen Jennings" <fluffy@cisco.com=
>
>> To: "sipping" <sipping@ietf.org>
>> Sent: Saturday, February 14, 2004 1:26 AM
>> Subject: [Sipping] draft-hwang-sipping-infomidcall-00.txt
>>
>>
>>
>>>    Why not just use MGCP instead of SIP if you want to do this. It=20
>>> will work
>>> much better.
>>>
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>> J*ix&j)b    bm?=0C0'~ffX)=DF=A3"8bX+=1FDY=D7=AFzZ)ayi'=1A*'=3DK=1E*b^=
?=1C%fin~-w.=C8=A9R=C7=AC++=1Dz(}+y"=0F
>=20
>=20

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



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From: "Scoggins, Sophia" <sscoggins@telogy.com>
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My group is looking for some security encryption protocol for both SIP and
RTP streams for CPE. We were wondering what encryption protocol that most
SIP proxy servers currently support or intend to support? Would you please
just send your reply to me?

 

Thanks.

 

Regards,

Sophia2@ti.com


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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>My group is looking for some security encryption protocol
for both SIP and RTP streams for CPE. We were wondering what encryption
protocol that most SIP proxy servers currently support or intend to support? Would
you please just send your reply to me?</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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_______________________________________________
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Subject: RE: [Sipping] SIP Proxy servers -- security encryption protocol
Date: Fri, 20 Feb 2004 10:56:55 +0530
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Hi
    For secured transactions you could use TLS i.e using sips: instead
of normal sip: url.
Alternatively IPSec also could be used at IP Layer
and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt)
protocol.
=20
Regards
Ranjit

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Scoggins, Sophia
Sent: Friday, February 20, 2004 10:45 AM
To: 'sipping@ietf.org'
Subject: [Sipping] SIP Proxy servers -- security encryption protocol



=20

My group is looking for some security encryption protocol for both SIP
and RTP streams for CPE. We were wondering what encryption protocol that
most SIP proxy servers currently support or intend to support? Would you
please just send your reply to me?

=20

Thanks.

=20

Regards,

Sophia2@ti.com


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<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2>Hi</FONT></SPAN></DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2>&nbsp;&nbsp;&nbsp; For secured transactions you could use TLS =
i.e using=20
sips: instead of normal sip: url.</FONT></SPAN></DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2>Alternatively IPSec also could be used at IP =
Layer</FONT></SPAN></DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =
size=3D2>and=20
for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt)=20
protocol.</FONT></SPAN></DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards</FONT></SPAN></DIV>
<DIV><SPAN class=3D419333105-20022004><FONT face=3DArial color=3D#0000ff =

size=3D2>Ranjit</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] <B>On Behalf Of =

  </B>Scoggins, Sophia<BR><B>Sent:</B> Friday, February 20, 2004 10:45=20
  AM<BR><B>To:</B> 'sipping@ietf.org'<BR><B>Subject:</B> [Sipping] SIP =
Proxy=20
  servers -- security encryption protocol<BR><BR></FONT></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">My group is looking for =
some=20
  security encryption protocol for both SIP and RTP streams for CPE. We =
were=20
  wondering what encryption protocol that most SIP proxy servers =
currently=20
  support or intend to support? Would you please just send your reply to =

  me?</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Thanks.</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT>&nbsp;</P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Regards,</SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Sophia2@ti.com</SPAN></FONT></P></DIV></BLOCKQUOTE></BODY></HTML>

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



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From: "Scoggins, Sophia" <sscoggins@telogy.com>
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Subject: [Sipping] VoIP (SIP) Outsourcing opportunities
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If this is not a proper place to post this email, I apologize.

 

I am with the Broadband Communications group at Texas Instruments. My group
is developing VoIP products (IP phones and small media gateways) using SIP.
We have a VoIP project that we are looking for outsourcing to a VoIP
experienced company in the U.S. (U.S. only, please). If you are interested,
please contact me. Thanks.

 

 

Regards,

Sophia2@ti.com

 


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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>If this is not a proper place to post this email, I apologize.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am with the Broadband Communications group at Texas
Instruments. My group is developing VoIP products (IP phones and small media
gateways) using SIP. We have a VoIP project that we are looking for outsourcing
to a VoIP experienced company in the </span></font><font size=2 face=Arial><span
  style='font-size:10.0pt;font-family:Arial'>U.S.</span></font><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'> (</span></font><font
  size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>U.S.</span></font><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'> only,
please). If you are interested, please contact me. Thanks.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Regards,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Sophia2@ti.com</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>&nbsp;</span></font></p>

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------_=_NextPart_001_01C3F774.1290C390--

_______________________________________________
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Hi:

One of the more cofusing things I found this draft is related to how the =
subscribed event is selected in the REFER request.

I believe that we want to explicitly indicate to the REFER-Recipient =
which event the REFER-Issuer wants to subscribe to. We may want to =
indicate also which events and formats are supported.

If we mimic the SUBSCRIBE behaviour, the event of the subscription is =
indicated in an Event header. The supported events are indicated in the =
Allow-events header. As for the format, you have the Accept header, as =
you point out in the draft.

So my question is: Have you consider to import the Event and =
Allow-events header in the extended REFER?

/Miguel

---
Miguel A. Garcia           tel:+358-50-4804586   =20
Nokia Research Center      Helsinki, Finland

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Date: Fri, 20 Feb 2004 09:41:42 -0500
To: Mpierce1@aol.com
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] Re: sos - dead or alive
Cc: jgunn6@csc.com, hgs@cs.columbia.edu, sipping@ietf.org
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At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>jgunn6@csc.com writes:
>
>
>>So any kind of regular, scheduled test should be designed so it does  NOT
>>tie up the operators.
>
>I believe the test must be devised so that it does not involve the 
>emergency center or any of the equipment directly supporting the emergency 
>center at all.
>
>Mike

Then there is not much point in doing any test.  As Brian points out, the 
question is how do you stage these reachability tests so that they scale, 
ie. you don't have millions of users pinging the ECC directly.  But, 
instead of the proxies pinging the ECC, which might reduce the load, but 
might not make the burstiness more deterministic, perhaps a presence 
subscription would be more appropriate.  The proxies could subscribe to the 
presence of the ECC, and the ECC could periodically Notify those proxies 
that it is still there and able to receive calls.  The ECC could then 
control the scheduling and load distribution for sending the 
Notifies.  Failing to receive the notifications beyond some interval could 
provide an indication that something is wrong and trigger an inquiry.  You 
might even be able to extend this to the users, where emergency help is a 
"buddy" you subscribe to on the local proxy.  I haven't thought this 
through, but thought I'd throw that out.

Mike H.


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Date: Fri, 20 Feb 2004 10:00:26 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Michael Hammer <mhammer@cisco.com>
CC: Mpierce1@aol.com, jgunn6@csc.com, sipping@ietf.org
Subject: Re: [Sipping] Re: sos - dead or alive
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I think there are five goals, somewhat contradictory:

(1) make it as realistic as possible, to make sure you are really 
testing the emergency calling system, not the testing system; after all, 
the fire safety officer doesn't go around ringing a cowbell instead of 
the real fire alarms.

(2) do not impose any additional operator involvement; this is solved by 
the OPTIONS mechanism proposed;

(3) allow for reasonable scaling;

(4) avoid interfering with real emergency calls

(5) allow user on-demand testing.

With the proxy-based testing I proposed, we get very close to all goals. 
In the worst case, i.e., every device does this once a day, we get 
roughly 100 million calls a day in the US, or 1157/second. That's about 
10 proxy servers worth of equipment, i.e., about $20k. This hardly seems 
worth spending inordinate engineering resources on avoiding, 
particularly since the likely load is going to be lower by probably a 
factor of 1000 or more (I'm assuming that most residential users will 
use a cable or DSL provider for this, for example).

With even a few hundred systems, you'll get almost perfect smoothing and 
very little burstiness.

I know protocol design is fun, but I'm very worried about adding 
complexity to solve largely a non-problem.

Michael Hammer wrote:

> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
> 
>> In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>> jgunn6@csc.com writes:
>>
>>
>>> So any kind of regular, scheduled test should be designed so it does  
>>> NOT
>>> tie up the operators.
>>
>>
>> I believe the test must be devised so that it does not involve the 
>> emergency center or any of the equipment directly supporting the 
>> emergency center at all.
>>
>> Mike
> 
> 
> Then there is not much point in doing any test.  As Brian points out, 
> the question is how do you stage these reachability tests so that they 
> scale, ie. you don't have millions of users pinging the ECC directly.  
> But, instead of the proxies pinging the ECC, which might reduce the 
> load, but might not make the burstiness more deterministic, perhaps a 
> presence subscription would be more appropriate.  The proxies could 
> subscribe to the presence of the ECC, and the ECC could periodically 
> Notify those proxies that it is still there and able to receive calls.  
> The ECC could then control the scheduling and load distribution for 
> sending the Notifies.  Failing to receive the notifications beyond some 
> interval could provide an indication that something is wrong and trigger 
> an inquiry.  You might even be able to extend this to the users, where 
> emergency help is a "buddy" you subscribe to on the local proxy.  I 
> haven't thought this through, but thought I'd throw that out.
> 
> Mike H.


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This list is for NEW development of the application of SIP
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Michael Hammer'" <mhammer@cisco.com>, Mpierce1@aol.com
Cc: jgunn6@csc.com, hgs@cs.columbia.edu, sipping@ietf.org
Subject: RE: [Sipping] Re: sos - dead or alive
Date: Fri, 20 Feb 2004 10:00:26 -0500
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I think Hennings idea is the gem of the answer.

Assume that there are a couple of proxies between you and the ERC.

If the proxy closest to the ERC assures itself that it can reach the ERC,
then it can assure any proxy that routes through it to the ERC that it
can reach the ERC without the second level proxy having to actually touch
the ERC itself.  Repeat as necessary.  That scales better than having
each UA doing something that results in messages hitting the ERC itself.

There is no explicit mechanism in SIP to do this.
  It could be a Notify against some kind of event package we define
  It could be a kind of presence subscription
  It could be a kind of INVITE that returns a new error code
  It could be a new method that returns a new response code

When considering what the best mechanism is, remember that you have
a URI of the ERC, but you don't have a URI to an intermediate proxy.
We could figure out how to get you one.

The question you are asking is "If I sent a (real) INVITE to this
URI, would it get there?"  If you are subscribing to something, how do 
you know what to subscribe to?

Brian

> -----Original Message-----
> From: Michael Hammer [mailto:mhammer@cisco.com]
> Sent: Friday, February 20, 2004 9:42 AM
> To: Mpierce1@aol.com
> Cc: jgunn6@csc.com; hgs@cs.columbia.edu; sipping@ietf.org
> Subject: Re: [Sipping] Re: sos - dead or alive
> 
> 
> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
> >In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
> >jgunn6@csc.com writes:
> >
> >
> >>So any kind of regular, scheduled test should be designed 
> so it does  NOT
> >>tie up the operators.
> >
> >I believe the test must be devised so that it does not involve the 
> >emergency center or any of the equipment directly supporting 
> the emergency 
> >center at all.
> >
> >Mike
> 
> Then there is not much point in doing any test.  As Brian 
> points out, the 
> question is how do you stage these reachability tests so that 
> they scale, 
> ie. you don't have millions of users pinging the ECC directly.  But, 
> instead of the proxies pinging the ECC, which might reduce 
> the load, but 
> might not make the burstiness more deterministic, perhaps a presence 
> subscription would be more appropriate.  The proxies could 
> subscribe to the 
> presence of the ECC, and the ECC could periodically Notify 
> those proxies 
> that it is still there and able to receive calls.  The ECC could then 
> control the scheduling and load distribution for sending the 
> Notifies.  Failing to receive the notifications beyond some 
> interval could 
> provide an indication that something is wrong and trigger an 
> inquiry.  You 
> might even be able to extend this to the users, where 
> emergency help is a 
> "buddy" you subscribe to on the local proxy.  I haven't thought this 
> through, but thought I'd throw that out.
> 
> Mike H.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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Date: Fri, 20 Feb 2004 10:41:45 -0500
To: Henning Schulzrinne <hgs@cs.columbia.edu>
From: Michael Hammer <mhammer@cisco.com>
Subject: Re: [Sipping] Re: sos - dead or alive
Cc: Mpierce1@aol.com, jgunn6@csc.com, sipping@ietf.org
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Henning,

The reality of deployment would be that every ECC would have it's own 
proxy, so that would be more like 1000 x $2k = $2 million.  But, the 
corresponding distribution of load across those servers means it is even 
more of a non-problem at any given site.

One question:  A real successful test to a wrong ECC doesn't buy you 
much.  I hope that mobility is addressed in the solution as well.  Many of 
those 100 million phones are mobile, which might become SIP-based some time 
in the future.  so I would add a 6) "works with mobile terminals" to your 
list below.

Mike

At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
>I think there are five goals, somewhat contradictory:
>
>(1) make it as realistic as possible, to make sure you are really testing 
>the emergency calling system, not the testing system; after all, the fire 
>safety officer doesn't go around ringing a cowbell instead of the real 
>fire alarms.
>
>(2) do not impose any additional operator involvement; this is solved by 
>the OPTIONS mechanism proposed;
>
>(3) allow for reasonable scaling;
>
>(4) avoid interfering with real emergency calls
>
>(5) allow user on-demand testing.
>
>With the proxy-based testing I proposed, we get very close to all goals. 
>In the worst case, i.e., every device does this once a day, we get roughly 
>100 million calls a day in the US, or 1157/second. That's about 10 proxy 
>servers worth of equipment, i.e., about $20k. This hardly seems worth 
>spending inordinate engineering resources on avoiding, particularly since 
>the likely load is going to be lower by probably a factor of 1000 or more 
>(I'm assuming that most residential users will use a cable or DSL provider 
>for this, for example).
>
>With even a few hundred systems, you'll get almost perfect smoothing and 
>very little burstiness.
>
>I know protocol design is fun, but I'm very worried about adding 
>complexity to solve largely a non-problem.
>
>Michael Hammer wrote:
>
>>At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
>>
>>>In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>>>jgunn6@csc.com writes:
>>>
>>>
>>>>So any kind of regular, scheduled test should be designed so it does
>>>>NOT
>>>>tie up the operators.
>>>
>>>
>>>I believe the test must be devised so that it does not involve the 
>>>emergency center or any of the equipment directly supporting the 
>>>emergency center at all.
>>>
>>>Mike
>>
>>Then there is not much point in doing any test.  As Brian points out, the 
>>question is how do you stage these reachability tests so that they scale, 
>>ie. you don't have millions of users pinging the ECC directly.
>>But, instead of the proxies pinging the ECC, which might reduce the load, 
>>but might not make the burstiness more deterministic, perhaps a presence 
>>subscription would be more appropriate.  The proxies could subscribe to 
>>the presence of the ECC, and the ECC could periodically Notify those 
>>proxies that it is still there and able to receive calls.
>>The ECC could then control the scheduling and load distribution for 
>>sending the Notifies.  Failing to receive the notifications beyond some 
>>interval could provide an indication that something is wrong and trigger 
>>an inquiry.  You might even be able to extend this to the users, where 
>>emergency help is a "buddy" you subscribe to on the local proxy.  I 
>>haven't thought this through, but thought I'd throw that out.
>>Mike H.


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



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Thread-Topic: event in draft-olson-sipping-refer-extensions-01
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From: "Orit Levin" <oritl@microsoft.com>
To: <Miguel.An.Garcia@nokia.com>, "Sean Olson" <seanol@windows.microsoft.com>
Cc: <sipping@ietf.org>
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Subject: [Sipping] RE: event in draft-olson-sipping-refer-extensions-01
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The draft is more a requirements draft (i.e. SIPPING) rather than the
mechanism. I totally agree that the event specification needs much
better definition and re-using SUBSCRIBE syntax makes a lot of sense to
me.

I would also like to hear comments what of the extensions proposed in
the draft we really need and want to promote towards SIP. In my opinion,
out of all the extensions, the implicit "extended" subscription is the
most questionable.

Thanks,
Orit.

-----Original Message-----
From: Miguel.An.Garcia@nokia.com [mailto:Miguel.An.Garcia@nokia.com]=20
Sent: Friday, February 20, 2004 2:06 AM
To: Sean Olson; Orit Levin
Cc: sipping@ietf.org
Subject: event in draft-olson-sipping-refer-extensions-01

Hi:

One of the more cofusing things I found this draft is related to how the
subscribed event is selected in the REFER request.

I believe that we want to explicitly indicate to the REFER-Recipient
which event the REFER-Issuer wants to subscribe to. We may want to
indicate also which events and formats are supported.

If we mimic the SUBSCRIBE behaviour, the event of the subscription is
indicated in an Event header. The supported events are indicated in the
Allow-events header. As for the format, you have the Accept header, as
you point out in the draft.

So my question is: Have you consider to import the Event and
Allow-events header in the extended REFER?

/Miguel

---
Miguel A. Garcia           tel:+358-50-4804586   =20
Nokia Research Center      Helsinki, Finland

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@mci.com>
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Subject: [Sipping] Language Preference in draft-sinnreich-sipdev-req-03.txt
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   International-1, language support: SIP devices SHOULD indicate the 
   preferred language using SIP Caller Preferences. The setting for 
   this header MUST be provisioned.  

Might this be expanded a bit?  The MUST be provisioned implies you can't
have a UI choice.  Maybe a default must be provisioned.

Please could we also have a note that this choice may be used to guide
how the call is answered as well as affect text on a UI.  Imagine a call 
center with some bilingual call takers.  Language preference might guide 
which call taker got the call.  Clearly it would also guide how softkeys 
were labeled, what announcements were played, etc.

I'd actually like to see the first SHOULD be a MUST.  Emergency calls,
in particular, need to be routed to someone who can actually communicate
with the caller.

Brian


       

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The mobile issue is interesting. I would think that a mobile phone might 
want to test connectivity every time it moved into a different 
geographical area, or every time it roamed to a cell under different 
administration. That could up the load considerably.

	Paul

Michael Hammer wrote:
> Henning,
> 
> The reality of deployment would be that every ECC would have it's own 
> proxy, so that would be more like 1000 x $2k = $2 million.  But, the 
> corresponding distribution of load across those servers means it is even 
> more of a non-problem at any given site.
> 
> One question:  A real successful test to a wrong ECC doesn't buy you 
> much.  I hope that mobility is addressed in the solution as well.  Many 
> of those 100 million phones are mobile, which might become SIP-based 
> some time in the future.  so I would add a 6) "works with mobile 
> terminals" to your list below.
> 
> Mike
> 
> At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
> 
>> I think there are five goals, somewhat contradictory:
>>
>> (1) make it as realistic as possible, to make sure you are really 
>> testing the emergency calling system, not the testing system; after 
>> all, the fire safety officer doesn't go around ringing a cowbell 
>> instead of the real fire alarms.
>>
>> (2) do not impose any additional operator involvement; this is solved 
>> by the OPTIONS mechanism proposed;
>>
>> (3) allow for reasonable scaling;
>>
>> (4) avoid interfering with real emergency calls
>>
>> (5) allow user on-demand testing.
>>
>> With the proxy-based testing I proposed, we get very close to all 
>> goals. In the worst case, i.e., every device does this once a day, we 
>> get roughly 100 million calls a day in the US, or 1157/second. That's 
>> about 10 proxy servers worth of equipment, i.e., about $20k. This 
>> hardly seems worth spending inordinate engineering resources on 
>> avoiding, particularly since the likely load is going to be lower by 
>> probably a factor of 1000 or more (I'm assuming that most residential 
>> users will use a cable or DSL provider for this, for example).
>>
>> With even a few hundred systems, you'll get almost perfect smoothing 
>> and very little burstiness.
>>
>> I know protocol design is fun, but I'm very worried about adding 
>> complexity to solve largely a non-problem.
>>
>> Michael Hammer wrote:
>>
>>> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
>>>
>>>> In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>>>> jgunn6@csc.com writes:
>>>>
>>>>
>>>>> So any kind of regular, scheduled test should be designed so it does
>>>>> NOT
>>>>> tie up the operators.
>>>>
>>>>
>>>>
>>>> I believe the test must be devised so that it does not involve the 
>>>> emergency center or any of the equipment directly supporting the 
>>>> emergency center at all.
>>>>
>>>> Mike
>>>
>>>
>>> Then there is not much point in doing any test.  As Brian points out, 
>>> the question is how do you stage these reachability tests so that 
>>> they scale, ie. you don't have millions of users pinging the ECC 
>>> directly.
>>> But, instead of the proxies pinging the ECC, which might reduce the 
>>> load, but might not make the burstiness more deterministic, perhaps a 
>>> presence subscription would be more appropriate.  The proxies could 
>>> subscribe to the presence of the ECC, and the ECC could periodically 
>>> Notify those proxies that it is still there and able to receive calls.
>>> The ECC could then control the scheduling and load distribution for 
>>> sending the Notifies.  Failing to receive the notifications beyond 
>>> some interval could provide an indication that something is wrong and 
>>> trigger an inquiry.  You might even be able to extend this to the 
>>> users, where emergency help is a "buddy" you subscribe to on the 
>>> local proxy.  I haven't thought this through, but thought I'd throw 
>>> that out.
>>> Mike H.
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>,
        Michael Hammer
	 <mhammer@cisco.com>
Cc: Henning Schulzrinne <hgs@cs.columbia.edu>, Mpierce1@aol.com,
        jgunn6@csc.com, sipping@ietf.org
Subject: RE: [Sipping] Re: sos - dead or alive
Date: Fri, 20 Feb 2004 14:11:25 -0500
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Depending on how we end up implementing routing to the
correct ERC, the service boundary information will be available
to either the UA or a proxy.  When the mobile device roams
beyond the boundary, it could trigger a new test.

Agree that it would up the load.  Not clear that it will make
a tremendous increase, especially if Henning's idea prevails.

Brian

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, February 20, 2004 2:04 PM
> To: Michael Hammer
> Cc: Henning Schulzrinne; Mpierce1@aol.com; jgunn6@csc.com;
> sipping@ietf.org
> Subject: Re: [Sipping] Re: sos - dead or alive
> 
> 
> The mobile issue is interesting. I would think that a mobile 
> phone might 
> want to test connectivity every time it moved into a different 
> geographical area, or every time it roamed to a cell under different 
> administration. That could up the load considerably.
> 
> 	Paul
> 
> Michael Hammer wrote:
> > Henning,
> > 
> > The reality of deployment would be that every ECC would 
> have it's own 
> > proxy, so that would be more like 1000 x $2k = $2 million.  
> But, the 
> > corresponding distribution of load across those servers 
> means it is even 
> > more of a non-problem at any given site.
> > 
> > One question:  A real successful test to a wrong ECC 
> doesn't buy you 
> > much.  I hope that mobility is addressed in the solution as 
> well.  Many 
> > of those 100 million phones are mobile, which might become 
> SIP-based 
> > some time in the future.  so I would add a 6) "works with mobile 
> > terminals" to your list below.
> > 
> > Mike
> > 
> > At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
> > 
> >> I think there are five goals, somewhat contradictory:
> >>
> >> (1) make it as realistic as possible, to make sure you are really 
> >> testing the emergency calling system, not the testing 
> system; after 
> >> all, the fire safety officer doesn't go around ringing a cowbell 
> >> instead of the real fire alarms.
> >>
> >> (2) do not impose any additional operator involvement; 
> this is solved 
> >> by the OPTIONS mechanism proposed;
> >>
> >> (3) allow for reasonable scaling;
> >>
> >> (4) avoid interfering with real emergency calls
> >>
> >> (5) allow user on-demand testing.
> >>
> >> With the proxy-based testing I proposed, we get very close to all 
> >> goals. In the worst case, i.e., every device does this 
> once a day, we 
> >> get roughly 100 million calls a day in the US, or 
> 1157/second. That's 
> >> about 10 proxy servers worth of equipment, i.e., about $20k. This 
> >> hardly seems worth spending inordinate engineering resources on 
> >> avoiding, particularly since the likely load is going to 
> be lower by 
> >> probably a factor of 1000 or more (I'm assuming that most 
> residential 
> >> users will use a cable or DSL provider for this, for example).
> >>
> >> With even a few hundred systems, you'll get almost perfect 
> smoothing 
> >> and very little burstiness.
> >>
> >> I know protocol design is fun, but I'm very worried about adding 
> >> complexity to solve largely a non-problem.
> >>
> >> Michael Hammer wrote:
> >>
> >>> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
> >>>
> >>>> In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
> >>>> jgunn6@csc.com writes:
> >>>>
> >>>>
> >>>>> So any kind of regular, scheduled test should be 
> designed so it does
> >>>>> NOT
> >>>>> tie up the operators.
> >>>>
> >>>>
> >>>>
> >>>> I believe the test must be devised so that it does not 
> involve the 
> >>>> emergency center or any of the equipment directly supporting the 
> >>>> emergency center at all.
> >>>>
> >>>> Mike
> >>>
> >>>
> >>> Then there is not much point in doing any test.  As Brian 
> points out, 
> >>> the question is how do you stage these reachability tests so that 
> >>> they scale, ie. you don't have millions of users pinging the ECC 
> >>> directly.
> >>> But, instead of the proxies pinging the ECC, which might 
> reduce the 
> >>> load, but might not make the burstiness more 
> deterministic, perhaps a 
> >>> presence subscription would be more appropriate.  The 
> proxies could 
> >>> subscribe to the presence of the ECC, and the ECC could 
> periodically 
> >>> Notify those proxies that it is still there and able to 
> receive calls.
> >>> The ECC could then control the scheduling and load 
> distribution for 
> >>> sending the Notifies.  Failing to receive the 
> notifications beyond 
> >>> some interval could provide an indication that something 
> is wrong and 
> >>> trigger an inquiry.  You might even be able to extend this to the 
> >>> users, where emergency help is a "buddy" you subscribe to on the 
> >>> local proxy.  I haven't thought this through, but thought 
> I'd throw 
> >>> that out.
> >>> Mike H.
> >>
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] updated dialog package
Date: Fri, 20 Feb 2004 13:07:41 -0800
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Thread-Topic: [Sipping] updated dialog package
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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
To: "Rohan Mahy" <rohan@cisco.com>, sipping <sipping@ietf.org>
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Hi:

From a shared line application perspective:

* Altering the dialog state draft to support sending NOTIFY before phone =
actually goes off-hook was required !

* There are a piece missing from a implementation perspective for shared =
line appearances.=20

   - Ability to signal Hold. I know there have been a few discussions in =
the SIPPING mailing list around this and it there was no conclusion =
about what "hold" actually is ! However, one of the important aspects of =
shared lines is the ability to communicate a held call to the other =
parties in the group. Currently the anil-sipping-bla draft uses the SDP =
information to help other UA's decide a dialog is transitioning to hold =
and off hold. It would be definitely useful to have the dialog state =
express this state to the far end rather than having to send out SDP =
information. For the purpose of SLA, instead of going down defining what =
is Hold, I was wondering if we can create  a state/or an attribute in =
the dialog state package say "pickable" dialog to indicate the same. =
I've heard comments saying a UA can block relevant fields of a dialog =
whenever it wants others to be able to pick or not to pick a dialog, =
however the same is not clean and it breaks when say the same UA that =
put the dialog on Hold, retrieves the dialog. Nothing in the dialog =
identifiers have changed.

The anil-sipping-bla-draft covers other various aspects of SLA from an =
application implementation perspective with call flow examples.

regards,
Venkatesh

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Rohan Mahy
Sent: Thursday, February 19, 2004 10:00 AM
To: sipping
Cc: rohan@cisco.com
Subject: [Sipping] updated dialog package


Hi Everyone,

I've updated the dialog packages with the changes we agreed to in=20
Minneapolis (removing CSeq and the Route-Set).

I've also provided a detailed example of using the dialog package for a=20
"shared-line" application.  I apologize for my tardiness here.  I was=20
supposed to post something shortly after the meeting.  I any case, I=20
believe this demonstrates that the current syntax is at least=20
sufficient for the use cases we've discussed.  I would like to hear=20
from all of you if this does the trick.

many thanks,
-rohan


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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



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Subject: Re: [Sipping] Re: sos - dead or alive
References: <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220103319.02fbf370@cia.cisco.com> <40365A1A.5030102@cisco.com>
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Clearly, this has to work with mobile terminals. However, the mobility 
indication leads me to re-think the testing approach a bit and consider 
a more modular solution:

(1) Make sure that a device has recent location information;

(2) Make sure that it can look up a DNS record for that information or, 
ask its outbound proxy to do that (MaxForwards=1). There is no 
particular reason for everybody who resolves psap.ny.ny.gov to go and 
send it a SIP request - the result is likely the same.

(3) Have a testing application (e.g., run by some government agency or 
large providers) that walks the sos.arpa DNS tree and does sanity 
checks. For example, we might want to include in the OPTIONS response 
the area that a PSAP covers. If the DNS entry and the OPTIONS response 
differ on that, there's a problem that needs to be fixed - but that's 
probably best dealt with not by grandma or Vonage, but by an agency that 
knows whom to contact to fix such issues. (Such systematic tests could 
presumably also be used to gather availability statistics and discover 
routing problems.)

Having an end-to-end test doesn't really add much value beyond the 
component tests except for testing network reachability. However, 
network reachability changes fairly quickly, so the fact that I can or 
can't reach an IP address now doesn't say much as to whether I can reach 
it a few hours from now.

Doing a true end-to-end check is helpful for NATted devices, but then 
we'd also have to do an RTP echo test. Doing a NAT test isn't really 
necessary on a daily basis, as the configuration doesn't tend to change 
that often.

Thus, I think we should either use component testing or think of a more 
comprehensive test.

I think a manual 'test' function is useful in any event, probably with 
an RTP echo. (This could be done via INVITE with caller preferences 
indication of sos.test; the remote site would then 'pick up' and say 
'Your test call to the Hickstown emergency call center has been 
successful. If this is a real emergency, please hang up and dial 911 - 
and please remember to change your smoke detector batteries.')

Henning

Paul Kyzivat wrote:

> The mobile issue is interesting. I would think that a mobile phone might 
> want to test connectivity every time it moved into a different 
> geographical area, or every time it roamed to a cell under different 
> administration. That could up the load considerably.
> 
>     Paul
> 
> Michael Hammer wrote:
> 
>> Henning,
>>
>> The reality of deployment would be that every ECC would have it's own 
>> proxy, so that would be more like 1000 x $2k = $2 million.  But, the 
>> corresponding distribution of load across those servers means it is 
>> even more of a non-problem at any given site.
>>
>> One question:  A real successful test to a wrong ECC doesn't buy you 
>> much.  I hope that mobility is addressed in the solution as well.  
>> Many of those 100 million phones are mobile, which might become 
>> SIP-based some time in the future.  so I would add a 6) "works with 
>> mobile terminals" to your list below.
>>
>> Mike
>>
>> At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
>>
>>> I think there are five goals, somewhat contradictory:
>>>
>>> (1) make it as realistic as possible, to make sure you are really 
>>> testing the emergency calling system, not the testing system; after 
>>> all, the fire safety officer doesn't go around ringing a cowbell 
>>> instead of the real fire alarms.
>>>
>>> (2) do not impose any additional operator involvement; this is solved 
>>> by the OPTIONS mechanism proposed;
>>>
>>> (3) allow for reasonable scaling;
>>>
>>> (4) avoid interfering with real emergency calls
>>>
>>> (5) allow user on-demand testing.
>>>
>>> With the proxy-based testing I proposed, we get very close to all 
>>> goals. In the worst case, i.e., every device does this once a day, we 
>>> get roughly 100 million calls a day in the US, or 1157/second. That's 
>>> about 10 proxy servers worth of equipment, i.e., about $20k. This 
>>> hardly seems worth spending inordinate engineering resources on 
>>> avoiding, particularly since the likely load is going to be lower by 
>>> probably a factor of 1000 or more (I'm assuming that most residential 
>>> users will use a cable or DSL provider for this, for example).
>>>
>>> With even a few hundred systems, you'll get almost perfect smoothing 
>>> and very little burstiness.
>>>
>>> I know protocol design is fun, but I'm very worried about adding 
>>> complexity to solve largely a non-problem.
>>>
>>> Michael Hammer wrote:
>>>
>>>> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
>>>>
>>>>> In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>>>>> jgunn6@csc.com writes:
>>>>>
>>>>>
>>>>>> So any kind of regular, scheduled test should be designed so it does
>>>>>> NOT
>>>>>> tie up the operators.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> I believe the test must be devised so that it does not involve the 
>>>>> emergency center or any of the equipment directly supporting the 
>>>>> emergency center at all.
>>>>>
>>>>> Mike
>>>>
>>>>
>>>>
>>>> Then there is not much point in doing any test.  As Brian points 
>>>> out, the question is how do you stage these reachability tests so 
>>>> that they scale, ie. you don't have millions of users pinging the 
>>>> ECC directly.
>>>> But, instead of the proxies pinging the ECC, which might reduce the 
>>>> load, but might not make the burstiness more deterministic, perhaps 
>>>> a presence subscription would be more appropriate.  The proxies 
>>>> could subscribe to the presence of the ECC, and the ECC could 
>>>> periodically Notify those proxies that it is still there and able to 
>>>> receive calls.
>>>> The ECC could then control the scheduling and load distribution for 
>>>> sending the Notifies.  Failing to receive the notifications beyond 
>>>> some interval could provide an indication that something is wrong 
>>>> and trigger an inquiry.  You might even be able to extend this to 
>>>> the users, where emergency help is a "buddy" you subscribe to on the 
>>>> local proxy.  I haven't thought this through, but thought I'd throw 
>>>> that out.
>>>> Mike H.
>>>
>>>
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>

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From: "Elwell, John" <john.elwell@siemens.com>
To: Cullen Jennings <fluffy@cisco.com>, sipping@ietf.org
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Cullen,

See in-line.

John

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com] 
> Sent: 17 February 2004 16:00
> To: sipping@ietf.org; Elwell, John
> Subject: comments on :draft-ietf-sipping-qsig2sip-04.txt
> 
> 
> 
> One major concern, Section 9.1.3 and 9.2.3 - PAI won't work 
> for this as
> described in draft-jennings-sipping-connected-00. I highly 
> recommend we talk
> about this on a phone call or in Korea and get it sorted out. 
> Suggesting a
> solution here will just make solving this problem more 
> complex. Even if you
> think that it does work, it is definitely a new semantic 
> meaning of PAI that
> is not covered in any the current documents. This would 
> require the qsig2sip
> document to be dealt with in the sip WG instead of sipping it 
> it is going to
> extend the meaning of PAI. 
[JRE]  See attached email thread, in which I received only support for the
use of PAI in a response. For the reasons given in the first email in the
thread, and also for the additional reason stated in the response from Keith
Drage, I don't believe the qsig2sip document is extending the meaning of
PAI.

> It is recognized that SIP is 
> lacking in this
> area, there have been multiple drafts proposing solutions 
> (including the
> work in draft-elwell-sip-state-update). I think it would be 
> better to get
> one of theses solutions moving and just use it here.
> 
> Few minor things:
> 
> If the GW signs a sipfrag, what should be in the SubjectAltName of the
> certificate?
[JRE] It would have to be something that identifies the gateway.

> 
> Section 9.1 (and other places) The terminology "the gw trusts 
> the next hop"
> is wrong and should be something more like "the GW is in the 
> same trust
> domain (as defined in RFC 3325) as the next hop"
[JRE] OK

> 
> Section 9.2.2 If the From is Anonymous, and there is no 
> privacy value, then
> it should be treated as if there was a privacy value of id.
[JRE] OK

> 
> NIT:
> 
> I find reference style of "as described in [10]" much harder 
> to read than "
> as described in RFC 3261 [10]"
[JRE] I don't mind. Is there a documented preference? RFCs such as 3325 and
3261 are certainly not consistent in this respect.

> 
> Cullen
> 
> 
> On 1/20/04 7:05 AM, "Internet-Drafts@ietf.org" 
> <Internet-Drafts@ietf.org>
> wrote:
> 
> > A New Internet-Draft is available from the on-line Internet-Drafts
> > directories.
> > This draft is a work item of the Session Initiation 
> Proposal Investigation
> > Working Group of the IETF.
> > 
> > Title        : Interworking between SIP and QSIG
> > Author(s)    : J. Elwell
> > Filename    : draft-ietf-sipping-qsig2sip-04.txt
> > Pages        : 61
> > Date        : 2004-1-19
> > 
> > This document specifies interworking between the Session Initiation
> >  Protocol (SIP) and QSIG within corporate telecommunication networks
> >  (also known as enterprise networks). SIP is an Internet 
> application-
> >  layer control (signalling) protocol for creating, modifying, and
> >  terminating sessions with one or more participants. These sessions
> >  include, in particular, telephone calls. QSIG is a signalling
> >  protocol for creating, modifying and terminating circuit-switched
> >  calls, in particular telephone calls, within Private Integrated
> >  Services Networks (PISNs). QSIG is specified in a number of ECMA
> >  Standards and published also as ISO/IEC standards.
> > 
> > A URL for this Internet-Draft is:
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-04.txt
> > 
> > To remove yourself from the IETF Announcement list, send a 
> message to
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> > 
> > Internet-Drafts are also available by anonymous FTP. Login 
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> > "get draft-ietf-sipping-qsig2sip-04.txt".
> > 
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > 
> > 
> > Internet-Drafts can also be obtained by e-mail.
> > 
> > Send a message to:
> > mailserv@ietf.org.
> > In the body type:
> > "FILE /internet-drafts/draft-ietf-sipping-qsig2sip-04.txt".
> > 
> > NOTE:    The mail server at ietf.org can return the document in
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> 


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Content-type: message/rfc822
Content-description: RE: [Sip] Use of P-Asserted-Identity in a response

Date: Mon, 19 Jan 2004 12:59:31 +0000
From: Mark Watson <mwatson@nortelnetworks.com>
Subject: RE: [Sip] Use of P-Asserted-Identity in a response
To: "'Elwell, John'" <john.elwell@siemens.com>, "'SIP@ietf.org'" <SIP@ietf.org>
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John, 

It was certainly the intention when 3324/3325 were written that PAI could
appear in responses. The term message was used deliberately. I am not sure
where opinions to the contrary have come from - has someone claimed that a
'response' is not a 'message' ?

...Mark 

-----Original Message----- 
From: Elwell, John [ mailto:john.elwell@siemens.com
<mailto:john.elwell@siemens.com> ] 
Sent: 02 January 2004 16:11 
To: SIP@ietf.org 
Subject: [Sip] Use of P-Asserted-Identity in a response 


There seem to be some differences of opinion concerning the use of
P-Asserted-Identity (PAI) in a response. In particular, this is delaying the
progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a 200 OK
to convey the connected party identity received from the QSIG network. The
same situation is likely to be encountered when interworking with other
legacy signalling systems.

It is my opinion that PAI can be used in a 200 OK to assert the identity of
the connect party. Moreover, it can be inserted by a gateway UAS and, unless
subject to privacy, can be passed on even to untrusted entities including
the UAC.

My evidence for assuming this is as follows: 

1. RFC 3325: In section 5, 2nd paragraph it states: 
"If the proxy receives a message (request or response) from a node that it
trusts, it can use the information in the P-Asserted-Identity header field,
if any, as if it had authenticated the user itself." There is nowhere that
forbids the use of P-Asserted-ID in a response, although there are places in
the RFC that mention procedures for a request without mentioning a response.
In many places it talks about "message", which, according to the text above,
means request or response.

2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
calling user, PAI in a 180 response the ringing user and PAI in 200 OK the
user who answered the call. RFC 3325 is supposed to meet the requirement of
RFC3324.

I have seen a number of arguments against the use of PAI in a response: 

a) It has been suggested that a PAI in a response represents the identity of
the entity that issued the request. This does not hold water. It clearly
would be in contradiction to clause 5 of RFC3324 and there is nowhere in
RFC3324 or RFC3325 that suggests echoing in a response the PAI in the
request.

b)It has been suggested that there would need to be corresponding privacy
requirements for identity information in a response. However, RFC 3323 does
indeed address privacy in responses. See last paragraph of section 3: "It
may be desirable to restrict identity information on both requests and
responses.  Initially, it might seem unusual to suggest that a response has
privacy concerns - presumably, the originator of the request knows who they
were attempting to contact, so the identity of the respondent can hardly be
confidential.  However, some personal information in responses (such as the
contact address at which the respondent is currently registered) is subject
to privacy concerns and can be addressed by these mechanisms." Also see
section 4.2, which states: "...this document defines a new SIP header,
Privacy, that can be used to specify privacy handling for requests and
responses."

c)It has been argued that using PAI in a response may cause an existing
implementation to do the wrong thing with respect to privacy, e.g., if
privacy is required by the caller, a proxy may apply this to the connected
user's identity in the response as well. I don't think this is a likely
scenario in view of RFC 3323's coverage of privacy in requests and
responses.

d)Also it has been suggested that an existing UAS may not take care to
request privacy in a response, not realising that a proxy might add a PAI.
Again, since RFC 3323 covers privacy in both requests and responses, any UA
implementation that supports privacy in requests should also support privacy
in responses.

e) It has been pointed out that a proxy cannot reject a response if the PAI
is wrong. I don't believe this matters - it should be sufficient for the
proxy just to strip out a PAI that is wrong.

In conclusion, I see evidence that PAI can be used in a response to assert
the ID of the connected party and no solid evidence to the contrary.

John Elwell (john.elwell@siemens.com) 

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Content-type: message/rfc822
Content-description: RE: [Sip] Use of P-Asserted-Identity in a response

Date: Sun, 04 Jan 2004 00:57:31 +0000
From: "Drage, Keith (Keith)" <drage@lucent.com>
Subject: RE: [Sip] Use of P-Asserted-Identity in a response
To: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
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 <475FF955A05DD411980D00508B6D5FB00AA7817B@en0033exch001u.uk.lucent.com>
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RFC 3325 section 9.1 and section 9.2 both extend table 2 of RFC 3261 with
the new header.

Both entires leave the "where" column empty.

The key to RFC 3261 table 2 states:

      An empty entry in the "where" column indicates that the header
           field may be present in all requests and responses.

I would therefore conclude from the existing specification that both headers
may appear in any response to BYE, INV, OPT, SUB, NOT, REF, i.e. those
methods that either create a dialog, or are used outside a dialog.

As MESSAGE appeared after RFC 3325 I would also assume that it is allowed in
the MESSAGE method, in both requests and all responses.

Note that these tables do not restrict it to 18x and 2xx responses, although
for ISDN mapping purposes those responses are the useful ones. It is
perfectly valid in 3xx, 4xx and 5xx responses.

Any assertion of identity in responses (and its value) is of course
dependent on the specification of privacy by the UAS and on the requirements
of the first proxy in the trust domain.

This usage corresponds to what has been specified in 3GPP TS 24.229 and I
believe in ITU-T Q.1912.5.

regards

Keith 

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: 02 January 2004 16:11
> To: SIP@ietf.org
> Subject: [Sip] Use of P-Asserted-Identity in a response
> 
> 
> There seem to be some differences of opinion concerning the use of
> P-Asserted-Identity (PAI) in a response. In particular, this 
> is delaying the
> progress of draft-ietf-sipping-qsig2sip, which makes use of 
> PAI in a 200 OK
> to convey the connected party identity received from the QSIG 
> network. The
> same situation is likely to be encountered when interworking 
> with other
> legacy signalling systems.
> 
> It is my opinion that PAI can be used in a 200 OK to assert 
> the identity of
> the connect party. Moreover, it can be inserted by a gateway 
> UAS and, unless
> subject to privacy, can be passed on even to untrusted 
> entities including
> the UAC.
> 
> My evidence for assuming this is as follows:
> 
> 1. RFC 3325: In section 5, 2nd paragraph it states:
> "If the proxy receives a message (request or response) from a 
> node that it
> trusts, it can use the information in the P-Asserted-Identity 
> header field,
> if any, as if it had authenticated the user itself."
> There is nowhere that forbids the use of P-Asserted-ID in a response,
> although there are places in the RFC that mention procedures 
> for a request
> without mentioning a response. In many places it talks about 
> "message",
> which, according to the text above, means request or response.
> 
> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
> calling user, PAI in a 180 response the ringing user and PAI 
> in 200 OK the
> user who answered the call. RFC 3325 is supposed to meet the 
> requirement of
> RFC3324.
> 
> I have seen a number of arguments against the use of PAI in a 
> response:
> 
> a) It has been suggested that a PAI in a response represents 
> the identity of
> the entity that issued the request. This does not hold water. 
> It clearly
> would be in contradiction to clause 5 of RFC3324 and there is 
> nowhere in
> RFC3324 or RFC3325 that suggests echoing in a response the PAI in the
> request.
> 
> b)It has been suggested that there would need to be 
> corresponding privacy
> requirements for identity information in a response. However, 
> RFC 3323 does
> indeed address privacy in responses. See last paragraph of section 3:
> "It may be desirable to restrict identity information on both 
> requests and
> responses.  Initially, it might seem unusual to suggest that 
> a response has
> privacy concerns - presumably, the originator of the request 
> knows who they
> were attempting to contact, so the identity of the respondent 
> can hardly be
> confidential.  However, some personal information in 
> responses (such as the
> contact address at which the respondent is currently 
> registered) is subject
> to privacy concerns and can be addressed by these mechanisms."
> Also see section 4.2, which states:
> "...this document defines a new SIP header, Privacy, that can 
> be used to
> specify privacy handling for requests and responses."
> 
> c)It has been argued that using PAI in a response may cause 
> an existing
> implementation to do the wrong thing with respect to privacy, e.g., if
> privacy is required by the caller, a proxy may apply this to 
> the connected
> user's identity in the response as well. I don't think this 
> is a likely
> scenario in view of RFC 3323's coverage of privacy in requests and
> responses.
> 
> d)Also it has been suggested that an existing UAS may not take care to
> request privacy in a response, not realising that a proxy 
> might add a PAI.
> Again, since RFC 3323 covers privacy in both requests and 
> responses, any UA
> implementation that supports privacy in requests should also 
> support privacy
> in responses.
> 
> e) It has been pointed out that a proxy cannot reject a 
> response if the PAI
> is wrong. I don't believe this matters - it should be 
> sufficient for the
> proxy just to strip out a PAI that is wrong.
> 
> In conclusion, I see evidence that PAI can be used in a 
> response to assert
> the ID of the connected party and no solid evidence to the contrary.
> 
> John Elwell (john.elwell@siemens.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
> 

--Boundary_(ID_KjWIVHTBo5sNlbvaG2Fi5Q)
Content-type: message/rfc822
Content-description: Re: [Sip] Use of P-Asserted-Identity in a response

Date: Sun, 04 Jan 2004 00:27:28 +0000
From: Tom Taylor <taylor@nortelnetworks.com>
Subject: Re: [Sip] Use of P-Asserted-Identity in a response
To: David Oran <oran@cisco.com>
Cc: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
Message-id: <3FF75DF0.7080402@nortelnetworks.com>
MIME-version: 1.0
X-Mailer: Internet Mail Service (5.5.2650.21)
Content-type: text/plain
Content-Transfer-Encoding: 7BIT

I would also support this usage.

David Oran wrote:

> These arguments are good ones.
> Even if they weren't good I'd still be in favor of what John proposes, 
> because it's eminently reasonable and provides desperately needed 
> functionality for which we currently have no alternative machinery.
> 
> Dave.
> 
> --On Friday, January 2, 2004 4:10 PM +0000 "Elwell, John" 
> <john.elwell@siemens.com> wrote:
> 
>> There seem to be some differences of opinion concerning the use of
>> P-Asserted-Identity (PAI) in a response. In particular, this is delaying
>> the progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a
>> 200 OK to convey the connected party identity received from the QSIG
>> network. The same situation is likely to be encountered when interworking
>> with other legacy signalling systems.
>>
>> It is my opinion that PAI can be used in a 200 OK to assert the identity
>> of the connect party. Moreover, it can be inserted by a gateway UAS and,
>> unless subject to privacy, can be passed on even to untrusted entities
>> including the UAC.
>>
>> My evidence for assuming this is as follows:
>>
>> 1. RFC 3325: In section 5, 2nd paragraph it states:
>> "If the proxy receives a message (request or response) from a node 
>> that it
>> trusts, it can use the information in the P-Asserted-Identity header
>> field, if any, as if it had authenticated the user itself."
>> There is nowhere that forbids the use of P-Asserted-ID in a response,
>> although there are places in the RFC that mention procedures for a 
>> request
>> without mentioning a response. In many places it talks about "message",
>> which, according to the text above, means request or response.
>>
>> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
>> calling user, PAI in a 180 response the ringing user and PAI in 200 OK 
>> the
>> user who answered the call. RFC 3325 is supposed to meet the requirement
>> of RFC3324.
>>
>> I have seen a number of arguments against the use of PAI in a response:
>>
>> a) It has been suggested that a PAI in a response represents the identity
>> of the entity that issued the request. This does not hold water. It
>> clearly would be in contradiction to clause 5 of RFC3324 and there is
>> nowhere in RFC3324 or RFC3325 that suggests echoing in a response the PAI
>> in the request.
>>
>> b)It has been suggested that there would need to be corresponding privacy
>> requirements for identity information in a response. However, RFC 3323
>> does indeed address privacy in responses. See last paragraph of 
>> section 3:
>> "It may be desirable to restrict identity information on both requests 
>> and
>> responses.  Initially, it might seem unusual to suggest that a response
>> has privacy concerns - presumably, the originator of the request knows
>> who they were attempting to contact, so the identity of the respondent
>> can hardly be confidential.  However, some personal information in
>> responses (such as the contact address at which the respondent is
>> currently registered) is subject to privacy concerns and can be addressed
>> by these mechanisms."
>> Also see section 4.2, which states:
>> "...this document defines a new SIP header, Privacy, that can be used to
>> specify privacy handling for requests and responses."
>>
>> c)It has been argued that using PAI in a response may cause an existing
>> implementation to do the wrong thing with respect to privacy, e.g., if
>> privacy is required by the caller, a proxy may apply this to the 
>> connected
>> user's identity in the response as well. I don't think this is a likely
>> scenario in view of RFC 3323's coverage of privacy in requests and
>> responses.
>>
>> d)Also it has been suggested that an existing UAS may not take care to
>> request privacy in a response, not realising that a proxy might add a 
>> PAI.
>> Again, since RFC 3323 covers privacy in both requests and responses, any
>> UA implementation that supports privacy in requests should also support
>> privacy in responses.
>>
>> e) It has been pointed out that a proxy cannot reject a response if the
>> PAI is wrong. I don't believe this matters - it should be sufficient for
>> the proxy just to strip out a PAI that is wrong.
>>
>> In conclusion, I see evidence that PAI can be used in a response to 
>> assert
>> the ID of the connected party and no solid evidence to the contrary.
>>
>> John Elwell (john.elwell@siemens.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
> 
> 
> 
> 
> 
> 
> _______________________________________________
> 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
> 

--Boundary_(ID_KjWIVHTBo5sNlbvaG2Fi5Q)
Content-type: message/rfc822
Content-description: Re: [Sip] Use of P-Asserted-Identity in a response

Date: Sat, 03 Jan 2004 19:13:24 +0000
From: David Oran <oran@cisco.com>
Subject: Re: [Sip] Use of P-Asserted-Identity in a response
To: "Elwell, John" <john.elwell@siemens.com>, SIP@ietf.org
Message-id: <2147483647.1073139204@[10.32.245.157]>
MIME-version: 1.0
X-Mailer: Internet Mail Service (5.5.2650.21)
Content-type: text/plain
Content-Transfer-Encoding: 7BIT

These arguments are good ones.
Even if they weren't good I'd still be in favor of what John proposes, 
because it's eminently reasonable and provides desperately needed 
functionality for which we currently have no alternative machinery.

Dave.

--On Friday, January 2, 2004 4:10 PM +0000 "Elwell, John" 
<john.elwell@siemens.com> wrote:

> There seem to be some differences of opinion concerning the use of
> P-Asserted-Identity (PAI) in a response. In particular, this is delaying
> the progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a
> 200 OK to convey the connected party identity received from the QSIG
> network. The same situation is likely to be encountered when interworking
> with other legacy signalling systems.
>
> It is my opinion that PAI can be used in a 200 OK to assert the identity
> of the connect party. Moreover, it can be inserted by a gateway UAS and,
> unless subject to privacy, can be passed on even to untrusted entities
> including the UAC.
>
> My evidence for assuming this is as follows:
>
> 1. RFC 3325: In section 5, 2nd paragraph it states:
> "If the proxy receives a message (request or response) from a node that it
> trusts, it can use the information in the P-Asserted-Identity header
> field, if any, as if it had authenticated the user itself."
> There is nowhere that forbids the use of P-Asserted-ID in a response,
> although there are places in the RFC that mention procedures for a request
> without mentioning a response. In many places it talks about "message",
> which, according to the text above, means request or response.
>
> 2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
> calling user, PAI in a 180 response the ringing user and PAI in 200 OK the
> user who answered the call. RFC 3325 is supposed to meet the requirement
> of RFC3324.
>
> I have seen a number of arguments against the use of PAI in a response:
>
> a) It has been suggested that a PAI in a response represents the identity
> of the entity that issued the request. This does not hold water. It
> clearly would be in contradiction to clause 5 of RFC3324 and there is
> nowhere in RFC3324 or RFC3325 that suggests echoing in a response the PAI
> in the request.
>
> b)It has been suggested that there would need to be corresponding privacy
> requirements for identity information in a response. However, RFC 3323
> does indeed address privacy in responses. See last paragraph of section 3:
> "It may be desirable to restrict identity information on both requests and
> responses.  Initially, it might seem unusual to suggest that a response
> has privacy concerns - presumably, the originator of the request knows
> who they were attempting to contact, so the identity of the respondent
> can hardly be confidential.  However, some personal information in
> responses (such as the contact address at which the respondent is
> currently registered) is subject to privacy concerns and can be addressed
> by these mechanisms."
> Also see section 4.2, which states:
> "...this document defines a new SIP header, Privacy, that can be used to
> specify privacy handling for requests and responses."
>
> c)It has been argued that using PAI in a response may cause an existing
> implementation to do the wrong thing with respect to privacy, e.g., if
> privacy is required by the caller, a proxy may apply this to the connected
> user's identity in the response as well. I don't think this is a likely
> scenario in view of RFC 3323's coverage of privacy in requests and
> responses.
>
> d)Also it has been suggested that an existing UAS may not take care to
> request privacy in a response, not realising that a proxy might add a PAI.
> Again, since RFC 3323 covers privacy in both requests and responses, any
> UA implementation that supports privacy in requests should also support
> privacy in responses.
>
> e) It has been pointed out that a proxy cannot reject a response if the
> PAI is wrong. I don't believe this matters - it should be sufficient for
> the proxy just to strip out a PAI that is wrong.
>
> In conclusion, I see evidence that PAI can be used in a response to assert
> the ID of the connected party and no solid evidence to the contrary.
>
> John Elwell (john.elwell@siemens.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




--Boundary_(ID_KjWIVHTBo5sNlbvaG2Fi5Q)
Content-type: message/rfc822
Content-description: Use of P-Asserted-Identity in a response

Date: Fri, 02 Jan 2004 16:11:21 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: Use of P-Asserted-Identity in a response
To: "'SIP@ietf.org'" <SIP@ietf.org>
Message-id: <6B2AC5A8F90C654980890BEFA88DCD53E70CEF@ntht201e>
MIME-version: 1.0
X-Mailer: Internet Mail Service (5.5.2650.21)
Content-type: text/plain
Content-Transfer-Encoding: 7BIT

There seem to be some differences of opinion concerning the use of
P-Asserted-Identity (PAI) in a response. In particular, this is delaying the
progress of draft-ietf-sipping-qsig2sip, which makes use of PAI in a 200 OK
to convey the connected party identity received from the QSIG network. The
same situation is likely to be encountered when interworking with other
legacy signalling systems.

It is my opinion that PAI can be used in a 200 OK to assert the identity of
the connect party. Moreover, it can be inserted by a gateway UAS and, unless
subject to privacy, can be passed on even to untrusted entities including
the UAC.

My evidence for assuming this is as follows:

1. RFC 3325: In section 5, 2nd paragraph it states:
"If the proxy receives a message (request or response) from a node that it
trusts, it can use the information in the P-Asserted-Identity header field,
if any, as if it had authenticated the user itself."
There is nowhere that forbids the use of P-Asserted-ID in a response,
although there are places in the RFC that mention procedures for a request
without mentioning a response. In many places it talks about "message",
which, according to the text above, means request or response.

2. RFC 3324 clause 5 clearly states that PAI in INVITE identifies the
calling user, PAI in a 180 response the ringing user and PAI in 200 OK the
user who answered the call. RFC 3325 is supposed to meet the requirement of
RFC3324.

I have seen a number of arguments against the use of PAI in a response:

a) It has been suggested that a PAI in a response represents the identity of
the entity that issued the request. This does not hold water. It clearly
would be in contradiction to clause 5 of RFC3324 and there is nowhere in
RFC3324 or RFC3325 that suggests echoing in a response the PAI in the
request.

b)It has been suggested that there would need to be corresponding privacy
requirements for identity information in a response. However, RFC 3323 does
indeed address privacy in responses. See last paragraph of section 3:
"It may be desirable to restrict identity information on both requests and
responses.  Initially, it might seem unusual to suggest that a response has
privacy concerns - presumably, the originator of the request knows who they
were attempting to contact, so the identity of the respondent can hardly be
confidential.  However, some personal information in responses (such as the
contact address at which the respondent is currently registered) is subject
to privacy concerns and can be addressed by these mechanisms."
Also see section 4.2, which states:
"...this document defines a new SIP header, Privacy, that can be used to
specify privacy handling for requests and responses."

c)It has been argued that using PAI in a response may cause an existing
implementation to do the wrong thing with respect to privacy, e.g., if
privacy is required by the caller, a proxy may apply this to the connected
user's identity in the response as well. I don't think this is a likely
scenario in view of RFC 3323's coverage of privacy in requests and
responses.

d)Also it has been suggested that an existing UAS may not take care to
request privacy in a response, not realising that a proxy might add a PAI.
Again, since RFC 3323 covers privacy in both requests and responses, any UA
implementation that supports privacy in requests should also support privacy
in responses.

e) It has been pointed out that a proxy cannot reject a response if the PAI
is wrong. I don't believe this matters - it should be sufficient for the
proxy just to strip out a PAI that is wrong.

In conclusion, I see evidence that PAI can be used in a response to assert
the ID of the connected party and no solid evidence to the contrary.

John Elwell (john.elwell@siemens.com)

--Boundary_(ID_KjWIVHTBo5sNlbvaG2Fi5Q)--

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



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User-Agent: Microsoft-Entourage/10.1.4.030702.0
Date: Sun, 22 Feb 2004 12:39:57 -0800
Subject: Re: [Sipping] SIP Proxy servers -- security encryption protocol
From: Cullen Jennings <fluffy@cisco.com>
To: <ranjit.avasarala@wipro.com>, <sscoggins@telogy.com>, <sipping@ietf.org>
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And the crypto that is typically being recommended and folks are
implementing is RSA for asymmetric key, SHA1 for hash, and 3DES or AES for
encryption. Some of the early stuff said 3DES because the RFC would have
blocked on AES work. It=B9s a long debate with no clear answer between AES an=
d
3DES (and let=B9s not repeat it on this list) but most SIP implementers seem
to be doing AES or both AES and 3DES.

On 2/19/04 9:26 PM, "ranjit.avasarala@wipro.com"
<ranjit.avasarala@wipro.com> wrote:

> Hi
>     For secured transactions you could use TLS i.e using sips: instead of
> normal sip: url.
> Alternatively IPSec also could be used at IP Layer
> and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt) prot=
ocol.
> =20
> Regards
> Ranjit
>> =20
>> =20
>> -----Original Message-----
>> From:  sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of
>> Scoggins, Sophia
>> Sent: Friday, February 20, 2004 10:45  AM
>> To: 'sipping@ietf.org'
>> Subject: [Sipping] SIP Proxy  servers -- security encryption protocol
>>=20
>> =20
>> =20
>>=20
>> =20
>> =20
>> My group is looking for some  security encryption protocol for both SIP =
and
>> RTP streams for CPE. We were  wondering what encryption protocol that mo=
st
>> SIP proxy servers currently  support or intend to support? Would you ple=
ase
>> just send your reply to  me?
>> =20
>> =20
>> =20
>> Thanks.
>> =20
>> =20
>> =20
>> Regards,
>> =20
>> Sophia2@ti.com
>=20



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</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
And the crypto that is typically being recommended and folks are implementi=
ng is RSA for asymmetric key, SHA1 for hash, and 3DES or AES for encryption.=
 Some of the early stuff said 3DES because the RFC would have blocked on AES=
 work. It&#8217;s a long debate with no clear answer between AES and 3DES (a=
nd let&#8217;s not repeat it on this list) but most SIP implementers seem to=
 be doing AES or both AES and 3DES. &nbsp;<BR>
<BR>
On 2/19/04 9:26 PM, &quot;ranjit.avasarala@wipro.com&quot; &lt;ranjit.avasa=
rala@wipro.com&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT COLOR=3D"#0000=
FF"><FONT FACE=3D"Arial">Hi<BR>
&nbsp;&nbsp;&nbsp;&nbsp;For secured transactions you could use TLS i.e usin=
g sips: instead of normal sip: url.<BR>
Alternatively IPSec also could be used at IP Layer<BR>
and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt) protoc=
ol.<BR>
</FONT></FONT><FONT FACE=3D"Verdana"> <BR>
</FONT><FONT COLOR=3D"#0000FF"><FONT FACE=3D"Arial">Regards<BR>
Ranjit<BR>
</FONT></FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D=
"Verdana"> <BR>
&nbsp;<BR>
</FONT><FONT FACE=3D"Tahoma">-----Original Message-----<BR>
<B>From:</B> &nbsp;sipping-admin@ietf.org [<a href=3D"mailto:sipping-admin@ie=
tf.org]">mailto:sipping-admin@ietf.org]</a> <B>On Behalf Of &nbsp;</B>Scoggi=
ns, Sophia<BR>
<B>Sent:</B> Friday, February 20, 2004 10:45 &nbsp;AM<BR>
<B>To:</B> 'sipping@ietf.org'<BR>
<B>Subject:</B> [Sipping] SIP Proxy &nbsp;servers -- security encryption pr=
otocol<BR>
<BR>
</FONT><FONT FACE=3D"Verdana"> <BR>
&nbsp;<BR>
<BR>
&nbsp;<BR>
&nbsp;<BR>
My group is looking for some &nbsp;security encryption protocol for both SI=
P and RTP streams for CPE. We were &nbsp;wondering what encryption protocol =
that most SIP proxy servers currently &nbsp;support or intend to support? Wo=
uld you please just send your reply to &nbsp;me?<BR>
&nbsp;<BR>
&nbsp;<BR>
&nbsp;<BR>
Thanks.<BR>
&nbsp;<BR>
&nbsp;<BR>
&nbsp;<BR>
Regards,<BR>
&nbsp;<BR>
Sophia2@ti.com<BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verda=
na"><BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:14.0px'><FONT FACE=3D"Verda=
na"><BR>
</FONT></SPAN>
</BODY>
</HTML>


--B_3160331029_775979--


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Sun, 22 Feb 2004 22:24:21 -0800
Subject: Re: [Sipping] RE: comments on :draft-ietf-sipping-qsig2sip-04.txt
From: Cullen Jennings <fluffy@cisco.com>
To: "Elwell, John" <john.elwell@siemens.com>, <sipping@ietf.org>
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I think the attached email thread more or less makes exactly my point. I am
not at all arguing that 3325 forbids PAI in response.  That was never the
intention - but 3325 certainly does not say what it means in a response or
how to make it work to something useful or what the useful thing you can do
is. It does not say if PAI in a response is the identity of the UAS or the
UAC. It does not deal with the issue of how you authenticate a response.
There is a bunch of other things it does not deal with.  The pure amount of
language lawyer conjecture in the email thread makes it clear that 3325 does
not define what it means.

If you want to clearly define what PAI in a response means in some document
(qsig or another one), I've got no problem with that - I want to solve this
problem - but 3325 does not define what it means or how to make it work to
accomplish some useful goal.

Cullen



On 2/22/04 10:17 AM, "Elwell, John" <john.elwell@siemens.com> wrote:

>> One major concern, Section 9.1.3 and 9.2.3 - PAI won't work
>> for this as
>> described in draft-jennings-sipping-connected-00. I highly
>> recommend we talk
>> about this on a phone call or in Korea and get it sorted out.
>> Suggesting a
>> solution here will just make solving this problem more
>> complex. Even if you
>> think that it does work, it is definitely a new semantic
>> meaning of PAI that
>> is not covered in any the current documents. This would
>> require the qsig2sip
>> document to be dealt with in the sip WG instead of sipping it
>> it is going to
>> extend the meaning of PAI.
> [JRE]  See attached email thread, in which I received only support for the
> use of PAI in a response. For the reasons given in the first email in the
> thread, and also for the additional reason stated in the response from Keith
> Drage, I don't believe the qsig2sip document is extending the meaning of
> PAI.


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



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Thread-Topic: [Simple] Advanced IM requirements
Thread-Index: AcP5C/kBvwKHvMXJS6urOHqe9sfskgA3pGvQ
From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Simple WG" <simple@ietf.org>
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Hi Jonathan et al,

	I have an extremely early version (so be gentle) of a draft
aimed at supplying IM delivery reports.  It hasn't been formally
submitted yet but I have attached a copy - so would be grateful if the
list could take a look and see if it's worth progressing.  I will submit
to the archives in the next few days.  I will also put this on the
internet somewhere in case this attachment doesn't survive its journey
(Will send link later).

Best Regards,

Chris.
=20

>-----Original Message-----
>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>Sent: 22 February 2004 06:17
>To: Simple WG
>Subject: [Simple] Advanced IM requirements
>
>Folks,
>
>As I noted earlier, I resubmitted the advanced IM requirements draft.
It
>specifies a bunch of advanced IM functions we havent yet tackled here.
>They are:
>
>* delivery reports for IM
>* is-typing indicator (Henning had an I-D on this that has expired)
>* capabilities indications of maximum desired message sizes for IM
>* page mode groups (under discussion in the context of exploders in
>sipping)
>* invitations to non-real time sessions
>
>
>The last of these is particularly interesting. One use case is "call
>alerts" in PTT. There, I send you a request to have a PTT chat. The
>request is not answered in real time. Rather, its stored by the
>recipient as if it were an IM, and can be answered at any time later.
>"Answering it" merely involves setting up a PTT call back to the
caller.
>Its also possible to cancel the call alert at any time.
>
>There are some interesting ways to accomplish this function. The one
>I've been thinking about is taking a SIP INVITE message, and sending it
>in the body of a MESSAGE request as a message/sip content. I'm sure
>there are other ways.
>
>Are these topics for which there is still group interest in working on?
>I think all of them remain quite important.
>
>Thanks,
>Jonathan R.
>--
>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>Chief Technology Officer                    Parsippany, NJ 07054-2711
>dynamicsoft
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com
>
>
>_______________________________________________
>Simple mailing list
>Simple@ietf.org
>https://www1.ietf.org/mailman/listinfo/simple


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From: "Harry Behrens" <harry.behrens@snom.de>
To: <sipping@ietf.org>
Cc: <knuettel@fokus.fraunhofer.de>
Date: Mon, 23 Feb 2004 12:35:42 +0100
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Subject: [Sipping] Roaming Architecture for SIP
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We have submitted a draft to the IETF
(draft-behrens-sipping-roaming-architecture-00.txt),
in which we describe a mechanism to enable VoIP roaming based on a
unique ID - the NAI - which constitutes
a handle into the billing relationship for a user.
The original draft has some mistakes in the discussion of the call flows
as well as not being 100% correctly formatted.
We will therefore submit an updated version after the cutoff date, which
we attach in this mail.
Comments etc. are very welcome.

Best regards,

	Harry Behrens
	Karsten Knuettel

Abstract 
 
This draft describes an authentication mechanism which can be used to 
enable users of VoIP to globally roam between any number of ITSPs 
(Internet Telephony Service Providers) without needing to have a prior 
customer or billing relationship with all of them. This enables 
profiles and one-line-billing. 
Providers using this authentication mechanism do neither need full-mesh 
trust relationships or roaming agreements with every other possible 
provider nor do they need to rely on a centralized brokerage entity to 
process calls. 
The Home Network (HN), which handles all AAA for a user attempting to 
use an ITSP's service is determined through a unique identifier 
submitted by the user. This Network Access Identifier [RFC2486] 
contains information about the trust domain or trust realm the user 
belongs to. Based on a simple discovery mechanism a provider can 
establish a trust path to this home network. Once this is established, 
the provider can authenticate the user and check his credit with the 
home network. Accounting and rate information will be sent to the home 
network for mediation and processing.

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Internet Engineering Task Force				     H. Behrens=20
Internet Draft                                       snom technology AG=20
                                                            K. Knuettel=20
                                                       Fraunhofer FOKUS=20
=20
 =20
draft-behrens-sipping-roaming-architecture-01.txt=20
February 16, 2003=20
Expires: August, 2004=20
=20
 Interdomain Trust-Relationships for SIP-based Roaming=20
=20
=20
STATUS OF THIS MEMO=20
 =20
This document is an Internet-Draft and is subject to all provisions of=20
Section 10 of RFC2026.=20
 =20
Internet-Drafts are working documents of the Internet Engineering Task=20
Force (IETF), its areas, and its working groups. Note that other groups=20
may also distribute working documents as Internet- Drafts.=20
=20
Internet-Drafts are draft documents valid for a maximum of six months=20
and may be updated, replaced, or obsoleted by other documents at any=20
time. It is inappropriate to use Internet-Drafts as reference material=20
or to cite them other than as "work in progress".=20
 =20
The list of current Internet-Drafts can be accessed at=20
http://www.ietf.org/ietf/1id-abstracts.txt=20
 =20
To view the list Internet-Draft Shadow Directories, see=20
http://www.ietf.org/shadow.html.=20
=20
=20
Abstract=20
=20
This draft describes an authentication mechanism which can be used to=20
enable users of VoIP to globally roam between any number of ITSPs=20
(Internet Telephony Service Providers) without needing to have a prior=20
customer or billing relationship with all of them. This enables=20
profiles and one-line-billing.=20
Providers using this authentication mechanism do neither need full-mesh=20
trust relationships or roaming agreements with every other possible=20
provider nor do they need to rely on a centralized brokerage entity to=20
process calls.=20
The Home Network (HN), which handles all AAA for a user attempting to=20
use an ITSP's service is determined through a unique identifier=20
submitted by the user. This Network Access Identifier [RFC2486]=20
contains information about the trust domain or trust realm the user=20
belongs to. Based on a simple discovery mechanism a provider can=20
establish a trust path to this home network. Once this is established,=20
the provider can authenticate the user and check his credit with the=20
home network. Accounting and rate information will be sent to the home=20
network for mediation and processing.=20
 =20
H. Behrens, K. Knuettel                                  [Page 1]=0C
=20
1. Introduction..............................................2=20
2. Requirements..............................................2=20
3. Architecture..............................................3=20
4. Terminology...............................................3=20
5. Conventions used in this document.........................3=20
6. Security Architecture.....................................4=20
6.1 Trust Relationships......................................4


Internet Draft SIP February 8, 2003=20


6.2 Trust Relationships between Providers and Settlement.....4=20
7. Actors and Mechanisms.....................................5=20
7.1 Actors...................................................5=20
7.2 Mechanisms...............................................5=20
8. Call Flow.................................................6=20
10. Bibliography............................................10=20
=20
1. Introduction=20
=20
More and more ISPs are beginning to add VoIP to their service=20
portfolio. The last years have seen the merging of IP-based telephony=20
and switched line telephony (both mobile and fixed-line) in the sense=20
that users of VoIP want to be able to call a PSTN or PLMN number as=20
well as being reachable under a home PSTN or PLMN number. Roaming=20
between ISPs has not developed to a large degree.=20
With the advent of voice communication becoming an important service,=20
this will become more relevant. Users will want to use VoIP for their=20
telephony and they will want to do this wherever they are located. The=20
authors believe that one way to address this is to leverage a user's=20
existing customer and billing relationship with his home telephony=20
provider. Users make use of this when signing up for ad-hoc service=20
with a visited ITSP. The ITSP verifies the existing billing=20
relationship and receives authorization of a certain amount of credit=20
the home network is willing to underwrite.=20
Based on the security architecture described here, mutual trust can be=20
established and the user can both place VoIP calls through his visited=20
network as well as being reachable under home number and URI.=20
=20
2. Requirements=20
=20
The term 'Roaming Capability' has been defined in [RFC2486]. =20
One aim of this document is to provide an architecture by which Roaming=20
Capability can be assured by a federation of independent VoIP=20
providers. In the world of IP-related protocols little attention has=20
been given to this issue.=20
Requirements for global roaming for WLAN users have been described by=20
J. Caron in [CARON.1]. He also outlined a simple DNS-based method in=20
[CARON.2].=20
The Roaming Scenario addressed here is as follows:=20
 -  The principal with the NAI of alice@biloxi.com sends a REGISTER=20
    request to the SIP registrar of atlanta.com. atlanta.com in this=20
    case is Alice's 'Visited Network' (VN). biloxi.com is her home=20
    network.=20
 - Alice wants to use her VN to place authenticated and    secure=20
   VoIP calls.=20
 - Alice wants to be reachable both under her home URI as well as=20
   under a URI assigned by the VN.=20
 - Alice provides her NAI to her VN's SIP registrar.=20
 - The VN=20
      o asserts that Alice's Home Network signs Alice's identity and=20
        authorizes a credit amount greater 0=20
      o asserts that Alice's HN is a trusted entity=20
 - atlanta.com assigns Alice a temporary Address of Record (AOR) URI=20
   as well as a valid contact address from within its own name space.=20
 - Alice registers with her HN using the newly assigned Contact=20
   address as the Contact.=20
 - Alice's SUA can place outgoing call either under the temporary URI=20
   assigned by the VN or her home URI at biloxi.com.=20
 - Alice is reachable both under her home-AOR as well as the visiting-
   AOR. Assuming Alice's HN provides PSTN break-in, Alice is also=20
   reachable under her telephone number known to her buddies.=20
 - All calls placed by Alice for the duration of this registration are=20
   accounted to Alice's HN.=20

H. Behrens, K. Knuettel                                  [Page 2]=0C


Internet Draft SIP February 8, 2003=20


 =20
3. Architecture=20
 =20
The architecture is based on a model that is user-centric in that it=20
assumes for each principal or user to have one Home Network with which=20
he has a valid billing relationship. =20
This entity can authenticate the user based on his NAI and is willing=20
to underwrite credit as well as receive CDRs for further settlement.=20
This HN corresponds to the 'Authentication Service' described in=20
[PETERSON.1]. Basing the architecture on a user-centric model based on=20
a NAI, greatly reduces the number of billing relationships in a roaming=20
environment as well allowing incumbent HNs, such PSTN or PLMN providers=20
[RFC2486], to leverage their existing customer relationships. =20
New emerging operators can benefit from this, because they can deliver=20
their services to any user with a valid NAI, knowing that they can bill=20
the user's HN up to the credit amount underwritten. =20
HNs can benefit by charging for this service. This potentially creates=20
a win-win situation between incumbent operators, who hold the trust and=20
billing relationships, and emerging ITSPs who wish to be able to=20
attract as many users as possible, without the financial and=20
bureaucratic overhead involved in setting up a documented and secure=20
billing relation with each of their users.=20
=20
The architecture consists of=20
 1. Peers: entities participating in a roamable VoIP infrastructure=20
 2. Security Architecture: the Security architecture, which describes=20
   the trust relationships as well as security mechanisms used.=20
 3. Signalling Protocol: Signalling is based on the Session Initiation=20
   Protocol (SIP) [RFC3261]. =20
 4. Accounting infrastructure: Both Authentication and Authorization=20
   are covered by the Security Architecture and/or the Signalling=20
   Protocol. Accounting is based on=20
       a. the Authentication and Authorization signed by the HN and=20
          accepted by the VN.=20
       b. hub-and-spoke trust relationships from the participating=20
          operators to a Settlement Entity (SE).=20
This document proposes an architecture, which fulfils these=20
requirements by making use of pre-existing trust relationships.=20
The architecture is strongly influenced by work done by J. Peterson=20
[PETERSON.1] and [RFC3325] on one hand and the work of Aboba et. al=20
[RFC2486],[RFC2194] on the other hand.=20
=20
4. Terminology=20
=20
AAA    Authentication, Authorization, Accounting=20
AOR    Address of Record=20
CDR    Call Detail Record=20
CRM    Customer Relationship Management=20
DoT    Domain of Trust=20
ETSI   European Telecommunication Standard Institute=20
HN    Home Network=20
ITSP   Internet Telephony Service Provider=20
NAI   Network Access Identifier=20
OSP    Open Settlement Protocol=20
PKI    Public Key Infrastructure=20
PLMN   Public Line Mobile Network=20
PSTN   Public Switched Telephone Network=20
URI    Uniform Resource Identifier=20
VN    Visited Network=20
VoIP   Voice over IP=20
=20
5. Conventions used in this document=20
=20

H. Behrens, K. Knuettel                                  [Page 3]=0C


Internet Draft SIP February 8, 2003=20


The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",=20
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this=20
document are to be interpreted as described in RFC-2119 [RFC2119].=20
 =20
6. Security Architecture=20
=20
The proposed architecture is based on the assumption that all entities,=20
participating in the execution of a call, establish a mutual Trust=20
Domain (TD) for at least the duration of that call. The term 'Trust=20
Domain' is used in the sense described in [RFC3324].=20
Such a TD can be of permanent nature or established within the context=20
of one or more transactions.=20
Based on this definition of Trust Domain the architecture described=20
here, can be separated into two distinct layers:=20
 1. Verification or Establishment of a common Trust Domain based on=20
   existing and derived direct Trust Relationships.=20
 2. Use the mechanisms provided by this TD  to assert the=20
   validity of VoIP calls and their related document trail.=20
Section 6.1 and 6.2 will outline the direct Trust Relationships assumed=20
in the scope of this document. Section 7 outlines how a common Trust=20
Domain is established. Section 8 provides example call flows that=20
combine the mechanisms to demonstrate how the requirements are=20
fulfilled.=20
 =20
6.1 Trust Relationships=20
For SIP-based VoIP to become commercially accepted, providers need to=20
be able to=20
 -  account and bill for their services=20
 -  be able to cooperate with other VoIP providers in routing and=20
    signalling calls.=20
Both of these requirements assume trust relationships between the=20
user(s) being served as well as between the operators cooperating=20
during the call.=20
In order to achieve critical mass, it needs to be easy for providers to=20
cooperate in a flexible ad-hoc fashion in placing and routing their=20
calls. Authentication should be based on a globally unique identifier,=20
so one principal has one AAA and billing relationship. This unique=20
identifier needs to specify his home domain of trust to other entities.=20
Within his DoT, the identifier needs to identify his specific principal=20
identity.=20
Authentication and authorization of a user can be transparent because=20
the user uses only one set of credentials wherever he may be. Within=20
this document we assume credentials based on exchange of PKI=20
certificates.=20
The model operates on direct trust relationships. A user has an a-
priori trust relationship with his home network.=20
Visited network and home network establish trust based on exchange and=20
verification of certificates.=20
The user and the visited network establish trust through the mechanisms=20
described in this document.=20
Once the user and the VN have established a trust relationship, which=20
is based on the VN knowing that the HN can be charged for the credit=20
amount authorized, the user is fully authenticated and authorized and=20
can use the VN's services.=20
This is even though a-priori trust or billing relationship exists=20
between the authenticated user and the serving provider.=20
 =20
6.2 Trust Relationships between Providers and Settlement=20
 Entity=20
 =20
Inter-operator accounting is still within the scope of the architecture=20
described here. We assume that billing is based on a settlement and=20

H. Behrens, K. Knuettel                                  [Page 4]=0C


Internet Draft SIP February 8, 2003=20


clearing entity. This entity provides CDR-aggregation, settlement,=20
clearing and billing functionalities to the participating providers.=20
Where there is more than one such entity, we assume an existing trust=20
relationship between them. This architecture assumes the existence of=20
such an entity, e.g. based on the Open Settlement Protocol (OSP)=20
defined by the ETSI [TS102.321]. It is also assumed that each provider=20
has its own trust relationship with the Settlement Agency. It should be=20
noted that the model explicitly does not use any of the authentication=20
or routing functionalities proposed in that specification.=20
In the architecture presented here, OSP is strictly used for=20
aggregation of CDRs and settlement among operators.=20
The authors think that the fully integrated position taken by the ETSI=20
does not fulfil the more organic and fluid environment of the upcoming=20
VoIP industry.=20
=20
7. Actors and Mechanisms=20
=20
7.1 Actors=20
=20
The actors (players) are as follows:=20
 =20
  -  The Service User (User): A user wanting to use VoIP telephony=20
      both for inbound as well as outbound calls. A user is uniquely=20
      identified based on his NAI, which identifies him as a user of=20
      his home domain. A mechanism of the location of the AAA server=20
      serving that home domain is assumed.=20
  -  The Visited Network (VN): the VN is the network a user uses when=20
      roaming. He wants to use that networks services while being=20
      billed by his HN. Roaming should be transparent. This means that=20
      the user should not need an a-priori trust relationship with the=20
      VN. It also means that he should be reachable under his home AOR=20
      and home telephone number no matter who is roaming provider is.=20
  -  The Home Network (HN): The Home Network has two functions:=20
       o act as a home AAA server for the User.=20
       o act as the home VoIP network for the User.=20
=20
7.2 Mechanisms=20
=20
Two mechanisms to establish a mutual Trust Domain are proposed. =20
  1. One mechanism relies on transparent pass-through of registrations =
to=20
     achive registrations both with the Visited and with the Home =
Network.=20
  2. The other mechanism uses two independent registrations. One for=20
     the VN one for the HN.=20
Whichever one is chosen, at the end of either one =20
  -  The User has registered with the VN.=20
  -  The VN has established that he can bill services delivered to=20
      the User to the User's HN.=20
  -  The User has signed off that VN may bill to the HN=20
=20
7.2.1 Establishing trust using pass-through registrations=20

  -  A user registers with the VN (atlanta.com in the call flow=20
      given). The user submits his NAI in new header field 'P-NAI'.=20
      The user submits a bogus URI in both the From and the To header.=20
      (F1 in section 8.2.1)
  -  The VN recognizes the registration as a P-NAI registration and=20
      sends back a "401 Unauthorized" response, with both the 'From'=20
      and the 'To' URI set to the temporary URI that will be assigned=20
      to the user upon successful authentication and authorization (F2). =


H. Behrens, K. Knuettel                                  [Page 5]=0C

Internet Draft SIP February 8, 2003=20

  -  The User submits the credentials belonging to the NAI specified,=20
      e.g. his signed public key in the next register message (F3).=20
  -  The VN sends a registration to the HN on behalf of the user. The VN
      sets the 'From' and 'To' headers to "sip:NAI", i.e.=20
      "sip:alice@biloxi.com", where the NAI is "alice@biloxi.com" (F5).=20
  -  The HN sends a "407 Proxy Authentication Required" back to the VN.
  -  The VN forwards the response to the user (F7).
  -  The User submits his credentials for the identity represented by =
his
      NAI in a new registration message (F8).
  -  The VN forwards this to the HN. The HN authenticates the user and =
includes
      a 'P-NAI-Token' encrypted using the user's public key in the =
response to=20
      the user's registration request.=20
  -  The VN forwards this response to the user. The user decrypts the =
'P-NAI-Token'=20
     with his private key. The user is now registered with the HN and =
the HN has=20
     the authorized user to use any VN's services based on the encrypted =

     P-NAI-Token (F11).

  -  The user encrypts the P-NAI-Token with the public key of the VN and =
sends
     a registration message to register his contact with the 'From' and =
'To'
     URIs received in F2. Both 'P-NAI' and 'P-NAI-Token' are set in this =

     request (F12).=20
  -  The VN decrypts the token and sends a "200 OK" (F13).
  -  The user is now registered with the VN as well as the HN and
      all necessary trust relationships have been established.=20
       o The HN has authenticated the user.
       o The VN now trusts the user based on the VN's trusting   =20
          the HN.=20
       o The user trusts the VN. This is established by the VN=20
          possessing the clear text P-NAI-Token.=20
       o The token can be used in accounting, billing and settlement
	  to establish the link to this ad-hoc Trust Domain.

7.2.2 Establishing trust using two indpendent registrations=20
=20
The mechanism is as follows:=20
  -  A user registers with the VN (atlanta.com in the call flow=20
      given). The user submits his NAI in new header field 'P-NAI'.=20
      The user submits a bogus URI in both the From and the To header
      (F1 in section 8.2.2).=20
  -  The VN recognizes the registration as a P-NAI registration and=20
      sends back a "401 Unauthorized" response, with both the 'From'=20
      and the 'To' URI set to the temporary URI that will be assigned=20
      to the user upon successful authentication and authorization (F2). =

  -  The User submits the credentials belonging to the NAI specified,=20
      e.g. his signed public key in the next register message (F3).=20
  -  The VN requests authentication and authorization of the user=20
      from the home network. This can either be done through the AAA=20
      architecture or future extensions to the SIP protocol (F5).=20
  -  The HN authenticates the user and authorizes a certain amount of=20
      credit. The HN includes a one-time token, which he encrypts=20
      using the user's public key in the return message. The HN signs=20
      the user's credentials and the encrypted token together with the=20
      amount authorized. At this point authentication is established.=20
      Authorization is still pending (F6).=20
  -  The VN sends the user the encrypted token in a new header field=20
      'P-NAI-Token' in the response to the user's registration=20
      request (F7).=20
  -  The user decrypts the 'P-NAI-Token' with his private key. He=20
      then encrypts it with his VN's public key. =20
  -  The user is now registered with the VN.=20
  -  The user registers with the HN with the 'P- NAI-Token' (encrypted=20
      with the VN's certificate) inserted in the header (F8).=20
  -  The VN decrypts the token and encrypts it using the HN's public=20
      key. It substitutes the two and forwards this to the HN (F9).
  -  The HN decrypts the received token and compares it with the=20
      locally generated token. If they match, the user is now registered =

      with the HN as well and all necessary trust=20
      relationships have been established.=20
       o The HN has authenticated the user to the VN.=20
       o The VN now trusts the user based on the VN's trusting   =20
          the HN.=20
       o The user trusts the VN. This is established by the VN=20
          possessing the clear text P-NAI-Token.=20
       o The HN receives proof of this trust and can therefore=20
          authorize the VN to bill up to the credit amount given.=20
  -  The HN sends and VN forwards the OK to the second registration =
(F10, F11)
=20

=20
H. Behrens, K. Knuettel                                  [Page 6]=0C


Internet Draft SIP February 8, 2003=20


8. Call Flows=20
=20
8.2.1 Call Flow for pass-through registration =20
(PAR: Proxy Authentication Required)
=20
                          atlanta.com  . . .  .    biloxi.com=20
                           ITSP/GME                ITSP/GME =20
                                                 =20
       Alice's  . . . . . . . . . . . . . . . . . . . . .=20
      softphone                                   =20
     |                      |                     |      =20
     |     REGISTER F1      |                     |      =20
     | with P-NAI header    |                     |
     |--------------------> |                     |      =20
     | 401 Unauthorized F2  |                     | =20
     |<-------------------- |                     |=20
     |     REGISTER F3      |                     |      =20
     |--------------------> |                     |      =20
     | 100 Trying F4        | REGISTER F5         |         =20
     |<-------------------- |---------------->    |      =20
     |                      | 407 PAR F6          |=20
     | 407 PAR F7           |<------------------- |
     |<-------------------- |                     |=20
     |--------------------> |                     |=20
     |     REGISTER F8      |                     |=20
     | contains credentials |                     |=20
     | for user's NAI       |                     |
     |--------------------> |                     |
     |                      |      REGISTER   F9  |=20
     |                      |-------------------> |=20
     |                      |       200 OK F10    |=20
     |                      |<------------------- |=20
     |    200 OK F11        |                     |=20
     |<-------------------- |                     |=20
     |     REGISTER F12     |                     |=20
     | contains token       |                     |=20
     | encrypted for VN     |                     |
     |--------------------> |                     |
     |   200 OK F13         |                     |
     |<---------------------|                     |=20

H. Behrens, K. Knuettel                                  [Page 7]=0C

8.2.2 Call Flow for two independent registrations=20
=20
Client is located in the foreign ITSP-Network and wants to be=20
authenticated by his home network.=20
Within that flow a trust relationship between the ITSPs is established.=20
(PAR: Proxy Authentication Required)
                          atlanta.com  . . .  .    biloxi.com=20
                           ITSP/GME                ITSP/GME =20
                                                 =20
       Alice's  . . . . . . . . . . . . . . . . . . . . .=20
      softphone                                   =20
     |                      |                     |      =20
     |     REGISTER F1      |                     |      =20
     | with P-NAI header    |                     |
     |--------------------> |                     |      =20
     | 407 PAR F2           |                     | =20
     |<-------------------- |                     |=20
     |     REGISTER F3      |                     |      =20
     |--------------------> |                     |      =20
     |                      |                     |=20
     | 100 Trying F4        | Auth-Request(NAI) F5|         =20
     |<-------------------- |------------------>  |      =20
     |                      |                     |=20
     |                      | Auth-OK       F6    |
     |                      | contains P-NAI-Token|
     |                      | encrypted for Alice |
     |                      |<------------------- |
     |                      |                     |
     | 407 Proxy            |                     |=20
     | Authentication       |                     |=20
     |  Required        F7  |                     |=20
     |<-------------------- |                     |=20
     |     REGISTER F8      |                     |=20
     | contains token       |                     |=20
     | encrypted for VN     |                     |
     |--------------------> |                     |
     |                      |      REGISTER   F9  |=20
     |                      | contains token      |=20
     |                      | encrypted for HN    | =20
     |                      |-------------------> |=20
     |                      |       200 OK F10    |=20
     |                      |<------------------- |=20
     |    200 OK F11        |                     |=20
     |<-------------------- |                     |=20
         =20

                                                                         =
                                     =20
=20
IANA Considerations=20
=20
 SIP headers defined: P-NAI, P-NAI-Token=20
=20
 Normative description: This document=20
=20
=20
9. Authors' Addresses=20
=20
 Dr. Harry Behrens=20
 snom technology AG=20
 Pascalstr. 10B=20
 10587 Berlin=20
 Germany=20
 Email: hb@snom.com=20
=20
 Karsten Knuettel=20
 Fraunhofer FOKUS=20

H. Behrens, K. Knuettel                                  [Page 8]=0C


Internet Draft SIP February 8, 2003=20


 Kaiserin-Augusta-Allee 31=20
 10589 Berlin=20
 Email: knuettel@fokus.fraunhofer.de=20
=20
=20
10. Bibliography=20
=20
[CARON.1]Caron, J., =93Public Wireless LAN roaming issues=94,=20
 <draft-caron-public-wlan-roaming-issues-00.txt>,=20
 WIP,=20
 February 2002.=20
[CARON.2]Caron, J., =93DNS-based Roaming=94,=20
 <draft-caron-dns-based-roaming-00.txt>, WIP,=20
 April 2002.=20
[PETERSON.1] Peterson, J. =93Enhancements for=20
 Authenticated Identity=20
 Management in the Session Initiation Protocol=20
 (SIP)=94,=20
 <draft-ietf-sip-identity-01>, WIP, February=20
 2003.=20
[RFC2194] Aboba, B. et al., "Review of Roaming=20
 Implementations", RFC2194, September 1997=20
 [RFC2119] Bradner, S., "Key words for use in RFCs to=20
 Indicate Requirement Levels", RFC2119, March=20
 1997=20
[RFC2486]Aboba, B. et al., "The Network Access=20
 Identifier",=20
 January 1999=20
[RFC3261]Rosenberg, J., Schulzrinne H., Camarillo G.,=20
 Johnston, A.R., Peterson, J., Sparks, R.,=20
 Handley, M. and E. Schooler, "SIP: Session=20
 Initiation Protocol", RFC 3261, Internet=20
 Engineering Task Force, June 2002.=20
[RFC3325] Jennings, C. et al. "Private Extensions to the=20
 Session Initiation Protocol (SIP) for Asserted=20
 Identity within Trusted Networks", November=20
 2002=20
[TS102.321] ETSI TS, =93Telecommunications and Internet=20
 Protocol Harmonization Over Networks (TIPHON);=20
 Open Settlement Protocol (OSP) for Interdomain=20
 Pricing, authorization and usage exchange=94,=20
 Technical Specification 101 321. =20
[RFC3261]=20
 J. Rosenberg, H. Schulzrinne, G. Camarillo, A.=20
 R. Johnston, J.Peterson, R. Sparks, M. Handley,=20
 and E. Schooler, "SIP: Session Initiation=20
 Protocol", RFC 3261, Internet Engineering Task=20
 Force, June 2002.=20
 =20
The IETF takes no position regarding the validity or scope of any=20
intellectual property or other rights that might be claimed to pertain=20
to the implementation or use of the technology described in this=20
document or the extent to which any license under such rights might or=20
might not be available; neither does it represent that it has made any=20
effort to identify any such rights.=20
Information on the IETF's procedures with respect to rights in=20
standards-track and standards-related documentation can be found in=20
BCP-11. Copies of claims of rights made available for publication and=20
any assurances of licenses to be made available, or the result of an=20
attempt made to obtain a general license or permission for the use of=20
such proprietary rights by implementors or users of this specification=20
can be obtained from the IETF Secretariat.=20
 =20

H. Behrens, K. Knuettel                                 [Page 9]=0C


Internet Draft SIP February 8, 2003=20


The IETF invites any interested party to bring to its attention any=20
copyrights, patents or patent applications, or other proprietary rights=20
which may cover technology that may be required to practice this=20
standard. Please address the information to the IETF Executive=20
Director.=20
 =20
Full Copyright Statement=20
 =20
Copyright (c) The Internet Society (2003). All Rights Reserved.=20
 =20
This document and translations of it may be copied and furnished to=20
others, and derivative works that comment on or otherwise explain it or=20
assist in its implementation may be prepared, copied, published and=20
distributed, in whole or in part, without restriction of any kind,=20
provided that the above copyright notice and this paragraph are=20
included on all such copies and derivative works. However, this=20
document itself may not be modified in any way, such as by removing the=20
copyright notice or references to the Internet Society or other=20
Internet organizations, except as needed for the purpose of developing=20
Internet standards in which case the procedures for copyrights defined=20
in the Internet Standards process must be followed, or as required to=20
translate it into languages other than English.=20
 =20
The limited permissions granted above are perpetual and will not be=20
revoked by the Internet Society or its successors or assigns.=20
 =20
This document and the information contained herein is provided on an=20
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING=20
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT=20
NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL=20
NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR=20
FITNESS FOR A PARTICULAR PURPOSE.=20






























           H. Behrens, K. Knuettel                                 [Page =
10]=0C=C1
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hello,

The following draft defines a framework for session policies in SIP
http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt

It defines an event package and a data format to carry session policy 
information. Now, the following draft defines a framework for sending 
configuration information and user profiles to SIP user agents.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt

We need to decide whether session policies should use the package 
defined by the configuration framework or, on the other hand, continue 
defining its own package. The XML based data format would remain the 
same on both cases.

I would like to know what type of information folks intend to store as 
user profile. Would it make sense to consider session policies as a 
subset of the user profile, or it would be better to define a new 
profile-name value: "session-policy" (in addition to the already defined 
"user" and "device").

I tend to think that it would be better to re-use the same package for 
session policies as for configuration, defining the new "session-policy" 
value for the profile-name Event header field parameter, but I would 
like to hear other opinions.

Comments?

A further question of the config draft. Is the intention to establish 
two subscriptions, one for device configuration and another for user 
profiles, or the idea is to use a single subscription and multiplex 
somehow both data types into it.

Thanks,

Gonzalo



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My general concern is that the current config framework is far too 
device-centric, presumably based on the origin and interests of the 
authors. Generally, configuration seems to consist of at least three 
sources that can all contain pretty much the same data, ranging from 
protocol parameters to media policies to user configuration information 
such as speed dial buttons and address lists (or IM equivalents).

The three sources are:
(1) The local network that I happen to be in, represented by the 
outbound proxy or other information conveyed by DHCP, identified by my AOR;

(2) My home provider/network, where I am identified by my AOR;

(3) My device, identified by some permanent device identifier, typically 
a MAC-address like thing, but some of the other identifiers discussed 
here recently may also be possible.

So far, only a subset of (3) seems to have been discussed.

I had raised this issue earlier, but have gotten no response. It seems 
fundamental to the whole architecture discussion. The same problem that 
Gonzalo raises (one or more events) also applies to the question as to 
how this relates to watcher lists and other IM configuration information.

Gonzalo Camarillo wrote:

> Hello,
> 
> The following draft defines a framework for session policies in SIP
> http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt 
> 
> 
> It defines an event package and a data format to carry session policy 
> information. Now, the following draft defines a framework for sending 
> configuration information and user profiles to SIP user agents.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt 
> 
> 
> We need to decide whether session policies should use the package 
> defined by the configuration framework or, on the other hand, continue 
> defining its own package. The XML based data format would remain the 
> same on both cases.
> 
> I would like to know what type of information folks intend to store as 
> user profile. Would it make sense to consider session policies as a 
> subset of the user profile, or it would be better to define a new 
> profile-name value: "session-policy" (in addition to the already defined 
> "user" and "device").
> 
> I tend to think that it would be better to re-use the same package for 
> session policies as for configuration, defining the new "session-policy" 
> value for the profile-name Event header field parameter, but I would 
> like to hear other opinions.
> 
> Comments?
> 
> A further question of the config draft. Is the intention to establish 
> two subscriptions, one for device configuration and another for user 
> profiles, or the idea is to use a single subscription and multiplex 
> somehow both data types into it.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> 
> This communication is confidential and intended solely for the 
> addressee(s). Any unauthorized review, use, disclosure or distribution 
> is prohibited. If you believe this message has been sent to you in 
> error, please notify the sender by replying to this transmission and 
> delete the message without disclosing it. Thank you.
> 
> E-mail including attachments is susceptible to data corruption, 
> interruption, unauthorized amendment, tampering and viruses, and we only 
> send and receive e-mails on the basis that we are not liable for any 
> such corruption, interception, amendment, tampering or viruses or any 
> consequences thereof.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hello,

Regarding requirement number 5 of the following draft:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-e2m-sec-reqs-01.txt

*****
5.  It SHOULD allow a UA to request the recipient UA to disclose the
        same information that the requesting UA is providing to the proxy
        server to the same proxy server. The request itself SHOULD be
        secure.
*****

We have had some discussions on this issue when working on the session 
policies drafts:

http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-spec-policy-00.txt

Our conclusions so far are that mechanisms were *one* of the UAs inform 
the proxy about the session are simpler. Of course, one UA can keep its 
trusted proxies up to date while the other does the same with its own 
trusted proxies.

Unfortunatelly, we did not get around to releasing a new revision of the 
draft above, but have been doing some brainstorming on the, so called, 
re-direct model, where UAs establish separate dialogs with the proxies. 
These dialogs are related to the main dialog between the UAs, but are 
different dialogs.

Having an ad-hoc meeting in Korea to discuss these issues may be useful.

As a chair, I would encourage the authors of both drafts to get in touch 
with each other.

Thanks,

Gonzalo



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From exim@www1.ietf.org  Mon Feb 23 09:00:56 2004
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] E2M security solution
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Hello,

The e2m security solution, described in the draft below,
http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-01.txt

attempts to resolve some of the problems that the session-specific 
policies draft addresses:
http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-spec-policy-00.txt

The need of a UA to obtain the policies that a particular proxy has is 
dealt with in both drafts. In addition, the e2m security draft talks 
about disclosing information to intermediaries. The policy draft 
addresses this issue and defines a means to disclose *only* the 
information needed by the proxy.

My comments on the e2m requirements draft apply also to this one. We 
should try to align both efforts (session policies and e2m security).

Thanks,

Gonzalo


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From: "Scoggins, Sophia" <sscoggins@telogy.com>
To: sipping@ietf.org
Subject: RE: [Sipping] SIP Proxy servers -- security encryption protocol
Date: Mon, 23 Feb 2004 09:49:38 -0500
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Thanks for the information. For each security association establishment,
there is a significant delay. Depending on the DH group, both phases (phase
1 authentication, phase 2 key exchange) have delays ranging from 2 seconds
to 10 seconds using ISKIKE. That is before any signaling messages started.
This is not acceptable in call processing. 

 

Is any key exchange protocol recommended? What is the performance (delays)
for key exchange?

 

Why is AES blocked? AES is supposed to have better performance than 3DES.

 

 

Regards,

Sophia

 

-----Original Message-----
From: Cullen Jennings [mailto:fluffy@cisco.com] 
Sent: Sunday, February 22, 2004 3:40 PM
To: ranjit.avasarala@wipro.com; sscoggins@telogy.com; sipping@ietf.org
Subject: Re: [Sipping] SIP Proxy servers -- security encryption protocol

 


And the crypto that is typically being recommended and folks are
implementing is RSA for asymmetric key, SHA1 for hash, and 3DES or AES for
encryption. Some of the early stuff said 3DES because the RFC would have
blocked on AES work. It's a long debate with no clear answer between AES and
3DES (and let's not repeat it on this list) but most SIP implementers seem
to be doing AES or both AES and 3DES.  

On 2/19/04 9:26 PM, "ranjit.avasarala@wipro.com"
<ranjit.avasarala@wipro.com> wrote:

Hi
    For secured transactions you could use TLS i.e using sips: instead of
normal sip: url.
Alternatively IPSec also could be used at IP Layer
and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt)
protocol.

Regards
Ranjit


 
-----Original Message-----
From:  sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
<mailto:sipping-admin@ietf.org%5d>  On Behalf Of  Scoggins, Sophia
Sent: Friday, February 20, 2004 10:45  AM
To: 'sipping@ietf.org'
Subject: [Sipping] SIP Proxy  servers -- security encryption protocol


 

 
 
My group is looking for some  security encryption protocol for both SIP and
RTP streams for CPE. We were  wondering what encryption protocol that most
SIP proxy servers currently  support or intend to support? Would you please
just send your reply to  me?
 
 
 
Thanks.
 
 
 
Regards,
 
Sophia2@ti.com

 

 


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<title>Re: [Sipping] SIP Proxy servers -- security encryption protocol</title>

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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks for the information. For each
security association establishment, there is a significant delay. Depending on
the DH group, both phases (phase 1 authentication, phase 2 key exchange) have
delays ranging from 2 seconds to 10 seconds using ISKIKE. That is before any
signaling messages started. This is not acceptable in call processing. </span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Is any key exchange protocol recommended?
What is the performance (delays) for key exchange?</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Why is AES blocked? AES is supposed to
have better performance than 3DES.</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Regards,</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Sophia</span></font></p>

<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> Cullen Jennings
[mailto:fluffy@cisco.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Sunday, February 22, 2004
3:40 PM<br>
<b><span style='font-weight:bold'>To:</span></b> ranjit.avasarala@wipro.com;
sscoggins@telogy.com; sipping@ietf.org<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Sipping] SIP Proxy
servers -- security encryption protocol</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

<p class=MsoNormal style='margin-right:0in;margin-bottom:12.0pt;margin-left:
.5in'><font size=2 face=Verdana><span style='font-size:10.5pt;font-family:Verdana'><br>
And the crypto that is typically being recommended and folks are implementing
is RSA for asymmetric key, SHA1 for hash, and 3DES or AES for encryption. Some
of the early stuff said 3DES because the RFC would have blocked on AES work.
It's a long debate with no clear answer between AES and 3DES (and
let's not repeat it on this list) but most SIP implementers seem to be
doing AES or both AES and 3DES. &nbsp;<br>
<br>
On 2/19/04 9:26 PM, &quot;ranjit.avasarala@wipro.com&quot;
&lt;ranjit.avasarala@wipro.com&gt; wrote:</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 color=blue face=Arial><span
style='font-size:10.5pt;font-family:Arial;color:blue'>Hi<br>
&nbsp;&nbsp;&nbsp;&nbsp;For secured transactions you could use TLS i.e using
sips: instead of normal sip: url.<br>
Alternatively IPSec also could be used at IP Layer<br>
and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt) protocol.<br>
</span></font><font size=2 face=Verdana><span style='font-size:10.5pt;
font-family:Verdana'><br>
</span></font><font size=2 color=blue face=Arial><span style='font-size:10.5pt;
font-family:Arial;color:blue'>Regards<br>
Ranjit</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Verdana><span
style='font-size:10.5pt;font-family:Verdana'><br>
&nbsp;<br>
</span></font><font size=2 face=Tahoma><span style='font-size:10.5pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b> &nbsp;sipping-admin@ietf.org
[<a href="mailto:sipping-admin@ietf.org%5d">mailto:sipping-admin@ietf.org]</a> <b><span
style='font-weight:bold'>On Behalf Of &nbsp;</span></b>Scoggins, Sophia<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, February 20, 2004
10:45 &nbsp;AM<br>
<b><span style='font-weight:bold'>To:</span></b> 'sipping@ietf.org'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Sipping] SIP Proxy
&nbsp;servers -- security encryption protocol<br>
<br>
</span></font><font size=2 face=Verdana><span style='font-size:10.5pt;
font-family:Verdana'><br>
&nbsp;<br>
<br>
&nbsp;<br>
&nbsp;<br>
My group is looking for some &nbsp;security encryption protocol for both SIP
and RTP streams for CPE. We were &nbsp;wondering what encryption protocol that
most SIP proxy servers currently &nbsp;support or intend to support? Would you please
just send your reply to &nbsp;me?<br>
&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
Thanks.<br>
&nbsp;<br>
&nbsp;<br>
&nbsp;<br>
Regards,<br>
&nbsp;<br>
Sophia2@ti.com</span></font></p>

<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>&nbsp;</span></font></p>

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style='font-size:12.0pt'>&nbsp;</span></font></p>

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_______________________________________________
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From exim@www1.ietf.org  Mon Feb 23 10:18:06 2004
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Subject: [Sipping] Re: Config framework
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I have not gotten through all of draft-hilt-sipping-session-indep-policy-00
yet, but I certainly consider policies such as access and authorization to
be part of  user and device profile data.  Whether directly or by reference
the policy data applies to the user or the device.  I would hope that this
could all be decovered and delivered using the same mechanizm.

Ideally, it would be nice if all user profile data (including policies) or
all device data could optionally be delivered via one URL.  Using XML
name spaces the subscriber can specify what it wants and parse out the
parts of interest.
Cheers,
Dan

Gonzalo Camarillo wrote:

> Hello,
>
> The following draft defines a framework for session policies in SIP
> http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt
>
> It defines an event package and a data format to carry session policy
> information. Now, the following draft defines a framework for sending
> configuration information and user profiles to SIP user agents.
>
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt
>
> We need to decide whether session policies should use the package
> defined by the configuration framework or, on the other hand, continue
> defining its own package. The XML based data format would remain the
> same on both cases.
>
> I would like to know what type of information folks intend to store as
> user profile. Would it make sense to consider session policies as a
> subset of the user profile, or it would be better to define a new
> profile-name value: "session-policy" (in addition to the already defined
> "user" and "device").
>
> I tend to think that it would be better to re-use the same package for
> session policies as for configuration, defining the new "session-policy"
> value for the profile-name Event header field parameter, but I would
> like to hear other opinions.
>
> Comments?
>
> A further question of the config draft. Is the intention to establish
> two subscriptions, one for device configuration and another for user
> profiles, or the idea is to use a single subscription and multiplex
> somehow both data types into it.
>
> Thanks,
>
> Gonzalo
>
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>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping <sipping@ietf.org>, Volker Hilt <volkerh@bell-labs.com>,
        Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sipping] Config framework
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Comments below:

Henning Schulzrinne wrote:

> My general concern is that the current config framework is far too
> device-centric, presumably based on the origin and interests of the

This is certainly not the intension.  The draft specifies a way to separately
subscribe for device or user profile data.  I would be happy to clarify this
where you think the draft is vague or implies this restriction.

Section 3.2 describes parameters that are device specific as even in the
ideal world, purely user specific data may need to consider the UA
implementation that it is to be applied to.  Section 3.2 also specifies a
means to indicate the desire for user or device data.

Section 3.13 specifies how to form the SUBSCRIBE URI based upon
the interest for either a device or user profile.



>
> authors. Generally, configuration seems to consist of at least three
> sources that can all contain pretty much the same data, ranging from
> protocol parameters to media policies to user configuration information
> such as speed dial buttons and address lists (or IM equivalents).
>
> The three sources are:
> (1) The local network that I happen to be in, represented by the
> outbound proxy or other information conveyed by DHCP, identified by my AOR;
>
> (2) My home provider/network, where I am identified by my AOR;

>
>
> (3) My device, identified by some permanent device identifier, typically
> a MAC-address like thing, but some of the other identifiers discussed
> here recently may also be possible.
>

These 3 important concepts are intended to be supported in
draft-ietf-sipping-config-framework-02.  It is however difficult in
practice to keep these three things separate.  In many environments
it is easier to keep two of the three separate.

For example:
In an enterprise with physical devices that do not move much, it becomes
easier to combine the properties of 1 related to location with the properties
of 3.

When a user does not know their AOR or does not want to have to provide
it to the UA (to provide a plug and play experience), the user's AOR can be
associated with 3 (the device).  2 is may then be subscribe implicitly or explicitly.

In the automated discover phase for a new UA, the distinction between
1 and 2 become blurred as the device can only discover or derive
AORs based upon the local domain (without user input).

I think that this illustrates a very good point that the separation between
user and device profile data cannot be made concrete.  Data such as
out bound proxy, or policy may be dictated by any or all of the 3
realms defined above.  Only the application and local administrative
policies can determine the realm to which data should be associated.
In addition it may be desirable to override some of the user or device
profile data based upon other local discovery data (e.g. DHCP options
for out bound proxy, STUN for NAT traversal).

At best we can define a precedence by which the device data
can be overridden by user data (or the other way around)
along with the ability for administrators to dictate policy (perhaps indicated
in the profile data) over what can be overridden.





>
> So far, only a subset of (3) seems to have been discussed.
>
> I had raised this issue earlier, but have gotten no response. It seems
> fundamental to the whole architecture discussion. The same problem that
> Gonzalo raises (one or more events) also applies to the question as to
> how this relates to watcher lists and other IM configuration information.
>
> Gonzalo Camarillo wrote:
>
> > Hello,
> >
> > The following draft defines a framework for session policies in SIP
> > http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt
> >
> >
> > It defines an event package and a data format to carry session policy
> > information. Now, the following draft defines a framework for sending
> > configuration information and user profiles to SIP user agents.
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt
> >
> >
> > We need to decide whether session policies should use the package
> > defined by the configuration framework or, on the other hand, continue
> > defining its own package. The XML based data format would remain the
> > same on both cases.
> >
> > I would like to know what type of information folks intend to store as
> > user profile. Would it make sense to consider session policies as a
> > subset of the user profile, or it would be better to define a new
> > profile-name value: "session-policy" (in addition to the already defined
> > "user" and "device").
> >
> > I tend to think that it would be better to re-use the same package for
> > session policies as for configuration, defining the new "session-policy"
> > value for the profile-name Event header field parameter, but I would
> > like to hear other opinions.
> >
> > Comments?
> >
> > A further question of the config draft. Is the intention to establish
> > two subscriptions, one for device configuration and another for user
> > profiles, or the idea is to use a single subscription and multiplex
> > somehow both data types into it.
> >
> > Thanks,
> >
> > Gonzalo
> >
> >
> >
> > This communication is confidential and intended solely for the
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> > delete the message without disclosing it. Thank you.
> >
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> > consequences thereof.
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP


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



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Date: Mon, 23 Feb 2004 11:17:25 -0500
From: David Oran <oran@cisco.com>
To: "Scoggins, Sophia" <sscoggins@telogy.com>, sipping@ietf.org
Subject: RE: [Sipping] SIP Proxy servers -- security encryption protocol
Message-ID: <2147483647.1077535045@[10.32.245.156]>
In-Reply-To: <37A3C2F21006D611995100B0D0F9B73C04C3E444@tnint11.telogy.design.ti.com>
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--On Monday, February 23, 2004 9:49 AM -0500 "Scoggins, Sophia" 
<sscoggins@telogy.com> wrote:

>
>
> Thanks for the information. For each security association establishment,
> there is a significant delay. Depending on the DH group, both phases
> (phase 1 authentication, phase 2 key exchange) have delays ranging from 2
> seconds to 10 seconds using ISKIKE. That is before any signaling messages
> started. This is not acceptable in call processing.
>
so, set up the TLS connection ahead of time. Do you think you'll have so 
many inter-provider signaling associations that this would be a scalability 
concern?

You should also look into TLS "fast restart" which caches the results of 
the first phase of key management and is only one extra RTT since the 
results of the prior PK computations are kept.

>
>
> Is any key exchange protocol recommended? What is the performance
> (delays) for key exchange?
>
>
>
> Why is AES blocked? AES is supposed to have better performance than 3DES.
>
>
>
>
>
> Regards,
>
> Sophia
>
>
>
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Sunday, February 22, 2004 3:40 PM
> To: ranjit.avasarala@wipro.com; sscoggins@telogy.com; sipping@ietf.org
> Subject: Re: [Sipping] SIP Proxy servers -- security encryption protocol
>
>
>
>
> And the crypto that is typically being recommended and folks are
> implementing is RSA for asymmetric key, SHA1 for hash, and 3DES or AES
> for encryption. Some of the early stuff said 3DES because the RFC would
> have blocked on AES work. It's a long debate with no clear answer between
> AES and 3DES (and let's not repeat it on this list) but most SIP
> implementers seem to be doing AES or both AES and 3DES.
>
> On 2/19/04 9:26 PM, "ranjit.avasarala@wipro.com"
> <ranjit.avasarala@wipro.com> wrote:
>
> Hi
>     For secured transactions you could use TLS i.e using sips: instead of
> normal sip: url.
> Alternatively IPSec also could be used at IP Layer
> and for RTP security you could use SRTP (draft-ietf-avt-srtp-09.txt)
> protocol.
>
> Regards
> Ranjit
>
>
>
> -----Original Message-----
> From:  sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
> Of  Scoggins, Sophia
> Sent: Friday, February 20, 2004 10:45  AM
> To: 'sipping@ietf.org'
> Subject: [Sipping] SIP Proxy  servers -- security encryption protocol
>
>
>
>
>
>
> My group is looking for some  security encryption protocol for both SIP
> and RTP streams for CPE. We were  wondering what encryption protocol that
> most SIP proxy servers currently  support or intend to support? Would you
> please just send your reply to  me?
>
>
>
> Thanks.
>
>
>
> Regards,
>
> Sophia2@ti.com
>
>
>
>





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



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Thread-Topic: [Simple] Advanced IM requirements
Thread-Index: AcP5C/kBvwKHvMXJS6urOHqe9sfskgA3pGvQABD6gKA=
From: "Chris Boulton" <cboulton@ubiquity.net>
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As promised.

https://www.ubiquity.net/draft-boulton-sipping-message-receipt-00.txt

Chris.


>-----Original Message-----
>From: Chris Boulton
>Sent: 23 February 2004 10:40
>To: 'Jonathan Rosenberg'; Simple WG
>Cc: 'sipping@ietf.org'
>Subject: RE: [Simple] Advanced IM requirements
>
>Hi Jonathan et al,
>
>	I have an extremely early version (so be gentle) of a draft
aimed at
>supplying IM delivery reports.  It hasn't been formally submitted yet
but I
>have attached a copy - so would be grateful if the list could take a
look
>and see if it's worth progressing.  I will submit to the archives in
the
>next few days.  I will also put this on the internet somewhere in case
this
>attachment doesn't survive its journey (Will send link later).
>
>Best Regards,
>
>Chris.
>
>
>>-----Original Message-----
>>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>Sent: 22 February 2004 06:17
>>To: Simple WG
>>Subject: [Simple] Advanced IM requirements
>>
>>Folks,
>>
>>As I noted earlier, I resubmitted the advanced IM requirements draft.
It
>>specifies a bunch of advanced IM functions we havent yet tackled here.
>>They are:
>>
>>* delivery reports for IM
>>* is-typing indicator (Henning had an I-D on this that has expired)
>>* capabilities indications of maximum desired message sizes for IM
>>* page mode groups (under discussion in the context of exploders in
>>sipping)
>>* invitations to non-real time sessions
>>
>>
>>The last of these is particularly interesting. One use case is "call
>>alerts" in PTT. There, I send you a request to have a PTT chat. The
>>request is not answered in real time. Rather, its stored by the
>>recipient as if it were an IM, and can be answered at any time later.
>>"Answering it" merely involves setting up a PTT call back to the
caller.
>>Its also possible to cancel the call alert at any time.
>>
>>There are some interesting ways to accomplish this function. The one
>>I've been thinking about is taking a SIP INVITE message, and sending
it
>>in the body of a MESSAGE request as a message/sip content. I'm sure
>>there are other ways.
>>
>>Are these topics for which there is still group interest in working
on?
>>I think all of them remain quite important.
>>
>>Thanks,
>>Jonathan R.
>>--
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>
>>_______________________________________________
>>Simple mailing list
>>Simple@ietf.org
>>https://www1.ietf.org/mailman/listinfo/simple


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Date: Mon, 23 Feb 2004 12:07:43 -0500
From: Volker Hilt <volkerh@bell-labs.com>
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To: "Daniel G. Petrie" <dpetrie@pingtel.com>
CC: Henning Schulzrinne <hgs@cs.columbia.edu>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping <sipping@ietf.org>,
        Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com>
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Policies and configuration information seem to share a number of 
commonalities. Thus, I think it would make sense to use the same 
framework for both. The framework should probably be able to transfer 
any kind of information from a network entity to a user agent.

More comments inline.

Daniel G. Petrie wrote:
> Henning Schulzrinne wrote:
>>My general concern is that the current config framework is far too
>>device-centric, presumably based on the origin and interests of the
> 
> This is certainly not the intension.  The draft specifies a way to separately
> subscribe for device or user profile data.  I would be happy to clarify this
> where you think the draft is vague or implies this restriction.
> 
Is it safe to assume that policy information is always part of the 
device or user configuration or can it be an independent piece of 
information a UA wants to subscribe to? Thus, would it make sense to 
define a profile-name "session-policies"? This could be useful if a 
device is configured by other means and simply wants to keep track of 
the policies in a network.

>>
>>The three sources are:
>>(1) The local network that I happen to be in, represented by the
>>outbound proxy or other information conveyed by DHCP, identified by my AOR;
>>
>>(2) My home provider/network, where I am identified by my AOR;
>>
>>(3) My device, identified by some permanent device identifier, typically
>>a MAC-address like thing, but some of the other identifiers discussed
>>here recently may also be possible.
>
> These 3 important concepts are intended to be supported in
> draft-ietf-sipping-config-framework-02.  It is however difficult in
> practice to keep these three things separate.  In many environments
> it is easier to keep two of the three separate.
> 
> For example:
> In an enterprise with physical devices that do not move much, it becomes
> easier to combine the properties of 1 related to location with the properties
> of 3.
> 
> When a user does not know their AOR or does not want to have to provide
> it to the UA (to provide a plug and play experience), the user's AOR can be
> associated with 3 (the device).  2 is may then be subscribe implicitly or explicitly.
>
> In the automated discover phase for a new UA, the distinction between
> 1 and 2 become blurred as the device can only discover or derive
> AORs based upon the local domain (without user input).
> 
A difference between configuration data and other information (such as 
policies) seems to be that subscriptions to configuration data must work 
on an "out-of-the-box" device that has not been configured yet. Once the 
configuration is done, its probably easier to subscribe to the different 
sources of information.

Is there information that applies to a device besides the device 
configuration? If not, would it make sense to differentiate into the 
device configuration (requiring a discovery process) and subscriptions 
to the local network using my AOR as well as subscriptions to my home 
network (also using my AOR)?

Thanks,

Volker




> I think that this illustrates a very good point that the separation between
> user and device profile data cannot be made concrete.  Data such as
> out bound proxy, or policy may be dictated by any or all of the 3
> realms defined above.  Only the application and local administrative
> policies can determine the realm to which data should be associated.
> In addition it may be desirable to override some of the user or device
> profile data based upon other local discovery data (e.g. DHCP options
> for out bound proxy, STUN for NAT traversal).
> 
> At best we can define a precedence by which the device data
> can be overridden by user data (or the other way around)
> along with the ability for administrators to dictate policy (perhaps indicated
> in the profile data) over what can be overridden.
> 
> 
>>So far, only a subset of (3) seems to have been discussed.
>>
>>I had raised this issue earlier, but have gotten no response. It seems
>>fundamental to the whole architecture discussion. The same problem that
>>Gonzalo raises (one or more events) also applies to the question as to
>>how this relates to watcher lists and other IM configuration information.
>>
>>Gonzalo Camarillo wrote:
>>
>>
>>>Hello,
>>>
>>>The following draft defines a framework for session policies in SIP
>>>http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt
>>>
>>>
>>>It defines an event package and a data format to carry session policy
>>>information. Now, the following draft defines a framework for sending
>>>configuration information and user profiles to SIP user agents.
>>>
>>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt
>>>
>>>
>>>We need to decide whether session policies should use the package
>>>defined by the configuration framework or, on the other hand, continue
>>>defining its own package. The XML based data format would remain the
>>>same on both cases.
>>>
>>>I would like to know what type of information folks intend to store as
>>>user profile. Would it make sense to consider session policies as a
>>>subset of the user profile, or it would be better to define a new
>>>profile-name value: "session-policy" (in addition to the already defined
>>>"user" and "device").
>>>
>>>I tend to think that it would be better to re-use the same package for
>>>session policies as for configuration, defining the new "session-policy"
>>>value for the profile-name Event header field parameter, but I would
>>>like to hear other opinions.
>>>
>>>Comments?
>>>
>>>A further question of the config draft. Is the intention to establish
>>>two subscriptions, one for device configuration and another for user
>>>profiles, or the idea is to use a single subscription and multiplex
>>>somehow both data types into it.
>>>
>>>Thanks,
>>>
>>>Gonzalo
>>>
>>>
>>>
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>>>_______________________________________________
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> 
> 


-- 
Volker Hilt                      Bell Labs / Lucent Technologies
phone: +1 732 332 6432           101 Crawfords Corner Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733

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Date: Mon, 23 Feb 2004 11:27:17 -0600
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Cc: "Rosen, Brian" <Brian.Rosen@marconi.com>
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Subject: [Sipping] New version of Location Conveyance Reqs ID out
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All

This is to update the list on the submission of the latest version of the 
Location Conveyance ID posted recently.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-location-requirements-00.txt

Open issues since the last version that are addressed in this version are:

1. Location should always be transmitted, even if not considered reliable.

2. UAs SHOULD be able to inform the ERC if it considers the location
         info "probably wrong".

         should a flag saying this LI is probably wrong be included?
         This could be a listed open item.

3. In the situation of a VPN, the UA SHOULD keep the first DHCP
         Location Reply, and not the one sent in the VPN set-up.

4. Requirement to indicate in the INVITE where the LI came from (GPS,
         Cell Tower Triangulation, WiFi, DHCP, manual entry as examples).

         And possibly include all that is available

5. Use TLS in emergency calls (eliminates the issue of self signed
         certs and S/MIME), and use S/MIME in normal location
         conveyance scenarios between UAs.

6. Title should have the work "Requirements" in it.

Not addressed and looking for more clarity on each requirement

N1. Requirement to allow ERCs to do "after the fact" investigation on
         why a location was provided (that ended up being wrong).

N2. MUST be able to communicate an MSAG-like equivalent location
         field.

Part that is still open:
O1. Proxies SHOULD be able to insert Location into an emergency
         call in a multipart body or return an LO in a Redirect Response

New open issues are

New-1 - surprised the requirement doesn't exist for a self signed cert
         for S/MIME?

         Current status - Not sure if this should be MUST, SHOULD or
         MAY (or even mentioned, but not disallowed)

New-2 - - there needs to be a "confidence value" included with the LI
         in a SIP message. Currently Cell tower triangulation provides this,
         and isn't sure how many other technologies do.

New-3 - - there should be a flag of where the LI was derived (.11,
         GPS, manual entry, etc)

Any discussion prior to the SIPPING meeting would be appreciated

Thanks





cheers,
James

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


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Subject: RE: [Sipping] updated dialog package
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Hi,

I work for a company that has developed handset gateway devices that
enable legacy, digital, business-grade telephones to be driven using
SIP.  For example, this enables an existing legacy PBX user to retain
their telephones and cabling, but to replace the PBX by a Softswitch or
IP Centrex system.  As most business-grade telephone systems make use of
a shared line facility, the addition of a shared line example to the
dialog event package ID is of great interest to us, and others who
support these types of feature.

Although incorporating a shared line example is a step towards a common
approach to shared line implementation, I don't believe it goes far
enough.  Our current implementation of shared lines has shown that the
following issues need to be addressed by the spec, and should be shown
in the examples:

o Line seize, an essential part of any shared line implementation, is
not adequately described. When more than one user agent attempts to
seize a line at approximately the same instant, a race condition arises.
No description of how to handle the race is given - an example
application/dialog-info+xml NOTIFY payload is provided, but this alone
is inadequate, as message sequencing is an important part of seizing a
line.  The Line Seize condition will be covered in the
"anil-sipping-bla-draft" that Venkatesh refers to.

o Handling of multiple instances of a shared line has been overlooked.
On business-grade PBX systems a line will roughly correspond to an AOR.
However, many such systems permit multiple "instances" of a shared line
to appear on a single endpoint. For example, a number of instances of a
"product support" shared line might appear on all the endpoints of the
product support group, thereby allowing multiple calls to be taken and
passed around within the group.

o The mechanism used to convey that a call has been placed on hold
appears to rely upon implementation-specific parameters, found in the
xml's <param> element. I can't see any reference to definitions of the
values used by this element's pname and pval attributes
(pname=3D"+activity" pval=3D"noninteractive"), so I assume they're
non-standard.

Thanks,

Paul.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]=20
> Sent: Thursday, February 19, 2004 6:00 PM
> To: sipping
> Cc: rohan@cisco.com
> Subject: [Sipping] updated dialog package
>=20
>=20
> Hi Everyone,
>=20
> I've updated the dialog packages with the changes we agreed to in=20
> Minneapolis (removing CSeq and the Route-Set).
>=20
> I've also provided a detailed example of using the dialog=20
> package for a=20
> "shared-line" application.  I apologize for my tardiness here.  I was=20
> supposed to post something shortly after the meeting.  I any case, I=20
> believe this demonstrates that the current syntax is at least=20
> sufficient for the use cases we've discussed.  I would like to hear=20
> from all of you if this does the trick.
>=20
> many thanks,
> -rohan
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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Subject: Re: [Sipping] Roaming Architecture for SIP
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Harry Behrens wrote:
> We have submitted a draft to the IETF
> (draft-behrens-sipping-roaming-architecture-00.txt),
> in which we describe a mechanism to enable VoIP roaming based on a
> unique ID - the NAI - which constitutes
> a handle into the billing relationship for a user.
> The original draft has some mistakes in the discussion of the call flows
> as well as not being 100% correctly formatted.
> We will therefore submit an updated version after the cutoff date, which
> we attach in this mail.
> Comments etc. are very welcome.

I have specific comment inline below, and then a question.

	Paul

>   -  A user registers with the VN (atlanta.com in the call flow 
>       given). The user submits his NAI in new header field 'P-NAI'. 
>       The user submits a bogus URI in both the From and the To header. 
>       (F1 in section 8.2.1)
>   -  The VN recognizes the registration as a P-NAI registration and 
>       sends back a "401 Unauthorized" response, with both the 'From' 
>       and the 'To' URI set to the temporary URI that will be assigned 
>       to the user upon successful authentication and authorization (F2). 

This isn't legal. The From and To (with exception of To-tag) of response 
must be same as in the request.

The question is: why would the UAC want to register with the VN? 
Assuming it is able to send sip messages, (which apparently it can), why 
wouldn't it simply send a REGISTER to its HN?

I presume there must be some service that you intend the VN won't 
perform, that will prevent the UAC from doing this. But it isn't 
apparent what service that is.

	Paul


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



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Chris Boulton wrote:
> As promised.
> 
> https://www.ubiquity.net/draft-boulton-sipping-message-receipt-00.txt

Chris - just a couple of comments:

1) I don't see any particular reason to link this draft to GRUUs. The 
way receipts work, they may not be delivered until long after the 
message is sent. Anything that forces the receipt to be delivered to the 
same UA instance that sent the message is likely to cause problems. The 
response may well need to be captured by a recording device that 
wouldn't share a GRUU with the sending device. So I suspect that 
normally Receipt-To should mention the AOR of the sender. Of course the 
sender is still free to use a GRUU if desired, but it certainly isn't 
required.

2) I have some doubts about the need and value of the message-receipt 
option tag. Even if I sent you a message with Require: message-receipt, 
you are still free to not send me a receipt. (In the case of receipts 
with email I generally refuse to send them to parties unknown to me.) As 
a result, the option tag adds little or nothing. It should be sufficient 
to simply use the Receipt-To header without any option tag, and hope the 
recipient is willing and able to send a response.

	Paul


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



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In section 4.1 a change was made around the descriptin of handling a 
subscription. The new wording seems wrong:

    The SUBSCRIBE should be sent outside the INVITE-initiated dialog.

    Participants leaving the conference send a BYE to the focus.  If they
    have a current subscription to the conference package, they also must
    send a SUBSCRIBE with an Expires:0 header field to terminate both
    dialogs.

There is nothing for "The SUBSCRIBE" to refer back to. I wonder if this 
sentence wasn't intended to be in section 3.4, where it would make more 
sense.

	Paul


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Subject: Re: [Sipping] Re: sos - dead or alive
References: <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220103319.02fbf370@cia.cisco.com>
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This discussion is relevant to the case where the ECC is IP-reachable, but what 
about legacy ECCs reachable through gateways?  What happens when some node along 
the chain tries to run a reachability test in this case?  Bear in mind that only 
the gateway is likely to know that the call is going to the PSTN.

Michael Hammer wrote:

> Henning,
> 
> The reality of deployment would be that every ECC would have it's own 
> proxy, so that would be more like 1000 x $2k = $2 million.  But, the 
> corresponding distribution of load across those servers means it is even 
> more of a non-problem at any given site.
> 
> One question:  A real successful test to a wrong ECC doesn't buy you 
> much.  I hope that mobility is addressed in the solution as well.  Many 
> of those 100 million phones are mobile, which might become SIP-based 
> some time in the future.  so I would add a 6) "works with mobile 
> terminals" to your list below.
> 
> Mike
> 
> At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
> 
>> I think there are five goals, somewhat contradictory:
>>
>> (1) make it as realistic as possible, to make sure you are really 
>> testing the emergency calling system, not the testing system; after 
>> all, the fire safety officer doesn't go around ringing a cowbell 
>> instead of the real fire alarms.
>>
>> (2) do not impose any additional operator involvement; this is solved 
>> by the OPTIONS mechanism proposed;
>>
>> (3) allow for reasonable scaling;
>>
>> (4) avoid interfering with real emergency calls
>>
>> (5) allow user on-demand testing.
>>
>> With the proxy-based testing I proposed, we get very close to all 
>> goals. In the worst case, i.e., every device does this once a day, we 
>> get roughly 100 million calls a day in the US, or 1157/second. That's 
>> about 10 proxy servers worth of equipment, i.e., about $20k. This 
>> hardly seems worth spending inordinate engineering resources on 
>> avoiding, particularly since the likely load is going to be lower by 
>> probably a factor of 1000 or more (I'm assuming that most residential 
>> users will use a cable or DSL provider for this, for example).
>>
>> With even a few hundred systems, you'll get almost perfect smoothing 
>> and very little burstiness.
>>
>> I know protocol design is fun, but I'm very worried about adding 
>> complexity to solve largely a non-problem.
>>
>> Michael Hammer wrote:
>>
>>> At 09:03 AM 2/19/2004 -0500, Mpierce1@aol.com wrote:
>>>
>>>> In a message dated 2/19/2004 1:58:18 AM Eastern Standard Time, 
>>>> jgunn6@csc.com writes:
>>>>
>>>>
>>>>> So any kind of regular, scheduled test should be designed so it does
>>>>> NOT
>>>>> tie up the operators.
>>>>
>>>>
>>>>
>>>> I believe the test must be devised so that it does not involve the 
>>>> emergency center or any of the equipment directly supporting the 
>>>> emergency center at all.
>>>>
>>>> Mike
>>>
>>>
>>> Then there is not much point in doing any test.  As Brian points out, 
>>> the question is how do you stage these reachability tests so that 
>>> they scale, ie. you don't have millions of users pinging the ECC 
>>> directly.
>>> But, instead of the proxies pinging the ECC, which might reduce the 
>>> load, but might not make the burstiness more deterministic, perhaps a 
>>> presence subscription would be more appropriate.  The proxies could 
>>> subscribe to the presence of the ECC, and the ECC could periodically 
>>> Notify those proxies that it is still there and able to receive calls.
>>> The ECC could then control the scheduling and load distribution for 
>>> sending the Notifies.  Failing to receive the notifications beyond 
>>> some interval could provide an indication that something is wrong and 
>>> trigger an inquiry.  You might even be able to extend this to the 
>>> users, where emergency help is a "buddy" you subscribe to on the 
>>> local proxy.  I haven't thought this through, but thought I'd throw 
>>> that out.
>>> Mike H.
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

_______________________________________________
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CC: Michael Hammer <mhammer@cisco.com>, Mpierce1@aol.com, jgunn6@csc.com,
        sipping@ietf.org
Subject: Re: [Sipping] Re: sos - dead or alive
References: <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220093002.00b283f0@cia.cisco.com> <4.3.2.7.2.20040220103319.02fbf370@cia.cisco.com> <403A5FEE.9020205@nortelnetworks.com>
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If it's an OPTIONS and/or sos.test request, it will get to the last SIP 
hop, but no further. I don't think we can do more than that unless ISUP 
or Q.931 has a 'just testing' request.

Tom Taylor wrote:

> This discussion is relevant to the case where the ECC is IP-reachable, 
> but what about legacy ECCs reachable through gateways?  What happens 
> when some node along the chain tries to run a reachability test in this 
> case?  Bear in mind that only the gateway is likely to know that the 
> call is going to the PSTN.


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



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Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
Date: Mon, 23 Feb 2004 17:34:22 -0500
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Yep, yep, yep, we can (and will) use GRUU.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, February 05, 2004 2:46 AM
> To: sipping@ietf.org
> Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
>=20
>=20
> Eric,
>=20
> The draft talks about sending the subscribe requests on the=20
> same dialog=20
> as the INVITE. For many reasons that have been discussed ad=20
> nauseam on=20
> the list, dialog sharing is a bad idea. GRUU provides a solution that=20
> does not require dialog sharing. As I recall, there was discussion on=20
> this point in the last ietf, and this seemed to be the=20
> preferred direction.
>=20
> Is there any reason we cant change this to use gruu's rather=20
> than dialog=20
> sharing?
>=20
> Also, you are missing the IANA registration for the kpml=20
> event package.
>=20
> Thanks,
> Jonathan R.
>=20
> Internet-Drafts@ietf.org wrote:
>=20
> > A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> > This draft is a work item of the Session Initiation=20
> Proposal Investigation Working Group of the IETF.
> >=20
> > 	Title		: Keypad Stimulus Protocol (KPML)
> > 	Author(s)	: E. Burger
> > 	Filename	: draft-ietf-sipping-kpml-02.txt
> > 	Pages		: 42
> > 	Date		: 2004-2-4
> > =09
> > The Key Press Stimulus Protocol uses the SIP SUBSCRIBE/NOTIFY
> >    mechanism and Keypad Markup Language (KPML) to provide=20
> instructions
> >    to SIP User Agents for the reporting of user key presses.
> >=20
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-02.txt
> >=20
> > To remove yourself from the IETF Announcement list, send a=20
> message to=20
> > ietf-announce-request with the word unsubscribe in the body=20
> of the message.
> >=20
> > Internet-Drafts are also available by anonymous FTP. Login=20
> with the username
> > "anonymous" and a password of your e-mail address. After logging in,
> > type "cd internet-drafts" and then
> > 	"get draft-ietf-sipping-kpml-02.txt".
> >=20
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html=20
> > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >=20
> >=20
> > Internet-Drafts can also be obtained by e-mail.
> >=20
> > Send a message to:
> > 	mailserv@ietf.org.
> > In the body type:
> > 	"FILE /internet-drafts/draft-ietf-sipping-kpml-02.txt".
> > =09
> > NOTE:	The mail server at ietf.org can return the document in
> > 	MIME-encoded form by using the "mpack" utility.  To use this
> > 	feature, insert the command "ENCODING mime" before the "FILE"
> > 	command.  To decode the response(s), you will need "munpack" or
> > 	a MIME-compliant mail reader.  Different MIME-compliant=20
> mail readers
> > 	exhibit different behavior, especially when dealing with
> > 	"multipart" MIME messages (i.e. documents which have been split
> > 	up into multiple messages), so check your local documentation on
> > 	how to manipulate these messages.
> > 	=09
> > 	=09
> > Below is the data which will enable a MIME compliant mail reader
> > implementation to automatically retrieve the ASCII version of the
> > Internet-Draft.
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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> This is certainly not the intension.  The draft specifies a way to separately
> subscribe for device or user profile data.  I would be happy to clarify this
> where you think the draft is vague or implies this restriction.

> 
> Section 3.2 describes parameters that are device specific as even in the
> ideal world, purely user specific data may need to consider the UA
> implementation that it is to be applied to.  Section 3.2 also specifies a
> means to indicate the desire for user or device data.
> 
> Section 3.13 specifies how to form the SUBSCRIBE URI based upon
> the interest for either a device or user profile.

3.13 and 4.1 don't really spell out that user (as opposed to device) 
data may need to go elsewhere. It only has a single set of steps that 
aborts as soon as (say) DHCP-based discovery succeeds.

I think the two modes, user and device, are somewhat dissimilar. For 
device, I may need to combine local information (e.g., information about 
outbound proxies and local dialplans) with home information (local 
dialplan, device-specific UI configuration, etc.).

For user configuration, I probably need to assume that the device has an 
AOR, either initially or discovered via device configuration, which is 
realistic for laptops and other personal devices.

4.1 does not seem to spell out what to do if either the device or the 
user knows the AOR, but wants to get the remainder of the device and 
user configuration automatically.

Thus, I'd suggest separating the two discovery models and possibly 
logically separating user configuration as being derived (among other 
sources of AORs) from the device configuration, as in:

DHCP -> device configuration query -> AOR -> user-based query



> These 3 important concepts are intended to be supported in
> draft-ietf-sipping-config-framework-02.  It is however difficult in
> practice to keep these three things separate.  In many environments
> it is easier to keep two of the three separate.
> 
> For example:
> In an enterprise with physical devices that do not move much, it becomes
> easier to combine the properties of 1 related to location with the properties
> of 3.

I'd rather be more general; if the user configuration spans multiple 
devices, it's a lot cleaner if I only have to update one server rather 
than each device-based server. (Device-based servers may well be more 
local than user-based servers.)


> 
> When a user does not know their AOR or does not want to have to provide
> it to the UA (to provide a plug and play experience), the user's AOR can be
> associated with 3 (the device).  2 is may then be subscribe implicitly or explicitly.

In my two-stage model, this is supported.


> I think that this illustrates a very good point that the separation between
> user and device profile data cannot be made concrete.  Data such as
> out bound proxy, or policy may be dictated by any or all of the 3
> realms defined above.  Only the application and local administrative
> policies can determine the realm to which data should be associated.
> In addition it may be desirable to override some of the user or device
> profile data based upon other local discovery data (e.g. DHCP options
> for out bound proxy, STUN for NAT traversal).

There may well be information that could be considered both, but in my 
book, user-based configuration is independent of where I plug in my 
device and does not depend on the device itself. Thus, an outbound proxy 
and related transport parameters are clearly device-related, while speed 
dial functionality, media preferences, home dial plans and buddy lists 
are probably not.

Some parameters may be hierarchical, in that local network configuration 
overrides my general configuration. If my visited network doesn't allow 
video, that takes precedence over my user configuration that prefers 
video calls.


> 
> At best we can define a precedence by which the device data
> can be overridden by user data (or the other way around)
> along with the ability for administrators to dictate policy (perhaps indicated
> in the profile data) over what can be overridden.

I agree. I think we'll have both directions of override. For example, a 
CIA agent user configuration may mandate use of TLS and AES encryption 
and accept that a call will fail in some local environments that don't 
allow this.

Having a sample set of configuration parameters that we can work through 
might test our assumptions.  draft-stredicke-sipping-ep-config-00.txt 
seems to be the last one on this topic, or is there something else?

It unfortunately conflates local parameters (e.g., outbound proxy, 
3.1.4.3) with user-based parameters (registration period, 3.1.4.1).


Henning

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



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Cc: Rohan Mahy <rohan@cisco.com>, sipping <sipping@ietf.org>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] updated dialog package
Date: Mon, 23 Feb 2004 18:54:47 -0800
To: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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Hi,

On Feb 23, 2004, at 5:43 AM, Venkatesh Venkataramanan wrote:

> Resending the same as my first posting somehow was empty in the 
> mailing list !
>
> From a shared line application perspective:
>
> There is a piece missing from a implementation perspective for shared 
> line appearances.
>
>    - Ability to signal Hold. I know there have been a few discussions 
> in the SIPPING mailing list around this and it there was no conclusion 
> about what "hold" actually is ! However, one of the important aspects 
> of shared lines is the ability to communicate a held call to the other 
> parties in the group. Currently the anil-sipping-bla draft uses the 
> SDP information to help other UA's decide a dialog is transitioning to 
> hold and off hold. It would be definitely useful to have the dialog 
> state express this state to the far end rather than having to send out 
> SDP information.

I'm sorry.  I dropped the ball on this.  I was going to propose a new 
callee-caps style "media feature parameter" called "interactive".  I 
think this satisfies the requirement for what folks are actually 
interested in for "hold", but I need to send out the proposal tomorrow 
to verify this with everyone on the list.

> For the purpose of SLA, instead of going down defining what is Hold, I 
> was wondering if we can create  a state/or an attribute in the dialog 
> state package say "pickable" dialog to indicate the same. I've heard 
> comments saying a UA can block relevant fields of a dialog whenever it 
> wants others to be able to pick or not to pick a dialog, however the 
> same is not clean

why isn't this clean?

> and it breaks when say the same UA that put the dialog on Hold, 
> retrieves the dialog. Nothing in the dialog identifiers have changed.

I don't see why this is a problem.  Also, if I'm already in a dialog, 
the implicit filtering of the dialog package prevent me from receiving 
notifications about state that I already know about directly.  Not 
exactly sure what you mean.

In any case, lets take two cases, cooperating and non-cooperating UAs.  
Cooperating UAs can use the presence or existence of dialog identifiers 
in notifications that they receive to convey that a call is pickable by 
them. Non-cooperating UAs that happen to learn the dialog identifiers 
but aren't authorized, can try to pick calls all they want, but will 
just get 403 Forbidden responses for their requests.

thanks,
-rohan


> The anil-sipping-bla-draft covers other various aspects of SLA from an 
> application implementation perspective with call flow examples.
>
> regards,
> Venkatesh
>
> 	-----Original Message-----
> 	From: sipping-admin@ietf.org on behalf of Rohan Mahy
> 	Sent: Thu 2/19/2004 9:59 AM
> 	To: sipping
> 	Cc: rohan@cisco.com
> 	Subject: [Sipping] updated dialog package
> 	
> 	
>
> 	Hi Everyone,
>
> 	I've updated the dialog packages with the changes we agreed to in
> 	Minneapolis (removing CSeq and the Route-Set).
>
> 	I've also provided a detailed example of using the dialog package for 
> a
> 	"shared-line" application.  I apologize for my tardiness here.  I was
> 	supposed to post something shortly after the meeting.  I any case, I
> 	believe this demonstrates that the current syntax is at least
> 	sufficient for the use cases we've discussed.  I would like to hear
> 	from all of you if this does the trick.
>
> 	many thanks,
> 	-rohan
>
>
> 	_______________________________________________
> 	Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> 	This list is for NEW development of the application of SIP
> 	Use sip-implementors@cs.columbia.edu for questions on current sip
> 	Use sip@ietf.org for new developments of core SIP
>


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



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Cc: Rohan Mahy <rohan@cisco.com>, "sipping" <sipping@ietf.org>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] updated dialog package
Date: Mon, 23 Feb 2004 19:14:24 -0800
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On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:

> Hi,
>
> I work for a company that has developed handset gateway devices that
> enable legacy, digital, business-grade telephones to be driven using
> SIP.  For example, this enables an existing legacy PBX user to retain
> their telephones and cabling, but to replace the PBX by a Softswitch or
> IP Centrex system.  As most business-grade telephone systems make use 
> of
> a shared line facility, the addition of a shared line example to the
> dialog event package ID is of great interest to us, and others who
> support these types of feature.
>
> Although incorporating a shared line example is a step towards a common
> approach to shared line implementation, I don't believe it goes far
> enough.  Our current implementation of shared lines has shown that the
> following issues need to be addressed by the spec, and should be shown
> in the examples:
>
> o Line seize, an essential part of any shared line implementation, is
> not adequately described. When more than one user agent attempts to
> seize a line at approximately the same instant, a race condition 
> arises.
> No description of how to handle the race is given - an example
> application/dialog-info+xml NOTIFY payload is provided, but this alone
> is inadequate, as message sequencing is an important part of seizing a
> line.  The Line Seize condition will be covered in the
> "anil-sipping-bla-draft" that Venkatesh refers to.

Line seizure is well beyond what would be appropriate to standardize in 
SIPPING because for many types of UAs that concept does not even exist. 
  I was just trying to demonstrate a non-normative description of how it 
could be implemented. How you judge who "wins" a seizure race condition 
is an implementation specific issue.  In fact, its quite likely that 
different behaviors will be desired by different customers.  When 
talking about implementing this at Cisco, we concluded that a 
configuration knob was needed to satisfy essentially opposite 
requirements from two different customers.  Each customer was equally 
adamant that he was "right" and the other was "wrong".  As a result, I 
find it unlikely that SIPPING could come to come to consensus on what 
"proper seizing behavior" is even if we ignored the fact that the bulk 
of deployed SIP UAs are software UAs which have no lines to seize.

> o Handling of multiple instances of a shared line has been overlooked.
> On business-grade PBX systems a line will roughly correspond to an AOR.
> However, many such systems permit multiple "instances" of a shared line
> to appear on a single endpoint. For example, a number of instances of a
> "product support" shared line might appear on all the endpoints of the
> product support group, thereby allowing multiple calls to be taken and
> passed around within the group.

This has not been overlooked at all.  You can have multiple 
subscriptions (one per AOR), or you can subscribe to a list of AORs 
with the event list extension:

	http://www.ietf.org/internet-drafts/draft-ietf-simple-event-list-04.txt

> o The mechanism used to convey that a call has been placed on hold
> appears to rely upon implementation-specific parameters, found in the
> xml's <param> element. I can't see any reference to definitions of the
> values used by this element's pname and pval attributes
> (pname="+activity" pval="noninteractive"), so I assume they're
> non-standard.

This was due to distraction on my part.  I had intended to include a 
discussion of this in the current draft, but somehow left it out at 2am 
the day of the draft deadline.  I'm terribly sorry about that.  I'll 
send out more about the hold issue tomorrow.

thanks,
-rohan

> Thanks,
>
> Paul.
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@cisco.com]
>> Sent: Thursday, February 19, 2004 6:00 PM
>> To: sipping
>> Cc: rohan@cisco.com
>> Subject: [Sipping] updated dialog package
>>
>>
>> Hi Everyone,
>>
>> I've updated the dialog packages with the changes we agreed to in
>> Minneapolis (removing CSeq and the Route-Set).
>>
>> I've also provided a detailed example of using the dialog
>> package for a
>> "shared-line" application.  I apologize for my tardiness here.  I was
>> supposed to post something shortly after the meeting.  I any case, I
>> believe this demonstrates that the current syntax is at least
>> sufficient for the use cases we've discussed.  I would like to hear
>> from all of you if this does the trick.
>>
>> many thanks,
>> -rohan
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>


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



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Date: Tue, 24 Feb 2004 14:49:51 +0900
From: Kumiko Ono <ono.kumiko@lab.ntt.co.jp>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping <sipping@ietf.org>
Cc: volkerh@bell-labs.com, tachimoto.shinya@lab.ntt.co.jp
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Subject: [Sipping] Re: E2M security solution
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Hi,

I agree with you about the e2m security solution could partly solve the 
session-specific policies. 

In my understanding, the policy draft proposes new headers to disclose 
only the information to *all* proxies in the path. Some mechanism are 
needed in order to disclose the information to *specific* proxies. The 
e2m security can be applied as one of the mechanisms.

Regards,

Kumiko

>Hello,
>
>The e2m security solution, described in the draft below,
>http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-
>01.txt
>
>attempts to resolve some of the problems that the session-specific 
>policies draft addresses:
>http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-spec-policy-
>00.txt
>
>The need of a UA to obtain the policies that a particular proxy has is 
>dealt with in both drafts. In addition, the e2m security draft talks 
>about disclosing information to intermediaries. The policy draft 
>addresses this issue and defines a means to disclose *only* the 
>information needed by the proxy.
>
>My comments on the e2m requirements draft apply also to this one. We 
>should try to align both efforts (session policies and e2m security).
>
>Thanks,
>
>Gonzalo
>
>
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--
Kumiko Ono
NTT Network Service Systems Labs.
Tel +81 422 59 4508/Fax +81 422 60 4012


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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Cullen Jennings <fluffy@cisco.com>
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        Eric Burger <eburger@snowshore.com>
Subject: Re: [Sipping] SDP in draft-ietf-sipping-transc-3pcc-00
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Cullen,

inline:

Cullen Jennings wrote:
> SDP is a scary place so using SDP to separate the SDP bodies for A and B is
> scary. Consider the following use case:
> 
> I have a transcoder that takes a audio from a UA connected to a microphone
> and another camera UA that sends video. I am going to feed both of these to
> a transcoder and produce a single mpeg output stream that contains both
> audio video. Here I need so specify that the input SDP has two IP addresses
> and the output has one. This would be even more complicated if one UA did
> the offer/ans in INV/200 while the other used the 200/ACK.
> 
> Also related to all this, would be nice to be confident that pre conditions
> stuff worked while using all this.
> 
> Anyways, I like the overall solution to all this stuff. I wish there was a
> better solution to separating the A+B, TA+TB information to the transcoder
> than adding more semantics to SDP.

Previous version of this draft use to provide this information using SDP 
attributes, which were defined in the draft below:

http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-source-sink-01.txt

Nevertheless, the feedback we got from, among others, MMUSIC and 
SIPPING, was that we did not need those parameters. The Request-URI will 
indentify the service the transcoder provides, making unnecessary more 
information.

In your example, you would have a URI that indentifies the service 
(taking both inputs and creating the mpeg).

Thanks,

Gonzalo


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From exim@www1.ietf.org  Tue Feb 24 03:38:24 2004
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Subject: [Sipping] UA indentification and HIP
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Folks,

the following two drafts describe how to indentify a particular UA using 
globally unique identifiers:

http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt
http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt

Cullen pointed out that this work may be able to take advantage of the 
work carried out in the HIP (Host Identity Protocol) WG, which 
supplemental web page is:

http://hip.piuha.net/

Here you have a *very* brief introduction to HIP. I will also be 
providing a brief introduction to the drafts above to the HIP community. 
Hopefully, we will have a useful exchange of ideas between both WGs.

HIP consists of a layer between the network and the tranport layers 
(e.g., between IP and TCP). It uses HIs (Host Identifiers) to identify 
hosts. The idea is that, when a host changes its IP address, it keeps 
its HI. This facilitates mobility, for instance.

HIs (Host Identifiers) are public keys, which may have different lengths 
and formats. To resolve this, HIP defines the HITs (Host Identity Tags), 
which are hashes of the HI. That is, a HIT is a hash of a public key. 
They are 128 bit long, and can be used instead of IPv6 addresses in 
socket calls. HIP also defines LSIs (Local Scope Indetifiers), that are 
32 bit long and can be used instead of IPv4 addresses in socket calls.

Regards,

Gonzalo


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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] ToIP Requirements
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Hello,

the draft below provides a set of requirements for real-time 
conversations using SIP.
http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-toip-01.txt

Nevertheless, I have found a lot of device requirements in the text, but 
I fail to understand which particular protocol requirements we need to 
fulfill with new protocol extensions.

While SIPPING is the right place to handle requirements that will lead 
to new SIP protocol extensions, SIPPING may not be the right place to 
deal with device requirements (e.g., text phones MUST have a display of 
at least 100 characters).

Thanks,

Gonzalo


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Hi All.
=20
Sorry if this is a much asked and much answered question, but haven't
the basic problems being solved in a "Roaming Architecture for SIP",
i.e. allowing a SIP host to roam to foreign networks, already been
solved using Mobile IP? What else does "Roaming Architecture for SIP"
give me?
=20

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



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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Steve Langstaff'" <steve.langstaff@citel.com>, <sipping@ietf.org>
Subject: AW: [Sipping] Roaming Architecture for SIP vs Mobile IP
Date: Tue, 24 Feb 2004 11:27:08 +0100
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Dear Steve,

I do not think one can always assume the existence of Mobile IP on SIP
UAs,
in fact I have yet to see one instance of such a combination - unless
custom-made.
While MIP is a good concept in theory, it is less than widely spread and
cannot be assumed as given in any standard application or IP stack.
Furthermore - depending on the exact configuration - MIP introduces
additional latencies, which is something to be avoided for VoIP.
You are however correct in stating that MIP is one approach to solving
roaming issues. I believe its most obvious application seems to be
WLAN/GSM/3GPP, i.e. layer 2 or layer 3 rather than layer 5-7.
The focus on this draft is to have a pure SIP-based approach,
specifically geared towards VoIP.
As you are probably aware, we are still far away from having completely
vertically integrated AAA infrastructures - though an NAI-based approach
could eventually lead there.

Regards,

	Harry


> -----Urspr=FCngliche Nachricht-----
> Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
> Im Auftrag von Steve Langstaff
> Gesendet: Dienstag, 24. Februar 2004 10:31
> An: sipping@ietf.org
> Betreff: [Sipping] Roaming Architecture for SIP vs Mobile IP
>=20
>=20
> Hi All.
> =20
> Sorry if this is a much asked and much answered question, but=20
> haven't the basic problems being solved in a "Roaming=20
> Architecture for SIP", i.e. allowing a SIP host to roam to=20
> foreign networks, already been solved using Mobile IP? What=20
> else does "Roaming Architecture for SIP" give me?
> =20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20


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



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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Cc: <sipping@ietf.org>, <knuettel@fokus.fraunhofer.de>
Subject: AW: [Sipping] Roaming Architecture for SIP
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Dear Paul,

inline

> -----Urspr=FCngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Gesendet: Montag, 23. Februar 2004 19:47
> An: Harry Behrens
> Cc: sipping@ietf.org; knuettel@fokus.fraunhofer.de
> Betreff: Re: [Sipping] Roaming Architecture for SIP
>=20
>=20
> I have specific comment inline below, and then a question.
>=20
> 	Paul
>=20
> >   -  A user registers with the VN (atlanta.com in the call flow=20
> >       given). The user submits his NAI in new header field 'P-NAI'.=20
> >       The user submits a bogus URI in both the From and the=20
> To header.=20
> >       (F1 in section 8.2.1)
> >   -  The VN recognizes the registration as a P-NAI registration and=20
> >       sends back a "401 Unauthorized" response, with both=20
> the 'From'=20
> >       and the 'To' URI set to the temporary URI that will=20
> be assigned=20
> >       to the user upon successful authentication and authorization=20
> > (F2).
>=20
> This isn't legal. The From and To (with exception of To-tag)=20
> of response=20
> must be same as in the request.
>=20

This is true. The reason we have this crutch in there is to allow the VN
server to pass a user ID to the client, which the client can use to
register to the VN. This would otherwise have to happen out-of-band,
which we were trying to avoid.
We are thinking about alternative approaches to this requirement and are
grateful for any input.

> The question is: why would the UAC want to register with the VN?=20
> Assuming it is able to send sip messages, (which apparently=20
> it can), why=20
> wouldn't it simply send a REGISTER to its HN?
>=20
> I presume there must be some service that you intend the VN won't=20
> perform, that will prevent the UAC from doing this. But it isn't=20
> apparent what service that is.
>=20

The scenario is of a user based e.g. in Berlin/Germany on a business
trip in Beijing/China.
The user will want to both be reachable under his home AOR (for that he
registers with his HN) as well as make use of the VN for local calls,
especially local PSTN breakout (this is why he wants to have a valid
registration with the VN).
Furthermore, in a number of cases, one encounters RTP-proxying, where
around-the-world loops are to be avoided.
Another issue which we foresee is that in the near future hotels,
airports or other semi-closed (walled garden) Internet access points,
will force users through their own VoIP (SIP) proxy infrastructure,
similar to what a lot of sites aleady do in the case of HTTP.


	Harry

> 	Paul
>=20


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



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Hello,

some comments on the following draft:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-03.txt

The focus of a conference returns the conference URI in the contact 
header field of the 200 OK for the INVITE. The UAC uses this URI to 
SUBSCRIBE to the conf package (using a new dialog)...

Why are we forcing the focus to use the same URI for receiving the 
INVITEs from all the users and for receiving the SUBSCRIBEs? I believe 
the solution would be cleaner if we defined a new header field (e.g., 
Associated-Conf-State) that carried a globally routable URI. The 200 OK 
for the initial INVITE (or the INVITE in a dial-out case) would carry 
this header field. This would keep separate the URI used for joining the 
conference and the URI used for subscribing to the state of the conference.

This is actually the approach that the draft below follows:
http://www.ietf.org/internet-drafts/draft-camarillo-sipping-transac-package-00.txt

We define the Associated-Transactions-State to be returned, for example, 
in the 200 OK of a MESSAGE sent to an exploder. Note that we had to 
create this new header because MESSAGE does not carry a Contact 
header... following the same approach in both, the conferencing work and 
the exploders work would be good, in my opinion.

Comments?

Thanks,

Gonzalo


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From: Miguel.An.Garcia@nokia.com
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Date: Tue, 24 Feb 2004 14:11:20 +0200
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Thread-Topic: [Simple] Advanced IM requirements
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To: <cboulton@ubiquity.net>, <simple@ietf.org>, <sipping@ietf.org>
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Subject: [Sipping] RE: Advanced IM requirements
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A comment on the usage of the Require header in the MESSAGE receipt =
draft.

As far as I know, the Require header is used to indicate the peer that =
the request should not be processed if the extension is not supported.

You apply the Require header to the response... which is a bit weird, =
because you tell me what should be the action of the UAC if it does not =
support the functionality.

You can argue that we have already used Require headers in SIP responses =
(e.g.., 100rel). But there it had a clear meaning: there is an action =
that has to be taken upon the reception of the response: to generate a =
PRACK. But in your draft, I didn't see an equivalence.

In my view, this mechanism should only be applicable to Support headers =
in requests. If the message is no delivered, then a DSN can be =
generated, because of the Support header tag in the request.

And that comes to my next question. Do you intend proxies to generate =
DSNs to non delivered messages? I guess this is the idea, but I didn't =
find a "Proxy behavior" section in the document.

/Miguel



> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Chris Boulton
> Sent: 23 February, 2004 19:01
> To: Simple WG; sipping
> Subject: [Sipping] FW: [Simple] Advanced IM requirements
>=20
>=20
>=20
>=20
> As promised.
>=20
> https://www.ubiquity.net/draft-boulton-sipping-message-receipt-00.txt
>=20
> Chris.
>=20
>=20
> >-----Original Message-----
> >From: Chris Boulton
> >Sent: 23 February 2004 10:40
> >To: 'Jonathan Rosenberg'; Simple WG
> >Cc: 'sipping@ietf.org'
> >Subject: RE: [Simple] Advanced IM requirements
> >
> >Hi Jonathan et al,
> >
> >	I have an extremely early version (so be gentle) of a draft
> aimed at
> >supplying IM delivery reports.  It hasn't been formally submitted yet
> but I
> >have attached a copy - so would be grateful if the list could take a
> look
> >and see if it's worth progressing.  I will submit to the archives in
> the
> >next few days.  I will also put this on the internet=20
> somewhere in case
> this
> >attachment doesn't survive its journey (Will send link later).
> >
> >Best Regards,
> >
> >Chris.
> >
> >
> >>-----Original Message-----
> >>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> >>Sent: 22 February 2004 06:17
> >>To: Simple WG
> >>Subject: [Simple] Advanced IM requirements
> >>
> >>Folks,
> >>
> >>As I noted earlier, I resubmitted the advanced IM=20
> requirements draft.
> It
> >>specifies a bunch of advanced IM functions we havent yet=20
> tackled here.
> >>They are:
> >>
> >>* delivery reports for IM
> >>* is-typing indicator (Henning had an I-D on this that has expired)
> >>* capabilities indications of maximum desired message sizes for IM
> >>* page mode groups (under discussion in the context of exploders in
> >>sipping)
> >>* invitations to non-real time sessions
> >>
> >>
> >>The last of these is particularly interesting. One use case is "call
> >>alerts" in PTT. There, I send you a request to have a PTT chat. The
> >>request is not answered in real time. Rather, its stored by the
> >>recipient as if it were an IM, and can be answered at any=20
> time later.
> >>"Answering it" merely involves setting up a PTT call back to the
> caller.
> >>Its also possible to cancel the call alert at any time.
> >>
> >>There are some interesting ways to accomplish this function. The one
> >>I've been thinking about is taking a SIP INVITE message, and sending
> it
> >>in the body of a MESSAGE request as a message/sip content. I'm sure
> >>there are other ways.
> >>
> >>Are these topics for which there is still group interest in working
> on?
> >>I think all of them remain quite important.
> >>
> >>Thanks,
> >>Jonathan R.
> >>--
> >>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> >>Chief Technology Officer                    Parsippany, NJ=20
> 07054-2711
> >>dynamicsoft
> >>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> >>http://www.jdrosen.net                      PHONE: (973) 952-5000
> >>http://www.dynamicsoft.com
> >>
> >>
> >>_______________________________________________
> >>Simple mailing list
> >>Simple@ietf.org
> >>https://www1.ietf.org/mailman/listinfo/simple
>=20
>=20
> This message has been scanned for viruses by MailControl -=20
www.mailcontrol.com

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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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



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Thread-Topic: Conferencing for UAs
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Gonzalo:

The idea is fine, but I am afraid we may end up defining =
Associated-Transaction-State, Associated-Conference-State, =
Associated-XXX-State.

Aren't we heading to defining a generic Associated-State header with a =
"role" parameter that contains a value "transaction", "conference", =
"xxx", etc...

/Miguel

> -----Original Message-----
> From: ext Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: 24 February, 2004 14:06
> To: sipping
> Cc: Alan Johnston; Orit Levin; Garcia Miguel.An (Nokia-NRC/Helsinki)
> Subject: Conferencing for UAs
>=20
>=20
> Hello,
>=20
> some comments on the following draft:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conf
> erencing-03.txt
>=20
> The focus of a conference returns the conference URI in the contact=20
> header field of the 200 OK for the INVITE. The UAC uses this URI to=20
> SUBSCRIBE to the conf package (using a new dialog)...
>=20
> Why are we forcing the focus to use the same URI for receiving the=20
> INVITEs from all the users and for receiving the SUBSCRIBEs?=20
> I believe=20
> the solution would be cleaner if we defined a new header field (e.g.,=20
> Associated-Conf-State) that carried a globally routable URI.=20
> The 200 OK=20
> for the initial INVITE (or the INVITE in a dial-out case) would carry=20
> this header field. This would keep separate the URI used for=20
> joining the=20
> conference and the URI used for subscribing to the state of=20
> the conference.
>=20
> This is actually the approach that the draft below follows:
> http://www.ietf.org/internet-drafts/draft-camarillo-sipping-tr
ansac-package-00.txt

We define the Associated-Transactions-State to be returned, for example, =

in the 200 OK of a MESSAGE sent to an exploder. Note that we had to=20
create this new header because MESSAGE does not carry a Contact=20
header... following the same approach in both, the conferencing work and =

the exploders work would be good, in my opinion.

Comments?

Thanks,

Gonzalo


This communication is confidential and intended solely for the =
addressee(s). Any unauthorized review, use, disclosure or distribution =
is prohibited. If you believe this message has been sent to you in =
error, please notify the sender by replying to this transmission and =
delete the message without disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption, =
interruption, unauthorized amendment, tampering and viruses, and we only =
send and receive e-mails on the basis that we are not liable for any =
such corruption, interception, amendment, tampering or viruses or any =
consequences thereof.


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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi,

> Aren't we heading to defining a generic Associated-State header with
> a "role" parameter that contains a value "transaction", "conference",
> "xxx", etc...

I actually do not care how we encode this information; in a header with 
several parameters or in multiple headers (in any case, you may be right 
that using a single header could be cleaner).

What I want to know is whether or not people think that it is worthwhile 
keeping separate the URI used for the INVITE dialog and the one (or 
ones) used for subscriptions to some piece of state that is associated 
somehow with that dialog. My personal opinion is that we should keep 
them separate.

Gonzalo


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Subject: RE: [Sipping] Conferencing for UAs
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From: "Chris Boulton" <cboulton@ubiquity.net>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>,
        "sipping" <sipping@ietf.org>
Cc: "Alan Johnston" <alan.johnston@mci.com>,
        "Orit Levin" <oritl@microsoft.com>,
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>-----Original Message-----
>From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
>Sent: 24 February 2004 12:06
>To: sipping
>Cc: Alan Johnston; Orit Levin; Miguel Angel Garcia Martin
>Subject: [Sipping] Conferencing for UAs
>
>Hello,
>
>some comments on the following draft:
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-
>03.txt
>
>The focus of a conference returns the conference URI in the contact
>header field of the 200 OK for the INVITE. The UAC uses this URI to
>SUBSCRIBE to the conf package (using a new dialog)...
>
>Why are we forcing the focus to use the same URI for receiving the
>INVITEs from all the users and for receiving the SUBSCRIBEs? I believe
>the solution would be cleaner if we defined a new header field (e.g.,
>Associated-Conf-State) that carried a globally routable URI. The 200 OK
>for the initial INVITE (or the INVITE in a dial-out case) would carry
>this header field. This would keep separate the URI used for joining
the
>conference and the URI used for subscribing to the state of the
conference.

[Chris Boulton] I agree that it would be better to convey this
information separately for instances where the presence module is a
separate entity to the conference focus.




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From: "A vWijk" <A.vWijk@viataal.nl>
To: <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <gunnar.hellstrom@omnitor.se>, <paulej@packetizer.com>,
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Subject: Betr.: [Sipping] ToIP Requirements
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Hello Gonzalo,
While there are device requirements in the text. Those requirements and =
the explanation on the different analogue text telephone systems is =
included to give the reader an understanding what kind of devices will =
connect with ToIP. And the need for the text gateways. And what the =
different behaviour can/may be compared to end to end IP devices.
The most important part will be the ToIP framwork and the text gateways =
and how such gateways will be used and work with the ToIP framework.

There is a ToIP framework where a terminal uses streaming text as defined =
in RFC2793 to connect with any other Interactive texting/real-time =
streaming text device.
This can be end-to-end IP, and a direct T140/RTP stream is established =
with SIP. But in other cases, a text gateway must be used to allow =
T.140/RTP to connect with a non IP text telephony device.
It can be an analogue text telephone (TDD/TTY) that uses a text gateway to =
be able to understand RFC2793 text. (PSTN-IP)
It can be a mobile phone running GPRS that uses the text gateway to =
understand RFC2793 text. (this exist right NOW using SIP! And those =
gateways will be used in mobile networks this year!)
Or a mobile text telephone using CTM via the text gateway to communicate =
with RFC2793 text.
Or even one analogue text telephone connecting to another analogue text =
telephone in a different country, and the PSTN's use IP to connect the =
call.

The question is for example, how do we use that gateway from the IP side? =
Is it invisible for the IP side? Or do we need special extensions in SIP =
to that the IP terminal knows it is communicating with the end-device =
through the text gateway?
There may be a slightly different call setup flow when a text gateway is =
included.
Or additional SIP messages indicating gateway activity. When you connect =
for example to an analogue text telephone with an RFC2793 capable IP =
terminal the capabilities will be less then IP to IP terminal directly =
(different transmission speeds, half/full duplex etc).
Or what to do with emergency calls that are in RFC 2793 text only?=20

That are requirements that will be defined and worked out in this draft.
Also, we want to make everybody in SIP and SIPPING aware about those text =
gateways. So that complicated matters like how to send analogue text =
telephone text through a PSTN and connect with a SIP telephone using =
RFC2793 interactive texting. When using the text gateway, there is then no =
need for RFC 2833 events at the PSTN for text telephones.=20

So, yes. I believe that this should be in the SIPPING WG. And it will fit =
in the charter - support for hearing-/speech-impaired calling.

greetz

Arnoud



Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

>>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 24-02-04 10:21 >>>
Hello,

the draft below provides a set of requirements for real-time=20
conversations using SIP.
http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-toip-01.txt=20

Nevertheless, I have found a lot of device requirements in the text, =
but=20
I fail to understand which particular protocol requirements we need to=20
fulfill with new protocol extensions.

While SIPPING is the right place to handle requirements that will lead=20
to new SIP protocol extensions, SIPPING may not be the right place to=20
deal with device requirements (e.g., text phones MUST have a display of=20
at least 100 characters).

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s)=
. Any unauthorized review, use, disclosure or distribution is prohibited. =
If you believe this message has been sent to you in error, please notify =
the sender by replying to this transmission and delete the message without =
disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption, interruptio=
n, unauthorized amendment, tampering and viruses, and we only send and =
receive e-mails on the basis that we are not liable for any such corruption=
, interception, amendment, tampering or viruses or any consequences =
thereof.


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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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Thread-Topic: Advanced IM requirements
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From: "Chris Boulton" <cboulton@ubiquity.net>
To: <Miguel.An.Garcia@nokia.com>, <simple@ietf.org>, <sipping@ietf.org>
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Subject: [Sipping] RE: Advanced IM requirements
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Miguel - thanks for taking the time to read this - I have included a
couple of comments in-line.


>-----Original Message-----
>From: Miguel.An.Garcia@nokia.com [mailto:Miguel.An.Garcia@nokia.com]
>Sent: 24 February 2004 12:11
>To: Chris Boulton; simple@ietf.org; sipping@ietf.org
>Subject: RE: Advanced IM requirements
>
>A comment on the usage of the Require header in the MESSAGE receipt
draft.
>
>As far as I know, the Require header is used to indicate the peer that
the
>request should not be processed if the extension is not supported.
>
>You apply the Require header to the response... which is a bit weird,
>because you tell me what should be the action of the UAC if it does not
>support the functionality.
>
>You can argue that we have already used Require headers in SIP
responses
>(e.g.., 100rel). But there it had a clear meaning: there is an action
that
>has to be taken upon the reception of the response: to generate a
PRACK.
>But in your draft, I didn't see an equivalence.
>
>In my view, this mechanism should only be applicable to Support headers
in
>requests. If the message is no delivered, then a DSN can be generated,
>because of the Support header tag in the request.

[Chris Boulton] Seems reasonable to me.  This is an early version and I
will note this as a point for my discussions in Korea.  My intention was
to encourage a stateless mechanism that allowed for state-full
processing if required when used with a GRUU.  In that if a UAC issued a
MESSAGE with a GRUU + grid parameter, it would need state for the grid
association on the incoming receipt(Even though this is discouraged).
So the intended action for the UAC was to remain state-full for that
particular grid parameter - otherwise dispose of state (But I agree -
not a SIP action).

>
>And that comes to my next question. Do you intend proxies to generate
DSNs
>to non delivered messages? I guess this is the idea, but I didn't find
a
>"Proxy behavior" section in the document.

[Chris Boulton] In this initial draft, the mechanism only applies to
successful MESSAGE requests (2xx class recpose).  It covered scenarios
where the message had arrived at a store-forward device OR delivered to
the handset but had not been read etc.  Again, I am interested in the
lists thoughts on this so please let me know.  I will also note this
down for discussion.

Chris.


>
>/Miguel
>
>
>
>> -----Original Message-----
>> From: sipping-admin@ietf.org
>> [mailto:sipping-admin@ietf.org]On Behalf Of
>> ext Chris Boulton
>> Sent: 23 February, 2004 19:01
>> To: Simple WG; sipping
>> Subject: [Sipping] FW: [Simple] Advanced IM requirements
>>
>>
>>
>>
>> As promised.
>>
>> https://www.ubiquity.net/draft-boulton-sipping-message-receipt-00.txt
>>
>> Chris.
>>
>>
>> >-----Original Message-----
>> >From: Chris Boulton
>> >Sent: 23 February 2004 10:40
>> >To: 'Jonathan Rosenberg'; Simple WG
>> >Cc: 'sipping@ietf.org'
>> >Subject: RE: [Simple] Advanced IM requirements
>> >
>> >Hi Jonathan et al,
>> >
>> >	I have an extremely early version (so be gentle) of a draft
>> aimed at
>> >supplying IM delivery reports.  It hasn't been formally submitted
yet
>> but I
>> >have attached a copy - so would be grateful if the list could take a
>> look
>> >and see if it's worth progressing.  I will submit to the archives in
>> the
>> >next few days.  I will also put this on the internet
>> somewhere in case
>> this
>> >attachment doesn't survive its journey (Will send link later).
>> >
>> >Best Regards,
>> >
>> >Chris.
>> >
>> >
>> >>-----Original Message-----
>> >>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>> >>Sent: 22 February 2004 06:17
>> >>To: Simple WG
>> >>Subject: [Simple] Advanced IM requirements
>> >>
>> >>Folks,
>> >>
>> >>As I noted earlier, I resubmitted the advanced IM
>> requirements draft.
>> It
>> >>specifies a bunch of advanced IM functions we havent yet
>> tackled here.
>> >>They are:
>> >>
>> >>* delivery reports for IM
>> >>* is-typing indicator (Henning had an I-D on this that has expired)
>> >>* capabilities indications of maximum desired message sizes for IM
>> >>* page mode groups (under discussion in the context of exploders in
>> >>sipping)
>> >>* invitations to non-real time sessions
>> >>
>> >>
>> >>The last of these is particularly interesting. One use case is
"call
>> >>alerts" in PTT. There, I send you a request to have a PTT chat. The
>> >>request is not answered in real time. Rather, its stored by the
>> >>recipient as if it were an IM, and can be answered at any
>> time later.
>> >>"Answering it" merely involves setting up a PTT call back to the
>> caller.
>> >>Its also possible to cancel the call alert at any time.
>> >>
>> >>There are some interesting ways to accomplish this function. The
one
>> >>I've been thinking about is taking a SIP INVITE message, and
sending
>> it
>> >>in the body of a MESSAGE request as a message/sip content. I'm sure
>> >>there are other ways.
>> >>
>> >>Are these topics for which there is still group interest in working
>> on?
>> >>I think all of them remain quite important.
>> >>
>> >>Thanks,
>> >>Jonathan R.
>> >>--
>> >>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>> >>Chief Technology Officer                    Parsippany, NJ
>> 07054-2711
>> >>dynamicsoft
>> >>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>> >>http://www.jdrosen.net                      PHONE: (973) 952-5000
>> >>http://www.dynamicsoft.com
>> >>
>> >>
>> >>_______________________________________________
>> >>Simple mailing list
>> >>Simple@ietf.org
>> >>https://www1.ietf.org/mailman/listinfo/simple
>>
>>
>> This message has been scanned for viruses by MailControl -
>www.mailcontrol.com
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


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



From exim@www1.ietf.org  Tue Feb 24 08:58:34 2004
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Subject: Re: [Sipping] Re: sos - dead or alive
To: Michael Hammer <mhammer@cisco.com>
Cc: Henning Schulzrinne <hgs@cs.columbia.edu>, Mpierce1@aol.com,
        sipping@ietf.org
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From a different perspective-  and this is a US centric comment but the
principle is applicable worldwide.

In the US, the Government "mandated" E-911  (911 with location info for
mobile phones, among other things) by imposing requirements on the
manufacturers of the switching equipment, and on the service providers.
However, there was no mechanism to require the PSAPs to do anything with
the information, as there was no direct jurisdiction.

As a result, the vast majority of the PSAPs simply "throw it on the floor".
They have no GIS, and no mechanism to DO anything with the location
information.  Furthermore , a large number of the PSAPs have no plans to do
anything with the location information.

Therefore, I would be VERY cautious about basing any solution  on the
expectation, that the PSAPs will make ANY additional investment that is not
required by law.

For instance, when I saw  "$2k per proxy, one proxy per  ECC", it threw up
a red flag for me.

Janet

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                      Michael Hammer                                                                                             
                      <mhammer                 To:      Henning Schulzrinne <hgs@cs.columbia.edu>                                
                      @cisco.com>              cc:      Mpierce1@aol.com, Janet P Gunn/FED/CSC@CSC, sipping@ietf.org             
                                               Subject: Re: [Sipping] Re: sos - dead or alive                                    
                      02/20/2004 10:41                                                                                           
                      AM                                                                                                         
                                                                                                                                 
                                                                                                                                 




Henning,

The reality of deployment would be that every ECC would have it's own
proxy, so that would be more like 1000 x $2k = $2 million.  But, the
corresponding distribution of load across those servers means it is even
more of a non-problem at any given site.

One question:  A real successful test to a wrong ECC doesn't buy you
much.  I hope that mobility is addressed in the solution as well.  Many of
those 100 million phones are mobile, which might become SIP-based some time

in the future.  so I would add a 6) "works with mobile terminals" to your
list below.

Mike

At 10:00 AM 2/20/2004 -0500, Henning Schulzrinne wrote:
>I think there are five goals, somewhat contradictory:
>
>(1) make it as realistic as possible, to make sure you are really testing
>the emergency calling system, not the testing system; after all, the fire
>safety officer doesn't go around ringing a cowbell instead of the real
>fire alarms.
>
>(2) do not impose any additional operator involvement; this is solved by
>the OPTIONS mechanism proposed;
>
>(3) allow for reasonable scaling;
>
>(4) avoid interfering with real emergency calls
>
>(5) allow user on-demand testing.
>
>With the proxy-based testing I proposed, we get very close to all goals.
>In the worst case, i.e., every device does this once a day, we get roughly

>100 million calls a day in the US, or 1157/second. That's about 10 proxy
>servers worth of equipment, i.e., about $20k. This hardly seems worth
>spending inordinate engineering resources on avoiding, particularly since
>the likely load is going to be lower by probably a factor of 1000 or more
>(I'm assuming that most residential users will use a cable or DSL provider

>for this, for example).
>
>With even a few hundred systems, you'll get almost perfect smoothing and
>very little burstiness.
>
>I know protocol design is fun, but I'm very worried about adding
>complexity to solve largely a non-problem.
>






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Subject: RE: [Sipping] Conferencing for UAs
Date: Tue, 24 Feb 2004 16:00:03 +0200
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017977C9@esebe019.ntc.nokia.com>
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> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Gonzalo Camarillo
> Sent: 24.February.2004 14:06
> To: sipping
> Cc: Alan Johnston; Orit Levin; Garcia Miguel.An (Nokia-NRC/Helsinki)
> Subject: [Sipping] Conferencing for UAs
>=20
>=20
>=20
> Hello,
>=20
> some comments on the following draft:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conf
> erencing-03.txt
>=20
> The focus of a conference returns the conference URI in the contact=20
> header field of the 200 OK for the INVITE. The UAC uses this URI to=20
> SUBSCRIBE to the conf package (using a new dialog)...
>=20
> Why are we forcing the focus to use the same URI for receiving the=20
> INVITEs from all the users and for receiving the SUBSCRIBEs?=20
> I believe=20
> the solution would be cleaner if we defined a new header field (e.g.,=20
> Associated-Conf-State) that carried a globally routable URI.=20
> The 200 OK=20
> for the initial INVITE (or the INVITE in a dial-out case) would carry=20
> this header field. This would keep separate the URI used for=20
> joining the=20
> conference and the URI used for subscribing to the state of=20
> the conference.

I don't see what the problem is here and why it is a problem? Why do =
they need to be separate. Both the INVITEs and SUBSCRIBEs are destined =
to the same resource.

You have a resource (in this case a conference ID). You invite yourself =
to that resource. You also subscribe to that resource, using the same =
URI. Note that the Request-URI on the SUBSCRIBE request carries the =
resource you are subscribing to.

/Hisham

>=20
> This is actually the approach that the draft below follows:
> http://www.ietf.org/internet-drafts/draft-camarillo-sipping-tr
> ansac-package-00.txt
>=20
> We define the Associated-Transactions-State to be returned,=20
> for example,=20
> in the 200 OK of a MESSAGE sent to an exploder. Note that we had to=20
> create this new header because MESSAGE does not carry a Contact=20
> header... following the same approach in both, the=20
> conferencing work and=20
> the exploders work would be good, in my opinion.
>=20
> Comments?
>=20
> Thanks,
>=20
> Gonzalo
>=20
>=20
> This communication is confidential and intended solely for=20
> the addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is prohibited. If you believe this message has=20
> been sent to you in error, please notify the sender by=20
> replying to this transmission and delete the message without=20
> disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data=20
> corruption, interruption, unauthorized amendment, tampering=20
> and viruses, and we only send and receive e-mails on the=20
> basis that we are not liable for any such corruption,=20
> interception, amendment, tampering or viruses or any=20
> consequences thereof.
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [Sipping] Conferencing for UAs
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Hello,

> I don't see what the problem is here and why it is a problem? Why do
> they need to be separate. Both the INVITEs and SUBSCRIBEs are
> destined to the same resource.

You send an INVITE to a conference server that will INVITE the rest of 
the participants in the conference. This INVITE is creating:

1) a dialog
2) an INVITE explosion
3) a conference

I find it cleaner to have three separate URIs instead of a single URI 
where the server figures out what the UAC is trying to do by looking at 
the contents of the request (in this case, the event header).

In addition, the application server handling the presence for the 
conference may not be co-located with the entity handling the SIP 
signalling for the conference.

Furthermore, when you deal with requests that do not establish a dialog 
(e.g., MESSAGE), the server needs to convey the URI of the resource to 
the UAC using a different mechanism (Contact is not allowed). It would 
be good to use the same mechanism for requests that create a dialog and 
for requests that do not create a dialog. This avoids overloading the 
Contact header field.

I believe that having something like this:

Associated-Resource: sip:abc@a.com;purpose=conference-state
Associated-Resource: sip:def@a.com;purpose=explosion-state

makes it easier for the UAC to know what can be done; it is more explicit.

Gonzalo


This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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Date: Tue, 24 Feb 2004 08:14:33 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conferencing for UAs
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At 02:06 PM 2/24/2004 +0200, Gonzalo Camarillo wrote:
>Hello,
>
>some comments on the following draft:
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-03.txt
>
>The focus of a conference returns the conference URI in the contact header 
>field of the 200 OK for the INVITE. The UAC uses this URI to SUBSCRIBE to 
>the conf package (using a new dialog)...
>
>Why are we forcing the focus to use the same URI for receiving the INVITEs 
>from all the users and for receiving the SUBSCRIBEs? I believe the 
>solution would be cleaner if we defined a new header field (e.g., 
>Associated-Conf-State) that carried a globally routable URI. The 200 OK 
>for the initial INVITE (or the INVITE in a dial-out case) would carry this 
>header field. This would keep separate the URI used for joining the 
>conference and the URI used for subscribing to the state of the conference.

Gonzalo,

This is a fundamental design decision made in SIP conferencing nearly two 
years ago, and I believe it is a good one.  A requirement for conferencing 
is that a UA must be able to find out information about a conference (by 
subscribing to the conference package, for example) by only knowing the 
conference URI.

Why is this very useful property a problem?

>This is actually the approach that the draft below follows:
>http://www.ietf.org/internet-drafts/draft-camarillo-sipping-transac-package-00.txt
>
>We define the Associated-Transactions-State to be returned, for example, 
>in the 200 OK of a MESSAGE sent to an exploder. Note that we had to create 
>this new header because MESSAGE does not carry a Contact header... 
>following the same approach in both, the conferencing work and the 
>exploders work would be good, in my opinion.

With all due respect, the exploders work should follow the approaches 
described in the conferencing work.  Note that the exploders work should 
not also contradict the XCON work in related areas, as well...

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com


>Comments?
>
>Thanks,
>
>Gonzalo
>
>
>This communication is confidential and intended solely for the 
>addressee(s). Any unauthorized review, use, disclosure or distribution is 
>prohibited. If you believe this message has been sent to you in error, 
>please notify the sender by replying to this transmission and delete the 
>message without disclosing it. Thank you.
>
>E-mail including attachments is susceptible to data corruption, 
>interruption, unauthorized amendment, tampering and viruses, and we only 
>send and receive e-mails on the basis that we are not liable for any such 
>corruption, interception, amendment, tampering or viruses or any 
>consequences thereof.
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


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



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Date: Tue, 24 Feb 2004 08:49:05 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conferencing for UAs
In-reply-to: <403B5F02.50404@ericsson.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        hisham.khartabil@nokia.com
Cc: sipping@ietf.org, oritl@microsoft.com, Miguel.An.Garcia@nokia.com
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At 04:26 PM 2/24/2004 +0200, Gonzalo Camarillo wrote:
>Hello,
>
>>I don't see what the problem is here and why it is a problem? Why do
>>they need to be separate. Both the INVITEs and SUBSCRIBEs are
>>destined to the same resource.
>
>You send an INVITE to a conference server that will INVITE the rest of the 
>participants in the conference. This INVITE is creating:
>
>1) a dialog
>2) an INVITE explosion
>3) a conference

No - this is the problem.  You are looking at your single narrow 
application.  Conferencing is a very broad application, and we have 
developed a very general framework and set of primitives.  It is a mistake 
to invent a whole new set to fit a single application, or change our 
framework to fit your application.

Sending an INVITE creates a dialog.  Period.  Everything else that happens, 
happens because of properties (we call it policy) of the URI.  Your 
application (conference policy of a URI) may cause a conference (called a 
focus) to be created, and an INVITE explosion (called a dial out 
participant list).


>I find it cleaner to have three separate URIs instead of a single URI 
>where the server figures out what the UAC is trying to do by looking at 
>the contents of the request (in this case, the event header).
>
>In addition, the application server handling the presence for the 
>conference may not be co-located with the entity handling the SIP 
>signalling for the conference.


>Furthermore, when you deal with requests that do not establish a dialog 
>(e.g., MESSAGE), the server needs to convey the URI of the resource to the 
>UAC using a different mechanism (Contact is not allowed). It would be good 
>to use the same mechanism for requests that create a dialog and for 
>requests that do not create a dialog. This avoids overloading the Contact 
>header field.
>
>I believe that having something like this:
>
>Associated-Resource: sip:abc@a.com;purpose=conference-state
>Associated-Resource: sip:def@a.com;purpose=explosion-state

No.  There is policy related to the conference URI.  Not an endless set of 
Associated-Something header fields that try to describe a set of 
application specific information.  And we need general mechanisms to find 
out and modify the policy associated with that URI.  This the work being 
done in XCON.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com


>makes it easier for the UAC to know what can be done; it is more explicit.
>
>Gonzalo
>
>
>This communication is confidential and intended solely for the 
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'eburger@snowshore.com'" <eburger@snowshore.com>,
        "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>
Subject: [Fwd: RE: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt]
Date: Tue, 24 Feb 2004 16:25:57 +0100
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Hi,

I guess this is a clarification question on kpml, as well as on gruu (and functions where gruu is used) in general.

If you have a node that doesn't register, but eg wants to use kpml (a gateway, for example), could it generate the gruu on its own, or is it "mandated" to use the register mechanism? If the register mechanism is not used, are there any issues which could come up?

Regards,

Christer Holmberg
Ericsson Finland





> -------- Original Message --------
> Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
> Date: Mon, 23 Feb 2004 17:34:22 -0500
> From: Eric Burger <eburger@snowshore.com>
> To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, <sipping@ietf.org>
> 
> Yep, yep, yep, we can (and will) use GRUU.
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, February 05, 2004 2:46 AM
> > To: sipping@ietf.org
> > Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
> > 
> > 
> > Eric,
> > 
> > The draft talks about sending the subscribe requests on the 
> > same dialog 
> > as the INVITE. For many reasons that have been discussed ad 
> > nauseam on 
> > the list, dialog sharing is a bad idea. GRUU provides a 
> solution that 
> > does not require dialog sharing. As I recall, there was 
> discussion on 
> > this point in the last ietf, and this seemed to be the 
> > preferred direction.
> > 
> > Is there any reason we cant change this to use gruu's rather 
> > than dialog 
> > sharing?
> > 
> > Also, you are missing the IANA registration for the kpml 
> > event package.
> > 
> > Thanks,
> > Jonathan R.
> > 
> > Internet-Drafts@ietf.org wrote:
> > 
> > > A New Internet-Draft is available from the on-line 
> > Internet-Drafts directories.
> > > This draft is a work item of the Session Initiation 
> > Proposal Investigation Working Group of the IETF.
> > > 
> > > 	Title		: Keypad Stimulus Protocol (KPML)
> > > 	Author(s)	: E. Burger
> > > 	Filename	: draft-ietf-sipping-kpml-02.txt
> > > 	Pages		: 42
> > > 	Date		: 2004-2-4
> > > 	
> > > The Key Press Stimulus Protocol uses the SIP SUBSCRIBE/NOTIFY
> > >    mechanism and Keypad Markup Language (KPML) to provide 
> > instructions
> > >    to SIP User Agents for the reporting of user key presses.
> > > 
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-02.txt
> > > 
> > > To remove yourself from the IETF Announcement list, send a 
> > message to 
> > > ietf-announce-request with the word unsubscribe in the body 
> > of the message.
> > > 
> > > Internet-Drafts are also available by anonymous FTP. Login 
> > with the username
> > > "anonymous" and a password of your e-mail address. After 
> logging in,
> > > type "cd internet-drafts" and then
> > > 	"get draft-ietf-sipping-kpml-02.txt".
> > > 
> > > A list of Internet-Drafts directories can be found in
> > > http://www.ietf.org/shadow.html 
> > > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > 
> > > 
> > > Internet-Drafts can also be obtained by e-mail.
> > > 
> > > Send a message to:
> > > 	mailserv@ietf.org.
> > > In the body type:
> > > 	"FILE /internet-drafts/draft-ietf-sipping-kpml-02.txt".
> > > 	
> > > NOTE:	The mail server at ietf.org can return the document in
> > > 	MIME-encoded form by using the "mpack" utility.  To use this
> > > 	feature, insert the command "ENCODING mime" before the "FILE"
> > > 	command.  To decode the response(s), you will need "munpack" or
> > > 	a MIME-compliant mail reader.  Different MIME-compliant 
> > mail readers
> > > 	exhibit different behavior, especially when dealing with
> > > 	"multipart" MIME messages (i.e. documents which have been split
> > > 	up into multiple messages), so check your local documentation on
> > > 	how to manipulate these messages.
> > > 		
> > > 		
> > > Below is the data which will enable a MIME compliant mail reader
> > > implementation to automatically retrieve the ASCII version of the
> > > Internet-Draft.
> > > 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > Chief Technology Officer                    Parsippany, NJ 
> 07054-2711
> > dynamicsoft
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> -- 
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
> 

This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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From: Curro <curro_dominguez@terra.es>
To: sipping@ietf.org
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Hello all

I would like to make you some questions about SIP and IPv6.Please,
excuse me if this is not the right list. 

I'm interested in SIP and testing behaviour with IPv6/IPv4 interworking.
I know IETF recommend dual-stack. But it describes a SIP/SDP ALG in
draft-ietf-v6ops-3gpp-analysis-08.txt like a transition mechanism. 


I want to have an IPv6-only SIP network and a 'SIP Edge Proxy' and
'Media translator' to communicate IPv6 and IPv4, as described in
mentioned draft.

I would like to know whether someone is working on this. Is there some
implementation of this SIP/SDP ALG? Is SIP/SDP ALG worth or is better
waiting to have all legacy IPv4 hosts with dual-stack?  

Or maybe in my scenario, it could be better other type of IPv6/IPv4
interworking? 

By the way, should the SIP/SDP ALG modify SDP body or not? 

Also, draft-elmalki-sipping-3gpp-translator-00.txt proposes a SIP
extension called ALT. But, if I have understand, it implies modify
end-points, doesn't? 

Please, help me to clarify mi ideas. Thank you very much for your
time and responses.

Curro


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



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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
CC: sipping <sipping@ietf.org>, Cullen Jennings <fluffy@cisco.com>,
        bstucker@nortelnetworks.com
Subject: Re: [Sipping] UA indentification and HIP
References: <403B0C82.20407@ericsson.com>
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Does this really work for us?

It sounds like it only works for *hosts*. That doesn't sound like it 
covers the case of a UA (e.g. softphone) that may move from host to host 
while retaining its identity. Nor does it seem to cover the case of 
multiple UAs that share a host.

	Paul

Gonzalo Camarillo wrote:
> Folks,
> 
> the following two drafts describe how to indentify a particular UA using 
> globally unique identifiers:
> 
> http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt
> http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt 
> 
> 
> Cullen pointed out that this work may be able to take advantage of the 
> work carried out in the HIP (Host Identity Protocol) WG, which 
> supplemental web page is:
> 
> http://hip.piuha.net/
> 
> Here you have a *very* brief introduction to HIP. I will also be 
> providing a brief introduction to the drafts above to the HIP community. 
> Hopefully, we will have a useful exchange of ideas between both WGs.
> 
> HIP consists of a layer between the network and the tranport layers 
> (e.g., between IP and TCP). It uses HIs (Host Identifiers) to identify 
> hosts. The idea is that, when a host changes its IP address, it keeps 
> its HI. This facilitates mobility, for instance.
> 
> HIs (Host Identifiers) are public keys, which may have different lengths 
> and formats. To resolve this, HIP defines the HITs (Host Identity Tags), 
> which are hashes of the HI. That is, a HIT is a hash of a public key. 
> They are 128 bit long, and can be used instead of IPv6 addresses in 
> socket calls. HIP also defines LSIs (Local Scope Indetifiers), that are 
> 32 bit long and can be used instead of IPv4 addresses in socket calls.
> 
> Regards,
> 
> Gonzalo
> 
> 
> This communication is confidential and intended solely for the 
> addressee(s). Any unauthorized review, use, disclosure or distribution 
> is prohibited. If you believe this message has been sent to you in 
> error, please notify the sender by replying to this transmission and 
> delete the message without disclosing it. Thank you.
> 
> E-mail including attachments is susceptible to data corruption, 
> interruption, unauthorized amendment, tampering and viruses, and we only 
> send and receive e-mails on the basis that we are not liable for any 
> such corruption, interception, amendment, tampering or viruses or any 
> consequences thereof.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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



From exim@www1.ietf.org  Tue Feb 24 11:44:31 2004
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From: "Brian Stucker" <bstucker@nortelnetworks.com>
To: Paul Kyzivat <pkyzivat@cisco.com>,
        Gonzalo Camarillo
	 <Gonzalo.Camarillo@ericsson.com>
Cc: sipping <sipping@ietf.org>, Cullen Jennings <fluffy@cisco.com>
Subject: RE: [Sipping] UA indentification and HIP
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I'm inclined to say that it may work for us in some cases (where HIP-like
behavior is desired), but I agree with you Paul that it does not seem to
cover the case of multiple UAs that share a host at the very least. It also
seems a bit heavy for environments that don't have easy access to the layer
2/3
information regarding HIP.

Regards,

Brian

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, February 24, 2004 10:11 AM
> To: Gonzalo Camarillo
> Cc: sipping; Cullen Jennings; Stucker, Brian [NGC:B617:EXCH]
> Subject: Re: [Sipping] UA indentification and HIP
> 
> 
> Does this really work for us?
> 
> It sounds like it only works for *hosts*. That doesn't sound like it 
> covers the case of a UA (e.g. softphone) that may move from 
> host to host 
> while retaining its identity. Nor does it seem to cover the case of 
> multiple UAs that share a host.
> 
> 	Paul
> 
> Gonzalo Camarillo wrote:
> > Folks,
> > 
> > the following two drafts describe how to indentify a 
> particular UA using 
> > globally unique identifiers:
> > 
> > http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt
> > 
> http://www.ietf.org/internet-drafts/draft-jennings-sipping-ins
> tance-id-00.txt 
> > 
> > 
> > Cullen pointed out that this work may be able to take 
> advantage of the 
> > work carried out in the HIP (Host Identity Protocol) WG, which 
> > supplemental web page is:
> > 
> > http://hip.piuha.net/
> > 
> > Here you have a *very* brief introduction to HIP. I will also be 
> > providing a brief introduction to the drafts above to the 
> HIP community. 
> > Hopefully, we will have a useful exchange of ideas between both WGs.
> > 
> > HIP consists of a layer between the network and the tranport layers 
> > (e.g., between IP and TCP). It uses HIs (Host Identifiers) 
> to identify 
> > hosts. The idea is that, when a host changes its IP 
> address, it keeps 
> > its HI. This facilitates mobility, for instance.
> > 
> > HIs (Host Identifiers) are public keys, which may have 
> different lengths 
> > and formats. To resolve this, HIP defines the HITs (Host 
> Identity Tags), 
> > which are hashes of the HI. That is, a HIT is a hash of a 
> public key. 
> > They are 128 bit long, and can be used instead of IPv6 addresses in 
> > socket calls. HIP also defines LSIs (Local Scope 
> Indetifiers), that are 
> > 32 bit long and can be used instead of IPv4 addresses in 
> socket calls.
> > 
> > Regards,
> > 
> > Gonzalo
> > 
> > 
> > This communication is confidential and intended solely for the 
> > addressee(s). Any unauthorized review, use, disclosure or 
> distribution 
> > is prohibited. If you believe this message has been sent to you in 
> > error, please notify the sender by replying to this 
> transmission and 
> > delete the message without disclosing it. Thank you.
> > 
> > E-mail including attachments is susceptible to data corruption, 
> > interruption, unauthorized amendment, tampering and 
> viruses, and we only 
> > send and receive e-mails on the basis that we are not 
> liable for any 
> > such corruption, interception, amendment, tampering or 
> viruses or any 
> > consequences thereof.
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> 
> 

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<TITLE>RE: [Sipping] UA indentification and HIP</TITLE>
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<P><FONT SIZE=3D2>I'm inclined to say that it may work for us in some =
cases (where HIP-like</FONT>
<BR><FONT SIZE=3D2>behavior is desired), but I agree with you Paul that =
it does not seem to</FONT>
<BR><FONT SIZE=3D2>cover the case of multiple UAs that share a host at =
the very least. It also</FONT>
<BR><FONT SIZE=3D2>seems a bit heavy for environments that don't have =
easy access to the layer 2/3</FONT>
<BR><FONT SIZE=3D2>information regarding HIP.</FONT>
</P>

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

<P><FONT SIZE=3D2>Brian</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Paul Kyzivat [<A =
HREF=3D"mailto:pkyzivat@cisco.com">mailto:pkyzivat@cisco.com</A>]</FONT>=

<BR><FONT SIZE=3D2>&gt; Sent: Tuesday, February 24, 2004 10:11 =
AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Gonzalo Camarillo</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sipping; Cullen Jennings; Stucker, Brian =
[NGC:B617:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sipping] UA indentification and =
HIP</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Does this really work for us?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; It sounds like it only works for *hosts*. That =
doesn't sound like it </FONT>
<BR><FONT SIZE=3D2>&gt; covers the case of a UA (e.g. softphone) that =
may move from </FONT>
<BR><FONT SIZE=3D2>&gt; host to host </FONT>
<BR><FONT SIZE=3D2>&gt; while retaining its identity. Nor does it seem =
to cover the case of </FONT>
<BR><FONT SIZE=3D2>&gt; multiple UAs that share a host.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Gonzalo Camarillo wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; the following two drafts describe how to =
indentify a </FONT>
<BR><FONT SIZE=3D2>&gt; particular UA using </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; globally unique identifiers:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.tx=
t" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-stucker-sip-=
guid-00.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-jennings-sipping-ins" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-jennings-sip=
ping-ins</A></FONT>
<BR><FONT SIZE=3D2>&gt; tance-id-00.txt </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cullen pointed out that this work may be =
able to take </FONT>
<BR><FONT SIZE=3D2>&gt; advantage of the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; work carried out in the HIP (Host Identity =
Protocol) WG, which </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; supplemental web page is:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://hip.piuha.net/" =
TARGET=3D"_blank">http://hip.piuha.net/</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Here you have a *very* brief introduction =
to HIP. I will also be </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; providing a brief introduction to the =
drafts above to the </FONT>
<BR><FONT SIZE=3D2>&gt; HIP community. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Hopefully, we will have a useful exchange =
of ideas between both WGs.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; HIP consists of a layer between the =
network and the tranport layers </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (e.g., between IP and TCP). It uses HIs =
(Host Identifiers) </FONT>
<BR><FONT SIZE=3D2>&gt; to identify </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; hosts. The idea is that, when a host =
changes its IP </FONT>
<BR><FONT SIZE=3D2>&gt; address, it keeps </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; its HI. This facilitates mobility, for =
instance.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; HIs (Host Identifiers) are public keys, =
which may have </FONT>
<BR><FONT SIZE=3D2>&gt; different lengths </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; and formats. To resolve this, HIP defines =
the HITs (Host </FONT>
<BR><FONT SIZE=3D2>&gt; Identity Tags), </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; which are hashes of the HI. That is, a HIT =
is a hash of a </FONT>
<BR><FONT SIZE=3D2>&gt; public key. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; They are 128 bit long, and can be used =
instead of IPv6 addresses in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; socket calls. HIP also defines LSIs (Local =
Scope </FONT>
<BR><FONT SIZE=3D2>&gt; Indetifiers), that are </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 32 bit long and can be used instead of =
IPv4 addresses in </FONT>
<BR><FONT SIZE=3D2>&gt; socket calls.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Gonzalo</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This communication is confidential and =
intended solely for the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; addressee(s). Any unauthorized review, =
use, disclosure or </FONT>
<BR><FONT SIZE=3D2>&gt; distribution </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; is prohibited. If you believe this message =
has been sent to you in </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; error, please notify the sender by =
replying to this </FONT>
<BR><FONT SIZE=3D2>&gt; transmission and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; delete the message without disclosing it. =
Thank you.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; E-mail including attachments is =
susceptible to data corruption, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; interruption, unauthorized amendment, =
tampering and </FONT>
<BR><FONT SIZE=3D2>&gt; viruses, and we only </FONT>
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<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
application of SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip@ietf.org for new developments of =
core SIP</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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------_=_NextPart_001_01C3FAF4.9C0B69C6--

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
CC: sipping <sipping@ietf.org>, Cullen Jennings <fluffy@cisco.com>,
        bstucker@nortelnetworks.com
Subject: Re: [Sipping] UA indentification and HIP
References: <403B0C82.20407@ericsson.com> <403B7790.4020507@cisco.com>
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Paul,

 > That doesn't sound like it
> covers the case of a UA (e.g. softphone) that may move from host to host 
> while retaining its identity.

The drafts mention things like MAC addresses.... my impression is that 
the indentity would be bundled to the hardware the phone is running on, 
but it would be good to clarify the purpose of the identity in this 
situation.

> Nor does it seem to cover the case of 
> multiple UAs that share a host.

I believe HIP covers this case, because different UAs would use 
different public keys, and thus, different HITs.

Gonzalo


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Harry Behrens wrote:
> Dear Paul,
>=20
> inline
>=20
>=20
>>-----Urspr=FCngliche Nachricht-----
>>Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
>>Gesendet: Montag, 23. Februar 2004 19:47
>>An: Harry Behrens
>>Cc: sipping@ietf.org; knuettel@fokus.fraunhofer.de
>>Betreff: Re: [Sipping] Roaming Architecture for SIP
>>
>>
>>I have specific comment inline below, and then a question.
>>
>>	Paul
>>
>>
>>>  -  A user registers with the VN (atlanta.com in the call flow=20
>>>      given). The user submits his NAI in new header field 'P-NAI'.=20
>>>      The user submits a bogus URI in both the From and the=20
>>
>>To header.=20
>>
>>>      (F1 in section 8.2.1)
>>>  -  The VN recognizes the registration as a P-NAI registration and=20
>>>      sends back a "401 Unauthorized" response, with both=20
>>
>>the 'From'=20
>>
>>>      and the 'To' URI set to the temporary URI that will=20
>>
>>be assigned=20
>>
>>>      to the user upon successful authentication and authorization=20
>>>(F2).
>>
>>This isn't legal. The From and To (with exception of To-tag)=20
>>of response=20
>>must be same as in the request.
>=20
> This is true. The reason we have this crutch in there is to allow the V=
N
> server to pass a user ID to the client, which the client can use to
> register to the VN. This would otherwise have to happen out-of-band,
> which we were trying to avoid.
> We are thinking about alternative approaches to this requirement and ar=
e
> grateful for any input.

No good idea comes immediately to mind. I'm still not convinced of why I=20
would want this at all.

>>The question is: why would the UAC want to register with the VN?=20
>>Assuming it is able to send sip messages, (which apparently=20
>>it can), why=20
>>wouldn't it simply send a REGISTER to its HN?
>>
>>I presume there must be some service that you intend the VN won't=20
>>perform, that will prevent the UAC from doing this. But it isn't=20
>>apparent what service that is.
>=20
> The scenario is of a user based e.g. in Berlin/Germany on a business
> trip in Beijing/China.
> The user will want to both be reachable under his home AOR (for that he
> registers with his HN)

I'm ok with that.

 > as well as make use of the VN for local calls,
> especially local PSTN breakout (this is why he wants to have a valid
> registration with the VN).

Why? His billing relationship is with his HN. Why not let the HN figure=20
out the best way to route this calls? If it is via the VN's gateways,=20
then it can route the calls to the VN.

> Furthermore, in a number of cases, one encounters RTP-proxying, where
> around-the-world loops are to be avoided.

Certainly that is likely to be a bad thing. But does it require a=20
signaling identity with the VN to achieve that?

> Another issue which we foresee is that in the near future hotels,
> airports or other semi-closed (walled garden) Internet access points,
> will force users through their own VoIP (SIP) proxy infrastructure,
> similar to what a lot of sites aleady do in the case of HTTP.

Sure. But as long as those things are *proxies*, the UA should still be=20
able to register with his HN.

Dealing with that depends a bit on how users will be forced. The way=20
hotels do it today depends on details of http that won't necessarily=20
work for sip.

	Paul


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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
CC: sipping <sipping@ietf.org>, Alan Johnston <alan.johnston@mci.com>,
        Orit Levin <oritl@microsoft.com>,
        Miguel Angel Garcia Martin <Miguel.An.Garcia@nokia.com>
Subject: Re: [Sipping] Conferencing for UAs
References: <403B3E35.9010304@ericsson.com>
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Gonzalo Camarillo wrote:
> Hello,
> 
> some comments on the following draft:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-03.txt 
> 
> 
> The focus of a conference returns the conference URI in the contact 
> header field of the 200 OK for the INVITE. The UAC uses this URI to 
> SUBSCRIBE to the conf package (using a new dialog)...
> 
> Why are we forcing the focus to use the same URI for receiving the 
> INVITEs from all the users and for receiving the SUBSCRIBEs?

Why? Because that URI identifies the entity that I want information 
about. It is the same reason that you might send a subscription for the 
dialog package, or the presence package to the same address you send an 
INVITE or MESSAGE to.

One of the advantages of sip is that it simplifies addressing.

 > I believe
> the solution would be cleaner if we defined a new header field (e.g., 
> Associated-Conf-State) that carried a globally routable URI. The 200 OK 
> for the initial INVITE (or the INVITE in a dial-out case) would carry 
> this header field. This would keep separate the URI used for joining the 
> conference and the URI used for subscribing to the state of the conference.

Why complicate life for users by introducing more addresses to deal with?

If the conference server wants to decompose the service, it can always 
return an AOR as its contact address, with a proxy routing subscribe 
requests one way, invite requests a different way, etc.

	Paul


_______________________________________________
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Subject: Re: [Sipping] Conferencing for UAs
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Gonzalo Camarillo wrote:
> Hello,
> 
>> I don't see what the problem is here and why it is a problem? Why do
>> they need to be separate. Both the INVITEs and SUBSCRIBEs are
>> destined to the same resource.

I agree.

> You send an INVITE to a conference server that will INVITE the rest of 
> the participants in the conference. This INVITE is creating:
> 
> 1) a dialog
> 2) an INVITE explosion
> 3) a conference

You just jumped to the most complex of cases that combines a bunch of 
things. Not every invite to a focus generates an invite explosion.

As I see it, I send an INVITE to the address of some entity that I want 
to talk to. Sometimes it turns out to be a conference focus, sometimes 
not. Either way I am communicating with that entity.

If I want to know more about that entity than I get by virtue of my 
invitation, I send some other request to the address of that entity, 
defined by the remote contact address of my dialog. I could send OPTIONS 
for some info. Or I can send SUBSCRIBE if I want to request event updates.

Taken to extreme, what you suggest would lead to a different address for 
OPTIONS, for MESSAGE, for PUBLISH, for REGISTER, etc. And maybe 
different addresses for each kind of subscription.

I don't want to go there.

	Paul


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From: "Scoggins, Sophia" <sscoggins@telogy.com>
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My group islooking for SIP proxy servers with either TLS, or IPSec, or any
other standard based security protocol implementation to perform interop
testing. If any one knows any SIP proxy server with the security feature,
please let me know. Thanks in advance.

 

 

Regards,

Sophia2@ti.com


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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>My group islooking for SIP proxy servers with either TLS, or
IPSec, or any other standard based security protocol implementation to perform interop
testing. If any one knows any SIP proxy server with the security feature,
please let me know. Thanks in advance.</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Regards,</span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Sophia2@ti.com</span></font></p>

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From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] ToIP Requirements
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At 11:21 AM 2/24/2004 +0200, Gonzalo Camarillo wrote:


>While SIPPING is the right place to handle requirements that will lead to 
>new SIP protocol extensions, SIPPING may not be the right place to deal 
>with device requirements (e.g., text phones MUST have a display of at 
>least 100 characters).

In the US, Orgs like the TIA do this device requirement type of work 
(specifically the TR41.4 group within TIA)


>Thanks,
>
>Gonzalo




cheers,
James

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


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Subject: [Sipping] SIPPING Agenda, IETF 59
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Hyperlinked version at:
http://www.softarmor.com/sipping/meets/ietf59/agenda.html

Please get any slides in to Gonzalo well in advance

Note that there are GRUU-related entries on both SIPPING and SIP 
agendae. The SIPPING session is for requirements, and the SIP session 
for mechanism.


MONDAY, March 1, 2004, 0900-1130, Sapphire 4
0900 Angenda Bash
             Chairs
0905 Status of Work
             Chairs
0920 Event Package for Configuration Profiles
           Dan Petrie
           draft-ietf-sipping-config-framework-02.txt
0935 Session Policies
           Volker Hilt
           draft-hilt-sipping-session-indep-policy-00.txt
           draft-hilt-sipping-session-spec-policy-00.txt
           Additional Reading
           draft-ietf-sipping-session-policy-req-01.txt
0945 The dialog package
           Rohan Mahy
           draft-ietf-sipping-dialog-package-04 [txt] [html]
0955 End-To-Middle Security
           Kumiko Ono
           draft-ietf-sipping-e2m-sec-reqs-01.txt
           draft-ono-sipping-end2middle-security-01.txt
           draft-ono-sipping-smime-keyreuse-00.txt
1010 Real-Time Text Conversation Using SIP
           Arnoud van Wijk
           draft-manyfolks-sipping-toip-01.txt
1025 QSIG SIP Call Transfer
           Jean-Francois Rey
           draft-rey-sipping-qsig2sip-transfer-00.txt
1035 S/MIME Certificates in SIP
           Cullen Jennings
           draft-jennings-sipping-certs-02.txt [html]
1045 KPML
           Eric Burger
           draft-ietf-sipping-kpml-02.txt
1055  Basic Network Media Services with SIP
           Eric Burger
           draft-burger-sipping-netann-08.txt
  1105 Instance ID, GRID, GRUU, Resource ID Requirements
          Brian Stucker, Cullen Jennings
          draft-stucker-sip-guid-00.txt
          draft-jennings-sipping-instance-id-00.txt [html]
          draft-ietf-xmpp-core-22.txt section 2.3
WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
1300 Agenda Bash
           Chairs

Design Teams
1305 Application Interaction
           Jonathan Rosenberg
           draft-ietf-sipping-app-interaction-framework-01.txt
1315 SIP Conferencing
           Alan Johnston
           draft-ietf-sipping-cc-conferencing-03.txt
           draft-ietf-sipping-conference-package-03.txt
           draft-ietf-sipping-conferencing-requirements-00.txt
           draft-ietf-sipping-conferencing-framework-01.txt
1325 Transcoding
           Gonzalo Camarillo
           draft-camarillo-sipping-transc-b2bua-01.txt [PDF]
1335 SIP Emergency
           Henning Schulzrinne
           draft-ietf-sipping-sos-00.txt
           draft-schulzrinne-sipping-emergency-arch-00.txt
1345 Location Conveyance Requirements
           James Polk
           draft-ietf-sipping-location-requirements-00.txt
1355 REFER Extensions
           Orit Levin, Rohan Mahy, and Gonzalo Camarillo
           draft-olson-sipping-refer-extensions-01.txt
           draft-mahy-sip-remote-cc-01.txt
           draft-camarillo-sipping-multiple-refer-00.txt
1425 Exploders
           Gonzalo Camarillo
           draft-camarillo-sipping-exploders-02.txt
           draft-camarillo-sipping-uri-list-01.txt
           draft-camarillo-sipping-adhoc-management-00.txt
           draft-camarillo-sipping-transac-package-00.txt
           draft-hiller-uri-list-index-00.txt
           draft-levin-simple-adhoc-list-01.txt
1445 RTCP Summary Report Event Package
           Alan Johnston
           draft-johnston-sipping-rtcp-summary-02.txt

Additional requests will be accomodated if time allows


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



From exim@www1.ietf.org  Tue Feb 24 14:13:42 2004
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Steve Langstaff wrote:
> Hi All.
>  
> Sorry if this is a much asked and much answered question, but haven't
> the basic problems being solved in a "Roaming Architecture for SIP",
> i.e. allowing a SIP host to roam to foreign networks, already been
> solved using Mobile IP? What else does "Roaming Architecture for SIP"
> give me?
>  

Multiple concurrent appearance with capability and preference matched 
request routing.

--
Dean

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To: Dean Willis <dean.willis@softarmor.com>, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] SIPPING Agenda, IETF 59
Cc: Rohan Mahy <rohan@cisco.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:

>Hyperlinked version at:
>http://www.softarmor.com/sipping/meets/ietf59/agenda.html
>
>WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
>
>1335 SIP Emergency
>           Henning Schulzrinne
>           draft-ietf-sipping-sos-00.txt
>           draft-schulzrinne-sipping-emergency-arch-00.txt
>1345 Location Conveyance Requirements
>           James Polk
>           draft-ietf-sipping-location-requirements-00.txt


I believe these presentations should be reversed in order because the main 
points of discussion in the Location Requirements ID are for Emergency 
location dealing with call routing by proxies, adding LI (location 
information) by the proxy (whether or not the UA inserted it already) and 
the security of the LI in transit. Discussing these points after the 
Emer-Arch doc doesn't seem to make much sense due to potential rehash while 
the appropriate presenter is not at the podium (in either direction).

comments?



cheers,
James

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


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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James M. Polk wrote:
> At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
> 
>> Hyperlinked version at:
>> http://www.softarmor.com/sipping/meets/ietf59/agenda.html
>>
>> WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
>>
>> 1335 SIP Emergency
>>           Henning Schulzrinne
>>           draft-ietf-sipping-sos-00.txt
>>           draft-schulzrinne-sipping-emergency-arch-00.txt
>> 1345 Location Conveyance Requirements
>>           James Polk
>>           draft-ietf-sipping-location-requirements-00.txt
> 
> 
> 
> I believe these presentations should be reversed in order because the 
> main points of discussion in the Location Requirements ID are for 
> Emergency location dealing with call routing by proxies, adding LI 
> (location information) by the proxy (whether or not the UA inserted it 
> already) and the security of the LI in transit. Discussing these points 
> after the Emer-Arch doc doesn't seem to make much sense due to potential 
> rehash while the appropriate presenter is not at the podium (in either 
> direction).
> 
> comments?

Works for me. Henning?

--
Dean

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



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Date: Tue, 24 Feb 2004 15:03:36 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Re:draft-ietf-sipping-cc-conferencing-03.txt
In-reply-to: <403A5B84.3040103@cisco.com>
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At 02:59 PM 2/23/2004 -0500, Paul Kyzivat wrote:
>In section 4.1 a change was made around the descriptin of handling a 
>subscription. The new wording seems wrong:
>
>    The SUBSCRIBE should be sent outside the INVITE-initiated dialog.
>
>    Participants leaving the conference send a BYE to the focus.  If they
>    have a current subscription to the conference package, they also must
>    send a SUBSCRIBE with an Expires:0 header field to terminate both
>    dialogs.
>
>There is nothing for "The SUBSCRIBE" to refer back to. I wonder if this 
>sentence wasn't intended to be in section 3.4, where it would make more sense.

Hi Paul,

Thanks for the comment.  The SUBSCRIBE mentioned is message F5 in Figure 1, 
but I agree that the wording is bad.  You are right - the statement should 
be in 3.4.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com


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


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



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        Rohan Mahy <rohan@cisco.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        hgxs@cs.columbia.edu
Subject: Re: [Sipping] SIPPING Agenda, IETF 59
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Reversal is fine with me.

Dean Willis wrote:

> James M. Polk wrote:
> 
>> At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
>>
>>> Hyperlinked version at:
>>> http://www.softarmor.com/sipping/meets/ietf59/agenda.html
>>>
>>> WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
>>>
>>> 1335 SIP Emergency
>>>           Henning Schulzrinne
>>>           draft-ietf-sipping-sos-00.txt
>>>           draft-schulzrinne-sipping-emergency-arch-00.txt
>>> 1345 Location Conveyance Requirements
>>>           James Polk
>>>           draft-ietf-sipping-location-requirements-00.txt
>>
>>
>>
>>
>> I believe these presentations should be reversed in order because the 
>> main points of discussion in the Location Requirements ID are for 
>> Emergency location dealing with call routing by proxies, adding LI 
>> (location information) by the proxy (whether or not the UA inserted it 
>> already) and the security of the LI in transit. Discussing these 
>> points after the Emer-Arch doc doesn't seem to make much sense due to 
>> potential rehash while the appropriate presenter is not at the podium 
>> (in either direction).
>>
>> comments?
> 
> 
> Works for me. Henning?
> 
> -- 
> Dean
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

_______________________________________________
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CC: Henning Schulzrinne <hgs@cs.columbia.edu>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping <sipping@ietf.org>,
        Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com> <403A335F.8090403@bell-labs.com>
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Volker Hilt wrote:

> Policies and configuration information seem to share a number of
> commonalities. Thus, I think it would make sense to use the same
> framework for both. The framework should probably be able to transfer
> any kind of information from a network entity to a user agent.
>
> More comments inline.
>
> Daniel G. Petrie wrote:
> > Henning Schulzrinne wrote:
> >>My general concern is that the current config framework is far too
> >>device-centric, presumably based on the origin and interests of the
> >
> > This is certainly not the intension.  The draft specifies a way to separately
> > subscribe for device or user profile data.  I would be happy to clarify this
> > where you think the draft is vague or implies this restriction.
> >
> Is it safe to assume that policy information is always part of the
> device or user configuration or can it be an independent piece of
> information a UA wants to subscribe to? Thus, would it make sense to
> define a profile-name "session-policies"? This could be useful if a
> device is configured by other means and simply wants to keep track of
> the policies in a network.

I can think of applications where policy information is useful by itself (e.g. in a
profile server UI which facilitates profile data input) as well as applications were
the profile data is needed in conjuntion with the policy data.  I would suggest
using something like XML name spaces so that everyting may be contained
in the same or separate XML documents.

>
>
> >>
> >>The three sources are:
> >>(1) The local network that I happen to be in, represented by the
> >>outbound proxy or other information conveyed by DHCP, identified by my AOR;
> >>
> >>(2) My home provider/network, where I am identified by my AOR;
> >>
> >>(3) My device, identified by some permanent device identifier, typically
> >>a MAC-address like thing, but some of the other identifiers discussed
> >>here recently may also be possible.
> >
> > These 3 important concepts are intended to be supported in
> > draft-ietf-sipping-config-framework-02.  It is however difficult in
> > practice to keep these three things separate.  In many environments
> > it is easier to keep two of the three separate.
> >
> > For example:
> > In an enterprise with physical devices that do not move much, it becomes
> > easier to combine the properties of 1 related to location with the properties
> > of 3.
> >
> > When a user does not know their AOR or does not want to have to provide
> > it to the UA (to provide a plug and play experience), the user's AOR can be
> > associated with 3 (the device).  2 is may then be subscribe implicitly or explicitly.
> >
> > In the automated discover phase for a new UA, the distinction between
> > 1 and 2 become blurred as the device can only discover or derive
> > AORs based upon the local domain (without user input).
> >
> A difference between configuration data and other information (such as
> policies) seems to be that subscriptions to configuration data must work
> on an "out-of-the-box" device that has not been configured yet. Once the
> configuration is done, its probably easier to subscribe to the different
> sources of information.
>
> Is there information that applies to a device besides the device
> configuration? If not, would it make sense to differentiate into the
> device configuration (requiring a discovery process) and subscriptions

It is an interesting thought that discover may only be needed for device
profiles.

>
> to the local network using my AOR as well as subscriptions to my home
> network (also using my AOR)?
>

I am not sure I follow how things get routed differently to the local or
home network.  Are there two different AORs?

>
> Thanks,
>
> Volker
>
> > I think that this illustrates a very good point that the separation between
> > user and device profile data cannot be made concrete.  Data such as
> > out bound proxy, or policy may be dictated by any or all of the 3
> > realms defined above.  Only the application and local administrative
> > policies can determine the realm to which data should be associated.
> > In addition it may be desirable to override some of the user or device
> > profile data based upon other local discovery data (e.g. DHCP options
> > for out bound proxy, STUN for NAT traversal).
> >
> > At best we can define a precedence by which the device data
> > can be overridden by user data (or the other way around)
> > along with the ability for administrators to dictate policy (perhaps indicated
> > in the profile data) over what can be overridden.
> >
> >
> >>So far, only a subset of (3) seems to have been discussed.
> >>
> >>I had raised this issue earlier, but have gotten no response. It seems
> >>fundamental to the whole architecture discussion. The same problem that
> >>Gonzalo raises (one or more events) also applies to the question as to
> >>how this relates to watcher lists and other IM configuration information.
> >>
> >>Gonzalo Camarillo wrote:
> >>
> >>
> >>>Hello,
> >>>
> >>>The following draft defines a framework for session policies in SIP
> >>>http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-indep-policy-00.txt
> >>>
> >>>
> >>>It defines an event package and a data format to carry session policy
> >>>information. Now, the following draft defines a framework for sending
> >>>configuration information and user profiles to SIP user agents.
> >>>
> >>>http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-02.txt
> >>>
> >>>
> >>>We need to decide whether session policies should use the package
> >>>defined by the configuration framework or, on the other hand, continue
> >>>defining its own package. The XML based data format would remain the
> >>>same on both cases.
> >>>
> >>>I would like to know what type of information folks intend to store as
> >>>user profile. Would it make sense to consider session policies as a
> >>>subset of the user profile, or it would be better to define a new
> >>>profile-name value: "session-policy" (in addition to the already defined
> >>>"user" and "device").
> >>>
> >>>I tend to think that it would be better to re-use the same package for
> >>>session policies as for configuration, defining the new "session-policy"
> >>>value for the profile-name Event header field parameter, but I would
> >>>like to hear other opinions.
> >>>
> >>>Comments?
> >>>
> >>>A further question of the config draft. Is the intention to establish
> >>>two subscriptions, one for device configuration and another for user
> >>>profiles, or the idea is to use a single subscription and multiplex
> >>>somehow both data types into it.
> >>>
> >>>Thanks,
> >>>
> >>>Gonzalo
> >>>
> >>>
> >>>
> >>>This communication is confidential and intended solely for the
> >>>addressee(s). Any unauthorized review, use, disclosure or distribution
> >>>is prohibited. If you believe this message has been sent to you in
> >>>error, please notify the sender by replying to this transmission and
> >>>delete the message without disclosing it. Thank you.
> >>>
> >>>E-mail including attachments is susceptible to data corruption,
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> >>>send and receive e-mails on the basis that we are not liable for any
> >>>such corruption, interception, amendment, tampering or viruses or any
> >>>consequences thereof.
> >>>
> >>>
> >>>_______________________________________________
> >>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>>This list is for NEW development of the application of SIP
> >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>Use sip@ietf.org for new developments of core SIP
> >
> >
>
> --
> Volker Hilt                      Bell Labs / Lucent Technologies
> phone: +1 732 332 6432           101 Crawfords Corner Rd
> email: volkerh@bell-labs.com     Holmdel, NJ 07733


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Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com> <403A889F.4010601@cs.columbia.edu>
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I split this into two responses.  This is good input.  I think making
two distict discover approaches makes sense.  I will try to incorporate
this into the next draft.

Henning Schulzrinne wrote:

> > This is certainly not the intension.  The draft specifies a way to separately
> > subscribe for device or user profile data.  I would be happy to clarify this
> > where you think the draft is vague or implies this restriction.
>
> >
> > Section 3.2 describes parameters that are device specific as even in the
> > ideal world, purely user specific data may need to consider the UA
> > implementation that it is to be applied to.  Section 3.2 also specifies a
> > means to indicate the desire for user or device data.
> >
> > Section 3.13 specifies how to form the SUBSCRIBE URI based upon
> > the interest for either a device or user profile.
>
> 3.13 and 4.1 don't really spell out that user (as opposed to device)
> data may need to go elsewhere. It only has a single set of steps that
> aborts as soon as (say) DHCP-based discovery succeeds.
>
> I think the two modes, user and device, are somewhat dissimilar. For
> device, I may need to combine local information (e.g., information about
> outbound proxies and local dialplans) with home information (local
> dialplan, device-specific UI configuration, etc.).
>
> For user configuration, I probably need to assume that the device has an
> AOR, either initially or discovered via device configuration, which is
> realistic for laptops and other personal devices.
>
> 4.1 does not seem to spell out what to do if either the device or the
> user knows the AOR, but wants to get the remainder of the device and
> user configuration automatically.
>
> Thus, I'd suggest separating the two discovery models and possibly
> logically separating user configuration as being derived (among other
> sources of AORs) from the device configuration, as in:
>
> DHCP -> device configuration query -> AOR -> user-based query


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Date: Wed, 25 Feb 2004 00:37:17 -0500
From: "Daniel G. Petrie" <dpetrie@pingtel.com>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping <sipping@ietf.org>
Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com> <403A889F.4010601@cs.columbia.edu>
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Henning Schulzrinne wrote:

snipped

> > These 3 important concepts are intended to be supported in
> > draft-ietf-sipping-config-framework-02.  It is however difficult in
> > practice to keep these three things separate.  In many environments
> > it is easier to keep two of the three separate.
> >
> > For example:
> > In an enterprise with physical devices that do not move much, it becomes
> > easier to combine the properties of 1 related to location with the properties
> > of 3.
>
> I'd rather be more general; if the user configuration spans multiple
> devices, it's a lot cleaner if I only have to update one server rather
> than each device-based server. (Device-based servers may well be more
> local than user-based servers.)

I am not sure I follow this.  Perhaps if I elaborate, you can help me realize
my confusion.  The three types of profile data are: local network, user and
device data.  In the above example I was suggesting that in many environments
the UA is not mobile or at least not frequently moved.  Hense the UA can
be associated with a specific location.  In these environments it is convenient
to combine the local network data and the device specific data.  I was not
suggesting that the user data should be denormalized.  A user may have
many devices.  The devices can all use the same user profile data.

I am also questioning whether people think we really need three distinct
types of profiles (local network, user and device).  Is there a real life
environment where we need all three or can it be accomlished with
two by combining parts of the third with the other two.  If people think
we need all three then we need to have three different discovery models
and ways to form three different AORs (for each profile type).  This is not
difficult, I just want to be sure the extra moving parts are necessary.

>
>
> >
> > When a user does not know their AOR or does not want to have to provide
> > it to the UA (to provide a plug and play experience), the user's AOR can be
> > associated with 3 (the device).  2 is may then be subscribe implicitly or explicitly.
>
> In my two-stage model, this is supported.
>
> > I think that this illustrates a very good point that the separation between
> > user and device profile data cannot be made concrete.  Data such as
> > out bound proxy, or policy may be dictated by any or all of the 3
> > realms defined above.  Only the application and local administrative
> > policies can determine the realm to which data should be associated.
> > In addition it may be desirable to override some of the user or device
> > profile data based upon other local discovery data (e.g. DHCP options
> > for out bound proxy, STUN for NAT traversal).
>
> There may well be information that could be considered both, but in my
> book, user-based configuration is independent of where I plug in my
> device and does not depend on the device itself. Thus, an outbound proxy
> and related transport parameters are clearly device-related, while speed
> dial functionality, media preferences, home dial plans and buddy lists
> are probably not.
>
> Some parameters may be hierarchical, in that local network configuration
> overrides my general configuration. If my visited network doesn't allow
> video, that takes precedence over my user configuration that prefers
> video calls.
>
> >
> > At best we can define a precedence by which the device data
> > can be overridden by user data (or the other way around)
> > along with the ability for administrators to dictate policy (perhaps indicated
> > in the profile data) over what can be overridden.
>
> I agree. I think we'll have both directions of override. For example, a
> CIA agent user configuration may mandate use of TLS and AES encryption
> and accept that a call will fail in some local environments that don't
> allow this.
>
> Having a sample set of configuration parameters that we can work through
> might test our assumptions.  draft-stredicke-sipping-ep-config-00.txt
> seems to be the last one on this topic, or is there something else?
>

There is nothing newer than this expired draft.  I was hoping to shake
the dust off and restructure this document once the framework was
last called.  I think the document identifies some good profile data
requirements, but I think they should be sliced up a little differently.

>
> It unfortunately conflates local parameters (e.g., outbound proxy,
> 3.1.4.3) with user-based parameters (registration period, 3.1.4.1).
>
> Henning


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To: "Daniel G. Petrie" <dpetrie@pingtel.com>
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        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
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Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com> <403A335F.8090403@bell-labs.com> <403C265D.38884E9A@pingtel.com>
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Daniel G. Petrie wrote:
>>Is it safe to assume that policy information is always part of the
>>device or user configuration or can it be an independent piece of
>>information a UA wants to subscribe to? Thus, would it make sense to
>>define a profile-name "session-policies"? This could be useful if a
>>device is configured by other means and simply wants to keep track of
>>the policies in a network.
> 
> I can think of applications where policy information is useful by itself (e.g. in a
> profile server UI which facilitates profile data input) as well as applications were
> the profile data is needed in conjuntion with the policy data.  I would suggest
> using something like XML name spaces so that everyting may be contained
> in the same or separate XML documents.
> 
Yes, this would allow the UA to get the relevant information. I'm not 
sure if it would enable a UA to subscribe to a certain type of 
information (e.g. policies). Thus, the notifier might have to return all 
URIs it has available for the subscriber in the NOTIFY. Unless the UA 
can determine the exact content of a document (e.g. based on MIME type 
or URI structure) it would have to download all documents to find the 
relevant pieces of information. I'm not sure though if this poses a real 
problem.

>>Is there information that applies to a device besides the device
>>configuration? If not, would it make sense to differentiate into the
>>device configuration (requiring a discovery process) and subscriptions
> 
> It is an interesting thought that discover may only be needed for device
> profiles.
> 
>>to the local network using my AOR as well as subscriptions to my home
>>network (also using my AOR)?
> 
> I am not sure I follow how things get routed differently to the local or
> home network.  Are there two different AORs?
> 
One possibility could be that the UA subscribes separately to the AOR 
and to a discovered local profile server (both times using the same AOR 
in the From header).

Volker


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Gonzalo:

I see your idea as a future proof solution. Basically, with the current =
framework, we are assuming that subscriptions and other state =
information is kept in the same entity that has established the dialog, =
and that probably is the common case today. I don't see any application =
today that would need to split and send pointers to different entities, =
perhaps in the future, but I haven't seen that one.

Having said that, I still think that from the point of view of the =
protocol design it is a clear solution to separate items, and avoid =
assumptions. Perhaps one of the most frustrating things in protocol =
design is to find out that "a couple of years ago we design feature foo =
based on assupmption bar, which is not valid anymore; now we need to =
patch the protocol". So I would support your proposal.

/Miguel

> -----Original Message-----
> From: ext Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: 24 February, 2004 16:26
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: sipping@ietf.org; alan.johnston@mci.com;=20
> oritl@microsoft.com; Garcia
> Miguel.An (Nokia-NRC/Helsinki)
> Subject: Re: [Sipping] Conferencing for UAs
>=20
>=20
> Hello,
>=20
> > I don't see what the problem is here and why it is a problem? Why do
> > they need to be separate. Both the INVITEs and SUBSCRIBEs are
> > destined to the same resource.
>=20
> You send an INVITE to a conference server that will INVITE=20
> the rest of=20
> the participants in the conference. This INVITE is creating:
>=20
> 1) a dialog
> 2) an INVITE explosion
> 3) a conference
>=20
> I find it cleaner to have three separate URIs instead of a single URI=20
> where the server figures out what the UAC is trying to do by=20
> looking at=20
> the contents of the request (in this case, the event header).
>=20
> In addition, the application server handling the presence for the=20
> conference may not be co-located with the entity handling the SIP=20
> signalling for the conference.
>=20
> Furthermore, when you deal with requests that do not=20
> establish a dialog=20
> (e.g., MESSAGE), the server needs to convey the URI of the=20
> resource to=20
> the UAC using a different mechanism (Contact is not allowed).=20
> It would=20
> be good to use the same mechanism for requests that create a=20
> dialog and=20
> for requests that do not create a dialog. This avoids overloading the=20
> Contact header field.
>=20
> I believe that having something like this:
>=20
> Associated-Resource: sip:abc@a.com;purpose=3Dconference-state
> Associated-Resource: sip:def@a.com;purpose=3Dexplosion-state
>=20
> makes it easier for the UAC to know what can be done; it is=20
> more explicit.
>=20
> Gonzalo
>=20
>=20
> This communication is confidential and intended solely for=20
> the addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is prohibited. If you believe this message has=20
> been sent to you in error, please notify the sender by=20
> replying to this transmission and delete the message without=20
> disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data=20
> corruption, interruption, unauthorized amendment, tampering=20
> and viruses, and we only send and receive e-mails on the=20
> basis that we are not liable for any such corruption,=20
> interception, amendment, tampering or viruses or any=20
> consequences thereof.
>=20
>=20

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Subject: RE: [Sipping] Conferencing for UAs
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> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 24 February, 2004 19:47
> To: Gonzalo Camarillo
> Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org;
> alan.johnston@mci.com; oritl@microsoft.com; Garcia Miguel.An
> (Nokia-NRC/Helsinki)
> Subject: Re: [Sipping] Conferencing for UAs

[snap]

> Taken to extreme, what you suggest would lead to a different=20
> address for=20
> OPTIONS, for MESSAGE, for PUBLISH, for REGISTER, etc. And maybe=20
> different addresses for each kind of subscription.
>=20
> I don't want to go there.
>=20
> 	Paul
>=20

Aren't already there with Caller Preferences?

/Miguel

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To: Alan Johnston <alan.johnston@mci.com>
CC: hisham.khartabil@nokia.com, sipping@ietf.org, oritl@microsoft.com,
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Subject: Re: [Sipping] Conferencing for UAs
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Hi Alan,

I understand that my comments come late in the game and that it might be 
impossible to change the conferencing framework at this point. In any 
event, I do not agree with your statement below:

> No - this is the problem.  You are looking at your single narrow 
> application.

I believe I am actually looking at a much wider range of applications 
than just conferencing. So, to me, the problem is that the conferencing 
framework is too focused on a "single narrow application"; conferencing.

In a conferencing service, there is a single resource created by the 
initial INVITE; a conference. That's why the server can use the Contact 
header field to send it back to the UAC. If you think of services that 
create more than one resource (e.g., an explosion and a conference), the 
server may not want to use the same URI for both. Moreover, 
conceptually, is is not a good thing to use the same URI to identify 
both resources.

So, my main concern is the semantics overload of the Contact header. (We 
abused this header when design REGISTER as well, by the way. That's why 
I do not want to do it again.)

In my opinion, the Contact header is the URI where you send subsequent 
requests within the dialog. If the server wants to tell the client that 
a resource (e.g., a conference) has been created, it can use a different 
header to include the URI of such a resource. Note that, if you want 
both URIs to be the same (like in the conferencing case), you can do it.

Contact: sip:1234@a.com;isfocus
Resource: sip:1234@a.com;isfocus

An entity receiving these headers would use the URI in the Contact 
header to send subsequent requests *within* the dialog, and the URI in 
the other header to send requests *outside* that dialog (e.g., SUBSCRIBE 
to the conference package).

In brief, instead of sticking the conference URI in a Contact header and 
label that URI with an isfocus parameter so that the UAC figures what is 
going on, I find it more explicit to use a different header. Although, 
as I said, I understand that this comment comes late and that may not be 
useful at this point.

Thanks,

Gonzalo


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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: sipping <sipping@ietf.org>
CC: jmpolk@cisco.com, Brian.rosen@marconi.com, ono.kumiko@lab.ntt.co.jp,
        tachimoto.shinya@lab.ntt.co.jp
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Subject: [Sipping] E2M security and location conveyance
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Hello,

in case the authors of the folloing drafts are not in touch yet, I 
believe you should have a look at each other's work to make sure that 
the e2m security stuff is taking into account the location conveyance 
requirements. Specifically, Section 2 of the location conveyance draft 
directly talks about e2m security.

http://www.ietf.org/internet-drafts/draft-ietf-sipping-e2m-sec-reqs-01.txt

http://www.ietf.org/internet-drafts/draft-ietf-sipping-location-requirements-00.txt

Thanks,

Gonzalo


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Date: Wed, 25 Feb 2004 18:35:38 +0900
From: Kumiko Ono <ono.kumiko@lab.ntt.co.jp>
Subject: Re: [Sipping] Re: E2M security solution
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, volkerh@bell-labs.com
Cc: sipping <sipping@ietf.org>, tachimoto.shinya@lab.ntt.co.jp
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Hi Volker,

I have a question about security of the MIO. Let's consider for firewall 
control located with a proxy. The MIO header that you propose contains 
parts of SDP information. When UAs encrypt SDP for end-to-end in order 
to conceal IP addresses and port numbers, does the MIO also contain the 
same information to need to use e2e encryption?

Thanks,
Kumiko
>Hi,
>
>I agree with you about the e2m security solution could partly solve the 
>session-specific policies. 
>
>In my understanding, the policy draft proposes new headers to disclose 
>only the information to *all* proxies in the path. Some mechanism are 
>needed in order to disclose the information to *specific* proxies. The 
>e2m security can be applied as one of the mechanisms.
>
>Regards,
>
>Kumiko
>
>>Hello,
>>
>>The e2m security solution, described in the draft below,
>>http://www.ietf.org/internet-drafts/draft-ono-sipping-end2middle-security-
>>01.txt
>>
>>attempts to resolve some of the problems that the session-specific 
>>policies draft addresses:
>>http://www.ietf.org/internet-drafts/draft-hilt-sipping-session-spec-policy-
>>00.txt
>>
>>The need of a UA to obtain the policies that a particular proxy has is 
>>dealt with in both drafts. In addition, the e2m security draft talks 
>>about disclosing information to intermediaries. The policy draft 
>>addresses this issue and defines a means to disclose *only* the 
>>information needed by the proxy.
>>
>>My comments on the e2m requirements draft apply also to this one. We 
>>should try to align both efforts (session policies and e2m security).
>>
>>Thanks,
>>
>>Gonzalo
>>
>>
>>This communication is confidential and intended solely for the addressee(s)
>>. Any unauthorized review, use, disclosure or distribution is prohibited. 
>>If you believe this message has been sent to you in error, please notify 
>>the sender by replying to this transmission and delete the message without 
>>disclosing it. Thank you.
>>
>>E-mail including attachments is susceptible to data corruption, 
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>>send and receive e-mails on the basis that we are not liable for any such 
>>corruption, interception, amendment, tampering or viruses or any 
>>consequences thereof.
>
>--
>Kumiko Ono
>NTT Network Service Systems Labs.
>Tel +81 422 59 4508/Fax +81 422 60 4012
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP

--
Kumiko Ono
NTT Network Service Systems Labs.
Tel +81 422 59 4508/Fax +81 422 60 4012


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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
Cc: <sipping@ietf.org>, <knuettel@fokus.fraunhofer.de>
Subject: AW: AW: [Sipping] Roaming Architecture for SIP
Date: Wed, 25 Feb 2004 10:54:13 +0100
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Paul,

inline.

> > The scenario is of a user based e.g. in Berlin/Germany on a 
> business 
> > trip in Beijing/China. The user will want to both be 
> reachable under 
> > his home AOR (for that he registers with his HN)
> 
> I'm ok with that.
>

That's nice.
 
>  > as well as make use of the VN for local calls,
> > especially local PSTN breakout (this is why he wants to 
> have a valid 
> > registration with the VN).
> 
> Why? His billing relationship is with his HN. Why not let the 
> HN figure 
> out the best way to route this calls? If it is via the VN's gateways, 
> then it can route the calls to the VN.
> 

In all practical experience: if you and your VN is in China and your HN
is in Germany (or any similar setup), who would you want to route your
calls within China?
The HN has an important role to fulfil as the anchor of trust for the
billing relationship, however we don't think it neither feasible nor
realistic nor even desirable from an IETF point of view at this point in
time to have the HN fulfil routing decisions at the HN.
The OSP as defined by ETSI goes into this direction. However this seems
a far too carrier-centric model to be of use in the somewhat more
flexible world of the Internet and specifically ITSPs.
Furthermore, even if my HN has my billing relationship, why should we be
able to assume that this HN knows anything about the topography of VoIP
connections in China?

> > Furthermore, in a number of cases, one encounters 
> RTP-proxying, where 
> > around-the-world loops are to be avoided.
> 
> Certainly that is likely to be a bad thing. But does it require a 
> signaling identity with the VN to achieve that?
> 

If you put the question like that, no. However, in practice the
RTP-Proxy is normally in the domain/realm of the carrier handling the
PSTN break-out.
How would you suggest for the signalling to happen at the HN and the
media conversion at the VN if break-out is happening at the VN?
Pls. note that we do not assume a VoIP peering agreement between the two
networks (VN, HN). In fact the HN being a VoIP carrier is actually
optional. The emphasis is on the HN to be an "authorized billing
entity".

> > Another issue which we foresee is that in the near future hotels, 
> > airports or other semi-closed (walled garden) Internet 
> access points, 
> > will force users through their own VoIP (SIP) proxy infrastructure, 
> > similar to what a lot of sites aleady do in the case of HTTP.
> 
> Sure. But as long as those things are *proxies*, the UA 
> should still be 
> able to register with his HN.
> 
> Dealing with that depends a bit on how users will be forced. The way 
> hotels do it today depends on details of http that won't necessarily 
> work for sip.
> 

Either way, the "walled garden proxy" would probably/possibly want the
user to be registered - even if further registrations such as with the
HN are allowed. 
Again: there is nothing to be gained by putting routing and signalling
decisions for the local domain (i.e. China, VN in the example) to the
HN.
Conversely a lot is to be gained in terms of flexibility in following
the model.

	Harry


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


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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Alan Johnston <alan.johnston@mci.com>
CC: hisham.khartabil@nokia.com, sipping@ietf.org, oritl@microsoft.com,
        Miguel.An.Garcia@nokia.com
Subject: Re: [Sipping] Conferencing for UAs
References: <2038BCC78B1AD641891A0D1AE133DBB7017977C9@esebe019.ntc.nokia.com> <2038BCC78B1AD641891A0D1AE133DBB7017977C9@esebe019.ntc.nokia.com> <5.2.1.1.0.20040224084212.02819ef8@pop.mcit.com>
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Hi,

as an additional note, the following draft
http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00

proposes to carry a policy URI for the conference in a Call-Info 
header... what I am proposing is a more general mechanism that can carry 
all the URIs of the resources created by a request.

Otherwise, we would place the policy URI in the call-info header, the 
explosion URI in the associated-explosion header, the conference URI in 
the contact header, etc, etc... I want a cleaner way (e.g., a single 
header) to provide the URIs of the resources associated to a request.

Associated-Resource: sip:abc@a.com;purpose=conference-state
Associated-Resource: sip:def@a.com;purpose=explosion-state
Associated-Resource: sip:def@a.com;purpose=conference-policy


Thanks,

Gonzalo


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Date: Wed, 25 Feb 2004 09:29:11 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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CC: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
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Subject: Re: [Sipping] Config framework
References: <4039FEAD.5070708@ericsson.com> <403A01BD.7080905@cs.columbia.edu> <403A250F.9B29CE10@pingtel.com> <403A889F.4010601@cs.columbia.edu> <403C348D.21C43F72@pingtel.com>
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> I am not sure I follow this.  Perhaps if I elaborate, you can help me realize
> my confusion.  The three types of profile data are: local network, user and
> device data.  In the above example I was suggesting that in many environments
> the UA is not mobile or at least not frequently moved.  Hense the UA can
> be associated with a specific location.  In these environments it is convenient
> to combine the local network data and the device specific data.  I was not
> suggesting that the user data should be denormalized.  A user may have
> many devices.  The devices can all use the same user profile data.

I think we're in agreement. My suggestion is to simply logically 
separate the local network, user and device data and link those by 
pointers (URIs), rather than by inclusion. Thus, even if a stationary 
device is always associated a single user, it would reference rather 
than include the user information.

> 
> I am also questioning whether people think we really need three distinct
> types of profiles (local network, user and device).  Is there a real life
> environment where we need all three or can it be accomlished with
> two by combining parts of the third with the other two.  If people think
> we need all three then we need to have three different discovery models
> and ways to form three different AORs (for each profile type).  This is not
> difficult, I just want to be sure the extra moving parts are necessary.

While many devices will have one device = one user = one location, even 
within these enterprises there may well be classes of SIP endpoints that 
don't. Variations of hoteling, both the Hilton kind and the 
intra-company desk loaning, come to mind.

If I carry a device (laptop, PDA) with me from one branch to another, 
there will be device-based information that only depends on the device 
characteristics (media capabilities, user interface, network speed, 
etc.), some that depend on where I plug in the device (emergency call 
numbers, local dial plan, outbound proxy configuration, local media 
policy) and some that are the same for all my devices and only depend on 
my user identity (general media preferences, speed dial, buddy lists).

Note, for example, that media policy appears in all three configurations.

Since network, user and device are clearly distinguishable in practice, 
I'd argue that proponents of unifying two of them need to prove that 
they should *never* be treated separately, rather than the opposite.

> 
> There is nothing newer than this expired draft.  I was hoping to shake
> the dust off and restructure this document once the framework was
> last called.  I think the document identifies some good profile data
> requirements, but I think they should be sliced up a little differently.

The discussion here illustrates that it would be useful to have at least 
a rough set of parameters that we can use as test cases to see if they 
are accommodated well within the framework.

Henning

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CC: Alan Johnston <alan.johnston@mci.com>, hisham.khartabil@nokia.com,
        sipping@ietf.org, oritl@microsoft.com, Miguel.An.Garcia@nokia.com
Subject: Re: [Sipping] Conferencing for UAs
References: <2038BCC78B1AD641891A0D1AE133DBB7017977C9@esebe019.ntc.nokia.com> <2038BCC78B1AD641891A0D1AE133DBB7017977C9@esebe019.ntc.nokia.com> <5.2.1.1.0.20040224084212.02819ef8@pop.mcit.com> <403C76D9.2010401@ericsson.com>
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Bonzalo,

I am *slightly* warming to your point, but I am concerned that there is 
a slippery slope here and it will require a great deal of care to avoid 
heading down that slope.

I agree that policy URI for conferences is an analogous kind of usage. 
I'm not yet convinced that such an approach should be used often, or 
that we need a more general way to return such things. For cases where 
it is appropriate, call-info may be a sufficient mechanism.

I am definitely not convinced that we need separate URIs for requests 
within a dialog and outside of one. We are concurrently working on the 
GRUU proposal with the intent that we can use the same URI for both.

Nor am I convinced that we need separate URIs for INVITE and SUBSCRIBE, 
when the logical resource is the same for both. (E.g. join a conference 
vs request info (via sip) for the same conference.)

The key is determining when things are truly distinct resources. One 
hint is that they are addressed with different protocols. That is one 
justification for separating conference policy from conference focus.

Operations that do too much at once (e.g. create multiple distinct 
resources) raise a red flag for me - they aren't necessarily wrong, but 
they require extra consideration because they are inclined to be complex 
to understand. So hopefully there won't be many requests that need to 
reference multiple "output" resources.

	Paul

Gonzalo Camarillo wrote:
> Hi,
> 
> as an additional note, the following draft
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00 
> 
> 
> proposes to carry a policy URI for the conference in a Call-Info 
> header... what I am proposing is a more general mechanism that can carry 
> all the URIs of the resources created by a request.
> 
> Otherwise, we would place the policy URI in the call-info header, the 
> explosion URI in the associated-explosion header, the conference URI in 
> the contact header, etc, etc... I want a cleaner way (e.g., a single 
> header) to provide the URIs of the resources associated to a request.
> 
> Associated-Resource: sip:abc@a.com;purpose=conference-state
> Associated-Resource: sip:def@a.com;purpose=explosion-state
> Associated-Resource: sip:def@a.com;purpose=conference-policy
> 
> 
> Thanks,
> 
> Gonzalo
> 
> 
> This communication is confidential and intended solely for the 
> addressee(s). Any unauthorized review, use, disclosure or distribution 
> is prohibited. If you believe this message has been sent to you in 
> error, please notify the sender by replying to this transmission and 
> delete the message without disclosing it. Thank you.
> 
> E-mail including attachments is susceptible to data corruption, 
> interruption, unauthorized amendment, tampering and viruses, and we only 
> send and receive e-mails on the basis that we are not liable for any 
> such corruption, interception, amendment, tampering or viruses or any 
> consequences thereof.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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Subject: RE: [Sipping] updated dialog package
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Hi,

Comments in-line.

Thanks,

Paul.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]=20
> Sent: Tuesday, February 24, 2004 3:14 AM
> To: Paul Pepper
> Cc: Rohan Mahy; sipping
> Subject: Re: [Sipping] updated dialog package
>=20
>=20
>=20
> On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:
>=20
> > Hi,
> >
> > I work for a company that has developed handset gateway devices that
> > enable legacy, digital, business-grade telephones to be driven using
> > SIP.  For example, this enables an existing legacy PBX user=20
> > to retain
> > their telephones and cabling, but to replace the PBX by a=20
> > Softswitch or
> > IP Centrex system.  As most business-grade telephone=20
> > systems make use=20
> > of
> > a shared line facility, the addition of a shared line example to the
> > dialog event package ID is of great interest to us, and others who
> > support these types of feature.
> >
> > Although incorporating a shared line example is a step=20
> > towards a common
> > approach to shared line implementation, I don't believe it goes far
> > enough.  Our current implementation of shared lines has=20
> > shown that the
> > following issues need to be addressed by the spec, and=20
> > should be shown
> > in the examples:
> >
> > o Line seize, an essential part of any shared line=20
> > implementation, is
> > not adequately described. When more than one user agent attempts to
> > seize a line at approximately the same instant, a race condition=20
> > arises.
> > No description of how to handle the race is given - an example
> > application/dialog-info+xml NOTIFY payload is provided, but=20
> > this alone
> > is inadequate, as message sequencing is an important part=20
> > of seizing a
> > line.  The Line Seize condition will be covered in the
> > "anil-sipping-bla-draft" that Venkatesh refers to.
>=20
> Line seizure is well beyond what would be appropriate to=20
> standardize in=20
> SIPPING because for many types of UAs that concept does not=20
> even exist.=20
>   I was just trying to demonstrate a non-normative=20
> description of how it=20
> could be implemented. How you judge who "wins" a seizure race=20
> condition=20
> is an implementation specific issue.  In fact, its quite likely that=20
> different behaviors will be desired by different customers.  When=20
> talking about implementing this at Cisco, we concluded that a=20
> configuration knob was needed to satisfy essentially opposite=20
> requirements from two different customers.  Each customer was equally=20
> adamant that he was "right" and the other was "wrong".  As a=20
> result, I=20
> find it unlikely that SIPPING could come to come to consensus on what=20
> "proper seizing behavior" is even if we ignored the fact that=20
> the bulk=20
> of deployed SIP UAs are software UAs which have no lines to seize.
>


Yes, determining which user agents should win seize races is a matter of
policy, and implementation specific. However, the mechanism used to
convey whether a seize race has been won seems to be a matter for
standardisation (e.g. a non-200 final response to the NOTIFY requesting
the seize could indicate that the seize operation has failed).

> > o Handling of multiple instances of a shared line has been=20
> > overlooked.
> > On business-grade PBX systems a line will roughly=20
> > correspond to an AOR.
> > However, many such systems permit multiple "instances" of a=20
> > shared line
> > to appear on a single endpoint. For example, a number of=20
> > instances of a
> > "product support" shared line might appear on all the=20
> > endpoints of the
> > product support group, thereby allowing multiple calls to=20
> > be taken and
> > passed around within the group.
>=20
> This has not been overlooked at all.  You can have multiple=20
> subscriptions (one per AOR), or you can subscribe to a list of AORs=20
> with the event list extension:
>=20
> =09
> http://www.ietf.org/internet-drafts/draft-ietf-simple-event-li
> st-04.txt
>=20

I haven't been sufficiently clear on this point. As you point out, each
shared AOR will have its own subscription. Take an example AOR of
support@example. Although a single registration and two subscriptions
(one in each direction, as subscriber and notifier, if a state agent is
used) are used per AOR on each user agent, multiple calls to/from that
AOR may be active at any time. If a user agent has the shared line
capability then all calls to support@example will be presented on it by,
say, illuminating one button per call. (Of course this is constrained to
varying degrees by the user agent's display capabilities.) As each
button shows a call to support@example the calls must be presented on
those buttons in a consistent manner across all user agents on the
network. This will allow Harry to put a call to support@example on hold
and verbally tell Thelma which button to press in order to pick that
call up. If all the calls to support@example are presented in an
arbitrary manner on different user agents, then the coherency of call
presentation is lost. This is a problem as coherent call presentation is
a requirement of existing PBX users, and gives the only meaningful use
for this feature.

One way of achieving a consistent mapping of calls to buttons (if
buttons are the user interface mechanism used) is to index buttons on
all the user agents. That way, a call can be mapped to the specific
button on all user agents of an installation.


> > o The mechanism used to convey that a call has been placed on hold
> > appears to rely upon implementation-specific parameters,=20
> > found in the
> > xml's <param> element. I can't see any reference to=20
> > definitions of the
> > values used by this element's pname and pval attributes
> > (pname=3D"+activity" pval=3D"noninteractive"), so I assume they're
> > non-standard.
>=20
> This was due to distraction on my part.  I had intended to include a=20
> discussion of this in the current draft, but somehow left it=20
> out at 2am=20
> the day of the draft deadline.  I'm terribly sorry about that.  I'll=20
> send out more about the hold issue tomorrow.
>=20
> thanks,
> -rohan
>=20
> > Thanks,
> >
> > Paul.
> >
> >> -----Original Message-----
> >> From: Rohan Mahy [mailto:rohan@cisco.com]
> >> Sent: Thursday, February 19, 2004 6:00 PM
> >> To: sipping
> >> Cc: rohan@cisco.com
> >> Subject: [Sipping] updated dialog package
> >>
> >>
> >> Hi Everyone,
> >>
> >> I've updated the dialog packages with the changes we agreed to in
> >> Minneapolis (removing CSeq and the Route-Set).
> >>
> >> I've also provided a detailed example of using the dialog
> >> package for a
> >> "shared-line" application.  I apologize for my tardiness=20
> >> here.  I was
> >> supposed to post something shortly after the meeting.  I=20
> >> any case, I
> >> believe this demonstrates that the current syntax is at least
> >> sufficient for the use cases we've discussed.  I would like to hear
> >> from all of you if this does the trick.
> >>
> >> many thanks,
> >> -rohan
> >>
> >>
> >> _______________________________________________
> >> Sipping mailing list =20
> >> https://www1.ietf.org/mailman/listinfo/sipping
> >> This list=20
> >> is for NEW development of the application of SIP
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sip@ietf.org for new developments of core SIP
> >>
>=20
>=20

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



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Subject: Re: [Sipping] Conferencing for UAs
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Paul,

I believe we are in agreement. You use different URIs for different 
resources. Additionally, it is not wise to overload too much a request 
so that it creates too many resources. We agree on these two points.

In any event, an INVITE that creates a conference, an explosion, and a 
policy URI is not such a strange example. Another example that covers 
what you said about different protocols would be an INVITE that, in 
addition to creating a conference (identified by a SIP URI), creates a 
mailing list for the conference (indentified by a mailto: URI).

In addition, the mechanism used to convery the URIs of these resources 
to the UAC should not be INVITE specific, because requests such as 
MESSAGE can also trigger the creation of new resources (e.g., a 
explosion of MESSAGEs).

Thanks,

Gonzalo

Paul Kyzivat wrote:

> Bonzalo,
> 
> I am *slightly* warming to your point, but I am concerned that there is 
> a slippery slope here and it will require a great deal of care to avoid 
> heading down that slope.
> 
> I agree that policy URI for conferences is an analogous kind of usage. 
> I'm not yet convinced that such an approach should be used often, or 
> that we need a more general way to return such things. For cases where 
> it is appropriate, call-info may be a sufficient mechanism.
> 
> I am definitely not convinced that we need separate URIs for requests 
> within a dialog and outside of one. We are concurrently working on the 
> GRUU proposal with the intent that we can use the same URI for both.
> 
> Nor am I convinced that we need separate URIs for INVITE and SUBSCRIBE, 
> when the logical resource is the same for both. (E.g. join a conference 
> vs request info (via sip) for the same conference.)
> 
> The key is determining when things are truly distinct resources. One 
> hint is that they are addressed with different protocols. That is one 
> justification for separating conference policy from conference focus.
> 
> Operations that do too much at once (e.g. create multiple distinct 
> resources) raise a red flag for me - they aren't necessarily wrong, but 
> they require extra consideration because they are inclined to be complex 
> to understand. So hopefully there won't be many requests that need to 
> reference multiple "output" resources.
> 
>     Paul
> 
> Gonzalo Camarillo wrote:
> 
>> Hi,
>>
>> as an additional note, the following draft
>> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri-call-info-purpose-00 
>>
>>
>> proposes to carry a policy URI for the conference in a Call-Info 
>> header... what I am proposing is a more general mechanism that can 
>> carry all the URIs of the resources created by a request.
>>
>> Otherwise, we would place the policy URI in the call-info header, the 
>> explosion URI in the associated-explosion header, the conference URI 
>> in the contact header, etc, etc... I want a cleaner way (e.g., a 
>> single header) to provide the URIs of the resources associated to a 
>> request.
>>
>> Associated-Resource: sip:abc@a.com;purpose=conference-state
>> Associated-Resource: sip:def@a.com;purpose=explosion-state
>> Associated-Resource: sip:def@a.com;purpose=conference-policy
>>
>>
>> Thanks,
>>
>> Gonzalo
>>
>>
>> This communication is confidential and intended solely for the 
>> addressee(s). Any unauthorized review, use, disclosure or distribution 
>> is prohibited. If you believe this message has been sent to you in 
>> error, please notify the sender by replying to this transmission and 
>> delete the message without disclosing it. Thank you.
>>
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>> only send and receive e-mails on the basis that we are not liable for 
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>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>

-- 
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Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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Date: Wed, 25 Feb 2004 09:44:18 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conferencing for UAs
In-reply-to: <403C530E.1070204@ericsson.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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At 09:47 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
>Hi Alan,
>
>I understand that my comments come late in the game and that it might be 
>impossible to change the conferencing framework at this point. In any 
>event, I do not agree with your statement below:
>
>>No - this is the problem.  You are looking at your single narrow application.
>
>I believe I am actually looking at a much wider range of applications than 
>just conferencing. So, to me, the problem is that the conferencing 
>framework is too focused on a "single narrow application"; conferencing.

Gonzalo,

If you are saying that some of your exploder services do not have a 
conferencing component, then I agree.  Some of the exploder services seem 
to just be a "request to fork" to a set of URIs.  Others definitely have 
conferencing semantics.  The ones with conferencing semantics, I believe, 
can be, and should be modeled using the conferencing framework.

The problem is that each method seems to have a different set of semantics 
as to what exploding that method means - INVITE is handled differently from 
SUBSCRIBE, from MESSAGE.   From a high level, exploders may seem to be a 
consistent set of services.  However, as I delve into the details and think 
about call flows (something  absent from any of the exploder drafts), a 
different story emerges.

I think we need to explore exactly the services and call flows that are 
being envisioned for exploders before proposing major changes to our 
conferencing framework, which I believe is well understood and documented.

>In a conferencing service, there is a single resource created by the 
>initial INVITE; a conference. That's why the server can use the Contact 
>header field to send it back to the UAC. If you think of services that 
>create more than one resource (e.g., an explosion and a conference), the 
>server may not want to use the same URI for both. Moreover, conceptually, 
>is is not a good thing to use the same URI to identify both resources.

As Paul has pointed out - they are the same resource.  Additionally, it is 
a trivial SIP routing issue to fix this - many proxies do method-based 
request routing - that would completely solve your problem of sending 
SUBSCRIBEs to a different UA than the initial request.

>So, my main concern is the semantics overload of the Contact header. (We 
>abused this header when design REGISTER as well, by the way. That's why I 
>do not want to do it again.)
>
>In my opinion, the Contact header is the URI where you send subsequent 
>requests within the dialog. If the server wants to tell the client that a 
>resource (e.g., a conference) has been created, it can use a different 
>header to include the URI of such a resource. Note that, if you want both 
>URIs to be the same (like in the conferencing case), you can do it.
>
>Contact: sip:1234@a.com;isfocus
>Resource: sip:1234@a.com;isfocus
>
>An entity receiving these headers would use the URI in the Contact header 
>to send subsequent requests *within* the dialog, and the URI in the other 
>header to send requests *outside* that dialog (e.g., SUBSCRIBE to the 
>conference package).

No.  A GRUU should be used in the Contact, and its routing is valid outside 
the dialog.


>In brief, instead of sticking the conference URI in a Contact header and 
>label that URI with an isfocus parameter so that the UAC figures what is 
>going on, I find it more explicit to use a different header. Although, as 
>I said, I understand that this comment comes late and that may not be 
>useful at this point.

I think much more work is needed before the working group can evaluate the 
exploders work and see how and if it fits in with existing concepts and 
approaches.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

>Thanks,
>
>Gonzalo
>
>
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At 10:50 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
>Hello,
>
>in case the authors of the folloing drafts are not in touch yet, I believe 
>you should have a look at each other's work to make sure that the e2m 
>security stuff is taking into account the location conveyance 
>requirements. Specifically, Section 2 of the location conveyance draft 
>directly talks about e2m security.

yep - it appears that if the e2m security allows the hiding of the location 
information (LI) from a proxy that needs that information in order to 
properly route the call (to the ERC) - then this is a problem (and 
inconsistency between the goals of the 2 IDs/efforts).

I can imagine a case in which the proxy inserts LI into the message body 
thinking it knows where the UA is - leaving two LIs in the body (perhaps 
not labelled with who inserted each) for the ERC to figure out which to 
trust for that call

Do we suggests a special case where the UA trust all proxies in a scenario 
that the UA understands the new INVITE is to an ERC (via the sos@ 
indication or some other emergency number dialed on the (hard or soft) phone)?

We did have this special case written into the Geopriv Requirements doc 
(RFC 3693, Section 8.3) with Location Conveyance in mind.

comments


>http://www.ietf.org/internet-drafts/draft-ietf-sipping-e2m-sec-reqs-01.txt
>
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-location-requirements-00.txt
>
>Thanks,
>
>Gonzalo
>
>
>This communication is confidential and intended solely for the 
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>prohibited. If you believe this message has been sent to you in error, 
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>


cheers,
James

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


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On Feb 23, 2004, at 3:11 PM, Henning Schulzrinne wrote:
>
>> This is certainly not the intension.  The draft specifies a way to 
>> separately
>> subscribe for device or user profile data.  I would be happy to 
>> clarify this
>> where you think the draft is vague or implies this restriction.
>
>> Section 3.2 describes parameters that are device specific as even in 
>> the
>> ideal world, purely user specific data may need to consider the UA
>> implementation that it is to be applied to.  Section 3.2 also 
>> specifies a
>> means to indicate the desire for user or device data.
>> Section 3.13 specifies how to form the SUBSCRIBE URI based upon
>> the interest for either a device or user profile.
>
> 3.13 and 4.1 don't really spell out that user (as opposed to device) 
> data may need to go elsewhere. It only has a single set of steps that 
> aborts as soon as (say) DHCP-based discovery succeeds.
>
> I think the two modes, user and device, are somewhat dissimilar. For 
> device, I may need to combine local information (e.g., information 
> about outbound proxies and local dialplans) with home information 
> (local dialplan, device-specific UI configuration, etc.).

Henning,

You are presenting an extremely rigid and limited view of what 
constitutes local vs. home information.  There is certainly a need for 
*specific implementations* to sort various parameters into what they 
will allow to be configured in each instance, but you will certainly 
not gain consensus on this list that the things listed above are one of 
the other.

Case in point, there are valid reasons why outbound proxy should be set 
via discovery, device, and user.  Likewise, in a thread on emergency 
services, we discussed why local dialplan (sos numbers) can be based on 
where you are OR who you are, but not both simultaneously.

> For user configuration, I probably need to assume that the device has 
> an AOR, either initially or discovered via device configuration, which 
> is realistic for laptops and other personal devices.

no, you don't need to assume that at all.  If you have an AOR, feel 
free to use it.  If you don't, but you can discover one, you should 
also feel free to do so.

> 4.1 does not seem to spell out what to do if either the device or the 
> user knows the AOR, but wants to get the remainder of the device and 
> user configuration automatically.
>
> Thus, I'd suggest separating the two discovery models and possibly 
> logically separating user configuration as being derived (among other 
> sources of AORs) from the device configuration, as in:
>
> DHCP -> device configuration query -> AOR -> user-based query
>
>
>
>> These 3 important concepts are intended to be supported in
>> draft-ietf-sipping-config-framework-02.  It is however difficult in
>> practice to keep these three things separate.  In many environments
>> it is easier to keep two of the three separate.
>> For example:
>> In an enterprise with physical devices that do not move much, it 
>> becomes
>> easier to combine the properties of 1 related to location with the 
>> properties
>> of 3.
>
> I'd rather be more general; if the user configuration spans multiple 
> devices, it's a lot cleaner if I only have to update one server rather 
> than each device-based server. (Device-based servers may well be more 
> local than user-based servers.)
>
>
>> When a user does not know their AOR or does not want to have to 
>> provide
>> it to the UA (to provide a plug and play experience), the user's AOR 
>> can be
>> associated with 3 (the device).  2 is may then be subscribe 
>> implicitly or explicitly.
>
> In my two-stage model, this is supported.

I don't think you have really articulated what constitutes your 2-stage 
model.  Can you elaborate please?

>
>> I think that this illustrates a very good point that the separation 
>> between
>> user and device profile data cannot be made concrete.  Data such as
>> out bound proxy, or policy may be dictated by any or all of the 3
>> realms defined above.  Only the application and local administrative
>> policies can determine the realm to which data should be associated.
>> In addition it may be desirable to override some of the user or device
>> profile data based upon other local discovery data (e.g. DHCP options
>> for out bound proxy, STUN for NAT traversal).
>
> There may well be information that could be considered both, but in my 
> book, user-based configuration is independent of where I plug in my 
> device and does not depend on the device itself. Thus, an outbound 
> proxy and related transport parameters are clearly device-related, 
> while speed dial functionality, media preferences, home dial plans and 
> buddy lists are probably not.

I can present a counter-example for each of your examples.  For 
example, I can have an outbound proxy that records my call history 
independent of my device.  Speed dial functionality is not completely 
device independent as I may render speed dials for example using soft 
buttons on some devices, and as sequences of digits on another.  
Finally media preferences depend intimately on the capabilities of my 
device.

> Some parameters may be hierarchical, in that local network 
> configuration overrides my general configuration. If my visited 
> network doesn't allow video, that takes precedence over my user 
> configuration that prefers video calls.
>
>
>> At best we can define a precedence by which the device data
>> can be overridden by user data (or the other way around)
>> along with the ability for administrators to dictate policy (perhaps 
>> indicated
>> in the profile data) over what can be overridden.
>
> I agree. I think we'll have both directions of override. For example, 
> a CIA agent user configuration may mandate use of TLS and AES 
> encryption and accept that a call will fail in some local environments 
> that don't allow this.
>
> Having a sample set of configuration parameters that we can work 
> through might test our assumptions.  
> draft-stredicke-sipping-ep-config-00.txt seems to be the last one on 
> this topic, or is there something else?
>
> It unfortunately conflates local parameters (e.g., outbound proxy, 
> 3.1.4.3) with user-based parameters (registration period, 3.1.4.1).

again, your case for local vs. user parameters is very thin.  you may 
want to reconsider if your taxonomy if real or just an artifact of the 
fact that these things were divided in the past.

thanks,
-rohan

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


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Subject: Re: [Sipping] updated dialog package
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On Feb 25, 2004, at 7:32 AM, Paul Pepper wrote:

> Hi,
>
> Comments in-line.
>
> Thanks,
>
> Paul.
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@cisco.com]
>> Sent: Tuesday, February 24, 2004 3:14 AM
>> To: Paul Pepper
>> Cc: Rohan Mahy; sipping
>> Subject: Re: [Sipping] updated dialog package
>>
>>
>>
>> On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:
>>
>>> Hi,
>>>
>>> I work for a company that has developed handset gateway devices that
>>> enable legacy, digital, business-grade telephones to be driven using
>>> SIP.  For example, this enables an existing legacy PBX user
>>> to retain
>>> their telephones and cabling, but to replace the PBX by a
>>> Softswitch or
>>> IP Centrex system.  As most business-grade telephone
>>> systems make use
>>> of
>>> a shared line facility, the addition of a shared line example to the
>>> dialog event package ID is of great interest to us, and others who
>>> support these types of feature.
>>>
>>> Although incorporating a shared line example is a step
>>> towards a common
>>> approach to shared line implementation, I don't believe it goes far
>>> enough.  Our current implementation of shared lines has
>>> shown that the
>>> following issues need to be addressed by the spec, and
>>> should be shown
>>> in the examples:
>>>
>>> o Line seize, an essential part of any shared line
>>> implementation, is
>>> not adequately described. When more than one user agent attempts to
>>> seize a line at approximately the same instant, a race condition
>>> arises.
>>> No description of how to handle the race is given - an example
>>> application/dialog-info+xml NOTIFY payload is provided, but
>>> this alone
>>> is inadequate, as message sequencing is an important part
>>> of seizing a
>>> line.  The Line Seize condition will be covered in the
>>> "anil-sipping-bla-draft" that Venkatesh refers to.
>>
>> Line seizure is well beyond what would be appropriate to
>> standardize in
>> SIPPING because for many types of UAs that concept does not
>> even exist.
>>   I was just trying to demonstrate a non-normative
>> description of how it
>> could be implemented. How you judge who "wins" a seizure race
>> condition
>> is an implementation specific issue.  In fact, its quite likely that
>> different behaviors will be desired by different customers.  When
>> talking about implementing this at Cisco, we concluded that a
>> configuration knob was needed to satisfy essentially opposite
>> requirements from two different customers.  Each customer was equally
>> adamant that he was "right" and the other was "wrong".  As a
>> result, I
>> find it unlikely that SIPPING could come to come to consensus on what
>> "proper seizing behavior" is even if we ignored the fact that
>> the bulk
>> of deployed SIP UAs are software UAs which have no lines to seize.
>>
>
>
> Yes, determining which user agents should win seize races is a matter 
> of
> policy, and implementation specific. However, the mechanism used to
> convey whether a seize race has been won seems to be a matter for
> standardisation (e.g. a non-200 final response to the NOTIFY requesting
> the seize could indicate that the seize operation has failed).

Now I see what you are getting at.  The particular mechanism that you 
mention is unlikely to go over favorably because it would have the side 
effect of terminating your subscription.  However, I agree it would be 
useful to have a standard way to convey something like this.  I'm not 
convinced that this belongs in the dialog package, but I'll think about 
this.


>>> o Handling of multiple instances of a shared line has been
>>> overlooked.
>>> On business-grade PBX systems a line will roughly
>>> correspond to an AOR.
>>> However, many such systems permit multiple "instances" of a
>>> shared line
>>> to appear on a single endpoint. For example, a number of
>>> instances of a
>>> "product support" shared line might appear on all the
>>> endpoints of the
>>> product support group, thereby allowing multiple calls to
>>> be taken and
>>> passed around within the group.
>>
>> This has not been overlooked at all.  You can have multiple
>> subscriptions (one per AOR), or you can subscribe to a list of AORs
>> with the event list extension:
>>
>> 	
>> http://www.ietf.org/internet-drafts/draft-ietf-simple-event-li
>> st-04.txt
>>
>
> I haven't been sufficiently clear on this point. As you point out, each
> shared AOR will have its own subscription. Take an example AOR of
> support@example. Although a single registration and two subscriptions
> (one in each direction, as subscriber and notifier, if a state agent is
> used) are used per AOR on each user agent, multiple calls to/from that
> AOR may be active at any time. If a user agent has the shared line
> capability then all calls to support@example will be presented on it 
> by,
> say, illuminating one button per call. (Of course this is constrained 
> to
> varying degrees by the user agent's display capabilities.) As each
> button shows a call to support@example the calls must be presented on
> those buttons in a consistent manner across all user agents on the
> network. This will allow Harry to put a call to support@example on hold
> and verbally tell Thelma which button to press in order to pick that
> call up. If all the calls to support@example are presented in an
> arbitrary manner on different user agents, then the coherency of call
> presentation is lost. This is a problem as coherent call presentation 
> is
> a requirement of existing PBX users, and gives the only meaningful use
> for this feature.
>
> One way of achieving a consistent mapping of calls to buttons (if
> buttons are the user interface mechanism used) is to index buttons on
> all the user agents. That way, a call can be mapped to the specific
> button on all user agents of an installation.

I expect that any such index would be conveyed as a target URI 
parameter using a device ID of some sort.  There is agenda time 
scheduled in SIPPING to discuss Instance ID requirements with the 
following reading list:

draft-stucker-sip-guid-00.txt
draft-jennings-sipping-instance-id-00.txt
draft-ietf-xmpp-core-22.txt (section 2.3)

thanks,
-rohan


>>> o The mechanism used to convey that a call has been placed on hold
>>> appears to rely upon implementation-specific parameters,
>>> found in the
>>> xml's <param> element. I can't see any reference to
>>> definitions of the
>>> values used by this element's pname and pval attributes
>>> (pname="+activity" pval="noninteractive"), so I assume they're
>>> non-standard.
>>
>> This was due to distraction on my part.  I had intended to include a
>> discussion of this in the current draft, but somehow left it
>> out at 2am
>> the day of the draft deadline.  I'm terribly sorry about that.  I'll
>> send out more about the hold issue tomorrow.
>>
>> thanks,
>> -rohan
>>
>>> Thanks,
>>>
>>> Paul.
>>>
>>>> -----Original Message-----
>>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>>> Sent: Thursday, February 19, 2004 6:00 PM
>>>> To: sipping
>>>> Cc: rohan@cisco.com
>>>> Subject: [Sipping] updated dialog package
>>>>
>>>>
>>>> Hi Everyone,
>>>>
>>>> I've updated the dialog packages with the changes we agreed to in
>>>> Minneapolis (removing CSeq and the Route-Set).
>>>>
>>>> I've also provided a detailed example of using the dialog
>>>> package for a
>>>> "shared-line" application.  I apologize for my tardiness
>>>> here.  I was
>>>> supposed to post something shortly after the meeting.  I
>>>> any case, I
>>>> believe this demonstrates that the current syntax is at least
>>>> sufficient for the use cases we've discussed.  I would like to hear
>>>> from all of you if this does the trick.
>>>>
>>>> many thanks,
>>>> -rohan
>>>>
>>>>
>>>> _______________________________________________
>>>> Sipping mailing list
>>>> https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list
>>>> is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>>
>>
>>


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



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Miguel.An.Garcia@nokia.com wrote:
>>From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>Sent: 24 February, 2004 19:47
>>To: Gonzalo Camarillo
>>Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org;
>>alan.johnston@mci.com; oritl@microsoft.com; Garcia Miguel.An
>>(Nokia-NRC/Helsinki)
>>Subject: Re: [Sipping] Conferencing for UAs
> 
> [snap]
> 
>>Taken to extreme, what you suggest would lead to a different 
>>address for 
>>OPTIONS, for MESSAGE, for PUBLISH, for REGISTER, etc. And maybe 
>>different addresses for each kind of subscription.
>>
>>I don't want to go there.
>>
>>	Paul
> 
> Aren't already there with Caller Preferences?

No - I don't see the parallel at all.

	Paul


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Christer Holmberg (JO/LMF) wrote:

> Hi,
> 
> I guess this is a clarification question on kpml, as well as on gruu
> (and functions where gruu is used) in general.
> 
> If you have a node that doesn't register, but eg wants to use kpml (a
> gateway, for example), could it generate the gruu on its own, or is
> it "mandated" to use the register mechanism? If the register
> mechanism is not used, are there any issues which could come up?

Use of the registration mechanism is not mandated; the spec says that 
this is one way, but not the only possible way, to get a gruu. However, 
no other mechanisms are standardized at this time.

Generating your own gruu is probably a bad idea, since it is extremely 
hard to guarantee that it is indeed globally routable.


-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: [Sipping] Re: Conferencing for UAs
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I think the same URI makes sense. At the core, there really is one 
resource here - the conference. All that differs is the methods that are 
invoked on it and how that resource behaves when users contact it. 
Subscribing to it and initiating a session to it are two different 
methods, but both against fundamentally the same resource.

-Jonathan R.

Gonzalo Camarillo wrote:

> Hi,
> 
>> Aren't we heading to defining a generic Associated-State header with
>> a "role" parameter that contains a value "transaction", "conference",
>> "xxx", etc...
> 
> 
> I actually do not care how we encode this information; in a header with 
> several parameters or in multiple headers (in any case, you may be right 
> that using a single header could be cleaner).
> 
> What I want to know is whether or not people think that it is worthwhile 
> keeping separate the URI used for the INVITE dialog and the one (or 
> ones) used for subscriptions to some piece of state that is associated 
> somehow with that dialog. My personal opinion is that we should keep 
> them separate.
> 
> Gonzalo
> 
> 
> This communication is confidential and intended solely for the 
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> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Subject: RE: [Sipping] updated dialog package
Date: Wed, 25 Feb 2004 11:51:09 -0800
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Thread-Topic: [Sipping] updated dialog package
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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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Also while we are at the topic of being able to reject NOTIFY's without =
terminating subscriptions, another useful capability to have would be =
for a UA to reject a NOTIFY with a specific response code so that any =
NOTIFY's for that dialog alone can be supressed. For example with SLA =
there may be UA's in the group that are of varying capabilities. For =
example UA1 can have capabilities to display say 2 active dialogs at any =
point, while UA2 can have capabilities that permit it to show upto 5 =
active dialogs. When both the UA's are subscribed to dialog state, they =
receive dialog states of all dialogs. Now, if UA1 had an ability to =
respond to a NOTIFY for a third active dialog saying "Dont terminate my =
subscriptions; I want to continue getting notifications for all dialogs, =
but supress anything for this specific dialog as I have reached my =
limits", then it would avoid un-necessary NOTIFY overheads being sent =
towards UA1 and it having to accept it and throwing it away; keep track =
of what dialogs it has accepted without UI, so that all new NOTIFY's =
towards this dialog id are continued to be discarded silently ......

Venkatesh

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Rohan Mahy
Sent: Wednesday, February 25, 2004 9:29 AM
To: Paul Pepper
Cc: Rohan Mahy; sipping
Subject: Re: [Sipping] updated dialog package



On Feb 25, 2004, at 7:32 AM, Paul Pepper wrote:

> Hi,
>
> Comments in-line.
>
> Thanks,
>
> Paul.
>
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan@cisco.com]
>> Sent: Tuesday, February 24, 2004 3:14 AM
>> To: Paul Pepper
>> Cc: Rohan Mahy; sipping
>> Subject: Re: [Sipping] updated dialog package
>>
>>
>>
>> On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:
>>
>>> Hi,
>>>
>>> I work for a company that has developed handset gateway devices that
>>> enable legacy, digital, business-grade telephones to be driven using
>>> SIP.  For example, this enables an existing legacy PBX user
>>> to retain
>>> their telephones and cabling, but to replace the PBX by a
>>> Softswitch or
>>> IP Centrex system.  As most business-grade telephone
>>> systems make use
>>> of
>>> a shared line facility, the addition of a shared line example to the
>>> dialog event package ID is of great interest to us, and others who
>>> support these types of feature.
>>>
>>> Although incorporating a shared line example is a step
>>> towards a common
>>> approach to shared line implementation, I don't believe it goes far
>>> enough.  Our current implementation of shared lines has
>>> shown that the
>>> following issues need to be addressed by the spec, and
>>> should be shown
>>> in the examples:
>>>
>>> o Line seize, an essential part of any shared line
>>> implementation, is
>>> not adequately described. When more than one user agent attempts to
>>> seize a line at approximately the same instant, a race condition
>>> arises.
>>> No description of how to handle the race is given - an example
>>> application/dialog-info+xml NOTIFY payload is provided, but
>>> this alone
>>> is inadequate, as message sequencing is an important part
>>> of seizing a
>>> line.  The Line Seize condition will be covered in the
>>> "anil-sipping-bla-draft" that Venkatesh refers to.
>>
>> Line seizure is well beyond what would be appropriate to
>> standardize in
>> SIPPING because for many types of UAs that concept does not
>> even exist.
>>   I was just trying to demonstrate a non-normative
>> description of how it
>> could be implemented. How you judge who "wins" a seizure race
>> condition
>> is an implementation specific issue.  In fact, its quite likely that
>> different behaviors will be desired by different customers.  When
>> talking about implementing this at Cisco, we concluded that a
>> configuration knob was needed to satisfy essentially opposite
>> requirements from two different customers.  Each customer was equally
>> adamant that he was "right" and the other was "wrong".  As a
>> result, I
>> find it unlikely that SIPPING could come to come to consensus on what
>> "proper seizing behavior" is even if we ignored the fact that
>> the bulk
>> of deployed SIP UAs are software UAs which have no lines to seize.
>>
>
>
> Yes, determining which user agents should win seize races is a matter=20
> of
> policy, and implementation specific. However, the mechanism used to
> convey whether a seize race has been won seems to be a matter for
> standardisation (e.g. a non-200 final response to the NOTIFY =
requesting
> the seize could indicate that the seize operation has failed).

Now I see what you are getting at.  The particular mechanism that you=20
mention is unlikely to go over favorably because it would have the side=20
effect of terminating your subscription.  However, I agree it would be=20
useful to have a standard way to convey something like this.  I'm not=20
convinced that this belongs in the dialog package, but I'll think about=20
this.


>>> o Handling of multiple instances of a shared line has been
>>> overlooked.
>>> On business-grade PBX systems a line will roughly
>>> correspond to an AOR.
>>> However, many such systems permit multiple "instances" of a
>>> shared line
>>> to appear on a single endpoint. For example, a number of
>>> instances of a
>>> "product support" shared line might appear on all the
>>> endpoints of the
>>> product support group, thereby allowing multiple calls to
>>> be taken and
>>> passed around within the group.
>>
>> This has not been overlooked at all.  You can have multiple
>> subscriptions (one per AOR), or you can subscribe to a list of AORs
>> with the event list extension:
>>
>> =09
>> http://www.ietf.org/internet-drafts/draft-ietf-simple-event-li
>> st-04.txt
>>
>
> I haven't been sufficiently clear on this point. As you point out, =
each
> shared AOR will have its own subscription. Take an example AOR of
> support@example. Although a single registration and two subscriptions
> (one in each direction, as subscriber and notifier, if a state agent =
is
> used) are used per AOR on each user agent, multiple calls to/from that
> AOR may be active at any time. If a user agent has the shared line
> capability then all calls to support@example will be presented on it=20
> by,
> say, illuminating one button per call. (Of course this is constrained=20
> to
> varying degrees by the user agent's display capabilities.) As each
> button shows a call to support@example the calls must be presented on
> those buttons in a consistent manner across all user agents on the
> network. This will allow Harry to put a call to support@example on =
hold
> and verbally tell Thelma which button to press in order to pick that
> call up. If all the calls to support@example are presented in an
> arbitrary manner on different user agents, then the coherency of call
> presentation is lost. This is a problem as coherent call presentation=20
> is
> a requirement of existing PBX users, and gives the only meaningful use
> for this feature.
>
> One way of achieving a consistent mapping of calls to buttons (if
> buttons are the user interface mechanism used) is to index buttons on
> all the user agents. That way, a call can be mapped to the specific
> button on all user agents of an installation.

I expect that any such index would be conveyed as a target URI=20
parameter using a device ID of some sort.  There is agenda time=20
scheduled in SIPPING to discuss Instance ID requirements with the=20
following reading list:

draft-stucker-sip-guid-00.txt
draft-jennings-sipping-instance-id-00.txt
draft-ietf-xmpp-core-22.txt (section 2.3)

thanks,
-rohan


>>> o The mechanism used to convey that a call has been placed on hold
>>> appears to rely upon implementation-specific parameters,
>>> found in the
>>> xml's <param> element. I can't see any reference to
>>> definitions of the
>>> values used by this element's pname and pval attributes
>>> (pname=3D"+activity" pval=3D"noninteractive"), so I assume they're
>>> non-standard.
>>
>> This was due to distraction on my part.  I had intended to include a
>> discussion of this in the current draft, but somehow left it
>> out at 2am
>> the day of the draft deadline.  I'm terribly sorry about that.  I'll
>> send out more about the hold issue tomorrow.
>>
>> thanks,
>> -rohan
>>
>>> Thanks,
>>>
>>> Paul.
>>>
>>>> -----Original Message-----
>>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>>> Sent: Thursday, February 19, 2004 6:00 PM
>>>> To: sipping
>>>> Cc: rohan@cisco.com
>>>> Subject: [Sipping] updated dialog package
>>>>
>>>>
>>>> Hi Everyone,
>>>>
>>>> I've updated the dialog packages with the changes we agreed to in
>>>> Minneapolis (removing CSeq and the Route-Set).
>>>>
>>>> I've also provided a detailed example of using the dialog
>>>> package for a
>>>> "shared-line" application.  I apologize for my tardiness
>>>> here.  I was
>>>> supposed to post something shortly after the meeting.  I
>>>> any case, I
>>>> believe this demonstrates that the current syntax is at least
>>>> sufficient for the use cases we've discussed.  I would like to hear
>>>> from all of you if this does the trick.
>>>>
>>>> many thanks,
>>>> -rohan
>>>>
>>>>
>>>> _______________________________________________
>>>> Sipping mailing list
>>>> https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list
>>>> is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>>
>>
>>


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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_______________________________________________
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Date: Wed, 25 Feb 2004 15:01:18 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
CC: sipping <sipping@ietf.org>, Cullen Jennings <fluffy@cisco.com>,
        bstucker@nortelnetworks.com
Subject: Re: [Sipping] UA indentification and HIP
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Thanks for the brief intro, Gonzalo. I need to read more on HIP and its 
on my now-very-length reading list for the plane.

Let me anyway ask the main question I have not understood, which is how 
address resolution is done. So, lets say I want to send a message to a 
host, and I have obtained its HIT somehow. Presumably I need to 
translate that to the IP for the host. How is that done? If its DNS, 
then what is the difference between HIP and just using dynamic DNS and 
assigning a hostname to each host?

Or, is the proposal that a HIT is an address, not a name, in which case 
routes to it are injected into the IP routing infrastructure. I hope, 
and assume, that this is not the case, since thats equivalent to a host 
route for every host on the Internet... egads!

-Jonathan R.

Gonzalo Camarillo wrote:

> Folks,
> 
> the following two drafts describe how to indentify a particular UA using 
> globally unique identifiers:
> 
> http://www.ietf.org/internet-drafts/draft-stucker-sip-guid-00.txt
> http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-00.txt 
> 
> 
> Cullen pointed out that this work may be able to take advantage of the 
> work carried out in the HIP (Host Identity Protocol) WG, which 
> supplemental web page is:
> 
> http://hip.piuha.net/
> 
> Here you have a *very* brief introduction to HIP. I will also be 
> providing a brief introduction to the drafts above to the HIP community. 
> Hopefully, we will have a useful exchange of ideas between both WGs.
> 
> HIP consists of a layer between the network and the tranport layers 
> (e.g., between IP and TCP). It uses HIs (Host Identifiers) to identify 
> hosts. The idea is that, when a host changes its IP address, it keeps 
> its HI. This facilitates mobility, for instance.
> 
> HIs (Host Identifiers) are public keys, which may have different lengths 
> and formats. To resolve this, HIP defines the HITs (Host Identity Tags), 
> which are hashes of the HI. That is, a HIT is a hash of a public key. 
> They are 128 bit long, and can be used instead of IPv6 addresses in 
> socket calls. HIP also defines LSIs (Local Scope Indetifiers), that are 
> 32 bit long and can be used instead of IPv4 addresses in socket calls.
> 
> Regards,
> 
> Gonzalo
> 
> 
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> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: RE: [Sipping] updated dialog package
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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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What I was trying to say is as below:

1. UA1 and UA2 are part of an SLA group.
2. UA3 calls the group.=20
3. UA1 answers the call.
4. UA2 gets a NOTIFY with the payload only containing a dialog id, =
version number and the state of the dialog as "confirmed" so that any =
pickup can be prevented.
5. UA1 places the call on Hold.
6. UA2 receives a NOTIFY with payload that contains the dialog id; =
updated version number, the call-id; local and remote-tags so that UA2 =
is free to pick the dialog.
7. UA1 now retrieve the call. What does UA1 send in the new NOTIFY? Same =
dialog-id, updated version number but no callid, remote-tag/local-tag? =
If yes, does this mean UA2 upon receipt of this dialog state remove any =
existing callids/localtag/remote tag values it got in step 6?=20


regards,
Venkatesh

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Monday, February 23, 2004 6:55 PM
To: Venkatesh Venkataramanan
Cc: Rohan Mahy; sipping
Subject: Re: [Sipping] updated dialog package


Hi,

On Feb 23, 2004, at 5:43 AM, Venkatesh Venkataramanan wrote:

> Resending the same as my first posting somehow was empty in the=20
> mailing list !
>
> From a shared line application perspective:
>
> There is a piece missing from a implementation perspective for shared=20
> line appearances.
>
>    - Ability to signal Hold. I know there have been a few discussions=20
> in the SIPPING mailing list around this and it there was no conclusion =

> about what "hold" actually is ! However, one of the important aspects=20
> of shared lines is the ability to communicate a held call to the other =

> parties in the group. Currently the anil-sipping-bla draft uses the=20
> SDP information to help other UA's decide a dialog is transitioning to =

> hold and off hold. It would be definitely useful to have the dialog=20
> state express this state to the far end rather than having to send out =

> SDP information.

I'm sorry.  I dropped the ball on this.  I was going to propose a new=20
callee-caps style "media feature parameter" called "interactive".  I=20
think this satisfies the requirement for what folks are actually=20
interested in for "hold", but I need to send out the proposal tomorrow=20
to verify this with everyone on the list.

> For the purpose of SLA, instead of going down defining what is Hold, I =

> was wondering if we can create  a state/or an attribute in the dialog=20
> state package say "pickable" dialog to indicate the same. I've heard=20
> comments saying a UA can block relevant fields of a dialog whenever it =

> wants others to be able to pick or not to pick a dialog, however the=20
> same is not clean

why isn't this clean?

> and it breaks when say the same UA that put the dialog on Hold,=20
> retrieves the dialog. Nothing in the dialog identifiers have changed.

I don't see why this is a problem.  Also, if I'm already in a dialog,=20
the implicit filtering of the dialog package prevent me from receiving=20
notifications about state that I already know about directly.  Not=20
exactly sure what you mean.

In any case, lets take two cases, cooperating and non-cooperating UAs. =20
Cooperating UAs can use the presence or existence of dialog identifiers=20
in notifications that they receive to convey that a call is pickable by=20
them. Non-cooperating UAs that happen to learn the dialog identifiers=20
but aren't authorized, can try to pick calls all they want, but will=20
just get 403 Forbidden responses for their requests.

thanks,
-rohan


> The anil-sipping-bla-draft covers other various aspects of SLA from an =

> application implementation perspective with call flow examples.
>
> regards,
> Venkatesh
>
> 	-----Original Message-----
> 	From: sipping-admin@ietf.org on behalf of Rohan Mahy
> 	Sent: Thu 2/19/2004 9:59 AM
> 	To: sipping
> 	Cc: rohan@cisco.com
> 	Subject: [Sipping] updated dialog package
> =09
> =09
>
> 	Hi Everyone,
>
> 	I've updated the dialog packages with the changes we agreed to in
> 	Minneapolis (removing CSeq and the Route-Set).
>
> 	I've also provided a detailed example of using the dialog package for =

> a
> 	"shared-line" application.  I apologize for my tardiness here.  I was
> 	supposed to post something shortly after the meeting.  I any case, I
> 	believe this demonstrates that the current syntax is at least
> 	sufficient for the use cases we've discussed.  I would like to hear
> 	from all of you if this does the trick.
>
> 	many thanks,
> 	-rohan
>
>
> 	_______________________________________________
> 	Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> 	This list is for NEW development of the application of SIP
> 	Use sip-implementors@cs.columbia.edu for questions on current sip
> 	Use sip@ietf.org for new developments of core SIP
>




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



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Cc: Rohan Mahy <rohan@cisco.com>
From: Rohan Mahy <rohan@cisco.com>
Date: Wed, 25 Feb 2004 12:44:10 -0800
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Hi Folks,

If you are writing a draft using the xml2rfc tool, please make sure 
that the following line appears just after the line which begins <?rfc 
toc=

	<?rfc compact="yes"?>

if you don't do this, the tool will generate each top-level section on 
its own page, so you end up wasting a whole page to print for example:

	11. IANA Considerations

	   This document requires no action by IANA.

many thanks for saving trees, etc.. and making my reading list lighter 
(at least by weight).

thanks,
-rohan



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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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To: Kumiko Ono <ono.kumiko@lab.ntt.co.jp>
CC: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
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Subject: Re: [Sipping] Re: E2M security solution
References: <6AC3FA9A0545F4ono.kumiko@lab.ntt.co.jp> <7EC3FB82BA6357ono.kumiko@lab.ntt.co.jp> <403C6DA1.5020008@ericsson.com>
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Kumiko,
> 
>> I have a question about security of the MIO. Let's consider for 
>> firewall control located with a proxy. The MIO header that you propose 
>> contains parts of SDP information. When UAs encrypt SDP for end-to-end 
>> in order to conceal IP addresses and port numbers, does the MIO also 
>> contain the same information to need to use e2e encryption?
 >
> the idea is that you place in these headers only the information that 
> the proxy needs. That is, the information that the UAC wants to disclose 
> to the proxy. Right now, if the UAC chooses to disclose some information 
> to a particular proxy, it has to disclose it to every proxy in the 
> path... this may not always be desirable. This is where e2m security 
> comes into the scene.
> 
Yes, I fully agree with Gonzalo. The two drafts should probably be 
aligned so that e2m security can be used in conjunction with session 
policies.

A related question might be "m2e" security: in some scenarios, a proxy 
may not want to disclose policies to other proxies. I would assume that 
the proxy could use the keys of the recipient (UAS/UAC) to encrypt these 
policies. However, it might not have these keys handy. Would it make 
sense to use the e2m infrastructure to also *encrypt* information 
inserted by a proxy?

This of course also raises the question whether a proxy may only insert 
headers, or if it can be enabled to add bodies to a SIP message (e.g. to 
a multipart/mime body). This would probably require the definition of a 
new mechanism for adding bodies.

Thanks,

Volker

-- 
Volker Hilt                      Bell Labs / Lucent Technologies
phone: +1 732 332 6432           101 Crawfords Corner Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733

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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: sipping <sipping@ietf.org>, Cullen Jennings <fluffy@cisco.com>,
        bstucker@nortelnetworks.com
Subject: Re: [Sipping] UA indentification and HIP
References: <403B0C82.20407@ericsson.com> <403CFF0E.5070705@dynamicsoft.com>
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Jonathan,

> Thanks for the brief intro, Gonzalo. I need to read more on HIP and its 
> on my now-very-length reading list for the plane.

You can find the drafts at http://hip.piuha.net
If you want some extra slideware, drop me a private note.

> Let me anyway ask the main question I have not understood, which is how 
> address resolution is done. So, lets say I want to send a message to a 
> host, and I have obtained its HIT somehow. Presumably I need to 
> translate that to the IP for the host. How is that done? 

We have a working group and a research group. The working group focuses 
on short-term stuff, while the research group focuses on long-term 
stuff. In the WG, the idea is that you get the FQDN of the remote host 
and perform a DNS lookup. You will get a HIT and a bunch of IP addresses.

The kernel stores the mapping between the HIT and the IP addresses and 
passes the HIT to the application. The application uses the HIT to 
perform socket calls. When a socket is created, the kernel uses the 
mapping it stored before to send IP packets to one of the IP addresses 
of the remote host. That is, standard IP routing is used.

The first exchange between the host consists of a four-way handshake. 
which includes a DH key exchange. Afterwards, mobility is handled by HIP 
messages between the hosts, and traffic is protected using IPSec.

So, once you perform the initial handshake, the only situation where you 
would need to map a HIT to a new IP address would be if both hosts move 
simultaneously and their connectivity is lost.

 > If its DNS,
 > then what is the difference between HIP and just using dynamic DNS and
 > assigning a hostname to each host?

For the initial discovery and for the simultaneous move of both hosts, 
both mechanisms would be identical regarding remote IP address 
discovery. Still, HIP provides you additional features because host 
identifiers are cryptographycally generated. This is somehow similar to 
the SEND (secure neighbor discovery) approach to address ownership 
(i.e., CGAs; cryptographically generated addresses). HIP handles 
mobility as well.


That much about the WG. The research group will be working on a more 
complex rendezvous mechanism which includes, for instance, HIP to 
non-HIP scenarios. Rendezvous servers will take care of mapping between 
HITs and IP addresses... but this is still research, rather than 
engineering.

> Or, is the proposal that a HIT is an address, not a name, in which case 
> routes to it are injected into the IP routing infrastructure. I hope, 
> and assume, that this is not the case, since thats equivalent to a host 
> route for every host on the Internet... egads!

No, HITs are not routable. HIP messages use IP addresses, and once the 
HIP exchange is finished, the IPSec packets carry IP addresses as well.

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "A vWijk" <A.vWijk@viataal.nl>, <Gonzalo.Camarillo@ericsson.com>,
        <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Wed, 25 Feb 2004 22:54:12 +0100
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I think PSTN textphone gateway invocation is one area where we either need
protocol extensions, or need agreements on what current mechanisms to use.

Another is invocation of various kinds of relay services or translation
services.

A third is calling emergency services and reach positions capable of
handling the type of communication that the user prefers.



Gunnar
-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
A vWijk
Sent: Tuesday, February 24, 2004 1:22 PM
To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
gunnar.hellstrom@omnitor.se; paulej@packetizer.com; gv@trace.wisc.edu
Subject: Betr.: [Sipping] ToIP Requirements


Hello Gonzalo,
While there are device requirements in the text. Those requirements and the
explanation on the different analogue text telephone systems is included to
give the reader an understanding what kind of devices will connect with
ToIP. And the need for the text gateways. And what the different behaviour
can/may be compared to end to end IP devices.
The most important part will be the ToIP framwork and the text gateways and
how such gateways will be used and work with the ToIP framework.

There is a ToIP framework where a terminal uses streaming text as defined in
RFC2793 to connect with any other Interactive texting/real-time streaming
text device.
This can be end-to-end IP, and a direct T140/RTP stream is established with
SIP. But in other cases, a text gateway must be used to allow T.140/RTP to
connect with a non IP text telephony device.
It can be an analogue text telephone (TDD/TTY) that uses a text gateway to
be able to understand RFC2793 text. (PSTN-IP)
It can be a mobile phone running GPRS that uses the text gateway to
understand RFC2793 text. (this exist right NOW using SIP! And those gateways
will be used in mobile networks this year!)
Or a mobile text telephone using CTM via the text gateway to communicate
with RFC2793 text.
Or even one analogue text telephone connecting to another analogue text
telephone in a different country, and the PSTN's use IP to connect the call.

The question is for example, how do we use that gateway from the IP side? Is
it invisible for the IP side? Or do we need special extensions in SIP to
that the IP terminal knows it is communicating with the end-device through
the text gateway?
There may be a slightly different call setup flow when a text gateway is
included.
Or additional SIP messages indicating gateway activity. When you connect for
example to an analogue text telephone with an RFC2793 capable IP terminal
the capabilities will be less then IP to IP terminal directly (different
transmission speeds, half/full duplex etc).
Or what to do with emergency calls that are in RFC 2793 text only?

That are requirements that will be defined and worked out in this draft.
Also, we want to make everybody in SIP and SIPPING aware about those text
gateways. So that complicated matters like how to send analogue text
telephone text through a PSTN and connect with a SIP telephone using RFC2793
interactive texting. When using the text gateway, there is then no need for
RFC 2833 events at the PSTN for text telephones.

So, yes. I believe that this should be in the SIPPING WG. And it will fit in
the charter - support for hearing-/speech-impaired calling.

greetz

Arnoud



Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408

>>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 24-02-04 10:21 >>>
Hello,

the draft below provides a set of requirements for real-time
conversations using SIP.
http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-toip-01.txt

Nevertheless, I have found a lot of device requirements in the text, but
I fail to understand which particular protocol requirements we need to
fulfill with new protocol extensions.

While SIPPING is the right place to handle requirements that will lead
to new SIP protocol extensions, SIPPING may not be the right place to
deal with device requirements (e.g., text phones MUST have a display of
at least 100 characters).

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you.

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interruption, unauthorized amendment, tampering and viruses, and we only
send and receive e-mails on the basis that we are not liable for any such
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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




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



From exim@www1.ietf.org  Wed Feb 25 17:13:39 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        A vWijk
	 <A.vWijk@viataal.nl>, Gonzalo.Camarillo@ericsson.com,
        sipping@ietf.org
Cc: rrroy@att.com, barry.dingle@bigfoot.com.au, henry.sinnreich@mci.com,
        paulej@packetizer.com, gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Wed, 25 Feb 2004 17:07:33 -0500
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I'm particularly interested in the last one, emergency calling.

Given that you propose an appropriate SDP indicating support for text,
do you think there is anything else we need to do to in SIP to make this
happen?  I don't see anything.

I know that the problem of using a legacy TDD device on any kind of
adapter is as problematic as current TDD; the call will answered
as if it was a voice call, the baudot/whatever tones will be detected, 
and the operator will look for text.  
Can't think of any way to make that better.

Do note that language preference headers should be set!!!

Brian

> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> Sent: Wednesday, February 25, 2004 4:54 PM
> To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> I think PSTN textphone gateway invocation is one area where 
> we either need
> protocol extensions, or need agreements on what current 
> mechanisms to use.
> 
> Another is invocation of various kinds of relay services or 
> translation
> services.
> 
> A third is calling emergency services and reach positions capable of
> handling the type of communication that the user prefers.
> 
> 
> 
> Gunnar
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Original Message-----
> From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org]On Behalf Of
> A vWijk
> Sent: Tuesday, February 24, 2004 1:22 PM
> To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> gunnar.hellstrom@omnitor.se; paulej@packetizer.com; gv@trace.wisc.edu
> Subject: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hello Gonzalo,
> While there are device requirements in the text. Those 
> requirements and the
> explanation on the different analogue text telephone systems 
> is included to
> give the reader an understanding what kind of devices will 
> connect with
> ToIP. And the need for the text gateways. And what the 
> different behaviour
> can/may be compared to end to end IP devices.
> The most important part will be the ToIP framwork and the 
> text gateways and
> how such gateways will be used and work with the ToIP framework.
> 
> There is a ToIP framework where a terminal uses streaming 
> text as defined in
> RFC2793 to connect with any other Interactive 
> texting/real-time streaming
> text device.
> This can be end-to-end IP, and a direct T140/RTP stream is 
> established with
> SIP. But in other cases, a text gateway must be used to allow 
> T.140/RTP to
> connect with a non IP text telephony device.
> It can be an analogue text telephone (TDD/TTY) that uses a 
> text gateway to
> be able to understand RFC2793 text. (PSTN-IP)
> It can be a mobile phone running GPRS that uses the text gateway to
> understand RFC2793 text. (this exist right NOW using SIP! And 
> those gateways
> will be used in mobile networks this year!)
> Or a mobile text telephone using CTM via the text gateway to 
> communicate
> with RFC2793 text.
> Or even one analogue text telephone connecting to another 
> analogue text
> telephone in a different country, and the PSTN's use IP to 
> connect the call.
> 
> The question is for example, how do we use that gateway from 
> the IP side? Is
> it invisible for the IP side? Or do we need special 
> extensions in SIP to
> that the IP terminal knows it is communicating with the 
> end-device through
> the text gateway?
> There may be a slightly different call setup flow when a text 
> gateway is
> included.
> Or additional SIP messages indicating gateway activity. When 
> you connect for
> example to an analogue text telephone with an RFC2793 capable 
> IP terminal
> the capabilities will be less then IP to IP terminal directly 
> (different
> transmission speeds, half/full duplex etc).
> Or what to do with emergency calls that are in RFC 2793 text only?
> 
> That are requirements that will be defined and worked out in 
> this draft.
> Also, we want to make everybody in SIP and SIPPING aware 
> about those text
> gateways. So that complicated matters like how to send analogue text
> telephone text through a PSTN and connect with a SIP 
> telephone using RFC2793
> interactive texting. When using the text gateway, there is 
> then no need for
> RFC 2833 events at the PSTN for text telephones.
> 
> So, yes. I believe that this should be in the SIPPING WG. And 
> it will fit in
> the charter - support for hearing-/speech-impaired calling.
> 
> greetz
> 
> Arnoud
> 
> 
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> 24-02-04 10:21 >>>
> Hello,
> 
> the draft below provides a set of requirements for real-time
> conversations using SIP.
> http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
ip-01.txt

Nevertheless, I have found a lot of device requirements in the text, but
I fail to understand which particular protocol requirements we need to
fulfill with new protocol extensions.

While SIPPING is the right place to handle requirements that will lead
to new SIP protocol extensions, SIPPING may not be the right place to
deal with device requirements (e.g., text phones MUST have a display of
at least 100 characters).

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s).
Any unauthorized review, use, disclosure or distribution is prohibited. If
you believe this message has been sent to you in error, please notify the
sender by replying to this transmission and delete the message without
disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption,
interruption, unauthorized amendment, tampering and viruses, and we only
send and receive e-mails on the basis that we are not liable for any such
corruption, interception, amendment, tampering or viruses or any
consequences thereof.


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

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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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



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Cc: "Paul Pepper" <paul.pepper@citel.com>, Rohan Mahy <rohan@cisco.com>,
        sipping <sipping@ietf.org>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] updated dialog package
Date: Wed, 25 Feb 2004 14:38:43 -0800
To: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
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Hi,

We can't just redefine the meaning of error responses to a NOTIFY.  The 
behavior is very explicit in RFC3265 and its not package specific.  It 
means "terminate my subscription immediately".

What you seem to want to accomplish is a type of filtering.  We have 
requirements for filtering and a proposed mechanism even.  I think it 
would be useful to compare your requirements to that work.  However, I 
am usually fairly skeptical about the value of asking for filtering 
when it is easy to throw away the information you don't want.  
Obviously in the case of filtering spam, you want someone upstream to 
be able to do that, but in many other cases, sending around the 
filtering rules themselves generates more traffic than the traffic the 
filter would have removed...

thanks,
-rohan


On Feb 25, 2004, at 11:51 AM, Venkatesh Venkataramanan wrote:

> Also while we are at the topic of being able to reject NOTIFY's 
> without terminating subscriptions, another useful capability to have 
> would be for a UA to reject a NOTIFY with a specific response code so 
> that any NOTIFY's for that dialog alone can be supressed. For example 
> with SLA there may be UA's in the group that are of varying 
> capabilities. For example UA1 can have capabilities to display say 2 
> active dialogs at any point, while UA2 can have capabilities that 
> permit it to show upto 5 active dialogs. When both the UA's are 
> subscribed to dialog state, they receive dialog states of all dialogs. 
> Now, if UA1 had an ability to respond to a NOTIFY for a third active 
> dialog saying "Dont terminate my subscriptions; I want to continue 
> getting notifications for all dialogs, but supress anything for this 
> specific dialog as I have reached my limits", then it would avoid 
> un-necessary NOTIFY overheads being sent towards UA1 and it having to 
> accept it and throwing it away; keep track of what dialogs it has 
> accepted without UI, so that all new NOTIFY's towards this dialog id 
> are continued to be discarded silently ......
>
> Venkatesh
>
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf 
> Of
> Rohan Mahy
> Sent: Wednesday, February 25, 2004 9:29 AM
> To: Paul Pepper
> Cc: Rohan Mahy; sipping
> Subject: Re: [Sipping] updated dialog package
>
>
>
> On Feb 25, 2004, at 7:32 AM, Paul Pepper wrote:
>
>> Hi,
>>
>> Comments in-line.
>>
>> Thanks,
>>
>> Paul.
>>
>>> -----Original Message-----
>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>> Sent: Tuesday, February 24, 2004 3:14 AM
>>> To: Paul Pepper
>>> Cc: Rohan Mahy; sipping
>>> Subject: Re: [Sipping] updated dialog package
>>>
>>>
>>>
>>> On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:
>>>
>>>> Hi,
>>>>
>>>> I work for a company that has developed handset gateway devices that
>>>> enable legacy, digital, business-grade telephones to be driven using
>>>> SIP.  For example, this enables an existing legacy PBX user
>>>> to retain
>>>> their telephones and cabling, but to replace the PBX by a
>>>> Softswitch or
>>>> IP Centrex system.  As most business-grade telephone
>>>> systems make use
>>>> of
>>>> a shared line facility, the addition of a shared line example to the
>>>> dialog event package ID is of great interest to us, and others who
>>>> support these types of feature.
>>>>
>>>> Although incorporating a shared line example is a step
>>>> towards a common
>>>> approach to shared line implementation, I don't believe it goes far
>>>> enough.  Our current implementation of shared lines has
>>>> shown that the
>>>> following issues need to be addressed by the spec, and
>>>> should be shown
>>>> in the examples:
>>>>
>>>> o Line seize, an essential part of any shared line
>>>> implementation, is
>>>> not adequately described. When more than one user agent attempts to
>>>> seize a line at approximately the same instant, a race condition
>>>> arises.
>>>> No description of how to handle the race is given - an example
>>>> application/dialog-info+xml NOTIFY payload is provided, but
>>>> this alone
>>>> is inadequate, as message sequencing is an important part
>>>> of seizing a
>>>> line.  The Line Seize condition will be covered in the
>>>> "anil-sipping-bla-draft" that Venkatesh refers to.
>>>
>>> Line seizure is well beyond what would be appropriate to
>>> standardize in
>>> SIPPING because for many types of UAs that concept does not
>>> even exist.
>>>   I was just trying to demonstrate a non-normative
>>> description of how it
>>> could be implemented. How you judge who "wins" a seizure race
>>> condition
>>> is an implementation specific issue.  In fact, its quite likely that
>>> different behaviors will be desired by different customers.  When
>>> talking about implementing this at Cisco, we concluded that a
>>> configuration knob was needed to satisfy essentially opposite
>>> requirements from two different customers.  Each customer was equally
>>> adamant that he was "right" and the other was "wrong".  As a
>>> result, I
>>> find it unlikely that SIPPING could come to come to consensus on what
>>> "proper seizing behavior" is even if we ignored the fact that
>>> the bulk
>>> of deployed SIP UAs are software UAs which have no lines to seize.
>>>
>>
>>
>> Yes, determining which user agents should win seize races is a matter
>> of
>> policy, and implementation specific. However, the mechanism used to
>> convey whether a seize race has been won seems to be a matter for
>> standardisation (e.g. a non-200 final response to the NOTIFY 
>> requesting
>> the seize could indicate that the seize operation has failed).
>
> Now I see what you are getting at.  The particular mechanism that you
> mention is unlikely to go over favorably because it would have the side
> effect of terminating your subscription.  However, I agree it would be
> useful to have a standard way to convey something like this.  I'm not
> convinced that this belongs in the dialog package, but I'll think about
> this.
>
>
>>>> o Handling of multiple instances of a shared line has been
>>>> overlooked.
>>>> On business-grade PBX systems a line will roughly
>>>> correspond to an AOR.
>>>> However, many such systems permit multiple "instances" of a
>>>> shared line
>>>> to appear on a single endpoint. For example, a number of
>>>> instances of a
>>>> "product support" shared line might appear on all the
>>>> endpoints of the
>>>> product support group, thereby allowing multiple calls to
>>>> be taken and
>>>> passed around within the group.
>>>
>>> This has not been overlooked at all.  You can have multiple
>>> subscriptions (one per AOR), or you can subscribe to a list of AORs
>>> with the event list extension:
>>>
>>> 	
>>> http://www.ietf.org/internet-drafts/draft-ietf-simple-event-li
>>> st-04.txt
>>>
>>
>> I haven't been sufficiently clear on this point. As you point out, 
>> each
>> shared AOR will have its own subscription. Take an example AOR of
>> support@example. Although a single registration and two subscriptions
>> (one in each direction, as subscriber and notifier, if a state agent 
>> is
>> used) are used per AOR on each user agent, multiple calls to/from that
>> AOR may be active at any time. If a user agent has the shared line
>> capability then all calls to support@example will be presented on it
>> by,
>> say, illuminating one button per call. (Of course this is constrained
>> to
>> varying degrees by the user agent's display capabilities.) As each
>> button shows a call to support@example the calls must be presented on
>> those buttons in a consistent manner across all user agents on the
>> network. This will allow Harry to put a call to support@example on 
>> hold
>> and verbally tell Thelma which button to press in order to pick that
>> call up. If all the calls to support@example are presented in an
>> arbitrary manner on different user agents, then the coherency of call
>> presentation is lost. This is a problem as coherent call presentation
>> is
>> a requirement of existing PBX users, and gives the only meaningful use
>> for this feature.
>>
>> One way of achieving a consistent mapping of calls to buttons (if
>> buttons are the user interface mechanism used) is to index buttons on
>> all the user agents. That way, a call can be mapped to the specific
>> button on all user agents of an installation.
>
> I expect that any such index would be conveyed as a target URI
> parameter using a device ID of some sort.  There is agenda time
> scheduled in SIPPING to discuss Instance ID requirements with the
> following reading list:
>
> draft-stucker-sip-guid-00.txt
> draft-jennings-sipping-instance-id-00.txt
> draft-ietf-xmpp-core-22.txt (section 2.3)
>
> thanks,
> -rohan
>
>
>>>> o The mechanism used to convey that a call has been placed on hold
>>>> appears to rely upon implementation-specific parameters,
>>>> found in the
>>>> xml's <param> element. I can't see any reference to
>>>> definitions of the
>>>> values used by this element's pname and pval attributes
>>>> (pname="+activity" pval="noninteractive"), so I assume they're
>>>> non-standard.
>>>
>>> This was due to distraction on my part.  I had intended to include a
>>> discussion of this in the current draft, but somehow left it
>>> out at 2am
>>> the day of the draft deadline.  I'm terribly sorry about that.  I'll
>>> send out more about the hold issue tomorrow.
>>>
>>> thanks,
>>> -rohan
>>>
>>>> Thanks,
>>>>
>>>> Paul.
>>>>
>>>>> -----Original Message-----
>>>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>>>> Sent: Thursday, February 19, 2004 6:00 PM
>>>>> To: sipping
>>>>> Cc: rohan@cisco.com
>>>>> Subject: [Sipping] updated dialog package
>>>>>
>>>>>
>>>>> Hi Everyone,
>>>>>
>>>>> I've updated the dialog packages with the changes we agreed to in
>>>>> Minneapolis (removing CSeq and the Route-Set).
>>>>>
>>>>> I've also provided a detailed example of using the dialog
>>>>> package for a
>>>>> "shared-line" application.  I apologize for my tardiness
>>>>> here.  I was
>>>>> supposed to post something shortly after the meeting.  I
>>>>> any case, I
>>>>> believe this demonstrates that the current syntax is at least
>>>>> sufficient for the use cases we've discussed.  I would like to hear
>>>>> from all of you if this does the trick.
>>>>>
>>>>> many thanks,
>>>>> -rohan
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sipping mailing list
>>>>> https://www1.ietf.org/mailman/listinfo/sipping
>>>>> This list
>>>>> is for NEW development of the application of SIP
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sip@ietf.org for new developments of core SIP
>>>>>
>>>
>>>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>
>


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Gonzalo Camarillo wrote:
> No, HITs are not routable. HIP messages use IP addresses, and once the 
> HIP exchange is finished, the IPSec packets carry IP addresses as well

The Nifty Thing about this, as I understand it, is that it provides to 
the application an assurance of identity-binding under IPSEC that can 
today be provided only by implementing TLS in the application istelf.

That's way cool.

--
Dean


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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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Cc: Rohan Mahy <rohan@cisco.com>, sipping <sipping@ietf.org>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] updated dialog package
Date: Wed, 25 Feb 2004 14:54:25 -0800
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On Feb 25, 2004, at 12:13 PM, Venkatesh Venkataramanan wrote:

> What I was trying to say is as below:
>
> 1. UA1 and UA2 are part of an SLA group.
> 2. UA3 calls the group.
> 3. UA1 answers the call.
> 4. UA2 gets a NOTIFY with the payload only containing a dialog id, 
> version number and the state of the dialog as "confirmed" so that any 
> pickup can be prevented.

it doesn't even have to contain the dialog information.  UA2 will also 
get a CANCEL for the call it receives.

> 5. UA1 places the call on Hold.
> 6. UA2 receives a NOTIFY with payload that contains the dialog id; 
> updated version number, the call-id; local and remote-tags so that UA2 
> is free to pick the dialog.
> 7. UA1 now retrieve the call. What does UA1 send in the new NOTIFY? 
> Same dialog-id, updated version number but no callid, 
> remote-tag/local-tag? If yes, does this mean UA2 upon receipt of this 
> dialog state remove any existing callids/localtag/remote tag values it 
> got in step 6?

sure, if you'd like to filter this information from cooperating UAs.

thanks,
-rohan

>
> regards,
> Venkatesh
>
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Monday, February 23, 2004 6:55 PM
> To: Venkatesh Venkataramanan
> Cc: Rohan Mahy; sipping
> Subject: Re: [Sipping] updated dialog package
>
>
> Hi,
>
> On Feb 23, 2004, at 5:43 AM, Venkatesh Venkataramanan wrote:
>
>> Resending the same as my first posting somehow was empty in the
>> mailing list !
>>
>> From a shared line application perspective:
>>
>> There is a piece missing from a implementation perspective for shared
>> line appearances.
>>
>>    - Ability to signal Hold. I know there have been a few discussions
>> in the SIPPING mailing list around this and it there was no conclusion
>> about what "hold" actually is ! However, one of the important aspects
>> of shared lines is the ability to communicate a held call to the other
>> parties in the group. Currently the anil-sipping-bla draft uses the
>> SDP information to help other UA's decide a dialog is transitioning to
>> hold and off hold. It would be definitely useful to have the dialog
>> state express this state to the far end rather than having to send out
>> SDP information.
>
> I'm sorry.  I dropped the ball on this.  I was going to propose a new
> callee-caps style "media feature parameter" called "interactive".  I
> think this satisfies the requirement for what folks are actually
> interested in for "hold", but I need to send out the proposal tomorrow
> to verify this with everyone on the list.
>
>> For the purpose of SLA, instead of going down defining what is Hold, I
>> was wondering if we can create  a state/or an attribute in the dialog
>> state package say "pickable" dialog to indicate the same. I've heard
>> comments saying a UA can block relevant fields of a dialog whenever it
>> wants others to be able to pick or not to pick a dialog, however the
>> same is not clean
>
> why isn't this clean?
>
>> and it breaks when say the same UA that put the dialog on Hold,
>> retrieves the dialog. Nothing in the dialog identifiers have changed.
>
> I don't see why this is a problem.  Also, if I'm already in a dialog,
> the implicit filtering of the dialog package prevent me from receiving
> notifications about state that I already know about directly.  Not
> exactly sure what you mean.
>
> In any case, lets take two cases, cooperating and non-cooperating UAs.
> Cooperating UAs can use the presence or existence of dialog identifiers
> in notifications that they receive to convey that a call is pickable by
> them. Non-cooperating UAs that happen to learn the dialog identifiers
> but aren't authorized, can try to pick calls all they want, but will
> just get 403 Forbidden responses for their requests.
>
> thanks,
> -rohan
>
>
>> The anil-sipping-bla-draft covers other various aspects of SLA from an
>> application implementation perspective with call flow examples.
>>
>> regards,
>> Venkatesh
>>
>> 	-----Original Message-----
>> 	From: sipping-admin@ietf.org on behalf of Rohan Mahy
>> 	Sent: Thu 2/19/2004 9:59 AM
>> 	To: sipping
>> 	Cc: rohan@cisco.com
>> 	Subject: [Sipping] updated dialog package
>> 	
>> 	
>>
>> 	Hi Everyone,
>>
>> 	I've updated the dialog packages with the changes we agreed to in
>> 	Minneapolis (removing CSeq and the Route-Set).
>>
>> 	I've also provided a detailed example of using the dialog package for
>> a
>> 	"shared-line" application.  I apologize for my tardiness here.  I was
>> 	supposed to post something shortly after the meeting.  I any case, I
>> 	believe this demonstrates that the current syntax is at least
>> 	sufficient for the use cases we've discussed.  I would like to hear
>> 	from all of you if this does the trick.
>>
>> 	many thanks,
>> 	-rohan
>>
>>
>> 	_______________________________________________
>> 	Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> 	This list is for NEW development of the application of SIP
>> 	Use sip-implementors@cs.columbia.edu for questions on current sip
>> 	Use sip@ietf.org for new developments of core SIP
>>
>
>
>


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



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From: "Mary Barnes" <mary.barnes@nortelnetworks.com>
To: "'Volker Hilt'" <volkerh@bell-labs.com>,
        Kumiko Ono
	 <ono.kumiko@lab.ntt.co.jp>
Cc: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        sipping
	 <sipping@ietf.org>, tachimoto.shinya@lab.ntt.co.jp
Subject: RE: [Sipping] Re: E2M security solution
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Volker,

Yes, there is also a related m2e (and m2m) security problem that must be
solved for the History-Info header and is intended to apply to any
information inserted by intermediaries; right now basic requirements are
identified in:
http://www.ietf.org/internet-drafts/draft-barnes-sipping-sec-inserted-info-0
1.txt
(Note, the draft was not revved for IETF-59 as there's been little
discussion and I've not had a chance to make much progress).

And, yes, this does introduce the need for a mechanism for intermediaries to
add message bodies and to secure that information. I does seem that similar
security mechanisms could be used for both e2m and m2e. But, this is where
we need to dig through some more details. 

Also, with regards to policy, the document currently leaves whether or not
the intermediary is authorized to secure the information as something
implicitly associated with providing the service whereby it is adding the
information, but we need to decide if we want to explicitly define how
Session Policy might be used here, as well. 

Mary.

-----Original Message-----
From: Volker Hilt [mailto:volkerh@bell-labs.com] 
Sent: Wednesday, February 25, 2004 3:01 PM
To: Kumiko Ono
Cc: Gonzalo Camarillo; sipping; tachimoto.shinya@lab.ntt.co.jp
Subject: Re: [Sipping] Re: E2M security solution


Kumiko,
> 
>> I have a question about security of the MIO. Let's consider for 
>> firewall control located with a proxy. The MIO header that you 
>> propose contains parts of SDP information. When UAs encrypt SDP for 
>> end-to-end in order to conceal IP addresses and port numbers, does 
>> the MIO also contain the same information to need to use e2e 
>> encryption?
 >
> the idea is that you place in these headers only the information that 
> the proxy needs. That is, the information that the UAC wants to 
> disclose to the proxy. Right now, if the UAC chooses to disclose some 
> information to a particular proxy, it has to disclose it to every 
> proxy in the path... this may not always be desirable. This is where 
> e2m security comes into the scene.
> 
Yes, I fully agree with Gonzalo. The two drafts should probably be 
aligned so that e2m security can be used in conjunction with session 
policies.

A related question might be "m2e" security: in some scenarios, a proxy 
may not want to disclose policies to other proxies. I would assume that 
the proxy could use the keys of the recipient (UAS/UAC) to encrypt these 
policies. However, it might not have these keys handy. Would it make 
sense to use the e2m infrastructure to also *encrypt* information 
inserted by a proxy?

This of course also raises the question whether a proxy may only insert 
headers, or if it can be enabled to add bodies to a SIP message (e.g. to 
a multipart/mime body). This would probably require the definition of a 
new mechanism for adding bodies.

Thanks,

Volker

-- 
Volker Hilt                      Bell Labs / Lucent Technologies
phone: +1 732 332 6432           101 Crawfords Corner Rd
email: volkerh@bell-labs.com     Holmdel, NJ 07733

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

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charset=3Dus-ascii">
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5.5.2656.31">
<TITLE>RE: [Sipping] Re: E2M security solution</TITLE>
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<BODY>

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

<P><FONT SIZE=3D2>Yes, there is also a related m2e (and m2m) security =
problem that must be solved for the History-Info header and is intended =
to apply to any information inserted by intermediaries; right now basic =
requirements are identified in:</FONT></P>

<P><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-barnes-sipping-sec-ins=
erted-info-01.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-barnes-sippi=
ng-sec-inserted-info-01.txt</A></FONT>
<BR><FONT SIZE=3D2>(Note, the draft was not revved for IETF-59 as =
there's been little discussion and I've not had a chance to make much =
progress).</FONT></P>

<P><FONT SIZE=3D2>And, yes, this does introduce the need for a =
mechanism for intermediaries to add message bodies and to secure that =
information. I does seem that similar security mechanisms could be used =
for both e2m and m2e. But, this is where we need to dig through some =
more details. </FONT></P>

<P><FONT SIZE=3D2>Also, with regards to policy, the document currently =
leaves whether or not the intermediary is authorized to secure the =
information as something implicitly associated with providing the =
service whereby it is adding the information, but we need to decide if =
we want to explicitly define how Session Policy might be used here, as =
well. </FONT></P>

<P><FONT SIZE=3D2>Mary.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Volker Hilt [<A =
HREF=3D"mailto:volkerh@bell-labs.com">mailto:volkerh@bell-labs.com</A>] =
</FONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, February 25, 2004 3:01 PM</FONT>
<BR><FONT SIZE=3D2>To: Kumiko Ono</FONT>
<BR><FONT SIZE=3D2>Cc: Gonzalo Camarillo; sipping; =
tachimoto.shinya@lab.ntt.co.jp</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sipping] Re: E2M security =
solution</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Kumiko,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; I have a question about security of the =
MIO. Let's consider for </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; firewall control located with a proxy. The =
MIO header that you </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; propose contains parts of SDP information. =
When UAs encrypt SDP for </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; end-to-end in order to conceal IP addresses =
and port numbers, does </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; the MIO also contain the same information =
to need to use e2e </FONT>
<BR><FONT SIZE=3D2>&gt;&gt; encryption?</FONT>
<BR><FONT SIZE=3D2>&nbsp;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; the idea is that you place in these headers =
only the information that </FONT>
<BR><FONT SIZE=3D2>&gt; the proxy needs. That is, the information that =
the UAC wants to </FONT>
<BR><FONT SIZE=3D2>&gt; disclose to the proxy. Right now, if the UAC =
chooses to disclose some </FONT>
<BR><FONT SIZE=3D2>&gt; information to a particular proxy, it has to =
disclose it to every </FONT>
<BR><FONT SIZE=3D2>&gt; proxy in the path... this may not always be =
desirable. This is where </FONT>
<BR><FONT SIZE=3D2>&gt; e2m security comes into the scene.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>Yes, I fully agree with Gonzalo. The two drafts =
should probably be </FONT>
<BR><FONT SIZE=3D2>aligned so that e2m security can be used in =
conjunction with session </FONT>
<BR><FONT SIZE=3D2>policies.</FONT>
</P>

<P><FONT SIZE=3D2>A related question might be &quot;m2e&quot; security: =
in some scenarios, a proxy </FONT>
<BR><FONT SIZE=3D2>may not want to disclose policies to other proxies. =
I would assume that </FONT>
<BR><FONT SIZE=3D2>the proxy could use the keys of the recipient =
(UAS/UAC) to encrypt these </FONT>
<BR><FONT SIZE=3D2>policies. However, it might not have these keys =
handy. Would it make </FONT>
<BR><FONT SIZE=3D2>sense to use the e2m infrastructure to also =
*encrypt* information </FONT>
<BR><FONT SIZE=3D2>inserted by a proxy?</FONT>
</P>

<P><FONT SIZE=3D2>This of course also raises the question whether a =
proxy may only insert </FONT>
<BR><FONT SIZE=3D2>headers, or if it can be enabled to add bodies to a =
SIP message (e.g. to </FONT>
<BR><FONT SIZE=3D2>a multipart/mime body). This would probably require =
the definition of a </FONT>
<BR><FONT SIZE=3D2>new mechanism for adding bodies.</FONT>
</P>

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

<P><FONT SIZE=3D2>Volker</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Volker =
Hilt&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Bell Labs / =
Lucent Technologies</FONT>
<BR><FONT SIZE=3D2>phone: +1 732 332 =
6432&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 101 =
Crawfords Corner Rd</FONT>
<BR><FONT SIZE=3D2>email: volkerh@bell-labs.com&nbsp;&nbsp;&nbsp;&nbsp; =
Holmdel, NJ 07733</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>This list is for NEW development of the application =
of SIP</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip Use sip@ietf.org for new developments of core SIP</FONT>
</P>

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------_=_NextPart_001_01C3FBF8.6FF3C536--

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Cc: Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>,
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From: Rohan Mahy <rohan@cisco.com>
Date: Wed, 25 Feb 2004 17:28:13 -0800
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Hi Folks,

A single, properly-paginated, suitable for double-sided printing, PDF 
file of the entire reading list for the SIPPING WGs upcoming meeting is 
now available on the supplemental website. Links are available from the 
agenda page:

http://www.softarmor.com/sipping/meets/ietf59/agenda.html

There are actually 4 PDFs available.  You can select from PDFs for A4 
or US-Letter paper, and one-page per side or two-pages per side 
layouts.

Enjoy and see many of you in Korea.
thanks,
-rohan


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



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Rohan -

Thanks for putting this together! This will prove quite invaluable for 
the long plane ride.

-Jonathan R.

Rohan Mahy wrote:

> Hi Folks,
> 
> A single, properly-paginated, suitable for double-sided printing, PDF 
> file of the entire reading list for the SIPPING WGs upcoming meeting is 
> now available on the supplemental website. Links are available from the 
> agenda page:
> 
> http://www.softarmor.com/sipping/meets/ietf59/agenda.html
> 
> There are actually 4 PDFs available.  You can select from PDFs for A4 or 
> US-Letter paper, and one-page per side or two-pages per side layouts.
> 
> Enjoy and see many of you in Korea.
> thanks,
> -rohan
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



From exim@www1.ietf.org  Thu Feb 26 05:38:04 2004
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	Thu, 26 Feb 2004 11:33:43 +0100 (MET)
From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Dean Willis'" <dean.willis@softarmor.com>
Cc: "'James M. Polk'" <jmpolk@cisco.com>, <sipping@ietf.org>,
        "'Rohan Mahy'" <rohan@cisco.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        <hgxs@cs.columbia.edu>
Subject: AW: [Sipping] SIPPING Agenda, IETF 59
Date: Thu, 26 Feb 2004 11:35:22 +0100
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May I suggest a SIP, AAA and Billing Track?
I would assume this to be of interest as we are seeing tremendous growth
in the number of companies entering the ITSP market these days.

Regards,

	Harry


> -----Urspr=FCngliche Nachricht-----
> Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
> Im Auftrag von Henning Schulzrinne
> Gesendet: Dienstag, 24. Februar 2004 22:11
> An: Dean Willis
> Cc: James M. Polk; sipping@ietf.org; Rohan Mahy; Gonzalo=20
> Camarillo; hgxs@cs.columbia.edu
> Betreff: Re: [Sipping] SIPPING Agenda, IETF 59
>=20
>=20
> Reversal is fine with me.
>=20
> Dean Willis wrote:
>=20
> > James M. Polk wrote:
> >=20
> >> At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
> >>
> >>> Hyperlinked version at:=20
> >>> http://www.softarmor.com/sipping/meets/ietf59/agenda.html
> >>>
> >>> WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
> >>>
> >>> 1335 SIP Emergency
> >>>           Henning Schulzrinne
> >>>           draft-ietf-sipping-sos-00.txt
> >>>           draft-schulzrinne-sipping-emergency-arch-00.txt
> >>> 1345 Location Conveyance Requirements
> >>>           James Polk
> >>>           draft-ietf-sipping-location-requirements-00.txt
> >>
> >>
> >>
> >>
> >> I believe these presentations should be reversed in order=20
> because the
> >> main points of discussion in the Location Requirements ID are for=20
> >> Emergency location dealing with call routing by proxies, adding LI=20
> >> (location information) by the proxy (whether or not the UA=20
> inserted it=20
> >> already) and the security of the LI in transit. Discussing these=20
> >> points after the Emer-Arch doc doesn't seem to make much=20
> sense due to=20
> >> potential rehash while the appropriate presenter is not at=20
> the podium=20
> >> (in either direction).
> >>
> >> comments?
> >=20
> >=20
> > Works for me. Henning?
> >=20
> > --
> > Dean
> >=20
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sip@ietf.org for new developments of core SIP
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20


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



From exim@www1.ietf.org  Thu Feb 26 06:11:07 2004
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Date: Thu, 26 Feb 2004 13:06:45 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Harry Behrens <harry.behrens@snom.de>
CC: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Dean Willis'" <dean.willis@softarmor.com>,
        "'James M. Polk'" <jmpolk@cisco.com>, sipping@ietf.org,
        "'Rohan Mahy'" <rohan@cisco.com>, hgxs@cs.columbia.edu
Subject: Re: AW: [Sipping] SIPPING Agenda, IETF 59
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Harry,

do you mean that you would like to request a time slot to present your=20
draft in SIPPING?

Gonzalo

Harry Behrens wrote:

> May I suggest a SIP, AAA and Billing Track?
> I would assume this to be of interest as we are seeing tremendous growt=
h
> in the number of companies entering the ITSP market these days.
>=20
> Regards,
>=20
> 	Harry
>=20
>=20
>=20
>>-----Urspr=FCngliche Nachricht-----
>>Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
>>Im Auftrag von Henning Schulzrinne
>>Gesendet: Dienstag, 24. Februar 2004 22:11
>>An: Dean Willis
>>Cc: James M. Polk; sipping@ietf.org; Rohan Mahy; Gonzalo=20
>>Camarillo; hgxs@cs.columbia.edu
>>Betreff: Re: [Sipping] SIPPING Agenda, IETF 59
>>
>>
>>Reversal is fine with me.
>>
>>Dean Willis wrote:
>>
>>
>>>James M. Polk wrote:
>>>
>>>
>>>>At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
>>>>
>>>>
>>>>>Hyperlinked version at:=20
>>>>>http://www.softarmor.com/sipping/meets/ietf59/agenda.html
>>>>>
>>>>>WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
>>>>>
>>>>>1335 SIP Emergency
>>>>>          Henning Schulzrinne
>>>>>          draft-ietf-sipping-sos-00.txt
>>>>>          draft-schulzrinne-sipping-emergency-arch-00.txt
>>>>>1345 Location Conveyance Requirements
>>>>>          James Polk
>>>>>          draft-ietf-sipping-location-requirements-00.txt
>>>>
>>>>
>>>>
>>>>
>>>>I believe these presentations should be reversed in order=20
>>
>>because the
>>
>>>>main points of discussion in the Location Requirements ID are for=20
>>>>Emergency location dealing with call routing by proxies, adding LI=20
>>>>(location information) by the proxy (whether or not the UA=20
>>
>>inserted it=20
>>
>>>>already) and the security of the LI in transit. Discussing these=20
>>>>points after the Emer-Arch doc doesn't seem to make much=20
>>
>>sense due to=20
>>
>>>>potential rehash while the appropriate presenter is not at=20
>>
>>the podium=20
>>
>>>>(in either direction).
>>>>
>>>>comments?
>>>
>>>
>>>Works for me. Henning?
>>>
>>>--
>>>Dean
>>>
>>>_______________________________________________
>>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>This list is for NEW development of the application of SIP Use=20
>>>sip-implementors@cs.columbia.edu for questions on current sip Use=20
>>>sip@ietf.org for new developments of core SIP
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current=20
>>sip Use sip@ietf.org for new developments of core SIP
>=20
>=20

--=20
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


This communication is confidential and intended solely for the addressee(=
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d. If you believe this message has been sent to you in error, please noti=
fy the sender by replying to this transmission and delete the message wit=
hout disclosing it. Thank you.

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ion, unauthorized amendment, tampering and viruses, and we only send and =
receive e-mails on the basis that we are not liable for any such corrupti=
on, interception, amendment, tampering or viruses or any consequences the=
reof.


_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Thu, 26 Feb 2004 12:07:03 +0100
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Actually,
When using the text gateway, there is no need to detect the TDD/Analogue
text telephone tones/signals. That is the job of the text gateway. 

And for the emergency calling. We have to look closely at this in
combination with a text gateway. Possible behaviour/action. And possible
action in SIP.

Right now I cannot say what has to happen, I must study the emergency calls
in depth. But as I see it, using the text gateway, all SIP originating and
receiving emergency calls will be possible for any textphone user.

Cheers

Arnoud


Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands. 
Mobile: +31651921948
International text telephone: +31735588408

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf Of
Rosen, Brian
Sent: woensdag 25 februari 2004 23:08
To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
paulej@packetizer.com; gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements

I'm particularly interested in the last one, emergency calling.

Given that you propose an appropriate SDP indicating support for text,
do you think there is anything else we need to do to in SIP to make this
happen?  I don't see anything.

I know that the problem of using a legacy TDD device on any kind of
adapter is as problematic as current TDD; the call will answered
as if it was a voice call, the baudot/whatever tones will be detected, 
and the operator will look for text.  
Can't think of any way to make that better.

Do note that language preference headers should be set!!!

Brian

> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> Sent: Wednesday, February 25, 2004 4:54 PM
> To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> I think PSTN textphone gateway invocation is one area where 
> we either need
> protocol extensions, or need agreements on what current 
> mechanisms to use.
> 
> Another is invocation of various kinds of relay services or 
> translation
> services.
> 
> A third is calling emergency services and reach positions capable of
> handling the type of communication that the user prefers.
> 
> 
> 
> Gunnar
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Original Message-----
> From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org]On Behalf Of
> A vWijk
> Sent: Tuesday, February 24, 2004 1:22 PM
> To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> gunnar.hellstrom@omnitor.se; paulej@packetizer.com; gv@trace.wisc.edu
> Subject: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hello Gonzalo,
> While there are device requirements in the text. Those 
> requirements and the
> explanation on the different analogue text telephone systems 
> is included to
> give the reader an understanding what kind of devices will 
> connect with
> ToIP. And the need for the text gateways. And what the 
> different behaviour
> can/may be compared to end to end IP devices.
> The most important part will be the ToIP framwork and the 
> text gateways and
> how such gateways will be used and work with the ToIP framework.
> 
> There is a ToIP framework where a terminal uses streaming 
> text as defined in
> RFC2793 to connect with any other Interactive 
> texting/real-time streaming
> text device.
> This can be end-to-end IP, and a direct T140/RTP stream is 
> established with
> SIP. But in other cases, a text gateway must be used to allow 
> T.140/RTP to
> connect with a non IP text telephony device.
> It can be an analogue text telephone (TDD/TTY) that uses a 
> text gateway to
> be able to understand RFC2793 text. (PSTN-IP)
> It can be a mobile phone running GPRS that uses the text gateway to
> understand RFC2793 text. (this exist right NOW using SIP! And 
> those gateways
> will be used in mobile networks this year!)
> Or a mobile text telephone using CTM via the text gateway to 
> communicate
> with RFC2793 text.
> Or even one analogue text telephone connecting to another 
> analogue text
> telephone in a different country, and the PSTN's use IP to 
> connect the call.
> 
> The question is for example, how do we use that gateway from 
> the IP side? Is
> it invisible for the IP side? Or do we need special 
> extensions in SIP to
> that the IP terminal knows it is communicating with the 
> end-device through
> the text gateway?
> There may be a slightly different call setup flow when a text 
> gateway is
> included.
> Or additional SIP messages indicating gateway activity. When 
> you connect for
> example to an analogue text telephone with an RFC2793 capable 
> IP terminal
> the capabilities will be less then IP to IP terminal directly 
> (different
> transmission speeds, half/full duplex etc).
> Or what to do with emergency calls that are in RFC 2793 text only?
> 
> That are requirements that will be defined and worked out in 
> this draft.
> Also, we want to make everybody in SIP and SIPPING aware 
> about those text
> gateways. So that complicated matters like how to send analogue text
> telephone text through a PSTN and connect with a SIP 
> telephone using RFC2793
> interactive texting. When using the text gateway, there is 
> then no need for
> RFC 2833 events at the PSTN for text telephones.
> 
> So, yes. I believe that this should be in the SIPPING WG. And 
> it will fit in
> the charter - support for hearing-/speech-impaired calling.
> 
> greetz
> 
> Arnoud
> 
> 
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> 24-02-04 10:21 >>>
> Hello,
> 
> the draft below provides a set of requirements for real-time
> conversations using SIP.
> http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
ip-01.txt

Nevertheless, I have found a lot of device requirements in the text, but
I fail to understand which particular protocol requirements we need to
fulfill with new protocol extensions.

While SIPPING is the right place to handle requirements that will lead
to new SIP protocol extensions, SIPPING may not be the right place to
deal with device requirements (e.g., text phones MUST have a display of
at least 100 characters).

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s).
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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

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



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



From exim@www1.ietf.org  Thu Feb 26 06:26:06 2004
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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Dean Willis'" <dean.willis@softarmor.com>,
        "'James M. Polk'" <jmpolk@cisco.com>, <sipping@ietf.org>,
        "'Rohan Mahy'" <rohan@cisco.com>, <hgxs@cs.columbia.edu>
Subject: AW: AW: [Sipping] SIPPING Agenda, IETF 59
Date: Thu, 26 Feb 2004 12:23:14 +0100
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That too.
However presenting the latest relevant AAA and Billing related drafts
seemed like a good idea in general.
It appears to me to be a topic of interest and I am not presumptuous
enough to assume that my draft is the only document worthy of attention
in that field ;-)

So: if such a "track" would exist I would like to reqeust to have one
slot allocated to my draft.
If however, the notion of such a track is not agreeable I would probably
just like to request a slot.


	Harry


> -----Urspr=FCngliche Nachricht-----
> Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Gesendet: Donnerstag, 26. Februar 2004 12:07
> An: Harry Behrens
> Cc: 'Henning Schulzrinne'; 'Dean Willis'; 'James M. Polk';=20
> sipping@ietf.org; 'Rohan Mahy'; hgxs@cs.columbia.edu
> Betreff: Re: AW: [Sipping] SIPPING Agenda, IETF 59
>=20
>=20
> Harry,
>=20
> do you mean that you would like to request a time slot to=20
> present your=20
> draft in SIPPING?
>=20
> Gonzalo
>=20
> Harry Behrens wrote:
>=20
> > May I suggest a SIP, AAA and Billing Track?
> > I would assume this to be of interest as we are seeing tremendous=20
> > growth in the number of companies entering the ITSP market=20
> these days.
> >=20
> > Regards,
> >=20
> > 	Harry
> >=20
> >=20
> >=20
> >>-----Urspr=FCngliche Nachricht-----
> >>Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> >>Im Auftrag von Henning Schulzrinne
> >>Gesendet: Dienstag, 24. Februar 2004 22:11
> >>An: Dean Willis
> >>Cc: James M. Polk; sipping@ietf.org; Rohan Mahy; Gonzalo=20
> >>Camarillo; hgxs@cs.columbia.edu
> >>Betreff: Re: [Sipping] SIPPING Agenda, IETF 59
> >>
> >>
> >>Reversal is fine with me.
> >>
> >>Dean Willis wrote:
> >>
> >>
> >>>James M. Polk wrote:
> >>>
> >>>
> >>>>At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
> >>>>
> >>>>
> >>>>>Hyperlinked version at:
> >>>>>http://www.softarmor.com/sipping/meets/ietf59/agenda.html
> >>>>>
> >>>>>WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
> >>>>>
> >>>>>1335 SIP Emergency
> >>>>>          Henning Schulzrinne
> >>>>>          draft-ietf-sipping-sos-00.txt
> >>>>>          draft-schulzrinne-sipping-emergency-arch-00.txt
> >>>>>1345 Location Conveyance Requirements
> >>>>>          James Polk
> >>>>>          draft-ietf-sipping-location-requirements-00.txt
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>I believe these presentations should be reversed in order
> >>
> >>because the
> >>
> >>>>main points of discussion in the Location Requirements ID are for
> >>>>Emergency location dealing with call routing by proxies,=20
> adding LI=20
> >>>>(location information) by the proxy (whether or not the UA=20
> >>
> >>inserted it
> >>
> >>>>already) and the security of the LI in transit. Discussing these
> >>>>points after the Emer-Arch doc doesn't seem to make much=20
> >>
> >>sense due to
> >>
> >>>>potential rehash while the appropriate presenter is not at
> >>
> >>the podium
> >>
> >>>>(in either direction).
> >>>>
> >>>>comments?
> >>>
> >>>
> >>>Works for me. Henning?
> >>>
> >>>--
> >>>Dean
> >>>
> >>>_______________________________________________
> >>>Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >>>This list=20
> is for NEW development of the application of SIP Use
> >>>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>>sip@ietf.org for new developments of core SIP
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP Use=20
> >>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>sip@ietf.org for new developments of core SIP
> >=20
> >=20
>=20
> --=20
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
>=20
>=20
> This communication is confidential and intended solely for=20
> the addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is prohibited. If you believe this message has=20
> been sent to you in error, please notify the sender by=20
> replying to this transmission and delete the message without=20
> disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data=20
> corruption, interruption, unauthorized amendment, tampering=20
> and viruses, and we only send and receive e-mails on the=20
> basis that we are not liable for any such corruption,=20
> interception, amendment, tampering or viruses or any=20
> consequences thereof.
>=20


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



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From: "Stephen James" <sjames@carriustech.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'sipping'" <sipping@ietf.org>, "'Cullen Jennings'" <fluffy@cisco.com>,
        <bstucker@nortelnetworks.com>
Subject: RE: [Sipping] UA indentification and HIP
Date: Thu, 26 Feb 2004 05:52:19 -0600
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We are kind of drifing off the SIP topic, but what is the prognosis for
realizing HIP in the real network? Are we looking at something in terms
of months, years, etc. 

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Dean Willis
Sent: Wednesday, February 25, 2004 4:53 PM
To: Gonzalo Camarillo
Cc: Jonathan Rosenberg; sipping; Cullen Jennings;
bstucker@nortelnetworks.com
Subject: Re: [Sipping] UA indentification and HIP


Gonzalo Camarillo wrote:
> No, HITs are not routable. HIP messages use IP addresses, and once the
> HIP exchange is finished, the IPSec packets carry IP addresses as well

The Nifty Thing about this, as I understand it, is that it provides to 
the application an assurance of identity-binding under IPSEC that can 
today be provided only by implementing TLS in the application istelf.

That's way cool.

--
Dean


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


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



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CC: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'sipping'" <sipping@ietf.org>, "'Cullen Jennings'" <fluffy@cisco.com>,
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Subject: Re: [Sipping] UA indentification and HIP
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Hi,

> We are kind of drifing off the SIP topic, but what is the prognosis for
> realizing HIP in the real network? Are we looking at something in terms
> of months, years, etc. 

I would like to finish the standardization of the base protocol, the 
basic redezvous mechanism, the mobility and multi-homing stuff, and some 
other issues by the end of this year.

Regarding deployment, we will probably have early adopters in the 
following months... we may get a critical mass in around two years... 
and then, it may become something like SSH, or it may be completely 
abandoned, or it may be so successful that the whole Internet starts 
migrating to HIP...

The thing with HIP is that it makes a lot of things easier, but we still 
have not found the killer application that will make HIP a great success.

Gonzalo


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Date: Thu, 26 Feb 2004 14:13:01 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Alan Johnston <alan.johnston@mci.com>
CC: hisham.khartabil@nokia.com, sipping@ietf.org, oritl@microsoft.com,
        Miguel.An.Garcia@nokia.com
Subject: Re: [Sipping] Conferencing for UAs
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Alan,

I agree with you that we do not want to change the conferencing 
framework at this point in time (I already said in my previous mail that 
I understood that my comments related to this came too late). Still, I 
found the discussion useful to understand your point of view.

In any event, the topic I am interested in is the creation of different 
resources (identified by URIs) by a single request. In particular, how 
to communicate those URIs to the UAC. (I sent separate comments on 
Hisham's draft.)

I hope we can discuss these issues in Korea.

Thanks,

Gonzalo


Alan Johnston wrote:

> At 09:47 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
> 
>> Hi Alan,
>>
>> I understand that my comments come late in the game and that it might 
>> be impossible to change the conferencing framework at this point. In 
>> any event, I do not agree with your statement below:
>>
>>> No - this is the problem.  You are looking at your single narrow 
>>> application.
>>
>>
>> I believe I am actually looking at a much wider range of applications 
>> than just conferencing. So, to me, the problem is that the 
>> conferencing framework is too focused on a "single narrow 
>> application"; conferencing.
> 
> 
> Gonzalo,
> 
> If you are saying that some of your exploder services do not have a 
> conferencing component, then I agree.  Some of the exploder services 
> seem to just be a "request to fork" to a set of URIs.  Others definitely 
> have conferencing semantics.  The ones with conferencing semantics, I 
> believe, can be, and should be modeled using the conferencing framework.
> 
> The problem is that each method seems to have a different set of 
> semantics as to what exploding that method means - INVITE is handled 
> differently from SUBSCRIBE, from MESSAGE.   From a high level, exploders 
> may seem to be a consistent set of services.  However, as I delve into 
> the details and think about call flows (something  absent from any of 
> the exploder drafts), a different story emerges.
> 
> I think we need to explore exactly the services and call flows that are 
> being envisioned for exploders before proposing major changes to our 
> conferencing framework, which I believe is well understood and documented.
> 
>> In a conferencing service, there is a single resource created by the 
>> initial INVITE; a conference. That's why the server can use the 
>> Contact header field to send it back to the UAC. If you think of 
>> services that create more than one resource (e.g., an explosion and a 
>> conference), the server may not want to use the same URI for both. 
>> Moreover, conceptually, is is not a good thing to use the same URI to 
>> identify both resources.
> 
> 
> As Paul has pointed out - they are the same resource.  Additionally, it 
> is a trivial SIP routing issue to fix this - many proxies do 
> method-based request routing - that would completely solve your problem 
> of sending SUBSCRIBEs to a different UA than the initial request.
> 
>> So, my main concern is the semantics overload of the Contact header. 
>> (We abused this header when design REGISTER as well, by the way. 
>> That's why I do not want to do it again.)
>>
>> In my opinion, the Contact header is the URI where you send subsequent 
>> requests within the dialog. If the server wants to tell the client 
>> that a resource (e.g., a conference) has been created, it can use a 
>> different header to include the URI of such a resource. Note that, if 
>> you want both URIs to be the same (like in the conferencing case), you 
>> can do it.
>>
>> Contact: sip:1234@a.com;isfocus
>> Resource: sip:1234@a.com;isfocus
>>
>> An entity receiving these headers would use the URI in the Contact 
>> header to send subsequent requests *within* the dialog, and the URI in 
>> the other header to send requests *outside* that dialog (e.g., 
>> SUBSCRIBE to the conference package).
> 
> 
> No.  A GRUU should be used in the Contact, and its routing is valid 
> outside the dialog.
> 
> 
>> In brief, instead of sticking the conference URI in a Contact header 
>> and label that URI with an isfocus parameter so that the UAC figures 
>> what is going on, I find it more explicit to use a different header. 
>> Although, as I said, I understand that this comment comes late and 
>> that may not be useful at this point.
> 
> 
> I think much more work is needed before the working group can evaluate 
> the exploders work and see how and if it fits in with existing concepts 
> and approaches.
> 
> Thanks,
> Alan Johnston
> MCI
> sip:alan@sipstation.com
> 
>> Thanks,
>>
>> Gonzalo
>>
>>
>> This communication is confidential and intended solely for the 
>> addressee(s). Any unauthorized review, use, disclosure or distribution 
>> is prohibited. If you believe this message has been sent to you in 
>> error, please notify the sender by replying to this transmission and 
>> delete the message without disclosing it. Thank you.
>>
>> E-mail including attachments is susceptible to data corruption, 
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>> only send and receive e-mails on the basis that we are not liable for 
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>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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_______________________________________________
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This list is for NEW development of the application of SIP
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From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Thu, 26 Feb 2004 15:16:34 +0100
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Hello Brian,
That is also why we need to look at the emergency call system.
This text gateway is only at the PSTN side, thus another requirement for a
T140/RTP gateway for the PSTN is that when a local emergency number is
dialled (112, or 911 etc). That the gateway must be able to receive that
call (even dropping existing calls if necessary). Such a call is easy to
recognize as an emergency call. (PSTN side).
From that gateway on, it is all IP with RFC2793 interactive text. And then
it is just all existing SIP as if it was from an IP end-point.

This gives emergency centers an advantage, all calls will be T.140/RTP, so
integration in their computer systems will be easy instead being forced to
have analog textphones available for a textphone call. (or a separate text
telephone emergency center).

/Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands. 
Mobile: +31651921948
International text telephone: +31735588408


-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com] 
Sent: donderdag 26 februari 2004 15:04
To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; Gonzalo.Camarillo@ericsson.com;
sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
paulej@packetizer.com; gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements

Arnoud

Generally, you don't want to rely on very many SIP elements to complete
an emergency call.  Present systems accept TDD at all answering
points because they don't know before the calls is answered that it
is a TDD call.  I certainly don't want the user to have to do anything
other than dial his local emergency number, or complete a device
specific key sequence to complete an emergency call, and I don't
want a gateway busy to prevent an emergency call.

Brian

> -----Original Message-----
> From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> Sent: Thursday, February 26, 2004 6:07 AM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Actually,
> When using the text gateway, there is no need to detect the 
> TDD/Analogue
> text telephone tones/signals. That is the job of the text gateway. 
> 
> And for the emergency calling. We have to look closely at this in
> combination with a text gateway. Possible behaviour/action. 
> And possible
> action in SIP.
> 
> Right now I cannot say what has to happen, I must study the 
> emergency calls
> in depth. But as I see it, using the text gateway, all SIP 
> originating and
> receiving emergency calls will be possible for any textphone user.
> 
> Cheers
> 
> Arnoud
> 
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands. 
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of
> Rosen, Brian
> Sent: woensdag 25 februari 2004 23:08
> To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> I'm particularly interested in the last one, emergency calling.
> 
> Given that you propose an appropriate SDP indicating support for text,
> do you think there is anything else we need to do to in SIP 
> to make this
> happen?  I don't see anything.
> 
> I know that the problem of using a legacy TDD device on any kind of
> adapter is as problematic as current TDD; the call will answered
> as if it was a voice call, the baudot/whatever tones will be 
> detected, 
> and the operator will look for text.  
> Can't think of any way to make that better.
> 
> Do note that language preference headers should be set!!!
> 
> Brian
> 
> > -----Original Message-----
> > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > Sent: Wednesday, February 25, 2004 4:54 PM
> > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > I think PSTN textphone gateway invocation is one area where 
> > we either need
> > protocol extensions, or need agreements on what current 
> > mechanisms to use.
> > 
> > Another is invocation of various kinds of relay services or 
> > translation
> > services.
> > 
> > A third is calling emergency services and reach positions capable of
> > handling the type of communication that the user prefers.
> > 
> > 
> > 
> > Gunnar
> > -------------------------------------------
> > Gunnar Hellstrom
> > Omnitor AB
> > Renathvagen 2
> > SE 121 37 Johanneshov
> > SWEDEN
> > +46 8 556 002 03
> > Mob: +46 708 204 288
> > www.omnitor.se
> > Gunnar.Hellstrom@Omnitor.se
> > --------------------------------------------
> > 
> > 
> > -----Original Message-----
> > From: sipping-admin@ietf.org 
> > [mailto:sipping-admin@ietf.org]On Behalf Of
> > A vWijk
> > Sent: Tuesday, February 24, 2004 1:22 PM
> > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > gunnar.hellstrom@omnitor.se; paulej@packetizer.com; 
> gv@trace.wisc.edu
> > Subject: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > Hello Gonzalo,
> > While there are device requirements in the text. Those 
> > requirements and the
> > explanation on the different analogue text telephone systems 
> > is included to
> > give the reader an understanding what kind of devices will 
> > connect with
> > ToIP. And the need for the text gateways. And what the 
> > different behaviour
> > can/may be compared to end to end IP devices.
> > The most important part will be the ToIP framwork and the 
> > text gateways and
> > how such gateways will be used and work with the ToIP framework.
> > 
> > There is a ToIP framework where a terminal uses streaming 
> > text as defined in
> > RFC2793 to connect with any other Interactive 
> > texting/real-time streaming
> > text device.
> > This can be end-to-end IP, and a direct T140/RTP stream is 
> > established with
> > SIP. But in other cases, a text gateway must be used to allow 
> > T.140/RTP to
> > connect with a non IP text telephony device.
> > It can be an analogue text telephone (TDD/TTY) that uses a 
> > text gateway to
> > be able to understand RFC2793 text. (PSTN-IP)
> > It can be a mobile phone running GPRS that uses the text gateway to
> > understand RFC2793 text. (this exist right NOW using SIP! And 
> > those gateways
> > will be used in mobile networks this year!)
> > Or a mobile text telephone using CTM via the text gateway to 
> > communicate
> > with RFC2793 text.
> > Or even one analogue text telephone connecting to another 
> > analogue text
> > telephone in a different country, and the PSTN's use IP to 
> > connect the call.
> > 
> > The question is for example, how do we use that gateway from 
> > the IP side? Is
> > it invisible for the IP side? Or do we need special 
> > extensions in SIP to
> > that the IP terminal knows it is communicating with the 
> > end-device through
> > the text gateway?
> > There may be a slightly different call setup flow when a text 
> > gateway is
> > included.
> > Or additional SIP messages indicating gateway activity. When 
> > you connect for
> > example to an analogue text telephone with an RFC2793 capable 
> > IP terminal
> > the capabilities will be less then IP to IP terminal directly 
> > (different
> > transmission speeds, half/full duplex etc).
> > Or what to do with emergency calls that are in RFC 2793 text only?
> > 
> > That are requirements that will be defined and worked out in 
> > this draft.
> > Also, we want to make everybody in SIP and SIPPING aware 
> > about those text
> > gateways. So that complicated matters like how to send analogue text
> > telephone text through a PSTN and connect with a SIP 
> > telephone using RFC2793
> > interactive texting. When using the text gateway, there is 
> > then no need for
> > RFC 2833 events at the PSTN for text telephones.
> > 
> > So, yes. I believe that this should be in the SIPPING WG. And 
> > it will fit in
> > the charter - support for hearing-/speech-impaired calling.
> > 
> > greetz
> > 
> > Arnoud
> > 
> > 
> > 
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
> > 
> > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> > 24-02-04 10:21 >>>
> > Hello,
> > 
> > the draft below provides a set of requirements for real-time
> > conversations using SIP.
> > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> ip-01.txt
> 
> Nevertheless, I have found a lot of device requirements in 
> the text, but
> I fail to understand which particular protocol requirements we need to
> fulfill with new protocol extensions.
> 
> While SIPPING is the right place to handle requirements that will lead
> to new SIP protocol extensions, SIPPING may not be the right place to
> deal with device requirements (e.g., text phones MUST have a 
> display of
> at least 100 characters).
> 
> Thanks,
> 
> Gonzalo
> 
> 
> This communication is confidential and intended solely for 
> the addressee(s).
> Any unauthorized review, use, disclosure or distribution is 
> prohibited. If
> you believe this message has been sent to you in error, 
> please notify the
> sender by replying to this transmission and delete the message without
> disclosing it. Thank you.
> 
> E-mail including attachments is susceptible to data corruption,
> interruption, unauthorized amendment, tampering and viruses, 
> and we only
> send and receive e-mails on the basis that we are not liable 
> for any such
> corruption, interception, amendment, tampering or viruses or any
> consequences thereof.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 



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



From exim@www1.ietf.org  Thu Feb 26 09:22:03 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>,
        "'Gunnar Hellstrom'"
	 <gunnar.hellstrom@omnitor.se>,
        Gonzalo.Camarillo@ericsson.com, sipping@ietf.org
Cc: rrroy@att.com, barry.dingle@bigfoot.com.au, henry.sinnreich@mci.com,
        paulej@packetizer.com, gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Thu, 26 Feb 2004 09:03:49 -0500
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	autolearn=no version=2.60

Arnoud

Generally, you don't want to rely on very many SIP elements to complete
an emergency call.  Present systems accept TDD at all answering
points because they don't know before the calls is answered that it
is a TDD call.  I certainly don't want the user to have to do anything
other than dial his local emergency number, or complete a device
specific key sequence to complete an emergency call, and I don't
want a gateway busy to prevent an emergency call.

Brian

> -----Original Message-----
> From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> Sent: Thursday, February 26, 2004 6:07 AM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Actually,
> When using the text gateway, there is no need to detect the 
> TDD/Analogue
> text telephone tones/signals. That is the job of the text gateway. 
> 
> And for the emergency calling. We have to look closely at this in
> combination with a text gateway. Possible behaviour/action. 
> And possible
> action in SIP.
> 
> Right now I cannot say what has to happen, I must study the 
> emergency calls
> in depth. But as I see it, using the text gateway, all SIP 
> originating and
> receiving emergency calls will be possible for any textphone user.
> 
> Cheers
> 
> Arnoud
> 
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands. 
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> On Behalf Of
> Rosen, Brian
> Sent: woensdag 25 februari 2004 23:08
> To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> I'm particularly interested in the last one, emergency calling.
> 
> Given that you propose an appropriate SDP indicating support for text,
> do you think there is anything else we need to do to in SIP 
> to make this
> happen?  I don't see anything.
> 
> I know that the problem of using a legacy TDD device on any kind of
> adapter is as problematic as current TDD; the call will answered
> as if it was a voice call, the baudot/whatever tones will be 
> detected, 
> and the operator will look for text.  
> Can't think of any way to make that better.
> 
> Do note that language preference headers should be set!!!
> 
> Brian
> 
> > -----Original Message-----
> > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > Sent: Wednesday, February 25, 2004 4:54 PM
> > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > I think PSTN textphone gateway invocation is one area where 
> > we either need
> > protocol extensions, or need agreements on what current 
> > mechanisms to use.
> > 
> > Another is invocation of various kinds of relay services or 
> > translation
> > services.
> > 
> > A third is calling emergency services and reach positions capable of
> > handling the type of communication that the user prefers.
> > 
> > 
> > 
> > Gunnar
> > -------------------------------------------
> > Gunnar Hellstrom
> > Omnitor AB
> > Renathvagen 2
> > SE 121 37 Johanneshov
> > SWEDEN
> > +46 8 556 002 03
> > Mob: +46 708 204 288
> > www.omnitor.se
> > Gunnar.Hellstrom@Omnitor.se
> > --------------------------------------------
> > 
> > 
> > -----Original Message-----
> > From: sipping-admin@ietf.org 
> > [mailto:sipping-admin@ietf.org]On Behalf Of
> > A vWijk
> > Sent: Tuesday, February 24, 2004 1:22 PM
> > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > gunnar.hellstrom@omnitor.se; paulej@packetizer.com; 
> gv@trace.wisc.edu
> > Subject: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > Hello Gonzalo,
> > While there are device requirements in the text. Those 
> > requirements and the
> > explanation on the different analogue text telephone systems 
> > is included to
> > give the reader an understanding what kind of devices will 
> > connect with
> > ToIP. And the need for the text gateways. And what the 
> > different behaviour
> > can/may be compared to end to end IP devices.
> > The most important part will be the ToIP framwork and the 
> > text gateways and
> > how such gateways will be used and work with the ToIP framework.
> > 
> > There is a ToIP framework where a terminal uses streaming 
> > text as defined in
> > RFC2793 to connect with any other Interactive 
> > texting/real-time streaming
> > text device.
> > This can be end-to-end IP, and a direct T140/RTP stream is 
> > established with
> > SIP. But in other cases, a text gateway must be used to allow 
> > T.140/RTP to
> > connect with a non IP text telephony device.
> > It can be an analogue text telephone (TDD/TTY) that uses a 
> > text gateway to
> > be able to understand RFC2793 text. (PSTN-IP)
> > It can be a mobile phone running GPRS that uses the text gateway to
> > understand RFC2793 text. (this exist right NOW using SIP! And 
> > those gateways
> > will be used in mobile networks this year!)
> > Or a mobile text telephone using CTM via the text gateway to 
> > communicate
> > with RFC2793 text.
> > Or even one analogue text telephone connecting to another 
> > analogue text
> > telephone in a different country, and the PSTN's use IP to 
> > connect the call.
> > 
> > The question is for example, how do we use that gateway from 
> > the IP side? Is
> > it invisible for the IP side? Or do we need special 
> > extensions in SIP to
> > that the IP terminal knows it is communicating with the 
> > end-device through
> > the text gateway?
> > There may be a slightly different call setup flow when a text 
> > gateway is
> > included.
> > Or additional SIP messages indicating gateway activity. When 
> > you connect for
> > example to an analogue text telephone with an RFC2793 capable 
> > IP terminal
> > the capabilities will be less then IP to IP terminal directly 
> > (different
> > transmission speeds, half/full duplex etc).
> > Or what to do with emergency calls that are in RFC 2793 text only?
> > 
> > That are requirements that will be defined and worked out in 
> > this draft.
> > Also, we want to make everybody in SIP and SIPPING aware 
> > about those text
> > gateways. So that complicated matters like how to send analogue text
> > telephone text through a PSTN and connect with a SIP 
> > telephone using RFC2793
> > interactive texting. When using the text gateway, there is 
> > then no need for
> > RFC 2833 events at the PSTN for text telephones.
> > 
> > So, yes. I believe that this should be in the SIPPING WG. And 
> > it will fit in
> > the charter - support for hearing-/speech-impaired calling.
> > 
> > greetz
> > 
> > Arnoud
> > 
> > 
> > 
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
> > 
> > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> > 24-02-04 10:21 >>>
> > Hello,
> > 
> > the draft below provides a set of requirements for real-time
> > conversations using SIP.
> > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> ip-01.txt
> 
> Nevertheless, I have found a lot of device requirements in 
> the text, but
> I fail to understand which particular protocol requirements we need to
> fulfill with new protocol extensions.
> 
> While SIPPING is the right place to handle requirements that will lead
> to new SIP protocol extensions, SIPPING may not be the right place to
> deal with device requirements (e.g., text phones MUST have a 
> display of
> at least 100 characters).
> 
> Thanks,
> 
> Gonzalo
> 
> 
> This communication is confidential and intended solely for 
> the addressee(s).
> Any unauthorized review, use, disclosure or distribution is 
> prohibited. If
> you believe this message has been sent to you in error, 
> please notify the
> sender by replying to this transmission and delete the message without
> disclosing it. Thank you.
> 
> E-mail including attachments is susceptible to data corruption,
> interruption, unauthorized amendment, tampering and viruses, 
> and we only
> send and receive e-mails on the basis that we are not liable 
> for any such
> corruption, interception, amendment, tampering or viruses or any
> consequences thereof.
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 

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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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Message-ID: <313680C9A886D511A06000204840E1CF070B646A@whq-msgusr-02.pit.comms.marconi.com>
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>,
        "'Gunnar Hellstrom'"
	 <gunnar.hellstrom@omnitor.se>,
        Gonzalo.Camarillo@ericsson.com, sipping@ietf.org
Cc: rrroy@att.com, barry.dingle@bigfoot.com.au, henry.sinnreich@mci.com,
        paulej@packetizer.com, gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Thu, 26 Feb 2004 09:36:23 -0500
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	autolearn=no version=2.60

Arnoud

I do not believe this can work.  A TDD user dials 112 or 911.  Today, this
will be routed directly to a "selective router" (a TDM switch) which sends
the
call to the correct Emergency Response Center.  You can't insert a gateway,
because you don't know that it's a TDD call until it is answered.  To make
it work, you would have to establish a special PSTN number for emergency
TDD,
which would route to the gateway, which would then call the ERC.  I don't
believe that is practical.

Brian



> -----Original Message-----
> From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> Sent: Thursday, February 26, 2004 9:17 AM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hello Brian,
> That is also why we need to look at the emergency call system.
> This text gateway is only at the PSTN side, thus another 
> requirement for a
> T140/RTP gateway for the PSTN is that when a local emergency number is
> dialled (112, or 911 etc). That the gateway must be able to 
> receive that
> call (even dropping existing calls if necessary). Such a call 
> is easy to
> recognize as an emergency call. (PSTN side).
> From that gateway on, it is all IP with RFC2793 interactive 
> text. And then
> it is just all existing SIP as if it was from an IP end-point.
> 
> This gives emergency centers an advantage, all calls will be 
> T.140/RTP, so
> integration in their computer systems will be easy instead 
> being forced to
> have analog textphones available for a textphone call. (or a 
> separate text
> telephone emergency center).
> 
> /Arnoud
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands. 
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> 
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com] 
> Sent: donderdag 26 februari 2004 15:04
> To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> Arnoud
> 
> Generally, you don't want to rely on very many SIP elements 
> to complete
> an emergency call.  Present systems accept TDD at all answering
> points because they don't know before the calls is answered that it
> is a TDD call.  I certainly don't want the user to have to do anything
> other than dial his local emergency number, or complete a device
> specific key sequence to complete an emergency call, and I don't
> want a gateway busy to prevent an emergency call.
> 
> Brian
> 
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 6:07 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > Actually,
> > When using the text gateway, there is no need to detect the 
> > TDD/Analogue
> > text telephone tones/signals. That is the job of the text gateway. 
> > 
> > And for the emergency calling. We have to look closely at this in
> > combination with a text gateway. Possible behaviour/action. 
> > And possible
> > action in SIP.
> > 
> > Right now I cannot say what has to happen, I must study the 
> > emergency calls
> > in depth. But as I see it, using the text gateway, all SIP 
> > originating and
> > receiving emergency calls will be possible for any textphone user.
> > 
> > Cheers
> > 
> > Arnoud
> > 
> > 
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands. 
> > Mobile: +31651921948
> > International text telephone: +31735588408
> > 
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> > On Behalf Of
> > Rosen, Brian
> > Sent: woensdag 25 februari 2004 23:08
> > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > I'm particularly interested in the last one, emergency calling.
> > 
> > Given that you propose an appropriate SDP indicating 
> support for text,
> > do you think there is anything else we need to do to in SIP 
> > to make this
> > happen?  I don't see anything.
> > 
> > I know that the problem of using a legacy TDD device on any kind of
> > adapter is as problematic as current TDD; the call will answered
> > as if it was a voice call, the baudot/whatever tones will be 
> > detected, 
> > and the operator will look for text.  
> > Can't think of any way to make that better.
> > 
> > Do note that language preference headers should be set!!!
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > 
> > > 
> > > I think PSTN textphone gateway invocation is one area where 
> > > we either need
> > > protocol extensions, or need agreements on what current 
> > > mechanisms to use.
> > > 
> > > Another is invocation of various kinds of relay services or 
> > > translation
> > > services.
> > > 
> > > A third is calling emergency services and reach positions 
> capable of
> > > handling the type of communication that the user prefers.
> > > 
> > > 
> > > 
> > > Gunnar
> > > -------------------------------------------
> > > Gunnar Hellstrom
> > > Omnitor AB
> > > Renathvagen 2
> > > SE 121 37 Johanneshov
> > > SWEDEN
> > > +46 8 556 002 03
> > > Mob: +46 708 204 288
> > > www.omnitor.se
> > > Gunnar.Hellstrom@Omnitor.se
> > > --------------------------------------------
> > > 
> > > 
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org 
> > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > A vWijk
> > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > > henry.sinnreich@mci.com;
> > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com; 
> > gv@trace.wisc.edu
> > > Subject: Betr.: [Sipping] ToIP Requirements
> > > 
> > > 
> > > Hello Gonzalo,
> > > While there are device requirements in the text. Those 
> > > requirements and the
> > > explanation on the different analogue text telephone systems 
> > > is included to
> > > give the reader an understanding what kind of devices will 
> > > connect with
> > > ToIP. And the need for the text gateways. And what the 
> > > different behaviour
> > > can/may be compared to end to end IP devices.
> > > The most important part will be the ToIP framwork and the 
> > > text gateways and
> > > how such gateways will be used and work with the ToIP framework.
> > > 
> > > There is a ToIP framework where a terminal uses streaming 
> > > text as defined in
> > > RFC2793 to connect with any other Interactive 
> > > texting/real-time streaming
> > > text device.
> > > This can be end-to-end IP, and a direct T140/RTP stream is 
> > > established with
> > > SIP. But in other cases, a text gateway must be used to allow 
> > > T.140/RTP to
> > > connect with a non IP text telephony device.
> > > It can be an analogue text telephone (TDD/TTY) that uses a 
> > > text gateway to
> > > be able to understand RFC2793 text. (PSTN-IP)
> > > It can be a mobile phone running GPRS that uses the text 
> gateway to
> > > understand RFC2793 text. (this exist right NOW using SIP! And 
> > > those gateways
> > > will be used in mobile networks this year!)
> > > Or a mobile text telephone using CTM via the text gateway to 
> > > communicate
> > > with RFC2793 text.
> > > Or even one analogue text telephone connecting to another 
> > > analogue text
> > > telephone in a different country, and the PSTN's use IP to 
> > > connect the call.
> > > 
> > > The question is for example, how do we use that gateway from 
> > > the IP side? Is
> > > it invisible for the IP side? Or do we need special 
> > > extensions in SIP to
> > > that the IP terminal knows it is communicating with the 
> > > end-device through
> > > the text gateway?
> > > There may be a slightly different call setup flow when a text 
> > > gateway is
> > > included.
> > > Or additional SIP messages indicating gateway activity. When 
> > > you connect for
> > > example to an analogue text telephone with an RFC2793 capable 
> > > IP terminal
> > > the capabilities will be less then IP to IP terminal directly 
> > > (different
> > > transmission speeds, half/full duplex etc).
> > > Or what to do with emergency calls that are in RFC 2793 text only?
> > > 
> > > That are requirements that will be defined and worked out in 
> > > this draft.
> > > Also, we want to make everybody in SIP and SIPPING aware 
> > > about those text
> > > gateways. So that complicated matters like how to send 
> analogue text
> > > telephone text through a PSTN and connect with a SIP 
> > > telephone using RFC2793
> > > interactive texting. When using the text gateway, there is 
> > > then no need for
> > > RFC 2833 events at the PSTN for text telephones.
> > > 
> > > So, yes. I believe that this should be in the SIPPING WG. And 
> > > it will fit in
> > > the charter - support for hearing-/speech-impaired calling.
> > > 
> > > greetz
> > > 
> > > Arnoud
> > > 
> > > 
> > > 
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > > 
> > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> > > 24-02-04 10:21 >>>
> > > Hello,
> > > 
> > > the draft below provides a set of requirements for real-time
> > > conversations using SIP.
> > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > ip-01.txt
> > 
> > Nevertheless, I have found a lot of device requirements in 
> > the text, but
> > I fail to understand which particular protocol requirements 
> we need to
> > fulfill with new protocol extensions.
> > 
> > While SIPPING is the right place to handle requirements 
> that will lead
> > to new SIP protocol extensions, SIPPING may not be the 
> right place to
> > deal with device requirements (e.g., text phones MUST have a 
> > display of
> > at least 100 characters).
> > 
> > Thanks,
> > 
> > Gonzalo
> > 
> > 
> > This communication is confidential and intended solely for 
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> > sender by replying to this transmission and delete the 
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> > disclosing it. Thank you.
> > 
> > E-mail including attachments is susceptible to data corruption,
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> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > 
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > 
> 
> 

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



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Message-Id: <200402261607.i1QG72rb012677@smtp.eweka.nl>
From: "Arnoud van Wijk" <a.vwijk@viataal.nl>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Thu, 26 Feb 2004 16:56:48 +0100
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Hello Brian,

Yes you are right about that the PSTN cannot detect if it is a voice call or
TDD/Textphone call. They are both audio.
The PSTN would if it can detect it is a textphone call, route it to the text
gateway. And transcode the call into T.140 over RTP as decribed in RFC 2793.
This is for PSTNs which use IP to interconnect with other PSTNs or with any
SIP telephones. So it is for analoge telephony into SIP telephony here.

In the ToIP draft:
http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-toip-01.txt
A possible solution is a prefix for textphone users before dialling the
selected number.
It indeed would mean that you dial e.g. 600-112 instead 112 directly.
Is that bad?
I believe it is not.

Most textphones have the emergency number pre programmed. And when textphone
users have to use a pre-fix for any textphone call, then the same prefix
before 112 or 911 will be fine.
In the Netherlands, we have for emergency calls 800-112 instead of 112. It
works fine, only almost no deaf users call 112, since they are then mostly
in a situation where they do not have a textphone nearby!

Maybe there are other solutions that allow the PSTN to detect that the
incoming call is a textphone user and acts on it.

For example it is possible to create a database where the PSTN can check if
the incoming number is registered as a textphone.

But using a text gateway to connect PSTN textphones/TDDs to ToIP is still
the best solution, even the only solution in my opinion.
There should be no attempts to transport the text telephone tones (e.g.
Baudot) over the IP voice channel! I believe that will open Pandora's box.

I am not a PSTN expert, and we can surely look into this matter to find a
solution. Let it be prefix or any other solution that lets the PSTN
immediately route the call to the text gateway.

What do you PSTN people think? How would you tackle this problem? Your input
will be very welcome.

Greetz

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands. 
Mobile: +31651921948
International text telephone: +31735588408


-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com] 
Sent: donderdag 26 februari 2004 15:36
To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; Gonzalo.Camarillo@ericsson.com;
sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
paulej@packetizer.com; gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements

Arnoud

I do not believe this can work.  A TDD user dials 112 or 911.  Today, this
will be routed directly to a "selective router" (a TDM switch) which sends
the
call to the correct Emergency Response Center.  You can't insert a gateway,
because you don't know that it's a TDD call until it is answered.  To make
it work, you would have to establish a special PSTN number for emergency
TDD,
which would route to the gateway, which would then call the ERC.  I don't
believe that is practical.

Brian



> -----Original Message-----
> From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> Sent: Thursday, February 26, 2004 9:17 AM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hello Brian,
> That is also why we need to look at the emergency call system.
> This text gateway is only at the PSTN side, thus another 
> requirement for a
> T140/RTP gateway for the PSTN is that when a local emergency number is
> dialled (112, or 911 etc). That the gateway must be able to 
> receive that
> call (even dropping existing calls if necessary). Such a call 
> is easy to
> recognize as an emergency call. (PSTN side).
> From that gateway on, it is all IP with RFC2793 interactive 
> text. And then
> it is just all existing SIP as if it was from an IP end-point.
> 
> This gives emergency centers an advantage, all calls will be 
> T.140/RTP, so
> integration in their computer systems will be easy instead 
> being forced to
> have analog textphones available for a textphone call. (or a 
> separate text
> telephone emergency center).
> 
> /Arnoud
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands. 
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> 
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com] 
> Sent: donderdag 26 februari 2004 15:04
> To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> Arnoud
> 
> Generally, you don't want to rely on very many SIP elements 
> to complete
> an emergency call.  Present systems accept TDD at all answering
> points because they don't know before the calls is answered that it
> is a TDD call.  I certainly don't want the user to have to do anything
> other than dial his local emergency number, or complete a device
> specific key sequence to complete an emergency call, and I don't
> want a gateway busy to prevent an emergency call.
> 
> Brian
> 
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 6:07 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > 
> > Actually,
> > When using the text gateway, there is no need to detect the 
> > TDD/Analogue
> > text telephone tones/signals. That is the job of the text gateway. 
> > 
> > And for the emergency calling. We have to look closely at this in
> > combination with a text gateway. Possible behaviour/action. 
> > And possible
> > action in SIP.
> > 
> > Right now I cannot say what has to happen, I must study the 
> > emergency calls
> > in depth. But as I see it, using the text gateway, all SIP 
> > originating and
> > receiving emergency calls will be possible for any textphone user.
> > 
> > Cheers
> > 
> > Arnoud
> > 
> > 
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands. 
> > Mobile: +31651921948
> > International text telephone: +31735588408
> > 
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] 
> > On Behalf Of
> > Rosen, Brian
> > Sent: woensdag 25 februari 2004 23:08
> > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > 
> > I'm particularly interested in the last one, emergency calling.
> > 
> > Given that you propose an appropriate SDP indicating 
> support for text,
> > do you think there is anything else we need to do to in SIP 
> > to make this
> > happen?  I don't see anything.
> > 
> > I know that the problem of using a legacy TDD device on any kind of
> > adapter is as problematic as current TDD; the call will answered
> > as if it was a voice call, the baudot/whatever tones will be 
> > detected, 
> > and the operator will look for text.  
> > Can't think of any way to make that better.
> > 
> > Do note that language preference headers should be set!!!
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > 
> > > 
> > > I think PSTN textphone gateway invocation is one area where 
> > > we either need
> > > protocol extensions, or need agreements on what current 
> > > mechanisms to use.
> > > 
> > > Another is invocation of various kinds of relay services or 
> > > translation
> > > services.
> > > 
> > > A third is calling emergency services and reach positions 
> capable of
> > > handling the type of communication that the user prefers.
> > > 
> > > 
> > > 
> > > Gunnar
> > > -------------------------------------------
> > > Gunnar Hellstrom
> > > Omnitor AB
> > > Renathvagen 2
> > > SE 121 37 Johanneshov
> > > SWEDEN
> > > +46 8 556 002 03
> > > Mob: +46 708 204 288
> > > www.omnitor.se
> > > Gunnar.Hellstrom@Omnitor.se
> > > --------------------------------------------
> > > 
> > > 
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org 
> > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > A vWijk
> > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> > > henry.sinnreich@mci.com;
> > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com; 
> > gv@trace.wisc.edu
> > > Subject: Betr.: [Sipping] ToIP Requirements
> > > 
> > > 
> > > Hello Gonzalo,
> > > While there are device requirements in the text. Those 
> > > requirements and the
> > > explanation on the different analogue text telephone systems 
> > > is included to
> > > give the reader an understanding what kind of devices will 
> > > connect with
> > > ToIP. And the need for the text gateways. And what the 
> > > different behaviour
> > > can/may be compared to end to end IP devices.
> > > The most important part will be the ToIP framwork and the 
> > > text gateways and
> > > how such gateways will be used and work with the ToIP framework.
> > > 
> > > There is a ToIP framework where a terminal uses streaming 
> > > text as defined in
> > > RFC2793 to connect with any other Interactive 
> > > texting/real-time streaming
> > > text device.
> > > This can be end-to-end IP, and a direct T140/RTP stream is 
> > > established with
> > > SIP. But in other cases, a text gateway must be used to allow 
> > > T.140/RTP to
> > > connect with a non IP text telephony device.
> > > It can be an analogue text telephone (TDD/TTY) that uses a 
> > > text gateway to
> > > be able to understand RFC2793 text. (PSTN-IP)
> > > It can be a mobile phone running GPRS that uses the text 
> gateway to
> > > understand RFC2793 text. (this exist right NOW using SIP! And 
> > > those gateways
> > > will be used in mobile networks this year!)
> > > Or a mobile text telephone using CTM via the text gateway to 
> > > communicate
> > > with RFC2793 text.
> > > Or even one analogue text telephone connecting to another 
> > > analogue text
> > > telephone in a different country, and the PSTN's use IP to 
> > > connect the call.
> > > 
> > > The question is for example, how do we use that gateway from 
> > > the IP side? Is
> > > it invisible for the IP side? Or do we need special 
> > > extensions in SIP to
> > > that the IP terminal knows it is communicating with the 
> > > end-device through
> > > the text gateway?
> > > There may be a slightly different call setup flow when a text 
> > > gateway is
> > > included.
> > > Or additional SIP messages indicating gateway activity. When 
> > > you connect for
> > > example to an analogue text telephone with an RFC2793 capable 
> > > IP terminal
> > > the capabilities will be less then IP to IP terminal directly 
> > > (different
> > > transmission speeds, half/full duplex etc).
> > > Or what to do with emergency calls that are in RFC 2793 text only?
> > > 
> > > That are requirements that will be defined and worked out in 
> > > this draft.
> > > Also, we want to make everybody in SIP and SIPPING aware 
> > > about those text
> > > gateways. So that complicated matters like how to send 
> analogue text
> > > telephone text through a PSTN and connect with a SIP 
> > > telephone using RFC2793
> > > interactive texting. When using the text gateway, there is 
> > > then no need for
> > > RFC 2833 events at the PSTN for text telephones.
> > > 
> > > So, yes. I believe that this should be in the SIPPING WG. And 
> > > it will fit in
> > > the charter - support for hearing-/speech-impaired calling.
> > > 
> > > greetz
> > > 
> > > Arnoud
> > > 
> > > 
> > > 
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > > 
> > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com> 
> > > 24-02-04 10:21 >>>
> > > Hello,
> > > 
> > > the draft below provides a set of requirements for real-time
> > > conversations using SIP.
> > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > ip-01.txt
> > 
> > Nevertheless, I have found a lot of device requirements in 
> > the text, but
> > I fail to understand which particular protocol requirements 
> we need to
> > fulfill with new protocol extensions.
> > 
> > While SIPPING is the right place to handle requirements 
> that will lead
> > to new SIP protocol extensions, SIPPING may not be the 
> right place to
> > deal with device requirements (e.g., text phones MUST have a 
> > display of
> > at least 100 characters).
> > 
> > Thanks,
> > 
> > Gonzalo
> > 
> > 
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> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
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> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > 
> > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> > 
> > 
> 
> 



_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Venkatesh Venkataramanan" <Venkatesh.Venkataramanan@sylantro.com>
To: "Rohan Mahy" <rohan@cisco.com>
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Hi,

Filtering as a mechanism would work well if the phone is receiving the =
NOTIFY's from one single entity. In applications based on dialog state =
where the UA could be receiving NOTIFY's from various entities, this =
would amount to the UA having to establish filters at each such entity =
that is providing the UA notifications which like you rightly comment =
would amount to more filtering install traffic than any meaningful =
exchange of information. Definitely like I said in my earlier email and =
your comment, an end UA can simply discard NOTIFY's once it reaches it =
limits, but limiting NOTIFY's at the source if possible would be nice to =
have feature.=20

Also, what I am requesting for is not radically redefining the error =
response for a NOTIFY. The 4xx response is STILL terminating =
subscriptions, but with a narrower scope (just subscriptions to the =
dialog that was presented to it in the payload of the NOTIFY).

thanks
Venkatesh

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Wednesday, February 25, 2004 2:39 PM
To: Venkatesh Venkataramanan
Cc: Paul Pepper; Rohan Mahy; sipping
Subject: Re: [Sipping] updated dialog package


Hi,

We can't just redefine the meaning of error responses to a NOTIFY.  The=20
behavior is very explicit in RFC3265 and its not package specific.  It=20
means "terminate my subscription immediately".

What you seem to want to accomplish is a type of filtering.  We have=20
requirements for filtering and a proposed mechanism even.  I think it=20
would be useful to compare your requirements to that work.  However, I=20
am usually fairly skeptical about the value of asking for filtering=20
when it is easy to throw away the information you don't want. =20
Obviously in the case of filtering spam, you want someone upstream to=20
be able to do that, but in many other cases, sending around the=20
filtering rules themselves generates more traffic than the traffic the=20
filter would have removed...

thanks,
-rohan


On Feb 25, 2004, at 11:51 AM, Venkatesh Venkataramanan wrote:

> Also while we are at the topic of being able to reject NOTIFY's=20
> without terminating subscriptions, another useful capability to have=20
> would be for a UA to reject a NOTIFY with a specific response code so=20
> that any NOTIFY's for that dialog alone can be supressed. For example=20
> with SLA there may be UA's in the group that are of varying=20
> capabilities. For example UA1 can have capabilities to display say 2=20
> active dialogs at any point, while UA2 can have capabilities that=20
> permit it to show upto 5 active dialogs. When both the UA's are=20
> subscribed to dialog state, they receive dialog states of all dialogs. =

> Now, if UA1 had an ability to respond to a NOTIFY for a third active=20
> dialog saying "Dont terminate my subscriptions; I want to continue=20
> getting notifications for all dialogs, but supress anything for this=20
> specific dialog as I have reached my limits", then it would avoid=20
> un-necessary NOTIFY overheads being sent towards UA1 and it having to=20
> accept it and throwing it away; keep track of what dialogs it has=20
> accepted without UI, so that all new NOTIFY's towards this dialog id=20
> are continued to be discarded silently ......
>
> Venkatesh
>
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf=20
> Of
> Rohan Mahy
> Sent: Wednesday, February 25, 2004 9:29 AM
> To: Paul Pepper
> Cc: Rohan Mahy; sipping
> Subject: Re: [Sipping] updated dialog package
>
>
>
> On Feb 25, 2004, at 7:32 AM, Paul Pepper wrote:
>
>> Hi,
>>
>> Comments in-line.
>>
>> Thanks,
>>
>> Paul.
>>
>>> -----Original Message-----
>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>> Sent: Tuesday, February 24, 2004 3:14 AM
>>> To: Paul Pepper
>>> Cc: Rohan Mahy; sipping
>>> Subject: Re: [Sipping] updated dialog package
>>>
>>>
>>>
>>> On Feb 23, 2004, at 9:40 AM, Paul Pepper wrote:
>>>
>>>> Hi,
>>>>
>>>> I work for a company that has developed handset gateway devices =
that
>>>> enable legacy, digital, business-grade telephones to be driven =
using
>>>> SIP.  For example, this enables an existing legacy PBX user
>>>> to retain
>>>> their telephones and cabling, but to replace the PBX by a
>>>> Softswitch or
>>>> IP Centrex system.  As most business-grade telephone
>>>> systems make use
>>>> of
>>>> a shared line facility, the addition of a shared line example to =
the
>>>> dialog event package ID is of great interest to us, and others who
>>>> support these types of feature.
>>>>
>>>> Although incorporating a shared line example is a step
>>>> towards a common
>>>> approach to shared line implementation, I don't believe it goes far
>>>> enough.  Our current implementation of shared lines has
>>>> shown that the
>>>> following issues need to be addressed by the spec, and
>>>> should be shown
>>>> in the examples:
>>>>
>>>> o Line seize, an essential part of any shared line
>>>> implementation, is
>>>> not adequately described. When more than one user agent attempts to
>>>> seize a line at approximately the same instant, a race condition
>>>> arises.
>>>> No description of how to handle the race is given - an example
>>>> application/dialog-info+xml NOTIFY payload is provided, but
>>>> this alone
>>>> is inadequate, as message sequencing is an important part
>>>> of seizing a
>>>> line.  The Line Seize condition will be covered in the
>>>> "anil-sipping-bla-draft" that Venkatesh refers to.
>>>
>>> Line seizure is well beyond what would be appropriate to
>>> standardize in
>>> SIPPING because for many types of UAs that concept does not
>>> even exist.
>>>   I was just trying to demonstrate a non-normative
>>> description of how it
>>> could be implemented. How you judge who "wins" a seizure race
>>> condition
>>> is an implementation specific issue.  In fact, its quite likely that
>>> different behaviors will be desired by different customers.  When
>>> talking about implementing this at Cisco, we concluded that a
>>> configuration knob was needed to satisfy essentially opposite
>>> requirements from two different customers.  Each customer was =
equally
>>> adamant that he was "right" and the other was "wrong".  As a
>>> result, I
>>> find it unlikely that SIPPING could come to come to consensus on =
what
>>> "proper seizing behavior" is even if we ignored the fact that
>>> the bulk
>>> of deployed SIP UAs are software UAs which have no lines to seize.
>>>
>>
>>
>> Yes, determining which user agents should win seize races is a matter
>> of
>> policy, and implementation specific. However, the mechanism used to
>> convey whether a seize race has been won seems to be a matter for
>> standardisation (e.g. a non-200 final response to the NOTIFY=20
>> requesting
>> the seize could indicate that the seize operation has failed).
>
> Now I see what you are getting at.  The particular mechanism that you
> mention is unlikely to go over favorably because it would have the =
side
> effect of terminating your subscription.  However, I agree it would be
> useful to have a standard way to convey something like this.  I'm not
> convinced that this belongs in the dialog package, but I'll think =
about
> this.
>
>
>>>> o Handling of multiple instances of a shared line has been
>>>> overlooked.
>>>> On business-grade PBX systems a line will roughly
>>>> correspond to an AOR.
>>>> However, many such systems permit multiple "instances" of a
>>>> shared line
>>>> to appear on a single endpoint. For example, a number of
>>>> instances of a
>>>> "product support" shared line might appear on all the
>>>> endpoints of the
>>>> product support group, thereby allowing multiple calls to
>>>> be taken and
>>>> passed around within the group.
>>>
>>> This has not been overlooked at all.  You can have multiple
>>> subscriptions (one per AOR), or you can subscribe to a list of AORs
>>> with the event list extension:
>>>
>>> =09
>>> http://www.ietf.org/internet-drafts/draft-ietf-simple-event-li
>>> st-04.txt
>>>
>>
>> I haven't been sufficiently clear on this point. As you point out,=20
>> each
>> shared AOR will have its own subscription. Take an example AOR of
>> support@example. Although a single registration and two subscriptions
>> (one in each direction, as subscriber and notifier, if a state agent=20
>> is
>> used) are used per AOR on each user agent, multiple calls to/from =
that
>> AOR may be active at any time. If a user agent has the shared line
>> capability then all calls to support@example will be presented on it
>> by,
>> say, illuminating one button per call. (Of course this is constrained
>> to
>> varying degrees by the user agent's display capabilities.) As each
>> button shows a call to support@example the calls must be presented on
>> those buttons in a consistent manner across all user agents on the
>> network. This will allow Harry to put a call to support@example on=20
>> hold
>> and verbally tell Thelma which button to press in order to pick that
>> call up. If all the calls to support@example are presented in an
>> arbitrary manner on different user agents, then the coherency of call
>> presentation is lost. This is a problem as coherent call presentation
>> is
>> a requirement of existing PBX users, and gives the only meaningful =
use
>> for this feature.
>>
>> One way of achieving a consistent mapping of calls to buttons (if
>> buttons are the user interface mechanism used) is to index buttons on
>> all the user agents. That way, a call can be mapped to the specific
>> button on all user agents of an installation.
>
> I expect that any such index would be conveyed as a target URI
> parameter using a device ID of some sort.  There is agenda time
> scheduled in SIPPING to discuss Instance ID requirements with the
> following reading list:
>
> draft-stucker-sip-guid-00.txt
> draft-jennings-sipping-instance-id-00.txt
> draft-ietf-xmpp-core-22.txt (section 2.3)
>
> thanks,
> -rohan
>
>
>>>> o The mechanism used to convey that a call has been placed on hold
>>>> appears to rely upon implementation-specific parameters,
>>>> found in the
>>>> xml's <param> element. I can't see any reference to
>>>> definitions of the
>>>> values used by this element's pname and pval attributes
>>>> (pname=3D"+activity" pval=3D"noninteractive"), so I assume they're
>>>> non-standard.
>>>
>>> This was due to distraction on my part.  I had intended to include a
>>> discussion of this in the current draft, but somehow left it
>>> out at 2am
>>> the day of the draft deadline.  I'm terribly sorry about that.  I'll
>>> send out more about the hold issue tomorrow.
>>>
>>> thanks,
>>> -rohan
>>>
>>>> Thanks,
>>>>
>>>> Paul.
>>>>
>>>>> -----Original Message-----
>>>>> From: Rohan Mahy [mailto:rohan@cisco.com]
>>>>> Sent: Thursday, February 19, 2004 6:00 PM
>>>>> To: sipping
>>>>> Cc: rohan@cisco.com
>>>>> Subject: [Sipping] updated dialog package
>>>>>
>>>>>
>>>>> Hi Everyone,
>>>>>
>>>>> I've updated the dialog packages with the changes we agreed to in
>>>>> Minneapolis (removing CSeq and the Route-Set).
>>>>>
>>>>> I've also provided a detailed example of using the dialog
>>>>> package for a
>>>>> "shared-line" application.  I apologize for my tardiness
>>>>> here.  I was
>>>>> supposed to post something shortly after the meeting.  I
>>>>> any case, I
>>>>> believe this demonstrates that the current syntax is at least
>>>>> sufficient for the use cases we've discussed.  I would like to =
hear
>>>>> from all of you if this does the trick.
>>>>>
>>>>> many thanks,
>>>>> -rohan
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sipping mailing list
>>>>> https://www1.ietf.org/mailman/listinfo/sipping
>>>>> This list
>>>>> is for NEW development of the application of SIP
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sip@ietf.org for new developments of core SIP
>>>>>
>>>
>>>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>
>




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



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Date: Thu, 26 Feb 2004 13:04:56 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conferencing for UAs
In-reply-to: <403DE2CD.1060708@ericsson.com>
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At 02:13 PM 2/26/2004 +0200, Gonzalo Camarillo wrote:
>Alan,
>
>I agree with you that we do not want to change the conferencing framework 
>at this point in time (I already said in my previous mail that I 
>understood that my comments related to this came too late). Still, I found 
>the discussion useful to understand your point of view.

Gonzalo,

Glad to hear that.  I'd be happy to think about how to extend the 
conferencing work to cover the exploder scenarios you are proposing with 
MESSAGE, INVITE, and even REFER.    With the current drafts, I have a hard 
time understanding how to do this, but I look forward to 
clarifications.   The SUBSCRIBE exploder does not seem to have any 
conferencing semantics, but seems to be a SIMPLE application since it is 
tightly coupled to the presence application.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com


>In any event, the topic I am interested in is the creation of different 
>resources (identified by URIs) by a single request. In particular, how to 
>communicate those URIs to the UAC. (I sent separate comments on Hisham's 
>draft.)
>
>I hope we can discuss these issues in Korea.
>
>Thanks,
>
>Gonzalo
>
>
>Alan Johnston wrote:
>
>>At 09:47 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
>>
>>>Hi Alan,
>>>
>>>I understand that my comments come late in the game and that it might be 
>>>impossible to change the conferencing framework at this point. In any 
>>>event, I do not agree with your statement below:
>>>
>>>>No - this is the problem.  You are looking at your single narrow 
>>>>application.
>>>
>>>
>>>I believe I am actually looking at a much wider range of applications 
>>>than just conferencing. So, to me, the problem is that the conferencing 
>>>framework is too focused on a "single narrow application"; conferencing.
>>
>>Gonzalo,
>>If you are saying that some of your exploder services do not have a 
>>conferencing component, then I agree.  Some of the exploder services seem 
>>to just be a "request to fork" to a set of URIs.  Others definitely have 
>>conferencing semantics.  The ones with conferencing semantics, I believe, 
>>can be, and should be modeled using the conferencing framework.
>>The problem is that each method seems to have a different set of 
>>semantics as to what exploding that method means - INVITE is handled 
>>differently from SUBSCRIBE, from MESSAGE.   From a high level, exploders 
>>may seem to be a consistent set of services.  However, as I delve into 
>>the details and think about call flows (something  absent from any of the 
>>exploder drafts), a different story emerges.
>>I think we need to explore exactly the services and call flows that are 
>>being envisioned for exploders before proposing major changes to our 
>>conferencing framework, which I believe is well understood and documented.
>>
>>>In a conferencing service, there is a single resource created by the 
>>>initial INVITE; a conference. That's why the server can use the Contact 
>>>header field to send it back to the UAC. If you think of services that 
>>>create more than one resource (e.g., an explosion and a conference), the 
>>>server may not want to use the same URI for both. Moreover, 
>>>conceptually, is is not a good thing to use the same URI to identify 
>>>both resources.
>>
>>As Paul has pointed out - they are the same resource.  Additionally, it 
>>is a trivial SIP routing issue to fix this - many proxies do method-based 
>>request routing - that would completely solve your problem of sending 
>>SUBSCRIBEs to a different UA than the initial request.
>>
>>>So, my main concern is the semantics overload of the Contact header. (We 
>>>abused this header when design REGISTER as well, by the way. That's why 
>>>I do not want to do it again.)
>>>
>>>In my opinion, the Contact header is the URI where you send subsequent 
>>>requests within the dialog. If the server wants to tell the client that 
>>>a resource (e.g., a conference) has been created, it can use a different 
>>>header to include the URI of such a resource. Note that, if you want 
>>>both URIs to be the same (like in the conferencing case), you can do it.
>>>
>>>Contact: sip:1234@a.com;isfocus
>>>Resource: sip:1234@a.com;isfocus
>>>
>>>An entity receiving these headers would use the URI in the Contact 
>>>header to send subsequent requests *within* the dialog, and the URI in 
>>>the other header to send requests *outside* that dialog (e.g., SUBSCRIBE 
>>>to the conference package).
>>
>>No.  A GRUU should be used in the Contact, and its routing is valid 
>>outside the dialog.
>>
>>>In brief, instead of sticking the conference URI in a Contact header and 
>>>label that URI with an isfocus parameter so that the UAC figures what is 
>>>going on, I find it more explicit to use a different header. Although, 
>>>as I said, I understand that this comment comes late and that may not be 
>>>useful at this point.
>>
>>I think much more work is needed before the working group can evaluate 
>>the exploders work and see how and if it fits in with existing concepts 
>>and approaches.
>>Thanks,
>>Alan Johnston
>>MCI
>>sip:alan@sipstation.com
>>
>>>Thanks,
>>>
>>>Gonzalo
>>>
>>>
>>>This communication is confidential and intended solely for the 
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>>>is prohibited. If you believe this message has been sent to you in 
>>>error, please notify the sender by replying to this transmission and 
>>>delete the message without disclosing it. Thank you.
>>>
>>>E-mail including attachments is susceptible to data corruption, 
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>>>
>>>
>>>_______________________________________________
>>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>This list is for NEW development of the application of SIP
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sip@ietf.org for new developments of core SIP
>
>--
>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>Telecom R&D               Fax   :  +358  9 299 30 52
>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>Finland                   http://www.hut.fi/~gonzalo
>
>
>This communication is confidential and intended solely for the 
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>please notify the sender by replying to this transmission and delete the 
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>
>E-mail including attachments is susceptible to data corruption, 
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Subject: RE: [Fwd: RE: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt]
Date: Thu, 26 Feb 2004 17:45:14 -0500
Message-ID: <4A3384433CE2AB46A63468CB207E209DB1A1B8@zoe.office.snowshore.com>
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Thread-Index: AcP66oS0Zw8ftexMSIakj2wwJ/uKQgArUIdw
From: "Eric Burger" <eburger@snowshore.com>
To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>,
        <jdrosen@dynamicsoft.com>
Cc: <sipping@ietf.org>
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Not registering does not stand in the way of using gruu: =
draft-ietf-sip-gruu-01 offers that gruu generation is pure magic.  It is =
*suggested* to be done at a registrar to ensure routability, uniqueness, =
etc.  However, "other means" work just as well (just not as likely to =
work globally).  The gateway can generate its own gruu without =
registering.

That said, a typical deployment scenario for KPML is signaled digit =
transport between media gateways/controllers and proxies in private =
networks and IP Phones and IP PBXen in private networks.  In these =
scenarios, gruu doesn't buy you anything.  Moreover, it would clearly be =
"easier" to just make the request on the INVITE dialog, as that is =
supposed to go to the right place by definition.

So...

What direction should we go with KPML? =20

Option 1:
"You MUST use gruu (even if you fake gruu generation) and you MUST =
create independent SUBSCRIBE dialogs for the subscription."  If your UA =
doesn't support gruu, you can't use KPML.  If you are in the gateway or =
IP PBX scenario, feel free to cheat at your own risk.  My guess is IP =
PBX guys will make this optimization as a proprietary (not necessarily =
interoperable) extension.

Option 2:
"You SHOULD use gruu and SHOULD create independent SUBSCRIBE dialogs for =
the subscription."  KPML keeps the documentation for how to do the =
subscription on the INVITE dialog.  KPML documents the evils and dangers =
of issuing subscriptions on INVITE dialogs.  The problem here is there =
is no protocol way of knowing for certain that reusing the INVITE dialog =
will work.  You only know from your network topology whether it is a =
good idea or not.  The good news is for the scenarios described above, =
it will work.


> -----Original Message-----
> From: Christer Holmberg (JO/LMF)=20
> [mailto:christer.holmberg@ericsson.com]
> Sent: Tuesday, February 24, 2004 10:26 AM
> To: Eric Burger; 'jdrosen@dynamicsoft.com'; 'sipping@ietf.org'
> Subject: [Fwd: RE: [Sipping] I-D=20
> ACTION:draft-ietf-sipping-kpml-02.txt]
>=20
>=20
>=20
> Hi,
>=20
> I guess this is a clarification question on kpml, as well as=20
> on gruu (and functions where gruu is used) in general.
>=20
> If you have a node that doesn't register, but eg wants to use=20
> kpml (a gateway, for example), could it generate the gruu on=20
> its own, or is it "mandated" to use the register mechanism?=20
> If the register mechanism is not used, are there any issues=20
> which could come up?
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
>=20
>=20
>=20
>=20
>=20
> > -------- Original Message --------
> > Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
> > Date: Mon, 23 Feb 2004 17:34:22 -0500
> > From: Eric Burger <eburger@snowshore.com>
> > To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, <sipping@ietf.org>
> >=20
> > Yep, yep, yep, we can (and will) use GRUU.
> >=20
> > > -----Original Message-----
> > > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > > Sent: Thursday, February 05, 2004 2:46 AM
> > > To: sipping@ietf.org
> > > Subject: Re: [Sipping] I-D ACTION:draft-ietf-sipping-kpml-02.txt
> > >=20
> > >=20
> > > Eric,
> > >=20
> > > The draft talks about sending the subscribe requests on the=20
> > > same dialog=20
> > > as the INVITE. For many reasons that have been discussed ad=20
> > > nauseam on=20
> > > the list, dialog sharing is a bad idea. GRUU provides a=20
> > solution that=20
> > > does not require dialog sharing. As I recall, there was=20
> > discussion on=20
> > > this point in the last ietf, and this seemed to be the=20
> > > preferred direction.
> > >=20
> > > Is there any reason we cant change this to use gruu's rather=20
> > > than dialog=20
> > > sharing?
> > >=20
> > > Also, you are missing the IANA registration for the kpml=20
> > > event package.
> > >=20
> > > Thanks,
> > > Jonathan R.
> > >=20
> > > Internet-Drafts@ietf.org wrote:
> > >=20
> > > > A New Internet-Draft is available from the on-line=20
> > > Internet-Drafts directories.
> > > > This draft is a work item of the Session Initiation=20
> > > Proposal Investigation Working Group of the IETF.
> > > >=20
> > > > 	Title		: Keypad Stimulus Protocol (KPML)
> > > > 	Author(s)	: E. Burger
> > > > 	Filename	: draft-ietf-sipping-kpml-02.txt
> > > > 	Pages		: 42
> > > > 	Date		: 2004-2-4
> > > > =09
> > > > The Key Press Stimulus Protocol uses the SIP SUBSCRIBE/NOTIFY
> > > >    mechanism and Keypad Markup Language (KPML) to provide=20
> > > instructions
> > > >    to SIP User Agents for the reporting of user key presses.
> > > >=20
> > > > A URL for this Internet-Draft is:
> > > >=20
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-02.txt
> > > >=20
> > > > To remove yourself from the IETF Announcement list, send a=20
> > > message to=20
> > > > ietf-announce-request with the word unsubscribe in the body=20
> > > of the message.
> > > >=20
> > > > Internet-Drafts are also available by anonymous FTP. Login=20
> > > with the username
> > > > "anonymous" and a password of your e-mail address. After=20
> > logging in,
> > > > type "cd internet-drafts" and then
> > > > 	"get draft-ietf-sipping-kpml-02.txt".
> > > >=20
> > > > A list of Internet-Drafts directories can be found in
> > > > http://www.ietf.org/shadow.html=20
> > > > or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > >=20
> > > >=20
> > > > Internet-Drafts can also be obtained by e-mail.
> > > >=20
> > > > Send a message to:
> > > > 	mailserv@ietf.org.
> > > > In the body type:
> > > > 	"FILE /internet-drafts/draft-ietf-sipping-kpml-02.txt".
> > > > =09
> > > > NOTE:	The mail server at ietf.org can return the document in
> > > > 	MIME-encoded form by using the "mpack" utility.=20
>  To use this
> > > > 	feature, insert the command "ENCODING mime"=20
> before the "FILE"
> > > > 	command.  To decode the response(s), you will=20
> need "munpack" or
> > > > 	a MIME-compliant mail reader.  Different MIME-compliant=20
> > > mail readers
> > > > 	exhibit different behavior, especially when dealing with
> > > > 	"multipart" MIME messages (i.e. documents which=20
> have been split
> > > > 	up into multiple messages), so check your local=20
> documentation on
> > > > 	how to manipulate these messages.
> > > > 	=09
> > > > 	=09
> > > > Below is the data which will enable a MIME compliant mail reader
> > > > implementation to automatically retrieve the ASCII=20
> version of the
> > > > Internet-Draft.
> > > >=20
> > >=20
> > > --=20
> > > Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> > > Chief Technology Officer                    Parsippany, NJ=20
> > 07054-2711
> > > dynamicsoft
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > >=20
> > > _______________________________________________
> > > Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >=20
> > >=20
> >=20
> >=20
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >=20
> >=20
> >=20
> > --=20
> > Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > Telecom R&D               Fax   :  +358  9 299 30 52
> > FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > Finland                   http://www.hut.fi/~gonzalo
> >=20
>=20
> This communication is confidential and intended solely for=20
> the addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is prohibited. If you believe this message has=20
> been sent to you in error, please notify the sender by=20
> replying to this transmission and delete the message without=20
> disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data=20
> corruption, interruption, unauthorized amendment, tampering=20
> and viruses, and we only send and receive e-mails on the=20
> basis that we are not liable for any such corruption,=20
> interception, amendment, tampering or viruses or any=20
> consequences thereof.
>=20
>=20
>=20


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



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Arnoud van Wijk" <a.vwijk@viataal.nl>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Fri, 27 Feb 2004 07:45:27 +0100
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The INCOM group of the European commission made recommendations that the
same emergency number shall be used for all kinds of terminals.
It is dangerous to have different prefixes or other tricks in different
countries. How shall you keep track of that if you are a visitor?

From PSTN, it is only the old textphones that cause problems. A calling V.18
sends a very distinct calling identifier that can be catched by a gateway or
an emergency centre and be properly handled. On the SIP side, the options
are even better.

I do not mean that all problems are solved by this statement, but I do not
think that the prefix is the only solution, and I do not think we shall
recommend it in connection to emergency.


Gunnar

-------------------------------------------
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Arnoud van Wijk
Sent: Thursday, February 26, 2004 4:57 PM
To: 'Rosen, Brian'; 'Gunnar Hellstrom'; Gonzalo.Camarillo@ericsson.com;
sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
paulej@packetizer.com; gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements


Hello Brian,

Yes you are right about that the PSTN cannot detect if it is a voice call or
TDD/Textphone call. They are both audio.
The PSTN would if it can detect it is a textphone call, route it to the text
gateway. And transcode the call into T.140 over RTP as decribed in RFC 2793.
This is for PSTNs which use IP to interconnect with other PSTNs or with any
SIP telephones. So it is for analoge telephony into SIP telephony here.

In the ToIP draft:
http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-toip-01.txt
A possible solution is a prefix for textphone users before dialling the
selected number.
It indeed would mean that you dial e.g. 600-112 instead 112 directly.
Is that bad?
I believe it is not.

Most textphones have the emergency number pre programmed. And when textphone
users have to use a pre-fix for any textphone call, then the same prefix
before 112 or 911 will be fine.
In the Netherlands, we have for emergency calls 800-112 instead of 112. It
works fine, only almost no deaf users call 112, since they are then mostly
in a situation where they do not have a textphone nearby!

Maybe there are other solutions that allow the PSTN to detect that the
incoming call is a textphone user and acts on it.

For example it is possible to create a database where the PSTN can check if
the incoming number is registered as a textphone.

But using a text gateway to connect PSTN textphones/TDDs to ToIP is still
the best solution, even the only solution in my opinion.
There should be no attempts to transport the text telephone tones (e.g.
Baudot) over the IP voice channel! I believe that will open Pandora's box.

I am not a PSTN expert, and we can surely look into this matter to find a
solution. Let it be prefix or any other solution that lets the PSTN
immediately route the call to the text gateway.

What do you PSTN people think? How would you tackle this problem? Your input
will be very welcome.

Greetz

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands.
Mobile: +31651921948
International text telephone: +31735588408


-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
Sent: donderdag 26 februari 2004 15:36
To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; Gonzalo.Camarillo@ericsson.com;
sipping@ietf.org
Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; henry.sinnreich@mci.com;
paulej@packetizer.com; gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements

Arnoud

I do not believe this can work.  A TDD user dials 112 or 911.  Today, this
will be routed directly to a "selective router" (a TDM switch) which sends
the
call to the correct Emergency Response Center.  You can't insert a gateway,
because you don't know that it's a TDD call until it is answered.  To make
it work, you would have to establish a special PSTN number for emergency
TDD,
which would route to the gateway, which would then call the ERC.  I don't
believe that is practical.

Brian



> -----Original Message-----
> From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> Sent: Thursday, February 26, 2004 9:17 AM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
>
>
> Hello Brian,
> That is also why we need to look at the emergency call system.
> This text gateway is only at the PSTN side, thus another
> requirement for a
> T140/RTP gateway for the PSTN is that when a local emergency number is
> dialled (112, or 911 etc). That the gateway must be able to
> receive that
> call (even dropping existing calls if necessary). Such a call
> is easy to
> recognize as an emergency call. (PSTN side).
> From that gateway on, it is all IP with RFC2793 interactive
> text. And then
> it is just all existing SIP as if it was from an IP end-point.
>
> This gives emergency centers an advantage, all calls will be
> T.140/RTP, so
> integration in their computer systems will be easy instead
> being forced to
> have analog textphones available for a textphone call. (or a
> separate text
> telephone emergency center).
>
> /Arnoud
>
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
>
>
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: donderdag 26 februari 2004 15:04
> To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
>
> Arnoud
>
> Generally, you don't want to rely on very many SIP elements
> to complete
> an emergency call.  Present systems accept TDD at all answering
> points because they don't know before the calls is answered that it
> is a TDD call.  I certainly don't want the user to have to do anything
> other than dial his local emergency number, or complete a device
> specific key sequence to complete an emergency call, and I don't
> want a gateway busy to prevent an emergency call.
>
> Brian
>
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 6:07 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> >
> > Actually,
> > When using the text gateway, there is no need to detect the
> > TDD/Analogue
> > text telephone tones/signals. That is the job of the text gateway.
> >
> > And for the emergency calling. We have to look closely at this in
> > combination with a text gateway. Possible behaviour/action.
> > And possible
> > action in SIP.
> >
> > Right now I cannot say what has to happen, I must study the
> > emergency calls
> > in depth. But as I see it, using the text gateway, all SIP
> > originating and
> > receiving emergency calls will be possible for any textphone user.
> >
> > Cheers
> >
> > Arnoud
> >
> >
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
> >
> > -----Original Message-----
> > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > On Behalf Of
> > Rosen, Brian
> > Sent: woensdag 25 februari 2004 23:08
> > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> > I'm particularly interested in the last one, emergency calling.
> >
> > Given that you propose an appropriate SDP indicating
> support for text,
> > do you think there is anything else we need to do to in SIP
> > to make this
> > happen?  I don't see anything.
> >
> > I know that the problem of using a legacy TDD device on any kind of
> > adapter is as problematic as current TDD; the call will answered
> > as if it was a voice call, the baudot/whatever tones will be
> > detected,
> > and the operator will look for text.
> > Can't think of any way to make that better.
> >
> > Do note that language preference headers should be set!!!
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > I think PSTN textphone gateway invocation is one area where
> > > we either need
> > > protocol extensions, or need agreements on what current
> > > mechanisms to use.
> > >
> > > Another is invocation of various kinds of relay services or
> > > translation
> > > services.
> > >
> > > A third is calling emergency services and reach positions
> capable of
> > > handling the type of communication that the user prefers.
> > >
> > >
> > >
> > > Gunnar
> > > -------------------------------------------
> > > Gunnar Hellstrom
> > > Omnitor AB
> > > Renathvagen 2
> > > SE 121 37 Johanneshov
> > > SWEDEN
> > > +46 8 556 002 03
> > > Mob: +46 708 204 288
> > > www.omnitor.se
> > > Gunnar.Hellstrom@Omnitor.se
> > > --------------------------------------------
> > >
> > >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org
> > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > A vWijk
> > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > gv@trace.wisc.edu
> > > Subject: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > Hello Gonzalo,
> > > While there are device requirements in the text. Those
> > > requirements and the
> > > explanation on the different analogue text telephone systems
> > > is included to
> > > give the reader an understanding what kind of devices will
> > > connect with
> > > ToIP. And the need for the text gateways. And what the
> > > different behaviour
> > > can/may be compared to end to end IP devices.
> > > The most important part will be the ToIP framwork and the
> > > text gateways and
> > > how such gateways will be used and work with the ToIP framework.
> > >
> > > There is a ToIP framework where a terminal uses streaming
> > > text as defined in
> > > RFC2793 to connect with any other Interactive
> > > texting/real-time streaming
> > > text device.
> > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > established with
> > > SIP. But in other cases, a text gateway must be used to allow
> > > T.140/RTP to
> > > connect with a non IP text telephony device.
> > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > text gateway to
> > > be able to understand RFC2793 text. (PSTN-IP)
> > > It can be a mobile phone running GPRS that uses the text
> gateway to
> > > understand RFC2793 text. (this exist right NOW using SIP! And
> > > those gateways
> > > will be used in mobile networks this year!)
> > > Or a mobile text telephone using CTM via the text gateway to
> > > communicate
> > > with RFC2793 text.
> > > Or even one analogue text telephone connecting to another
> > > analogue text
> > > telephone in a different country, and the PSTN's use IP to
> > > connect the call.
> > >
> > > The question is for example, how do we use that gateway from
> > > the IP side? Is
> > > it invisible for the IP side? Or do we need special
> > > extensions in SIP to
> > > that the IP terminal knows it is communicating with the
> > > end-device through
> > > the text gateway?
> > > There may be a slightly different call setup flow when a text
> > > gateway is
> > > included.
> > > Or additional SIP messages indicating gateway activity. When
> > > you connect for
> > > example to an analogue text telephone with an RFC2793 capable
> > > IP terminal
> > > the capabilities will be less then IP to IP terminal directly
> > > (different
> > > transmission speeds, half/full duplex etc).
> > > Or what to do with emergency calls that are in RFC 2793 text only?
> > >
> > > That are requirements that will be defined and worked out in
> > > this draft.
> > > Also, we want to make everybody in SIP and SIPPING aware
> > > about those text
> > > gateways. So that complicated matters like how to send
> analogue text
> > > telephone text through a PSTN and connect with a SIP
> > > telephone using RFC2793
> > > interactive texting. When using the text gateway, there is
> > > then no need for
> > > RFC 2833 events at the PSTN for text telephones.
> > >
> > > So, yes. I believe that this should be in the SIPPING WG. And
> > > it will fit in
> > > the charter - support for hearing-/speech-impaired calling.
> > >
> > > greetz
> > >
> > > Arnoud
> > >
> > >
> > >
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > >
> > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > 24-02-04 10:21 >>>
> > > Hello,
> > >
> > > the draft below provides a set of requirements for real-time
> > > conversations using SIP.
> > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > ip-01.txt
> >
> > Nevertheless, I have found a lot of device requirements in
> > the text, but
> > I fail to understand which particular protocol requirements
> we need to
> > fulfill with new protocol extensions.
> >
> > While SIPPING is the right place to handle requirements
> that will lead
> > to new SIP protocol extensions, SIPPING may not be the
> right place to
> > deal with device requirements (e.g., text phones MUST have a
> > display of
> > at least 100 characters).
> >
> > Thanks,
> >
> > Gonzalo
> >
> >
> > This communication is confidential and intended solely for
> > the addressee(s).
> > Any unauthorized review, use, disclosure or distribution is
> > prohibited. If
> > you believe this message has been sent to you in error,
> > please notify the
> > sender by replying to this transmission and delete the
> message without
> > disclosing it. Thank you.
> >
> > E-mail including attachments is susceptible to data corruption,
> > interruption, unauthorized amendment, tampering and viruses,
> > and we only
> > send and receive e-mails on the basis that we are not liable
> > for any such
> > corruption, interception, amendment, tampering or viruses or any
> > consequences thereof.
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
> >
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
> >
>
>



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




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



From exim@www1.ietf.org  Fri Feb 27 02:17:36 2004
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Folks,

those of you who will be presenting in any of the two SIPPING sessions, 
please, send us your slides so that we can make them available to 
everyone at:

http://www.softarmor.com/sipping/meets/ietf59/slides/

Thanks,

Gonzalo


This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>
Cc: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Dean Willis'" <dean.willis@softarmor.com>,
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        "'Rohan Mahy'" <rohan@cisco.com>, <hgxs@cs.columbia.edu>
Subject: AW: AW: AW: [Sipping] SIPPING Agenda, IETF 59
Date: Fri, 27 Feb 2004 12:40:05 +0100
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Gonzalo,

understood. Regarding "Additional requests will be=20
> accomodated=20
> if time allows"

how would that work?

	-h

> -----Urspr=FCngliche Nachricht-----
> Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
> Gesendet: Freitag, 27. Februar 2004 07:57
> An: Harry Behrens
> Cc: 'Henning Schulzrinne'; 'Dean Willis'; 'James M. Polk';=20
> sipping@ietf.org; 'Rohan Mahy'; hgxs@cs.columbia.edu
> Betreff: Re: AW: AW: [Sipping] SIPPING Agenda, IETF 59
>=20
>=20
> Harry,
>=20
> our agenda is pretty much full at this point, and, as you know, we=20
> cannot allocate time for draft presentations. In any event,=20
> if there are=20
> any open issues that have been discussed in the mailing list that you=20
> would like to further discuss, we can draw the attention of the WG to=20
> your work.
>=20
> Of course, as the agenda says, "Additional requests will be=20
> accomodated=20
> if time allows".
>=20
> Thanks,
>=20
> Gonzalo
>=20
> Harry Behrens wrote:
>=20
> > That too.
> > However presenting the latest relevant AAA and Billing=20
> related drafts=20
> > seemed like a good idea in general. It appears to me to be=20
> a topic of=20
> > interest and I am not presumptuous enough to assume that my=20
> draft is=20
> > the only document worthy of attention in that field ;-)
> >=20
> > So: if such a "track" would exist I would like to reqeust=20
> to have one=20
> > slot allocated to my draft. If however, the notion of such=20
> a track is=20
> > not agreeable I would probably just like to request a slot.
> >=20
> >=20
> > 	Harry
> >=20
> >=20
> >=20
> >>-----Urspr=FCngliche Nachricht-----
> >>Von: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> >>Gesendet: Donnerstag, 26. Februar 2004 12:07
> >>An: Harry Behrens
> >>Cc: 'Henning Schulzrinne'; 'Dean Willis'; 'James M. Polk';=20
> >>sipping@ietf.org; 'Rohan Mahy'; hgxs@cs.columbia.edu
> >>Betreff: Re: AW: [Sipping] SIPPING Agenda, IETF 59
> >>
> >>
> >>Harry,
> >>
> >>do you mean that you would like to request a time slot to
> >>present your=20
> >>draft in SIPPING?
> >>
> >>Gonzalo
> >>
> >>Harry Behrens wrote:
> >>
> >>
> >>>May I suggest a SIP, AAA and Billing Track?
> >>>I would assume this to be of interest as we are seeing tremendous
> >>>growth in the number of companies entering the ITSP market=20
> >>
> >>these days.
> >>
> >>>Regards,
> >>>
> >>>	Harry
> >>>
> >>>
> >>>
> >>>
> >>>>-----Urspr=FCngliche Nachricht-----
> >>>>Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] Im=20
> >>>>Auftrag von Henning Schulzrinne
> >>>>Gesendet: Dienstag, 24. Februar 2004 22:11
> >>>>An: Dean Willis
> >>>>Cc: James M. Polk; sipping@ietf.org; Rohan Mahy; Gonzalo
> >>>>Camarillo; hgxs@cs.columbia.edu
> >>>>Betreff: Re: [Sipping] SIPPING Agenda, IETF 59
> >>>>
> >>>>
> >>>>Reversal is fine with me.
> >>>>
> >>>>Dean Willis wrote:
> >>>>
> >>>>
> >>>>
> >>>>>James M. Polk wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>>At 01:02 PM 2/24/2004 -0600, Dean Willis wrote:
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>>Hyperlinked version at:=20
> >>>>>>>http://www.softarmor.com/sipping/meets/ietf59/agenda.html
> >>>>>>>
> >>>>>>>WEDNESDAY, March 3, 2004, 1300-1500, Crystal 3
> >>>>>>>
> >>>>>>>1335 SIP Emergency
> >>>>>>>         Henning Schulzrinne
> >>>>>>>         draft-ietf-sipping-sos-00.txt
> >>>>>>>         draft-schulzrinne-sipping-emergency-arch-00.txt
> >>>>>>>1345 Location Conveyance Requirements
> >>>>>>>         James Polk
> >>>>>>>         draft-ietf-sipping-location-requirements-00.txt
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>I believe these presentations should be reversed in order
> >>>>
> >>>>because the
> >>>>
> >>>>
> >>>>>>main points of discussion in the Location Requirements=20
> ID are for=20
> >>>>>>Emergency location dealing with call routing by proxies,
> >>
> >>adding LI
> >>
> >>>>>>(location information) by the proxy (whether or not the UA
> >>>>
> >>>>inserted it
> >>>>
> >>>>
> >>>>>>already) and the security of the LI in transit.=20
> Discussing these=20
> >>>>>>points after the Emer-Arch doc doesn't seem to make much
> >>>>
> >>>>sense due to
> >>>>
> >>>>
> >>>>>>potential rehash while the appropriate presenter is not at
> >>>>
> >>>>the podium
> >>>>
> >>>>
> >>>>>>(in either direction).
> >>>>>>
> >>>>>>comments?
> >>>>>
> >>>>>
> >>>>>Works for me. Henning?
> >>>>>
> >>>>>--
> >>>>>Dean
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sipping mailing list
> >>
> >>https://www1.ietf.org/mailman/listinfo/sipping
> >>
> >>>>>This list
> >>
> >>is for NEW development of the application of SIP Use
> >>
> >>>>>sip-implementors@cs.columbia.edu for questions on current sip Use
> >>>>>sip@ietf.org for new developments of core SIP
> >>>>
> >>>>_______________________________________________
> >>>>Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >>>>This list is for NEW development of the application of SIP Use
> >>>>sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>>>sip@ietf.org for new developments of core SIP
> >>>
> >>>
> >>--=20
> >>Gonzalo Camarillo         Phone :  +358  9 299 33 71
> >>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> >>Telecom R&D               Fax   :  +358  9 299 30 52
> >>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> >>Finland                   http://www.hut.fi/~gonzalo
> >>
> >>
> >>This communication is confidential and intended solely for
> >>the addressee(s). Any unauthorized review, use, disclosure or=20
> >>distribution is prohibited. If you believe this message has=20
> >>been sent to you in error, please notify the sender by=20
> >>replying to this transmission and delete the message without=20
> >>disclosing it. Thank you.
> >>
> >>E-mail including attachments is susceptible to data
> >>corruption, interruption, unauthorized amendment, tampering=20
> >>and viruses, and we only send and receive e-mails on the=20
> >>basis that we are not liable for any such corruption,=20
> >>interception, amendment, tampering or viruses or any=20
> >>consequences thereof.
> >=20
> >=20
>=20
> --=20
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland                   http://www.hut.fi/~gonzalo
>=20
>=20
> This communication is confidential and intended solely for=20
> the addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is prohibited. If you believe this message has=20
> been sent to you in error, please notify the sender by=20
> replying to this transmission and delete the message without=20
> disclosing it. Thank you.
>=20
> E-mail including attachments is susceptible to data=20
> corruption, interruption, unauthorized amendment, tampering=20
> and viruses, and we only send and receive e-mails on the=20
> basis that we are not liable for any such corruption,=20
> interception, amendment, tampering or viruses or any=20
> consequences thereof.
>=20


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



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Date: Fri, 27 Feb 2004 10:56:36 -0600
From: Dean Willis <dean.willis@softarmor.com>
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To: Harry Behrens <harry.behrens@snom.de>
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        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'James M. Polk'" <jmpolk@cisco.com>, sipping@ietf.org,
        "'Rohan Mahy'" <rohan@cisco.com>, hgxs@cs.columbia.edu
Subject: Re: AW: AW: AW: [Sipping] SIPPING Agenda, IETF 59
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Harry Behrens wrote:
> Gonzalo,
> 
> understood. Regarding "Additional requests will be 
> 
>>accomodated 
>>if time allows"
> 
> 
> how would that work?

It means if we get to the end of the agenda and still have time left, 
the chairs ask "Ok, we're at the end of the agenda. Anybody have 
anything else to talk about?", and you and other folks can pop up and 
say "I want to talk about . . .", and we judge the consensus of the room 
on what topic to talk about by priority.

--
Dean

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



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From: "Harry Behrens" <harry.behrens@snom.de>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Cc: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'James M. Polk'" <jmpolk@cisco.com>, <sipping@ietf.org>,
        "'Rohan Mahy'" <rohan@cisco.com>
Subject: AW: AW: AW: AW: [Sipping] SIPPING Agenda, IETF 59
Date: Fri, 27 Feb 2004 18:23:02 +0100
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Dear Dean,

understood. Thank you for the explanation.

See you in Seoul.

	Harry

> -----Urspr=FCngliche Nachricht-----
> Von: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]=20
> Im Auftrag von Dean Willis
> Gesendet: Freitag, 27. Februar 2004 17:57
> An: Harry Behrens
> Cc: 'Gonzalo Camarillo'; 'Henning Schulzrinne'; 'James M.=20
> Polk'; sipping@ietf.org; 'Rohan Mahy'; hgxs@cs.columbia.edu
> Betreff: Re: AW: AW: AW: [Sipping] SIPPING Agenda, IETF 59
>=20
>=20
> Harry Behrens wrote:
> > Gonzalo,
> >=20
> > understood. Regarding "Additional requests will be
> >=20
> >>accomodated
> >>if time allows"
> >=20
> >=20
> > how would that work?
>=20
> It means if we get to the end of the agenda and still have time left,=20
> the chairs ask "Ok, we're at the end of the agenda. Anybody have=20
> anything else to talk about?", and you and other folks can pop up and=20
> say "I want to talk about . . .", and we judge the consensus=20
> of the room=20
> on what topic to talk about by priority.
>=20
> --
> Dean
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20


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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: <sip@ietf.org>, "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: "Toip list" <toip@snowshore.com>, "'Sipping@Ietf. Org'" <sipping@ietf.org>
Date: Fri, 27 Feb 2004 22:22:48 +0100
Message-ID: <BHEHLFPKIPMLPFNFAHJKMEMLDLAA.gunnar.hellstrom@omnitor.se>
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Subject: [Sipping] [sip]  Comment to draft-agrawal-sip-h323-interworking-reqs-06.txt about text support
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In section 6.1.5 of the sip-h323-interworking spec, there is a sentence:

"The IWF SHOULD be able to map the common audio, video and application
   format names supported in H.323 to and from the equivalent RTP/AVP
   [RFC3550] names."

It is good practice to always mention how the three main media of video,
audio and text shall be handled in real time applications.

audio and video are mentioned explicitly.

It may not be evident to everybody that real time interactive text is
capability declared in H.323 Annex G / H.245 by
DataApplicationCapability.application =3D t140

So, in fact it is covered in the sentence requiring that "application"
format namnes shall be mapped.

The mapping should correspond to the RTP/AVP  name text/t140 and
corresponding names for inclusion of protection against  loss.

I seek your guidance how it can be made more clear that "text" capabiliti=
es
can be expressed in both H.323 and SIP systems and should be mapped.

( the spec is at
http://www.ietf.org/internet-drafts/draft-agrawal-sip-h323-interworking-r=
eqs
-06.txt )

Regards
Gunnar
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------




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From exim@www1.ietf.org  Fri Feb 27 18:40:21 2004
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Gonzalo Camarillo'" <Gonzalo.Camarillo@ericsson.com>,
        sipping
	 <sipping@ietf.org>
Cc: Dean Willis <dean.willis@softarmor.com>, Rohan Mahy <rohan@cisco.com>,
        "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Sipping] WGLC for draft-ietf-sipping-reason-header-for-preem
	ption-00.txt
Date: Fri, 27 Feb 2004 17:34:56 -0600
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The document looks good to me at a skim, but I believe
there are a couple of points of clarification that should
be added to 4.2.1.

First, I believe the label for the right-most bar in the
diagram should be "UA2" instead of "UA3," if my understanding
of the use case is correct.

Secondly, it is not clear how the "Resource-Priority: 2" in
the UA3-UA4 call comes into play. None of the UAs shown in
this flow are privy to any of the SIP signaling that was used
to establish the UA3-UA4 call. What would seem to matter is
the RSVP priorities used to establish the call. Correct?
This should probably be clarified in the text.

/a

> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: Sunday, February 08, 2004 15:29
> To: sipping
> Cc: Dean Willis; Rohan Mahy; James M. Polk
> Subject: [Sipping] WGLC for
> draft-ietf-sipping-reason-header-for-preemption-00.txt
> 
> 
> Folks,
> 
> we would like to announce the start of the WGLC for the 
> following draft.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-reason-
header-for-preemption-00.txt

This WGLC will end on Feb 29th.

Send your comments to this mailing list and to the author.

Thanks,

Gonzalo
SIPPING co-chair



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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Rohan Mahy <rohan@cisco.com>, hisham.khartabil@nokia.com, sipping@ietf.org
Subject: Re: [Sipping] Conference policy URI
References: <2038BCC78B1AD641891A0D1AE133DBB701797704@esebe019.ntc.nokia.com> <40293C6B.4060404@dynamicsoft.com> <26C87A2E-621D-11D8-9B6B-0003938AF740@cisco.com> <404036BC.9020001@dynamicsoft.com>
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Hi,

Jonathan wrote:
> As for the expense of creating registries - I dont think we should let 
> current administrative difficulties preclude us from doing what we think 
> the right thing is.

This is my view as well.

Gonzalo


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Date: Fri, 27 Feb 2004 09:15:42 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
CC: Arnoud van Wijk <a.vwijk@viataal.nl>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        Gonzalo.Camarillo@ericsson.com, sipping@ietf.org, rrroy@att.com,
        barry.dingle@bigfoot.com.au, henry.sinnreich@mci.com,
        paulej@packetizer.com, gv@trace.wisc.edu
Subject: Re: Betr.: [Sipping] ToIP Requirements
References: <BHEHLFPKIPMLPFNFAHJKKELMDLAA.gunnar.hellstrom@omnitor.se>
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Gunnar Hellstrom wrote:

> The INCOM group of the European commission made recommendations that the
> same emergency number shall be used for all kinds of terminals.
> It is dangerous to have different prefixes or other tricks in different
> countries. How shall you keep track of that if you are a visitor?

There are no different 'tricks'. Please read the 'sos' and 
emergency-arch drafts; we would appreciate comments, but general worry 
not grounded on actual proposals is somewhat less helpful.

Indeed, the proposals improve on the current system in that

- all countries will use the same identifier, sos@
- visitors can use their home and visited country identifiers
- discovery of these identifiers is automated
- callers MAY, but do not have to, indicate preferences for the kind of 
emergency service and the kind of media desired; the latter is most 
likely automatic, not something a user would type in (indeed, they can't)

> 
>>From PSTN, it is only the old textphones that cause problems. A calling V.18
> sends a very distinct calling identifier that can be catched by a gateway or
> an emergency centre and be properly handled. On the SIP side, the options
> are even better.
> 
> I do not mean that all problems are solved by this statement, but I do not
> think that the prefix is the only solution, and I do not think we shall
> recommend it in connection to emergency.

As far as I can tell, nobody is recommending a 'prefix'.

> 
> 
> Gunnar
> 
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: "Barry Dingle" <barry.dingle@bigfoot.com.au>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "'Arnoud van Wijk'" <a.vwijk@viataal.nl>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
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References: <313680C9A886D511A06000204840E1CF070B646A@whq-msgusr-02.pit.comms.marconi.com>
Subject: Re: Betr.: [Sipping] ToIP Requirements
Date: Fri, 27 Feb 2004 10:48:41 +1100
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Brian,

In Australia, we have a separate emergency number for Textphone users. It is
'106' and is regulated by legislation.

We have a separate '106' emergency call centre. This makes special support
for a range of text services much easier. (See
http://www.aceinfo.net.au/Services/NRS/Services/ers.html).  I understand
that this may be difficult to implement for historical reasons in some
countries - but it is possible and works well.

As an aside, we are seriously investigating a 'Text Server' to support a
whole range of text interworking scenarios - and ToIP is planned to be
supported in Stage 2. I believe that we will eventually move to a single
emergency number - but not until we can easily and reliably identify text
emergency calls.

Cheers,

Barry DINGLE
Telecommunications Consultant
Project Manager - ACIF Any-to-Any Text Connectivity Options WG
Fellow of the University of Melbourne
   barry.dingle@bigfoot.com.au
Mob:   +61 (0)41 911 7578
Fixed: +61 (0)3 9725 3937
----- Original Message ----- 
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>; "'Gunnar Hellstrom'"
<gunnar.hellstrom@omnitor.se>; <Gonzalo.Camarillo@ericsson.com>;
<sipping@ietf.org>
Cc: <rrroy@att.com>; <barry.dingle@bigfoot.com.au>;
<henry.sinnreich@mci.com>; <paulej@packetizer.com>; <gv@trace.wisc.edu>
Sent: Friday, February 27, 2004 1:36 AM
Subject: RE: Betr.: [Sipping] ToIP Requirements


> Arnoud
>
> I do not believe this can work.  A TDD user dials 112 or 911.  Today, this
> will be routed directly to a "selective router" (a TDM switch) which sends
> the
> call to the correct Emergency Response Center.  You can't insert a
gateway,
> because you don't know that it's a TDD call until it is answered.  To make
> it work, you would have to establish a special PSTN number for emergency
> TDD,
> which would route to the gateway, which would then call the ERC.  I don't
> believe that is practical.
>
> Brian
>
>
>
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 9:17 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> >
> > Hello Brian,
> > That is also why we need to look at the emergency call system.
> > This text gateway is only at the PSTN side, thus another
> > requirement for a
> > T140/RTP gateway for the PSTN is that when a local emergency number is
> > dialled (112, or 911 etc). That the gateway must be able to
> > receive that
> > call (even dropping existing calls if necessary). Such a call
> > is easy to
> > recognize as an emergency call. (PSTN side).
> > From that gateway on, it is all IP with RFC2793 interactive
> > text. And then
> > it is just all existing SIP as if it was from an IP end-point.
> >
> > This gives emergency centers an advantage, all calls will be
> > T.140/RTP, so
> > integration in their computer systems will be easy instead
> > being forced to
> > have analog textphones available for a textphone call. (or a
> > separate text
> > telephone emergency center).
> >
> > /Arnoud
> >
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
> >
> >
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: donderdag 26 februari 2004 15:04
> > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> > Arnoud
> >
> > Generally, you don't want to rely on very many SIP elements
> > to complete
> > an emergency call.  Present systems accept TDD at all answering
> > points because they don't know before the calls is answered that it
> > is a TDD call.  I certainly don't want the user to have to do anything
> > other than dial his local emergency number, or complete a device
> > specific key sequence to complete an emergency call, and I don't
> > want a gateway busy to prevent an emergency call.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > Sent: Thursday, February 26, 2004 6:07 AM
> > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > Actually,
> > > When using the text gateway, there is no need to detect the
> > > TDD/Analogue
> > > text telephone tones/signals. That is the job of the text gateway.
> > >
> > > And for the emergency calling. We have to look closely at this in
> > > combination with a text gateway. Possible behaviour/action.
> > > And possible
> > > action in SIP.
> > >
> > > Right now I cannot say what has to happen, I must study the
> > > emergency calls
> > > in depth. But as I see it, using the text gateway, all SIP
> > > originating and
> > > receiving emergency calls will be possible for any textphone user.
> > >
> > > Cheers
> > >
> > > Arnoud
> > >
> > >
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > On Behalf Of
> > > Rosen, Brian
> > > Sent: woensdag 25 februari 2004 23:08
> > > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > > I'm particularly interested in the last one, emergency calling.
> > >
> > > Given that you propose an appropriate SDP indicating
> > support for text,
> > > do you think there is anything else we need to do to in SIP
> > > to make this
> > > happen?  I don't see anything.
> > >
> > > I know that the problem of using a legacy TDD device on any kind of
> > > adapter is as problematic as current TDD; the call will answered
> > > as if it was a voice call, the baudot/whatever tones will be
> > > detected,
> > > and the operator will look for text.
> > > Can't think of any way to make that better.
> > >
> > > Do note that language preference headers should be set!!!
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > I think PSTN textphone gateway invocation is one area where
> > > > we either need
> > > > protocol extensions, or need agreements on what current
> > > > mechanisms to use.
> > > >
> > > > Another is invocation of various kinds of relay services or
> > > > translation
> > > > services.
> > > >
> > > > A third is calling emergency services and reach positions
> > capable of
> > > > handling the type of communication that the user prefers.
> > > >
> > > >
> > > >
> > > > Gunnar
> > > > -------------------------------------------
> > > > Gunnar Hellstrom
> > > > Omnitor AB
> > > > Renathvagen 2
> > > > SE 121 37 Johanneshov
> > > > SWEDEN
> > > > +46 8 556 002 03
> > > > Mob: +46 708 204 288
> > > > www.omnitor.se
> > > > Gunnar.Hellstrom@Omnitor.se
> > > > --------------------------------------------
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org
> > > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > > A vWijk
> > > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > > gv@trace.wisc.edu
> > > > Subject: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > Hello Gonzalo,
> > > > While there are device requirements in the text. Those
> > > > requirements and the
> > > > explanation on the different analogue text telephone systems
> > > > is included to
> > > > give the reader an understanding what kind of devices will
> > > > connect with
> > > > ToIP. And the need for the text gateways. And what the
> > > > different behaviour
> > > > can/may be compared to end to end IP devices.
> > > > The most important part will be the ToIP framwork and the
> > > > text gateways and
> > > > how such gateways will be used and work with the ToIP framework.
> > > >
> > > > There is a ToIP framework where a terminal uses streaming
> > > > text as defined in
> > > > RFC2793 to connect with any other Interactive
> > > > texting/real-time streaming
> > > > text device.
> > > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > > established with
> > > > SIP. But in other cases, a text gateway must be used to allow
> > > > T.140/RTP to
> > > > connect with a non IP text telephony device.
> > > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > > text gateway to
> > > > be able to understand RFC2793 text. (PSTN-IP)
> > > > It can be a mobile phone running GPRS that uses the text
> > gateway to
> > > > understand RFC2793 text. (this exist right NOW using SIP! And
> > > > those gateways
> > > > will be used in mobile networks this year!)
> > > > Or a mobile text telephone using CTM via the text gateway to
> > > > communicate
> > > > with RFC2793 text.
> > > > Or even one analogue text telephone connecting to another
> > > > analogue text
> > > > telephone in a different country, and the PSTN's use IP to
> > > > connect the call.
> > > >
> > > > The question is for example, how do we use that gateway from
> > > > the IP side? Is
> > > > it invisible for the IP side? Or do we need special
> > > > extensions in SIP to
> > > > that the IP terminal knows it is communicating with the
> > > > end-device through
> > > > the text gateway?
> > > > There may be a slightly different call setup flow when a text
> > > > gateway is
> > > > included.
> > > > Or additional SIP messages indicating gateway activity. When
> > > > you connect for
> > > > example to an analogue text telephone with an RFC2793 capable
> > > > IP terminal
> > > > the capabilities will be less then IP to IP terminal directly
> > > > (different
> > > > transmission speeds, half/full duplex etc).
> > > > Or what to do with emergency calls that are in RFC 2793 text only?
> > > >
> > > > That are requirements that will be defined and worked out in
> > > > this draft.
> > > > Also, we want to make everybody in SIP and SIPPING aware
> > > > about those text
> > > > gateways. So that complicated matters like how to send
> > analogue text
> > > > telephone text through a PSTN and connect with a SIP
> > > > telephone using RFC2793
> > > > interactive texting. When using the text gateway, there is
> > > > then no need for
> > > > RFC 2833 events at the PSTN for text telephones.
> > > >
> > > > So, yes. I believe that this should be in the SIPPING WG. And
> > > > it will fit in
> > > > the charter - support for hearing-/speech-impaired calling.
> > > >
> > > > greetz
> > > >
> > > > Arnoud
> > > >
> > > >
> > > >
> > > > Drs. Arnoud A. T. van Wijk
> > > > Viataal
> > > > Research & Development
> > > > Afdeling RDS
> > > > Theerestraat 42
> > > > 5271 GD Sint-Michielsgestel
> > > > The Netherlands.
> > > > Mobile: +31651921948
> > > > International text telephone: +31735588408
> > > >
> > > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > > 24-02-04 10:21 >>>
> > > > Hello,
> > > >
> > > > the draft below provides a set of requirements for real-time
> > > > conversations using SIP.
> > > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > > ip-01.txt
> > >
> > > Nevertheless, I have found a lot of device requirements in
> > > the text, but
> > > I fail to understand which particular protocol requirements
> > we need to
> > > fulfill with new protocol extensions.
> > >
> > > While SIPPING is the right place to handle requirements
> > that will lead
> > > to new SIP protocol extensions, SIPPING may not be the
> > right place to
> > > deal with device requirements (e.g., text phones MUST have a
> > > display of
> > > at least 100 characters).
> > >
> > > Thanks,
> > >
> > > Gonzalo
> > >
> > >
> > > This communication is confidential and intended solely for
> > > the addressee(s).
> > > Any unauthorized review, use, disclosure or distribution is
> > > prohibited. If
> > > you believe this message has been sent to you in error,
> > > please notify the
> > > sender by replying to this transmission and delete the
> > message without
> > > disclosing it. Thank you.
> > >
> > > E-mail including attachments is susceptible to data corruption,
> > > interruption, unauthorized amendment, tampering and viruses,
> > > and we only
> > > send and receive e-mails on the basis that we are not liable
> > > for any such
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> > > consequences thereof.
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > > _______________________________________________
> > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > > _______________________________________________
> > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > >
> >
> >
>



_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Sun, 29 Feb 2004 03:40:24 +0100
From: "A vWijk" <A.vWijk@viataal.nl>
To: <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>,
        <Brian.Rosen@marconi.com>, <gunnar.hellstrom@omnitor.se>
Cc: <len.bytheway@aceinfo.net.au>, <rrroy@att.com>,
        <barry.dingle@bigfoot.com.au>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: Betr.: Re: Betr.: [Sipping] ToIP Requirements
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Hi Barry and others,
the text server can actually be the text gateway described in the ToIP =
draft.

I think it is a good idea to communicate about those plans of the text =
server, so that we can include all the requirements into the ToIP draft.

About analogue textphones and PSTNs. The requirement should be that the =
PSTN MUST be able to recognize an incoming call as an analogue text =
telephone call (opposed to voice and fax). The PSTN MUST then invoke the =
text gateway for further call handling. In this way, the analogue text =
telephony is then immediately transcoded into RFC2793 interactive texting =
(T.140/RTP).
The emergency systems shouldn't need any special handling for analogue =
textphones since it is pure SIP telephony from then on.

Greetz

Arnoud

<<< "Barry Dingle" <barry.dingle@bigfoot.com.au> 29-02 02:34  >>>
Brian,

In Australia, we have a separate emergency number for Textphone users. It =
is
'106' and is regulated by legislation.

We have a separate '106' emergency call centre. This makes special support
for a range of text services much easier. (See
http://www.aceinfo.net.au/Services/NRS/Services/ers.html).  I understand
that this may be difficult to implement for historical reasons in some
countries - but it is possible and works well.

As an aside, we are seriously investigating a 'Text Server' to support a
whole range of text interworking scenarios - and ToIP is planned to be
supported in Stage 2. I believe that we will eventually move to a single
emergency number - but not until we can easily and reliably identify text
emergency calls.

Cheers,

Barry DINGLE
Telecommunications Consultant
Project Manager - ACIF Any-to-Any Text Connectivity Options WG
Fellow of the University of Melbourne
   barry.dingle@bigfoot.com.au
Mob:   +61 (0)41 911 7578
Fixed: +61 (0)3 9725 3937
----- Original Message -----=20
From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>; "'Gunnar Hellstrom'"
<gunnar.hellstrom@omnitor.se>; <Gonzalo.Camarillo@ericsson.com>;
<sipping@ietf.org>
Cc: <rrroy@att.com>; <barry.dingle@bigfoot.com.au>;
<henry.sinnreich@mci.com>; <paulej@packetizer.com>; <gv@trace.wisc.edu>
Sent: Friday, February 27, 2004 1:36 AM
Subject: RE: Betr.: [Sipping] ToIP Requirements


> Arnoud
>
> I do not believe this can work.  A TDD user dials 112 or 911.  Today, =
this
> will be routed directly to a "selective router" (a TDM switch) which =
sends
> the
> call to the correct Emergency Response Center.  You can't insert a
gateway,
> because you don't know that it's a TDD call until it is answered.  To =
make
> it work, you would have to establish a special PSTN number for emergency
> TDD,
> which would route to the gateway, which would then call the ERC.  I =
don't
> believe that is practical.
>
> Brian
>
>
>
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 9:17 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> >
> > Hello Brian,
> > That is also why we need to look at the emergency call system.
> > This text gateway is only at the PSTN side, thus another
> > requirement for a
> > T140/RTP gateway for the PSTN is that when a local emergency number is
> > dialled (112, or 911 etc). That the gateway must be able to
> > receive that
> > call (even dropping existing calls if necessary). Such a call
> > is easy to
> > recognize as an emergency call. (PSTN side).
> > From that gateway on, it is all IP with RFC2793 interactive
> > text. And then
> > it is just all existing SIP as if it was from an IP end-point.
> >
> > This gives eme

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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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Cc: hisham.khartabil@nokia.com, sipping@ietf.org, Rohan Mahy <rohan@cisco.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [Sipping] Conference policy URI
Date: Sat, 28 Feb 2004 23:50:14 -0800
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To summarize my position:

- I'm fine adding a new Call-Info purpose

- I don't think a registry of these purposes was or ever will be 
needed/"the right thing to do"

- To that end, I would be happy to log an errata against 3261 to remove 
the statement about such a registry

- Creating an unnecessary registry causes lots of people to lots of 
unnecessary work, and causes resentment toward the SIP and SIPPING 
working groups

thanks,
-rohan



On Feb 27, 2004, at 10:35 PM, Jonathan Rosenberg wrote:

> Call-Info purpose is not new. In fact, the idea of a registry for it 
> is not new either. RFC 3261 explicitly calls it out:
>
> 20.9 Call-Info
>
>    The Call-Info header field provides additional information about the
>    caller or callee, depending on whether it is found in a request or
>    response.  The purpose of the URI is described by the "purpose"
>    parameter.  The "icon" parameter designates an image suitable as an
>    iconic representation of the caller or callee.  The "info" parameter
>    describes the caller or callee in general, for example, through a 
> web
>    page.  The "card" parameter provides a business card, for example, 
> in
>    vCard [36] or LDIF [37] formats.  Additional tokens can be 
> registered
>    using IANA and the procedures in Section 27.
>
>
> The problem is that we forgot (i.e. - a bug) to include the definition 
> of this registry in RFC 3261. The draft is merely correcting an error 
> in RFC3261.
>
> As for the expense of creating registries - I dont think we should let 
> current administrative difficulties preclude us from doing what we 
> think the right thing is.
>
> -Jonathan R.
>
>
> Rohan Mahy wrote:
>> Hi,
>> I believe the SIPPING working group has come down with a bad case of  
>> Ianaus Registritus.  This rare, but preventable disease is caused by  
>> trying to register every possible field or parameter mentioned in a 
>> set  of standards track documents.   It is highly contagious to other 
>>  related working groups.  Treatment consists of deleting IANA  
>> Registration sections until the symptoms subside.  The patient should 
>>  begin feeling better immediately.
>> Seriously, I like the new Call-Info purpose, but I think the IANA  
>> Registry goes way too far overboard.  Remember that there is an  
>> administrative and political cost to setting up new IANA Registries.
>> thansk,
>> -rohan
>> On Feb 10, 2004, at 12:17 PM, Jonathan Rosenberg wrote:
>>> Thanks for doing this. Some comments.
>>>
>>> Firstly, on the registry itself. You need to give clearer guidance 
>>> to  IANA on the registry. I would suggest including a table that 
>>> shows how  it should look like. You need to tell IANA what 
>>> information needs to  be provided in future RFCs that register 
>>> additional values (value, RFC  number, registrant, etc.). Please 
>>> take a look at rfc2434  
>>> (ftp://ftp.rfc-editor.org/in-notes/rfc2434.txt).
>>>
>>> Secondly, on the "policy-uri" value - you need to specify, I think,  
>>> what the URI type is, and what the content is at that URI (at least 
>>> a  mandatory to implement one) in order to get interoperability. 
>>> Unlike  the other stuff, which is just rendered for user 
>>> information, this  resource is processed by an automata, and I think 
>>> you need to say more  about it.
>>>
>>> Thanks,
>>> Jonathan R.
>>>
>>> hisham.khartabil@nokia.com wrote:
>>>
>>>> Thanks Jonathan. A draft has been submitted to the SIP WG.
>>>> http://www.ietf.org/internet-drafts/draft-khartabil-sip-policy-uri- 
>>>> call-info-purpose-00.txt
>>>> Regards,
>>>> Hisham
>>>>
>>>>> -----Original Message-----
>>>>> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>>>> Sent: 24.January.2004 04:44
>>>>> To: Khartabil Hisham (Nokia-TP/Helsinki)
>>>>> Cc: sipping@ietf.org
>>>>> Subject: Re: [Sipping] Conference policy URI
>>>>>
>>>>>
>>>>> I think this is a reasonable usage of Call-Info. However, it can  
>>>>> only be done if a "purpose" tag value is registered with IANA, as  
>>>>> indicated by section 20.9. However, it ocurred to me, when reading 
>>>>>  this, that we never set up an IANA registry for call-info purpose 
>>>>>  tokens! Oops! Thats easily rectifiable; when the draft defining 
>>>>> the  'policy-uri' (or whatever its called) purpose tag is written, 
>>>>> it  should also create the registry and register the other tokens  
>>>>> defined in rfc3261.
>>>>>
>>>>> -Jonathan R.
>>>>>
>>>>> hisham.khartabil@nokia.com wrote:
>>>>>
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> There was consensus on the XCON mailing list that the conference
>>>>>> state event package needs to include in the notification the
>>>>>> conference policy URI.
>>>>>>
>>>>>> There was also consensus that some participants may not want to
>>>>>> subscribe to the conference package but would still like to learn
>>>>>> the conference policy URI. In this case, it was suggested that the
>>>>>> Call-info header carries such information. Perhaps the
>>>>>> cc-conferencing draft can include examples of that.
>>>>>>
>>>>>> Thanks, Hisham
>>>>>>
>>>>>> _______________________________________________ Sipping mailing
>>>>>> list  https://www1.ietf.org/mailman/listinfo/sipping This list is
>>>>>> for NEW development of the application of SIP Use
>>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>>>> sip@ietf.org for new developments of core SIP
>>>>>>
>>>>>
>>>>> -- 
>>>>> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>>>> Chief Technology Officer                    Parsippany, NJ 
>>>>> 07054-2711
>>>>> dynamicsoft
>>>>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>>>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>>>>> http://www.dynamicsoft.com
>>>>>
>>>>>
>>>
>>> -- 
>>> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>> Chief Technology Officer                    Parsippany, NJ 07054-2711
>>> dynamicsoft
>>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>>> http://www.dynamicsoft.com
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>


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



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Hi,

The benefit of having the same URI would be also that you could just =
SUBSCRIBE to the conference state without even needing to join it via =
INVITE. I think there could be some application use cases why you would =
want to do this: for instance a user wants to join a conference only =
when certain other user or certain amount of users are present. To do =
this, the user should only need to know the single conference URI.

(Supposedly you could find out the URI to subscribe to with OPTIONS too, =
but why bother, if it could be the same.)

Markus

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Jonathan Rosenberg
> Sent: 25 February, 2004 21:48
> To: Gonzalo Camarillo
> Cc: Garcia Miguel.An (Nokia-NRC/Helsinki); sipping@ietf.org;
> alan.johnston@mci.com; oritl@microsoft.com
> Subject: Re: [Sipping] Re: Conferencing for UAs
>=20
>=20
>=20
> I think the same URI makes sense. At the core, there really is one=20
> resource here - the conference. All that differs is the=20
> methods that are=20
> invoked on it and how that resource behaves when users contact it.=20
> Subscribing to it and initiating a session to it are two different=20
> methods, but both against fundamentally the same resource.
>=20
> -Jonathan R.
>=20
> Gonzalo Camarillo wrote:
>=20
> > Hi,
> >=20
> >> Aren't we heading to defining a generic Associated-State=20
> header with
> >> a "role" parameter that contains a value "transaction",=20
> "conference",
> >> "xxx", etc...
> >=20
> >=20
> > I actually do not care how we encode this information; in a=20
> header with=20
> > several parameters or in multiple headers (in any case, you=20
> may be right=20
> > that using a single header could be cleaner).
> >=20
> > What I want to know is whether or not people think that it=20
> is worthwhile=20
> > keeping separate the URI used for the INVITE dialog and the one (or=20
> > ones) used for subscriptions to some piece of state that is=20
> associated=20
> > somehow with that dialog. My personal opinion is that we=20
> should keep=20
> > them separate.
> >=20
> > Gonzalo
> >=20
> >=20
> > This communication is confidential and intended solely for the=20
> > addressee(s). Any unauthorized review, use, disclosure or=20
> distribution=20
> > is prohibited. If you believe this message has been sent to you in=20
> > error, please notify the sender by replying to this=20
> transmission and=20
> > delete the message without disclosing it. Thank you.
> >=20
> > E-mail including attachments is susceptible to data corruption,=20
> > interruption, unauthorized amendment, tampering and=20
> viruses, and we only=20
> > send and receive e-mails on the basis that we are not=20
> liable for any=20
> > such corruption, interception, amendment, tampering or=20
> viruses or any=20
> > consequences thereof.
> >=20
> >=20
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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Hi,

I also hope there could be more clarity on the relationship between =
exploders and conferencing work. Some thoughts below.

One clear use case for exploders is the establishment of a "dial-out =
ad-hoc" conference. One of Gonzalo's drafts proposes that the exploder =
B2BUA (which in this case becomes a conference focus) could return to =
the originating UA a reference (URI) to the list, after which the UA =
could use e.g. XCAP to manage the list out-of-band. The idea seems to be =
that the default format for this list indeed would be the XCAP resource =
list.

In conferencing framework this list clearly overlaps with the dial-out =
list concept to be defined in CPCP. Also, if the semantics of removing a =
user from the list will result in ejection of this user from the =
conference, this list may overlap with some other functionality of CPCP =
too.

My proposal to make the exploders and CPCP compliant with each other =
would be:
- When the conference is established by invoking an exploder, the =
conference server will use the exploder-list to populate a CPCP document =
in such a way that its semantics match with what was done with the =
exploder.
- URI for this CPCP document is provided in 200 OK using the mechanisms =
dicussed elsewhere (Call-info purpose, new header, ...).
- Conference participants can thereafter make conference policy =
operations by doing changes to this document, and the situation is as if =
the conference was originally created by CPCP in the first place. =20
- If ad hoc list management via re-INVITEs is still allowed, there needs =
to be a clear mapping how adding/removing list entries interacts with =
the CPCP document.

To enable this the CPCP semantics still need more clarity, and maybe =
also adopting the XCAP resource list format for the possible dial-out =
etc. lists in that document would help this mapping.

(We could still keep the simple non-CPCP controllable conference case, =
where the only manipulated object would be a simple resource list. In =
this case access lists and other such features woould not be supported.) =


Do people think that this would be feasible? Or would we be getting to =
an interaction that is too complex to model? In general I think there's =
room for both exploders AND CPCP, so it would be important to specify =
how they interwork.

Regards,
	Markus

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Alan Johnston
> Sent: 26 February, 2004 21:05
> To: Gonzalo Camarillo
> Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org;
> oritl@microsoft.com; Garcia Miguel.An (Nokia-NRC/Helsinki)
> Subject: Re: [Sipping] Conferencing for UAs
>=20
>=20
>=20
> At 02:13 PM 2/26/2004 +0200, Gonzalo Camarillo wrote:
> >Alan,
> >
> >I agree with you that we do not want to change the=20
> conferencing framework=20
> >at this point in time (I already said in my previous mail that I=20
> >understood that my comments related to this came too late).=20
> Still, I found=20
> >the discussion useful to understand your point of view.
>=20
> Gonzalo,
>=20
> Glad to hear that.  I'd be happy to think about how to extend the=20
> conferencing work to cover the exploder scenarios you are=20
> proposing with=20
> MESSAGE, INVITE, and even REFER.    With the current drafts,=20
> I have a hard=20
> time understanding how to do this, but I look forward to=20
> clarifications.   The SUBSCRIBE exploder does not seem to have any=20
> conferencing semantics, but seems to be a SIMPLE application=20
> since it is=20
> tightly coupled to the presence application.
>=20
> Thanks,
> Alan Johnston
> MCI
> sip:alan@sipstation.com
>=20
>=20
> >In any event, the topic I am interested in is the creation=20
> of different=20
> >resources (identified by URIs) by a single request. In=20
> particular, how to=20
> >communicate those URIs to the UAC. (I sent separate comments=20
> on Hisham's=20
> >draft.)
> >
> >I hope we can discuss these issues in Korea.
> >
> >Thanks,
> >
> >Gonzalo
> >
> >
> >Alan Johnston wrote:
> >
> >>At 09:47 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
> >>
> >>>Hi Alan,
> >>>
> >>>I understand that my comments come late in the game and=20
> that it might be=20
> >>>impossible to change the conferencing framework at this=20
> point. In any=20
> >>>event, I do not agree with your statement below:
> >>>
> >>>>No - this is the problem.  You are looking at your single narrow=20
> >>>>application.
> >>>
> >>>
> >>>I believe I am actually looking at a much wider range of=20
> applications=20
> >>>than just conferencing. So, to me, the problem is that the=20
> conferencing=20
> >>>framework is too focused on a "single narrow application";=20
> conferencing.
> >>
> >>Gonzalo,
> >>If you are saying that some of your exploder services do not have a=20
> >>conferencing component, then I agree.  Some of the exploder=20
> services seem=20
> >>to just be a "request to fork" to a set of URIs.  Others=20
> definitely have=20
> >>conferencing semantics.  The ones with conferencing=20
> semantics, I believe,=20
> >>can be, and should be modeled using the conferencing framework.
> >>The problem is that each method seems to have a different set of=20
> >>semantics as to what exploding that method means - INVITE=20
> is handled=20
> >>differently from SUBSCRIBE, from MESSAGE.   From a high=20
> level, exploders=20
> >>may seem to be a consistent set of services.  However, as I=20
> delve into=20
> >>the details and think about call flows (something  absent=20
> from any of the=20
> >>exploder drafts), a different story emerges.
> >>I think we need to explore exactly the services and call=20
> flows that are=20
> >>being envisioned for exploders before proposing major=20
> changes to our=20
> >>conferencing framework, which I believe is well understood=20
> and documented.
> >>
> >>>In a conferencing service, there is a single resource=20
> created by the=20
> >>>initial INVITE; a conference. That's why the server can=20
> use the Contact=20
> >>>header field to send it back to the UAC. If you think of=20
> services that=20
> >>>create more than one resource (e.g., an explosion and a=20
> conference), the=20
> >>>server may not want to use the same URI for both. Moreover,=20
> >>>conceptually, is is not a good thing to use the same URI=20
> to identify=20
> >>>both resources.
> >>
> >>As Paul has pointed out - they are the same resource. =20
> Additionally, it=20
> >>is a trivial SIP routing issue to fix this - many proxies=20
> do method-based=20
> >>request routing - that would completely solve your problem=20
> of sending=20
> >>SUBSCRIBEs to a different UA than the initial request.
> >>
> >>>So, my main concern is the semantics overload of the=20
> Contact header. (We=20
> >>>abused this header when design REGISTER as well, by the=20
> way. That's why=20
> >>>I do not want to do it again.)
> >>>
> >>>In my opinion, the Contact header is the URI where you=20
> send subsequent=20
> >>>requests within the dialog. If the server wants to tell=20
> the client that=20
> >>>a resource (e.g., a conference) has been created, it can=20
> use a different=20
> >>>header to include the URI of such a resource. Note that,=20
> if you want=20
> >>>both URIs to be the same (like in the conferencing case),=20
> you can do it.
> >>>
> >>>Contact: sip:1234@a.com;isfocus
> >>>Resource: sip:1234@a.com;isfocus
> >>>
> >>>An entity receiving these headers would use the URI in the Contact=20
> >>>header to send subsequent requests *within* the dialog,=20
> and the URI in=20
> >>>the other header to send requests *outside* that dialog=20
> (e.g., SUBSCRIBE=20
> >>>to the conference package).
> >>
> >>No.  A GRUU should be used in the Contact, and its routing is valid=20
> >>outside the dialog.
> >>
> >>>In brief, instead of sticking the conference URI in a=20
> Contact header and=20
> >>>label that URI with an isfocus parameter so that the UAC=20
> figures what is=20
> >>>going on, I find it more explicit to use a different=20
> header. Although,=20
> >>>as I said, I understand that this comment comes late and=20
> that may not be=20
> >>>useful at this point.
> >>
> >>I think much more work is needed before the working group=20
> can evaluate=20
> >>the exploders work and see how and if it fits in with=20
> existing concepts=20
> >>and approaches.
> >>Thanks,
> >>Alan Johnston
> >>MCI
> >>sip:alan@sipstation.com
> >>
> >>>Thanks,
> >>>
> >>>Gonzalo
> >>>
> >>>
> >>>This communication is confidential and intended solely for the=20
> >>>addressee(s). Any unauthorized review, use, disclosure or=20
> distribution=20
> >>>is prohibited. If you believe this message has been sent to you in=20
> >>>error, please notify the sender by replying to this=20
> transmission and=20
> >>>delete the message without disclosing it. Thank you.
> >>>
> >>>E-mail including attachments is susceptible to data corruption,=20
> >>>interruption, unauthorized amendment, tampering and=20
> viruses, and we only=20
> >>>send and receive e-mails on the basis that we are not=20
> liable for any=20
> >>>such corruption, interception, amendment, tampering or=20
> viruses or any=20
> >>>consequences thereof.
> >>>
> >>>
> >>>_______________________________________________
> >>>Sipping mailing list =20
> https://www1.ietf.org/mailman/listinfo/sipping
> >>>This list is for NEW development of the application of SIP
> >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>Use sip@ietf.org for new developments of core SIP
> >
> >--
> >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> >Telecom R&D               Fax   :  +358  9 299 30 52
> >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> >Finland                   http://www.hut.fi/~gonzalo
> >
> >
> >This communication is confidential and intended solely for the=20
> >addressee(s). Any unauthorized review, use, disclosure or=20
> distribution is=20
> >prohibited. If you believe this message has been sent to you=20
> in error,=20
> >please notify the sender by replying to this transmission=20
> and delete the=20
> >message without disclosing it. Thank you.
> >
> >E-mail including attachments is susceptible to data corruption,=20
> >interruption, unauthorized amendment, tampering and viruses,=20
> and we only=20
> >send and receive e-mails on the basis that we are not liable=20
> for any such=20
> >corruption, interception, amendment, tampering or viruses or any=20
> >consequences thereof.
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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From: "Eric Burger" <eburger@snowshore.com>
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        "Jonathan Rosenberg (E-mail)" <jdrosen@dynamicsoft.com>
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On sipping-spec-policy-00 -

Do we really need end-to-end, two-party acceptance of proxy policies?

Motivation: to the UAC, there is no difference between a proxy saying =
"you can only do G.711" and the UAS returning an answer saying "please =
only use G.711".

Conversely, to the UAS, there is no difference between a proxy saying =
"please only do G.711" and the UAC offering only G.711.

I *can* see letting the UAC know what is going on from a human factors =
perspective.  Let's say I bought the latest foobratz phone that does =
quadraphonic-better-than-CD-quality audio.  However, I can never use it, =
because everyone comes back answering with G.729ab.  It might be nice to =
know that it is because a proxy limited me, not that my fancy phone =
doesn't work.  As the owner of the UAC, I have expectations of the fancy =
stuff working.  That said, if my fancy phone is in the UAS role, I have =
expectations that other people *don't* have my fancy features, =
especially if my INVITES only have G.711 in them.


In another issue, consider section 3.1.2.  The document states that it =
is not recommended for a proxy to reject a request that doesn't honor =
the MFO.  It may be better to state the obvious alternative: If the =
proxy really has to enforce the policy, it flips into B2BUA mode.  By =
definition, end-to-end is broken (from the carrier's perspective), so =
the carrier will jump in and "fix it".  The idea being that if everyone =
plays by the rules (or suggestions), one won't have to look for another =
service provider that doesn't feel like billing :-)


I would also suggest stating that proxies MUST NOT mix its MFO's on a =
media-filter header populated with MFO's from another proxy.  That is, a =
proxy can add media-filter headers, but it cannot modify existing ones.  =
This is supported already in the syntax.  However, I would make this a =
MUST to make it easier to sign headers and ensure they don't break =
header signing.


Lastly, is there any reason to use different formats for =
session-specific versus session-independent policy (headers v. XML)?  =
XML is more expressive -- if we wanted to keep XML, how about using =
cid:'s for the session-specific MFO statements?


_______________________________________________
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Eric Burger" <eburger@snowshore.com>, <toip@snowshore.com>,
        "Sipping \(E-mail\)" <sipping@ietf.org>
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Subject: [Sipping] RE: manyfolks-sipping-ToIP-01
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Eric, the specified transmission method is designed for both satisfying t=
he
users and not straining networks:

In order to not strain networks, the transport level RFC2793 says that
characters should be buffered together in packets for shipping. The
recommendation for shipping packets with text is to use 300 ms intervals
between transmissions.

The revision of RFC2793, in draft-ietf-avt-rfc2793bis-02 is stronger and
more clear about this RECOMMENDATION.

With the 300 ms interval between packet transmissions, we have a text flo=
w
that is experienced to be close enough to character-by-character, and a
maximum network load of 3 packets per second containing around 3 kbit/s,
including redundancy. That is sustainable for most network connections.

Are there really typists that can type 60 characters per second????
I would expect that the maximum load would come from speech-to-text
arrangements, where I would guess that 20 characters per second would be =
a
practical maximum. Therefore, 30 characters/s throughput in transmission
sounded sufficient for me.

With the buffering described above, this figure is not critical, so if yo=
u
have reasons to think that there is a real requirement for 60 cps, we sho=
uld
document it.

Subtitling a radio report from a football match by using speech-to-text
might be among the more challenging applications..


Gunnar
-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: owner-toip@snowshore.com [mailto:owner-toip@snowshore.com]On
Behalf Of Eric Burger
Sent: Sunday, February 29, 2004 8:50 PM
To: toip@snowshore.com; Sipping (E-mail)
Subject: manyfolks-sipping-ToIP-01


I like the "network must support a minimum of 30cps" for text (section 6,
requirement (d)).  Then I thought about it: that's a ptime of 33ms.  That=
 is
OK.  However, if we said minimum of 30cps, but really like 60cps
(professional typist), that is a ptime of 16ms, which would start to stra=
in
many networks.

Why do we say that the file format for storage should be XHTML?  Are we
archiving marked-up text?  Why not just UTF-8?


-
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Snowshore is not to be held responsible for any actions or comments found
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From: Rohan Mahy <rohan@cisco.com>
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Subject: [Sipping] Re: State Agents in sipping-dialog-package-04
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Hi Eric,

The document already allows for state agents.  Is there something specific
you would want to put in the document in addition.

thanks,
-rohan


On Sun, 29 Feb 2004, Eric Burger wrote:

> I was thinking: would it not be useful to recommend a state agent for hiding devices behind a forking proxy?  Isn't the subscriber interested in the user / user+device state?
>
> Subscribing to the dialog package will let you see the topology of all of my endpoints behind a forking proxy.  You can figure out that I've got a home phone, a work phone, a soft phone, a phone at the ski slope, and a silly phone.  I might not want you to know that.  However, I may want you to use the cool features dialog package enables.
>
> If I have a stateless forking proxy, then yes, simply calling me will get you all of the 1xx's and 4xx's showing all of my devices.  However, if I have a stateful proxy that collects and sorts provisional responses, only sending back a real contact, then you don't get all of my device info.  Thus having a state agent would mirror the level of security a simple INVITE has.
>
>

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Eric Burger wrote:

> In the overview (2.3), the document states that the content delivery server MUST be in the public network or DMZ.

The intention of that sentence/paragraph was that it must be possible to have
profile delivery server anywhere on the Internet and that it must be possible for
UAs to access that server using this framework (e.g. traverse NAT/firewalls, etc.).

Thanks, I will fix the wording.

>
>
> I would offer that for a carrier, they will be loathe to put subscriber data on the open Internet.  Moreover, while it may be acceptable to have a MAC address float about in a "secure" network, service providers may be less sanguine in a sniffable, public network.

The MAC address was merely a suggestion.  The point was to have a unique
identifier for the device.

>
>
> How about some language about the perils of putting it in the public network, as well as the reasons for doing it.  For example, to support roaming - I need my profile when I'm away without having to establish a VPN first.

Based upon how the dicussion goes today, it is likely that I will be
beefing up the dicussion on hoteling, roaming and the concepts
of the local network, device and user profiles and subscriptions.

>
>
> I would also be more specific on the access protocol for retrieving the profile.  It is good to say, "any URI scheme", so long as one says something like "but MUST implement https at a minimum."  That way, there is a minimum baseline that all devices and content delivery servers will be able to interoperate with.

The URL scheme is to be defined by the profile data definintion which
is out of scope for this document.

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


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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At 05:34 PM 2/27/2004 -0600, Adam Roach wrote:
>The document looks good to me at a skim,

thanks

>but I believe
>there are a couple of points of clarification that should
>be added to 4.2.1.
>
>First, I believe the label for the right-most bar in the
>diagram should be "UA2" instead of "UA3," if my understanding
>of the use case is correct.

Actually, you point out that I didn't have Figure 3 correct (section 3.0), 
which I copied to Figure 5 (section 4.2.1). Thanks for the catch!


>Secondly, it is not clear how the "Resource-Priority: 2" in
>the UA3-UA4 call comes into play. None of the UAs shown in
>this flow are privy to any of the SIP signaling that was used
>to establish the UA3-UA4 call. What would seem to matter is
>the RSVP priorities used to establish the call. Correct?

correct

>This should probably be clarified in the text.

I was too simple in my explanation. I will add the text explaining that the 
SIP R-P values translate to RSVP priorities accordingly.

Thanks again.


>/a
>
> > -----Original Message-----
> > From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> > Sent: Sunday, February 08, 2004 15:29
> > To: sipping
> > Cc: Dean Willis; Rohan Mahy; James M. Polk
> > Subject: [Sipping] WGLC for
> > draft-ietf-sipping-reason-header-for-preemption-00.txt
> >
> >
> > Folks,
> >
> > we would like to announce the start of the WGLC for the
> > following draft.
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sipping-reason-
>header-for-preemption-00.txt
>
>This WGLC will end on Feb 29th.
>
>Send your comments to this mailing list and to the author.
>
>Thanks,
>
>Gonzalo
>SIPPING co-chair


cheers,
James

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


_______________________________________________
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Thread-Topic: State Agents in sipping-dialog-package-04
Thread-Index: AcP/DyAig4nr+7q4R9iREOqxoxn4iwADd4ow
From: "Eric Burger" <eburger@snowshore.com>
To: "Rohan Mahy" <rohan@cisco.com>
Cc: "Jonathan Rosenberg (E-mail)" <jdrosen@dynamicsoft.com>,
        <schulzrinne@cs.columbia.edu>, "Sipping (E-mail)" <sipping@ietf.org>
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Subject: [Sipping] RE: State Agents in sipping-dialog-package-04
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The State Agents section seems to me to read that State Agents are a bad =
idea ("Dialog state is ideally maintained in the user agents in which =
the dialog resides.").  I guess a softening of the language would be =
clearer.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Sunday, February 29, 2004 4:58 PM
> To: Eric Burger
> Cc: Jonathan Rosenberg (E-mail); schulzrinne@cs.columbia.edu; Sipping
> (E-mail)
> Subject: Re: State Agents in sipping-dialog-package-04
>=20
>=20
> Hi Eric,
>=20
> The document already allows for state agents.  Is there=20
> something specific
> you would want to put in the document in addition.
>=20
> thanks,
> -rohan
>=20
>=20
> On Sun, 29 Feb 2004, Eric Burger wrote:
>=20
> > I was thinking: would it not be useful to recommend a state=20
> agent for hiding devices behind a forking proxy?  Isn't the=20
> subscriber interested in the user / user+device state?
> >
> > Subscribing to the dialog package will let you see the=20
> topology of all of my endpoints behind a forking proxy.  You=20
> can figure out that I've got a home phone, a work phone, a=20
> soft phone, a phone at the ski slope, and a silly phone.  I=20
> might not want you to know that.  However, I may want you to=20
> use the cool features dialog package enables.
> >
> > If I have a stateless forking proxy, then yes, simply=20
> calling me will get you all of the 1xx's and 4xx's showing=20
> all of my devices.  However, if I have a stateful proxy that=20
> collects and sorts provisional responses, only sending back a=20
> real contact, then you don't get all of my device info.  Thus=20
> having a state agent would mirror the level of security a=20
> simple INVITE has.
> >
> >
>=20
>=20


_______________________________________________
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>,
        Arnoud van Wijk
	 <a.vwijk@viataal.nl>, Gonzalo.Camarillo@ericsson.com,
        sipping@ietf.org
Cc: rrroy@att.com, barry.dingle@bigfoot.com.au, henry.sinnreich@mci.com,
        paulej@packetizer.com, gv@trace.wisc.edu
Subject: RE: Betr.: [Sipping] ToIP Requirements
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How does V.18 work?  Is the "calling identifier" something in
(ISDN presumably) signalling?  Or is it a distinctive tone
pattern?

Brian

> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> Sent: Friday, February 27, 2004 1:45 AM
> To: Arnoud van Wijk; 'Rosen, Brian'; Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> The INCOM group of the European commission made 
> recommendations that the
> same emergency number shall be used for all kinds of terminals.
> It is dangerous to have different prefixes or other tricks in 
> different
> countries. How shall you keep track of that if you are a visitor?
> 
> From PSTN, it is only the old textphones that cause problems. 
> A calling V.18
> sends a very distinct calling identifier that can be catched 
> by a gateway or
> an emergency centre and be properly handled. On the SIP side, 
> the options
> are even better.
> 
> I do not mean that all problems are solved by this statement, 
> but I do not
> think that the prefix is the only solution, and I do not 
> think we shall
> recommend it in connection to emergency.
> 
> 
> Gunnar
> 
> -------------------------------------------
> Gunnar Hellstrom
> Omnitor AB
> Renathvagen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
> 
> 
> -----Original Message-----
> From: sipping-admin@ietf.org 
> [mailto:sipping-admin@ietf.org]On Behalf Of
> Arnoud van Wijk
> Sent: Thursday, February 26, 2004 4:57 PM
> To: 'Rosen, Brian'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> Hello Brian,
> 
> Yes you are right about that the PSTN cannot detect if it is 
> a voice call or
> TDD/Textphone call. They are both audio.
> The PSTN would if it can detect it is a textphone call, route 
> it to the text
> gateway. And transcode the call into T.140 over RTP as 
> decribed in RFC 2793.
> This is for PSTNs which use IP to interconnect with other 
> PSTNs or with any
> SIP telephones. So it is for analoge telephony into SIP 
> telephony here.
> 
> In the ToIP draft:
> http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> ip-01.txt
> A possible solution is a prefix for textphone users before 
> dialling the
> selected number.
> It indeed would mean that you dial e.g. 600-112 instead 112 directly.
> Is that bad?
> I believe it is not.
> 
> Most textphones have the emergency number pre programmed. And 
> when textphone
> users have to use a pre-fix for any textphone call, then the 
> same prefix
> before 112 or 911 will be fine.
> In the Netherlands, we have for emergency calls 800-112 
> instead of 112. It
> works fine, only almost no deaf users call 112, since they 
> are then mostly
> in a situation where they do not have a textphone nearby!
> 
> Maybe there are other solutions that allow the PSTN to detect that the
> incoming call is a textphone user and acts on it.
> 
> For example it is possible to create a database where the 
> PSTN can check if
> the incoming number is registered as a textphone.
> 
> But using a text gateway to connect PSTN textphones/TDDs to 
> ToIP is still
> the best solution, even the only solution in my opinion.
> There should be no attempts to transport the text telephone 
> tones (e.g.
> Baudot) over the IP voice channel! I believe that will open 
> Pandora's box.
> 
> I am not a PSTN expert, and we can surely look into this 
> matter to find a
> solution. Let it be prefix or any other solution that lets the PSTN
> immediately route the call to the text gateway.
> 
> What do you PSTN people think? How would you tackle this 
> problem? Your input
> will be very welcome.
> 
> Greetz
> 
> Arnoud
> 
> Drs. Arnoud A. T. van Wijk
> Viataal
> Research & Development
> Afdeling RDS
> Theerestraat 42
> 5271 GD Sint-Michielsgestel
> The Netherlands.
> Mobile: +31651921948
> International text telephone: +31735588408
> 
> 
> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> Sent: donderdag 26 februari 2004 15:36
> To: 'Arnoud van Wijk'; 'Gunnar Hellstrom'; 
> Gonzalo.Camarillo@ericsson.com;
> sipping@ietf.org
> Cc: rrroy@att.com; barry.dingle@bigfoot.com.au; 
> henry.sinnreich@mci.com;
> paulej@packetizer.com; gv@trace.wisc.edu
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> Arnoud
> 
> I do not believe this can work.  A TDD user dials 112 or 911. 
>  Today, this
> will be routed directly to a "selective router" (a TDM 
> switch) which sends
> the
> call to the correct Emergency Response Center.  You can't 
> insert a gateway,
> because you don't know that it's a TDD call until it is 
> answered.  To make
> it work, you would have to establish a special PSTN number 
> for emergency
> TDD,
> which would route to the gateway, which would then call the 
> ERC.  I don't
> believe that is practical.
> 
> Brian
> 
> 
> 
> > -----Original Message-----
> > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > Sent: Thursday, February 26, 2004 9:17 AM
> > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> >
> > Hello Brian,
> > That is also why we need to look at the emergency call system.
> > This text gateway is only at the PSTN side, thus another
> > requirement for a
> > T140/RTP gateway for the PSTN is that when a local 
> emergency number is
> > dialled (112, or 911 etc). That the gateway must be able to
> > receive that
> > call (even dropping existing calls if necessary). Such a call
> > is easy to
> > recognize as an emergency call. (PSTN side).
> > From that gateway on, it is all IP with RFC2793 interactive
> > text. And then
> > it is just all existing SIP as if it was from an IP end-point.
> >
> > This gives emergency centers an advantage, all calls will be
> > T.140/RTP, so
> > integration in their computer systems will be easy instead
> > being forced to
> > have analog textphones available for a textphone call. (or a
> > separate text
> > telephone emergency center).
> >
> > /Arnoud
> >
> > Drs. Arnoud A. T. van Wijk
> > Viataal
> > Research & Development
> > Afdeling RDS
> > Theerestraat 42
> > 5271 GD Sint-Michielsgestel
> > The Netherlands.
> > Mobile: +31651921948
> > International text telephone: +31735588408
> >
> >
> > -----Original Message-----
> > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > Sent: donderdag 26 februari 2004 15:04
> > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com;
> > sipping@ietf.org
> > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > henry.sinnreich@mci.com;
> > paulej@packetizer.com; gv@trace.wisc.edu
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> > Arnoud
> >
> > Generally, you don't want to rely on very many SIP elements
> > to complete
> > an emergency call.  Present systems accept TDD at all answering
> > points because they don't know before the calls is answered that it
> > is a TDD call.  I certainly don't want the user to have to 
> do anything
> > other than dial his local emergency number, or complete a device
> > specific key sequence to complete an emergency call, and I don't
> > want a gateway busy to prevent an emergency call.
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > Sent: Thursday, February 26, 2004 6:07 AM
> > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > Actually,
> > > When using the text gateway, there is no need to detect the
> > > TDD/Analogue
> > > text telephone tones/signals. That is the job of the text gateway.
> > >
> > > And for the emergency calling. We have to look closely at this in
> > > combination with a text gateway. Possible behaviour/action.
> > > And possible
> > > action in SIP.
> > >
> > > Right now I cannot say what has to happen, I must study the
> > > emergency calls
> > > in depth. But as I see it, using the text gateway, all SIP
> > > originating and
> > > receiving emergency calls will be possible for any textphone user.
> > >
> > > Cheers
> > >
> > > Arnoud
> > >
> > >
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > On Behalf Of
> > > Rosen, Brian
> > > Sent: woensdag 25 februari 2004 23:08
> > > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > > I'm particularly interested in the last one, emergency calling.
> > >
> > > Given that you propose an appropriate SDP indicating
> > support for text,
> > > do you think there is anything else we need to do to in SIP
> > > to make this
> > > happen?  I don't see anything.
> > >
> > > I know that the problem of using a legacy TDD device on 
> any kind of
> > > adapter is as problematic as current TDD; the call will answered
> > > as if it was a voice call, the baudot/whatever tones will be
> > > detected,
> > > and the operator will look for text.
> > > Can't think of any way to make that better.
> > >
> > > Do note that language preference headers should be set!!!
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > I think PSTN textphone gateway invocation is one area where
> > > > we either need
> > > > protocol extensions, or need agreements on what current
> > > > mechanisms to use.
> > > >
> > > > Another is invocation of various kinds of relay services or
> > > > translation
> > > > services.
> > > >
> > > > A third is calling emergency services and reach positions
> > capable of
> > > > handling the type of communication that the user prefers.
> > > >
> > > >
> > > >
> > > > Gunnar
> > > > -------------------------------------------
> > > > Gunnar Hellstrom
> > > > Omnitor AB
> > > > Renathvagen 2
> > > > SE 121 37 Johanneshov
> > > > SWEDEN
> > > > +46 8 556 002 03
> > > > Mob: +46 708 204 288
> > > > www.omnitor.se
> > > > Gunnar.Hellstrom@Omnitor.se
> > > > --------------------------------------------
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org
> > > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > > A vWijk
> > > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > > gv@trace.wisc.edu
> > > > Subject: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > Hello Gonzalo,
> > > > While there are device requirements in the text. Those
> > > > requirements and the
> > > > explanation on the different analogue text telephone systems
> > > > is included to
> > > > give the reader an understanding what kind of devices will
> > > > connect with
> > > > ToIP. And the need for the text gateways. And what the
> > > > different behaviour
> > > > can/may be compared to end to end IP devices.
> > > > The most important part will be the ToIP framwork and the
> > > > text gateways and
> > > > how such gateways will be used and work with the ToIP framework.
> > > >
> > > > There is a ToIP framework where a terminal uses streaming
> > > > text as defined in
> > > > RFC2793 to connect with any other Interactive
> > > > texting/real-time streaming
> > > > text device.
> > > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > > established with
> > > > SIP. But in other cases, a text gateway must be used to allow
> > > > T.140/RTP to
> > > > connect with a non IP text telephony device.
> > > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > > text gateway to
> > > > be able to understand RFC2793 text. (PSTN-IP)
> > > > It can be a mobile phone running GPRS that uses the text
> > gateway to
> > > > understand RFC2793 text. (this exist right NOW using SIP! And
> > > > those gateways
> > > > will be used in mobile networks this year!)
> > > > Or a mobile text telephone using CTM via the text gateway to
> > > > communicate
> > > > with RFC2793 text.
> > > > Or even one analogue text telephone connecting to another
> > > > analogue text
> > > > telephone in a different country, and the PSTN's use IP to
> > > > connect the call.
> > > >
> > > > The question is for example, how do we use that gateway from
> > > > the IP side? Is
> > > > it invisible for the IP side? Or do we need special
> > > > extensions in SIP to
> > > > that the IP terminal knows it is communicating with the
> > > > end-device through
> > > > the text gateway?
> > > > There may be a slightly different call setup flow when a text
> > > > gateway is
> > > > included.
> > > > Or additional SIP messages indicating gateway activity. When
> > > > you connect for
> > > > example to an analogue text telephone with an RFC2793 capable
> > > > IP terminal
> > > > the capabilities will be less then IP to IP terminal directly
> > > > (different
> > > > transmission speeds, half/full duplex etc).
> > > > Or what to do with emergency calls that are in RFC 2793 
> text only?
> > > >
> > > > That are requirements that will be defined and worked out in
> > > > this draft.
> > > > Also, we want to make everybody in SIP and SIPPING aware
> > > > about those text
> > > > gateways. So that complicated matters like how to send
> > analogue text
> > > > telephone text through a PSTN and connect with a SIP
> > > > telephone using RFC2793
> > > > interactive texting. When using the text gateway, there is
> > > > then no need for
> > > > RFC 2833 events at the PSTN for text telephones.
> > > >
> > > > So, yes. I believe that this should be in the SIPPING WG. And
> > > > it will fit in
> > > > the charter - support for hearing-/speech-impaired calling.
> > > >
> > > > greetz
> > > >
> > > > Arnoud
> > > >
> > > >
> > > >
> > > > Drs. Arnoud A. T. van Wijk
> > > > Viataal
> > > > Research & Development
> > > > Afdeling RDS
> > > > Theerestraat 42
> > > > 5271 GD Sint-Michielsgestel
> > > > The Netherlands.
> > > > Mobile: +31651921948
> > > > International text telephone: +31735588408
> > > >
> > > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > > 24-02-04 10:21 >>>
> > > > Hello,
> > > >
> > > > the draft below provides a set of requirements for real-time
> > > > conversations using SIP.
> > > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > > ip-01.txt
> > >
> > > Nevertheless, I have found a lot of device requirements in
> > > the text, but
> > > I fail to understand which particular protocol requirements
> > we need to
> > > fulfill with new protocol extensions.
> > >
> > > While SIPPING is the right place to handle requirements
> > that will lead
> > > to new SIP protocol extensions, SIPPING may not be the
> > right place to
> > > deal with device requirements (e.g., text phones MUST have a
> > > display of
> > > at least 100 characters).
> > >
> > > Thanks,
> > >
> > > Gonzalo
> > >
> > >
> > > This communication is confidential and intended solely for
> > > the addressee(s).
> > > Any unauthorized review, use, disclosure or distribution is
> > > prohibited. If
> > > you believe this message has been sent to you in error,
> > > please notify the
> > > sender by replying to this transmission and delete the
> > message without
> > > disclosing it. Thank you.
> > >
> > > E-mail including attachments is susceptible to data corruption,
> > > interruption, unauthorized amendment, tampering and viruses,
> > > and we only
> > > send and receive e-mails on the basis that we are not liable
> > > for any such
> > > corruption, interception, amendment, tampering or viruses or any
> > > consequences thereof.
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > > _______________________________________________
> > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > >
> >
> >
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 
> 
> 

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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: Barry Dingle <barry.dingle@bigfoot.com.au>,
        "'Arnoud van Wijk'"
	 <a.vwijk@viataal.nl>,
        "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        Gonzalo.Camarillo@ericsson.com, sipping@ietf.org
Cc: rrroy@att.com, henry.sinnreich@mci.com, paulej@packetizer.com,
        gv@trace.wisc.edu, Len Bytheway <len.bytheway@aceinfo.net.au>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Sun, 29 Feb 2004 19:16:13 -0500
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While we can, and have, specified a single worldwide URI
for emergency calls, I believe it is not even technically 
possible to find a single 3 digit number that is not
already in use somewhere.  Thus, I suspect anyone
dreaming that such a thing will happen probably is
going to be disappointed.

Since existing TDD devices do not provide an indication
that they are present until after the call is completed,
I think that we should not even consider use of any
kind of gateway for emergency calls.  This is true of 
V.18 devices also, correct?

Having a special number for text emergency calls does
have an impact on the sos draft however.  The draft
discusses use of local emergency numbers.  That text
may have to be expanded to cover such a case.

Are there other examples where there are special
numbers that are used for a subset of emergency
calls other than this kind of example (discriminate
based on media) and the examples of a direct number
for a specific responder, for example, police?

Brian

> -----Original Message-----
> From: Barry Dingle [mailto:barry.dingle@bigfoot.com.au]
> Sent: Thursday, February 26, 2004 6:49 PM
> To: Rosen, Brian; 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
> gv@trace.wisc.edu; Len Bytheway; Barry Dingle
> Subject: Re: Betr.: [Sipping] ToIP Requirements
> 
> 
> Brian,
> 
> In Australia, we have a separate emergency number for 
> Textphone users. It is
> '106' and is regulated by legislation.
> 
> We have a separate '106' emergency call centre. This makes 
> special support
> for a range of text services much easier. (See
> http://www.aceinfo.net.au/Services/NRS/Services/ers.html).  I 
> understand
> that this may be difficult to implement for historical reasons in some
> countries - but it is possible and works well.
> 
> As an aside, we are seriously investigating a 'Text Server' 
> to support a
> whole range of text interworking scenarios - and ToIP is planned to be
> supported in Stage 2. I believe that we will eventually move 
> to a single
> emergency number - but not until we can easily and reliably 
> identify text
> emergency calls.
> 
> Cheers,
> 
> Barry DINGLE
> Telecommunications Consultant
> Project Manager - ACIF Any-to-Any Text Connectivity Options WG
> Fellow of the University of Melbourne
>    barry.dingle@bigfoot.com.au
> Mob:   +61 (0)41 911 7578
> Fixed: +61 (0)3 9725 3937
> ----- Original Message ----- 
> From: "Rosen, Brian" <Brian.Rosen@marconi.com>
> To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>; "'Gunnar Hellstrom'"
> <gunnar.hellstrom@omnitor.se>; <Gonzalo.Camarillo@ericsson.com>;
> <sipping@ietf.org>
> Cc: <rrroy@att.com>; <barry.dingle@bigfoot.com.au>;
> <henry.sinnreich@mci.com>; <paulej@packetizer.com>; 
> <gv@trace.wisc.edu>
> Sent: Friday, February 27, 2004 1:36 AM
> Subject: RE: Betr.: [Sipping] ToIP Requirements
> 
> 
> > Arnoud
> >
> > I do not believe this can work.  A TDD user dials 112 or 
> 911.  Today, this
> > will be routed directly to a "selective router" (a TDM 
> switch) which sends
> > the
> > call to the correct Emergency Response Center.  You can't insert a
> gateway,
> > because you don't know that it's a TDD call until it is 
> answered.  To make
> > it work, you would have to establish a special PSTN number 
> for emergency
> > TDD,
> > which would route to the gateway, which would then call the 
> ERC.  I don't
> > believe that is practical.
> >
> > Brian
> >
> >
> >
> > > -----Original Message-----
> > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > Sent: Thursday, February 26, 2004 9:17 AM
> > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > Hello Brian,
> > > That is also why we need to look at the emergency call system.
> > > This text gateway is only at the PSTN side, thus another
> > > requirement for a
> > > T140/RTP gateway for the PSTN is that when a local 
> emergency number is
> > > dialled (112, or 911 etc). That the gateway must be able to
> > > receive that
> > > call (even dropping existing calls if necessary). Such a call
> > > is easy to
> > > recognize as an emergency call. (PSTN side).
> > > From that gateway on, it is all IP with RFC2793 interactive
> > > text. And then
> > > it is just all existing SIP as if it was from an IP end-point.
> > >
> > > This gives emergency centers an advantage, all calls will be
> > > T.140/RTP, so
> > > integration in their computer systems will be easy instead
> > > being forced to
> > > have analog textphones available for a textphone call. (or a
> > > separate text
> > > telephone emergency center).
> > >
> > > /Arnoud
> > >
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > >
> > >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: donderdag 26 februari 2004 15:04
> > > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > > Arnoud
> > >
> > > Generally, you don't want to rely on very many SIP elements
> > > to complete
> > > an emergency call.  Present systems accept TDD at all answering
> > > points because they don't know before the calls is 
> answered that it
> > > is a TDD call.  I certainly don't want the user to have 
> to do anything
> > > other than dial his local emergency number, or complete a device
> > > specific key sequence to complete an emergency call, and I don't
> > > want a gateway busy to prevent an emergency call.
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > > Sent: Thursday, February 26, 2004 6:07 AM
> > > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > > Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > Actually,
> > > > When using the text gateway, there is no need to detect the
> > > > TDD/Analogue
> > > > text telephone tones/signals. That is the job of the 
> text gateway.
> > > >
> > > > And for the emergency calling. We have to look closely 
> at this in
> > > > combination with a text gateway. Possible behaviour/action.
> > > > And possible
> > > > action in SIP.
> > > >
> > > > Right now I cannot say what has to happen, I must study the
> > > > emergency calls
> > > > in depth. But as I see it, using the text gateway, all SIP
> > > > originating and
> > > > receiving emergency calls will be possible for any 
> textphone user.
> > > >
> > > > Cheers
> > > >
> > > > Arnoud
> > > >
> > > >
> > > > Drs. Arnoud A. T. van Wijk
> > > > Viataal
> > > > Research & Development
> > > > Afdeling RDS
> > > > Theerestraat 42
> > > > 5271 GD Sint-Michielsgestel
> > > > The Netherlands.
> > > > Mobile: +31651921948
> > > > International text telephone: +31735588408
> > > >
> > > > -----Original Message-----
> > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > On Behalf Of
> > > > Rosen, Brian
> > > > Sent: woensdag 25 februari 2004 23:08
> > > > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > > I'm particularly interested in the last one, emergency calling.
> > > >
> > > > Given that you propose an appropriate SDP indicating
> > > support for text,
> > > > do you think there is anything else we need to do to in SIP
> > > > to make this
> > > > happen?  I don't see anything.
> > > >
> > > > I know that the problem of using a legacy TDD device on 
> any kind of
> > > > adapter is as problematic as current TDD; the call will answered
> > > > as if it was a voice call, the baudot/whatever tones will be
> > > > detected,
> > > > and the operator will look for text.
> > > > Can't think of any way to make that better.
> > > >
> > > > Do note that language preference headers should be set!!!
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > >
> > > > > I think PSTN textphone gateway invocation is one area where
> > > > > we either need
> > > > > protocol extensions, or need agreements on what current
> > > > > mechanisms to use.
> > > > >
> > > > > Another is invocation of various kinds of relay services or
> > > > > translation
> > > > > services.
> > > > >
> > > > > A third is calling emergency services and reach positions
> > > capable of
> > > > > handling the type of communication that the user prefers.
> > > > >
> > > > >
> > > > >
> > > > > Gunnar
> > > > > -------------------------------------------
> > > > > Gunnar Hellstrom
> > > > > Omnitor AB
> > > > > Renathvagen 2
> > > > > SE 121 37 Johanneshov
> > > > > SWEDEN
> > > > > +46 8 556 002 03
> > > > > Mob: +46 708 204 288
> > > > > www.omnitor.se
> > > > > Gunnar.Hellstrom@Omnitor.se
> > > > > --------------------------------------------
> > > > >
> > > > >
> > > > > -----Original Message-----
> > > > > From: sipping-admin@ietf.org
> > > > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > > > A vWijk
> > > > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > > > gv@trace.wisc.edu
> > > > > Subject: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > >
> > > > > Hello Gonzalo,
> > > > > While there are device requirements in the text. Those
> > > > > requirements and the
> > > > > explanation on the different analogue text telephone systems
> > > > > is included to
> > > > > give the reader an understanding what kind of devices will
> > > > > connect with
> > > > > ToIP. And the need for the text gateways. And what the
> > > > > different behaviour
> > > > > can/may be compared to end to end IP devices.
> > > > > The most important part will be the ToIP framwork and the
> > > > > text gateways and
> > > > > how such gateways will be used and work with the ToIP 
> framework.
> > > > >
> > > > > There is a ToIP framework where a terminal uses streaming
> > > > > text as defined in
> > > > > RFC2793 to connect with any other Interactive
> > > > > texting/real-time streaming
> > > > > text device.
> > > > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > > > established with
> > > > > SIP. But in other cases, a text gateway must be used to allow
> > > > > T.140/RTP to
> > > > > connect with a non IP text telephony device.
> > > > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > > > text gateway to
> > > > > be able to understand RFC2793 text. (PSTN-IP)
> > > > > It can be a mobile phone running GPRS that uses the text
> > > gateway to
> > > > > understand RFC2793 text. (this exist right NOW using SIP! And
> > > > > those gateways
> > > > > will be used in mobile networks this year!)
> > > > > Or a mobile text telephone using CTM via the text gateway to
> > > > > communicate
> > > > > with RFC2793 text.
> > > > > Or even one analogue text telephone connecting to another
> > > > > analogue text
> > > > > telephone in a different country, and the PSTN's use IP to
> > > > > connect the call.
> > > > >
> > > > > The question is for example, how do we use that gateway from
> > > > > the IP side? Is
> > > > > it invisible for the IP side? Or do we need special
> > > > > extensions in SIP to
> > > > > that the IP terminal knows it is communicating with the
> > > > > end-device through
> > > > > the text gateway?
> > > > > There may be a slightly different call setup flow when a text
> > > > > gateway is
> > > > > included.
> > > > > Or additional SIP messages indicating gateway activity. When
> > > > > you connect for
> > > > > example to an analogue text telephone with an RFC2793 capable
> > > > > IP terminal
> > > > > the capabilities will be less then IP to IP terminal directly
> > > > > (different
> > > > > transmission speeds, half/full duplex etc).
> > > > > Or what to do with emergency calls that are in RFC 
> 2793 text only?
> > > > >
> > > > > That are requirements that will be defined and worked out in
> > > > > this draft.
> > > > > Also, we want to make everybody in SIP and SIPPING aware
> > > > > about those text
> > > > > gateways. So that complicated matters like how to send
> > > analogue text
> > > > > telephone text through a PSTN and connect with a SIP
> > > > > telephone using RFC2793
> > > > > interactive texting. When using the text gateway, there is
> > > > > then no need for
> > > > > RFC 2833 events at the PSTN for text telephones.
> > > > >
> > > > > So, yes. I believe that this should be in the SIPPING WG. And
> > > > > it will fit in
> > > > > the charter - support for hearing-/speech-impaired calling.
> > > > >
> > > > > greetz
> > > > >
> > > > > Arnoud
> > > > >
> > > > >
> > > > >
> > > > > Drs. Arnoud A. T. van Wijk
> > > > > Viataal
> > > > > Research & Development
> > > > > Afdeling RDS
> > > > > Theerestraat 42
> > > > > 5271 GD Sint-Michielsgestel
> > > > > The Netherlands.
> > > > > Mobile: +31651921948
> > > > > International text telephone: +31735588408
> > > > >
> > > > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > > > 24-02-04 10:21 >>>
> > > > > Hello,
> > > > >
> > > > > the draft below provides a set of requirements for real-time
> > > > > conversations using SIP.
> > > > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > > > ip-01.txt
> > > >
> > > > Nevertheless, I have found a lot of device requirements in
> > > > the text, but
> > > > I fail to understand which particular protocol requirements
> > > we need to
> > > > fulfill with new protocol extensions.
> > > >
> > > > While SIPPING is the right place to handle requirements
> > > that will lead
> > > > to new SIP protocol extensions, SIPPING may not be the
> > > right place to
> > > > deal with device requirements (e.g., text phones MUST have a
> > > > display of
> > > > at least 100 characters).
> > > >
> > > > Thanks,
> > > >
> > > > Gonzalo
> > > >
> > > >
> > > > This communication is confidential and intended solely for
> > > > the addressee(s).
> > > > Any unauthorized review, use, disclosure or distribution is
> > > > prohibited. If
> > > > you believe this message has been sent to you in error,
> > > > please notify the
> > > > sender by replying to this transmission and delete the
> > > message without
> > > > disclosing it. Thank you.
> > > >
> > > > E-mail including attachments is susceptible to data corruption,
> > > > interruption, unauthorized amendment, tampering and viruses,
> > > > and we only
> > > > send and receive e-mails on the basis that we are not liable
> > > > for any such
> > > > corruption, interception, amendment, tampering or viruses or any
> > > > consequences thereof.
> > > >
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > > >
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > > >
> > > > _______________________________________________
> > > > Sipping mailing list  
> https://www1.ietf.org/mailman/listinfo/sipping
> > > > This list is for NEW development of the application of SIP
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sip@ietf.org for new developments of core SIP
> > > >
> > > >
> > >
> > >
> >
> 
> 

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



From exim@www1.ietf.org  Sun Feb 29 20:45:55 2004
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Thread-Topic: manyfolks-sipping-ToIP-01
Thread-Index: AcP/Dchf1VZgVdrrQT2p/XZJCZCb5wAHyKlA
From: "Eric Burger" <eburger@snowshore.com>
To: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>, <toip@snowshore.com>,
        "Sipping (E-mail)" <sipping@ietf.org>
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Works for me.

> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> Sent: Sunday, February 29, 2004 4:48 PM
> To: Eric Burger; TOIP Mailing List; Sipping (E-mail)
> Subject: RE: manyfolks-sipping-ToIP-01
>=20
>=20
> Eric, the specified transmission method is designed for both=20
> satisfying the
> users and not straining networks:
>=20
> In order to not strain networks, the transport level RFC2793 says that
> characters should be buffered together in packets for shipping. The
> recommendation for shipping packets with text is to use 300=20
> ms intervals
> between transmissions.
>=20
> The revision of RFC2793, in draft-ietf-avt-rfc2793bis-02 is=20
> stronger and
> more clear about this RECOMMENDATION.
>=20
> With the 300 ms interval between packet transmissions, we=20
> have a text flow
> that is experienced to be close enough to=20
> character-by-character, and a
> maximum network load of 3 packets per second containing=20
> around 3 kbit/s,
> including redundancy. That is sustainable for most network=20
> connections.
>=20
> Are there really typists that can type 60 characters per second????
> I would expect that the maximum load would come from speech-to-text
> arrangements, where I would guess that 20 characters per=20
> second would be a
> practical maximum. Therefore, 30 characters/s throughput in=20
> transmission
> sounded sufficient for me.
>=20
> With the buffering described above, this figure is not=20
> critical, so if you
> have reasons to think that there is a real requirement for 60=20
> cps, we should
> document it.
>=20
> Subtitling a radio report from a football match by using=20
> speech-to-text
> might be among the more challenging applications..
>=20
>=20
> Gunnar
> -------------------------------------------
> Gunnar Hellstr=F6m
> Omnitor AB
> Renathv=E4gen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>=20
>=20
> -----Original Message-----
> From: owner-toip@snowshore.com [mailto:owner-toip@snowshore.com]On
> Behalf Of Eric Burger
> Sent: Sunday, February 29, 2004 8:50 PM
> To: toip@snowshore.com; Sipping (E-mail)
> Subject: manyfolks-sipping-ToIP-01
>=20
>=20
> I like the "network must support a minimum of 30cps" for text=20
> (section 6,
> requirement (d)).  Then I thought about it: that's a ptime of=20
> 33ms.  That is
> OK.  However, if we said minimum of 30cps, but really like 60cps
> (professional typist), that is a ptime of 16ms, which would=20
> start to strain
> many networks.
>=20
> Why do we say that the file format for storage should be=20
> XHTML?  Are we
> archiving marked-up text?  Why not just UTF-8?
>=20
>=20
> -
> This list is maintained by Snowshore Networks -=20
> http://www.snowshore.com
> All comments on this list are the comments of the message=20
> originators and
> Snowshore is not to be held responsible for any actions or=20
> comments found
> on this list. The archives for this list can be found at
> http://flyingfox.snowshore.com/toip_archive/maillist.html
>=20
>=20
>=20
>=20
>=20


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



From exim@www1.ietf.org  Sun Feb 29 21:20:47 2004
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I think you're leaving a key element out of your argument of how unique this
type of mechanism is. UAs are not envisioned to be generating these IDs on a
whim, really it's something that they do at build time (ie embedded into the
hardware perhaps, ala MAC, etc) or something they do at installation time
(soft phone, etc). 

Because of this, I believe the uniqueness is much higher than you've
calculated because in your example, all of those billion UA's would have to
have been installed within that 15 minute period as well, typically. This is
highly unlikely to ever occur, and furthermore, if it should ever occur, I'd
be willing to eat my hat (given that the odds of me being struck dead by a
meteorite or dying of old age are vastly more likely). =)

Regards,

Brian

-----Original Message-----
From: Eric Burger [mailto:eburger@snowshore.com]
Sent: Sunday, February 29, 2004 1:50 PM
To: Stucker, Brian [NGC:B617:EXCH]
Cc: Sipping (E-mail)
Subject: Comments on sip-guid-00


First, a statistics issue.  I disagree that the time, in of itself, has
quite the properties asserted.  For example, given a billion UA's, if they
have a 15-minute clock drift with a Normal distribution, well over 3,000 of
them will think it is the same time in any given 5-minute window.

That is still pretty good odds, but not the one-in-a-trillion asserted.
This gets worse if the UA's have random number generators that have a
correlation with their notion of time (e.g., time is used as a salt, boot at
the same time with the same salt, etc.).  [Yes, I did Monte Carlo
simulations a lifetime ago...]

The text goes through the algorithm for generating a GUID, but then the
example has what looks like an IP address in it.  Is that a proper
interpretation?  Where does the 10.33.15.8 come from?


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<TITLE>RE: Comments on sip-guid-00</TITLE>
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<P><FONT SIZE=3D2>I think you're leaving a key element out of your =
argument of how unique this type of mechanism is. UAs are not =
envisioned to be generating these IDs on a whim, really it's something =
that they do at build time (ie embedded into the hardware perhaps, ala =
MAC, etc) or something they do at installation time (soft phone, etc). =
</FONT></P>

<P><FONT SIZE=3D2>Because of this, I believe the uniqueness is much =
higher than you've calculated because in your example, all of those =
billion UA's would have to have been installed within that 15 minute =
period as well, typically. This is highly unlikely to ever occur, and =
furthermore, if it should ever occur, I'd be willing to eat my hat =
(given that the odds of me being struck dead by a meteorite or dying of =
old age are vastly more likely). =3D)</FONT></P>

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

<P><FONT SIZE=3D2>Brian</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Eric Burger [<A =
HREF=3D"mailto:eburger@snowshore.com">mailto:eburger@snowshore.com</A>]<=
/FONT>
<BR><FONT SIZE=3D2>Sent: Sunday, February 29, 2004 1:50 PM</FONT>
<BR><FONT SIZE=3D2>To: Stucker, Brian [NGC:B617:EXCH]</FONT>
<BR><FONT SIZE=3D2>Cc: Sipping (E-mail)</FONT>
<BR><FONT SIZE=3D2>Subject: Comments on sip-guid-00</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>First, a statistics issue.&nbsp; I disagree that the =
time, in of itself, has quite the properties asserted.&nbsp; For =
example, given a billion UA's, if they have a 15-minute clock drift =
with a Normal distribution, well over 3,000 of them will think it is =
the same time in any given 5-minute window.</FONT></P>

<P><FONT SIZE=3D2>That is still pretty good odds, but not the =
one-in-a-trillion asserted.&nbsp; This gets worse if the UA's have =
random number generators that have a correlation with their notion of =
time (e.g., time is used as a salt, boot at the same time with the same =
salt, etc.).&nbsp; [Yes, I did Monte Carlo simulations a lifetime =
ago...]</FONT></P>

<P><FONT SIZE=3D2>The text goes through the algorithm for generating a =
GUID, but then the example has what looks like an IP address in =
it.&nbsp; Is that a proper interpretation?&nbsp; Where does the =
10.33.15.8 come from?</FONT></P>

</BODY>
</HTML>
------_=_NextPart_001_01C3FF32.82468C18--

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From exim@www1.ietf.org  Sun Feb 29 21:28:02 2004
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: sipping@ietf.org
Date: Mon, 1 Mar 2004 02:23:23 -0000 
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Subject: [Sipping] comments on draft-schulzrinne-sipping-emergency-arch
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In general, I am happy with this draft, but in reading it, I did have a few specific comments.

1)	Section 3 (Terminology). I am confused here, and in the rest of the document, about the distinction between ECC and PSAP. I think the issue here is that as defined by the associated text, these are actually different concepts in some countries, yet the text goes on to say that in USA, PSAP is the ECC. Maybe we need to clarify.

2)	Section 5.2. Mobile systems, possible in conjunction with the cell site location, may also transmit mobile country code and mobile network code of the host network. This MCC and MNC in itself does constitute location information, in that it tells that the user (with border constraints) is in a particular country) - which may be sufficient itself for determining the ECC to be used.

3)	Section 5.3. Somewhere around here, for any particular emergency call, we need to identify that it is possible to have multiple elements of location information generated by different entities involved in the call (e.g. terminal generated, outbound proxy generated, etc.). We need a statement that each location can be added to the call information, and that one does not overwrite another. It is up to the ECC to decide which (if any) location it wishes to believe, possibly in association with any location from the end user received via the media path.

4)	Section 5.4. Where information is delivered later, do we need an indicator within the INVITE that such information may be available later in the call? What is the mechanism for providing information later in the call (push or pull), i.e. is additional information just sent later in the dialog in another request, or does the location gathering entity indicate the availability of new information which can then be fetched.

5)	Section 7. This section seems to assume sips. As this is apparently a hop-by-hop mechanism, what is wrong with depending on any hop by hop security mechanism, e.g. IPsec - why does it have to be SIPs.

6)	Section 9. In addition to the generic mechanism mentioned here, there may be access transport specific mechanisms for downloading this information to the UE. For example, a 3GPP phone from release 5 onwards can have this information downloaded from visited network entities at network registration time. The existence of these mechanisms should be acknowledged in the document at this point.

regards

Keith

Keith Drage
Lucent Technologies
drage@lucent.com
tel: +44 1793 776249

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Agreed :-)

-----Original Message-----
From: Brian Stucker [mailto:bstucker@nortelnetworks.com]
Sent: Sunday, February 29, 2004 9:16 PM
To: Eric Burger
Cc: Sipping (E-mail)
Subject: RE: Comments on sip-guid-00


I think you're leaving a key element out of your argument of how unique =
this type of mechanism is. UAs are not envisioned to be generating these =
IDs on a whim, really it's something that they do at build time (ie =
embedded into the hardware perhaps, ala MAC, etc) or something they do =
at installation time (soft phone, etc).=20
Because of this, I believe the uniqueness is much higher than you've =
calculated because in your example, all of those billion UA's would have =
to have been installed within that 15 minute period as well, typically. =
This is highly unlikely to ever occur, and furthermore, if it should =
ever occur, I'd be willing to eat my hat (given that the odds of me =
being struck dead by a meteorite or dying of old age are vastly more =
likely). =3D)
Regards,=20
Brian=20
-----Original Message-----=20
From: Eric Burger [mailto:eburger@snowshore.com]=20
Sent: Sunday, February 29, 2004 1:50 PM=20
To: Stucker, Brian [NGC:B617:EXCH]=20
Cc: Sipping (E-mail)=20
Subject: Comments on sip-guid-00=20


First, a statistics issue.  I disagree that the time, in of itself, has =
quite the properties asserted.  For example, given a billion UA's, if =
they have a 15-minute clock drift with a Normal distribution, well over =
3,000 of them will think it is the same time in any given 5-minute =
window.
That is still pretty good odds, but not the one-in-a-trillion asserted.  =
This gets worse if the UA's have random number generators that have a =
correlation with their notion of time (e.g., time is used as a salt, =
boot at the same time with the same salt, etc.).  [Yes, I did Monte =
Carlo simulations a lifetime ago...]
The text goes through the algorithm for generating a GUID, but then the =
example has what looks like an IP address in it.  Is that a proper =
interpretation?  Where does the 10.33.15.8 come from?


_______________________________________________
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Comments on WGLC for reason header.
Sorry that I didn't review this version and comment by Feb 29. I had thought 
that this issue was resolved after our e-mail discussion some time ago.

The draft defines 2 reasons for preemption (access and network) and provides 
an indicatation to the UA of which one occurred.

The requirment for Assured Service (otherwise known as MLPP) is that the 
preempted party is not allowed to know why the preemption occurred. This would 
have the result of telling the person something they are not supposed to know, 
that the party they were talking to got a more important call.

Presuming that some network may want to notify the UA of the reason, this 
needs to be corrected by adding a third reason in Section 6.0 as follows (also in 
Section 4.0 as appropriate):

    In instances where this namespace is used to indicate preemption, 
    but it is not desirable to notify the UA at the end of the preempted 
    session the reason for the preemption, the following syntax shall be used:

       Reason: preemption ; cause=3 ;text="Unspecified_Preemption"

       This case may be used in place of either the UA_Preemption in 
       Section 4.1 or the RSVP_Preemption in Section 4.2.

If neccesary, a new Section 4.3 could be added to explain this 3rd value, 
that is, why/when it would be used. This would be one paragraph, no diagram 
needed. I'll provide it if needed.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>Comments on W=
GLC for reason header.
<BR>Sorry that I didn't review this version and comment by Feb 29. I had tho=
ught that this issue was resolved after our e-mail discussion some time ago.
<BR>
<BR>The draft defines 2 reasons for preemption (access and network) and prov=
ides an indicatation to the UA of which one occurred.
<BR>
<BR>The requirment for Assured Service (otherwise known as MLPP) is that the=
 preempted party is not allowed to know why the preemption occurred. This wo=
uld have the result of telling the person something they are not supposed to=
 know, that the party they were talking to got a more important call.
<BR>
<BR>Presuming that some network may want to notify the UA of the reason, thi=
s needs to be corrected by adding a third reason in Section 6.0 as follows (=
also in Section 4.0 as appropriate):
<BR>
<BR>    In instances where this namespace is used to indicate preemption,=20
<BR>    but it is not desirable to notify the UA at the end of the preempted=
=20
<BR>    session the reason for the preemption, the following syntax shall be=
 used:
<BR>
<BR>     &nbsp;&nbsp;Reason: preemption ; cause=3D3 ;text=3D"Unspecified_Pre=
emption"
<BR>
<BR>     &nbsp;&nbsp;This case may be used in place of either the UA_Preempt=
ion in=20
<BR>     &nbsp;&nbsp;Section 4.1 or the RSVP_Preemption in Section 4.2.
<BR>
<BR>If neccesary, a new Section 4.3 could be added to explain this 3rd value=
, that is, why/when it would be used. This would be one paragraph, no diagra=
m needed. I'll provide it if needed.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_15e.2bc7fd9f.2d73fa9e_boundary--

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Subject: Re: [Sipping] Comments on sip-voicemail-uri-01
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Eric Burger wrote:
> Big question: why bother if we get everything from history-info?  I know, "it's easier on the UAS", but do we care?

I think the big difference is that it moves the responsibility for who 
decides. If you *want* the voicemail system to decide which mailbox to 
use, then history-info is a plausible mechanism for giving it all the 
raw data, and then letting it decide.

But if you want to let something else (e.g. a proxy) decide which 
mailbox (perhaps based on policy for the AOR rather than policy for the 
VM system) then some mechanism to explicitly tell the VM system your 
choice seems more appropriate.

I suppose the proxy could lie in the history-info, based on a 
presumption of what algorithm the VM system uses to choose the mailbox, 
but I certainly hope nobody goes that way.

	Paul


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Date: Sun, 29 Feb 2004 22:12:31 -0600
From: Alan Johnston <alan.johnston@mci.com>
Subject: RE: [Sipping] Conferencing for UAs
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Markus,

I think you are on the right track.  Look for some movement on harmonizing 
exploders and conferencing this week  - stay tuned...

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

At 03:16 PM 2/29/2004 +0200, Markus.Isomaki@nokia.com wrote:
>Hi,
>
>I also hope there could be more clarity on the relationship between 
>exploders and conferencing work. Some thoughts below.
>
>One clear use case for exploders is the establishment of a "dial-out 
>ad-hoc" conference. One of Gonzalo's drafts proposes that the exploder 
>B2BUA (which in this case becomes a conference focus) could return to the 
>originating UA a reference (URI) to the list, after which the UA could use 
>e.g. XCAP to manage the list out-of-band. The idea seems to be that the 
>default format for this list indeed would be the XCAP resource list.
>
>In conferencing framework this list clearly overlaps with the dial-out 
>list concept to be defined in CPCP. Also, if the semantics of removing a 
>user from the list will result in ejection of this user from the 
>conference, this list may overlap with some other functionality of CPCP too.
>
>My proposal to make the exploders and CPCP compliant with each other would be:
>- When the conference is established by invoking an exploder, the 
>conference server will use the exploder-list to populate a CPCP document 
>in such a way that its semantics match with what was done with the exploder.
>- URI for this CPCP document is provided in 200 OK using the mechanisms 
>dicussed elsewhere (Call-info purpose, new header, ...).
>- Conference participants can thereafter make conference policy operations 
>by doing changes to this document, and the situation is as if the 
>conference was originally created by CPCP in the first place.
>- If ad hoc list management via re-INVITEs is still allowed, there needs 
>to be a clear mapping how adding/removing list entries interacts with the 
>CPCP document.
>
>To enable this the CPCP semantics still need more clarity, and maybe also 
>adopting the XCAP resource list format for the possible dial-out etc. 
>lists in that document would help this mapping.
>
>(We could still keep the simple non-CPCP controllable conference case, 
>where the only manipulated object would be a simple resource list. In this 
>case access lists and other such features woould not be supported.)
>
>Do people think that this would be feasible? Or would we be getting to an 
>interaction that is too complex to model? In general I think there's room 
>for both exploders AND CPCP, so it would be important to specify how they 
>interwork.
>
>Regards,
>         Markus
>
> > -----Original Message-----
> > From: sipping-admin@ietf.org
> > [mailto:sipping-admin@ietf.org]On Behalf Of
> > ext Alan Johnston
> > Sent: 26 February, 2004 21:05
> > To: Gonzalo Camarillo
> > Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org;
> > oritl@microsoft.com; Garcia Miguel.An (Nokia-NRC/Helsinki)
> > Subject: Re: [Sipping] Conferencing for UAs
> >
> >
> >
> > At 02:13 PM 2/26/2004 +0200, Gonzalo Camarillo wrote:
> > >Alan,
> > >
> > >I agree with you that we do not want to change the
> > conferencing framework
> > >at this point in time (I already said in my previous mail that I
> > >understood that my comments related to this came too late).
> > Still, I found
> > >the discussion useful to understand your point of view.
> >
> > Gonzalo,
> >
> > Glad to hear that.  I'd be happy to think about how to extend the
> > conferencing work to cover the exploder scenarios you are
> > proposing with
> > MESSAGE, INVITE, and even REFER.    With the current drafts,
> > I have a hard
> > time understanding how to do this, but I look forward to
> > clarifications.   The SUBSCRIBE exploder does not seem to have any
> > conferencing semantics, but seems to be a SIMPLE application
> > since it is
> > tightly coupled to the presence application.
> >
> > Thanks,
> > Alan Johnston
> > MCI
> > sip:alan@sipstation.com
> >
> >
> > >In any event, the topic I am interested in is the creation
> > of different
> > >resources (identified by URIs) by a single request. In
> > particular, how to
> > >communicate those URIs to the UAC. (I sent separate comments
> > on Hisham's
> > >draft.)
> > >
> > >I hope we can discuss these issues in Korea.
> > >
> > >Thanks,
> > >
> > >Gonzalo
> > >
> > >
> > >Alan Johnston wrote:
> > >
> > >>At 09:47 AM 2/25/2004 +0200, Gonzalo Camarillo wrote:
> > >>
> > >>>Hi Alan,
> > >>>
> > >>>I understand that my comments come late in the game and
> > that it might be
> > >>>impossible to change the conferencing framework at this
> > point. In any
> > >>>event, I do not agree with your statement below:
> > >>>
> > >>>>No - this is the problem.  You are looking at your single narrow
> > >>>>application.
> > >>>
> > >>>
> > >>>I believe I am actually looking at a much wider range of
> > applications
> > >>>than just conferencing. So, to me, the problem is that the
> > conferencing
> > >>>framework is too focused on a "single narrow application";
> > conferencing.
> > >>
> > >>Gonzalo,
> > >>If you are saying that some of your exploder services do not have a
> > >>conferencing component, then I agree.  Some of the exploder
> > services seem
> > >>to just be a "request to fork" to a set of URIs.  Others
> > definitely have
> > >>conferencing semantics.  The ones with conferencing
> > semantics, I believe,
> > >>can be, and should be modeled using the conferencing framework.
> > >>The problem is that each method seems to have a different set of
> > >>semantics as to what exploding that method means - INVITE
> > is handled
> > >>differently from SUBSCRIBE, from MESSAGE.   From a high
> > level, exploders
> > >>may seem to be a consistent set of services.  However, as I
> > delve into
> > >>the details and think about call flows (something  absent
> > from any of the
> > >>exploder drafts), a different story emerges.
> > >>I think we need to explore exactly the services and call
> > flows that are
> > >>being envisioned for exploders before proposing major
> > changes to our
> > >>conferencing framework, which I believe is well understood
> > and documented.
> > >>
> > >>>In a conferencing service, there is a single resource
> > created by the
> > >>>initial INVITE; a conference. That's why the server can
> > use the Contact
> > >>>header field to send it back to the UAC. If you think of
> > services that
> > >>>create more than one resource (e.g., an explosion and a
> > conference), the
> > >>>server may not want to use the same URI for both. Moreover,
> > >>>conceptually, is is not a good thing to use the same URI
> > to identify
> > >>>both resources.
> > >>
> > >>As Paul has pointed out - they are the same resource.
> > Additionally, it
> > >>is a trivial SIP routing issue to fix this - many proxies
> > do method-based
> > >>request routing - that would completely solve your problem
> > of sending
> > >>SUBSCRIBEs to a different UA than the initial request.
> > >>
> > >>>So, my main concern is the semantics overload of the
> > Contact header. (We
> > >>>abused this header when design REGISTER as well, by the
> > way. That's why
> > >>>I do not want to do it again.)
> > >>>
> > >>>In my opinion, the Contact header is the URI where you
> > send subsequent
> > >>>requests within the dialog. If the server wants to tell
> > the client that
> > >>>a resource (e.g., a conference) has been created, it can
> > use a different
> > >>>header to include the URI of such a resource. Note that,
> > if you want
> > >>>both URIs to be the same (like in the conferencing case),
> > you can do it.
> > >>>
> > >>>Contact: sip:1234@a.com;isfocus
> > >>>Resource: sip:1234@a.com;isfocus
> > >>>
> > >>>An entity receiving these headers would use the URI in the Contact
> > >>>header to send subsequent requests *within* the dialog,
> > and the URI in
> > >>>the other header to send requests *outside* that dialog
> > (e.g., SUBSCRIBE
> > >>>to the conference package).
> > >>
> > >>No.  A GRUU should be used in the Contact, and its routing is valid
> > >>outside the dialog.
> > >>
> > >>>In brief, instead of sticking the conference URI in a
> > Contact header and
> > >>>label that URI with an isfocus parameter so that the UAC
> > figures what is
> > >>>going on, I find it more explicit to use a different
> > header. Although,
> > >>>as I said, I understand that this comment comes late and
> > that may not be
> > >>>useful at this point.
> > >>
> > >>I think much more work is needed before the working group
> > can evaluate
> > >>the exploders work and see how and if it fits in with
> > existing concepts
> > >>and approaches.
> > >>Thanks,
> > >>Alan Johnston
> > >>MCI
> > >>sip:alan@sipstation.com
> > >>
> > >>>Thanks,
> > >>>
> > >>>Gonzalo
> > >>>
> > >>>
> > >>>This communication is confidential and intended solely for the
> > >>>addressee(s). Any unauthorized review, use, disclosure or
> > distribution
> > >>>is prohibited. If you believe this message has been sent to you in
> > >>>error, please notify the sender by replying to this
> > transmission and
> > >>>delete the message without disclosing it. Thank you.
> > >>>
> > >>>E-mail including attachments is susceptible to data corruption,
> > >>>interruption, unauthorized amendment, tampering and
> > viruses, and we only
> > >>>send and receive e-mails on the basis that we are not
> > liable for any
> > >>>such corruption, interception, amendment, tampering or
> > viruses or any
> > >>>consequences thereof.
> > >>>
> > >>>
> > >>>_______________________________________________
> > >>>Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > >>>This list is for NEW development of the application of SIP
> > >>>Use sip-implementors@cs.columbia.edu for questions on current sip
> > >>>Use sip@ietf.org for new developments of core SIP
> > >
> > >--
> > >Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > >Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > >Telecom R&D               Fax   :  +358  9 299 30 52
> > >FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > >Finland                   http://www.hut.fi/~gonzalo
> > >
> > >
> > >This communication is confidential and intended solely for the
> > >addressee(s). Any unauthorized review, use, disclosure or
> > distribution is
> > >prohibited. If you believe this message has been sent to you
> > in error,
> > >please notify the sender by replying to this transmission
> > and delete the
> > >message without disclosing it. Thank you.
> > >
> > >E-mail including attachments is susceptible to data corruption,
> > >interruption, unauthorized amendment, tampering and viruses,
> > and we only
> > >send and receive e-mails on the basis that we are not liable
> > for any such
> > >corruption, interception, amendment, tampering or viruses or any
> > >consequences thereof.
> >
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >


_______________________________________________
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This list is for NEW development of the application of SIP
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Use sip@ietf.org for new developments of core SIP



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: [Sipping] Re: Forwarding URI list requirement (sipping-exploders-02)
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Eric,

I am not sure if we need that requirement either, actually... this 
requirement deals with something similar to the cc: header in email...

Let's discuss whether or not we need it.

Opinions?

Gonzalo

Eric Burger wrote:

> Requirement 8 says it must be possible to forward a URI list to the recipients of a message sent using the list.
> 
> Where did the requirement come from?  I think that would help decide if it belongs.  I think it opens up a whole security issue in that just because parties B and C got blasted by my spam message, C still might not want B to get their address.

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.

E-mail including attachments is susceptible to data corruption, interruption, unauthorized amendment, tampering and viruses, and we only send and receive e-mails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.


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



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Date: Sun, 29 Feb 2004 22:38:03 -0600
To: Mpierce1@aol.com, sipping@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sipping] WGLC for
  draft-ietf-sipping-reason-header-for-preemption-00.txt
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comments in-line

At 09:31 PM 2/29/2004 -0500, Mpierce1@aol.com wrote:
>Comments on WGLC for reason header.
>Sorry that I didn't review this version and comment by Feb 29. I had 
>thought that this issue was resolved after our e-mail discussion some time 
>ago.
>
>The draft defines 2 reasons for preemption (access and network) and 
>provides an indicatation to the UA of which one occurred.
>
>The requirment for Assured Service (otherwise known as MLPP) is that the 
>preempted party is not allowed to know why the preemption occurred. This 
>would have the result of telling the person

how exactly is informing the UA translate into telling the user why 
something has happened always (which is the way I read your question)?

Communicating in the BYE "why" a session was preempted to all the SIP 
entities (including the remote UA) is a network mgmt requirement (SIP layer 
synchronization) and a function at least some want knowledge of, perhaps 
not the DSN at the moment.

>something they are not supposed to know, that the party they were talking 
>to got a more important call.
>
>Presuming that some network may want to notify the UA of the reason, this 
>needs to be corrected by adding a third reason in Section 6.0 as follows 
>(also in Section 4.0 as appropriate):

I believe this can be accomplished by using the following:

         Reason: Q.850; cause=8

which is a generic preemption indication for the circuit world

if others disagree - adding this 3rd "reason" is fine with me


>In instances where this namespace is used to indicate preemption,
>but it is not desirable to notify the UA at the end of the preempted
>session the reason for the preemption, the following syntax shall be used:
>
>   Reason: preemption ; cause=3 ;text="Unspecified_Preemption"
>
>   This case may be used in place of either the UA_Preemption in
>   Section 4.1 or the RSVP_Preemption in Section 4.2.
>
>If neccesary, a new Section 4.3 could be added to explain this 3rd value, 
>that is, why/when it would be used. This would be one paragraph, no 
>diagram needed. I'll provide it if needed.
>
>Mike Pierce


cheers,
James

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


_______________________________________________
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Eric Burger <eburger@snowshore.com>
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        "Jonathan Rosenberg (E-mail)" <jdrosen@dynamicsoft.com>,
        "Sipping (E-mail)" <sipping@ietf.org>
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Hi,

> Considering the security implications, a content server MUST support
> sips: and MUST support https for retrieval.

Yes, when we do the security considerations section we will talk about 
this... mandating TLs sounds sensible.

> Where do the codec names (6.1.2.1.1) come from?  RTP?  If so, say so.

Yes, they come from the RTP. They are the names of the payloads... we'll 
say so.

Thanks for the comments,

Gonzalo


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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Gunnar Hellstrom'" <gunnar.hellstrom@omnitor.se>,
        Barry Dingle
	 <barry.dingle@bigfoot.com.au>,
        "'Arnoud van Wijk'" <a.vwijk@viataal.nl>,
        Gonzalo.Camarillo@ericsson.com, sipping@ietf.org
Cc: rrroy@att.com, henry.sinnreich@mci.com, paulej@packetizer.com,
        gv@trace.wisc.edu, Len Bytheway <len.bytheway@aceinfo.net.au>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Sun, 29 Feb 2004 23:39:12 -0500
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This means that there is no way to detect a text call with V.18
without answering the call, which is the same as a BAUDOT
device.  All you are pointing out is that you can tell right
away with V.18, which is not the case for BAUDOT.

So, for emergency calling, there is no difference.  All calls
go to the answering point, which has to be text capable.
As was pointed out in previous messages, some jurisdictions
have separate phone numbers for text emergency calls.

So, from the emergency call perspective:
SIP calls marked with Text media would be routed to a text capable=20
  answering point.  This is the fundamental capability we can=20
  do something about.  It requires no new mechanisms.

Mechanisms that learn local emergency phone numbers need to
  be able to accommodate text-specific numbers.

Older V.18 and BAUDOT TDD devices would arrive at emergency
  centers in the way they do now, and are indistinguishable
  from voice calls until they are answered.

In the next revision of ToIP requirements, can we delete (c)
from Section 8.8?  There is no concept of an emergency
stream; a call can be distinguished as an emergency call
and which streams it offers is not germane.

Brian=20

> -----Original Message-----
> From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> Sent: Sunday, February 29, 2004 9:26 PM
> To: Rosen, Brian; Barry Dingle; 'Arnoud van Wijk';
> Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
> gv@trace.wisc.edu; Len Bytheway
> Subject: RE: Betr.: [Sipping] ToIP Requirements
>=20
>=20
> 1. How it is in PSTN:
> A calling V.18 sends a specific data pattern for text=20
> telephony in the V.8
> signal CI.
> It is sent in the connected voice channel.
> So, as with all calls from PSTN, there is no indication=20
> before the call is
> set up about what kind of media will be used.
>=20
> But it is far more easy to identify it as a textphone call=20
> than it is for
> some of the old systems. E.G the US system "Baudot" is silent=20
> when calling,
> and the answering side must guess that it is a textphone.
>=20
> There is not yet any really clear trend to move to V.18 for=20
> users in USA.
> Therefore I agree that the mechanisms for emergency calls=20
> must for a long
> time accomplish for the tedious task to try to respond in=20
> text on PSTN to
> all incoming silent calls to discriminate the text callers.=20
> That response
> should be done with V.18 signals, so that it at least is=20
> prepared for more
> convenient handling when people have moved to using V.18 textphones.
>=20
> 2. In SIP.
> In SIP, we have text in a separate media stream in the call.=20
> An emergency
> centre should have capability to use both voice and text=20
> simultaneously.
> There are mechanisms in discussion about indicating=20
> preference for handling
> the call with text as the main medium. I am not sure that we=20
> can require
> that preference to be set in a SIP terminal used for an=20
> emergency call by a
> text depending person.
> So, the safe handling would be for the emergency centre to offer text
> capability, and if text connection is achieved, an automatic=20
> text greeting
> is sent in that channel, while the emergency agent responds=20
> in voice. If the
> caller then starts using the text channel, it is evident=20
> enough that text is
> preferred.
>=20
>=20
> Gunnar
>=20
> -------------------------------------------
> Gunnar Hellstr=F6m
> Omnitor AB
> Renathv=E4gen 2
> SE 121 37 Johanneshov
> SWEDEN
> +46 8 556 002 03
> Mob: +46 708 204 288
> www.omnitor.se
> Gunnar.Hellstrom@Omnitor.se
> --------------------------------------------
>=20
>=20
> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> Rosen, Brian
> Sent: Monday, March 01, 2004 1:16 AM
> To: Barry Dingle; 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
> gv@trace.wisc.edu; Len Bytheway
> Subject: RE: Betr.: [Sipping] ToIP Requirements
>=20
>=20
> While we can, and have, specified a single worldwide URI
> for emergency calls, I believe it is not even technically
> possible to find a single 3 digit number that is not
> already in use somewhere.  Thus, I suspect anyone
> dreaming that such a thing will happen probably is
> going to be disappointed.
>=20
> Since existing TDD devices do not provide an indication
> that they are present until after the call is completed,
> I think that we should not even consider use of any
> kind of gateway for emergency calls.  This is true of
> V.18 devices also, correct?
>=20
> Having a special number for text emergency calls does
> have an impact on the sos draft however.  The draft
> discusses use of local emergency numbers.  That text
> may have to be expanded to cover such a case.
>=20
> Are there other examples where there are special
> numbers that are used for a subset of emergency
> calls other than this kind of example (discriminate
> based on media) and the examples of a direct number
> for a specific responder, for example, police?
>=20
> Brian
>=20
> > -----Original Message-----
> > From: Barry Dingle [mailto:barry.dingle@bigfoot.com.au]
> > Sent: Thursday, February 26, 2004 6:49 PM
> > To: Rosen, Brian; 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
> > gv@trace.wisc.edu; Len Bytheway; Barry Dingle
> > Subject: Re: Betr.: [Sipping] ToIP Requirements
> >
> >
> > Brian,
> >
> > In Australia, we have a separate emergency number for
> > Textphone users. It is
> > '106' and is regulated by legislation.
> >
> > We have a separate '106' emergency call centre. This makes
> > special support
> > for a range of text services much easier. (See
> > http://www.aceinfo.net.au/Services/NRS/Services/ers.html).  I
> > understand
> > that this may be difficult to implement for historical=20
> reasons in some
> > countries - but it is possible and works well.
> >
> > As an aside, we are seriously investigating a 'Text Server'
> > to support a
> > whole range of text interworking scenarios - and ToIP is=20
> planned to be
> > supported in Stage 2. I believe that we will eventually move
> > to a single
> > emergency number - but not until we can easily and reliably
> > identify text
> > emergency calls.
> >
> > Cheers,
> >
> > Barry DINGLE
> > Telecommunications Consultant
> > Project Manager - ACIF Any-to-Any Text Connectivity Options WG
> > Fellow of the University of Melbourne
> >    barry.dingle@bigfoot.com.au
> > Mob:   +61 (0)41 911 7578
> > Fixed: +61 (0)3 9725 3937
> > ----- Original Message -----
> > From: "Rosen, Brian" <Brian.Rosen@marconi.com>
> > To: "'Arnoud van Wijk'" <a.vwijk@viataal.nl>; "'Gunnar Hellstrom'"
> > <gunnar.hellstrom@omnitor.se>; <Gonzalo.Camarillo@ericsson.com>;
> > <sipping@ietf.org>
> > Cc: <rrroy@att.com>; <barry.dingle@bigfoot.com.au>;
> > <henry.sinnreich@mci.com>; <paulej@packetizer.com>;
> > <gv@trace.wisc.edu>
> > Sent: Friday, February 27, 2004 1:36 AM
> > Subject: RE: Betr.: [Sipping] ToIP Requirements
> >
> >
> > > Arnoud
> > >
> > > I do not believe this can work.  A TDD user dials 112 or
> > 911.  Today, this
> > > will be routed directly to a "selective router" (a TDM
> > switch) which sends
> > > the
> > > call to the correct Emergency Response Center.  You can't insert =
a
> > gateway,
> > > because you don't know that it's a TDD call until it is
> > answered.  To make
> > > it work, you would have to establish a special PSTN number
> > for emergency
> > > TDD,
> > > which would route to the gateway, which would then call the
> > ERC.  I don't
> > > believe that is practical.
> > >
> > > Brian
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > > Sent: Thursday, February 26, 2004 9:17 AM
> > > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > > Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > Hello Brian,
> > > > That is also why we need to look at the emergency call system.
> > > > This text gateway is only at the PSTN side, thus another
> > > > requirement for a
> > > > T140/RTP gateway for the PSTN is that when a local
> > emergency number is
> > > > dialled (112, or 911 etc). That the gateway must be able to
> > > > receive that
> > > > call (even dropping existing calls if necessary). Such a call
> > > > is easy to
> > > > recognize as an emergency call. (PSTN side).
> > > > From that gateway on, it is all IP with RFC2793 interactive
> > > > text. And then
> > > > it is just all existing SIP as if it was from an IP end-point.
> > > >
> > > > This gives emergency centers an advantage, all calls will be
> > > > T.140/RTP, so
> > > > integration in their computer systems will be easy instead
> > > > being forced to
> > > > have analog textphones available for a textphone call. (or a
> > > > separate text
> > > > telephone emergency center).
> > > >
> > > > /Arnoud
> > > >
> > > > Drs. Arnoud A. T. van Wijk
> > > > Viataal
> > > > Research & Development
> > > > Afdeling RDS
> > > > Theerestraat 42
> > > > 5271 GD Sint-Michielsgestel
> > > > The Netherlands.
> > > > Mobile: +31651921948
> > > > International text telephone: +31735588408
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > > Sent: donderdag 26 februari 2004 15:04
> > > > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > > > Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > > Arnoud
> > > >
> > > > Generally, you don't want to rely on very many SIP elements
> > > > to complete
> > > > an emergency call.  Present systems accept TDD at all answering
> > > > points because they don't know before the calls is
> > answered that it
> > > > is a TDD call.  I certainly don't want the user to have
> > to do anything
> > > > other than dial his local emergency number, or complete a =
device
> > > > specific key sequence to complete an emergency call, and I =
don't
> > > > want a gateway busy to prevent an emergency call.
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > > > Sent: Thursday, February 26, 2004 6:07 AM
> > > > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > > > Gonzalo.Camarillo@ericsson.com;
> > > > > sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > >
> > > > > Actually,
> > > > > When using the text gateway, there is no need to detect the
> > > > > TDD/Analogue
> > > > > text telephone tones/signals. That is the job of the
> > text gateway.
> > > > >
> > > > > And for the emergency calling. We have to look closely
> > at this in
> > > > > combination with a text gateway. Possible behaviour/action.
> > > > > And possible
> > > > > action in SIP.
> > > > >
> > > > > Right now I cannot say what has to happen, I must study the
> > > > > emergency calls
> > > > > in depth. But as I see it, using the text gateway, all SIP
> > > > > originating and
> > > > > receiving emergency calls will be possible for any
> > textphone user.
> > > > >
> > > > > Cheers
> > > > >
> > > > > Arnoud
> > > > >
> > > > >
> > > > > Drs. Arnoud A. T. van Wijk
> > > > > Viataal
> > > > > Research & Development
> > > > > Afdeling RDS
> > > > > Theerestraat 42
> > > > > 5271 GD Sint-Michielsgestel
> > > > > The Netherlands.
> > > > > Mobile: +31651921948
> > > > > International text telephone: +31735588408
> > > > >
> > > > > -----Original Message-----
> > > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > > On Behalf Of
> > > > > Rosen, Brian
> > > > > Sent: woensdag 25 februari 2004 23:08
> > > > > To: 'Gunnar Hellstrom'; A vWijk;=20
> Gonzalo.Camarillo@ericsson.com;
> > > > > sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > > I'm particularly interested in the last one,=20
> emergency calling.
> > > > >
> > > > > Given that you propose an appropriate SDP indicating
> > > > support for text,
> > > > > do you think there is anything else we need to do to in SIP
> > > > > to make this
> > > > > happen?  I don't see anything.
> > > > >
> > > > > I know that the problem of using a legacy TDD device on
> > any kind of
> > > > > adapter is as problematic as current TDD; the call=20
> will answered
> > > > > as if it was a voice call, the baudot/whatever tones will be
> > > > > detected,
> > > > > and the operator will look for text.
> > > > > Can't think of any way to make that better.
> > > > >
> > > > > Do note that language preference headers should be set!!!
> > > > >
> > > > > Brian
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > > > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > > > > To: A vWijk; Gonzalo.Camarillo@ericsson.com;=20
> sipping@ietf.org
> > > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > > henry.sinnreich@mci.com;
> > > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > > >
> > > > > >
> > > > > > I think PSTN textphone gateway invocation is one area where
> > > > > > we either need
> > > > > > protocol extensions, or need agreements on what current
> > > > > > mechanisms to use.
> > > > > >
> > > > > > Another is invocation of various kinds of relay services or
> > > > > > translation
> > > > > > services.
> > > > > >
> > > > > > A third is calling emergency services and reach positions
> > > > capable of
> > > > > > handling the type of communication that the user prefers.
> > > > > >
> > > > > >
> > > > > >
> > > > > > Gunnar
> > > > > > -------------------------------------------
> > > > > > Gunnar Hellstrom
> > > > > > Omnitor AB
> > > > > > Renathvagen 2
> > > > > > SE 121 37 Johanneshov
> > > > > > SWEDEN
> > > > > > +46 8 556 002 03
> > > > > > Mob: +46 708 204 288
> > > > > > www.omnitor.se
> > > > > > Gunnar.Hellstrom@Omnitor.se
> > > > > > --------------------------------------------
> > > > > >
> > > > > >
> > > > > > -----Original Message-----
> > > > > > From: sipping-admin@ietf.org
> > > > > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > > > > A vWijk
> > > > > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > > > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > > henry.sinnreich@mci.com;
> > > > > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > > > > gv@trace.wisc.edu
> > > > > > Subject: Betr.: [Sipping] ToIP Requirements
> > > > > >
> > > > > >
> > > > > > Hello Gonzalo,
> > > > > > While there are device requirements in the text. Those
> > > > > > requirements and the
> > > > > > explanation on the different analogue text telephone =
systems
> > > > > > is included to
> > > > > > give the reader an understanding what kind of devices will
> > > > > > connect with
> > > > > > ToIP. And the need for the text gateways. And what the
> > > > > > different behaviour
> > > > > > can/may be compared to end to end IP devices.
> > > > > > The most important part will be the ToIP framwork and the
> > > > > > text gateways and
> > > > > > how such gateways will be used and work with the ToIP
> > framework.
> > > > > >
> > > > > > There is a ToIP framework where a terminal uses streaming
> > > > > > text as defined in
> > > > > > RFC2793 to connect with any other Interactive
> > > > > > texting/real-time streaming
> > > > > > text device.
> > > > > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > > > > established with
> > > > > > SIP. But in other cases, a text gateway must be=20
> used to allow
> > > > > > T.140/RTP to
> > > > > > connect with a non IP text telephony device.
> > > > > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > > > > text gateway to
> > > > > > be able to understand RFC2793 text. (PSTN-IP)
> > > > > > It can be a mobile phone running GPRS that uses the text
> > > > gateway to
> > > > > > understand RFC2793 text. (this exist right NOW=20
> using SIP! And
> > > > > > those gateways
> > > > > > will be used in mobile networks this year!)
> > > > > > Or a mobile text telephone using CTM via the text gateway =
to
> > > > > > communicate
> > > > > > with RFC2793 text.
> > > > > > Or even one analogue text telephone connecting to another
> > > > > > analogue text
> > > > > > telephone in a different country, and the PSTN's use IP to
> > > > > > connect the call.
> > > > > >
> > > > > > The question is for example, how do we use that gateway =
from
> > > > > > the IP side? Is
> > > > > > it invisible for the IP side? Or do we need special
> > > > > > extensions in SIP to
> > > > > > that the IP terminal knows it is communicating with the
> > > > > > end-device through
> > > > > > the text gateway?
> > > > > > There may be a slightly different call setup flow=20
> when a text
> > > > > > gateway is
> > > > > > included.
> > > > > > Or additional SIP messages indicating gateway activity. =
When
> > > > > > you connect for
> > > > > > example to an analogue text telephone with an=20
> RFC2793 capable
> > > > > > IP terminal
> > > > > > the capabilities will be less then IP to IP=20
> terminal directly
> > > > > > (different
> > > > > > transmission speeds, half/full duplex etc).
> > > > > > Or what to do with emergency calls that are in RFC
> > 2793 text only?
> > > > > >
> > > > > > That are requirements that will be defined and worked out =
in
> > > > > > this draft.
> > > > > > Also, we want to make everybody in SIP and SIPPING aware
> > > > > > about those text
> > > > > > gateways. So that complicated matters like how to send
> > > > analogue text
> > > > > > telephone text through a PSTN and connect with a SIP
> > > > > > telephone using RFC2793
> > > > > > interactive texting. When using the text gateway, there is
> > > > > > then no need for
> > > > > > RFC 2833 events at the PSTN for text telephones.
> > > > > >
> > > > > > So, yes. I believe that this should be in the=20
> SIPPING WG. And
> > > > > > it will fit in
> > > > > > the charter - support for hearing-/speech-impaired calling.
> > > > > >
> > > > > > greetz
> > > > > >
> > > > > > Arnoud
> > > > > >
> > > > > >
> > > > > >
> > > > > > Drs. Arnoud A. T. van Wijk
> > > > > > Viataal
> > > > > > Research & Development
> > > > > > Afdeling RDS
> > > > > > Theerestraat 42
> > > > > > 5271 GD Sint-Michielsgestel
> > > > > > The Netherlands.
> > > > > > Mobile: +31651921948
> > > > > > International text telephone: +31735588408
> > > > > >
> > > > > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > > > > 24-02-04 10:21 >>>
> > > > > > Hello,
> > > > > >
> > > > > > the draft below provides a set of requirements for =
real-time
> > > > > > conversations using SIP.
> > > > > >=20
> http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > > > > ip-01.txt
> > > > >
> > > > > Nevertheless, I have found a lot of device requirements in
> > > > > the text, but
> > > > > I fail to understand which particular protocol requirements
> > > > we need to
> > > > > fulfill with new protocol extensions.
> > > > >
> > > > > While SIPPING is the right place to handle requirements
> > > > that will lead
> > > > > to new SIP protocol extensions, SIPPING may not be the
> > > > right place to
> > > > > deal with device requirements (e.g., text phones MUST have a
> > > > > display of
> > > > > at least 100 characters).
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Gonzalo
> > > > >
> > > > >
> > > > > This communication is confidential and intended solely for
> > > > > the addressee(s).
> > > > > Any unauthorized review, use, disclosure or distribution is
> > > > > prohibited. If
> > > > > you believe this message has been sent to you in error,
> > > > > please notify the
> > > > > sender by replying to this transmission and delete the
> > > > message without
> > > > > disclosing it. Thank you.
> > > > >
> > > > > E-mail including attachments is susceptible to data=20
> corruption,
> > > > > interruption, unauthorized amendment, tampering and viruses,
> > > > > and we only
> > > > > send and receive e-mails on the basis that we are not liable
> > > > > for any such
> > > > > corruption, interception, amendment, tampering or=20
> viruses or any
> > > > > consequences thereof.
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > >
> > > >
> > > >
> > >
> >
> >
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20
>=20
>=20

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



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Date: Mon, 01 Mar 2004 05:47:16 +0100
From: "A vWijk" <A.vWijk@viataal.nl>
To: <barry.dingle@bigfoot.com.au>, <Gonzalo.Camarillo@ericsson.com>,
        <sipping@ietf.org>, <Brian.Rosen@marconi.com>,
        <gunnar.hellstrom@omnitor.se>
Cc: <len.bytheway@aceinfo.net.au>, <rrroy@att.com>, <henry.sinnreich@mci.com>,
        <paulej@packetizer.com>, <gv@trace.wisc.edu>
Subject: Betr.: RE: Betr.: [Sipping] ToIP Requirements
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And that is why those 2 solutions would be best IMHO:
1. using a prefix number when using a textphone.
This prefix is for every textphone call you place! Thus not emergency only =
but any call made by the textphone.
600-911, 600-555-123456, 600-800-cheapshoe.

2. Label the phone number you use for the textphone as text telephone =
user.

In both cases the PSTN will immediately invoke the text gateway. Which =
then sends the V.18 reception/boudot reception signalling.

I know that it is not perfect, but the analogue textphones will disappear =
quickly. 2-3 years for most users and up to 10-15 years for the last few =
users.

I and many deaf people I know will throw the analogue textphone out of the =
window as soon as possible.
We will use mobile textphones via the textgateway and SIP phones with =
RFC2793 (and our pcs with an interactive text client).

greetz

Arnoud


<<< "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se> 01-03 03:27  >>>
1. How it is in PSTN:
A calling V.18 sends a specific data pattern for text telephony in the V.8
signal CI.
It is sent in the connected voice channel.
So, as with all calls from PSTN, there is no indication before the call is
set up about what kind of media will be used.

But it is far more easy to identify it as a textphone call than it is for
some of the old systems. E.G the US system "Baudot" is silent when =
calling,
and the answering side must guess that it is a textphone.

There is not yet any really clear trend to move to V.18 for users in USA.
Therefore I agree that the mechanisms for emergency calls must for a long
time accomplish for the tedious task to try to respond in text on PSTN to
all incoming silent calls to discriminate the text callers. That response
should be done with V.18 signals, so that it at least is prepared for more
convenient handling when people have moved to using V.18 textphones.

2. In SIP.
In SIP, we have text in a separate media stream in the call. An emergency
centre should have capability to use both voice and text simultaneously.
There are mechanisms in discussion about indicating preference for =
handling
the call with text as the main medium. I am not sure that we can require
that preference to be set in a SIP terminal used for an emergency call by =
a
text depending person.
So, the safe handling would be for the emergency centre to offer text
capability, and if text connection is achieved, an automatic text greeting
is sent in that channel, while the emergency agent responds in voice. If =
the
caller then starts using the text channel, it is evident enough that text =
is
preferred.


Gunnar

-------------------------------------------
Gunnar Hellstr=F6m
Omnitor AB
Renathv=E4gen 2
SE 121 37 Johanneshov
SWEDEN
+46 8 556 002 03
Mob: +46 708 204 288
www.omnitor.se
Gunnar.Hellstrom@Omnitor.se
--------------------------------------------


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Rosen, Brian
Sent: Monday, March 01, 2004 1:16 AM
To: Barry Dingle; 'Arnoud van Wijk'; 'Gunnar Hellstrom';
Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
gv@trace.wisc.edu; Len Bytheway
Subject: RE: Betr.: [Sipping] ToIP Requirements


While we can, and have, specified a single worldwide URI
for emergency calls, I believe it is not even technically
possible to find a single 3 digit number that is not
already in use somewhere.  Thus, I suspect anyone
dreaming that such a thing will happen probably is
going to be disappointed.

Since existing TDD devices do not provide an indication
that they are present until after the call is completed,
I think that we should not even consider use of any
kind of gateway for emergency calls.  This is true of
V.18 devices also, correct?

Having a special number for text emergency calls does
have

_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



