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Folks,

Question for clarification:

In the discussions here on how to populate sip URI in Infrastructure=20
ENUM to derive the Call Routing data for SPEERMINT  two
lines of thought exist:

The Enumservice SIP in Infrastructure ENUM should contain=20
1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
2. the ingress point into the provider network:
e.g. sip:+4319793321@sbc4711.provB.net

The originating UAC has in this case set the To: field to=20
tel:+4319793321 and the Request-URI to e.g.=20
sip:+4319793321provA.net;user=3Dphone. This caused the
ENUM query at the proxy of provider A

The question is how the proxy of provider A is now
setting up the request URI in cases mentioned above?

Does the UAS from provider B accept calls to sbc7411.provB.net,
or must the request URI set to the proper AoR as described in
8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.

I would also like to draw the attention to section 3 of
RFC3761 (SIP Enumservice) explaining the usage of AoR in
ENUM.

Regards

-sta


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> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 09:06
> To: speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Enum] How to correctly set up the request URI?
>=20
> Folks,
>=20
> Question for clarification:
>=20
> In the discussions here on how to populate sip URI in Infrastructure
> ENUM to derive the Call Routing data for SPEERMINT  two
> lines of thought exist:
>=20
> The Enumservice SIP in Infrastructure ENUM should contain
> 1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
> 2. the ingress point into the provider network:
> e.g. sip:+4319793321@sbc4711.provB.net
>=20
> The originating UAC has in this case set the To: field to
> tel:+4319793321 and the Request-URI to e.g.
> sip:+4319793321provA.net;user=3Dphone. This caused the
> ENUM query at the proxy of provider A
>=20
> The question is how the proxy of provider A is now
> setting up the request URI in cases mentioned above?
>=20
> Does the UAS from provider B accept calls to sbc7411.provB.net,
> or must the request URI set to the proper AoR as described in
> 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.

[Tom Creighton wrote:]=20
>From section 8.2.2.1 of RFC3261:	"If the Request-URI does not
identify an address that the UAS is willing to accept requests for, it
SHOULD reject the request with a 404 (Not Found) response."

As long as the UAS from provider B is "willing" to accept the request,
either should work.  Since provider B is populating Infrastructure ENUM,
one would hope that they would coordinate this data with the
configuration of their proxies.

Besides, how do you actually know that sbc4441.provB.net is actually a
hostname and isn't a subdomain?

In either case, after the provider A proxy performs an ENUM query and
rewrites the request URI, it should behave as a UAC and follow the
procedures outlined in Section 4 of RFC3263 to determine the IP address,
port, and transport protocol for the provider B UAS.

>=20
> I would also like to draw the attention to section 3 of
> RFC3761 (SIP Enumservice) explaining the usage of AoR in
> ENUM.
>=20
> Regards
>=20
> -sta
>=20
>=20
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum

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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
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Yes .. private ENUM would be in scope for the ENUM WG and in fact it already
is since the PSTN data draft that has now finished IETF last call
specifically deals with domains outside the scope of e164.apra visible on
invisible to the DNS


> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Tuesday, May 30, 2006 11:32 AM
> To: Andrew Newton
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
> 
> Andy wrote
> 
> >Stastny Richard wrote:
> >>
> >> IMHO SPEERMINT does not deal with E.164 numbers at all in routing, it
> is out of scope
> 
> >Would the "private" ENUM tree run by a peering point and specifying its
> >location (i.e. DNS suffix) be in scope?
> 
> IMHO yes, because any ENUM is dealing with E.164 numbers
> 
> -sta
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint


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From speermint-bounces@ietf.org Thu Jun 01 11:25:25 2006
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WG Update

As we look ahead to Montreal, Dave and I wanted to share with the WG
what our current objectives look like and what the draft agenda for
Montreal is.

*Please send us your feedback and suggest additional agenda items as
appropriate.*

For reference, see the WG charter at
http://www.ietf.org/html.charters/speermint-charter.html

Proposed WG agenda for Montreal:

0 - Agenda Bashing

1 - Terminology I-D review.

2 - Requirements I-D review. (+ Possible selected Use Case I-D reviews.)

3 - Architecture I-D review.

4 - Review Next Steps


>From Goals & Milestones: (My additional comments preceded by "JL-->".)
------------------------

1 - Jul 2006 Submit SPEERMINT terminology I-D

JL --> This draft is still in-process. We expect to present a reasonably
stable draft in Montreal and move to WGLC shortly afterwards.

*Please review this document again and submit your comments.* =20
http://www.ietf.org/internet-drafts/draft-ietf-speermint-terminology-00.
txt


2 - Sep 2006 Submit I-D defining the SPEERMINT routing architecture

JL --> Plan TBD.  (We expect to discuss our approach with key
contributors in Montreal.)


3 - Dec 2006 Submit I-D defining the message flows associated with the
SPEERMINT routing architecture

JL --> Plan TBD. It is likely that this document will mature out of one
or more of the use case I-Ds we hope to solicit (*see 2nd comment in #5
below*).


4 - Jan 2007 Submit I-D on the use of DNS SRV and NAPTR records as
specified by RFC 3263

JL --> Plan TBD. Same parenthetical note as in #3 above.


5 - Mar 2007 Submit I-D on the minimum set of requirements for SIP-based
VoIP interconnection

JL --> This document will be of major importance to the WG.
Jean-Francois Mule will be editing this WG document for us and leading
the effort.  We hope to see a basic -00 draft for Montreal, perhaps at
least carrying over the requirements that were split out of Dave's
earlier Requirements & Terminology draft.

JL --> Also *VERY* important to filling out the requirements draft (and
the architecture draft for that matter) are use cases.  The goal is to
solicit use case I-Ds.  These should be informational, as personal I-Ds
for the WG from people from different geographies, and with different
technologies.  Those use cases will need to comply with the high-level
requirements of course. However, in order to fully document all
requirements, these application-specific or industry-specific use cases
will be needed. We expect these will necessarily be concerned with and
constrained to a specific industry, market geography, or deployment
technology, so that they need not be all-encompassing or overly complex.
We would expect some of these uses cases to indicate a need for
additions to the requirements draft (based on WG consensus). And these
use cases will also share a large set of common requirements which we
can confirm as especially high priority, for example.=20

*Please email Dave and I if you may be interested in submitting use
cases.*


Thanks!!

Dave & Jason

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From speermint-bounces@ietf.org Thu Jun 01 11:26:19 2006
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Thank you, Tom

just a minor additional question:

> In either case, after the provider A proxy performs an ENUM query and
> rewrites the request URI, it should behave as a UAC and follow the
> procedures outlined in Section 4 of RFC3263 to determine the IP
address,
> port, and transport protocol for the provider B UAS.

If I have to do this according to RFC3263 anyway, why put
the sbc4711.provB.net into ENUM in the first place?

-sta


> -----Original Message-----
> From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Sent: Thursday, June 01, 2006 4:36 PM
> To: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>=20
>=20
>=20
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 09:06
> > To: speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Enum] How to correctly set up the request URI?
> >
> > Folks,
> >
> > Question for clarification:
> >
> > In the discussions here on how to populate sip URI in Infrastructure
> > ENUM to derive the Call Routing data for SPEERMINT  two
> > lines of thought exist:
> >
> > The Enumservice SIP in Infrastructure ENUM should contain
> > 1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
> > 2. the ingress point into the provider network:
> > e.g. sip:+4319793321@sbc4711.provB.net
> >
> > The originating UAC has in this case set the To: field to
> > tel:+4319793321 and the Request-URI to e.g.
> > sip:+4319793321provA.net;user=3Dphone. This caused the
> > ENUM query at the proxy of provider A
> >
> > The question is how the proxy of provider A is now
> > setting up the request URI in cases mentioned above?
> >
> > Does the UAS from provider B accept calls to sbc7411.provB.net,
> > or must the request URI set to the proper AoR as described in
> > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
>=20
> [Tom Creighton wrote:]
> From section 8.2.2.1 of RFC3261:	"If the Request-URI does not
> identify an address that the UAS is willing to accept requests for, it
> SHOULD reject the request with a 404 (Not Found) response."
>=20
> As long as the UAS from provider B is "willing" to accept the request,
> either should work.  Since provider B is populating Infrastructure
ENUM,
> one would hope that they would coordinate this data with the
> configuration of their proxies.
>=20
> Besides, how do you actually know that sbc4441.provB.net is actually a
> hostname and isn't a subdomain?
>=20
> In either case, after the provider A proxy performs an ENUM query and
> rewrites the request URI, it should behave as a UAC and follow the
> procedures outlined in Section 4 of RFC3263 to determine the IP
address,
> port, and transport protocol for the provider B UAS.
>=20
> >
> > I would also like to draw the attention to section 3 of
> > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> > ENUM.
> >
> > Regards
> >
> > -sta
> >
> >
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum

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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	<speermint@ietf.org>
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Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>=20
> Thank you, Tom
>=20
> just a minor additional question:
>=20
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
>=20
> If I have to do this according to RFC3263 anyway, why put
> the sbc4711.provB.net into ENUM in the first place?
>=20
> -sta
>=20
>=20
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two
> > > lines of thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain
> > > 1. the address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=3Dphone. This caused the
> > > ENUM query at the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now
> > > setting up the request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net,
> > > or must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:	"If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> > > ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum

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From speermint-bounces@ietf.org Thu Jun 01 12:01:19 2006
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So, the discovery of an ENUM service for an opportunistic peering 
exchange would be worked out in the ENUM working group, but all other 
aspects of an opportunistic peering exchange are to be worked out here. 
  Seems kinda weird.

-andy

Richard Shockey wrote:
> Yes .. private ENUM would be in scope for the ENUM WG and in fact it already
> is since the PSTN data draft that has now finished IETF last call
> specifically deals with domains outside the scope of e164.apra visible on
> invisible to the DNS
> 
> 
>> -----Original Message-----
>> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
>> Sent: Tuesday, May 30, 2006 11:32 AM
>> To: Andrew Newton
>> Cc: speermint@ietf.org
>> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>>
>> Andy wrote
>>
>>> Stastny Richard wrote:
>>>> IMHO SPEERMINT does not deal with E.164 numbers at all in routing, it
>> is out of scope
>>
>>> Would the "private" ENUM tree run by a peering point and specifying its
>>> location (i.e. DNS suffix) be in scope?
>> IMHO yes, because any ENUM is dealing with E.164 numbers



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From speermint-bounces@ietf.org Thu Jun 01 13:11:34 2006
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I see,
=20
but how do you solve this problem if the users can also be reached by =
AoRs direct?
=20
-sta

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 17:57
An: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?



Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>
> Thank you, Tom
>
> just a minor additional question:
>
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
>
> If I have to do this according to RFC3263 anyway, why put
> the sbc4711.provB.net into ENUM in the first place?
>
> -sta
>
>
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two
> > > lines of thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain
> > > 1. the address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=3Dphone. This caused the
> > > ENUM query at the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now
> > > setting up the request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net,
> > > or must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> > > ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum



_______________________________________________
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From: "Richard Shockey" <richard@shockey.us>
To: "'Andrew Newton'" <andy@hxr.us>
Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Thu, 1 Jun 2006 13:31:47 -0400
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No .. its not weird .. all you have to do is define the enumservice
registration E2U+"foo" with "foo" defined in Speermint

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, June 01, 2006 12:01 PM
> To: richard@shockey.us
> Cc: 'Stastny Richard'; speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
> 
> So, the discovery of an ENUM service for an opportunistic peering
> exchange would be worked out in the ENUM working group, but all other
> aspects of an opportunistic peering exchange are to be worked out here.
>   Seems kinda weird.
> 
> -andy
> 
> Richard Shockey wrote:
> > Yes .. private ENUM would be in scope for the ENUM WG and in fact it
> already
> > is since the PSTN data draft that has now finished IETF last call
> > specifically deals with domains outside the scope of e164.apra visible
> on
> > invisible to the DNS
> >
> >
> >> -----Original Message-----
> >> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> >> Sent: Tuesday, May 30, 2006 11:32 AM
> >> To: Andrew Newton
> >> Cc: speermint@ietf.org
> >> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
> >>
> >> Andy wrote
> >>
> >>> Stastny Richard wrote:
> >>>> IMHO SPEERMINT does not deal with E.164 numbers at all in routing, it
> >> is out of scope
> >>
> >>> Would the "private" ENUM tree run by a peering point and specifying
> its
> >>> location (i.e. DNS suffix) be in scope?
> >> IMHO yes, because any ENUM is dealing with E.164 numbers


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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 13:34:24 -0400
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From: "Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <speermint@ietf.org>
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In this case, I guess when the originating proxy resolves the domain
part of the AOR (sip:+4319793321@provB.net), it will return the IP
address of the root domain "provB.net" which may be a redirect proxy. So
"provB.net" has two choices:

(1) It accesses its local routing table and sends a 305 to the
originating proxy with "sbc4711.provB.net" in the contact header.

(2) It replaces the domain part in the r-uri to more specific like
"proxy.austria.provB.net" and forward the message to the next-hop.

Either case, the assumption is provB.net will have a way to find out
where the home proxy of user +4319793321.

This is only my guess.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 1:13 PM
To: Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Subject: [Speermint] Re: [Enum] How to correctly set up the request URI?

I see,
=20
but how do you solve this problem if the users can also be reached by
AoRs direct?
=20
-sta

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 17:57
An: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?



Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>
> Thank you, Tom
>
> just a minor additional question:
>
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the=20
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
>
> If I have to do this according to RFC3263 anyway, why put the=20
> sbc4711.provB.net into ENUM in the first place?
>
> -sta
>
>
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two lines of=20
> > > thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain 1. the=20
> > > address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM query =
at

> > > the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now setting up the=20
> > > request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net, or

> > > must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the=20
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the=20
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum



_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
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Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 13:45:40 -0400
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "Lee, Yiu" <Yiu_Lee@cable.comcast.com>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>, <speermint@ietf.org>
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Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.=20

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>=20
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>=20
> (2) It replaces the domain part in the r-uri to more specific like
> "proxy.austria.provB.net" and forward the message to the next-hop.
>=20
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>=20
> This is only my guess.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>=20
> I see,
>=20
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>=20
> -sta
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>=20
>=20
>=20
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a
> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress
> point.  This could very well save a service provider a lot of money in
> route proxy equipment although might cost more on the provisioning
side.
>=20
> That is my guess.
>=20
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the
> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of
> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the
> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the
> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 20:06:48 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>,
	"Lee, Yiu" <Yiu_Lee@cable.comcast.com>, <speermint@ietf.org>
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so basically the IMS I-CSCF -  HSS - S-CSCF way?
=20
Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a
> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress
> point.  This could very well save a service provider a lot of money in
> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the
> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of
> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the
> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the
> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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



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Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Thu, 1 Jun 2006 20:08:42 +0200
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	"Andrew Newton" <andy@hxr.us>
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Maybe it is me, but I am not quite able to follow this discussion:
I am a bit confused

Question for clarification
=20
What is discovered in which ENUM?
the E.164 to URI mapping club or the SIP peering club?
=20
-sta

________________________________

Von: Richard Shockey [mailto:richard@shockey.us]
Gesendet: Do 01.06.2006 19:31
An: 'Andrew Newton'
Cc: Stastny Richard; speermint@ietf.org
Betreff: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture



No .. its not weird .. all you have to do is define the enumservice
registration E2U+"foo" with "foo" defined in Speermint

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Thursday, June 01, 2006 12:01 PM
> To: richard@shockey.us
> Cc: 'Stastny Richard'; speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>
> So, the discovery of an ENUM service for an opportunistic peering
> exchange would be worked out in the ENUM working group, but all other
> aspects of an opportunistic peering exchange are to be worked out =
here.
>   Seems kinda weird.
>
> -andy
>
> Richard Shockey wrote:
> > Yes .. private ENUM would be in scope for the ENUM WG and in fact it
> already
> > is since the PSTN data draft that has now finished IETF last call
> > specifically deals with domains outside the scope of e164.apra =
visible
> on
> > invisible to the DNS
> >
> >
> >> -----Original Message-----
> >> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> >> Sent: Tuesday, May 30, 2006 11:32 AM
> >> To: Andrew Newton
> >> Cc: speermint@ietf.org
> >> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering =
Architecture
> >>
> >> Andy wrote
> >>
> >>> Stastny Richard wrote:
> >>>> IMHO SPEERMINT does not deal with E.164 numbers at all in =
routing, it
> >> is out of scope
> >>
> >>> Would the "private" ENUM tree run by a peering point and =
specifying
> its
> >>> location (i.e. DNS suffix) be in scope?
> >> IMHO yes, because any ENUM is dealing with E.164 numbers




_______________________________________________
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 14:10:06 -0400
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From: "Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>
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Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.=20

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?
=20
Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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



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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
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Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the
mapping to different hosting proxies in two places
(in this case ENUM and the HSS), if I can do it only
in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint





_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



From speermint-bounces@ietf.org Thu Jun 01 14:18:37 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Creighton, Tom'" <Tom_Creighton@cable.comcast.com>,
	"'Stastny Richard'" <Richard.Stastny@oefeg.at>, <speermint@ietf.org>
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Subject: [Speermint] RE: [Enum] How to correctly set up the request URI?
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I think this is not correct.

If the AoR is provB.net, then Provider B has technology to route incoming
calls addressed to provB.net to the sbc4711.provB.net, and it will use its
existing mechanism to do so.  The same AoR would be used for on-net as well
as off-net routing and the same routing path would be used.  The same
routing proxies would be used.  Capacity issues shouldn't be an issue with
this technology.

Brian

-----Original Message-----
From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com] 
Sent: Thursday, June 01, 2006 11:58 AM
To: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Enum] How to correctly set up the request URI?

Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
> 
> Thank you, Tom
> 
> just a minor additional question:
> 
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> 
> If I have to do this according to RFC3263 anyway, why put
> the sbc4711.provB.net into ENUM in the first place?
> 
> -sta
> 
> 
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two
> > > lines of thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain
> > > 1. the address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=phone. This caused the
> > > ENUM query at the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now
> > > setting up the request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net,
> > > or must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:	"If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> > > ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum

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


_______________________________________________
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 14:22:10 -0400
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From: "Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <speermint@ietf.org>
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Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint





_______________________________________________
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From speermint-bounces@ietf.org Thu Jun 01 14:41:45 2006
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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 20:43:15 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <speermint@ietf.org>
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Just out of curiosity:
=20
(2) It replaces the domain part in the r-uri to more specific like
"proxy.austria.provB.net" and forward the message to the next-hop.

Do I have to replace the request URI with "proxy.austria.provB.net,
or could I just leave it to "provB.net", assuming that =
"proxy.austria.provB.net
accepts "provB.net"?
=20
-sta

________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 19:34
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



In this case, I guess when the originating proxy resolves the domain
part of the AOR (sip:+4319793321@provB.net), it will return the IP
address of the root domain "provB.net" which may be a redirect proxy. So
"provB.net" has two choices:

(1) It accesses its local routing table and sends a 305 to the
originating proxy with "sbc4711.provB.net" in the contact header.

(2) It replaces the domain part in the r-uri to more specific like
"proxy.austria.provB.net" and forward the message to the next-hop.

Either case, the assumption is provB.net will have a way to find out
where the home proxy of user +4319793321.

This is only my guess.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 1:13 PM
To: Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Subject: [Speermint] Re: [Enum] How to correctly set up the request URI?

I see,

but how do you solve this problem if the users can also be reached by
AoRs direct?

-sta

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 17:57
An: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?



Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>
> Thank you, Tom
>
> just a minor additional question:
>
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
>
> If I have to do this according to RFC3263 anyway, why put the
> sbc4711.provB.net into ENUM in the first place?
>
> -sta
>
>
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two lines of
> > > thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain 1. the
> > > address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM query =
at

> > > the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now setting up the
> > > request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net, or

> > > must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum



_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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From speermint-bounces@ietf.org Thu Jun 01 14:50:34 2006
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Lee, Yiu" <Yiu_Lee@cable.comcast.com>, <speermint@ietf.org>
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Provider B could be using an internal ENUM database, an HSS-like
database, an LDAP database, or even static route table.  I was merely
using this as an example and certainly did not mean to imply or suggest
that provider B was or had to follow any particular religion. ;-)

Another way to look at it would be the case of a city government in the
U.S. that acted as its own PSTN provider and wanted to use its .us TLD
to provision its records into Infrastructure ENUM.

For example:

1. The originating UAC has the To: field set to tel:+12155551234 and the
Request-URI to sip:+12155551234@provA.net;user=3Dphone.

2. The provider A proxy performs a query to Infrastructure ENUM.

3. Infrastructure ENUM returns the following AoR:
sip:+12155551234@langhorne.pa.us

In this fictitious example, the entity that manages pa.us sub-delegated
langhorne.pa.us to the Township of Langhorne which gives them full
administrative and technical control of their domain.  They in turn
provision RFC3263 compliant NAPTR and SRV records that point to their
SIP proxy. =20

I am just trying to point out that it is impossible to determine the
difference between a hostname and a domain name if a provider's ingress
sip proxy is configured to accept what is provisioned on the right side
of the @ in Infrastructure ENUM.

Tom

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 14:07
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>=20
> Richard
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>=20
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
> > point.  This could very well save a service provider a lot of money
in
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20


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From speermint-bounces@ietf.org Thu Jun 01 15:19:05 2006
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Thu, 1 Jun 2006 15:18:35 -0400
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From: "Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <speermint@ietf.org>
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No, you don't have to. All I try to say is provB is responsible to route
the call to the user. If they decide to modify the r-uri to help the
downstream proxies, they can. If they decide to put the route set in the
header, they also can. This is local routing decision to provB. The
bottom line is the request will arrive in the user home proxy to replace
the user aor to the registered contact address to reach the user's
device.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:43 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Just out of curiosity:
=20
(2) It replaces the domain part in the r-uri to more specific like
"proxy.austria.provB.net" and forward the message to the next-hop.

Do I have to replace the request URI with "proxy.austria.provB.net, or
could I just leave it to "provB.net", assuming that
"proxy.austria.provB.net accepts "provB.net"?
=20
-sta

________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 19:34
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



In this case, I guess when the originating proxy resolves the domain
part of the AOR (sip:+4319793321@provB.net), it will return the IP
address of the root domain "provB.net" which may be a redirect proxy. So
"provB.net" has two choices:

(1) It accesses its local routing table and sends a 305 to the
originating proxy with "sbc4711.provB.net" in the contact header.

(2) It replaces the domain part in the r-uri to more specific like
"proxy.austria.provB.net" and forward the message to the next-hop.

Either case, the assumption is provB.net will have a way to find out
where the home proxy of user +4319793321.

This is only my guess.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 1:13 PM
To: Creighton, Tom; speermint@ietf.org
Cc: enum@ietf.org
Subject: [Speermint] Re: [Enum] How to correctly set up the request URI?

I see,

but how do you solve this problem if the users can also be reached by
AoRs direct?

-sta

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 17:57
An: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?



Maybe the service provider has some sort of segregated network
(physically or logically) and wants/needs to associate each its
subscribers with a particular network segment.  If the provider went
with the flat model of provB.net, each of its proxies would have to
accept requests destined to all of its customers and would then have to
route the request to the appropriate region or division of its network.
If the provider went with the hierarchical approach of
sbc4711.provB.net, all of the requests would already be routed to the
appropriate ingress point.  This could very well save a service provider
a lot of money in route proxy equipment although might cost more on the
provisioning side.

That is my guess.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 11:30
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Enum] How to correctly set up the request URI?
>
> Thank you, Tom
>
> just a minor additional question:
>
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the=20
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
>
> If I have to do this according to RFC3263 anyway, why put the=20
> sbc4711.provB.net into ENUM in the first place?
>
> -sta
>
>
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Thursday, June 01, 2006 4:36 PM
> > To: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 09:06
> > > To: speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Enum] How to correctly set up the request URI?
> > >
> > > Folks,
> > >
> > > Question for clarification:
> > >
> > > In the discussions here on how to populate sip URI in
Infrastructure
> > > ENUM to derive the Call Routing data for SPEERMINT  two lines of=20
> > > thought exist:
> > >
> > > The Enumservice SIP in Infrastructure ENUM should contain 1. the=20
> > > address-of-record of the user: e.g.
sip:+4319793321@provB.net
> > > 2. the ingress point into the provider network:
> > > e.g. sip:+4319793321@sbc4711.provB.net
> > >
> > > The originating UAC has in this case set the To: field to
> > > tel:+4319793321 and the Request-URI to e.g.
> > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM query =
at

> > > the proxy of provider A
> > >
> > > The question is how the proxy of provider A is now setting up the=20
> > > request URI in cases mentioned above?
> > >
> > > Does the UAS from provider B accept calls to sbc7411.provB.net, or

> > > must the request URI set to the proper AoR as described in
> > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > [Tom Creighton wrote:]
> > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > identify an address that the UAS is willing to accept requests for,
it
> > SHOULD reject the request with a 404 (Not Found) response."
> >
> > As long as the UAS from provider B is "willing" to accept the
request,
> > either should work.  Since provider B is populating Infrastructure
> ENUM,
> > one would hope that they would coordinate this data with the=20
> > configuration of their proxies.
> >
> > Besides, how do you actually know that sbc4441.provB.net is actually
a
> > hostname and isn't a subdomain?
> >
> > In either case, after the provider A proxy performs an ENUM query
and
> > rewrites the request URI, it should behave as a UAC and follow the=20
> > procedures outlined in Section 4 of RFC3263 to determine the IP
> address,
> > port, and transport protocol for the provider B UAS.
> >
> > >
> > > I would also like to draw the attention to section 3 of
> > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > >
> > > Regards
> > >
> > > -sta
> > >
> > >
> > > _______________________________________________
> > > enum mailing list
> > > enum@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/enum



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Stastny Richard wrote:
> Maybe it is me, but I am not quite able to follow this discussion:
> I am a bit confused
> 
> Question for clarification
>  
> What is discovered in which ENUM?
> the E.164 to URI mapping club or the SIP peering club?

I don't know of a single VoIP peering exchange that uses the e164.arpa. 
ENUM tree.  They might exist, but I've not seen them.  If ever there 
were to be opportunistic peering exchanges, there would need to be 
discovery of the ENUM tree for that peering exchange so that peers 
within the exchange can use the E.164 number namespace.

-andy


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From speermint-bounces@ietf.org Thu Jun 01 15:54:33 2006
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I have an I-D for the ENUM WG in the pipe which may solve your problem,
I will submit it after the weekend
=20
I am not quite sure what you mean with "opportunistic"
peering exchanges
=20
-sta
=20
=20

________________________________

Von: Andrew Newton [mailto:andy@hxr.us]
Gesendet: Do 01.06.2006 21:43
An: Stastny Richard
Cc: richard@shockey.us; speermint@ietf.org
Betreff: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture



Stastny Richard wrote:
> Maybe it is me, but I am not quite able to follow this discussion:
> I am a bit confused
>
> Question for clarification
>=20
> What is discovered in which ENUM?
> the E.164 to URI mapping club or the SIP peering club?

I don't know of a single VoIP peering exchange that uses the e164.arpa.
ENUM tree.  They might exist, but I've not seen them.  If ever there
were to be opportunistic peering exchanges, there would need to be
discovery of the ENUM tree for that peering exchange so that peers
within the exchange can use the E.164 number namespace.

-andy




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From speermint-bounces@ietf.org Thu Jun 01 18:00:37 2006
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Stastny Richard wrote:
>  
> I am not quite sure what you mean with "opportunistic"
> peering exchanges

I was using the term to mean a peering exchange (or federation or 
whatever you want to call them) that can be joined dynamically.

-andy


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>I was using the term to mean a peering exchange (or federation or
>whatever you want to call them) that can be joined dynamically.


On what terms? I still do nor understand. Joining a federation
always involves some legal negotiotions.
=20
In addition, a "federation" to do ENUM (i.e. E.164 to URI mapping =3D
finding out the Carrier-of-Record) and The "federation" to peer VoIP
traffic may be identical, but need not to be.

As I already pointed out on this list, it is still the choice of the
originating service provider to select the peering fabric
=20
-sta


________________________________

Von: Andrew Newton [mailto:andy@hxr.us]
Gesendet: Fr 02.06.2006 00:00
An: Stastny Richard
Cc: richard@shockey.us; speermint@ietf.org
Betreff: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture



Stastny Richard wrote:
>=20
> I am not quite sure what you mean with "opportunistic"
> peering exchanges

I was using the term to mean a peering exchange (or federation or
whatever you want to call them) that can be joined dynamically.

-andy




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On Jun 1, 2006, at 6:14 PM, Stastny Richard wrote:

>> I was using the term to mean a peering exchange (or federation or
>> whatever you want to call them) that can be joined dynamically.
>
>
> On what terms? I still do nor understand. Joining a federation
> always involves some legal negotiotions.

Well, they require an agreement.  The legality of that agreement is  
another matter.

If the execution of that agreement requires signed contracts, etc...  
all the configuration information for joining that federation does  
not need to be in machine readable form.  Indeed, much of it will not  
be given until an NDA is signed or a service contract is signed, so  
it won't even be available on the web.

If, however, there is intention to have the policies on which the  
peering federation agreement operates to be machine readable, such as  
in the DDDS application proposed by Otmar, then this seems to imply  
that joining the federation is more opportunistic in nature.

> In addition, a "federation" to do ENUM (i.e. E.164 to URI mapping =
> finding out the Carrier-of-Record) and The "federation" to peer VoIP
> traffic may be identical, but need not to be.

I guess that is one way to look at it.  I've always thought that an  
ENUM service was just a feature or service of a particular  
federation.  After all, many of the commercial ones today have  
mechanisms other than ENUM to convert E.164 numbers to URIs.  And it  
is possible that a federation may have multiple namespace management  
services as E.164 is but just one type of namespace.  A federation  
may decide to offer namespace management for ICQ numbers -- in  
addition to E.164.

> As I already pointed out on this list, it is still the choice of the
> originating service provider to select the peering fabric

Yes.  And to do it today requires no new protocols or work.

-andy

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From: Otmar Lendl <lendl@nic.at>
To: Andrew Newton <andy@hxr.us>
Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
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On 2006/06/02 02:06, Andrew Newton <andy@hxr.us> wrote:
> On Jun 1, 2006, at 6:14 PM, Stastny Richard wrote:

> >>I was using the term to mean a peering exchange (or federation or
> >>whatever you want to call them) that can be joined dynamically.
> >
> >
> >On what terms? I still do nor understand. Joining a federation
> >always involves some legal negotiotions.
> 
> Well, they require an agreement.  The legality of that agreement is  
> another matter.
> 
> If the execution of that agreement requires signed contracts, etc...  
> all the configuration information for joining that federation does  
> not need to be in machine readable form.  Indeed, much of it will not  
> be given until an NDA is signed or a service contract is signed, so  
> it won't even be available on the web.
> 
> If, however, there is intention to have the policies on which the  
> peering federation agreement operates to be machine readable, such as  
> in the DDDS application proposed by Otmar, then this seems to imply  
> that joining the federation is more opportunistic in nature.

My DDDS proposal allows both:

draft-lendl-speermint-federations-00 is targetted at the former
type of federation (we call it "named federation"). All the DNS
contains is the federation ID. Nothing else is required. From this
the draft (highlighted with !!):

-=-=-=-
3.  Federations

   The proposed method is based upon the concept of a "Federation".  A
   federation in this context is defined as follows:

      A Federation is a group of VoIP service providers which
      *  agree to receive calls from each other via SIP,
      *  agree on a set of administrative rules for such calls
         (settlement, abuse-handling, ...), and
      *  agree on specific rules for the technical details of the
         interconnection.

!! This document does not define what these rules can be and how they
!! are communicated to all members of a federation.  There is no
!! requirement to make those rules public.

   Federations shall use URIs as their identifiers.  It is RECOMMENDED
   that federations use URLs as identifiers which point to documents
   describing the federation.
-=-=-=-

The draft describing the overall algorithm (draft-lendl-domain-policy-ddds-00)
contains an "Examples" section which uses "ad-hoc" federations based
on references to standards. That section contains the following
warning:

-=-=-=-
5.  Examples

   In order to give meaningful examples, we need to define a few types
   of rules.  The definitions here are purely meant to illustrate the
   possibilities.  They MUST NOT be considered as valid examples of real
   entries.

...

   o  The policy-type "std" shall indicate that the URI indicates a
      standard the sender has to fulfill.

-=-=-=-

I haven't written an I-D defining the exact semantics of these
ad-hoc federations.

Our assumption is that named federation (which can hide their
internals from the public) are well suited for peerings between
service providers. 

For the Austrian I-ENUM trial, we're going to start with named
federations only.

Ad-hoc federations are IMHO more suitable for enterprise
environments.

Questions to the group:

* Is there interest in an I-D which defines the semantic of 
  policy rules which can be used to form ad-hoc federations?

* Is there interest in another presentation regarding my 
  domain policy DDDS proposal at the upcoming IETF meeting?

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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Richard:

All examples aside, I attempted to craft some words that would not set
any limitations on what can or cannot be used in the host portion of a
SIP URI while maintaining a consistent method for resolving whatever is
used in the host portion of a SIP URI.

When a SIP URI is used in Infrastructure ENUM, the procedures outlined
in RFC3263 MUST be followed for locating a UAS that will accept a
request for the given URI.  For providers assuming the role of a UAS,
this means that they MUST provision RFC3263 compliant NAPTR records and
SRV records that correspond to the host portion of the SIP URI
provisioned in Infrastructure ENUM.  For providers assuming the role of
a UAC, when a query to Infrastructure ENUM returns a SIP URI, they MUST
follow the procedures outlined in Section 4 of RFC3263 to determine the
IP address, port, and transport protocol of a UAS that will accept a
request for the given URI.

Please let me know what you think.

Regards,

Tom

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 09:06
> To: speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Enum] How to correctly set up the request URI?
>=20
> Folks,
>=20
> Question for clarification:
>=20
> In the discussions here on how to populate sip URI in Infrastructure
> ENUM to derive the Call Routing data for SPEERMINT  two
> lines of thought exist:
>=20
> The Enumservice SIP in Infrastructure ENUM should contain
> 1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
> 2. the ingress point into the provider network:
> e.g. sip:+4319793321@sbc4711.provB.net
>=20
> The originating UAC has in this case set the To: field to
> tel:+4319793321 and the Request-URI to e.g.
> sip:+4319793321provA.net;user=3Dphone. This caused the
> ENUM query at the proxy of provider A
>=20
> The question is how the proxy of provider A is now
> setting up the request URI in cases mentioned above?
>=20
> Does the UAS from provider B accept calls to sbc7411.provB.net,
> or must the request URI set to the proper AoR as described in
> 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
>=20
> I would also like to draw the attention to section 3 of
> RFC3761 (SIP Enumservice) explaining the usage of AoR in
> ENUM.
>=20
> Regards
>=20
> -sta
>=20
>=20
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum

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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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	"Stastny Richard" <Richard.Stastny@oefeg.at>,
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In the past it has been suggested that if Infrastructure ENUM is openly =
accessible, then for data privacy reasons it would not be acceptable to =
list the user AOR.  Do we still believe this is the case, or do we now =
think providing the user AOR would be acceptable?

James

-----Original Message-----
From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]=20
Sent: Thursday, June 01, 2006 2:22 PM
To: Stastny Richard; Creighton, Tom; speermint@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint





_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
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Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



From speermint-bounces@ietf.org Fri Jun 02 15:37:06 2006
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Tom,
=20
being in favor of the usage of an AoR in Infrastructure ENUM, I have no
problems with your proposed words. There are only two issues:
=20
1. There will be some people objecting, I assume.
2. Where should this words go?
-in the ENUM requirements or in the SPEERMINT requirements?
=20
could the esteemed chairs of the two WGs please give their advice?
=20
-sta

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Fr 02.06.2006 21:19
An: Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?



Richard:

All examples aside, I attempted to craft some words that would not set
any limitations on what can or cannot be used in the host portion of a
SIP URI while maintaining a consistent method for resolving whatever is
used in the host portion of a SIP URI.

When a SIP URI is used in Infrastructure ENUM, the procedures outlined
in RFC3263 MUST be followed for locating a UAS that will accept a
request for the given URI.  For providers assuming the role of a UAS,
this means that they MUST provision RFC3263 compliant NAPTR records and
SRV records that correspond to the host portion of the SIP URI
provisioned in Infrastructure ENUM.  For providers assuming the role of
a UAC, when a query to Infrastructure ENUM returns a SIP URI, they MUST
follow the procedures outlined in Section 4 of RFC3263 to determine the
IP address, port, and transport protocol of a UAS that will accept a
request for the given URI.

Please let me know what you think.

Regards,

Tom

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 09:06
> To: speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Enum] How to correctly set up the request URI?
>
> Folks,
>
> Question for clarification:
>
> In the discussions here on how to populate sip URI in Infrastructure
> ENUM to derive the Call Routing data for SPEERMINT  two
> lines of thought exist:
>
> The Enumservice SIP in Infrastructure ENUM should contain
> 1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
> 2. the ingress point into the provider network:
> e.g. sip:+4319793321@sbc4711.provB.net
>
> The originating UAC has in this case set the To: field to
> tel:+4319793321 and the Request-URI to e.g.
> sip:+4319793321provA.net;user=3Dphone. This caused the
> ENUM query at the proxy of provider A
>
> The question is how the proxy of provider A is now
> setting up the request URI in cases mentioned above?
>
> Does the UAS from provider B accept calls to sbc7411.provB.net,
> or must the request URI set to the proper AoR as described in
> 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
>
> I would also like to draw the attention to section 3 of
> RFC3761 (SIP Enumservice) explaining the usage of AoR in
> ENUM.
>
> Regards
>
> -sta
>
>
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum



_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



From speermint-bounces@ietf.org Fri Jun 02 15:46:16 2006
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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Fri, 2 Jun 2006 21:50:17 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Pfautz, Penn L, GBLAM" <ppfautz@att.com>,
	"James McEachern" <jmce@nortel.com>, <speermint@ietf.org>, <enum@ietf.org>
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Penn,
=20
I just wanted to give a similar reply:
the user part of the SIP URI should either be the E.164 phone number and =
an
identifier not allowing to derive any information about the identity of =
the
user e.g. an internal number or so.
=20
Regarding this issue, we need to add a security section to
draft-ietf-enum-infrastructure anyway, so this issue should be covered
there.
=20
-sta

________________________________

Von: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
Gesendet: Fr 02.06.2006 21:30
An: James McEachern; Lee, Yiu; Stastny Richard; Creighton, Tom; =
speermint@ietf.org; enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



James
I think what this means is that a provider should not provision an AoR
that would a) provide personally identifiable information beyond the
called phone number or b) allow direct access to the user's CPE.
Or are you referring to a user supplied SIP URI?

Penn

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]
Sent: Friday, June 02, 2006 3:12 PM
To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

In the past it has been suggested that if Infrastructure ENUM is openly
accessible, then for data privacy reasons it would not be acceptable to
list the user AOR.  Do we still believe this is the case, or do we now
think providing the user AOR would be acceptable?

James

-----Original Message-----
From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Sent: Thursday, June 01, 2006 2:22 PM
To: Stastny Richard; Creighton, Tom; speermint@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:

if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
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From speermint-bounces@ietf.org Fri Jun 02 15:49:28 2006
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Fri, 2 Jun 2006 15:48:59 -0400
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "James McEachern" <jmce@nortel.com>,
	"Lee, Yiu" <Yiu_Lee@cable.comcast.com>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>, <speermint@ietf.org>,
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I think we have two separate issues:

1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
defines an Address-of-Record as:

"Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user."

Based on that definition and the examples provided by Richard, a user's
privacy would not be compromised because it doesn't directly reach or
identify the customer.

2. There is confusion around what is or is not acceptable in the host
portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
matter what is placed in the host portion of the SIP URI as long as it
doesn't compromise a user's privacy.  So in the examples Richard
provided earlier, either are acceptable and would resolve in an
identical manner.

Tom

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 15:12
> To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> In the past it has been suggested that if Infrastructure ENUM is
openly
> accessible, then for data privacy reasons it would not be acceptable
to
> list the user AOR.  Do we still believe this is the case, or do we now
> think providing the user AOR would be acceptable?
>=20
> James
>=20
> -----Original Message-----
> From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Sent: Thursday, June 01, 2006 2:22 PM
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Actually, I think it is sufficient for the Enum to return the user AOR
> because the routing decision should happen in the provB.net network.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:18 PM
> To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, so I am back to my original question:
>=20
> if I implement both types of public user identities:
> E.164 numbers and SIP AoRs, why doing the mapping to different hosting
> proxies in two places (in this case ENUM and the HSS), if I can do it
> only in one place?
>=20
> -sta
>=20
>=20
>=20
> ________________________________
>=20
> Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Gesendet: Do 01.06.2006 20:10
> An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Yes. IMS is using Option 2. Option 1 is also possible if the operator
> decides to use just a redirect proxy.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:07 PM
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>=20
> Richard
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>=20
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
>=20
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
>=20
> > point.  This could very well save a service provider a lot of money
in
>=20
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
>=20
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
>=20
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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	"Lee, Yiu" <Yiu_Lee@Cable.Comcast.com>,
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James
I think what this means is that a provider should not provision an AoR
that would a) provide personally identifiable information beyond the
called phone number or b) allow direct access to the user's CPE.
Or are you referring to a user supplied SIP URI?

Penn=20

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]=20
Sent: Friday, June 02, 2006 3:12 PM
To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

In the past it has been suggested that if Infrastructure ENUM is openly
accessible, then for data privacy reasons it would not be acceptable to
list the user AOR.  Do we still believe this is the case, or do we now
think providing the user AOR would be acceptable?

James

-----Original Message-----
From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]=20
Sent: Thursday, June 01, 2006 2:22 PM
To: Stastny Richard; Creighton, Tom; speermint@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
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From speermint-bounces@ietf.org Fri Jun 02 15:59:55 2006
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Comments below.

> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Friday, June 02, 2006 15:41
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: Re: [Enum] How to correctly set up the request URI?
>=20
> Tom,
>=20
> being in favor of the usage of an AoR in Infrastructure ENUM, I have
no
> problems with your proposed words. There are only two issues:
>=20
> 1. There will be some people objecting, I assume.

[Tom Creighton wrote:]=20

I welcome any comments or recommendations that anyone has to help move
beyond this discussion. =20

> 2. Where should this words go?
> -in the ENUM requirements or in the SPEERMINT requirements?

[Tom Creighton wrote:]=20

My initial reaction would be to keep it in SPEERMINT as this is a
peering architecture issue and not a specific ENUM protocol issue, but
as you said, advice from the 4 chairs would be appreciated.

>=20
> could the esteemed chairs of the two WGs please give their advice?
>=20
> -sta
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Fr 02.06.2006 21:19
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>=20
>=20
>=20
> Richard:
>=20
> All examples aside, I attempted to craft some words that would not set
> any limitations on what can or cannot be used in the host portion of a
> SIP URI while maintaining a consistent method for resolving whatever
is
> used in the host portion of a SIP URI.
>=20
> When a SIP URI is used in Infrastructure ENUM, the procedures outlined
> in RFC3263 MUST be followed for locating a UAS that will accept a
> request for the given URI.  For providers assuming the role of a UAS,
> this means that they MUST provision RFC3263 compliant NAPTR records
and
> SRV records that correspond to the host portion of the SIP URI
> provisioned in Infrastructure ENUM.  For providers assuming the role
of
> a UAC, when a query to Infrastructure ENUM returns a SIP URI, they
MUST
> follow the procedures outlined in Section 4 of RFC3263 to determine
the
> IP address, port, and transport protocol of a UAS that will accept a
> request for the given URI.
>=20
> Please let me know what you think.
>=20
> Regards,
>=20
> Tom
>=20
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 09:06
> > To: speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Enum] How to correctly set up the request URI?
> >
> > Folks,
> >
> > Question for clarification:
> >
> > In the discussions here on how to populate sip URI in Infrastructure
> > ENUM to derive the Call Routing data for SPEERMINT  two
> > lines of thought exist:
> >
> > The Enumservice SIP in Infrastructure ENUM should contain
> > 1. the address-of-record of the user: e.g. sip:+4319793321@provB.net
> > 2. the ingress point into the provider network:
> > e.g. sip:+4319793321@sbc4711.provB.net
> >
> > The originating UAC has in this case set the To: field to
> > tel:+4319793321 and the Request-URI to e.g.
> > sip:+4319793321provA.net;user=3Dphone. This caused the
> > ENUM query at the proxy of provider A
> >
> > The question is how the proxy of provider A is now
> > setting up the request URI in cases mentioned above?
> >
> > Does the UAS from provider B accept calls to sbc7411.provB.net,
> > or must the request URI set to the proper AoR as described in
> > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> >
> > I would also like to draw the attention to section 3 of
> > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> > ENUM.
> >
> > Regards
> >
> > -sta
> >
> >
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum
>=20


_______________________________________________
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From speermint-bounces@ietf.org Fri Jun 02 16:40:11 2006
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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From: "James McEachern" <jmce@nortel.com>
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Penn,
I'm thinking of either user supplied SIP URIs, or carrier supplied SIP =
URIs other than those that are simply based on the E.164 number.
(e.g. sip:+4319793321provA.net;user=3Dphone)  I am assuming that these =
are possible, and that therefore we must provide a solution that deals =
with them.

James

-----Original Message-----
From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]=20
Sent: Friday, June 02, 2006 3:31 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard; =
Creighton, Tom; speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

James
I think what this means is that a provider should not provision an AoR
that would a) provide personally identifiable information beyond the
called phone number or b) allow direct access to the user's CPE.
Or are you referring to a user supplied SIP URI?

Penn=20

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]=20
Sent: Friday, June 02, 2006 3:12 PM
To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

In the past it has been suggested that if Infrastructure ENUM is openly
accessible, then for data privacy reasons it would not be acceptable to
list the user AOR.  Do we still believe this is the case, or do we now
think providing the user AOR would be acceptable?

James

-----Original Message-----
From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]=20
Sent: Thursday, June 01, 2006 2:22 PM
To: Stastny Richard; Creighton, Tom; speermint@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Fri, 2 Jun 2006 16:49:58 -0400
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 If a carrier agrees to provision a user-supplied URI and the user
chooses to use one including PII after proper caution, that's their
lookout.=20
I see no privacy issue with a carrier URI like the one you show below as
long as it points to a carrier network element rather than directly to
say, the user's terminal adapter. If the pointer is direct to CPE I'd
have some concerns, which is not to say a provider can be forbidden
(except by regulators) to do so.

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]=20
Sent: Friday, June 02, 2006 4:40 PM
To: Pfautz, Penn L, GBLAM; speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Penn,
I'm thinking of either user supplied SIP URIs, or carrier supplied SIP
URIs other than those that are simply based on the E.164 number.
(e.g. sip:+4319793321provA.net;user=3Dphone)  I am assuming that these =
are
possible, and that therefore we must provide a solution that deals with
them.

James

-----Original Message-----
From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]=20
Sent: Friday, June 02, 2006 3:31 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard;
Creighton, Tom; speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

James
I think what this means is that a provider should not provision an AoR
that would a) provide personally identifiable information beyond the
called phone number or b) allow direct access to the user's CPE.
Or are you referring to a user supplied SIP URI?

Penn=20

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]=20
Sent: Friday, June 02, 2006 3:12 PM
To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

In the past it has been suggested that if Infrastructure ENUM is openly
accessible, then for data privacy reasons it would not be acceptable to
list the user AOR.  Do we still believe this is the case, or do we now
think providing the user AOR would be acceptable?

James

-----Original Message-----
From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]=20
Sent: Thursday, June 01, 2006 2:22 PM
To: Stastny Richard; Creighton, Tom; speermint@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Actually, I think it is sufficient for the Enum to return the user AOR
because the routing decision should happen in the provB.net network.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Thursday, June 01, 2006 2:18 PM
To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, so I am back to my original question:
=20
if I implement both types of public user identities:
E.164 numbers and SIP AoRs, why doing the mapping to different hosting
proxies in two places (in this case ENUM and the HSS), if I can do it
only in one place?

-sta



________________________________

Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
Gesendet: Do 01.06.2006 20:10
An: Stastny Richard; Creighton, Tom; speermint@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Yes. IMS is using Option 2. Option 1 is also possible if the operator
decides to use just a redirect proxy.

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Thursday, June 01, 2006 2:07 PM
To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
Cc: enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

so basically the IMS I-CSCF -  HSS - S-CSCF way?

Richard

________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Do 01.06.2006 19:45
An: Lee, Yiu; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Agreed, in this case the proxy associated with provB.net would be an
intermediate device that would have to perform some sort of internal
lookup in order to route the request to the next hop.

> -----Original Message-----
> From: Lee, Yiu
> Sent: Thursday, June 01, 2006 13:34
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In this case, I guess when the originating proxy resolves the domain
part
> of the AOR (sip:+4319793321@provB.net), it will return the IP address
of
> the root domain "provB.net" which may be a redirect proxy. So
"provB.net"
> has two choices:
>
> (1) It accesses its local routing table and sends a 305 to the
originating
> proxy with "sbc4711.provB.net" in the contact header.
>
> (2) It replaces the domain part in the r-uri to more specific like=20
> "proxy.austria.provB.net" and forward the message to the next-hop.
>
> Either case, the assumption is provB.net will have a way to find out
where
> the home proxy of user +4319793321.
>
> This is only my guess.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 1:13 PM
> To: Creighton, Tom; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: [Speermint] Re: [Enum] How to correctly set up the request
URI?
>
> I see,
>
> but how do you solve this problem if the users can also be reached by
AoRs
> direct?
>
> -sta
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 17:57
> An: Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Enum] How to correctly set up the request URI?
>
>
>
> Maybe the service provider has some sort of segregated network
(physically
> or logically) and wants/needs to associate each its subscribers with a

> particular network segment.  If the provider went with the flat model
of
> provB.net, each of its proxies would have to accept requests destined
to
> all of its customers and would then have to route the request to the=20
> appropriate region or division of its network.
> If the provider went with the hierarchical approach of
sbc4711.provB.net,
> all of the requests would already be routed to the appropriate ingress

> point.  This could very well save a service provider a lot of money in

> route proxy equipment although might cost more on the provisioning
side.
>
> That is my guess.
>
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 11:30
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Enum] How to correctly set up the request URI?
> >
> > Thank you, Tom
> >
> > just a minor additional question:
> >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> >
> > If I have to do this according to RFC3263 anyway, why put the=20
> > sbc4711.provB.net into ENUM in the first place?
> >
> > -sta
> >
> >
> > > -----Original Message-----
> > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Sent: Thursday, June 01, 2006 4:36 PM
> > > To: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 09:06
> > > > To: speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Enum] How to correctly set up the request URI?
> > > >
> > > > Folks,
> > > >
> > > > Question for clarification:
> > > >
> > > > In the discussions here on how to populate sip URI in
> Infrastructure
> > > > ENUM to derive the Call Routing data for SPEERMINT  two lines of

> > > > thought exist:
> > > >
> > > > The Enumservice SIP in Infrastructure ENUM should contain 1. the

> > > > address-of-record of the user: e.g.
> sip:+4319793321@provB.net
> > > > 2. the ingress point into the provider network:
> > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > >
> > > > The originating UAC has in this case set the To: field to
> > > > tel:+4319793321 and the Request-URI to e.g.
> > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM =
query
at
> > > > the proxy of provider A
> > > >
> > > > The question is how the proxy of provider A is now setting up
the
> > > > request URI in cases mentioned above?
> > > >
> > > > Does the UAS from provider B accept calls to sbc7411.provB.net,
or
> > > > must the request URI set to the proper AoR as described in
> > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in ENUM.
> > >
> > > [Tom Creighton wrote:]
> > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > identify an address that the UAS is willing to accept requests
for,
> it
> > > SHOULD reject the request with a 404 (Not Found) response."
> > >
> > > As long as the UAS from provider B is "willing" to accept the
> request,
> > > either should work.  Since provider B is populating Infrastructure
> > ENUM,
> > > one would hope that they would coordinate this data with the=20
> > > configuration of their proxies.
> > >
> > > Besides, how do you actually know that sbc4441.provB.net is
actually
> a
> > > hostname and isn't a subdomain?
> > >
> > > In either case, after the provider A proxy performs an ENUM query
> and
> > > rewrites the request URI, it should behave as a UAC and follow the

> > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > address,
> > > port, and transport protocol for the provider B UAS.
> > >
> > > >
> > > > I would also like to draw the attention to section 3 of
> > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > >
> > > > Regards
> > > >
> > > > -sta
> > > >
> > > >
> > > > _______________________________________________
> > > > enum mailing list
> > > > enum@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/enum
>
>
>
> _______________________________________________
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Ok, I think I see now.  Presumably I can have a SIP URI that clearly =
identifies me, but this is not the AOR that would be populated in =
Infrastructure ENUM.  That AOR would either be based on my E.164 number, =
or on some other random string created by the carrier.  Have I got it =
right?

James

-----Original Message-----
From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]=20
Sent: Friday, June 02, 2006 3:49 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard; =
speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

I think we have two separate issues:

1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
defines an Address-of-Record as:

"Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user."

Based on that definition and the examples provided by Richard, a user's
privacy would not be compromised because it doesn't directly reach or
identify the customer.

2. There is confusion around what is or is not acceptable in the host
portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
matter what is placed in the host portion of the SIP URI as long as it
doesn't compromise a user's privacy.  So in the examples Richard
provided earlier, either are acceptable and would resolve in an
identical manner.

Tom

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 15:12
> To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> In the past it has been suggested that if Infrastructure ENUM is
openly
> accessible, then for data privacy reasons it would not be acceptable
to
> list the user AOR.  Do we still believe this is the case, or do we now
> think providing the user AOR would be acceptable?
>=20
> James
>=20
> -----Original Message-----
> From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Sent: Thursday, June 01, 2006 2:22 PM
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Actually, I think it is sufficient for the Enum to return the user AOR
> because the routing decision should happen in the provB.net network.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:18 PM
> To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, so I am back to my original question:
>=20
> if I implement both types of public user identities:
> E.164 numbers and SIP AoRs, why doing the mapping to different hosting
> proxies in two places (in this case ENUM and the HSS), if I can do it
> only in one place?
>=20
> -sta
>=20
>=20
>=20
> ________________________________
>=20
> Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Gesendet: Do 01.06.2006 20:10
> An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Yes. IMS is using Option 2. Option 1 is also possible if the operator
> decides to use just a redirect proxy.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:07 PM
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>=20
> Richard
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>=20
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
>=20
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
>=20
> > point.  This could very well save a service provider a lot of money
in
>=20
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
>=20
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
>=20
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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

_______________________________________________
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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Yes. Again, there might be exceptions, for example a business might want
a URI with their name to be used and some SPs might agree to that, but
generally the point of infrastructure ENUM is to be invisible to the
user=20

-----Original Message-----
From: James McEachern [mailto:jmce@nortel.com]=20
Sent: Friday, June 02, 2006 4:52 PM
To: Creighton, Tom; Lee, Yiu; Stastny Richard; speermint@ietf.org;
enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

Ok, I think I see now.  Presumably I can have a SIP URI that clearly
identifies me, but this is not the AOR that would be populated in
Infrastructure ENUM.  That AOR would either be based on my E.164 number,
or on some other random string created by the carrier.  Have I got it
right?

James

-----Original Message-----
From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]=20
Sent: Friday, June 02, 2006 3:49 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard;
speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?

I think we have two separate issues:

1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
defines an Address-of-Record as:

"Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user."

Based on that definition and the examples provided by Richard, a user's
privacy would not be compromised because it doesn't directly reach or
identify the customer.

2. There is confusion around what is or is not acceptable in the host
portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
matter what is placed in the host portion of the SIP URI as long as it
doesn't compromise a user's privacy.  So in the examples Richard
provided earlier, either are acceptable and would resolve in an
identical manner.

Tom

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 15:12
> To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> In the past it has been suggested that if Infrastructure ENUM is
openly
> accessible, then for data privacy reasons it would not be acceptable
to
> list the user AOR.  Do we still believe this is the case, or do we now
> think providing the user AOR would be acceptable?
>=20
> James
>=20
> -----Original Message-----
> From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Sent: Thursday, June 01, 2006 2:22 PM
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Actually, I think it is sufficient for the Enum to return the user AOR
> because the routing decision should happen in the provB.net network.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:18 PM
> To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, so I am back to my original question:
>=20
> if I implement both types of public user identities:
> E.164 numbers and SIP AoRs, why doing the mapping to different hosting
> proxies in two places (in this case ENUM and the HSS), if I can do it
> only in one place?
>=20
> -sta
>=20
>=20
>=20
> ________________________________
>=20
> Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Gesendet: Do 01.06.2006 20:10
> An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Yes. IMS is using Option 2. Option 1 is also possible if the operator
> decides to use just a redirect proxy.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:07 PM
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>=20
> Richard
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>=20
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
>=20
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
>=20
> > point.  This could very well save a service provider a lot of money
in
>=20
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
>=20
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
>=20
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
>=20
>=20
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> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Fri, 2 Jun 2006 16:58:19 -0400
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "James McEachern" <jmce@nortel.com>,
	"Lee, Yiu" <Yiu_Lee@cable.comcast.com>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>, <speermint@ietf.org>,
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That is correct.  The SIP URI that clearly identifies you, as you
described, would only be used within your provider's network.

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 16:52
> To: Creighton, Tom; Lee, Yiu; Stastny Richard; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, I think I see now.  Presumably I can have a SIP URI that clearly
> identifies me, but this is not the AOR that would be populated in
> Infrastructure ENUM.  That AOR would either be based on my E.164
number,
> or on some other random string created by the carrier.  Have I got it
> right?
>=20
> James
>=20
> -----Original Message-----
> From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Sent: Friday, June 02, 2006 3:49 PM
> To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard;
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> I think we have two separate issues:
>=20
> 1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
> defines an Address-of-Record as:
>=20
> "Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
>          that points to a domain with a location service that can map
>          the URI to another URI where the user might be available.
>          Typically, the location service is populated through
>          registrations.  An AOR is frequently thought of as the
"public
>          address" of the user."
>=20
> Based on that definition and the examples provided by Richard, a
user's
> privacy would not be compromised because it doesn't directly reach or
> identify the customer.
>=20
> 2. There is confusion around what is or is not acceptable in the host
> portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
> matter what is placed in the host portion of the SIP URI as long as it
> doesn't compromise a user's privacy.  So in the examples Richard
> provided earlier, either are acceptable and would resolve in an
> identical manner.
>=20
> Tom
>=20
> > -----Original Message-----
> > From: James McEachern [mailto:jmce@nortel.com]
> > Sent: Friday, June 02, 2006 15:12
> > To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> > enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In the past it has been suggested that if Infrastructure ENUM is
> openly
> > accessible, then for data privacy reasons it would not be acceptable
> to
> > list the user AOR.  Do we still believe this is the case, or do we
now
> > think providing the user AOR would be acceptable?
> >
> > James
> >
> > -----Original Message-----
> > From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > Sent: Thursday, June 01, 2006 2:22 PM
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > Actually, I think it is sufficient for the Enum to return the user
AOR
> > because the routing decision should happen in the provB.net network.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 2:18 PM
> > To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> > Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > Ok, so I am back to my original question:
> >
> > if I implement both types of public user identities:
> > E.164 numbers and SIP AoRs, why doing the mapping to different
hosting
> > proxies in two places (in this case ENUM and the HSS), if I can do
it
> > only in one place?
> >
> > -sta
> >
> >
> >
> > ________________________________
> >
> > Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > Gesendet: Do 01.06.2006 20:10
> > An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> >
> >
> > Yes. IMS is using Option 2. Option 1 is also possible if the
operator
> > decides to use just a redirect proxy.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 2:07 PM
> > To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > so basically the IMS I-CSCF -  HSS - S-CSCF way?
> >
> > Richard
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 19:45
> > An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> >
> >
> > Agreed, in this case the proxy associated with provB.net would be an
> > intermediate device that would have to perform some sort of internal
> > lookup in order to route the request to the next hop.
> >
> > > -----Original Message-----
> > > From: Lee, Yiu
> > > Sent: Thursday, June 01, 2006 13:34
> > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > In this case, I guess when the originating proxy resolves the
domain
> > part
> > > of the AOR (sip:+4319793321@provB.net), it will return the IP
> address
> > of
> > > the root domain "provB.net" which may be a redirect proxy. So
> > "provB.net"
> > > has two choices:
> > >
> > > (1) It accesses its local routing table and sends a 305 to the
> > originating
> > > proxy with "sbc4711.provB.net" in the contact header.
> > >
> > > (2) It replaces the domain part in the r-uri to more specific like
> > > "proxy.austria.provB.net" and forward the message to the next-hop.
> > >
> > > Either case, the assumption is provB.net will have a way to find
out
> > where
> > > the home proxy of user +4319793321.
> > >
> > > This is only my guess.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 1:13 PM
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> > >
> > > I see,
> > >
> > > but how do you solve this problem if the users can also be reached
> by
> > AoRs
> > > direct?
> > >
> > > -sta
> > >
> > > ________________________________
> > >
> > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Gesendet: Do 01.06.2006 17:57
> > > An: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Betreff: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > Maybe the service provider has some sort of segregated network
> > (physically
> > > or logically) and wants/needs to associate each its subscribers
with
> a
> >
> > > particular network segment.  If the provider went with the flat
> model
> > of
> > > provB.net, each of its proxies would have to accept requests
> destined
> > to
> > > all of its customers and would then have to route the request to
the
> > > appropriate region or division of its network.
> > > If the provider went with the hierarchical approach of
> > sbc4711.provB.net,
> > > all of the requests would already be routed to the appropriate
> ingress
> >
> > > point.  This could very well save a service provider a lot of
money
> in
> >
> > > route proxy equipment although might cost more on the provisioning
> > side.
> > >
> > > That is my guess.
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 11:30
> > > > To: Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > > Thank you, Tom
> > > >
> > > > just a minor additional question:
> > > >
> > > > > In either case, after the provider A proxy performs an ENUM
> query
> > > and
> > > > > rewrites the request URI, it should behave as a UAC and follow
> the
> >
> > > > > procedures outlined in Section 4 of RFC3263 to determine the
IP
> > > > address,
> > > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > If I have to do this according to RFC3263 anyway, why put the
> > > > sbc4711.provB.net into ENUM in the first place?
> > > >
> > > > -sta
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > > To: Stastny Richard; speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > >
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > > To: speermint@ietf.org
> > > > > > Cc: enum@ietf.org
> > > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > > >
> > > > > > Folks,
> > > > > >
> > > > > > Question for clarification:
> > > > > >
> > > > > > In the discussions here on how to populate sip URI in
> > > Infrastructure
> > > > > > ENUM to derive the Call Routing data for SPEERMINT  two
lines
> of
> >
> > > > > > thought exist:
> > > > > >
> > > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
> the
> >
> > > > > > address-of-record of the user: e.g.
> > > sip:+4319793321@provB.net
> > > > > > 2. the ingress point into the provider network:
> > > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > > >
> > > > > > The originating UAC has in this case set the To: field to
> > > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
> query
> > at
> > > > > > the proxy of provider A
> > > > > >
> > > > > > The question is how the proxy of provider A is now setting
up
> > the
> > > > > > request URI in cases mentioned above?
> > > > > >
> > > > > > Does the UAS from provider B accept calls to
> sbc7411.provB.net,
> > or
> > > > > > must the request URI set to the proper AoR as described in
> > > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
> ENUM.
> > > > >
> > > > > [Tom Creighton wrote:]
> > > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does
not
> > > > > identify an address that the UAS is willing to accept requests
> > for,
> > > it
> > > > > SHOULD reject the request with a 404 (Not Found) response."
> > > > >
> > > > > As long as the UAS from provider B is "willing" to accept the
> > > request,
> > > > > either should work.  Since provider B is populating
> Infrastructure
> > > > ENUM,
> > > > > one would hope that they would coordinate this data with the
> > > > > configuration of their proxies.
> > > > >
> > > > > Besides, how do you actually know that sbc4441.provB.net is
> > actually
> > > a
> > > > > hostname and isn't a subdomain?
> > > > >
> > > > > In either case, after the provider A proxy performs an ENUM
> query
> > > and
> > > > > rewrites the request URI, it should behave as a UAC and follow
> the
> >
> > > > > procedures outlined in Section 4 of RFC3263 to determine the
IP
> > > > address,
> > > > > port, and transport protocol for the provider B UAS.
> > > > >
> > > > > >
> > > > > > I would also like to draw the attention to section 3 of
> > > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
ENUM.
> > > > > >
> > > > > > Regards
> > > > > >
> > > > > > -sta
> > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > enum mailing list
> > > > > > enum@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/enum
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum
>=20
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum

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	"'Stastny Richard'" <Richard.Stastny@oefeg.at>, <speermint@ietf.org>
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This discussion is best kept in SPEERMINT. IMHO.

s/

your bewildered ENUM WG co-chair

> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Friday, June 02, 2006 15:41
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: Re: [Enum] How to correctly set up the request URI?
> >
> > Tom,
> >
> > being in favor of the usage of an AoR in Infrastructure ENUM, I have
> no
> > problems with your proposed words. There are only two issues:
> >
> > 1. There will be some people objecting, I assume.
> 
> [Tom Creighton wrote:]
> 
> I welcome any comments or recommendations that anyone has to help move
> beyond this discussion.
> 
> > 2. Where should this words go?
> > -in the ENUM requirements or in the SPEERMINT requirements?
> 
> [Tom Creighton wrote:]
> 
> My initial reaction would be to keep it in SPEERMINT as this is a
> peering architecture issue and not a specific ENUM protocol issue, but
> as you said, advice from the 4 chairs would be appreciated.


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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Fri, 2 Jun 2006 23:03:24 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "James McEachern" <jmce@nortel.com>,
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Exactly,
=20
A service provider will never populate Infrastructure ENUM with a user =
provided URI,
(even if the user may have one), it will always be the alias used by the
service provider.
=20
We should not mix this up with the discussion in SPEERMINT on user =
provided
AoR such as sip:user@company.com. This will always be an alias in =
addition
to the AoR used by the provider.
=20
Of couse the user may use this AoR in User ENUM.
=20
-sta

________________________________

Von: James McEachern [mailto:jmce@nortel.com]
Gesendet: Fr 02.06.2006 22:51
An: Creighton, Tom; Lee, Yiu; Stastny Richard; speermint@ietf.org; =
enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



Ok, I think I see now.  Presumably I can have a SIP URI that clearly =
identifies me, but this is not the AOR that would be populated in =
Infrastructure ENUM.  That AOR would either be based on my E.164 number, =
or on some other random string created by the carrier.  Have I got it =
right?

James

-----Original Message-----
From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Sent: Friday, June 02, 2006 3:49 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard; =
speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

I think we have two separate issues:

1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
defines an Address-of-Record as:

"Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user."

Based on that definition and the examples provided by Richard, a user's
privacy would not be compromised because it doesn't directly reach or
identify the customer.

2. There is confusion around what is or is not acceptable in the host
portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
matter what is placed in the host portion of the SIP URI as long as it
doesn't compromise a user's privacy.  So in the examples Richard
provided earlier, either are acceptable and would resolve in an
identical manner.

Tom

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 15:12
> To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> In the past it has been suggested that if Infrastructure ENUM is
openly
> accessible, then for data privacy reasons it would not be acceptable
to
> list the user AOR.  Do we still believe this is the case, or do we now
> think providing the user AOR would be acceptable?
>
> James
>
> -----Original Message-----
> From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Sent: Thursday, June 01, 2006 2:22 PM
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> Actually, I think it is sufficient for the Enum to return the user AOR
> because the routing decision should happen in the provB.net network.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:18 PM
> To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> Ok, so I am back to my original question:
>
> if I implement both types of public user identities:
> E.164 numbers and SIP AoRs, why doing the mapping to different hosting
> proxies in two places (in this case ENUM and the HSS), if I can do it
> only in one place?
>
> -sta
>
>
>
> ________________________________
>
> Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Gesendet: Do 01.06.2006 20:10
> An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
>
>
> Yes. IMS is using Option 2. Option 1 is also possible if the operator
> decides to use just a redirect proxy.
>
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:07 PM
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>
> Richard
>
> ________________________________
>
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
>
>
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
>
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
>
> > point.  This could very well save a service provider a lot of money
in
>
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
>
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
>
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>
> _______________________________________________
> Speermint mailing list
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>
>
>
>
>
> _______________________________________________
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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Tony Rutkowski" <trutkowski@verisign.com>,
	"Pfautz, Penn L, GBLAM" <ppfautz@att.com>,
	"James McEachern" <jmce@nortel.com>, <speermint@ietf.org>, <enum@ietf.org>
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Tony,
=20
I do not fully understand what the CALEA issue has to do with the =
question which
URI is put in ENUM. This may be a signalling requirement in SPEERMINT
=20
I have complete different concerns regarding CALEA, in principle I do=20
not understand why the service providers are bothered with this at all,
because the NSA is getting all data from Penn anyway ;-)
=20
http://news.com.com/AT38T+leaks+sensitive+info+in+NSA+suit/2100-1028_3-60=
77353.html?tag=3Dnefd.lede
=20
http://www.wired.com/news/technology/0,70944-0.html
=20
-sta

________________________________

Von: Tony Rutkowski [mailto:trutkowski@verisign.com]
Gesendet: Fr 02.06.2006 23:13
An: Pfautz, Penn L, GBLAM; James McEachern; speermint@ietf.org; =
enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



Hi Penn,

>I see no privacy issue with a carrier URI like the one you show below =
as
>long as it points to a carrier network element rather than directly to
>say, the user's terminal adapter. If the pointer is direct to CPE I'd
>have some concerns, which is not to say a provider can be forbidden
>(except by regulators) to do so.

The concerns primarily relate to network integrity as
well as support of law enforcement (CALEA/LI requirements)
in determining authoritative bindings of user/provider
information to identifiers.  In Europe and many countries,
this binding information is also subject to data retention
directives.

--tony


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This is also my preference, but it is basically what SPEERMINT
expects to be the CRD output from Infrastructure ENUM
(so it could also be a BCP what to put into Infrastucture ENUM ;-)
=20
-sta

________________________________

Von: Richard Shockey [mailto:richard@shockey.us]
Gesendet: Fr 02.06.2006 22:59
An: 'Creighton, Tom'; Stastny Richard; speermint@ietf.org
Cc: enum@ietf.org
Betreff: RE: [Enum] How to correctly set up the request URI?






This discussion is best kept in SPEERMINT. IMHO.

s/

your bewildered ENUM WG co-chair

> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Friday, June 02, 2006 15:41
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: Re: [Enum] How to correctly set up the request URI?
> >
> > Tom,
> >
> > being in favor of the usage of an AoR in Infrastructure ENUM, I have
> no
> > problems with your proposed words. There are only two issues:
> >
> > 1. There will be some people objecting, I assume.
>
> [Tom Creighton wrote:]
>
> I welcome any comments or recommendations that anyone has to help move
> beyond this discussion.
>
> > 2. Where should this words go?
> > -in the ENUM requirements or in the SPEERMINT requirements?
>
> [Tom Creighton wrote:]
>
> My initial reaction would be to keep it in SPEERMINT as this is a
> peering architecture issue and not a specific ENUM protocol issue, but
> as you said, advice from the 4 chairs would be appreciated.




_______________________________________________
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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I do not know whether I am writing in the right place. Please understand =
that users are  =20
Mobile to roam among providers and also prefer to maintain same identity =
when they switch providers.

Sohel=20

-----Original Message-----
From: "James McEachern" <jmce@nortel.com>
To: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>; "Lee, Yiu" =
<Yiu_Lee@cable.comcast.com>; "Stastny Richard" =
<Richard.Stastny@oefeg.at>; "speermint@ietf.org" <speermint@ietf.org>; =
"enum@ietf.org" <enum@ietf.org>
Sent: 6/2/06 3:51 PM
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

Ok, I think I see now.  Presumably I can have a SIP URI that clearly =
identifies me, but this is not the AOR that would be populated in =
Infrastructure ENUM.  That AOR would either be based on my E.164 number, =
or on some other random string created by the carrier.  Have I got it =
right?

James

-----Original Message-----
From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]=20
Sent: Friday, June 02, 2006 3:49 PM
To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard; =
speermint@ietf.org; enum@ietf.org
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?

I think we have two separate issues:

1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
defines an Address-of-Record as:

"Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
         that points to a domain with a location service that can map
         the URI to another URI where the user might be available.
         Typically, the location service is populated through
         registrations.  An AOR is frequently thought of as the "public
         address" of the user."

Based on that definition and the examples provided by Richard, a user's
privacy would not be compromised because it doesn't directly reach or
identify the customer.

2. There is confusion around what is or is not acceptable in the host
portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
matter what is placed in the host portion of the SIP URI as long as it
doesn't compromise a user's privacy.  So in the examples Richard
provided earlier, either are acceptable and would resolve in an
identical manner.

Tom

> -----Original Message-----
> From: James McEachern [mailto:jmce@nortel.com]
> Sent: Friday, June 02, 2006 15:12
> To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> In the past it has been suggested that if Infrastructure ENUM is
openly
> accessible, then for data privacy reasons it would not be acceptable
to
> list the user AOR.  Do we still believe this is the case, or do we now
> think providing the user AOR would be acceptable?
>=20
> James
>=20
> -----Original Message-----
> From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Sent: Thursday, June 01, 2006 2:22 PM
> To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Actually, I think it is sufficient for the Enum to return the user AOR
> because the routing decision should happen in the provB.net network.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:18 PM
> To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, so I am back to my original question:
>=20
> if I implement both types of public user identities:
> E.164 numbers and SIP AoRs, why doing the mapping to different hosting
> proxies in two places (in this case ENUM and the HSS), if I can do it
> only in one place?
>=20
> -sta
>=20
>=20
>=20
> ________________________________
>=20
> Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> Gesendet: Do 01.06.2006 20:10
> An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Yes. IMS is using Option 2. Option 1 is also possible if the operator
> decides to use just a redirect proxy.
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Thursday, June 01, 2006 2:07 PM
> To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> Cc: enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> so basically the IMS I-CSCF -  HSS - S-CSCF way?
>=20
> Richard
>=20
> ________________________________
>=20
> Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Gesendet: Do 01.06.2006 19:45
> An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> Cc: enum@ietf.org
> Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> Agreed, in this case the proxy associated with provB.net would be an
> intermediate device that would have to perform some sort of internal
> lookup in order to route the request to the next hop.
>=20
> > -----Original Message-----
> > From: Lee, Yiu
> > Sent: Thursday, June 01, 2006 13:34
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In this case, I guess when the originating proxy resolves the domain
> part
> > of the AOR (sip:+4319793321@provB.net), it will return the IP
address
> of
> > the root domain "provB.net" which may be a redirect proxy. So
> "provB.net"
> > has two choices:
> >
> > (1) It accesses its local routing table and sends a 305 to the
> originating
> > proxy with "sbc4711.provB.net" in the contact header.
> >
> > (2) It replaces the domain part in the r-uri to more specific like
> > "proxy.austria.provB.net" and forward the message to the next-hop.
> >
> > Either case, the assumption is provB.net will have a way to find out
> where
> > the home proxy of user +4319793321.
> >
> > This is only my guess.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 1:13 PM
> > To: Creighton, Tom; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> >
> > I see,
> >
> > but how do you solve this problem if the users can also be reached
by
> AoRs
> > direct?
> >
> > -sta
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 17:57
> > An: Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Enum] How to correctly set up the request URI?
> >
> >
> >
> > Maybe the service provider has some sort of segregated network
> (physically
> > or logically) and wants/needs to associate each its subscribers with
a
>=20
> > particular network segment.  If the provider went with the flat
model
> of
> > provB.net, each of its proxies would have to accept requests
destined
> to
> > all of its customers and would then have to route the request to the
> > appropriate region or division of its network.
> > If the provider went with the hierarchical approach of
> sbc4711.provB.net,
> > all of the requests would already be routed to the appropriate
ingress
>=20
> > point.  This could very well save a service provider a lot of money
in
>=20
> > route proxy equipment although might cost more on the provisioning
> side.
> >
> > That is my guess.
> >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 11:30
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Enum] How to correctly set up the request URI?
> > >
> > > Thank you, Tom
> > >
> > > just a minor additional question:
> > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > >
> > > If I have to do this according to RFC3263 anyway, why put the
> > > sbc4711.provB.net into ENUM in the first place?
> > >
> > > -sta
> > >
> > >
> > > > -----Original Message-----
> > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > To: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > To: speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Folks,
> > > > >
> > > > > Question for clarification:
> > > > >
> > > > > In the discussions here on how to populate sip URI in
> > Infrastructure
> > > > > ENUM to derive the Call Routing data for SPEERMINT  two lines
of
>=20
> > > > > thought exist:
> > > > >
> > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
the
>=20
> > > > > address-of-record of the user: e.g.
> > sip:+4319793321@provB.net
> > > > > 2. the ingress point into the provider network:
> > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > >
> > > > > The originating UAC has in this case set the To: field to
> > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
query
> at
> > > > > the proxy of provider A
> > > > >
> > > > > The question is how the proxy of provider A is now setting up
> the
> > > > > request URI in cases mentioned above?
> > > > >
> > > > > Does the UAS from provider B accept calls to
sbc7411.provB.net,
> or
> > > > > must the request URI set to the proper AoR as described in
> > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
ENUM.
> > > >
> > > > [Tom Creighton wrote:]
> > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does not
> > > > identify an address that the UAS is willing to accept requests
> for,
> > it
> > > > SHOULD reject the request with a 404 (Not Found) response."
> > > >
> > > > As long as the UAS from provider B is "willing" to accept the
> > request,
> > > > either should work.  Since provider B is populating
Infrastructure
> > > ENUM,
> > > > one would hope that they would coordinate this data with the
> > > > configuration of their proxies.
> > > >
> > > > Besides, how do you actually know that sbc4441.provB.net is
> actually
> > a
> > > > hostname and isn't a subdomain?
> > > >
> > > > In either case, after the provider A proxy performs an ENUM
query
> > and
> > > > rewrites the request URI, it should behave as a UAC and follow
the
>=20
> > > > procedures outlined in Section 4 of RFC3263 to determine the IP
> > > address,
> > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > >
> > > > > I would also like to draw the attention to section 3 of
> > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in ENUM.
> > > > >
> > > > > Regards
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > enum mailing list
> > > > > enum@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/enum
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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

_______________________________________________
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: "Khan, Sohel Q [CTO]" <Sohel.Q.Khan@sprint.com>,
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Sohel:

I don't believe that mobility has any bearing on this particular
discussion.  We are concentrating on the first part of the AoR
definition (referenced below) where a SIP or SIPS URI is populated in
Infrastructure ENUM and points to a domain (service provider) with a
location service.  In my mind, that provider-owned location service is
where the presence/mobility data would be consulted, thus allowing the
subscriber to maintain their identity regardless of where they roamed.
=20
Tom

> -----Original Message-----
> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> Sent: Friday, June 02, 2006 18:18
> To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> I do not know whether I am writing in the right place. Please
understand
> that users are
> Mobile to roam among providers and also prefer to maintain same
identity
> when they switch providers.
>=20
> Sohel
>=20
> -----Original Message-----
> From: "James McEachern" <jmce@nortel.com>
> To: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>; "Lee, Yiu"
> <Yiu_Lee@cable.comcast.com>; "Stastny Richard"
<Richard.Stastny@oefeg.at>;
> "speermint@ietf.org" <speermint@ietf.org>; "enum@ietf.org"
<enum@ietf.org>
> Sent: 6/2/06 3:51 PM
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> Ok, I think I see now.  Presumably I can have a SIP URI that clearly
> identifies me, but this is not the AOR that would be populated in
> Infrastructure ENUM.  That AOR would either be based on my E.164
number,
> or on some other random string created by the carrier.  Have I got it
> right?
>=20
> James
>=20
> -----Original Message-----
> From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Sent: Friday, June 02, 2006 3:49 PM
> To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard;
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
> I think we have two separate issues:
>=20
> 1. An issue of misunderstanding of the term AoR.  Section 6 of RFC3261
> defines an Address-of-Record as:
>=20
> "Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
>          that points to a domain with a location service that can map
>          the URI to another URI where the user might be available.
>          Typically, the location service is populated through
>          registrations.  An AOR is frequently thought of as the
"public
>          address" of the user."
>=20
> Based on that definition and the examples provided by Richard, a
user's
> privacy would not be compromised because it doesn't directly reach or
> identify the customer.
>=20
> 2. There is confusion around what is or is not acceptable in the host
> portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
> matter what is placed in the host portion of the SIP URI as long as it
> doesn't compromise a user's privacy.  So in the examples Richard
> provided earlier, either are acceptable and would resolve in an
> identical manner.
>=20
> Tom
>=20
> > -----Original Message-----
> > From: James McEachern [mailto:jmce@nortel.com]
> > Sent: Friday, June 02, 2006 15:12
> > To: Lee, Yiu; Stastny Richard; Creighton, Tom; speermint@ietf.org;
> > enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > In the past it has been suggested that if Infrastructure ENUM is
> openly
> > accessible, then for data privacy reasons it would not be acceptable
> to
> > list the user AOR.  Do we still believe this is the case, or do we
now
> > think providing the user AOR would be acceptable?
> >
> > James
> >
> > -----Original Message-----
> > From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > Sent: Thursday, June 01, 2006 2:22 PM
> > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > Actually, I think it is sufficient for the Enum to return the user
AOR
> > because the routing decision should happen in the provB.net network.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 2:18 PM
> > To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> > Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > Ok, so I am back to my original question:
> >
> > if I implement both types of public user identities:
> > E.164 numbers and SIP AoRs, why doing the mapping to different
hosting
> > proxies in two places (in this case ENUM and the HSS), if I can do
it
> > only in one place?
> >
> > -sta
> >
> >
> >
> > ________________________________
> >
> > Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > Gesendet: Do 01.06.2006 20:10
> > An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> >
> >
> > Yes. IMS is using Option 2. Option 1 is also possible if the
operator
> > decides to use just a redirect proxy.
> >
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Thursday, June 01, 2006 2:07 PM
> > To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> > Cc: enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > so basically the IMS I-CSCF -  HSS - S-CSCF way?
> >
> > Richard
> >
> > ________________________________
> >
> > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Gesendet: Do 01.06.2006 19:45
> > An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> > Cc: enum@ietf.org
> > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> >
> >
> > Agreed, in this case the proxy associated with provB.net would be an
> > intermediate device that would have to perform some sort of internal
> > lookup in order to route the request to the next hop.
> >
> > > -----Original Message-----
> > > From: Lee, Yiu
> > > Sent: Thursday, June 01, 2006 13:34
> > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > In this case, I guess when the originating proxy resolves the
domain
> > part
> > > of the AOR (sip:+4319793321@provB.net), it will return the IP
> address
> > of
> > > the root domain "provB.net" which may be a redirect proxy. So
> > "provB.net"
> > > has two choices:
> > >
> > > (1) It accesses its local routing table and sends a 305 to the
> > originating
> > > proxy with "sbc4711.provB.net" in the contact header.
> > >
> > > (2) It replaces the domain part in the r-uri to more specific like
> > > "proxy.austria.provB.net" and forward the message to the next-hop.
> > >
> > > Either case, the assumption is provB.net will have a way to find
out
> > where
> > > the home proxy of user +4319793321.
> > >
> > > This is only my guess.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 1:13 PM
> > > To: Creighton, Tom; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> > >
> > > I see,
> > >
> > > but how do you solve this problem if the users can also be reached
> by
> > AoRs
> > > direct?
> > >
> > > -sta
> > >
> > > ________________________________
> > >
> > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Gesendet: Do 01.06.2006 17:57
> > > An: Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Betreff: RE: [Enum] How to correctly set up the request URI?
> > >
> > >
> > >
> > > Maybe the service provider has some sort of segregated network
> > (physically
> > > or logically) and wants/needs to associate each its subscribers
with
> a
> >
> > > particular network segment.  If the provider went with the flat
> model
> > of
> > > provB.net, each of its proxies would have to accept requests
> destined
> > to
> > > all of its customers and would then have to route the request to
the
> > > appropriate region or division of its network.
> > > If the provider went with the hierarchical approach of
> > sbc4711.provB.net,
> > > all of the requests would already be routed to the appropriate
> ingress
> >
> > > point.  This could very well save a service provider a lot of
money
> in
> >
> > > route proxy equipment although might cost more on the provisioning
> > side.
> > >
> > > That is my guess.
> > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 11:30
> > > > To: Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > > Thank you, Tom
> > > >
> > > > just a minor additional question:
> > > >
> > > > > In either case, after the provider A proxy performs an ENUM
> query
> > > and
> > > > > rewrites the request URI, it should behave as a UAC and follow
> the
> >
> > > > > procedures outlined in Section 4 of RFC3263 to determine the
IP
> > > > address,
> > > > > port, and transport protocol for the provider B UAS.
> > > >
> > > > If I have to do this according to RFC3263 anyway, why put the
> > > > sbc4711.provB.net into ENUM in the first place?
> > > >
> > > > -sta
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > > To: Stastny Richard; speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > >
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > > To: speermint@ietf.org
> > > > > > Cc: enum@ietf.org
> > > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > > >
> > > > > > Folks,
> > > > > >
> > > > > > Question for clarification:
> > > > > >
> > > > > > In the discussions here on how to populate sip URI in
> > > Infrastructure
> > > > > > ENUM to derive the Call Routing data for SPEERMINT  two
lines
> of
> >
> > > > > > thought exist:
> > > > > >
> > > > > > The Enumservice SIP in Infrastructure ENUM should contain 1.
> the
> >
> > > > > > address-of-record of the user: e.g.
> > > sip:+4319793321@provB.net
> > > > > > 2. the ingress point into the provider network:
> > > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > > >
> > > > > > The originating UAC has in this case set the To: field to
> > > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > > sip:+4319793321provA.net;user=3Dphone. This caused the ENUM
> query
> > at
> > > > > > the proxy of provider A
> > > > > >
> > > > > > The question is how the proxy of provider A is now setting
up
> > the
> > > > > > request URI in cases mentioned above?
> > > > > >
> > > > > > Does the UAS from provider B accept calls to
> sbc7411.provB.net,
> > or
> > > > > > must the request URI set to the proper AoR as described in
> > > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
> ENUM.
> > > > >
> > > > > [Tom Creighton wrote:]
> > > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does
not
> > > > > identify an address that the UAS is willing to accept requests
> > for,
> > > it
> > > > > SHOULD reject the request with a 404 (Not Found) response."
> > > > >
> > > > > As long as the UAS from provider B is "willing" to accept the
> > > request,
> > > > > either should work.  Since provider B is populating
> Infrastructure
> > > > ENUM,
> > > > > one would hope that they would coordinate this data with the
> > > > > configuration of their proxies.
> > > > >
> > > > > Besides, how do you actually know that sbc4441.provB.net is
> > actually
> > > a
> > > > > hostname and isn't a subdomain?
> > > > >
> > > > > In either case, after the provider A proxy performs an ENUM
> query
> > > and
> > > > > rewrites the request URI, it should behave as a UAC and follow
> the
> >
> > > > > procedures outlined in Section 4 of RFC3263 to determine the
IP
> > > > address,
> > > > > port, and transport protocol for the provider B UAS.
> > > > >
> > > > > >
> > > > > > I would also like to draw the attention to section 3 of
> > > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
ENUM.
> > > > > >
> > > > > > Regards
> > > > > >
> > > > > > -sta
> > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > enum mailing list
> > > > > > enum@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/enum
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum
>=20
> _______________________________________________
> Speermint mailing list
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> https://www1.ietf.org/mailman/listinfo/speermint
>=20


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Date: Sat, 03 Jun 2006 08:11:24 -0500
From: Tim Dwight <timothy.dwight@verizon.com>
Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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	'James McEachern' <jmce@nortel.com>, "'Lee,
	Yiu'" <Yiu_Lee@cable.comcast.com>, 
	'Stastny Richard' <Richard.Stastny@oefeg.at>, speermint@ietf.org,
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What good does this globally-visible public-user-ID to private-user-ID
visible only within the serving network scheme do, if the means of
translating globally-visible public user ID to private user ID is publicly
accessible (public DNS)?  Or, are we assuming that it's not publicly
accessible, e.g., that the serving network has a split DNS and these "final
translations" are only visible within that network?

tim

> -----Original Message-----
> From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com] 
> Sent: Friday, June 02, 2006 3:58 PM
> To: James McEachern; Lee, Yiu; Stastny Richard; 
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up 
> the request URI?
> 
> That is correct.  The SIP URI that clearly identifies you, as 
> you described, would only be used within your provider's network.
> 
> > -----Original Message-----
> > From: James McEachern [mailto:jmce@nortel.com]
> > Sent: Friday, June 02, 2006 16:52
> > To: Creighton, Tom; Lee, Yiu; Stastny Richard; speermint@ietf.org; 
> > enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> > 
> > Ok, I think I see now.  Presumably I can have a SIP URI 
> that clearly 
> > identifies me, but this is not the AOR that would be populated in 
> > Infrastructure ENUM.  That AOR would either be based on my E.164
> number,
> > or on some other random string created by the carrier.  
> Have I got it 
> > right?
> > 
> > James
> > 
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Friday, June 02, 2006 3:49 PM
> > To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard; 
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> > 
> > I think we have two separate issues:
> > 
> > 1. An issue of misunderstanding of the term AoR.  Section 6 
> of RFC3261 
> > defines an Address-of-Record as:
> > 
> > "Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
> >          that points to a domain with a location service 
> that can map
> >          the URI to another URI where the user might be available.
> >          Typically, the location service is populated through
> >          registrations.  An AOR is frequently thought of as the
> "public
> >          address" of the user."
> > 
> > Based on that definition and the examples provided by Richard, a
> user's
> > privacy would not be compromised because it doesn't 
> directly reach or 
> > identify the customer.
> > 
> > 2. There is confusion around what is or is not acceptable 
> in the host 
> > portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't 
> > matter what is placed in the host portion of the SIP URI as 
> long as it 
> > doesn't compromise a user's privacy.  So in the examples Richard 
> > provided earlier, either are acceptable and would resolve in an 
> > identical manner.
> > 
> > Tom
> > 
> > > -----Original Message-----
> > > From: James McEachern [mailto:jmce@nortel.com]
> > > Sent: Friday, June 02, 2006 15:12
> > > To: Lee, Yiu; Stastny Richard; Creighton, Tom; 
> speermint@ietf.org; 
> > > enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > In the past it has been suggested that if Infrastructure ENUM is
> > openly
> > > accessible, then for data privacy reasons it would not be 
> acceptable
> > to
> > > list the user AOR.  Do we still believe this is the case, or do we
> now
> > > think providing the user AOR would be acceptable?
> > >
> > > James
> > >
> > > -----Original Message-----
> > > From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > > Sent: Thursday, June 01, 2006 2:22 PM
> > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > Actually, I think it is sufficient for the Enum to return the user
> AOR
> > > because the routing decision should happen in the 
> provB.net network.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 2:18 PM
> > > To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> > > Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > Ok, so I am back to my original question:
> > >
> > > if I implement both types of public user identities:
> > > E.164 numbers and SIP AoRs, why doing the mapping to different
> hosting
> > > proxies in two places (in this case ENUM and the HSS), if I can do
> it
> > > only in one place?
> > >
> > > -sta
> > >
> > >
> > >
> > > ________________________________
> > >
> > > Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > > Gesendet: Do 01.06.2006 20:10
> > > An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > >
> > >
> > > Yes. IMS is using Option 2. Option 1 is also possible if the
> operator
> > > decides to use just a redirect proxy.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 2:07 PM
> > > To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > so basically the IMS I-CSCF -  HSS - S-CSCF way?
> > >
> > > Richard
> > >
> > > ________________________________
> > >
> > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Gesendet: Do 01.06.2006 19:45
> > > An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > >
> > >
> > > Agreed, in this case the proxy associated with provB.net 
> would be an 
> > > intermediate device that would have to perform some sort 
> of internal 
> > > lookup in order to route the request to the next hop.
> > >
> > > > -----Original Message-----
> > > > From: Lee, Yiu
> > > > Sent: Thursday, June 01, 2006 13:34
> > > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > > request
> > > > URI?
> > > >
> > > > In this case, I guess when the originating proxy resolves the
> domain
> > > part
> > > > of the AOR (sip:+4319793321@provB.net), it will return the IP
> > address
> > > of
> > > > the root domain "provB.net" which may be a redirect proxy. So
> > > "provB.net"
> > > > has two choices:
> > > >
> > > > (1) It accesses its local routing table and sends a 305 to the
> > > originating
> > > > proxy with "sbc4711.provB.net" in the contact header.
> > > >
> > > > (2) It replaces the domain part in the r-uri to more 
> specific like 
> > > > "proxy.austria.provB.net" and forward the message to 
> the next-hop.
> > > >
> > > > Either case, the assumption is provB.net will have a way to find
> out
> > > where
> > > > the home proxy of user +4319793321.
> > > >
> > > > This is only my guess.
> > > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 1:13 PM
> > > > To: Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Speermint] Re: [Enum] How to correctly set up the
> request
> > > URI?
> > > >
> > > > I see,
> > > >
> > > > but how do you solve this problem if the users can also 
> be reached
> > by
> > > AoRs
> > > > direct?
> > > >
> > > > -sta
> > > >
> > > > ________________________________
> > > >
> > > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Gesendet: Do 01.06.2006 17:57
> > > > An: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Betreff: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > Maybe the service provider has some sort of segregated network
> > > (physically
> > > > or logically) and wants/needs to associate each its subscribers
> with
> > a
> > >
> > > > particular network segment.  If the provider went with the flat
> > model
> > > of
> > > > provB.net, each of its proxies would have to accept requests
> > destined
> > > to
> > > > all of its customers and would then have to route the request to
> the
> > > > appropriate region or division of its network.
> > > > If the provider went with the hierarchical approach of
> > > sbc4711.provB.net,
> > > > all of the requests would already be routed to the appropriate
> > ingress
> > >
> > > > point.  This could very well save a service provider a lot of
> money
> > in
> > >
> > > > route proxy equipment although might cost more on the 
> provisioning
> > > side.
> > > >
> > > > That is my guess.
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 11:30
> > > > > To: Creighton, Tom; speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Thank you, Tom
> > > > >
> > > > > just a minor additional question:
> > > > >
> > > > > > In either case, after the provider A proxy performs an ENUM
> > query
> > > > and
> > > > > > rewrites the request URI, it should behave as a UAC 
> and follow
> > the
> > >
> > > > > > procedures outlined in Section 4 of RFC3263 to determine the
> IP
> > > > > address,
> > > > > > port, and transport protocol for the provider B UAS.
> > > > >
> > > > > If I have to do this according to RFC3263 anyway, why put the 
> > > > > sbc4711.provB.net into ENUM in the first place?
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Creighton, Tom 
> [mailto:Tom_Creighton@cable.comcast.com]
> > > > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > > > To: Stastny Richard; speermint@ietf.org
> > > > > > Cc: enum@ietf.org
> > > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > > >
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > > > To: speermint@ietf.org
> > > > > > > Cc: enum@ietf.org
> > > > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > > > >
> > > > > > > Folks,
> > > > > > >
> > > > > > > Question for clarification:
> > > > > > >
> > > > > > > In the discussions here on how to populate sip URI in
> > > > Infrastructure
> > > > > > > ENUM to derive the Call Routing data for SPEERMINT  two
> lines
> > of
> > >
> > > > > > > thought exist:
> > > > > > >
> > > > > > > The Enumservice SIP in Infrastructure ENUM should 
> contain 1.
> > the
> > >
> > > > > > > address-of-record of the user: e.g.
> > > > sip:+4319793321@provB.net
> > > > > > > 2. the ingress point into the provider network:
> > > > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > > > >
> > > > > > > The originating UAC has in this case set the To: field to
> > > > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > > > sip:+4319793321provA.net;user=phone. This caused the ENUM
> > query
> > > at
> > > > > > > the proxy of provider A
> > > > > > >
> > > > > > > The question is how the proxy of provider A is now setting
> up
> > > the
> > > > > > > request URI in cases mentioned above?
> > > > > > >
> > > > > > > Does the UAS from provider B accept calls to
> > sbc7411.provB.net,
> > > or
> > > > > > > must the request URI set to the proper AoR as described in
> > > > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
> > ENUM.
> > > > > >
> > > > > > [Tom Creighton wrote:]
> > > > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does
> not
> > > > > > identify an address that the UAS is willing to 
> accept requests
> > > for,
> > > > it
> > > > > > SHOULD reject the request with a 404 (Not Found) response."
> > > > > >
> > > > > > As long as the UAS from provider B is "willing" to 
> accept the
> > > > request,
> > > > > > either should work.  Since provider B is populating
> > Infrastructure
> > > > > ENUM,
> > > > > > one would hope that they would coordinate this data 
> with the 
> > > > > > configuration of their proxies.
> > > > > >
> > > > > > Besides, how do you actually know that sbc4441.provB.net is
> > > actually
> > > > a
> > > > > > hostname and isn't a subdomain?
> > > > > >
> > > > > > In either case, after the provider A proxy performs an ENUM
> > query
> > > > and
> > > > > > rewrites the request URI, it should behave as a UAC 
> and follow
> > the
> > >
> > > > > > procedures outlined in Section 4 of RFC3263 to determine the
> IP
> > > > > address,
> > > > > > port, and transport protocol for the provider B UAS.
> > > > > >
> > > > > > >
> > > > > > > I would also like to draw the attention to section 3 of
> > > > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> ENUM.
> > > > > > >
> > > > > > > Regards
> > > > > > >
> > > > > > > -sta
> > > > > > >
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > enum mailing list
> > > > > > > enum@ietf.org
> > > > > > > https://www1.ietf.org/mailman/listinfo/enum
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > 
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum
> > 
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 


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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Sat, 3 Jun 2006 16:20:54 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
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Errors-To: speermint-bounces@ietf.org

I think this discussion is getting a bit confused in terminology
My I try to get this back in track from my knowledge:
=20
The terms Public User ID and Private User ID are from 3GPP =
specifications

A Public User ID is defined as an ID somebody may put on his business =
card:

Examples are:
a) tel:+4319793321 (a E.164 number)
b) sip:+4319793321@provider.com (an E.164 number and an =
address-of-record)
c) sip:richard.stastny@provider.com (also an AoR)
d) sip:richard@stastny.com
=20
All of these could be aliases linked to one Private User Identity
The private user identity is defined in 3GPP as a realm, used for =
registration
e.g. in the format richard.stastny @ provider.com
To be compatible with existing 2G systems, it could also be
derived from the IMSI with a predefined algorithm.
Private User Identities are our of scope in Speermint (I think)

a) cannot be user fro routing between SIP based networks, so it must be =
translated
to a SIP AoR. This can be done with ENUM. As we also stated here, not to =
disclose
and privacy, it should be in format b)
This mapping is publicly visible in Infrastructure ENUM

But this does not preclude me to put sip:richard.stastny@provider.com
or sip:richard@stastny.com on my business card, so it will
be used outside the providers network.

The routing of sip:richard.stastny@provider.com is
straightforward, for the mapping of the stastny.com part
sip:richard@stastny.com to provider.com an
entry (e.g. a NAPTR as Otmar described) is necessary
in the stastny.com domain to get the call proverly routed
to my network.

The mapping of the user part "richard" to "richard.stastny"
or +4319793321 takes place in the providers network
and is not visible outside

I hope this helps to clarify

-sta


________________________________

Von: Tim Dwight [mailto:timothy.dwight@verizon.com]
Gesendet: Sa 03.06.2006 15:11
An: 'Creighton, Tom'; 'James McEachern'; 'Lee, Yiu'; Stastny Richard; =
speermint@ietf.org; enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



What good does this globally-visible public-user-ID to private-user-ID
visible only within the serving network scheme do, if the means of
translating globally-visible public user ID to private user ID is =
publicly
accessible (public DNS)?  Or, are we assuming that it's not publicly
accessible, e.g., that the serving network has a split DNS and these =
"final
translations" are only visible within that network?

tim

> -----Original Message-----
> From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> Sent: Friday, June 02, 2006 3:58 PM
> To: James McEachern; Lee, Yiu; Stastny Richard;
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up
> the request URI?
>
> That is correct.  The SIP URI that clearly identifies you, as
> you described, would only be used within your provider's network.
>
> > -----Original Message-----
> > From: James McEachern [mailto:jmce@nortel.com]
> > Sent: Friday, June 02, 2006 16:52
> > To: Creighton, Tom; Lee, Yiu; Stastny Richard; speermint@ietf.org;
> > enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > Ok, I think I see now.  Presumably I can have a SIP URI
> that clearly
> > identifies me, but this is not the AOR that would be populated in
> > Infrastructure ENUM.  That AOR would either be based on my E.164
> number,
> > or on some other random string created by the carrier.=20
> Have I got it
> > right?
> >
> > James
> >
> > -----Original Message-----
> > From: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > Sent: Friday, June 02, 2006 3:49 PM
> > To: McEachern, James [CAR:5N00:EXCH]; Lee, Yiu; Stastny Richard;
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> request
> > URI?
> >
> > I think we have two separate issues:
> >
> > 1. An issue of misunderstanding of the term AoR.  Section 6
> of RFC3261
> > defines an Address-of-Record as:
> >
> > "Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI
> >          that points to a domain with a location service
> that can map
> >          the URI to another URI where the user might be available.
> >          Typically, the location service is populated through
> >          registrations.  An AOR is frequently thought of as the
> "public
> >          address" of the user."
> >
> > Based on that definition and the examples provided by Richard, a
> user's
> > privacy would not be compromised because it doesn't
> directly reach or
> > identify the customer.
> >
> > 2. There is confusion around what is or is not acceptable
> in the host
> > portion of a SIP URI.  As long as RFC3263 is followed, it shouldn't
> > matter what is placed in the host portion of the SIP URI as
> long as it
> > doesn't compromise a user's privacy.  So in the examples Richard
> > provided earlier, either are acceptable and would resolve in an
> > identical manner.
> >
> > Tom
> >
> > > -----Original Message-----
> > > From: James McEachern [mailto:jmce@nortel.com]
> > > Sent: Friday, June 02, 2006 15:12
> > > To: Lee, Yiu; Stastny Richard; Creighton, Tom;
> speermint@ietf.org;
> > > enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > In the past it has been suggested that if Infrastructure ENUM is
> > openly
> > > accessible, then for data privacy reasons it would not be
> acceptable
> > to
> > > list the user AOR.  Do we still believe this is the case, or do we
> now
> > > think providing the user AOR would be acceptable?
> > >
> > > James
> > >
> > > -----Original Message-----
> > > From: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > > Sent: Thursday, June 01, 2006 2:22 PM
> > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > Actually, I think it is sufficient for the Enum to return the user
> AOR
> > > because the routing decision should happen in the
> provB.net network.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 2:18 PM
> > > To: Lee, Yiu; Creighton, Tom; speermint@ietf.org
> > > Subject: Re: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > Ok, so I am back to my original question:
> > >
> > > if I implement both types of public user identities:
> > > E.164 numbers and SIP AoRs, why doing the mapping to different
> hosting
> > > proxies in two places (in this case ENUM and the HSS), if I can do
> it
> > > only in one place?
> > >
> > > -sta
> > >
> > >
> > >
> > > ________________________________
> > >
> > > Von: Lee, Yiu [mailto:Yiu_Lee@Cable.Comcast.com]
> > > Gesendet: Do 01.06.2006 20:10
> > > An: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > >
> > >
> > > Yes. IMS is using Option 2. Option 1 is also possible if the
> operator
> > > decides to use just a redirect proxy.
> > >
> > > -----Original Message-----
> > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > Sent: Thursday, June 01, 2006 2:07 PM
> > > To: Creighton, Tom; Lee, Yiu; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > > so basically the IMS I-CSCF -  HSS - S-CSCF way?
> > >
> > > Richard
> > >
> > > ________________________________
> > >
> > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > Gesendet: Do 01.06.2006 19:45
> > > An: Lee, Yiu; Stastny Richard; speermint@ietf.org
> > > Cc: enum@ietf.org
> > > Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the
> > request
> > > URI?
> > >
> > >
> > >
> > > Agreed, in this case the proxy associated with provB.net
> would be an
> > > intermediate device that would have to perform some sort
> of internal
> > > lookup in order to route the request to the next hop.
> > >
> > > > -----Original Message-----
> > > > From: Lee, Yiu
> > > > Sent: Thursday, June 01, 2006 13:34
> > > > To: Stastny Richard; Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
> > > request
> > > > URI?
> > > >
> > > > In this case, I guess when the originating proxy resolves the
> domain
> > > part
> > > > of the AOR (sip:+4319793321@provB.net), it will return the IP
> > address
> > > of
> > > > the root domain "provB.net" which may be a redirect proxy. So
> > > "provB.net"
> > > > has two choices:
> > > >
> > > > (1) It accesses its local routing table and sends a 305 to the
> > > originating
> > > > proxy with "sbc4711.provB.net" in the contact header.
> > > >
> > > > (2) It replaces the domain part in the r-uri to more
> specific like
> > > > "proxy.austria.provB.net" and forward the message to
> the next-hop.
> > > >
> > > > Either case, the assumption is provB.net will have a way to find
> out
> > > where
> > > > the home proxy of user +4319793321.
> > > >
> > > > This is only my guess.
> > > >
> > > > -----Original Message-----
> > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > Sent: Thursday, June 01, 2006 1:13 PM
> > > > To: Creighton, Tom; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Subject: [Speermint] Re: [Enum] How to correctly set up the
> request
> > > URI?
> > > >
> > > > I see,
> > > >
> > > > but how do you solve this problem if the users can also
> be reached
> > by
> > > AoRs
> > > > direct?
> > > >
> > > > -sta
> > > >
> > > > ________________________________
> > > >
> > > > Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
> > > > Gesendet: Do 01.06.2006 17:57
> > > > An: Stastny Richard; speermint@ietf.org
> > > > Cc: enum@ietf.org
> > > > Betreff: RE: [Enum] How to correctly set up the request URI?
> > > >
> > > >
> > > >
> > > > Maybe the service provider has some sort of segregated network
> > > (physically
> > > > or logically) and wants/needs to associate each its subscribers
> with
> > a
> > >
> > > > particular network segment.  If the provider went with the flat
> > model
> > > of
> > > > provB.net, each of its proxies would have to accept requests
> > destined
> > > to
> > > > all of its customers and would then have to route the request to
> the
> > > > appropriate region or division of its network.
> > > > If the provider went with the hierarchical approach of
> > > sbc4711.provB.net,
> > > > all of the requests would already be routed to the appropriate
> > ingress
> > >
> > > > point.  This could very well save a service provider a lot of
> money
> > in
> > >
> > > > route proxy equipment although might cost more on the
> provisioning
> > > side.
> > > >
> > > > That is my guess.
> > > >
> > > > > -----Original Message-----
> > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > Sent: Thursday, June 01, 2006 11:30
> > > > > To: Creighton, Tom; speermint@ietf.org
> > > > > Cc: enum@ietf.org
> > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > >
> > > > > Thank you, Tom
> > > > >
> > > > > just a minor additional question:
> > > > >
> > > > > > In either case, after the provider A proxy performs an ENUM
> > query
> > > > and
> > > > > > rewrites the request URI, it should behave as a UAC
> and follow
> > the
> > >
> > > > > > procedures outlined in Section 4 of RFC3263 to determine the
> IP
> > > > > address,
> > > > > > port, and transport protocol for the provider B UAS.
> > > > >
> > > > > If I have to do this according to RFC3263 anyway, why put the
> > > > > sbc4711.provB.net into ENUM in the first place?
> > > > >
> > > > > -sta
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Creighton, Tom
> [mailto:Tom_Creighton@cable.comcast.com]
> > > > > > Sent: Thursday, June 01, 2006 4:36 PM
> > > > > > To: Stastny Richard; speermint@ietf.org
> > > > > > Cc: enum@ietf.org
> > > > > > Subject: RE: [Enum] How to correctly set up the request URI?
> > > > > >
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > > > > > > Sent: Thursday, June 01, 2006 09:06
> > > > > > > To: speermint@ietf.org
> > > > > > > Cc: enum@ietf.org
> > > > > > > Subject: [Enum] How to correctly set up the request URI?
> > > > > > >
> > > > > > > Folks,
> > > > > > >
> > > > > > > Question for clarification:
> > > > > > >
> > > > > > > In the discussions here on how to populate sip URI in
> > > > Infrastructure
> > > > > > > ENUM to derive the Call Routing data for SPEERMINT  two
> lines
> > of
> > >
> > > > > > > thought exist:
> > > > > > >
> > > > > > > The Enumservice SIP in Infrastructure ENUM should
> contain 1.
> > the
> > >
> > > > > > > address-of-record of the user: e.g.
> > > > sip:+4319793321@provB.net
> > > > > > > 2. the ingress point into the provider network:
> > > > > > > e.g. sip:+4319793321@sbc4711.provB.net
> > > > > > >
> > > > > > > The originating UAC has in this case set the To: field to
> > > > > > > tel:+4319793321 and the Request-URI to e.g.
> > > > > > > sip:+4319793321provA.net;user=3Dphone. This caused the =
ENUM
> > query
> > > at
> > > > > > > the proxy of provider A
> > > > > > >
> > > > > > > The question is how the proxy of provider A is now setting
> up
> > > the
> > > > > > > request URI in cases mentioned above?
> > > > > > >
> > > > > > > Does the UAS from provider B accept calls to
> > sbc7411.provB.net,
> > > or
> > > > > > > must the request URI set to the proper AoR as described in
> > > > > > > 8.2.2.1 of RFC 3261? If yes, only option 1 is possible in
> > ENUM.
> > > > > >
> > > > > > [Tom Creighton wrote:]
> > > > > > From section 8.2.2.1 of RFC3261:    "If the Request-URI does
> not
> > > > > > identify an address that the UAS is willing to
> accept requests
> > > for,
> > > > it
> > > > > > SHOULD reject the request with a 404 (Not Found) response."
> > > > > >
> > > > > > As long as the UAS from provider B is "willing" to
> accept the
> > > > request,
> > > > > > either should work.  Since provider B is populating
> > Infrastructure
> > > > > ENUM,
> > > > > > one would hope that they would coordinate this data
> with the
> > > > > > configuration of their proxies.
> > > > > >
> > > > > > Besides, how do you actually know that sbc4441.provB.net is
> > > actually
> > > > a
> > > > > > hostname and isn't a subdomain?
> > > > > >
> > > > > > In either case, after the provider A proxy performs an ENUM
> > query
> > > > and
> > > > > > rewrites the request URI, it should behave as a UAC
> and follow
> > the
> > >
> > > > > > procedures outlined in Section 4 of RFC3263 to determine the
> IP
> > > > > address,
> > > > > > port, and transport protocol for the provider B UAS.
> > > > > >
> > > > > > >
> > > > > > > I would also like to draw the attention to section 3 of
> > > > > > > RFC3761 (SIP Enumservice) explaining the usage of AoR in
> ENUM.
> > > > > > >
> > > > > > > Regards
> > > > > > >
> > > > > > > -sta
> > > > > > >
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > enum mailing list
> > > > > > > enum@ietf.org
> > > > > > > https://www1.ietf.org/mailman/listinfo/enum
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> >
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum
> >
> > _______________________________________________
> > enum mailing list
> > enum@ietf.org
> > https://www1.ietf.org/mailman/listinfo/enum
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>




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From speermint-bounces@ietf.org Sat Jun 03 10:42:39 2006
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Sat, 3 Jun 2006 10:42:25 -0400
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Well Tony we continue to agree on some things after all ..

You point is well taken.  ENUM IS more than a flexible routing table. Its
value is in the ability to authenticate and maintain authoritative bindings
for E164 to URI bindings.

How those bindings are used may or may not involve the use of DNS
technology.

Consequently the critical issue is the TRUST created by and maintained by
the authoritative Registry of those bindings and that Registry MAY and in
all likelihood MUST maintain some mechanism for the maintenance of
information about providers and users.

Which is why we did the work on IRIS in the ENUM WG.

BTW Which EU directive you mention IYHO will require the maintenance of a
pin trap of ENUM queries?


> -----Original Message-----
> From: Tony Rutkowski [mailto:trutkowski@verisign.com]
> Sent: Friday, June 02, 2006 6:06 PM
> To: Stastny Richard; Pfautz, Penn L, GBLAM; James McEachern;
> speermint@ietf.org; enum@ietf.org
> Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> 
> Hi Richard,
> 
> >I do not fully understand what the CALEA issue has to do with the
> >question which
> >URI is put in ENUM.
> 
> ENUM is little more than a more flexible routing table
> - a more dynamic LERG.  Stable public infrastructures
> - for network integrity, law enforcement support, and
> a host of other requirements - require some ability
> to authenticate changes and maintain authoritative
> bindings to provider/user information.  Austrian
> police cannot do lawful interceptions if they can't
> trust the E.164 to URI bindings and do queries to
> get authoritative information about providers and
> users.
> 
> >I have complete different concerns regarding CALEA, in principle I do
> >not understand why the service providers are bothered with this at all,
> 
> European authorities will be much better supported
> under the EU Data Retention Directive - which will
> require capturing and timestamping every signalling
> query.
> 
> --tony
> 
> 
> _______________________________________________
> enum mailing list
> enum@ietf.org
> https://www1.ietf.org/mailman/listinfo/enum


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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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> -----Original Message-----
> From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
> Sent: Friday, June 02, 2006 4:56 PM
> To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request
> URI?
> 
> Yes. Again, there might be exceptions, for example a business might want
> a URI with their name to be used and some SPs might agree to that, but
> generally the point of infrastructure ENUM is to be invisible to the
> user
> 


Now the 64B question is the "infrastructure" URI invisible to the global
DNS?


> _______________________________________________
> Speermint mailing list
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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
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From: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>
To: <richard@shockey.us>, "Pfautz, Penn L, GBLAM" <ppfautz@att.com>,
	"James McEachern" <jmce@nortel.com>,
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Who has the authority to make that decision?  Is it the ITU, IETF, a
combination of the two, or each country that opts to participate in
Infrastructure ENUM?  I think that is the 64B question.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Saturday, June 03, 2006 10:48
> To: 'Pfautz, Penn L, GBLAM'; 'James McEachern'; Creighton, Tom; Lee,
Yiu;
> 'Stastny Richard'; speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>=20
>=20
>=20
> > -----Original Message-----
> > From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
> > Sent: Friday, June 02, 2006 4:56 PM
> > To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> >
> > Yes. Again, there might be exceptions, for example a business might
want
> > a URI with their name to be used and some SPs might agree to that,
but
> > generally the point of infrastructure ENUM is to be invisible to the
> > user
> >
>=20
>=20
> Now the 64B question is the "infrastructure" URI invisible to the
global
> DNS?
>=20
>=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint


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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <richard@shockey.us>,
	"Pfautz, Penn L, GBLAM" <ppfautz@att.com>, <speermint@ietf.org>,
	<enum@ietf.org>
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>> > Yes. Again, there might be exceptions, for example a business might =
want
>> > a URI with their name to be used and some SPs might agree to that, =
but
>> > generally the point of infrastructure ENUM is to be invisible to =
the
>> > user
>> >
>> Now the 64B question is the "infrastructure" URI invisible to the =
global
>> DNS?
>Who has the authority to make that decision?  Is it the ITU, IETF, a
>combination of the two, or each country that opts to participate in
>Infrastructure ENUM?  I think that is the 64B question

?
I am confused
Could anybody please explain to me what you are talking about
I cannot parse this thread

What is the "point of Infrastructure ENUM"?
What is an "Infrastructure URI" and why should it be invisible?
And what has this to do with an enterprise specific URI?

-sta



________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Mo 05.06.2006 17:15
An: richard@shockey.us; Pfautz, Penn L, GBLAM; James McEachern; Lee, =
Yiu; Stastny Richard; speermint@ietf.org; enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request =
URI?



Who has the authority to make that decision?  Is it the ITU, IETF, a
combination of the two, or each country that opts to participate in
Infrastructure ENUM?  I think that is the 64B question.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Saturday, June 03, 2006 10:48
> To: 'Pfautz, Penn L, GBLAM'; 'James McEachern'; Creighton, Tom; Lee,
Yiu;
> 'Stastny Richard'; speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
>
>
> > -----Original Message-----
> > From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
> > Sent: Friday, June 02, 2006 4:56 PM
> > To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> >
> > Yes. Again, there might be exceptions, for example a business might
want
> > a URI with their name to be used and some SPs might agree to that,
but
> > generally the point of infrastructure ENUM is to be invisible to the
> > user
> >
>
>
> Now the 64B question is the "infrastructure" URI invisible to the
global
> DNS?
>
>
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint




_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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Subject: RE: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Mon, 5 Jun 2006 11:57:18 -0400
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From: "Pfautz, Penn L, GBLAM" <ppfautz@att.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <richard@shockey.us>,
	<speermint@ietf.org>, <enum@ietf.org>
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My  original statement about invisibility was that the end user doesn't
need to specify/know the URI that the SP provides in infrastructure ENUM
- or even that infrastructure ENUM even exists at all.

The issue seems to have evolved to that of how much of the
infrastructure ENUM tree and its products will be openly visible. This
is a decision that many parties may have some level of input to.
National regulators may well be able to veto public visibility if they
so choose. Service providers may wish to control resolution to some
degree although making the infrastructure ENUM Tier 1 inaccessible would
not seem to be something the IETF would want to do other things being
equal.

Penn

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Monday, June 05, 2006 11:49 AM
To: Creighton, Tom; richard@shockey.us; Pfautz, Penn L, GBLAM;
speermint@ietf.org; enum@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

>> > Yes. Again, there might be exceptions, for example a business might
want
>> > a URI with their name to be used and some SPs might agree to that,
but
>> > generally the point of infrastructure ENUM is to be invisible to
the
>> > user
>> >
>> Now the 64B question is the "infrastructure" URI invisible to the
global
>> DNS?
>Who has the authority to make that decision?  Is it the ITU, IETF, a
>combination of the two, or each country that opts to participate in
>Infrastructure ENUM?  I think that is the 64B question

?
I am confused
Could anybody please explain to me what you are talking about
I cannot parse this thread

What is the "point of Infrastructure ENUM"?
What is an "Infrastructure URI" and why should it be invisible?
And what has this to do with an enterprise specific URI?

-sta



________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Mo 05.06.2006 17:15
An: richard@shockey.us; Pfautz, Penn L, GBLAM; James McEachern; Lee,
Yiu; Stastny Richard; speermint@ietf.org; enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Who has the authority to make that decision?  Is it the ITU, IETF, a
combination of the two, or each country that opts to participate in
Infrastructure ENUM?  I think that is the 64B question.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Saturday, June 03, 2006 10:48
> To: 'Pfautz, Penn L, GBLAM'; 'James McEachern'; Creighton, Tom; Lee,
Yiu;
> 'Stastny Richard'; speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
>
>
> > -----Original Message-----
> > From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
> > Sent: Friday, June 02, 2006 4:56 PM
> > To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> >
> > Yes. Again, there might be exceptions, for example a business might
want
> > a URI with their name to be used and some SPs might agree to that,
but
> > generally the point of infrastructure ENUM is to be invisible to the
> > user
> >
>
>
> Now the 64B question is the "infrastructure" URI invisible to the
global
> DNS?
>
>
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint




_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



From speermint-bounces@ietf.org Mon Jun 05 12:13:19 2006
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Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request URI?
Date: Mon, 5 Jun 2006 18:17:22 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
To: "Pfautz, Penn L, GBLAM" <ppfautz@att.com>,
	"Creighton, Tom" <Tom_Creighton@cable.comcast.com>, <richard@shockey.us>,
	<speermint@ietf.org>, <enum@ietf.org>
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>My  original statement about invisibility was that the end user doesn't
>need to specify/know the URI that the SP provides in infrastructure =
ENUM
>- or even that infrastructure ENUM even exists at all.

ok=20

>The issue seems to have evolved to that of how much of the
>infrastructure ENUM tree and its products will be openly visible. This
>is a decision that many parties may have some level of input to.
>National regulators may well be able to veto public visibility if they
>so choose.=20

I agree that national regulators may have some saying on what kind
of URIs should be placed in Infastructure ENUM (e.g. none that
jeopardizes the end-user privacy), but I do not see why the regulators
do not want to have the Infrastructure ENUM data not public at all.
=20
I already said that a security and privacy section is missing in
draft-ietf-enum-infrastructure

The Infrastructure ENUM data is basically providing a E.164 to SPID=20
mapping, and part of this is available on every regulators web-site
(the block assignment). What is not is the numbers ported out of
this blocks and this seems more to be a concern of the providers
and not of the regulators.

The problem here is that this information hiding is more to other
providers and not to the public.

But then a non public Infrastructure ENUM does not help either ;-)
=20
-sta
=20
>Service providers may wish to control resolution to some
>degree although making the infrastructure ENUM Tier 1 inaccessible =
would
>not seem to be something the IETF would want to do other things being
>equal.

Penn

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
Sent: Monday, June 05, 2006 11:49 AM
To: Creighton, Tom; richard@shockey.us; Pfautz, Penn L, GBLAM;
speermint@ietf.org; enum@ietf.org
Subject: Re: [Speermint] Re: [Enum] How to correctly set up the request
URI?

>> > Yes. Again, there might be exceptions, for example a business might
want
>> > a URI with their name to be used and some SPs might agree to that,
but
>> > generally the point of infrastructure ENUM is to be invisible to
the
>> > user
>> >
>> Now the 64B question is the "infrastructure" URI invisible to the
global
>> DNS?
>Who has the authority to make that decision?  Is it the ITU, IETF, a
>combination of the two, or each country that opts to participate in
>Infrastructure ENUM?  I think that is the 64B question

?
I am confused
Could anybody please explain to me what you are talking about
I cannot parse this thread

What is the "point of Infrastructure ENUM"?
What is an "Infrastructure URI" and why should it be invisible?
And what has this to do with an enterprise specific URI?

-sta



________________________________

Von: Creighton, Tom [mailto:Tom_Creighton@cable.comcast.com]
Gesendet: Mo 05.06.2006 17:15
An: richard@shockey.us; Pfautz, Penn L, GBLAM; James McEachern; Lee,
Yiu; Stastny Richard; speermint@ietf.org; enum@ietf.org
Betreff: RE: [Speermint] Re: [Enum] How to correctly set up the request
URI?



Who has the authority to make that decision?  Is it the ITU, IETF, a
combination of the two, or each country that opts to participate in
Infrastructure ENUM?  I think that is the 64B question.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Saturday, June 03, 2006 10:48
> To: 'Pfautz, Penn L, GBLAM'; 'James McEachern'; Creighton, Tom; Lee,
Yiu;
> 'Stastny Richard'; speermint@ietf.org; enum@ietf.org
> Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> URI?
>
>
>
> > -----Original Message-----
> > From: Pfautz, Penn L, GBLAM [mailto:ppfautz@att.com]
> > Sent: Friday, June 02, 2006 4:56 PM
> > To: James McEachern; Creighton, Tom; Lee, Yiu; Stastny Richard;
> > speermint@ietf.org; enum@ietf.org
> > Subject: RE: [Speermint] Re: [Enum] How to correctly set up the
request
> > URI?
> >
> > Yes. Again, there might be exceptions, for example a business might
want
> > a URI with their name to be used and some SPs might agree to that,
but
> > generally the point of infrastructure ENUM is to be invisible to the
> > user
> >
>
>
> Now the 64B question is the "infrastructure" URI invisible to the
global
> DNS?
>
>
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint





_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Tue, 6 Jun 2006 10:50:52 -0400
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All, I am back from vacation and find a very long and interesting
thread.  I was trying to hold off responding till I read all 130+
messages but am finding it difficult to remember and resist, so pardon
me if I restate something here.

First, the goal is to reach domains, so the routing to E.164 numbers
should be out of scope -- no reinventing TRIP.  We turn E.164 into sip
addresses and then route on the domains.

>From the domain we should be able to look up direct signaling points or
indirectly policy points.  In some cases, we may find federation
information, then need to think about how to route to that federation.

I think the discussion about number of domains and BGP-like stuff is
barking up the wrong tree.  Oh, BTW there are about 3000 carriers in the
USA alone and roughly 8000 public enterprises (publicly traded stock) in
the USA.  But, I think that misses the point.

I believe the reality will be that there will ultimately be two options:
 - from one SP (carrier/enterprise) to another there is direct peering,
 - from one SP to another there is no direct peering and a "Tier 1" SP
will be used for transit.

There will be some rather large SPs (think ATT, Sprint, name you
favorite PTT) that will be all too willing to be the transit carrier and
solve the bridging problem, and many enterprises or individuals will be
all too willing to let them do that.

The federations may thus be a means of aggregating the sip routing
problem and making it scalable.  The federation then becomes a default
route for the Tier 2 SP.  I suspect that the sequence will look
something like:

E.164 -> sip:E.164@termination-SP.com

   - at this point the originating SP can determine if it has a peering
throug internal means,
   - if not:

termination-SP.com -> terminating-federation.com

   - at this point the originating SP can determine if it belongs to
that same federation,
   - if not, the end result will be:

Request-URI: sip:E.164@termination-SP.com   -- points to target SP
Route: originating-federation.com           -- points to near side if
not common federation
Route: terminating-federation.com           -- points to far side
federation

originating-federation -> translates to my transit SP.

I may have abused the federation concept a bit, given that there is both
the qualitative aspect of having a common agreement on how to
interconnect, and the routing aspect which anticipates that a federation
may be formed around a large SP that has the IP network to connect to
all the SPs that sign up for that "federation".  I'll confess that there
is some legacy precedent here.  But, that may be the most viable short
term evolution path.

My thinking, if it is not obvious is that the optimal number of SPs is 2
(direct peering).
Second best is 3 (one transit). =20
Maximum number is 4 (one originating and one terminating transit). =20

I believe that hopping across 10 SIP signaling networks would be a sure
sign that our work here has failed, and that is what I fear the BGP
discussion would lead us to.

Mike


> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
> Sent: Saturday, May 27, 2006 7:26 AM
> To: Reinaldo Penno; Otmar Lendl
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> >I wonder how
> >LNP is solved within wireless...A specific entry for each=20
> phone? That=20
> >would be crazy.
>=20
> Ahem, ENUM?
>=20
> I thought that is all what ENUM is about?
>=20
> Richard
>=20
>=20
> ________________________________
>=20
> Von: Reinaldo Penno [mailto:rpenno@juniper.net]
> Gesendet: Sa 27.05.2006 11:04
> An: Otmar Lendl; Patrick Melampy
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org
> Betreff: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
>=20
>=20
> Hello,
>=20
> I believe the crux of the problem was missed.
>=20
> The issue is not really the number of domains. We have today=20
> some 400.000.000 million hosts on the Internet. BGP peers to=20
> do not exchange 400.000.000 million routes, they exchange in=20
> the low hundreds of thousands.
>=20
> And why is that? (I guess you know where I'm getting at).=20
> That's because IP addresses lend themselves very well to=20
> hierarchical deployment, compression through prefix/CIDR=20
> usage and the like. DNS names are strings and hence difficult=20
> to come up with a similar scheme.
>=20
>=20
> I believe that if domains need to be exchanged between peers:
>=20
> 1 - Just a limited set would be exchanged and a "default=20
> layer 5 route"
> (equivalent to a traditional default route) would be used.
>=20
> 2 - Compression schemes like *.company.com, regular=20
> expressions or the like would need to be used. Having=20
> specific routes to AORs would be hairy (maybe for the=20
> eventual roaming or visiting user). I wonder how LNP is=20
> solved within wireless...A specific entry for each phone?=20
> That would be crazy.
>=20
> 3 - Some new scheme that will make somebody rich....until then..
>=20
> I believe subscriptions would be used to exchange mostly=20
> "cost" and policy information and some domains. I use cost=20
> here in the layer 3 sense (number of calls, jitter,=20
> whatever). Cost would be an abstract value or a more specific=20
> one (up to the administrator).
>=20
> Regards,
>=20
> Reinaldo
>=20
> > -----Original Message-----
> > From: Otmar Lendl [mailto:lendl@nic.at]
> > Sent: Saturday, May 27, 2006 1:29 AM
> > To: Patrick Melampy
> > Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org
> > Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering=20
> Architecture
> >
> > On 2006/05/26 18:05, Patrick Melampy=20
> <PMelampy@acmepacket.com> wrote:
> > > I suppose like in any distributed routing protocol, there are some
> > chicken
> > > an egg issues. But consider the following.
> > >
> > > PREREQUISITES:
> > > 1.) You know the destination DOMAIN from either ENUM or=20
> other means.
> >
> > [...]
> > >
> > > This technique is very similar to TRIP, only its for=20
> DOMAINS and not
> > partial
> > > E.164 numbers.
> >
> > [...]
> >
> > > Obviously, the data set could get large, as the list of=20
> domains can
> be
> > > really big. There are some possibilities here...
> >
> > Let's do a quick estimate on the number of domains involved:
> >
> > Right now, operators usually give customers URIs of the form=20
> > <number>@<providerdomain>. As long as this is the state of affairs,=20
> > we're fine as the number of providers is quite finite (a few=20
> > thousands, perhaps).
> >
> > This is like the state of affairs with respect to email, anno 1990.
> > Back then customers started to notice that if they use their own
> domain
> > for email, then they can switch providers without having to change=20
> > their email address. Thus a *lot* of people opted to use their own=20
> > domain for email.
> >
> > My guess is that the same will happen to SIP. Once customers print=20
> > their SIP uris on business cards, they will want to use their own=20
> > domains. If providers won't support that feature then I=20
> suspect that=20
> > the regulators will step in and mandate an URI portability solution=20
> > just as they did in the PSTN with respect to numbers from number
> blocks.
> > (This could get really nasty protocol-wise, so I guess=20
> using customer-=20
> > owned domains is the far better and likelier solution to the porting
> > question.)
> >
> > So: Anything we come up with here in SPEERMINT needs to cope with a=20
> > scenario where the use of customer domains in SIP URIs is just as=20
> > widespread as it is for email.
> >
> > How many domains are used for email? That's hard to say. There are=20
> > about 60 million gTLD domains and probably more than 20 ccTLD ones.
> > Assuming that only a tenth of that is used for email we're at
> > 8 million domains. Given the exponential growth any system which=20
> > doesn't scale to at least 10 million domains will be=20
> obsolete before=20
> > the ink is dry on the RFC.
> >
> > So what happens if we do some sort of BGP or TRIP with domains?
> >
> > Let's say we need at least 100 bytes of state information=20
> per domain.
> > That makes a full routing table 1 GB worth of data. That also means=20
> > that a new border element needs to exchange that amount of=20
> data with=20
> > its peers before it has learned the current state of the routing=20
> > table.
> >
> > That is some serious amount of data.
> >
> > > 1.) Have notification point to a published document, and the
> document
> > lists
> > > all of the relevant reachability in real time. Thus any change to
> the
> > > document would generate a NOTIFICATION.
> >
> > + a download of up to 1 GB per peer.
> >
> > > 2.) User partial domains, broken up by the "dots". This may allow
> some
> > > wildcarding to reduce the number of domains. For=20
> instance, a carrier
> > like
> > > Verizon may have many 100's of domains attached to it, and version
> is
> > > willing to route and manage SIP traffic to any of them. The NOTIFY
> could
> > > contain something like: Verizon.com or *.verizon.com to indicate
> that
> > any
> > > domain in Verizon is reachable.
> >
> > That's helpful if a provider uses subdomains for internal purposes,=20
> > but that won't help you with customer owned domains.
> >
> > > This technique may work -- and fit nicely into an=20
> existing protocol.
> All
> > we
> > > would need to do is define the data models (XML?) and or documents
> for
> > > exchanging.
> >
> > I think this technique can work -- if you add an aggregation step=20
> > before the routing scheme.
> >
> > Take another look at BGP: there you have the concept of an=20
> Autonomous=20
> > System and not only that of prefixes. Maybe you need something=20
> > similar.
> >
> > One approach to cope with hosted SIP domains could be to use the=20
> > hostnames as found in SRV records. E.g. for settings like
> >
> > _sip._tcp.customer.domain IN SRV 10 10 5060 sip.provider.com
> >
> > run the routing algorithm on "sip.provider.com" instead of=20
> > "customer.domain".
> >
> > Another option is to store some AS-equivalent in the=20
> customer domain=20
> > and have the routing algorithm operate on that. In my domain-policy=20
> > framework this could be expressed e.g. like
> >
> > customers.domain IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > stating that this domain can be reached using the Speermint routing=20
> > logic by using the learned route to RoutingID 1042.
> >
> > Or, even better, use non-terminals, and thus refer to the ingress=20
> > policy stored in the provider's domain:
> >
> > customers.domain IN NAPTR "" 10 10 "D2P+SIP" "" provider.com
> >
> > provider.com. IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > (+ whatever other facts provider.com wants to announce about his
> > reachability.)
> >
> > This combination of our ideas can indeed work and scale to an=20
> > unlimited number of customer-owned domains.
> >
> > /ol
> > --
> > < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



From speermint-bounces@ietf.org Wed Jun 07 10:35:31 2006
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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
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To: "Michael Hammer (mhammer)" <mhammer@cisco.com>,
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Mike - I agree with what you are saying here, or more importantly how I
am reading what you are saying.  Some points below:

-It's important to avoid the "hopping across 10 SIP signaling networks"
scenario. AGREED.
-I think it's important to state and acknowledge that we aren't going to
ever see one big globally pertinent "CRD database" in the sky that is
mutually owned and shared.  Hence the value of "routing to a Federation"
instead of the ultimate terminating network. Does it make any difference
that access to the Federation is via a transitory relationship (your
"tier 1" provider example)?!?!  If both the originating and terminating
networks were members of the same Federation, then a direct relationship
can be established and or facilitated be the "authoritative" body of the
Federation.
-It's been identified on this list before that there is a finite limit
to the number of "peering relationships" that an orig domain can
maintain effective. With that in mind, there must be some summary
mechanism.  As you note, the ultimate goal is to reach domains.  That
reachability can either be direct, or through Federation alignment.  The
Federation concept alleviates the burden of both the central db in the
sky, as well as of the orig domain to peer with everyone to facilitate a
full mesh. =20

Otmar Lendl via draft-lendl-speermint-federations-00 suggests a method
to disseminate Federation membership, but this is via DNS, and therefore
out of scope for Speermint.  At the end of the day, Speermint routes to
either the terminating domain and/or the Federation where terminating
domain is a member. =20

Adam

-----Original Message-----
From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]=20
Sent: Tuesday, June 06, 2006 10:51 AM
To: Stastny Richard; Reinaldo Penno; Otmar Lendl
Cc: speermint@ietf.org
Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture

All, I am back from vacation and find a very long and interesting
thread.  I was trying to hold off responding till I read all 130+
messages but am finding it difficult to remember and resist, so pardon
me if I restate something here.

First, the goal is to reach domains, so the routing to E.164 numbers
should be out of scope -- no reinventing TRIP.  We turn E.164 into sip
addresses and then route on the domains.

>From the domain we should be able to look up direct signaling points or
indirectly policy points.  In some cases, we may find federation
information, then need to think about how to route to that federation.

I think the discussion about number of domains and BGP-like stuff is
barking up the wrong tree.  Oh, BTW there are about 3000 carriers in the
USA alone and roughly 8000 public enterprises (publicly traded stock) in
the USA.  But, I think that misses the point.

I believe the reality will be that there will ultimately be two options:
 - from one SP (carrier/enterprise) to another there is direct peering,
 - from one SP to another there is no direct peering and a "Tier 1" SP
will be used for transit.

There will be some rather large SPs (think ATT, Sprint, name you
favorite PTT) that will be all too willing to be the transit carrier and
solve the bridging problem, and many enterprises or individuals will be
all too willing to let them do that.

The federations may thus be a means of aggregating the sip routing
problem and making it scalable.  The federation then becomes a default
route for the Tier 2 SP.  I suspect that the sequence will look
something like:

E.164 -> sip:E.164@termination-SP.com

   - at this point the originating SP can determine if it has a peering
throug internal means,
   - if not:

termination-SP.com -> terminating-federation.com

   - at this point the originating SP can determine if it belongs to
that same federation,
   - if not, the end result will be:

Request-URI: sip:E.164@termination-SP.com   -- points to target SP
Route: originating-federation.com           -- points to near side if
not common federation
Route: terminating-federation.com           -- points to far side
federation

originating-federation -> translates to my transit SP.

I may have abused the federation concept a bit, given that there is both
the qualitative aspect of having a common agreement on how to
interconnect, and the routing aspect which anticipates that a federation
may be formed around a large SP that has the IP network to connect to
all the SPs that sign up for that "federation".  I'll confess that there
is some legacy precedent here.  But, that may be the most viable short
term evolution path.

My thinking, if it is not obvious is that the optimal number of SPs is 2
(direct peering).
Second best is 3 (one transit). =20
Maximum number is 4 (one originating and one terminating transit). =20

I believe that hopping across 10 SIP signaling networks would be a sure
sign that our work here has failed, and that is what I fear the BGP
discussion would lead us to.

Mike


> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Saturday, May 27, 2006 7:26 AM
> To: Reinaldo Penno; Otmar Lendl
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> >I wonder how
> >LNP is solved within wireless...A specific entry for each
> phone? That
> >would be crazy.
>=20
> Ahem, ENUM?
>=20
> I thought that is all what ENUM is about?
>=20
> Richard
>=20
>=20
> ________________________________
>=20
> Von: Reinaldo Penno [mailto:rpenno@juniper.net]
> Gesendet: Sa 27.05.2006 11:04
> An: Otmar Lendl; Patrick Melampy
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org
> Betreff: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
>=20
>=20
> Hello,
>=20
> I believe the crux of the problem was missed.
>=20
> The issue is not really the number of domains. We have today some=20
> 400.000.000 million hosts on the Internet. BGP peers to do not=20
> exchange 400.000.000 million routes, they exchange in the low hundreds

> of thousands.
>=20
> And why is that? (I guess you know where I'm getting at).=20
> That's because IP addresses lend themselves very well to hierarchical=20
> deployment, compression through prefix/CIDR usage and the like. DNS=20
> names are strings and hence difficult to come up with a similar=20
> scheme.
>=20
>=20
> I believe that if domains need to be exchanged between peers:
>=20
> 1 - Just a limited set would be exchanged and a "default layer 5=20
> route"
> (equivalent to a traditional default route) would be used.
>=20
> 2 - Compression schemes like *.company.com, regular expressions or the

> like would need to be used. Having specific routes to AORs would be=20
> hairy (maybe for the eventual roaming or visiting user). I wonder how=20
> LNP is solved within wireless...A specific entry for each phone?
> That would be crazy.
>=20
> 3 - Some new scheme that will make somebody rich....until then..
>=20
> I believe subscriptions would be used to exchange mostly "cost" and=20
> policy information and some domains. I use cost here in the layer 3=20
> sense (number of calls, jitter, whatever). Cost would be an abstract=20
> value or a more specific one (up to the administrator).
>=20
> Regards,
>=20
> Reinaldo
>=20
> > -----Original Message-----
> > From: Otmar Lendl [mailto:lendl@nic.at]
> > Sent: Saturday, May 27, 2006 1:29 AM
> > To: Patrick Melampy
> > Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org
> > Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering
> Architecture
> >
> > On 2006/05/26 18:05, Patrick Melampy
> <PMelampy@acmepacket.com> wrote:
> > > I suppose like in any distributed routing protocol, there are some
> > chicken
> > > an egg issues. But consider the following.
> > >
> > > PREREQUISITES:
> > > 1.) You know the destination DOMAIN from either ENUM or
> other means.
> >
> > [...]
> > >
> > > This technique is very similar to TRIP, only its for
> DOMAINS and not
> > partial
> > > E.164 numbers.
> >
> > [...]
> >
> > > Obviously, the data set could get large, as the list of
> domains can
> be
> > > really big. There are some possibilities here...
> >
> > Let's do a quick estimate on the number of domains involved:
> >
> > Right now, operators usually give customers URIs of the form=20
> > <number>@<providerdomain>. As long as this is the state of affairs,=20
> > we're fine as the number of providers is quite finite (a few=20
> > thousands, perhaps).
> >
> > This is like the state of affairs with respect to email, anno 1990.
> > Back then customers started to notice that if they use their own
> domain
> > for email, then they can switch providers without having to change=20
> > their email address. Thus a *lot* of people opted to use their own=20
> > domain for email.
> >
> > My guess is that the same will happen to SIP. Once customers print=20
> > their SIP uris on business cards, they will want to use their own=20
> > domains. If providers won't support that feature then I
> suspect that
> > the regulators will step in and mandate an URI portability solution=20
> > just as they did in the PSTN with respect to numbers from number
> blocks.
> > (This could get really nasty protocol-wise, so I guess
> using customer-
> > owned domains is the far better and likelier solution to the porting
> > question.)
> >
> > So: Anything we come up with here in SPEERMINT needs to cope with a=20
> > scenario where the use of customer domains in SIP URIs is just as=20
> > widespread as it is for email.
> >
> > How many domains are used for email? That's hard to say. There are=20
> > about 60 million gTLD domains and probably more than 20 ccTLD ones.
> > Assuming that only a tenth of that is used for email we're at
> > 8 million domains. Given the exponential growth any system which=20
> > doesn't scale to at least 10 million domains will be
> obsolete before
> > the ink is dry on the RFC.
> >
> > So what happens if we do some sort of BGP or TRIP with domains?
> >
> > Let's say we need at least 100 bytes of state information
> per domain.
> > That makes a full routing table 1 GB worth of data. That also means=20
> > that a new border element needs to exchange that amount of
> data with
> > its peers before it has learned the current state of the routing=20
> > table.
> >
> > That is some serious amount of data.
> >
> > > 1.) Have notification point to a published document, and the
> document
> > lists
> > > all of the relevant reachability in real time. Thus any change to
> the
> > > document would generate a NOTIFICATION.
> >
> > + a download of up to 1 GB per peer.
> >
> > > 2.) User partial domains, broken up by the "dots". This may allow
> some
> > > wildcarding to reduce the number of domains. For
> instance, a carrier
> > like
> > > Verizon may have many 100's of domains attached to it, and version
> is
> > > willing to route and manage SIP traffic to any of them. The NOTIFY
> could
> > > contain something like: Verizon.com or *.verizon.com to indicate
> that
> > any
> > > domain in Verizon is reachable.
> >
> > That's helpful if a provider uses subdomains for internal purposes,=20
> > but that won't help you with customer owned domains.
> >
> > > This technique may work -- and fit nicely into an
> existing protocol.
> All
> > we
> > > would need to do is define the data models (XML?) and or documents
> for
> > > exchanging.
> >
> > I think this technique can work -- if you add an aggregation step=20
> > before the routing scheme.
> >
> > Take another look at BGP: there you have the concept of an
> Autonomous
> > System and not only that of prefixes. Maybe you need something=20
> > similar.
> >
> > One approach to cope with hosted SIP domains could be to use the=20
> > hostnames as found in SRV records. E.g. for settings like
> >
> > _sip._tcp.customer.domain IN SRV 10 10 5060 sip.provider.com
> >
> > run the routing algorithm on "sip.provider.com" instead of=20
> > "customer.domain".
> >
> > Another option is to store some AS-equivalent in the
> customer domain
> > and have the routing algorithm operate on that. In my domain-policy=20
> > framework this could be expressed e.g. like
> >
> > customers.domain IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > stating that this domain can be reached using the Speermint routing=20
> > logic by using the learned route to RoutingID 1042.
> >
> > Or, even better, use non-terminals, and thus refer to the ingress=20
> > policy stored in the provider's domain:
> >
> > customers.domain IN NAPTR "" 10 10 "D2P+SIP" "" provider.com
> >
> > provider.com. IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > (+ whatever other facts provider.com wants to announce about his
> > reachability.)
> >
> > This combination of our ideas can indeed work and scale to an=20
> > unlimited number of customer-owned domains.
> >
> > /ol
> > --
> > < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
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> https://www1.ietf.org/mailman/listinfo/speermint
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>-I think it's important to state and acknowledge that we aren't going =
to
>ever see one big globally pertinent "CRD database" in the sky that is
>mutually owned and shared. =20
=20
I always thought that is the DNS, and how to use it is defined in RFC =
3263.
If you add draft-lendl-speermint-federations, you basically have it
=20
BTE, the DNS is not out of scope ;-)
=20
-sta

________________________________

Von: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]
Gesendet: Mi 07.06.2006 16:35
An: Michael Hammer (mhammer); Stastny Richard; Reinaldo Penno; Otmar =
Lendl
Cc: speermint@ietf.org
Betreff: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture



Mike - I agree with what you are saying here, or more importantly how I
am reading what you are saying.  Some points below:

-It's important to avoid the "hopping across 10 SIP signaling networks"
scenario. AGREED.
-I think it's important to state and acknowledge that we aren't going to
ever see one big globally pertinent "CRD database" in the sky that is
mutually owned and shared.  Hence the value of "routing to a Federation"
instead of the ultimate terminating network. Does it make any difference
that access to the Federation is via a transitory relationship (your
"tier 1" provider example)?!?!  If both the originating and terminating
networks were members of the same Federation, then a direct relationship
can be established and or facilitated be the "authoritative" body of the
Federation.
-It's been identified on this list before that there is a finite limit
to the number of "peering relationships" that an orig domain can
maintain effective. With that in mind, there must be some summary
mechanism.  As you note, the ultimate goal is to reach domains.  That
reachability can either be direct, or through Federation alignment.  The
Federation concept alleviates the burden of both the central db in the
sky, as well as of the orig domain to peer with everyone to facilitate a
full mesh.=20

Otmar Lendl via draft-lendl-speermint-federations-00 suggests a method
to disseminate Federation membership, but this is via DNS, and therefore
out of scope for Speermint.  At the end of the day, Speermint routes to
either the terminating domain and/or the Federation where terminating
domain is a member.=20

Adam

-----Original Message-----
From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
Sent: Tuesday, June 06, 2006 10:51 AM
To: Stastny Richard; Reinaldo Penno; Otmar Lendl
Cc: speermint@ietf.org
Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture

All, I am back from vacation and find a very long and interesting
thread.  I was trying to hold off responding till I read all 130+
messages but am finding it difficult to remember and resist, so pardon
me if I restate something here.

First, the goal is to reach domains, so the routing to E.164 numbers
should be out of scope -- no reinventing TRIP.  We turn E.164 into sip
addresses and then route on the domains.

>From the domain we should be able to look up direct signaling points or
indirectly policy points.  In some cases, we may find federation
information, then need to think about how to route to that federation.

I think the discussion about number of domains and BGP-like stuff is
barking up the wrong tree.  Oh, BTW there are about 3000 carriers in the
USA alone and roughly 8000 public enterprises (publicly traded stock) in
the USA.  But, I think that misses the point.

I believe the reality will be that there will ultimately be two options:
 - from one SP (carrier/enterprise) to another there is direct peering,
 - from one SP to another there is no direct peering and a "Tier 1" SP
will be used for transit.

There will be some rather large SPs (think ATT, Sprint, name you
favorite PTT) that will be all too willing to be the transit carrier and
solve the bridging problem, and many enterprises or individuals will be
all too willing to let them do that.

The federations may thus be a means of aggregating the sip routing
problem and making it scalable.  The federation then becomes a default
route for the Tier 2 SP.  I suspect that the sequence will look
something like:

E.164 -> sip:E.164@termination-SP.com

   - at this point the originating SP can determine if it has a peering
throug internal means,
   - if not:

termination-SP.com -> terminating-federation.com

   - at this point the originating SP can determine if it belongs to
that same federation,
   - if not, the end result will be:

Request-URI: sip:E.164@termination-SP.com   -- points to target SP
Route: originating-federation.com           -- points to near side if
not common federation
Route: terminating-federation.com           -- points to far side
federation

originating-federation -> translates to my transit SP.

I may have abused the federation concept a bit, given that there is both
the qualitative aspect of having a common agreement on how to
interconnect, and the routing aspect which anticipates that a federation
may be formed around a large SP that has the IP network to connect to
all the SPs that sign up for that "federation".  I'll confess that there
is some legacy precedent here.  But, that may be the most viable short
term evolution path.

My thinking, if it is not obvious is that the optimal number of SPs is 2
(direct peering).
Second best is 3 (one transit).=20
Maximum number is 4 (one originating and one terminating transit).=20

I believe that hopping across 10 SIP signaling networks would be a sure
sign that our work here has failed, and that is what I fear the BGP
discussion would lead us to.

Mike


> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Saturday, May 27, 2006 7:26 AM
> To: Reinaldo Penno; Otmar Lendl
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>
> >I wonder how
> >LNP is solved within wireless...A specific entry for each
> phone? That
> >would be crazy.
>
> Ahem, ENUM?
>
> I thought that is all what ENUM is about?
>
> Richard
>
>
> ________________________________
>
> Von: Reinaldo Penno [mailto:rpenno@juniper.net]
> Gesendet: Sa 27.05.2006 11:04
> An: Otmar Lendl; Patrick Melampy
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org
> Betreff: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>
>
>
> Hello,
>
> I believe the crux of the problem was missed.
>
> The issue is not really the number of domains. We have today some
> 400.000.000 million hosts on the Internet. BGP peers to do not
> exchange 400.000.000 million routes, they exchange in the low hundreds

> of thousands.
>
> And why is that? (I guess you know where I'm getting at).
> That's because IP addresses lend themselves very well to hierarchical
> deployment, compression through prefix/CIDR usage and the like. DNS
> names are strings and hence difficult to come up with a similar
> scheme.
>
>
> I believe that if domains need to be exchanged between peers:
>
> 1 - Just a limited set would be exchanged and a "default layer 5
> route"
> (equivalent to a traditional default route) would be used.
>
> 2 - Compression schemes like *.company.com, regular expressions or the

> like would need to be used. Having specific routes to AORs would be
> hairy (maybe for the eventual roaming or visiting user). I wonder how
> LNP is solved within wireless...A specific entry for each phone?
> That would be crazy.
>
> 3 - Some new scheme that will make somebody rich....until then..
>
> I believe subscriptions would be used to exchange mostly "cost" and
> policy information and some domains. I use cost here in the layer 3
> sense (number of calls, jitter, whatever). Cost would be an abstract
> value or a more specific one (up to the administrator).
>
> Regards,
>
> Reinaldo
>
> > -----Original Message-----
> > From: Otmar Lendl [mailto:lendl@nic.at]
> > Sent: Saturday, May 27, 2006 1:29 AM
> > To: Patrick Melampy
> > Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org
> > Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering
> Architecture
> >
> > On 2006/05/26 18:05, Patrick Melampy
> <PMelampy@acmepacket.com> wrote:
> > > I suppose like in any distributed routing protocol, there are some
> > chicken
> > > an egg issues. But consider the following.
> > >
> > > PREREQUISITES:
> > > 1.) You know the destination DOMAIN from either ENUM or
> other means.
> >
> > [...]
> > >
> > > This technique is very similar to TRIP, only its for
> DOMAINS and not
> > partial
> > > E.164 numbers.
> >
> > [...]
> >
> > > Obviously, the data set could get large, as the list of
> domains can
> be
> > > really big. There are some possibilities here...
> >
> > Let's do a quick estimate on the number of domains involved:
> >
> > Right now, operators usually give customers URIs of the form
> > <number>@<providerdomain>. As long as this is the state of affairs,
> > we're fine as the number of providers is quite finite (a few
> > thousands, perhaps).
> >
> > This is like the state of affairs with respect to email, anno 1990.
> > Back then customers started to notice that if they use their own
> domain
> > for email, then they can switch providers without having to change
> > their email address. Thus a *lot* of people opted to use their own
> > domain for email.
> >
> > My guess is that the same will happen to SIP. Once customers print
> > their SIP uris on business cards, they will want to use their own
> > domains. If providers won't support that feature then I
> suspect that
> > the regulators will step in and mandate an URI portability solution
> > just as they did in the PSTN with respect to numbers from number
> blocks.
> > (This could get really nasty protocol-wise, so I guess
> using customer-
> > owned domains is the far better and likelier solution to the porting
> > question.)
> >
> > So: Anything we come up with here in SPEERMINT needs to cope with a
> > scenario where the use of customer domains in SIP URIs is just as
> > widespread as it is for email.
> >
> > How many domains are used for email? That's hard to say. There are
> > about 60 million gTLD domains and probably more than 20 ccTLD ones.
> > Assuming that only a tenth of that is used for email we're at
> > 8 million domains. Given the exponential growth any system which
> > doesn't scale to at least 10 million domains will be
> obsolete before
> > the ink is dry on the RFC.
> >
> > So what happens if we do some sort of BGP or TRIP with domains?
> >
> > Let's say we need at least 100 bytes of state information
> per domain.
> > That makes a full routing table 1 GB worth of data. That also means
> > that a new border element needs to exchange that amount of
> data with
> > its peers before it has learned the current state of the routing
> > table.
> >
> > That is some serious amount of data.
> >
> > > 1.) Have notification point to a published document, and the
> document
> > lists
> > > all of the relevant reachability in real time. Thus any change to
> the
> > > document would generate a NOTIFICATION.
> >
> > + a download of up to 1 GB per peer.
> >
> > > 2.) User partial domains, broken up by the "dots". This may allow
> some
> > > wildcarding to reduce the number of domains. For
> instance, a carrier
> > like
> > > Verizon may have many 100's of domains attached to it, and version
> is
> > > willing to route and manage SIP traffic to any of them. The NOTIFY
> could
> > > contain something like: Verizon.com or *.verizon.com to indicate
> that
> > any
> > > domain in Verizon is reachable.
> >
> > That's helpful if a provider uses subdomains for internal purposes,
> > but that won't help you with customer owned domains.
> >
> > > This technique may work -- and fit nicely into an
> existing protocol.
> All
> > we
> > > would need to do is define the data models (XML?) and or documents
> for
> > > exchanging.
> >
> > I think this technique can work -- if you add an aggregation step
> > before the routing scheme.
> >
> > Take another look at BGP: there you have the concept of an
> Autonomous
> > System and not only that of prefixes. Maybe you need something
> > similar.
> >
> > One approach to cope with hosted SIP domains could be to use the
> > hostnames as found in SRV records. E.g. for settings like
> >
> > _sip._tcp.customer.domain IN SRV 10 10 5060 sip.provider.com
> >
> > run the routing algorithm on "sip.provider.com" instead of
> > "customer.domain".
> >
> > Another option is to store some AS-equivalent in the
> customer domain
> > and have the routing algorithm operate on that. In my domain-policy
> > framework this could be expressed e.g. like
> >
> > customers.domain IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > stating that this domain can be reached using the Speermint routing
> > logic by using the learned route to RoutingID 1042.
> >
> > Or, even better, use non-terminals, and thus refer to the ingress
> > policy stored in the provider's domain:
> >
> > customers.domain IN NAPTR "" 10 10 "D2P+SIP" "" provider.com
> >
> > provider.com. IN NAPTR 10 10 "u" "D2P+SIP:route-id"
> >       "!.*!urn:ietf:speermint:RID:1042!" .
> >
> > (+ whatever other facts provider.com wants to announce about his
> > reachability.)
> >
> > This combination of our ideas can indeed work and scale to an
> > unlimited number of customer-owned domains.
> >
> > /ol
> > --
> > < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>
> _______________________________________________
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>
>
>
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>

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I make the out of scope statement because I am of the mindset that the
SPEERMINT scope kicks in after resolution of the AoR - to me this means
that Speermint can assume knowledge of domain and federation from DNS,
and then use that CRD appropriately.  I might have this wrong though -
but that was what I was thinking.

-Adam=20

-----Original Message-----
From: Otmar Lendl [mailto:lendl@nic.at]=20
Sent: Wednesday, June 07, 2006 12:11 PM
To: Uzelac, Adam
Cc: Michael Hammer (mhammer); Stastny Richard; Reinaldo Penno;
speermint@ietf.org
Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture

On 2006/06/07 16:06, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
wrote:
>
> -It's been identified on this list before that there is a finite limit

> to the number of "peering relationships" that an orig domain can=20
> maintain effective. With that in mind, there must be some summary=20
> mechanism.  As you note, the ultimate goal is to reach domains.  That=20
> reachability can either be direct, or through Federation alignment. =20
> The Federation concept alleviates the burden of both the central db in

> the sky, as well as of the orig domain to peer with everyone to=20
> facilitate a full mesh.

Correct.

> Otmar Lendl via draft-lendl-speermint-federations-00 suggests a method

> to disseminate Federation membership, but this is via DNS, and=20
> therefore out of scope for Speermint.

ENUM is out of scope for speermint, as it deals with E.164 numbers.

draft-lendl-speermint-federations-00 operates on the domain part of SIP
URIs. I see no conflict with the SPEERMINT charter.

Can you elaborate why you think that
draft-lendl-speermint-federations-00
is out of scope?

/ol
--
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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From: Otmar Lendl <lendl@nic.at>
To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
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On 2006/06/07 16:06, "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com> wrote:
>
> -It's been identified on this list before that there is a finite limit
> to the number of "peering relationships" that an orig domain can
> maintain effective. With that in mind, there must be some summary
> mechanism.  As you note, the ultimate goal is to reach domains.  That
> reachability can either be direct, or through Federation alignment.  The
> Federation concept alleviates the burden of both the central db in the
> sky, as well as of the orig domain to peer with everyone to facilitate a
> full mesh.  

Correct.

> Otmar Lendl via draft-lendl-speermint-federations-00 suggests a method
> to disseminate Federation membership, but this is via DNS, and therefore
> out of scope for Speermint.  

ENUM is out of scope for speermint, as it deals with E.164 numbers.

draft-lendl-speermint-federations-00 operates on the domain part of
SIP URIs. I see no conflict with the SPEERMINT charter.

Can you elaborate why you think that draft-lendl-speermint-federations-00 
is out of scope?

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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Last week, Otmar asked the following questions to the working group:

* Is there interest in an I-D which defines the semantic of
   policy rules which can be used to form ad-hoc federations?

* Is there interest in another presentation regarding my
   domain policy DDDS proposal at the upcoming IETF meeting?

He got a resounding zero responses (perhaps because the question was 
buried in a very long thread).

I say "yes" to both questions.  Mostly because I'm unsure what other 
problems this working group is suppose to solve.

-andy


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From speermint-bounces@ietf.org Wed Jun 07 15:38:32 2006
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Andrew,

Question - are you saying that good points and questions getting buried
in very long threads?  Never seen that before in IETF mailing lists!  ;)
Great catch plucking these 2 questions out of that thread.  I would like
to understand better the "policy rules", and I think it would apply
beyond ad hoc. I am not sure I completely understand the "ad hoc" aspect
of the question, but it doesn't really matter. =20

As for the domain policy proposal presentation in Montreal, I believe
that would also kick a most lively discussion about policy exchange via
DNS or SIP. I personally see merit in both. =20

Adam

-----Original Message-----
From: Andrew Newton [mailto:andy@hxr.us]=20
Sent: Wednesday, June 07, 2006 1:52 PM
To: speermint@ietf.org
Subject: [Speermint] A question about federations

Last week, Otmar asked the following questions to the working group:

* Is there interest in an I-D which defines the semantic of
   policy rules which can be used to form ad-hoc federations?

* Is there interest in another presentation regarding my
   domain policy DDDS proposal at the upcoming IETF meeting?

He got a resounding zero responses (perhaps because the question was
buried in a very long thread).

I say "yes" to both questions.  Mostly because I'm unsure what other
problems this working group is suppose to solve.

-andy


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fyi=20

-----Original Message-----
From: IETF Administrative Director [mailto:iad@ietf.org]=20
Sent: Thursday, June 08, 2006 11:52 AM
To: IETF Announcement list
Subject: IETF 67 Location Announcement=20

The IETF is pleased to announce its selection of San Diego, California,
USA and the Sheraton San Diego Hotel & Marina as the site of IETF 67
being held November
5 - 10, 2006.

As we celebrate IETF's 20th year we return to the city where IETF 01
(with 21
attendees) was held, as well as IETFs' 9, 23, 49, and 60. The Sheraton
has added another 18,000 sq ft of meeting space for us to use this year.

Note Well: There are other conferences being held in San Diego so we
encourage you to reserve your hotel rooms right away.

We have arrangements on Harbor Island with:

Sheraton San Diego Hotel & Marina for 770 rooms and 50 rooms at the
Hilton San Diego Airport/Marina, both properties with free Internet in
the rooms.  More information can be found at:
http://www.ietf.org/meetings/hotels_67.html

Registration for San Diego will open July 12th.

We are looking forward to San Diego in November, but until then we will
see you in Montreal in 31 days!

And speaking of Montreal, Early Bird registration with its $550
registration fee expires at 1600 GMT on 30 June!  If you have not
registered yet, delay no longer!
http://www.ietf.org/meetings/IETF-66.html

Ray Pelletier
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> From: ietf-secretariat@ietf.org
> To: ietf-announce@ietf.org
> Cc:=20
> Subject: Internet-Drafts Submission Cutoff Dates for the 66th IETF Meeting
>  in Montreal, Quebec, Canada=20
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>=20
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This is TENTATIVE...

-----Original Message-----
From: IETF Secretariat=20
Subject: SPEERMINT - Requested session has been scheduled for IETF 66=20

The sessions that you have requested have been scheduled.
Below is the scheduled session information followed by the information
of sessions that you have requested.

SPEERMINT Session 1 (2 hours)
Monday, Afternoon Session I 1300-1500
Room Name: Room 520DEF
----------------------------------------------

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From speermint-bounces@ietf.org Mon Jun 12 21:34:33 2006
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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Mon, 12 Jun 2006 21:34:27 -0400
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I'd like to comment as follows:

You said:
>  I believe the reality will be that there will ultimately be two options:
>   - from one SP (carrier/enterprise) to another there is direct peering,
>   - from one SP to another there is no direct peering and a "Tier 1" SP
>  will be used for transit.

Unless somebody representing a carrier makes this statement, it's safe to
say we are just guessing. I find this statement your making to lack any
backing. How many Autonomous Systems do your communications go through
today? When you're at work? When you're at home? I'm guessing you have no
idea. Why should this group ever designate a role for a Tier 1 SP as a
required internetworking element? If you worked at a competitive Tier 2 you
might take exception to your statement.

> My thinking, if it is not obvious is that the optimal number of SPs is 2
> (direct peering).

Optimal in whose frame of reference? The internet's? The caller? The callee?
The carrier(s)? My guess is your thinking of the internet. 

> Second best is 3 (one transit).  
Why is one transit carrier better than 2 or 3? Is it cost? Quality? Where
did this conclusion come from? Are there any IP based models where a single
backbone carrier is considered optimal? Doesn't this limit competition and
choice? Wouldn't it be anti-competitive, and thus not be net-neutral?

>Maximum number is 4 (one originating and one terminating transit).

Why is this the maxium? 

I hate to argue with your statements, but they should be better defended.
There are a lot of managed networks that are both legislated and or
pre-existing. The owners of long haul fiber networks want to use them for
signaled media. My guess is we will not only have One Tier 1 to choose from,
but the following:

1.) Each Enterprise will desire at least 2 interconnect carriers to ensure
they have the lowest cost and best level of service with redundancy.

2.) Each Tier 1 Carrier will want to purchase termination services for its
customers and have multiple paths to each major desination domain

3.) Many Tier 2 and Tier 3 carriers will exist by selecting specific target
markets and verticals.

4.) Governments will continue to try to enforce regulations guaranteeing a
competitive landscape for communication interconnects (requiring access for
2nd and 3rd carrier options.

I could go on and on. I would suggest that we embrace the realities of our
world as requirements for this standards group.


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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Tue, 13 Jun 2006 10:38:09 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Patrick Melampy" <PMelampy@acmepacket.com>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Reinaldo Penno" <rpenno@juniper.net>, "Otmar Lendl" <lendl@nic.at>
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Patrick,

Let us take these one at a time.  Inline.=20

> -----Original Message-----
> From: Patrick Melampy [mailto:PMelampy@acmepacket.com]=20
> Sent: Monday, June 12, 2006 9:34 PM
> To: Michael Hammer (mhammer); 'Stastny Richard'; 'Reinaldo=20
> Penno'; 'Otmar Lendl'
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> I'd like to comment as follows:
>=20
> You said:
> >  I believe the reality will be that there will ultimately=20
> be two options:
> >   - from one SP (carrier/enterprise) to another there is=20
> direct peering,
> >   - from one SP to another there is no direct peering and a=20
> "Tier 1"=20
> > SP  will be used for transit.
>=20
> Unless somebody representing a carrier makes this statement,=20
> it's safe to say we are just guessing. I find this statement=20
> your making to lack any backing. How many Autonomous Systems=20
> do your communications go through today? When you're at work?=20
> When you're at home? I'm guessing you have no idea. Why=20
> should this group ever designate a role for a Tier 1 SP as a=20
> required internetworking element? If you worked at a=20
> competitive Tier 2 you might take exception to your statement.

The basis for this statement is an assumption that for security and
economic reasons, you must have rules of engagement in order to connect
to a peer.  If there is a business arrangement, you can directly
connect, if not then an intermediary is used.  This is rather simple and
straight forward.  This does not preclude you creating a
wide-open-to-the-Internet federation through which anyone can connect to
you.

I am simply stating there are two options:  direct and indirect.  Is
there a third possibility?

=20
> > My thinking, if it is not obvious is that the optimal=20
> number of SPs is=20
> > 2 (direct peering).
>=20
> Optimal in whose frame of reference? The internet's? The=20
> caller? The callee?
> The carrier(s)? My guess is your thinking of the internet.=20

The most optimal is directly from client to client, ie. true P2P.  But,
I am relaxing that here to the assumption that an end device may
"outsource" some degree of features and functionality to a SP, be that
enterprise PBX or carrier system.

It is optimal because we are talking about two parties, the caller and
called parties.  For that you could have an origination system and a
termination system.  Anything beyond that is extraneous middlemen that
add cost into the picture, but not much in the way of added value, and a
lot of possibility for unintended consequences to screw up the signaling
or media.

Case in point:  Every additional signaling point on the path will
exacerbate call setup delay and increase the complexity of the
interactions amongst all these components.

Case in point:  Every additional media anchor will add delay to the RTP
stream, thus negatively affecting QoS.

So, from the perspective of the caller and callee, the service
experience will be better if SPs can minimize the number of extra moving
parts in the system.

And, from the perspective of the SP, the fewer middlemen in the picture,
the fewer the profits to share amongst those middlemen.

I am just applying Occam's Razor to get the minimalist set.

=20
> > Second best is 3 (one transit). =20
> Why is one transit carrier better than 2 or 3? Is it cost?=20
> Quality? Where did this conclusion come from? Are there any=20
> IP based models where a single backbone carrier is considered=20
> optimal? Doesn't this limit competition and choice? Wouldn't=20
> it be anti-competitive, and thus not be net-neutral?

I am suggesting that if both caller and callee have the good fortune to
prefer the same transit carrier, then the savings from avoiding a second
SP can be shared.  This is what is experienced when a call stays within
a single enterprise.  This is what happens when you have a "friends and
family" scenario.

It is both a business advantage and a technical advantage to have only
one SP's policies to care about.  No interworking!

=20
> >Maximum number is 4 (one originating and one terminating transit).
>=20
> Why is this the maxium?=20

What would a third tranist carrier provide that could not be better
provided if transit carrier 1 and 2 had a direct signaling agreement
between them?  How is sharing revenues amongst 3 carriers better than
shared betweeen 2?


> I hate to argue with your statements, but they should be=20
> better defended.

Hopefully, I have done so. :)


> There are a lot of managed networks that are both legislated=20
> and or pre-existing.=20

I don't want our creativity to be limited by arbitrary government fiat,
nor by lack of imagination constrained to repeat simply what already
exists.


> The owners of long haul fiber networks=20
> want to use them for signaled media.=20

Nothing precludes this.  This is a layer 3 issue.  I am not saying that
to connect one signaling point to another that there can not be a VPN
over a third IP carrier.  Don't confuse Layer 5 with Layer 3 issues.


> My guess is we will not=20
> only have One Tier 1 to choose from, but the following:
>=20
> 1.) Each Enterprise will desire at least 2 interconnect=20
> carriers to ensure they have the lowest cost and best level=20
> of service with redundancy.

My statements don't preclude this.

=20
> 2.) Each Tier 1 Carrier will want to purchase termination=20
> services for its customers and have multiple paths to each=20
> major desination domain

Not precluded.

=20
> 3.) Many Tier 2 and Tier 3 carriers will exist by selecting=20
> specific target markets and verticals.

Define Tier 2 and 3.  I think you have descended into Layer 3 here.

=20
> 4.) Governments will continue to try to enforce regulations=20
> guaranteeing a competitive landscape for communication=20
> interconnects (requiring access for 2nd and 3rd carrier options.

See previous.

=20
> I could go on and on. I would suggest that we embrace the=20
> realities of our world as requirements for this standards group.

Belive me, I am being very realistic.  Allow room for arbitrage and it
will occur.  Economics will drive out an inefficiency in routing, so
long as a better path exists.

If you do go on and on, would you please also indicate what added value
any such intermediaries provide?  So far, you have only indicated that
they have existed in the past.  While their value may have been grounded
in the regulatory and business climate of the past, past history that
does not take into account new technical realities amounts to just
rebuilding the PSTN because that's the way we did it with TDM.  I don't
find such arguments persuasive.

Mike

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Date: Tue, 13 Jun 2006 23:09:05 +0200
From: Otmar Lendl <lendl@nic.at>
To: "Michael Hammer (mhammer)" <mhammer@cisco.com>
Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Message-ID: <20060613210905.GB9035@nic.at>
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[inline]

On 2006/06/13 16:06, "Michael Hammer (mhammer)" <mhammer@cisco.com> wrote:
> > From: Patrick Melampy [mailto:PMelampy@acmepacket.com] 
> > 
> > Unless somebody representing a carrier makes this statement, 
> > it's safe to say we are just guessing. I find this statement 
> > your making to lack any backing. How many Autonomous Systems 
> > do your communications go through today? When you're at work? 
> > When you're at home? I'm guessing you have no idea. Why 
> > should this group ever designate a role for a Tier 1 SP as a 
> > required internetworking element? If you worked at a 
> > competitive Tier 2 you might take exception to your statement.
> 
> The basis for this statement is an assumption that for security and
> economic reasons, you must have rules of engagement in order to connect
> to a peer.  If there is a business arrangement, you can directly
> connect, if not then an intermediary is used.  This is rather simple and
> straight forward.  This does not preclude you creating a
> wide-open-to-the-Internet federation through which anyone can connect to
> you.
> 
> I am simply stating there are two options:  direct and indirect.  Is
> there a third possibility?

Well, call it a variant of "direct": The business arrangement may
be via an intermediate, but the connection may be direct.
(e.g. when a some sort of certification authority qualifies peers
on behalf of some constituency.)

[...]

> It is optimal because we are talking about two parties, the caller and
> called parties.  For that you could have an origination system and a
> termination system.  Anything beyond that is extraneous middlemen that
> add cost into the picture, but not much in the way of added value, and a
> lot of possibility for unintended consequences to screw up the signaling
> or media.

[...]

> > I could go on and on. I would suggest that we embrace the 
> > realities of our world as requirements for this standards group.
> 
> Belive me, I am being very realistic.  Allow room for arbitrage and it
> will occur.  Economics will drive out an inefficiency in routing, so
> long as a better path exists.

Agreed. How fast and widestread this happens depends a lot on the
cost structure of layer 5 transits. If it is pure SIP signalling
(e.g. in order to provide trust) without media, then this can
be done with very little additional cost per call (and none per minute).

Whether some local SP from Hawaii will bother to establish peerings
with one from Finland will depend on calling patterns and the
administrative overhead of establishing the peering.

We just don't know yet what the cost factors will be in this business
in a few years. I just doubt that we can accurately predict what
business models will emerge and how many SIP proxies the average
call will traverse.

All we can do right now is to write standards which are not dependent
on a specific assumption regarding the interconnection structure. 
Our RFCs need to cope with both a quite flat hierarchy (large number
of peers per service provider) as well as 4-level one.

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
Date: Tue, 13 Jun 2006 21:48:51 -0400
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Michael Hammer (mhammer)" <mhammer@cisco.com>,
	"Patrick Melampy" <PMelampy@acmepacket.com>,
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Piping in here to make a point or two - Two realistic requirements that
oftentimes are overlooked in global peering bar-napkin discussions, and
possibly assumed in the PSTN world, are trust and scale.  I argue that
the P2P world of direct interconnection/peering, though preferred for
obvious technical reasons, can't be realized practically due to scale
limitations and an inherent lack of trust. Therefore these 2 items
should be kept in mind when scheming up a solution for global peering
dominance. (/me puts pinky finger to mouth and mutters, "1 million
dollars"!  muhahaha)  ;) =20

A service provide would be best served to provide its subscribers a way
to contact anyone else.  The burden to facilitate that session is on the
SP.  Will a SP in Topeka trust a SP in Oslo enough to establish a direct
peering arrangement?  Without that direct peer, how will the subscribers
of the Topeka SP access the folks in Oslo?  An intermediary most likely
will be involved with the value being reachability and trust that should
a call be sent to the transit provider, it will be completed as intended
by the calling party. At the end of the day, it will be the one that
will complete the call that gets the buck.  Unfortunately it's the PSTN
that facilitates that Topeka to Oslo call today more often than not.

-Adam

-----Original Message-----
From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]=20
Sent: Tuesday, June 13, 2006 10:38 AM
To: Patrick Melampy; Stastny Richard; Reinaldo Penno; Otmar Lendl
Cc: speermint@ietf.org
Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture

Patrick,

Let us take these one at a time.  Inline.=20

> -----Original Message-----
> From: Patrick Melampy [mailto:PMelampy@acmepacket.com]
> Sent: Monday, June 12, 2006 9:34 PM
> To: Michael Hammer (mhammer); 'Stastny Richard'; 'Reinaldo Penno';=20
> 'Otmar Lendl'
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> I'd like to comment as follows:
>=20
> You said:
> >  I believe the reality will be that there will ultimately
> be two options:
> >   - from one SP (carrier/enterprise) to another there is
> direct peering,
> >   - from one SP to another there is no direct peering and a
> "Tier 1"=20
> > SP  will be used for transit.
>=20
> Unless somebody representing a carrier makes this statement, it's safe

> to say we are just guessing. I find this statement your making to lack

> any backing. How many Autonomous Systems do your communications go=20
> through today? When you're at work?
> When you're at home? I'm guessing you have no idea. Why should this=20
> group ever designate a role for a Tier 1 SP as a required=20
> internetworking element? If you worked at a competitive Tier 2 you=20
> might take exception to your statement.

The basis for this statement is an assumption that for security and
economic reasons, you must have rules of engagement in order to connect
to a peer.  If there is a business arrangement, you can directly
connect, if not then an intermediary is used.  This is rather simple and
straight forward.  This does not preclude you creating a
wide-open-to-the-Internet federation through which anyone can connect to
you.

I am simply stating there are two options:  direct and indirect.  Is
there a third possibility?

=20
> > My thinking, if it is not obvious is that the optimal
> number of SPs is
> > 2 (direct peering).
>=20
> Optimal in whose frame of reference? The internet's? The caller? The=20
> callee?
> The carrier(s)? My guess is your thinking of the internet.=20

The most optimal is directly from client to client, ie. true P2P.  But,
I am relaxing that here to the assumption that an end device may
"outsource" some degree of features and functionality to a SP, be that
enterprise PBX or carrier system.

It is optimal because we are talking about two parties, the caller and
called parties.  For that you could have an origination system and a
termination system.  Anything beyond that is extraneous middlemen that
add cost into the picture, but not much in the way of added value, and a
lot of possibility for unintended consequences to screw up the signaling
or media.

Case in point:  Every additional signaling point on the path will
exacerbate call setup delay and increase the complexity of the
interactions amongst all these components.

Case in point:  Every additional media anchor will add delay to the RTP
stream, thus negatively affecting QoS.

So, from the perspective of the caller and callee, the service
experience will be better if SPs can minimize the number of extra moving
parts in the system.

And, from the perspective of the SP, the fewer middlemen in the picture,
the fewer the profits to share amongst those middlemen.

I am just applying Occam's Razor to get the minimalist set.

=20
> > Second best is 3 (one transit). =20
> Why is one transit carrier better than 2 or 3? Is it cost?=20
> Quality? Where did this conclusion come from? Are there any IP based=20
> models where a single backbone carrier is considered optimal? Doesn't=20
> this limit competition and choice? Wouldn't it be anti-competitive,=20
> and thus not be net-neutral?

I am suggesting that if both caller and callee have the good fortune to
prefer the same transit carrier, then the savings from avoiding a second
SP can be shared.  This is what is experienced when a call stays within
a single enterprise.  This is what happens when you have a "friends and
family" scenario.

It is both a business advantage and a technical advantage to have only
one SP's policies to care about.  No interworking!

=20
> >Maximum number is 4 (one originating and one terminating transit).
>=20
> Why is this the maxium?=20

What would a third tranist carrier provide that could not be better
provided if transit carrier 1 and 2 had a direct signaling agreement
between them?  How is sharing revenues amongst 3 carriers better than
shared betweeen 2?


> I hate to argue with your statements, but they should be better=20
> defended.

Hopefully, I have done so. :)


> There are a lot of managed networks that are both legislated and or=20
> pre-existing.

I don't want our creativity to be limited by arbitrary government fiat,
nor by lack of imagination constrained to repeat simply what already
exists.


> The owners of long haul fiber networks want to use them for signaled=20
> media.

Nothing precludes this.  This is a layer 3 issue.  I am not saying that
to connect one signaling point to another that there can not be a VPN
over a third IP carrier.  Don't confuse Layer 5 with Layer 3 issues.


> My guess is we will not
> only have One Tier 1 to choose from, but the following:
>=20
> 1.) Each Enterprise will desire at least 2 interconnect carriers to=20
> ensure they have the lowest cost and best level of service with=20
> redundancy.

My statements don't preclude this.

=20
> 2.) Each Tier 1 Carrier will want to purchase termination services for

> its customers and have multiple paths to each major desination domain

Not precluded.

=20
> 3.) Many Tier 2 and Tier 3 carriers will exist by selecting specific=20
> target markets and verticals.

Define Tier 2 and 3.  I think you have descended into Layer 3 here.

=20
> 4.) Governments will continue to try to enforce regulations=20
> guaranteeing a competitive landscape for communication interconnects=20
> (requiring access for 2nd and 3rd carrier options.

See previous.

=20
> I could go on and on. I would suggest that we embrace the realities of

> our world as requirements for this standards group.

Belive me, I am being very realistic.  Allow room for arbitrage and it
will occur.  Economics will drive out an inefficiency in routing, so
long as a better path exists.

If you do go on and on, would you please also indicate what added value
any such intermediaries provide?  So far, you have only indicated that
they have existed in the past.  While their value may have been grounded
in the regulatory and business climate of the past, past history that
does not take into account new technical realities amounts to just
rebuilding the PSTN because that's the way we did it with TDM.  I don't
find such arguments persuasive.

Mike

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Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
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I couldn't agree more.  Comments inline:

On Jun 13, 2006, at 9:48 PM, Uzelac, Adam wrote:

> A service provide would be best served to provide its subscribers a  
> way
> to contact anyone else.  The burden to facilitate that session is  
> on the
> SP.  Will a SP in Topeka trust a SP in Oslo enough to establish a  
> direct
> peering arrangement?  Without that direct peer, how will the  
> subscribers
> of the Topeka SP access the folks in Oslo?

One model that has been discussed on this list is the email model,  
which is that ITSPs just run open networks.  Given that I receive on  
the order of 900 spam emails a day, I do not think ITSPs will be  
opening up to this arrangement any time soon.

>   An intermediary most likely
> will be involved with the value being reachability and trust that  
> should
> a call be sent to the transit provider, it will be completed as  
> intended
> by the calling party.

There are entities that do this today, except they don't do transit  
(usually).  They are known as VoIP peering exchanges.

> At the end of the day, it will be the one that
> will complete the call that gets the buck.

And most don't have call termination charges.

-andy


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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; -kht=
ml-line-break: after-white-space; "><DIV>I couldn't agree more.=A0 Commen=
ts inline:</DIV><BR><DIV><DIV>On Jun 13, 2006, at 9:48 PM, Uzelac, Adam w=
rote:</DIV><BR class=3D"Apple-interchange-newline"><BLOCKQUOTE type=3D"ci=
te"><P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica"=
 size=3D"3" style=3D"font: 12.0px Helvetica">A service provide would be b=
est served to provide its subscribers a way</FONT></P> <P style=3D"margin=
: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3" style=3D"f=
ont: 12.0px Helvetica">to contact anyone else.<SPAN class=3D"Apple-conver=
ted-space">=A0 </SPAN>The burden to facilitate that session is on the</FO=
NT></P> <P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvet=
ica" size=3D"3" style=3D"font: 12.0px Helvetica">SP.<SPAN class=3D"Apple-=
converted-space">=A0 </SPAN>Will a SP in Topeka trust a SP in Oslo enough=
 to establish a direct</FONT></P> <P style=3D"margin: 0.0px 0.0px 0.0px 0=
.0px"><FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica=
">peering arrangement?<SPAN class=3D"Apple-converted-space">=A0 </SPAN>Wi=
thout that direct peer, how will the subscribers</FONT></P> <P style=3D"m=
argin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3" style=
=3D"font: 12.0px Helvetica">of the Topeka SP access the folks in Oslo?</F=
ONT></P></BLOCKQUOTE><DIV><BR class=3D"khtml-block-placeholder"></DIV><DI=
V>One model that has been discussed on this list is the email model, whic=
h is that ITSPs just run open networks.=A0 Given that I receive on the or=
der of 900 spam emails a day, I do not think ITSPs will be opening up to =
this arrangement any time soon.</DIV><BR><BLOCKQUOTE type=3D"cite"><P sty=
le=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3=
" style=3D"font: 12.0px Helvetica"><SPAN class=3D"Apple-converted-space">=
=A0 </SPAN>An intermediary most likely</FONT></P> <P style=3D"margin: 0.0=
px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3" style=3D"font: =
12.0px Helvetica">will be involved with the value being reachability and =
trust that should</FONT></P> <P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"=
><FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica">a c=
all be sent to the transit provider, it will be completed as intended</FO=
NT></P> <P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT face=3D"Helvet=
ica" size=3D"3" style=3D"font: 12.0px Helvetica">by the calling party.<BR=
></FONT></P></BLOCKQUOTE><DIV><BR class=3D"khtml-block-placeholder"></DIV=
><DIV>There are entities that do this today, except they don't do transit=
 (usually).=A0 They are known as VoIP peering exchanges.</DIV><BR><BLOCKQ=
UOTE type=3D"cite"><P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><FONT fac=
e=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"> At the end o=
f the day, it will be the one that</FONT></P> <P style=3D"margin: 0.0px 0=
.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0=
px Helvetica">will complete the call that gets the buck.<SPAN class=3D"Ap=
ple-converted-space">=A0</SPAN></FONT></P></BLOCKQUOTE><BR></DIV><DIV>And=
 most don't have call termination charges.</DIV><DIV><BR class=3D"khtml-b=
lock-placeholder"></DIV><DIV>-andy</DIV><BR></BODY></HTML>
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> Whether some local SP from Hawaii will bother to establish=20
> peerings with one from Finland will depend on calling=20
> patterns and the administrative overhead of establishing the peering.

I'd certainly like to see that we get that admin overhead as low /
minimal as possible.  To use a buzzword, it should be as 'frictionless'
as possible.  ;-)  Which is basically, easy, efficient, quick, etc.

Jason

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Very interesting response.  I think you have highlighted again why the
co-chairs feel that security and trust are major areas of concern and
work for the group.

Jason=20

> -----Original Message-----
> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]=20
> Sent: Tuesday, June 13, 2006 9:49 PM
> To: Michael Hammer (mhammer); Patrick Melampy; Stastny=20
> Richard; Reinaldo Penno; Otmar Lendl
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> Piping in here to make a point or two - Two realistic=20
> requirements that oftentimes are overlooked in global peering=20
> bar-napkin discussions, and possibly assumed in the PSTN=20
> world, are trust and scale.  I argue that the P2P world of=20
> direct interconnection/peering, though preferred for obvious=20
> technical reasons, can't be realized practically due to scale=20
> limitations and an inherent lack of trust. Therefore these 2=20
> items should be kept in mind when scheming up a solution for=20
> global peering dominance. (/me puts pinky finger to mouth and=20
> mutters, "1 million dollars"!  muhahaha)  ;) =20
>=20
> A service provide would be best served to provide its=20
> subscribers a way to contact anyone else.  The burden to=20
> facilitate that session is on the SP.  Will a SP in Topeka=20
> trust a SP in Oslo enough to establish a direct peering=20
> arrangement?  Without that direct peer, how will the=20
> subscribers of the Topeka SP access the folks in Oslo?  An=20
> intermediary most likely will be involved with the value=20
> being reachability and trust that should a call be sent to=20
> the transit provider, it will be completed as intended by the=20
> calling party. At the end of the day, it will be the one that=20
> will complete the call that gets the buck.  Unfortunately=20
> it's the PSTN that facilitates that Topeka to Oslo call today=20
> more often than not.
>=20
> -Adam
>=20
> -----Original Message-----
> From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> Sent: Tuesday, June 13, 2006 10:38 AM
> To: Patrick Melampy; Stastny Richard; Reinaldo Penno; Otmar Lendl
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering Architecture
>=20
> Patrick,
>=20
> Let us take these one at a time.  Inline.=20
>=20
> > -----Original Message-----
> > From: Patrick Melampy [mailto:PMelampy@acmepacket.com]
> > Sent: Monday, June 12, 2006 9:34 PM
> > To: Michael Hammer (mhammer); 'Stastny Richard'; 'Reinaldo Penno';=20
> > 'Otmar Lendl'
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] Updated Draft: SPEERMINT Peering=20
> Architecture
> >=20
> > I'd like to comment as follows:
> >=20
> > You said:
> > >  I believe the reality will be that there will ultimately
> > be two options:
> > >   - from one SP (carrier/enterprise) to another there is
> > direct peering,
> > >   - from one SP to another there is no direct peering and a
> > "Tier 1"=20
> > > SP  will be used for transit.
> >=20
> > Unless somebody representing a carrier makes this=20
> statement, it's safe
>=20
> > to say we are just guessing. I find this statement your=20
> making to lack
>=20
> > any backing. How many Autonomous Systems do your communications go=20
> > through today? When you're at work?
> > When you're at home? I'm guessing you have no idea. Why should this=20
> > group ever designate a role for a Tier 1 SP as a required=20
> > internetworking element? If you worked at a competitive Tier 2 you=20
> > might take exception to your statement.
>=20
> The basis for this statement is an assumption that for=20
> security and economic reasons, you must have rules of=20
> engagement in order to connect to a peer.  If there is a=20
> business arrangement, you can directly connect, if not then=20
> an intermediary is used.  This is rather simple and straight=20
> forward.  This does not preclude you creating a=20
> wide-open-to-the-Internet federation through which anyone can=20
> connect to you.
>=20
> I am simply stating there are two options:  direct and=20
> indirect.  Is there a third possibility?
>=20
> =20
> > > My thinking, if it is not obvious is that the optimal
> > number of SPs is
> > > 2 (direct peering).
> >=20
> > Optimal in whose frame of reference? The internet's? The=20
> caller? The=20
> > callee?
> > The carrier(s)? My guess is your thinking of the internet.=20
>=20
> The most optimal is directly from client to client, ie. true=20
> P2P.  But, I am relaxing that here to the assumption that an=20
> end device may "outsource" some degree of features and=20
> functionality to a SP, be that enterprise PBX or carrier system.
>=20
> It is optimal because we are talking about two parties, the=20
> caller and called parties.  For that you could have an=20
> origination system and a termination system.  Anything beyond=20
> that is extraneous middlemen that add cost into the picture,=20
> but not much in the way of added value, and a lot of=20
> possibility for unintended consequences to screw up the=20
> signaling or media.
>=20
> Case in point:  Every additional signaling point on the path=20
> will exacerbate call setup delay and increase the complexity=20
> of the interactions amongst all these components.
>=20
> Case in point:  Every additional media anchor will add delay=20
> to the RTP stream, thus negatively affecting QoS.
>=20
> So, from the perspective of the caller and callee, the=20
> service experience will be better if SPs can minimize the=20
> number of extra moving parts in the system.
>=20
> And, from the perspective of the SP, the fewer middlemen in=20
> the picture, the fewer the profits to share amongst those middlemen.
>=20
> I am just applying Occam's Razor to get the minimalist set.
>=20
> =20
> > > Second best is 3 (one transit). =20
> > Why is one transit carrier better than 2 or 3? Is it cost?=20
> > Quality? Where did this conclusion come from? Are there any=20
> IP based=20
> > models where a single backbone carrier is considered=20
> optimal? Doesn't=20
> > this limit competition and choice? Wouldn't it be anti-competitive,=20
> > and thus not be net-neutral?
>=20
> I am suggesting that if both caller and callee have the good=20
> fortune to prefer the same transit carrier, then the savings=20
> from avoiding a second SP can be shared.  This is what is=20
> experienced when a call stays within a single enterprise. =20
> This is what happens when you have a "friends and family" scenario.
>=20
> It is both a business advantage and a technical advantage to=20
> have only one SP's policies to care about.  No interworking!
>=20
> =20
> > >Maximum number is 4 (one originating and one terminating transit).
> >=20
> > Why is this the maxium?=20
>=20
> What would a third tranist carrier provide that could not be=20
> better provided if transit carrier 1 and 2 had a direct=20
> signaling agreement between them?  How is sharing revenues=20
> amongst 3 carriers better than shared betweeen 2?
>=20
>=20
> > I hate to argue with your statements, but they should be better=20
> > defended.
>=20
> Hopefully, I have done so. :)
>=20
>=20
> > There are a lot of managed networks that are both legislated and or=20
> > pre-existing.
>=20
> I don't want our creativity to be limited by arbitrary=20
> government fiat, nor by lack of imagination constrained to=20
> repeat simply what already exists.
>=20
>=20
> > The owners of long haul fiber networks want to use them for=20
> signaled=20
> > media.
>=20
> Nothing precludes this.  This is a layer 3 issue.  I am not=20
> saying that to connect one signaling point to another that=20
> there can not be a VPN over a third IP carrier.  Don't=20
> confuse Layer 5 with Layer 3 issues.
>=20
>=20
> > My guess is we will not
> > only have One Tier 1 to choose from, but the following:
> >=20
> > 1.) Each Enterprise will desire at least 2 interconnect carriers to=20
> > ensure they have the lowest cost and best level of service with=20
> > redundancy.
>=20
> My statements don't preclude this.
>=20
> =20
> > 2.) Each Tier 1 Carrier will want to purchase termination=20
> services for
>=20
> > its customers and have multiple paths to each major=20
> desination domain
>=20
> Not precluded.
>=20
> =20
> > 3.) Many Tier 2 and Tier 3 carriers will exist by selecting=20
> specific=20
> > target markets and verticals.
>=20
> Define Tier 2 and 3.  I think you have descended into Layer 3 here.
>=20
> =20
> > 4.) Governments will continue to try to enforce regulations=20
> > guaranteeing a competitive landscape for communication=20
> interconnects=20
> > (requiring access for 2nd and 3rd carrier options.
>=20
> See previous.
>=20
> =20
> > I could go on and on. I would suggest that we embrace the=20
> realities of
>=20
> > our world as requirements for this standards group.
>=20
> Belive me, I am being very realistic.  Allow room for=20
> arbitrage and it will occur.  Economics will drive out an=20
> inefficiency in routing, so long as a better path exists.
>=20
> If you do go on and on, would you please also indicate what=20
> added value any such intermediaries provide?  So far, you=20
> have only indicated that they have existed in the past. =20
> While their value may have been grounded in the regulatory=20
> and business climate of the past, past history that does not=20
> take into account new technical realities amounts to just=20
> rebuilding the PSTN because that's the way we did it with=20
> TDM.  I don't find such arguments persuasive.
>=20
> Mike
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

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From speermint-bounces@ietf.org Thu Jun 15 18:34:10 2006
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Subject: [speermint] minimum set of requirements for SIP-based VoIP
	interconnection and emergency calling
Date: Thu, 15 Jun 2006 16:34:01 -0600
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Should the special case of emergency calls be considered as part of the
use cases for the 'minimum' set of requirements for SIP-based VoIP
interconnection?

If so, what should the minimum set of requirements be for SIP voice
interconnects?

Unless this is done somewhere else in the context of voice
interconnects, I think it should be in scope given the definition of VSP
in the speermint terminology draft and the real use cases of VSPs
handing off 911 calls to other service provider or terminating calls to
911 psaps.

That said, those requirements should probably be left at the SIP level
and remain at a high-level given the work in sipping, ecrit and various
national bodies.=20
E.g.
	- a VSP should be able to recognize the SOS URN or any other
well known markers such as a local dialing string for emergency calls as
an emergency session
	- a VSP should invoke special SIP routing procedures to hand the
call off to the correct PSAP
	- put SIP Priority header
	- support SIP location conveyance I-D?
	- etc.

Any thoughts and comments are appreciated.

Jean-Francois.
email: jfm@cablelabs.com=20



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Subject: RE: [speermint] minimum set of requirements for SIP-based
	VoIPinterconnection and emergency calling
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=20
Hi Jean-Francois,

I agree with your general position on this and the first three of your
five bullet-list items.

(I don't know enough about the location conveyance I-D to have a
position and I'll wait on the etc item :)

We should probably do a close scrub of the latest ECRIT requirements doc
to ensure coverage and adopt appropriate terminology where applicable.

Cheers,
BobN

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
Sent: Thursday, June 15, 2006 6:34 PM
To: speermint@ietf.org
Subject: [speermint] minimum set of requirements for SIP-based
VoIPinterconnection and emergency calling


Should the special case of emergency calls be considered as part of the
use cases for the 'minimum' set of requirements for SIP-based VoIP
interconnection?

If so, what should the minimum set of requirements be for SIP voice
interconnects?

Unless this is done somewhere else in the context of voice
interconnects, I think it should be in scope given the definition of VSP
in the speermint terminology draft and the real use cases of VSPs
handing off 911 calls to other service provider or terminating calls to
911 psaps.

That said, those requirements should probably be left at the SIP level
and remain at a high-level given the work in sipping, ecrit and various
national bodies.=20
E.g.
	- a VSP should be able to recognize the SOS URN or any other
well known markers such as a local dialing string for emergency calls as
an emergency session
	- a VSP should invoke special SIP routing procedures to hand the
call off to the correct PSAP
	- put SIP Priority header
	- support SIP location conveyance I-D?
	- etc.

Any thoughts and comments are appreciated.

Jean-Francois.
email: jfm@cablelabs.com=20



_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint

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From speermint-bounces@ietf.org Fri Jun 16 04:06:58 2006
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Subject: Re: [speermint] minimum set of requirements for
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Sorry, but I doubt that emergency services is within the scope of =
speermint
=20
For emergency services normally either the UA itself or the first hop =
proxy is responsible,
so it never reaches the peering point at all
=20
-sta

________________________________

Von: Bob Natale [mailto:bnatale@nextone.com]
Gesendet: Fr 16.06.2006 00:41
An: Jean-Francois Mule
Cc: speermint@ietf.org
Betreff: RE: [speermint] minimum set of requirements for =
SIP-basedVoIPinterconnection and emergency calling




Hi Jean-Francois,

I agree with your general position on this and the first three of your
five bullet-list items.

(I don't know enough about the location conveyance I-D to have a
position and I'll wait on the etc item :)

We should probably do a close scrub of the latest ECRIT requirements doc
to ensure coverage and adopt appropriate terminology where applicable.

Cheers,
BobN

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
Sent: Thursday, June 15, 2006 6:34 PM
To: speermint@ietf.org
Subject: [speermint] minimum set of requirements for SIP-based
VoIPinterconnection and emergency calling


Should the special case of emergency calls be considered as part of the
use cases for the 'minimum' set of requirements for SIP-based VoIP
interconnection?

If so, what should the minimum set of requirements be for SIP voice
interconnects?

Unless this is done somewhere else in the context of voice
interconnects, I think it should be in scope given the definition of VSP
in the speermint terminology draft and the real use cases of VSPs
handing off 911 calls to other service provider or terminating calls to
911 psaps.

That said, those requirements should probably be left at the SIP level
and remain at a high-level given the work in sipping, ecrit and various
national bodies.
E.g.
        - a VSP should be able to recognize the SOS URN or any other
well known markers such as a local dialing string for emergency calls as
an emergency session
        - a VSP should invoke special SIP routing procedures to hand the
call off to the correct PSAP
        - put SIP Priority header
        - support SIP location conveyance I-D?
        - etc.

Any thoughts and comments are appreciated.

Jean-Francois.
email: jfm@cablelabs.com



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https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: [speermint] minimum set of requirements for
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Date: Fri, 16 Jun 2006 03:04:27 -0600
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To: "Stastny Richard" <Richard.Stastny@oefeg.at>
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Stastny Richard wrote:
> Sorry, but I doubt that emergency services is within the scope of
> speermint
>
> For emergency services normally either the UA itself or the first hop
> proxy is responsible,
> so it never reaches the peering point at all
        ^^^^^
There are examples of VSP providers that do offer sip-based VoIP =
interconnect services today including 9-1-1. So your assertion that =
emergency calls do not reach "peering points" between VSPs is simply not =
true (based on current practices and this is by no means =
cable-specific).

> For emergency services normally either the UA itself or the first hop
> proxy is responsible
I presume you mean responsible for "invoking the special routing =
procedures caused by an emergency dialed string or use of SOS URN".
This is ok but it does not necessarily mean that, upon determination =
that the call is an emergency call, the originating VSP does have to =
handle the SIP call to the PSAP within its own domain.=20
Some VSPs do have gateways connected to the PSTN and the whole selective =
routers thingy in the North American system for example to do this; some =
don't and hand off the call to another VSP. The question of who does the =
ECRIT mapping protocol resolution is also not relevant here.

My first question was:
   - should there be any minimal set of requirements (and ones =
applicable beyond national standards) for handling emergency calls that =
should be captured in a speermint requirement doc for SIP-based VoIP =
interconnect?

Jean-Fran=E7ois


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From speermint-bounces@ietf.org Fri Jun 16 05:21:42 2006
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Subject: Re: [speermint] minimum set of requirements for
	SIP-basedVoIPinterconnection and emergency calling
Date: Fri, 16 Jun 2006 11:22:55 +0200
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>Some VSPs do have gateways connected to the PSTN and the whole =
selective routers thingy in the North American system for >example to do =
this; some don't and hand off the call to another VSP. The question of =
who does the ECRIT mapping protocol >resolution is also not relevant =
here.

Ok, but you mention here two issues out of speermint scope: gateways to =
the PSTN and=20
North American selective router stuff
=20
-sta

________________________________

Von: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
Gesendet: Fr 16.06.2006 11:04
An: Stastny Richard
Cc: speermint@ietf.org
Betreff: RE: [speermint] minimum set of requirements for =
SIP-basedVoIPinterconnection and emergency calling



Stastny Richard wrote:
> Sorry, but I doubt that emergency services is within the scope of
> speermint
>
> For emergency services normally either the UA itself or the first hop
> proxy is responsible,
> so it never reaches the peering point at all
        ^^^^^
There are examples of VSP providers that do offer sip-based VoIP =
interconnect services today including 9-1-1. So your assertion that =
emergency calls do not reach "peering points" between VSPs is simply not =
true (based on current practices and this is by no means =
cable-specific).

> For emergency services normally either the UA itself or the first hop
> proxy is responsible
I presume you mean responsible for "invoking the special routing =
procedures caused by an emergency dialed string or use of SOS URN".
This is ok but it does not necessarily mean that, upon determination =
that the call is an emergency call, the originating VSP does have to =
handle the SIP call to the PSAP within its own domain.
Some VSPs do have gateways connected to the PSTN and the whole selective =
routers thingy in the North American system for example to do this; some =
don't and hand off the call to another VSP. The question of who does the =
ECRIT mapping protocol resolution is also not relevant here.

My first question was:
   - should there be any minimal set of requirements (and ones =
applicable beyond national standards) for handling emergency calls that =
should be captured in a speermint requirement doc for SIP-based VoIP =
interconnect?

Jean-Fran=E7ois




_______________________________________________
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Speermint@ietf.org
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From speermint-bounces@ietf.org Fri Jun 16 07:39:38 2006
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Stastny Richard wrote:
> >Some VSPs do have gateways connected to the PSTN and the whole
> selective routers thingy in the North American system for >example to
> do this; some don't and hand off the call to another VSP. The question
> of who does the ECRIT mapping protocol >resolution is also not
> relevant here.
>=20
> Ok, but you mention here two issues out of speermint scope: gateways
> to the PSTN and
> North American selective router stuff

And that is clearly out of scope.
Per my earlier email, the scope question was related to the other case
where emergency calls are not handled within a single VSP domain (or for
that matter, the case of an enterprise using a SIP interconnect service
and handing off all calls to the VSP).=20

Jean-Francois.


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Jean-Francois,

Is this to mean that you believe emergency calling should be addressed
within the scope of general peering arrangements, or you would like to
see emergency requirements addressed in a specific emergency-like
peering arrangement (possibly other services as well)?  I think I could
see that and one would not exclude the other.

-AU

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
Sent: Friday, June 16, 2006 7:39 AM
To: Stastny Richard
Cc: speermint@ietf.org
Subject: RE: [speermint] minimum set of requirements
forSIP-basedVoIPinterconnection and emergency calling

Stastny Richard wrote:
> >Some VSPs do have gateways connected to the PSTN and the whole
> selective routers thingy in the North American system for >example to=20
> do this; some don't and hand off the call to another VSP. The question

> of who does the ECRIT mapping protocol >resolution is also not=20
> relevant here.
>=20
> Ok, but you mention here two issues out of speermint scope: gateways=20
> to the PSTN and North American selective router stuff

And that is clearly out of scope.
Per my earlier email, the scope question was related to the other case
where emergency calls are not handled within a single VSP domain (or for
that matter, the case of an enterprise using a SIP interconnect service
and handing off all calls to the VSP).=20

Jean-Francois.


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I think we'd be better served NOT to take up the emergency use case for
now, until the other groups (GEOPRIV and ECRIT) solidify their
requirements.=20

Penn

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
Sent: Friday, June 16, 2006 7:39 AM
To: Stastny Richard
Cc: speermint@ietf.org
Subject: RE: [speermint] minimum set of requirements
forSIP-basedVoIPinterconnection and emergency calling

Stastny Richard wrote:
> >Some VSPs do have gateways connected to the PSTN and the whole
> selective routers thingy in the North American system for >example to
> do this; some don't and hand off the call to another VSP. The question
> of who does the ECRIT mapping protocol >resolution is also not
> relevant here.
>=20
> Ok, but you mention here two issues out of speermint scope: gateways
> to the PSTN and
> North American selective router stuff

And that is clearly out of scope.
Per my earlier email, the scope question was related to the other case
where emergency calls are not handled within a single VSP domain (or for
that matter, the case of an enterprise using a SIP interconnect service
and handing off all calls to the VSP).=20

Jean-Francois.


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The requirements work in ECRIT is finished. Here is the most recent 
document:
http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-10.txt

It might be good to clearly describe what you want to accomplish with 
emgerency services in the context of peering.

Ciao
Hannes


Pfautz, Penn L, GBLAM wrote:
> I think we'd be better served NOT to take up the emergency use case for
> now, until the other groups (GEOPRIV and ECRIT) solidify their
> requirements. 
> 
> Penn
> 
> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com] 
> Sent: Friday, June 16, 2006 7:39 AM
> To: Stastny Richard
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum set of requirements
> forSIP-basedVoIPinterconnection and emergency calling
> 
> Stastny Richard wrote:
> 
>>>Some VSPs do have gateways connected to the PSTN and the whole
>>
>>selective routers thingy in the North American system for >example to
>>do this; some don't and hand off the call to another VSP. The question
>>of who does the ECRIT mapping protocol >resolution is also not
>>relevant here.
>>
>>Ok, but you mention here two issues out of speermint scope: gateways
>>to the PSTN and
>>North American selective router stuff
> 
> 
> And that is clearly out of scope.
> Per my earlier email, the scope question was related to the other case
> where emergency calls are not handled within a single VSP domain (or for
> that matter, the case of an enterprise using a SIP interconnect service
> and handing off all calls to the VSP). 
> 
> Jean-Francois.
> 
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 
> 


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Adam,

   I'm generally interested in what folks believe should be in-scope and
out of scope for the wg deliverable identified as "the minimum set of
requirements for SIP-based VoIP interconnection (BCP)", and especially
re: emergency calling.

   If the goal is a BCP and if today, various providers do have SIP
interconnect agreements that include emergency calling, I thought it may
be in scope, at least from a SIP requirement point of view per my email
yesterday.

Jean-Francois

> -----Original Message-----
> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]
> Sent: Friday, June 16, 2006 5:50 AM
> To: Jean-Francois Mule; Stastny Richard
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum set of requirements forSIP-
> basedVoIPinterconnection and emergency calling
>=20
> Jean-Francois,
>=20
> Is this to mean that you believe emergency calling should be addressed
> within the scope of general peering arrangements, or you would like to
> see emergency requirements addressed in a specific emergency-like
> peering arrangement (possibly other services as well)?  I think I
> could
> see that and one would not exclude the other.
>=20
> -AU
>=20
> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Friday, June 16, 2006 7:39 AM
> To: Stastny Richard
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum set of requirements
> forSIP-basedVoIPinterconnection and emergency calling
>=20
> Stastny Richard wrote:
> > >Some VSPs do have gateways connected to the PSTN and the whole
> > selective routers thingy in the North American system for >example
> to
> > do this; some don't and hand off the call to another VSP. The
> question
>=20
> > of who does the ECRIT mapping protocol >resolution is also not
> > relevant here.
> >
> > Ok, but you mention here two issues out of speermint scope: gateways
> > to the PSTN and North American selective router stuff
>=20
> And that is clearly out of scope.
> Per my earlier email, the scope question was related to the other case
> where emergency calls are not handled within a single VSP domain (or
> for
> that matter, the case of an enterprise using a SIP interconnect
> service
> and handing off all calls to the VSP).
>=20
> Jean-Francois.
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint



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Stastny Richard wrote:
> Sorry, but I doubt that emergency services is within the scope of speermint

I can't speak to this statement because I'm completely confused about 
the purpose of the working being conducted in this working group.

> For emergency services normally either the UA itself or the first hop proxy is responsible,
> so it never reaches the peering point at all

But can you explain to me how you reached this conclusion?  Do you 
expect UAs to directly connect to PSAP SIP servers?  Are you ruling out 
a peering point because the ECRIT process results in a SIP URI and not a 
telephone number, and therefore a peering point won't be consulted?  I 
don't get it.

-andy


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	requirements	forSIP-basedVoIPinterconnection and emergency calling
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Penn and Hannes,

   Thanks for the input. Here is an attempt to clarify what could be in
scope of speermint.=20

Hannes wrote:
> It might be good to clearly describe what you want to accomplish with
> emgerency services in the context of peering.

The potential goal would be to write the minimum requirements a VSP
should comply with when handing off an emergency call to another
administrative domain over IP through a VoIP interconnect point (SIP
call termination to another third-party provider which in turn does
terminate the call to a PSAP).

   Folks assume the work in GEOPRIV and ECRIT takes care of the above.
The requirement gap I can see so far is based on:
a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena i3).
The ECRIT requirement document states:
| This document only focuses on end-to-end IP-based calls, i.e., where
| the emergency call originates from an IP end system and terminates
| into an IP-capable PSAP, conveyed entirely over an IP network.

b) GEOPRIV, ECRIT and SIPPING are concerned with location determination,
emergency context resolution, location-based routing of the SIP call to
the PSAP URI, location conveyance in SIP.

c) SPEERMINT is about "architectures to identify, signal, and route
delay-sensitive (real-time) communication sessions."... "These sessions
use the SIP signaling protocol to enable peering between two or more
administrative domains over IP networks."

So, for example, the speermint requirement doc could state things like:
	- any applicable ECRIT requirements for exchanging emergency sip
calls between 2 or more domains,
	- any SIP requirements that should be taken into consideration.
For e.g. if available, convey location using sipping location
conveyance. But also special handling of things like p-asserted-id when
handing off emergency calls (if privacy is on in the originating domain,
some VSPs remove p-asserted-id when handing out the call to another
provider). In the case of an emergency call, for psap callback,
p-asserted-id must/should be left in... so that the terminating VSP that
connects to the PSAPs can identify the ANI.

These are first thoughts to provide more examples. We could just have a
couple of sentences highlighting the key requirements and giving
pointers to consider when the handoff of emergency calls is possible.

Penn wrote:
> > I think we'd be better served NOT to take up the emergency use case
> for
> > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > requirements.
This is fine and I'm ok not taking if we think we should just not take
this for *now* but as Hannes points out, and as I tried to highlight
above, we should not assume there will be a lot more requirement work in
other IETF groups around the SIP interconnects and there may be some
gaps (hence the scope question).

Jean-Francois

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> Sent: Friday, June 16, 2006 6:10 AM
> To: Pfautz, Penn L, GBLAM
> Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> Subject: Re: [speermint] minimum set of requirements forSIP-
> basedVoIPinterconnection and emergency calling
>=20
> The requirements work in ECRIT is finished. Here is the most recent
> document:
> http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> 10.txt
>=20
> It might be good to clearly describe what you want to accomplish with
> emgerency services in the context of peering.
>=20
> Ciao
> Hannes
>=20
>=20
> Pfautz, Penn L, GBLAM wrote:
> > I think we'd be better served NOT to take up the emergency use case
> for
> > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > requirements.
> >
> > Penn
> >
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Friday, June 16, 2006 7:39 AM
> > To: Stastny Richard
> > Cc: speermint@ietf.org
> > Subject: RE: [speermint] minimum set of requirements
> > forSIP-basedVoIPinterconnection and emergency calling
> >
> > Stastny Richard wrote:
> >
> >>>Some VSPs do have gateways connected to the PSTN and the whole
> >>
> >>selective routers thingy in the North American system for >example
> to
> >>do this; some don't and hand off the call to another VSP. The
> question
> >>of who does the ECRIT mapping protocol >resolution is also not
> >>relevant here.
> >>
> >>Ok, but you mention here two issues out of speermint scope: gateways
> >>to the PSTN and
> >>North American selective router stuff
> >
> >
> > And that is clearly out of scope.
> > Per my earlier email, the scope question was related to the other
> case
> > where emergency calls are not handled within a single VSP domain (or
> for
> > that matter, the case of an enterprise using a SIP interconnect
> service
> > and handing off all calls to the VSP).
> >
> > Jean-Francois.
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> >



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----- Forwarded message from ietf-secretariat@ietf.org -----

> From: ietf-secretariat@ietf.org
> To: ietf-announce@ietf.org
> Cc:=20
> Subject: Internet-Drafts Submission Cutoff Dates for the 66th IETF Meeting
>  in Montreal, Quebec, Canada=20
> Date: Fri, 16 Jun 2006 00:00:01 -0400
>=20
>=20
> There are two (2) Internet-Draft cutoff dates for the 66th=20
> IETF Meeting in Montreal, Quebec, Canada:
>=20
> June 19th: Cutoff Date for Initial (i.e., version -00)=20
> Internet-Draft Submissions=20
>=20
> All initial Internet-Drafts (version -00) must be submitted by Monday,=20
> June 19th at 9:00 AM ET. As always, all initial submissions with a=20
> filename beginning with "draft-ietf" must be approved by the=20
> appropriate WG Chair before they can be processed or announced.  The=20
> Secretariat would appreciate receiving WG Chair approval by Monday,=20
> June 12th at 9:00 AM ET.
>=20
> June 26th: Cutoff Date for Revised (i.e., version -01 and higher)=20
> Internet-Draft Submissions=20
>=20
> All revised Internet-Drafts (version -01 and higher) must be submitted=20
> by Monday, June 26th at 9:00 AM ET.
>=20
> Initial and revised Internet-Drafts received after their respective=20
> cutoff dates will not be made available in the Internet-Drafts=20
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> AM ET, when Internet-Draft posting resumes.  Please do not wait until=20
> the last minute to submit.
>=20
> Thank you for your understanding and cooperation. If you have any=20
> questions or concerns, then please send a message to=20
> internet-drafts@ietf.org.
>=20
> The IETF Secretariat
>=20
> FYI: The Internet-Draft cutoff dates as well as other significant dates
> for the 66th IETF Meeting can be found at http://www.ietf.org/meetings/cu=
toff_dates_66.html.
>=20
> _______________________________________________
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Subject: RE: [speermint] minimum set
	ofrequirements	forSIP-basedVoIPinterconnection and emergency calling
Date: Fri, 16 Jun 2006 15:44:45 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>,
	"Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>,
	"Pfautz, Penn L, GBLAM" <ppfautz@att.com>
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All,

Let me ask the question a different way:

Will all PSAPs belong to the same federation (a special PSAP federation
that defines the rules for accessing PSAP domains), or

will all PSAPs belong to various federations, where the rules for
handling SIP messages and what is needed for SOS calls is defined by
each federations?

Put me in the column for holding off on special cases of calls being
addressed until we have worked out the basic call case.  By then,
perhaps the ECRIT WG will have made further progress on the SOS call
delivery mechanisms and associated protocol elements.

Mike

=20

> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
> Sent: Friday, June 16, 2006 9:39 AM
> To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> Cc: Stastny Richard; speermint@ietf.org
> Subject: RE: [speermint] minimum set ofrequirements=20
> forSIP-basedVoIPinterconnection and emergency calling
>=20
> Penn and Hannes,
>=20
>    Thanks for the input. Here is an attempt to clarify what=20
> could be in scope of speermint.=20
>=20
> Hannes wrote:
> > It might be good to clearly describe what you want to=20
> accomplish with=20
> > emgerency services in the context of peering.
>=20
> The potential goal would be to write the minimum requirements=20
> a VSP should comply with when handing off an emergency call=20
> to another administrative domain over IP through a VoIP=20
> interconnect point (SIP call termination to another=20
> third-party provider which in turn does terminate the call to a PSAP).
>=20
>    Folks assume the work in GEOPRIV and ECRIT takes care of the above.
> The requirement gap I can see so far is based on:
> a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena i3).
> The ECRIT requirement document states:
> | This document only focuses on end-to-end IP-based calls,=20
> i.e., where=20
> | the emergency call originates from an IP end system and terminates=20
> | into an IP-capable PSAP, conveyed entirely over an IP network.
>=20
> b) GEOPRIV, ECRIT and SIPPING are concerned with location=20
> determination, emergency context resolution, location-based=20
> routing of the SIP call to the PSAP URI, location conveyance in SIP.
>=20
> c) SPEERMINT is about "architectures to identify, signal, and=20
> route delay-sensitive (real-time) communication sessions."...=20
> "These sessions use the SIP signaling protocol to enable=20
> peering between two or more administrative domains over IP networks."
>=20
> So, for example, the speermint requirement doc could state=20
> things like:
> 	- any applicable ECRIT requirements for exchanging=20
> emergency sip calls between 2 or more domains,
> 	- any SIP requirements that should be taken into consideration.
> For e.g. if available, convey location using sipping location=20
> conveyance. But also special handling of things like=20
> p-asserted-id when handing off emergency calls (if privacy is=20
> on in the originating domain, some VSPs remove p-asserted-id=20
> when handing out the call to another provider). In the case=20
> of an emergency call, for psap callback, p-asserted-id=20
> must/should be left in... so that the terminating VSP that=20
> connects to the PSAPs can identify the ANI.
>=20
> These are first thoughts to provide more examples. We could=20
> just have a couple of sentences highlighting the key=20
> requirements and giving pointers to consider when the handoff=20
> of emergency calls is possible.
>=20
> Penn wrote:
> > > I think we'd be better served NOT to take up the=20
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their=20
> > > requirements.
> This is fine and I'm ok not taking if we think we should just=20
> not take this for *now* but as Hannes points out, and as I=20
> tried to highlight above, we should not assume there will be=20
> a lot more requirement work in other IETF groups around the=20
> SIP interconnects and there may be some gaps (hence the scope=20
> question).
>=20
> Jean-Francois
>=20
> > -----Original Message-----
> > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > Sent: Friday, June 16, 2006 6:10 AM
> > To: Pfautz, Penn L, GBLAM
> > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > Subject: Re: [speermint] minimum set of requirements forSIP-=20
> > basedVoIPinterconnection and emergency calling
> >=20
> > The requirements work in ECRIT is finished. Here is the most recent
> > document:
> > http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > 10.txt
> >=20
> > It might be good to clearly describe what you want to=20
> accomplish with=20
> > emgerency services in the context of peering.
> >=20
> > Ciao
> > Hannes
> >=20
> >=20
> > Pfautz, Penn L, GBLAM wrote:
> > > I think we'd be better served NOT to take up the=20
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their=20
> > > requirements.
> > >
> > > Penn
> > >
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 7:39 AM
> > > To: Stastny Richard
> > > Cc: speermint@ietf.org
> > > Subject: RE: [speermint] minimum set of requirements=20
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Stastny Richard wrote:
> > >
> > >>>Some VSPs do have gateways connected to the PSTN and the whole
> > >>
> > >>selective routers thingy in the North American system for >example
> > to
> > >>do this; some don't and hand off the call to another VSP. The
> > question
> > >>of who does the ECRIT mapping protocol >resolution is also not=20
> > >>relevant here.
> > >>
> > >>Ok, but you mention here two issues out of speermint=20
> scope: gateways=20
> > >>to the PSTN and North American selective router stuff
> > >
> > >
> > > And that is clearly out of scope.
> > > Per my earlier email, the scope question was related to the other
> > case
> > > where emergency calls are not handled within a single VSP=20
> domain (or
> > for
> > > that matter, the case of an enterprise using a SIP interconnect
> > service
> > > and handing off all calls to the VSP).
> > >
> > > Jean-Francois.
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

_______________________________________________
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From speermint-bounces@ietf.org Fri Jun 16 16:08:47 2006
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Subject: Re: [speermint] minimum set ofrequirements
	forSIP-basedVoIPinterconnection and emergency calling
Date: Fri, 16 Jun 2006 22:11:54 +0200
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From: "Stastny Richard" <Richard.Stastny@oefeg.at>
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	"Jean-Francois Mule" <jf.mule@cablelabs.com>
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I could live with this proposal
=20
-sta

________________________________

Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
Gesendet: Fr 16.06.2006 21:44
An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
Cc: Stastny Richard; speermint@ietf.org
Betreff: RE: [speermint] minimum set ofrequirements =
forSIP-basedVoIPinterconnection and emergency calling



All,

Let me ask the question a different way:

Will all PSAPs belong to the same federation (a special PSAP federation
that defines the rules for accessing PSAP domains), or

will all PSAPs belong to various federations, where the rules for
handling SIP messages and what is needed for SOS calls is defined by
each federations?

Put me in the column for holding off on special cases of calls being
addressed until we have worked out the basic call case.  By then,
perhaps the ECRIT WG will have made further progress on the SOS call
delivery mechanisms and associated protocol elements.

Mike



> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Friday, June 16, 2006 9:39 AM
> To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> Cc: Stastny Richard; speermint@ietf.org
> Subject: RE: [speermint] minimum set ofrequirements
> forSIP-basedVoIPinterconnection and emergency calling
>
> Penn and Hannes,
>
>    Thanks for the input. Here is an attempt to clarify what
> could be in scope of speermint.
>
> Hannes wrote:
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
>
> The potential goal would be to write the minimum requirements
> a VSP should comply with when handing off an emergency call
> to another administrative domain over IP through a VoIP
> interconnect point (SIP call termination to another
> third-party provider which in turn does terminate the call to a PSAP).
>
>    Folks assume the work in GEOPRIV and ECRIT takes care of the above.
> The requirement gap I can see so far is based on:
> a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena i3).
> The ECRIT requirement document states:
> | This document only focuses on end-to-end IP-based calls,
> i.e., where
> | the emergency call originates from an IP end system and terminates
> | into an IP-capable PSAP, conveyed entirely over an IP network.
>
> b) GEOPRIV, ECRIT and SIPPING are concerned with location
> determination, emergency context resolution, location-based
> routing of the SIP call to the PSAP URI, location conveyance in SIP.
>
> c) SPEERMINT is about "architectures to identify, signal, and
> route delay-sensitive (real-time) communication sessions."...
> "These sessions use the SIP signaling protocol to enable
> peering between two or more administrative domains over IP networks."
>
> So, for example, the speermint requirement doc could state
> things like:
>       - any applicable ECRIT requirements for exchanging
> emergency sip calls between 2 or more domains,
>       - any SIP requirements that should be taken into consideration.
> For e.g. if available, convey location using sipping location
> conveyance. But also special handling of things like
> p-asserted-id when handing off emergency calls (if privacy is
> on in the originating domain, some VSPs remove p-asserted-id
> when handing out the call to another provider). In the case
> of an emergency call, for psap callback, p-asserted-id
> must/should be left in... so that the terminating VSP that
> connects to the PSAPs can identify the ANI.
>
> These are first thoughts to provide more examples. We could
> just have a couple of sentences highlighting the key
> requirements and giving pointers to consider when the handoff
> of emergency calls is possible.
>
> Penn wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > requirements.
> This is fine and I'm ok not taking if we think we should just
> not take this for *now* but as Hannes points out, and as I
> tried to highlight above, we should not assume there will be
> a lot more requirement work in other IETF groups around the
> SIP interconnects and there may be some gaps (hence the scope
> question).
>
> Jean-Francois
>
> > -----Original Message-----
> > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > Sent: Friday, June 16, 2006 6:10 AM
> > To: Pfautz, Penn L, GBLAM
> > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > Subject: Re: [speermint] minimum set of requirements forSIP-
> > basedVoIPinterconnection and emergency calling
> >
> > The requirements work in ECRIT is finished. Here is the most recent
> > document:
> > http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > 10.txt
> >
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
> >
> > Ciao
> > Hannes
> >
> >
> > Pfautz, Penn L, GBLAM wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > requirements.
> > >
> > > Penn
> > >
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 7:39 AM
> > > To: Stastny Richard
> > > Cc: speermint@ietf.org
> > > Subject: RE: [speermint] minimum set of requirements
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Stastny Richard wrote:
> > >
> > >>>Some VSPs do have gateways connected to the PSTN and the whole
> > >>
> > >>selective routers thingy in the North American system for >example
> > to
> > >>do this; some don't and hand off the call to another VSP. The
> > question
> > >>of who does the ECRIT mapping protocol >resolution is also not
> > >>relevant here.
> > >>
> > >>Ok, but you mention here two issues out of speermint
> scope: gateways
> > >>to the PSTN and North American selective router stuff
> > >
> > >
> > > And that is clearly out of scope.
> > > Per my earlier email, the scope question was related to the other
> > case
> > > where emergency calls are not handled within a single VSP
> domain (or
> > for
> > > that matter, the case of an enterprise using a SIP interconnect
> > service
> > > and handing off all calls to the VSP).
> > >
> > > Jean-Francois.
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>



_______________________________________________
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Subject: RE: [speermint] minimum set
	ofrequirementsforSIP-basedVoIPinterconnection and emergency calling
Date: Fri, 16 Jun 2006 16:17:08 -0400
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Michael Hammer (mhammer)" <mhammer@cisco.com>,
	"Jean-Francois Mule" <jf.mule@cablelabs.com>
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I agree that focus on initial call flows is wise for the time being, but
what's the harm in citing the requirement with fairly high-level generic
wording for now?!?!  FWIW - I join Mike in the
"holding-off-a-bit-until-we-sort-out-basic-functionality" column.
Crawl-Walk-Run.

-AU

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Friday, June 16, 2006 4:12 PM
To: Michael Hammer (mhammer); Jean-Francois Mule
Cc: speermint@ietf.org
Subject: Re: [speermint] minimum set
ofrequirementsforSIP-basedVoIPinterconnection and emergency calling

I could live with this proposal
=20
-sta

________________________________

Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
Gesendet: Fr 16.06.2006 21:44
An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
Cc: Stastny Richard; speermint@ietf.org
Betreff: RE: [speermint] minimum set ofrequirements
forSIP-basedVoIPinterconnection and emergency calling



All,

Let me ask the question a different way:

Will all PSAPs belong to the same federation (a special PSAP federation
that defines the rules for accessing PSAP domains), or

will all PSAPs belong to various federations, where the rules for
handling SIP messages and what is needed for SOS calls is defined by
each federations?

Put me in the column for holding off on special cases of calls being
addressed until we have worked out the basic call case.  By then,
perhaps the ECRIT WG will have made further progress on the SOS call
delivery mechanisms and associated protocol elements.

Mike



> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Friday, June 16, 2006 9:39 AM
> To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> Cc: Stastny Richard; speermint@ietf.org
> Subject: RE: [speermint] minimum set ofrequirements=20
> forSIP-basedVoIPinterconnection and emergency calling
>
> Penn and Hannes,
>
>    Thanks for the input. Here is an attempt to clarify what could be=20
> in scope of speermint.
>
> Hannes wrote:
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
>
> The potential goal would be to write the minimum requirements a VSP=20
> should comply with when handing off an emergency call to another=20
> administrative domain over IP through a VoIP interconnect point (SIP=20
> call termination to another third-party provider which in turn does=20
> terminate the call to a PSAP).
>
>    Folks assume the work in GEOPRIV and ECRIT takes care of the above.
> The requirement gap I can see so far is based on:
> a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena i3).
> The ECRIT requirement document states:
> | This document only focuses on end-to-end IP-based calls,
> i.e., where
> | the emergency call originates from an IP end system and terminates=20
> | into an IP-capable PSAP, conveyed entirely over an IP network.
>
> b) GEOPRIV, ECRIT and SIPPING are concerned with location=20
> determination, emergency context resolution, location-based routing of

> the SIP call to the PSAP URI, location conveyance in SIP.
>
> c) SPEERMINT is about "architectures to identify, signal, and route=20
> delay-sensitive (real-time) communication sessions."...
> "These sessions use the SIP signaling protocol to enable peering=20
> between two or more administrative domains over IP networks."
>
> So, for example, the speermint requirement doc could state things=20
> like:
>       - any applicable ECRIT requirements for exchanging emergency sip

> calls between 2 or more domains,
>       - any SIP requirements that should be taken into consideration.
> For e.g. if available, convey location using sipping location=20
> conveyance. But also special handling of things like p-asserted-id=20
> when handing off emergency calls (if privacy is on in the originating=20
> domain, some VSPs remove p-asserted-id when handing out the call to=20
> another provider). In the case of an emergency call, for psap=20
> callback, p-asserted-id must/should be left in... so that the=20
> terminating VSP that connects to the PSAPs can identify the ANI.
>
> These are first thoughts to provide more examples. We could just have=20
> a couple of sentences highlighting the key requirements and giving=20
> pointers to consider when the handoff of emergency calls is possible.
>
> Penn wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their=20
> > > requirements.
> This is fine and I'm ok not taking if we think we should just not take

> this for *now* but as Hannes points out, and as I tried to highlight=20
> above, we should not assume there will be a lot more requirement work=20
> in other IETF groups around the SIP interconnects and there may be=20
> some gaps (hence the scope question).
>
> Jean-Francois
>
> > -----Original Message-----
> > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > Sent: Friday, June 16, 2006 6:10 AM
> > To: Pfautz, Penn L, GBLAM
> > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > Subject: Re: [speermint] minimum set of requirements forSIP-=20
> > basedVoIPinterconnection and emergency calling
> >
> > The requirements work in ECRIT is finished. Here is the most recent
> > document:
> > http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > 10.txt
> >
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
> >
> > Ciao
> > Hannes
> >
> >
> > Pfautz, Penn L, GBLAM wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their=20
> > > requirements.
> > >
> > > Penn
> > >
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 7:39 AM
> > > To: Stastny Richard
> > > Cc: speermint@ietf.org
> > > Subject: RE: [speermint] minimum set of requirements=20
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Stastny Richard wrote:
> > >
> > >>>Some VSPs do have gateways connected to the PSTN and the whole
> > >>
> > >>selective routers thingy in the North American system for >example
> > to
> > >>do this; some don't and hand off the call to another VSP. The
> > question
> > >>of who does the ECRIT mapping protocol >resolution is also not=20
> > >>relevant here.
> > >>
> > >>Ok, but you mention here two issues out of speermint
> scope: gateways
> > >>to the PSTN and North American selective router stuff
> > >
> > >
> > > And that is clearly out of scope.
> > > Per my earlier email, the scope question was related to the other
> > case
> > > where emergency calls are not handled within a single VSP
> domain (or
> > for
> > > that matter, the case of an enterprise using a SIP interconnect
> > service
> > > and handing off all calls to the VSP).
> > >
> > > Jean-Francois.
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>



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>> -----Original Message-----
>> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com] 

>> A service provide would be best served to provide its 
>> subscribers a way to contact anyone else.  The burden to 
>> facilitate that session is on the SP.  Will a SP in Topeka 
>> trust a SP in Oslo enough to establish a direct peering 
>> arrangement?  Without that direct peer, how will the 
>> subscribers of the Topeka SP access the folks in Oslo?  An 
>> intermediary most likely will be involved with the value 
>> being reachability and trust that should a call be sent to 
>> the transit provider, it will be completed as intended by the 
>> calling party. At the end of the day, it will be the one that 
>> will complete the call that gets the buck.  Unfortunately 
>> it's the PSTN that facilitates that Topeka to Oslo call today 
>> more often than not.

Assuming the provider for the caller can determine the provider for the 
callee, then I fail to see how introducing an intermediary can improve 
trust. No matter how many intermediaries are involved, ultimately the 
callee's provider will be involved in the call. Establishing a direct 
peering relationship seems at least as good, if not better.

	Paul

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From speermint-bounces@ietf.org Sun Jun 18 19:51:21 2006
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Paul,

It's clear that most of us Engineers (including me) don't understand what's
required to establish a business relationship for telephonic communications.
I'm guessing there are issues with taxation, regulation, service level
agreements, minimum usage, billing issues, currency issues, credit, and
more. Although I've never seen a peering agreement for voice traffic, I'm
guessing they are comprehensive. I personally am aware of IP Peering
relationships for voice that even have damage clauses for periods of service
interruption.

Maybe somebody who understands why these relationships are so difficult to
establish and maintain could shed some light on this. If anybody could
easily and quickly establish a new peering relationship, then your position
is correct. Maybe you can help resolve all of the business issues to enable
"easy and quick" peering with on contracts, no taxes or regulation, and no
service level agreements.

I do believe that Adam is correct that trust will depend on the signaling
infrastructure being open only to those networks that are trust worthy --
with pre-established contractual guarantees. Any network that would break
the SIPS chain of trust MUST be excluded, or we need to find a way to ensure
that every party in the communication knows that the call is leaving the
trusted world and entering into untrusted networks. As far as I know, there
is no way currently to know that a SIP communication has followed a SIPS
trusted set of proxies from end-to-end.
-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Saturday, June 17, 2006 4:12 PM
To: Livingood, Jason
Cc: speermint@ietf.org; Uzelac, Adam
Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture

>> -----Original Message-----
>> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com] 

>> A service provide would be best served to provide its 
>> subscribers a way to contact anyone else.  The burden to 
>> facilitate that session is on the SP.  Will a SP in Topeka 
>> trust a SP in Oslo enough to establish a direct peering 
>> arrangement?  Without that direct peer, how will the 
>> subscribers of the Topeka SP access the folks in Oslo?  An 
>> intermediary most likely will be involved with the value 
>> being reachability and trust that should a call be sent to 
>> the transit provider, it will be completed as intended by the 
>> calling party. At the end of the day, it will be the one that 
>> will complete the call that gets the buck.  Unfortunately 
>> it's the PSTN that facilitates that Topeka to Oslo call today 
>> more often than not.

Assuming the provider for the caller can determine the provider for the 
callee, then I fail to see how introducing an intermediary can improve 
trust. No matter how many intermediaries are involved, ultimately the 
callee's provider will be involved in the call. Establishing a direct 
peering relationship seems at least as good, if not better.

	Paul

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From: "Rohan Mahy" <rohan.mahy@gmail.com>
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Subject: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
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Hi Folks,

I just submitted a draft with a minimalist approach to peering.  I
thought this might be a useful way to have a productive discussion on
requirements.

Until it appears in the archive, you can get it here:

http://scm.sipfoundry.org/rep/ietf-drafts/rohan/direct-peering/draft-mahy-speermint-direct-peering-00.html
http://scm.sipfoundry.org/rep/ietf-drafts/rohan/direct-peering/draft-mahy-speermint-direct-peering-00.txt

thanks,
-rohan

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From speermint-bounces@ietf.org Mon Jun 19 09:40:24 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Stastny Richard'" <Richard.Stastny@oefeg.at>,
	"'Michael Hammer \(mhammer\)'" <mhammer@cisco.com>,
	"'Jean-Francois Mule'" <jf.mule@cablelabs.com>
Subject: RE: [speermint] minimum set
	ofrequirementsforSIP-basedVoIPinterconnection and emergency calling
Date: Mon, 19 Jun 2006 09:40:11 -0400
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I can too.

I do think it will be fairly common for an originating network to hand off
an emergency call to another network.  That network may be one operated by
or for the PSAP, but it will operate much like any other SP.  It probably
will have at least two different mechanisms for interconnect: one very much
a "peering" relationship, well within the scope of speermint, and the other
pretty much an open Internet connection most times accepting any call sent
to it.  The security profiles of these two mechanisms will be different.

However, it's not clear to me whether ecrit takes the speermint work and
runs with that, speermint takes ecrit work and puts words in its document,
or we leave it to the authorities and their own standards groups to do that
work, as we are talking about precisely how a call is handed from an SP to a
PSAP, and the PSAP side of that is governed by the emergency authorities.  I
think ecrit is definitely going to precisely specify the open Internet
interconnect.  It may be that this whole area collapses easily into the
basic speermint model, except for the location header and any priority/QoS
stuff that is peculiar to emergency calls.

Brian

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at] 
Sent: Friday, June 16, 2006 4:12 PM
To: Michael Hammer (mhammer); Jean-Francois Mule
Cc: speermint@ietf.org
Subject: Re: [speermint] minimum set
ofrequirementsforSIP-basedVoIPinterconnection and emergency calling

I could live with this proposal
 
-sta

________________________________

Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
Gesendet: Fr 16.06.2006 21:44
An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
Cc: Stastny Richard; speermint@ietf.org
Betreff: RE: [speermint] minimum set ofrequirements
forSIP-basedVoIPinterconnection and emergency calling



All,

Let me ask the question a different way:

Will all PSAPs belong to the same federation (a special PSAP federation
that defines the rules for accessing PSAP domains), or

will all PSAPs belong to various federations, where the rules for
handling SIP messages and what is needed for SOS calls is defined by
each federations?

Put me in the column for holding off on special cases of calls being
addressed until we have worked out the basic call case.  By then,
perhaps the ECRIT WG will have made further progress on the SOS call
delivery mechanisms and associated protocol elements.

Mike



> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Friday, June 16, 2006 9:39 AM
> To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> Cc: Stastny Richard; speermint@ietf.org
> Subject: RE: [speermint] minimum set ofrequirements
> forSIP-basedVoIPinterconnection and emergency calling
>
> Penn and Hannes,
>
>    Thanks for the input. Here is an attempt to clarify what
> could be in scope of speermint.
>
> Hannes wrote:
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
>
> The potential goal would be to write the minimum requirements
> a VSP should comply with when handing off an emergency call
> to another administrative domain over IP through a VoIP
> interconnect point (SIP call termination to another
> third-party provider which in turn does terminate the call to a PSAP).
>
>    Folks assume the work in GEOPRIV and ECRIT takes care of the above.
> The requirement gap I can see so far is based on:
> a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena i3).
> The ECRIT requirement document states:
> | This document only focuses on end-to-end IP-based calls,
> i.e., where
> | the emergency call originates from an IP end system and terminates
> | into an IP-capable PSAP, conveyed entirely over an IP network.
>
> b) GEOPRIV, ECRIT and SIPPING are concerned with location
> determination, emergency context resolution, location-based
> routing of the SIP call to the PSAP URI, location conveyance in SIP.
>
> c) SPEERMINT is about "architectures to identify, signal, and
> route delay-sensitive (real-time) communication sessions."...
> "These sessions use the SIP signaling protocol to enable
> peering between two or more administrative domains over IP networks."
>
> So, for example, the speermint requirement doc could state
> things like:
>       - any applicable ECRIT requirements for exchanging
> emergency sip calls between 2 or more domains,
>       - any SIP requirements that should be taken into consideration.
> For e.g. if available, convey location using sipping location
> conveyance. But also special handling of things like
> p-asserted-id when handing off emergency calls (if privacy is
> on in the originating domain, some VSPs remove p-asserted-id
> when handing out the call to another provider). In the case
> of an emergency call, for psap callback, p-asserted-id
> must/should be left in... so that the terminating VSP that
> connects to the PSAPs can identify the ANI.
>
> These are first thoughts to provide more examples. We could
> just have a couple of sentences highlighting the key
> requirements and giving pointers to consider when the handoff
> of emergency calls is possible.
>
> Penn wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > requirements.
> This is fine and I'm ok not taking if we think we should just
> not take this for *now* but as Hannes points out, and as I
> tried to highlight above, we should not assume there will be
> a lot more requirement work in other IETF groups around the
> SIP interconnects and there may be some gaps (hence the scope
> question).
>
> Jean-Francois
>
> > -----Original Message-----
> > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > Sent: Friday, June 16, 2006 6:10 AM
> > To: Pfautz, Penn L, GBLAM
> > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > Subject: Re: [speermint] minimum set of requirements forSIP-
> > basedVoIPinterconnection and emergency calling
> >
> > The requirements work in ECRIT is finished. Here is the most recent
> > document:
> > http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > 10.txt
> >
> > It might be good to clearly describe what you want to
> accomplish with
> > emgerency services in the context of peering.
> >
> > Ciao
> > Hannes
> >
> >
> > Pfautz, Penn L, GBLAM wrote:
> > > I think we'd be better served NOT to take up the
> emergency use case
> > for
> > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > requirements.
> > >
> > > Penn
> > >
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 7:39 AM
> > > To: Stastny Richard
> > > Cc: speermint@ietf.org
> > > Subject: RE: [speermint] minimum set of requirements
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Stastny Richard wrote:
> > >
> > >>>Some VSPs do have gateways connected to the PSTN and the whole
> > >>
> > >>selective routers thingy in the North American system for >example
> > to
> > >>do this; some don't and hand off the call to another VSP. The
> > question
> > >>of who does the ECRIT mapping protocol >resolution is also not
> > >>relevant here.
> > >>
> > >>Ok, but you mention here two issues out of speermint
> scope: gateways
> > >>to the PSTN and North American selective router stuff
> > >
> > >
> > > And that is clearly out of scope.
> > > Per my earlier email, the scope question was related to the other
> > case
> > > where emergency calls are not handled within a single VSP
> domain (or
> > for
> > > that matter, the case of an enterprise using a SIP interconnect
> > service
> > > and handing off all calls to the VSP).
> > >
> > > Jean-Francois.
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>



_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint


_______________________________________________
Speermint mailing list
Speermint@ietf.org
https://www1.ietf.org/mailman/listinfo/speermint



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Subject: RE: [speermint] minimum
	setofrequirementsforSIP-basedVoIPinterconnection and emergency calling
Date: Mon, 19 Jun 2006 07:59:54 -0600
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Brian Rosen" <br@brianrosen.net>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>,
	"Michael Hammer (mhammer)" <mhammer@cisco.com>
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Brian wrote:

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Monday, June 19, 2006 7:40 AM
> To: 'Stastny Richard'; 'Michael Hammer (mhammer)'; Jean-Francois Mule
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum setofrequirementsforSIP-
> basedVoIPinterconnection and emergency calling
>=20
> I can too.
>=20
> I do think it will be fairly common for an originating network to hand
> off
> an emergency call to another network.  That network may be one
> operated by
> or for the PSAP, but it will operate much like any other SP.  It
> probably
> will have at least two different mechanisms for interconnect: one very
> much
> a "peering" relationship, well within the scope of speermint, and the
> other
> pretty much an open Internet connection most times accepting any call
> sent
> to it.  The security profiles of these two mechanisms will be
> different.
>=20
> However, it's not clear to me whether ecrit takes the speermint work
> and
> runs with that, speermint takes ecrit work and puts words in its
> document,

I thought that wg charters could tell you that. I provided my reading; I
could care less about whether this is done in ECRIT or SPEERMINT but
something should be done for session peering of emergency calls.

> or we leave it to the authorities and their own standards groups to do
> that
> work, as we are talking about precisely how a call is handed from an
> SP to a
> PSAP, and the PSAP side of that is governed by the emergency
> authorities.  I
> think ecrit is definitely going to precisely specify the open Internet
> interconnect.  It may be that this whole area collapses easily into
> the
> basic speermint model, except for the location header and any
> priority/QoS
> stuff that is peculiar to emergency calls.

Ack.

>=20
> Brian
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Friday, June 16, 2006 4:12 PM
> To: Michael Hammer (mhammer); Jean-Francois Mule
> Cc: speermint@ietf.org
> Subject: Re: [speermint] minimum set
> ofrequirementsforSIP-basedVoIPinterconnection and emergency calling
>=20
> I could live with this proposal
>=20
> -sta
>=20
> ________________________________
>=20
> Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> Gesendet: Fr 16.06.2006 21:44
> An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
> Cc: Stastny Richard; speermint@ietf.org
> Betreff: RE: [speermint] minimum set ofrequirements
> forSIP-basedVoIPinterconnection and emergency calling
>=20
>=20
>=20
> All,
>=20
> Let me ask the question a different way:
>=20
> Will all PSAPs belong to the same federation (a special PSAP
> federation
> that defines the rules for accessing PSAP domains), or
>=20
> will all PSAPs belong to various federations, where the rules for
> handling SIP messages and what is needed for SOS calls is defined by
> each federations?
>=20
> Put me in the column for holding off on special cases of calls being
> addressed until we have worked out the basic call case.  By then,
> perhaps the ECRIT WG will have made further progress on the SOS call
> elivery mechanisms and associated protocol elements.
>=20
> Mike
>=20
>=20
>=20
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Friday, June 16, 2006 9:39 AM
> > To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> > Cc: Stastny Richard; speermint@ietf.org
> > Subject: RE: [speermint] minimum set ofrequirements
> > forSIP-basedVoIPinterconnection and emergency calling
> >
> > Penn and Hannes,
> >
> >    Thanks for the input. Here is an attempt to clarify what
> > could be in scope of speermint.
> >
> > Hannes wrote:
> > > It might be good to clearly describe what you want to
> > accomplish with
> > > emgerency services in the context of peering.
> >
> > The potential goal would be to write the minimum requirements
> > a VSP should comply with when handing off an emergency call
> > to another administrative domain over IP through a VoIP
> > interconnect point (SIP call termination to another
> > third-party provider which in turn does terminate the call to a
> PSAP).
> >
> >    Folks assume the work in GEOPRIV and ECRIT takes care of the
> above.
> > The requirement gap I can see so far is based on:
> > a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena
> i3).
> > The ECRIT requirement document states:
> > | This document only focuses on end-to-end IP-based calls,
> > i.e., where
> > | the emergency call originates from an IP end system and terminates
> > | into an IP-capable PSAP, conveyed entirely over an IP network.
> >
> > b) GEOPRIV, ECRIT and SIPPING are concerned with location
> > determination, emergency context resolution, location-based
> > routing of the SIP call to the PSAP URI, location conveyance in SIP.
> >
> > c) SPEERMINT is about "architectures to identify, signal, and
> > route delay-sensitive (real-time) communication sessions."...
> > "These sessions use the SIP signaling protocol to enable
> > peering between two or more administrative domains over IP
> networks."
> >
> > So, for example, the speermint requirement doc could state
> > things like:
> >       - any applicable ECRIT requirements for exchanging
> > emergency sip calls between 2 or more domains,
> >       - any SIP requirements that should be taken into
consideration.
> > For e.g. if available, convey location using sipping location
> > conveyance. But also special handling of things like
> > p-asserted-id when handing off emergency calls (if privacy is
> > on in the originating domain, some VSPs remove p-asserted-id
> > when handing out the call to another provider). In the case
> > of an emergency call, for psap callback, p-asserted-id
> > must/should be left in... so that the terminating VSP that
> > connects to the PSAPs can identify the ANI.
> >
> > These are first thoughts to provide more examples. We could
> > just have a couple of sentences highlighting the key
> > requirements and giving pointers to consider when the handoff
> > of emergency calls is possible.
> >
> > Penn wrote:
> > > > I think we'd be better served NOT to take up the
> > emergency use case
> > > for
> > > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > > requirements.
> > This is fine and I'm ok not taking if we think we should just
> > not take this for *now* but as Hannes points out, and as I
> > tried to highlight above, we should not assume there will be
> > a lot more requirement work in other IETF groups around the
> > SIP interconnects and there may be some gaps (hence the scope
> > question).
> >
> > Jean-Francois
> >
> > > -----Original Message-----
> > > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > > Sent: Friday, June 16, 2006 6:10 AM
> > > To: Pfautz, Penn L, GBLAM
> > > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > > Subject: Re: [speermint] minimum set of requirements forSIP-
> > > basedVoIPinterconnection and emergency calling
> > >
> > > The requirements work in ECRIT is finished. Here is the most
> recent
> > > document:
> > > http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > > 10.txt
> > >
> > > It might be good to clearly describe what you want to
> > accomplish with
> > > emgerency services in the context of peering.
> > >
> > > Ciao
> > > Hannes
> > >
> > >
> > > Pfautz, Penn L, GBLAM wrote:
> > > > I think we'd be better served NOT to take up the
> > emergency use case
> > > for
> > > > now, until the other groups (GEOPRIV and ECRIT) solidify their
> > > > requirements.
> > > >
> > > > Penn
> > > >
> > > > -----Original Message-----
> > > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > > Sent: Friday, June 16, 2006 7:39 AM
> > > > To: Stastny Richard
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: [speermint] minimum set of requirements
> > > > forSIP-basedVoIPinterconnection and emergency calling
> > > >
> > > > Stastny Richard wrote:
> > > >
> > > >>>Some VSPs do have gateways connected to the PSTN and the whole
> > > >>
> > > >>selective routers thingy in the North American system for
> >example
> > > to
> > > >>do this; some don't and hand off the call to another VSP. The
> > > question
> > > >>of who does the ECRIT mapping protocol >resolution is also not
> > > >>relevant here.
> > > >>
> > > >>Ok, but you mention here two issues out of speermint
> > scope: gateways
> > > >>to the PSTN and North American selective router stuff
> > > >
> > > >
> > > > And that is clearly out of scope.
> > > > Per my earlier email, the scope question was related to the
> other
> > > case
> > > > where emergency calls are not handled within a single VSP
> > domain (or
> > > for
> > > > that matter, the case of an enterprise using a SIP interconnect
> > > service
> > > > and handing off all calls to the VSP).
> > > >
> > > > Jean-Francois.
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
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To: "Michael Hammer (mhammer)" <mhammer@cisco.com>,
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Mike,

  Thanks for the input.

Mike wrote:
> All,
>=20
> Let me ask the question a different way:
>=20
> Will all PSAPs belong to the same federation (a special PSAP
> federation
> that defines the rules for accessing PSAP domains), or
>=20
> will all PSAPs belong to various federations, where the rules for
> handling SIP messages and what is needed for SOS calls is defined by
> each federations?

Each "PSAP federation" will likely define what is needed for SOS calls
(likely at a regional or national level, take NENA for e.g.).=20
But wouldn't it be a plus for all the actors (those regional PSAP
federations, multinational corporations connecting IP-PBX to SIP voice
providers, VSPs crossing multiple of those regional boundaries using a
common vendor solution) to point to a basic and minimum set of
applicable requirements for exchanging emergency calling between those
actors?
We have most of the requirement pieces, we just need the mint glue and a
document to stick them in.

> Put me in the column for holding off on special cases of calls being
> addressed until we have worked out the basic call case. =20
Ok, seems the right order anyway.

> By then,
> perhaps the ECRIT WG will have made further progress on the SOS call
> delivery mechanisms and associated protocol elements.
As Hannes reminded us, the requirement work in ECRIT is almost done.
This is where folks think there is no gap and take a 'wait and see'
position. Please re-read the ECRIT WG charter again.=20
I understand the following:
	- ECRIT will provide for how to identify a session initiation
request it for an emergency call and ECRIT will describe how "route an
emergency call based on location info",
	- once a SIP entity gets the PSAP URI back from the LoST mapping
protocol for e.g., this is where the speermint scope starts.

To draw a parallel, this is very similar to the relationship between
ENUM and SPEERMINT when an e.164 is used (ref. SPEERMINT Context in the
terminology draft): once call routing data is obtained from ENUM
(e.164-->URI) and you have the SIP URI, session-level signaling and the
associated requirements for SIP message exchange is in scope of
speermint.

What am I missing?
Jean-Francois.


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Subject: RE: [speermint] minimum
	setofrequirementsforSIP-basedVoIPinterconnection and emergency calling
Date: Mon, 19 Jun 2006 10:21:35 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>,
	"Brian Rosen" <br@brianrosen.net>,
	"Stastny Richard" <Richard.Stastny@oefeg.at>
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I need to look more closely at what this E-PSCF function is doing.  It
may for all I know be the ingress proxy for a PSAP domain.  Or maybe the
egress from an SP that needs to peer to a PSAP domain.

At some point we need to meld the two WG efforts together.

Mike


> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
> Sent: Monday, June 19, 2006 10:00 AM
> To: Brian Rosen; Stastny Richard; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum=20
> setofrequirementsforSIP-basedVoIPinterconnection and emergency calling
>=20
> Brian wrote:
>=20
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Monday, June 19, 2006 7:40 AM
> > To: 'Stastny Richard'; 'Michael Hammer (mhammer)';=20
> Jean-Francois Mule
> > Cc: speermint@ietf.org
> > Subject: RE: [speermint] minimum setofrequirementsforSIP-=20
> > basedVoIPinterconnection and emergency calling
> >=20
> > I can too.
> >=20
> > I do think it will be fairly common for an originating=20
> network to hand=20
> > off an emergency call to another network.  That network may be one=20
> > operated by or for the PSAP, but it will operate much like=20
> any other=20
> > SP.  It probably will have at least two different mechanisms for=20
> > interconnect: one very much a "peering" relationship, well=20
> within the=20
> > scope of speermint, and the other pretty much an open Internet=20
> > connection most times accepting any call sent to it.  The security=20
> > profiles of these two mechanisms will be different.
> >=20
> > However, it's not clear to me whether ecrit takes the=20
> speermint work=20
> > and runs with that, speermint takes ecrit work and puts=20
> words in its=20
> > document,
>=20
> I thought that wg charters could tell you that. I provided my=20
> reading; I could care less about whether this is done in=20
> ECRIT or SPEERMINT but something should be done for session=20
> peering of emergency calls.
>=20
> > or we leave it to the authorities and their own standards=20
> groups to do=20
> > that work, as we are talking about precisely how a call is=20
> handed from=20
> > an SP to a PSAP, and the PSAP side of that is governed by the=20
> > emergency authorities.  I think ecrit is definitely going=20
> to precisely=20
> > specify the open Internet interconnect.  It may be that this whole=20
> > area collapses easily into the basic speermint model,=20
> except for the=20
> > location header and any priority/QoS stuff that is peculiar to=20
> > emergency calls.
>=20
> Ack.
>=20
> >=20
> > Brian
> >=20
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Friday, June 16, 2006 4:12 PM
> > To: Michael Hammer (mhammer); Jean-Francois Mule
> > Cc: speermint@ietf.org
> > Subject: Re: [speermint] minimum set
> > ofrequirementsforSIP-basedVoIPinterconnection and emergency calling
> >=20
> > I could live with this proposal
> >=20
> > -sta
> >=20
> > ________________________________
> >=20
> > Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> > Gesendet: Fr 16.06.2006 21:44
> > An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
> > Cc: Stastny Richard; speermint@ietf.org
> > Betreff: RE: [speermint] minimum set ofrequirements=20
> > forSIP-basedVoIPinterconnection and emergency calling
> >=20
> >=20
> >=20
> > All,
> >=20
> > Let me ask the question a different way:
> >=20
> > Will all PSAPs belong to the same federation (a special PSAP=20
> > federation that defines the rules for accessing PSAP domains), or
> >=20
> > will all PSAPs belong to various federations, where the rules for=20
> > handling SIP messages and what is needed for SOS calls is=20
> defined by=20
> > each federations?
> >=20
> > Put me in the column for holding off on special cases of=20
> calls being=20
> > addressed until we have worked out the basic call case.  By then,=20
> > perhaps the ECRIT WG will have made further progress on the=20
> SOS call=20
> > elivery mechanisms and associated protocol elements.
> >=20
> > Mike
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 9:39 AM
> > > To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> > > Cc: Stastny Richard; speermint@ietf.org
> > > Subject: RE: [speermint] minimum set ofrequirements=20
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Penn and Hannes,
> > >
> > >    Thanks for the input. Here is an attempt to clarify=20
> what could be=20
> > > in scope of speermint.
> > >
> > > Hannes wrote:
> > > > It might be good to clearly describe what you want to
> > > accomplish with
> > > > emgerency services in the context of peering.
> > >
> > > The potential goal would be to write the minimum=20
> requirements a VSP=20
> > > should comply with when handing off an emergency call to another=20
> > > administrative domain over IP through a VoIP interconnect=20
> point (SIP=20
> > > call termination to another third-party provider which in=20
> turn does=20
> > > terminate the call to a
> > PSAP).
> > >
> > >    Folks assume the work in GEOPRIV and ECRIT takes care of the
> > above.
> > > The requirement gap I can see so far is based on:
> > > a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena
> > i3).
> > > The ECRIT requirement document states:
> > > | This document only focuses on end-to-end IP-based calls,
> > > i.e., where
> > > | the emergency call originates from an IP end system and=20
> terminates=20
> > > | into an IP-capable PSAP, conveyed entirely over an IP network.
> > >
> > > b) GEOPRIV, ECRIT and SIPPING are concerned with location=20
> > > determination, emergency context resolution,=20
> location-based routing=20
> > > of the SIP call to the PSAP URI, location conveyance in SIP.
> > >
> > > c) SPEERMINT is about "architectures to identify, signal,=20
> and route=20
> > > delay-sensitive (real-time) communication sessions."...
> > > "These sessions use the SIP signaling protocol to enable peering=20
> > > between two or more administrative domains over IP
> > networks."
> > >
> > > So, for example, the speermint requirement doc could state things=20
> > > like:
> > >       - any applicable ECRIT requirements for exchanging=20
> emergency=20
> > > sip calls between 2 or more domains,
> > >       - any SIP requirements that should be taken into
> consideration.
> > > For e.g. if available, convey location using sipping location=20
> > > conveyance. But also special handling of things like=20
> p-asserted-id=20
> > > when handing off emergency calls (if privacy is on in the=20
> > > originating domain, some VSPs remove p-asserted-id when=20
> handing out=20
> > > the call to another provider). In the case of an=20
> emergency call, for=20
> > > psap callback, p-asserted-id must/should be left in... so=20
> that the=20
> > > terminating VSP that connects to the PSAPs can identify the ANI.
> > >
> > > These are first thoughts to provide more examples. We could just=20
> > > have a couple of sentences highlighting the key requirements and=20
> > > giving pointers to consider when the handoff of emergency=20
> calls is=20
> > > possible.
> > >
> > > Penn wrote:
> > > > > I think we'd be better served NOT to take up the
> > > emergency use case
> > > > for
> > > > > now, until the other groups (GEOPRIV and ECRIT)=20
> solidify their=20
> > > > > requirements.
> > > This is fine and I'm ok not taking if we think we should just not=20
> > > take this for *now* but as Hannes points out, and as I tried to=20
> > > highlight above, we should not assume there will be a lot more=20
> > > requirement work in other IETF groups around the SIP=20
> interconnects=20
> > > and there may be some gaps (hence the scope question).
> > >
> > > Jean-Francois
> > >
> > > > -----Original Message-----
> > > > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > > > Sent: Friday, June 16, 2006 6:10 AM
> > > > To: Pfautz, Penn L, GBLAM
> > > > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > > > Subject: Re: [speermint] minimum set of requirements forSIP-=20
> > > > basedVoIPinterconnection and emergency calling
> > > >
> > > > The requirements work in ECRIT is finished. Here is the most
> > recent
> > > > document:
> > > >=20
> http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > > > 10.txt
> > > >
> > > > It might be good to clearly describe what you want to
> > > accomplish with
> > > > emgerency services in the context of peering.
> > > >
> > > > Ciao
> > > > Hannes
> > > >
> > > >
> > > > Pfautz, Penn L, GBLAM wrote:
> > > > > I think we'd be better served NOT to take up the
> > > emergency use case
> > > > for
> > > > > now, until the other groups (GEOPRIV and ECRIT)=20
> solidify their=20
> > > > > requirements.
> > > > >
> > > > > Penn
> > > > >
> > > > > -----Original Message-----
> > > > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > > > Sent: Friday, June 16, 2006 7:39 AM
> > > > > To: Stastny Richard
> > > > > Cc: speermint@ietf.org
> > > > > Subject: RE: [speermint] minimum set of requirements=20
> > > > > forSIP-basedVoIPinterconnection and emergency calling
> > > > >
> > > > > Stastny Richard wrote:
> > > > >
> > > > >>>Some VSPs do have gateways connected to the PSTN and=20
> the whole
> > > > >>
> > > > >>selective routers thingy in the North American system for
> > >example
> > > > to
> > > > >>do this; some don't and hand off the call to another VSP. The
> > > > question
> > > > >>of who does the ECRIT mapping protocol >resolution is=20
> also not=20
> > > > >>relevant here.
> > > > >>
> > > > >>Ok, but you mention here two issues out of speermint
> > > scope: gateways
> > > > >>to the PSTN and North American selective router stuff
> > > > >
> > > > >
> > > > > And that is clearly out of scope.
> > > > > Per my earlier email, the scope question was related to the
> > other
> > > > case
> > > > > where emergency calls are not handled within a single VSP
> > > domain (or
> > > > for
> > > > > that matter, the case of an enterprise using a SIP=20
> interconnect
> > > > service
> > > > > and handing off all calls to the VSP).
> > > > >
> > > > > Jean-Francois.
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >=20
> >=20
> >=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >=20
> >=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20

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An "E-CSCF" is an IMS term for a proxy in the originating network that
handles emergency calls.  It would probably be the exit proxy for the
originating domain.

An "ESRP" is a term used in ecrit for any intermediary in the domain the
PSAP is a part of (or a higher level domain when there is some hierarchy).
An ESRP could be the entrance proxy for the PSAP domain.

So the E-CSCF would 'peer' with the ESRP.

Brian

-----Original Message-----
From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com] 
Sent: Monday, June 19, 2006 10:22 AM
To: Jean-Francois Mule; Brian Rosen; Stastny Richard
Cc: speermint@ietf.org
Subject: RE: [speermint] minimum
setofrequirementsforSIP-basedVoIPinterconnection and emergency calling

I need to look more closely at what this E-PSCF function is doing.  It
may for all I know be the ingress proxy for a PSAP domain.  Or maybe the
egress from an SP that needs to peer to a PSAP domain.

At some point we need to meld the two WG efforts together.

Mike


> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com] 
> Sent: Monday, June 19, 2006 10:00 AM
> To: Brian Rosen; Stastny Richard; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum 
> setofrequirementsforSIP-basedVoIPinterconnection and emergency calling
> 
> Brian wrote:
> 
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Monday, June 19, 2006 7:40 AM
> > To: 'Stastny Richard'; 'Michael Hammer (mhammer)'; 
> Jean-Francois Mule
> > Cc: speermint@ietf.org
> > Subject: RE: [speermint] minimum setofrequirementsforSIP- 
> > basedVoIPinterconnection and emergency calling
> > 
> > I can too.
> > 
> > I do think it will be fairly common for an originating 
> network to hand 
> > off an emergency call to another network.  That network may be one 
> > operated by or for the PSAP, but it will operate much like 
> any other 
> > SP.  It probably will have at least two different mechanisms for 
> > interconnect: one very much a "peering" relationship, well 
> within the 
> > scope of speermint, and the other pretty much an open Internet 
> > connection most times accepting any call sent to it.  The security 
> > profiles of these two mechanisms will be different.
> > 
> > However, it's not clear to me whether ecrit takes the 
> speermint work 
> > and runs with that, speermint takes ecrit work and puts 
> words in its 
> > document,
> 
> I thought that wg charters could tell you that. I provided my 
> reading; I could care less about whether this is done in 
> ECRIT or SPEERMINT but something should be done for session 
> peering of emergency calls.
> 
> > or we leave it to the authorities and their own standards 
> groups to do 
> > that work, as we are talking about precisely how a call is 
> handed from 
> > an SP to a PSAP, and the PSAP side of that is governed by the 
> > emergency authorities.  I think ecrit is definitely going 
> to precisely 
> > specify the open Internet interconnect.  It may be that this whole 
> > area collapses easily into the basic speermint model, 
> except for the 
> > location header and any priority/QoS stuff that is peculiar to 
> > emergency calls.
> 
> Ack.
> 
> > 
> > Brian
> > 
> > -----Original Message-----
> > From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> > Sent: Friday, June 16, 2006 4:12 PM
> > To: Michael Hammer (mhammer); Jean-Francois Mule
> > Cc: speermint@ietf.org
> > Subject: Re: [speermint] minimum set
> > ofrequirementsforSIP-basedVoIPinterconnection and emergency calling
> > 
> > I could live with this proposal
> > 
> > -sta
> > 
> > ________________________________
> > 
> > Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]
> > Gesendet: Fr 16.06.2006 21:44
> > An: Jean-Francois Mule; Hannes Tschofenig; Pfautz, Penn L, GBLAM
> > Cc: Stastny Richard; speermint@ietf.org
> > Betreff: RE: [speermint] minimum set ofrequirements 
> > forSIP-basedVoIPinterconnection and emergency calling
> > 
> > 
> > 
> > All,
> > 
> > Let me ask the question a different way:
> > 
> > Will all PSAPs belong to the same federation (a special PSAP 
> > federation that defines the rules for accessing PSAP domains), or
> > 
> > will all PSAPs belong to various federations, where the rules for 
> > handling SIP messages and what is needed for SOS calls is 
> defined by 
> > each federations?
> > 
> > Put me in the column for holding off on special cases of 
> calls being 
> > addressed until we have worked out the basic call case.  By then, 
> > perhaps the ECRIT WG will have made further progress on the 
> SOS call 
> > elivery mechanisms and associated protocol elements.
> > 
> > Mike
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > Sent: Friday, June 16, 2006 9:39 AM
> > > To: Hannes Tschofenig; Pfautz, Penn L, GBLAM
> > > Cc: Stastny Richard; speermint@ietf.org
> > > Subject: RE: [speermint] minimum set ofrequirements 
> > > forSIP-basedVoIPinterconnection and emergency calling
> > >
> > > Penn and Hannes,
> > >
> > >    Thanks for the input. Here is an attempt to clarify 
> what could be 
> > > in scope of speermint.
> > >
> > > Hannes wrote:
> > > > It might be good to clearly describe what you want to
> > > accomplish with
> > > > emgerency services in the context of peering.
> > >
> > > The potential goal would be to write the minimum 
> requirements a VSP 
> > > should comply with when handing off an emergency call to another 
> > > administrative domain over IP through a VoIP interconnect 
> point (SIP 
> > > call termination to another third-party provider which in 
> turn does 
> > > terminate the call to a
> > PSAP).
> > >
> > >    Folks assume the work in GEOPRIV and ECRIT takes care of the
> > above.
> > > The requirement gap I can see so far is based on:
> > > a) the requirement work in ECRIT is for IP-enabled PSAPs (US/nena
> > i3).
> > > The ECRIT requirement document states:
> > > | This document only focuses on end-to-end IP-based calls,
> > > i.e., where
> > > | the emergency call originates from an IP end system and 
> terminates 
> > > | into an IP-capable PSAP, conveyed entirely over an IP network.
> > >
> > > b) GEOPRIV, ECRIT and SIPPING are concerned with location 
> > > determination, emergency context resolution, 
> location-based routing 
> > > of the SIP call to the PSAP URI, location conveyance in SIP.
> > >
> > > c) SPEERMINT is about "architectures to identify, signal, 
> and route 
> > > delay-sensitive (real-time) communication sessions."...
> > > "These sessions use the SIP signaling protocol to enable peering 
> > > between two or more administrative domains over IP
> > networks."
> > >
> > > So, for example, the speermint requirement doc could state things 
> > > like:
> > >       - any applicable ECRIT requirements for exchanging 
> emergency 
> > > sip calls between 2 or more domains,
> > >       - any SIP requirements that should be taken into
> consideration.
> > > For e.g. if available, convey location using sipping location 
> > > conveyance. But also special handling of things like 
> p-asserted-id 
> > > when handing off emergency calls (if privacy is on in the 
> > > originating domain, some VSPs remove p-asserted-id when 
> handing out 
> > > the call to another provider). In the case of an 
> emergency call, for 
> > > psap callback, p-asserted-id must/should be left in... so 
> that the 
> > > terminating VSP that connects to the PSAPs can identify the ANI.
> > >
> > > These are first thoughts to provide more examples. We could just 
> > > have a couple of sentences highlighting the key requirements and 
> > > giving pointers to consider when the handoff of emergency 
> calls is 
> > > possible.
> > >
> > > Penn wrote:
> > > > > I think we'd be better served NOT to take up the
> > > emergency use case
> > > > for
> > > > > now, until the other groups (GEOPRIV and ECRIT) 
> solidify their 
> > > > > requirements.
> > > This is fine and I'm ok not taking if we think we should just not 
> > > take this for *now* but as Hannes points out, and as I tried to 
> > > highlight above, we should not assume there will be a lot more 
> > > requirement work in other IETF groups around the SIP 
> interconnects 
> > > and there may be some gaps (hence the scope question).
> > >
> > > Jean-Francois
> > >
> > > > -----Original Message-----
> > > > From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net]
> > > > Sent: Friday, June 16, 2006 6:10 AM
> > > > To: Pfautz, Penn L, GBLAM
> > > > Cc: Jean-Francois Mule; Stastny Richard; speermint@ietf.org
> > > > Subject: Re: [speermint] minimum set of requirements forSIP- 
> > > > basedVoIPinterconnection and emergency calling
> > > >
> > > > The requirements work in ECRIT is finished. Here is the most
> > recent
> > > > document:
> > > > 
> http://www.ietf.org/internet-drafts/draft-ietf-ecrit-requirements-
> > > > 10.txt
> > > >
> > > > It might be good to clearly describe what you want to
> > > accomplish with
> > > > emgerency services in the context of peering.
> > > >
> > > > Ciao
> > > > Hannes
> > > >
> > > >
> > > > Pfautz, Penn L, GBLAM wrote:
> > > > > I think we'd be better served NOT to take up the
> > > emergency use case
> > > > for
> > > > > now, until the other groups (GEOPRIV and ECRIT) 
> solidify their 
> > > > > requirements.
> > > > >
> > > > > Penn
> > > > >
> > > > > -----Original Message-----
> > > > > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > > > > Sent: Friday, June 16, 2006 7:39 AM
> > > > > To: Stastny Richard
> > > > > Cc: speermint@ietf.org
> > > > > Subject: RE: [speermint] minimum set of requirements 
> > > > > forSIP-basedVoIPinterconnection and emergency calling
> > > > >
> > > > > Stastny Richard wrote:
> > > > >
> > > > >>>Some VSPs do have gateways connected to the PSTN and 
> the whole
> > > > >>
> > > > >>selective routers thingy in the North American system for
> > >example
> > > > to
> > > > >>do this; some don't and hand off the call to another VSP. The
> > > > question
> > > > >>of who does the ECRIT mapping protocol >resolution is 
> also not 
> > > > >>relevant here.
> > > > >>
> > > > >>Ok, but you mention here two issues out of speermint
> > > scope: gateways
> > > > >>to the PSTN and North American selective router stuff
> > > > >
> > > > >
> > > > > And that is clearly out of scope.
> > > > > Per my earlier email, the scope question was related to the
> > other
> > > > case
> > > > > where emergency calls are not handled within a single VSP
> > > domain (or
> > > > for
> > > > > that matter, the case of an enterprise using a SIP 
> interconnect
> > > > service
> > > > > and handing off all calls to the VSP).
> > > > >
> > > > > Jean-Francois.
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > 
> > 
> > 
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> > 
> > 
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> 


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Subject: [Speermint] use of RFC 3263 for locating peer's SIP server
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Hi,

   We have discussed the use of the DNS procedures specified in RFC3263
in speermint to resolve a SIP URI into a reachable host (IP address and
port), and transport protocol.

   To document the best current practices, could you contact me if you
have any implementation or operational experience in field use of 3263
mechanisms (full 3263 or partial NAPTR/SRV)?
	- based on Robert Sparks' presentation in IETF#65 SIPPING on the
status of the implementation in sip interoperability testing events:
	40% of implementers showing up in SIPit do NAPTR,=20
	50% do SRV.

	- when implemented on the products, how many folks use those
options? any feedback on the field use of 3263 for NAPTR lookup for SIP
transport protocol selection?
a) The RFC indicates that if the URI contains the transport protocol,
then it should be used (no lookup): is this common? Should it be?
b) In a few publicly-available product administration guides I have
looked at, NAPTR lookup for transport selection can be turned on/off
when it exists as anyone would expect. Do service providers use it? If
no, what are the issues?
=3D=3D> I'm told that some service providers mandate one transport =
protocol
(say TCP) for SIP termination and therefore, the transport protocol to
reach a peering partner is often statically configured along with the
sip route. Is this common?
=3D=3D=3D> some folks believe it is costly to do on a per query basis. =
Is this
a matter of recommending proper DNS caching and long enough TTL values
for those things?

	- what about SRV? Same questions.

If I get enough feedback, my goal is to provide a brief summary on the
above to the list, and possibly, draft one or two slides outline any
issues at IETF#66.

Thanks,
Jean-Francois.



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Date: Mon, 19 Jun 2006 17:11:58 +0200
From: Otmar Lendl <lendl@nic.at>
To: speermint@ietf.org
Subject: Re: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
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On 2006/06/19 10:06, Rohan Mahy <rohan.mahy@gmail.com> wrote:
> Hi Folks,
> 
> I just submitted a draft with a minimalist approach to peering.  I
> thought this might be a useful way to have a productive discussion on
> requirements.

Thanks for the contribution.

It's always helpful to have something concrete to talk about.

Some comments:

> 3.  Overview
> 
>    The approach described here assumes that the initiating peer first
>    discovers a SIP [2] target URI handled by the receiving peer, then
>    sends SIP requests directly to the receiving peer.

Perhaps you should also notice here that such ad-hoc peering implies
settlement-free peering.

>    Note that in a federation of domains, the initiating peer role can be
>    logically decomposed.  Domain A1 can forward requests to domain A2,
>    so that A2 actually opens a TLS connection to the receiving peer.
>    Likewise, the receiving peer function can be similarly decomposed.
>    How functions are distributed between or among domains or nodes is
>    not subject to this specification as long as the requirements
>    described in this document are met.

We have to be careful here to keep the terminology consistent between
the various documents.

I used "peering" for calls within a federation, you seem to use
"peering" only for inter-federation connects. The latter are not
covered by draft-lendl-speermint-federations-00.

A common understanding of this aspect will be important.

> 4.  Procedures

I like this section. It's very helpful for the discussion.

> 4.1.2  User ENUM lookup
> 
>    If an external E.164 address is the target, the initiating peer
>    consults the public "User ENUM" rooted at e164.arpa, according to the
>    procedures described in RFC 3761.  The peer MUST query for the "E2U+
>    sip" enumservice as described in RFC 3674 [11], but MAY check for
>    other enumservices.  The initiating peer MAY consult a cache or
>    alternate representation of the ENUM data rather than actual DNS
>    queries.  Also, the peer MAY skip actual DNS queries if the
>    initiating peer is sure that the target address country code is not
>    represented in e164.arpa.  If a sip: or sips: URI is chosen, the peer
>    skips to Section 4.1.5.

IMHO the URIs as returned by User-ENUM do not require TLS (except when
they are sips: ones) as described in 4.1.6.

They could point to end-user controlled devices (an Asterisk
installation, other VoIP-enabled PBX's or simple SIP services).  I see
no need to insist on anything more than pure RFC3263 for these cases.

> 4.1.6  Setup TLS connection
> 
>    Once a transport, port, and address are found, the initiating peer
>    will open or find a reusable TLS connection to the peer.  The
>    initiating provider MUST verify the server certificate which SHOULD
>    be rooted in a well-known certificate authority.  

That is going to be contentious. What are the "well-known certificate
authorities"? The ones which have their keys installed in MSIE and
Firefox?

Do you expect people to run white-lists based on the keys themselves
or based on special-purpose CAs?

Or blacklists to keep baddies out?

 If you allow the same CAs as used for TLS certs for websites then this
 is a huge community of potential peering partners. Not all of them
 are well-meaning and serious operators.

Do you expect to use CRLs?

What if the destination SIP server hosts multiple domains? Which
cert is he supposed to present to an incoming SIP/TLS connection?

(This is the same problem as hosting multiple https: servers on
a single server. Do you need one IP address per domain?)

>    The initiating
>    provider MUST be prepared to provide a TLS client certificate upon
>    request during the TLS handshake.  The client certificate MUST
>    contain a DNS or URI choice type in the subjectAltName which
>    corresponds to the domain asserted in the host production of the From
>    header URI.  The certificate SHOULD be valid and rooted in a well-
>    known certificate authority.

Same question again as above.

What about TLS connection reuse? If both SIP servers serve URIs
in multiple domains, can they reuse TLS connections which were
established based on calls to a different domain (but hosted
on the same server)?

>    Note that the client certificate MAY contain a list of entries in the
>    subjectAltName, only one of which has to match the domain in the From
>    header URI.

What are the scalability limits to that?

Any moderately sized ISP mail server hosts mails for thousands of
domains. Can this TLS setup scale to similar levels of domain hosting?

> 4.2  Receiving Peer
> 4.2.4  Receive SIP requests

>    The receiving peer MAY reject incoming requests due to local policy.
>    When a request is rejected because the initiating peer is not
>    authorized to peer, the receiving peer SHOULD respond with a 403
>    response with the reason phrase "Unsupported Peer".

You have to distinguish rejects based on TLS negotiations and those on
the SIP layer.

What is the sending peer supposed to do in that case?
Fall back to 4.1.4? Cache that result? If caching: per destination
URI or per SIP proxy? For how long?

How do you prevent unsuccessful TLS establishments to become a kind of
DoS attack? Firewalling could be an option on the destination side,
but dropped packets lead to timeouts on the sender side which lead to
unpredictable call-establishment delays.

--------

As I wrote above: this draft is a good starting point for discussions.

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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From speermint-bounces@ietf.org Mon Jun 19 11:44:37 2006
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Subject: RE: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
Date: Mon, 19 Jun 2006 11:44:29 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Otmar Lendl" <lendl@nic.at>, <speermint@ietf.org>
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To add to the discussion.

I think it might be useful that federations as part of their role in
peering to establish certificate authorities.

One aspect of the comments below that we have to keep very distinct is
that of authentication from authorization.  You could potentially know,
because a certificate is valid, exactly who the offending SP is.
Whether you then white list or black list is a separate authorization
issue.  In that case, if they subscribe to get your policy, you send
them to the ingress proxy at:  0.0.0.0  :)

Mike


> -----Original Message-----
> From: Otmar Lendl [mailto:lendl@nic.at]=20
> Sent: Monday, June 19, 2006 11:12 AM
> To: speermint@ietf.org
> Subject: Re: [Speermint] Just submitted=20
> draft-mahy-speermint-direct-peering-00
>=20
> On 2006/06/19 10:06, Rohan Mahy <rohan.mahy@gmail.com> wrote:
> > Hi Folks,
> >=20
> > I just submitted a draft with a minimalist approach to peering.  I=20
> > thought this might be a useful way to have a productive=20
> discussion on=20
> > requirements.
>=20
> Thanks for the contribution.
>=20
> It's always helpful to have something concrete to talk about.
>=20
> Some comments:
>=20
> > 3.  Overview
> >=20
> >    The approach described here assumes that the initiating=20
> peer first
> >    discovers a SIP [2] target URI handled by the receiving=20
> peer, then
> >    sends SIP requests directly to the receiving peer.
>=20
> Perhaps you should also notice here that such ad-hoc peering=20
> implies settlement-free peering.
>=20
> >    Note that in a federation of domains, the initiating=20
> peer role can be
> >    logically decomposed.  Domain A1 can forward requests to=20
> domain A2,
> >    so that A2 actually opens a TLS connection to the receiving peer.
> >    Likewise, the receiving peer function can be similarly=20
> decomposed.
> >    How functions are distributed between or among domains=20
> or nodes is
> >    not subject to this specification as long as the requirements
> >    described in this document are met.
>=20
> We have to be careful here to keep the terminology consistent=20
> between the various documents.
>=20
> I used "peering" for calls within a federation, you seem to=20
> use "peering" only for inter-federation connects. The latter=20
> are not covered by draft-lendl-speermint-federations-00.
>=20
> A common understanding of this aspect will be important.
>=20
> > 4.  Procedures
>=20
> I like this section. It's very helpful for the discussion.
>=20
> > 4.1.2  User ENUM lookup
> >=20
> >    If an external E.164 address is the target, the initiating peer
> >    consults the public "User ENUM" rooted at e164.arpa,=20
> according to the
> >    procedures described in RFC 3761.  The peer MUST query=20
> for the "E2U+
> >    sip" enumservice as described in RFC 3674 [11], but MAY check for
> >    other enumservices.  The initiating peer MAY consult a cache or
> >    alternate representation of the ENUM data rather than actual DNS
> >    queries.  Also, the peer MAY skip actual DNS queries if the
> >    initiating peer is sure that the target address country=20
> code is not
> >    represented in e164.arpa.  If a sip: or sips: URI is=20
> chosen, the peer
> >    skips to Section 4.1.5.
>=20
> IMHO the URIs as returned by User-ENUM do not require TLS=20
> (except when they are sips: ones) as described in 4.1.6.
>=20
> They could point to end-user controlled devices (an Asterisk=20
> installation, other VoIP-enabled PBX's or simple SIP=20
> services).  I see no need to insist on anything more than=20
> pure RFC3263 for these cases.
>=20
> > 4.1.6  Setup TLS connection
> >=20
> >    Once a transport, port, and address are found, the=20
> initiating peer
> >    will open or find a reusable TLS connection to the peer.  The
> >    initiating provider MUST verify the server certificate=20
> which SHOULD
> >    be rooted in a well-known certificate authority. =20
>=20
> That is going to be contentious. What are the "well-known=20
> certificate authorities"? The ones which have their keys=20
> installed in MSIE and Firefox?
>=20
> Do you expect people to run white-lists based on the keys=20
> themselves or based on special-purpose CAs?
>=20
> Or blacklists to keep baddies out?
>=20
>  If you allow the same CAs as used for TLS certs for websites=20
> then this  is a huge community of potential peering partners.=20
> Not all of them  are well-meaning and serious operators.
>=20
> Do you expect to use CRLs?
>=20
> What if the destination SIP server hosts multiple domains?=20
> Which cert is he supposed to present to an incoming SIP/TLS=20
> connection?
>=20
> (This is the same problem as hosting multiple https: servers=20
> on a single server. Do you need one IP address per domain?)
>=20
> >    The initiating
> >    provider MUST be prepared to provide a TLS client=20
> certificate upon
> >    request during the TLS handshake.  The client certificate MUST
> >    contain a DNS or URI choice type in the subjectAltName which
> >    corresponds to the domain asserted in the host=20
> production of the From
> >    header URI.  The certificate SHOULD be valid and rooted=20
> in a well-
> >    known certificate authority.
>=20
> Same question again as above.
>=20
> What about TLS connection reuse? If both SIP servers serve=20
> URIs in multiple domains, can they reuse TLS connections=20
> which were established based on calls to a different domain=20
> (but hosted on the same server)?
>=20
> >    Note that the client certificate MAY contain a list of=20
> entries in the
> >    subjectAltName, only one of which has to match the=20
> domain in the From
> >    header URI.
>=20
> What are the scalability limits to that?
>=20
> Any moderately sized ISP mail server hosts mails for=20
> thousands of domains. Can this TLS setup scale to similar=20
> levels of domain hosting?
>=20
> > 4.2  Receiving Peer
> > 4.2.4  Receive SIP requests
>=20
> >    The receiving peer MAY reject incoming requests due to=20
> local policy.
> >    When a request is rejected because the initiating peer is not
> >    authorized to peer, the receiving peer SHOULD respond with a 403
> >    response with the reason phrase "Unsupported Peer".
>=20
> You have to distinguish rejects based on TLS negotiations and=20
> those on the SIP layer.
>=20
> What is the sending peer supposed to do in that case?
> Fall back to 4.1.4? Cache that result? If caching: per=20
> destination URI or per SIP proxy? For how long?
>=20
> How do you prevent unsuccessful TLS establishments to become=20
> a kind of DoS attack? Firewalling could be an option on the=20
> destination side, but dropped packets lead to timeouts on the=20
> sender side which lead to unpredictable call-establishment delays.
>=20
> --------
>=20
> As I wrote above: this draft is a good starting point for discussions.
>=20
> /ol
> --
> < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

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From speermint-bounces@ietf.org Mon Jun 19 13:45:28 2006
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From: "Brian Rosen" <br@brianrosen.net>
To: "'Michael Hammer \(mhammer\)'" <mhammer@cisco.com>,
	"'Otmar Lendl'" <lendl@nic.at>, <speermint@ietf.org>
Subject: RE: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
Date: Mon, 19 Jun 2006 13:45:13 -0400
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Does this strike a chord with others?

A federation is DEFINED, at least partially, by a CA handing out certs to
club members useful to restrict calls to members of the club?

I do think many federations will have more restrictive security policies
than simply possession of a valid federation cert, but this idea could be
one of those rare cases where a PKI can work.

Remember: this is TLS.  The NEXT HOP is the one that counts.  So transit
networks would work here.  A transit network offering transit from A to B
would have a cert in A and a cert in B.  Originator X, in federation A, can
check the transit network's A cert, and Y, the terminating network in B, can
check the transit network's B cert.

X -> ( A cloud ) -> Transit -> ( B cloud ) -> Y

It's simple enough.  If peering is by federation, it might work pretty well.

Brian

-----Original Message-----
From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com] 
Sent: Monday, June 19, 2006 11:44 AM
To: Otmar Lendl; speermint@ietf.org
Subject: RE: [Speermint] Just submitted
draft-mahy-speermint-direct-peering-00

To add to the discussion.

I think it might be useful that federations as part of their role in
peering to establish certificate authorities.

One aspect of the comments below that we have to keep very distinct is
that of authentication from authorization.  You could potentially know,
because a certificate is valid, exactly who the offending SP is.
Whether you then white list or black list is a separate authorization
issue.  In that case, if they subscribe to get your policy, you send
them to the ingress proxy at:  0.0.0.0  :)

Mike


> -----Original Message-----
> From: Otmar Lendl [mailto:lendl@nic.at] 
> Sent: Monday, June 19, 2006 11:12 AM
> To: speermint@ietf.org
> Subject: Re: [Speermint] Just submitted 
> draft-mahy-speermint-direct-peering-00
> 
> On 2006/06/19 10:06, Rohan Mahy <rohan.mahy@gmail.com> wrote:
> > Hi Folks,
> > 
> > I just submitted a draft with a minimalist approach to peering.  I 
> > thought this might be a useful way to have a productive 
> discussion on 
> > requirements.
> 
> Thanks for the contribution.
> 
> It's always helpful to have something concrete to talk about.
> 
> Some comments:
> 
> > 3.  Overview
> > 
> >    The approach described here assumes that the initiating 
> peer first
> >    discovers a SIP [2] target URI handled by the receiving 
> peer, then
> >    sends SIP requests directly to the receiving peer.
> 
> Perhaps you should also notice here that such ad-hoc peering 
> implies settlement-free peering.
> 
> >    Note that in a federation of domains, the initiating 
> peer role can be
> >    logically decomposed.  Domain A1 can forward requests to 
> domain A2,
> >    so that A2 actually opens a TLS connection to the receiving peer.
> >    Likewise, the receiving peer function can be similarly 
> decomposed.
> >    How functions are distributed between or among domains 
> or nodes is
> >    not subject to this specification as long as the requirements
> >    described in this document are met.
> 
> We have to be careful here to keep the terminology consistent 
> between the various documents.
> 
> I used "peering" for calls within a federation, you seem to 
> use "peering" only for inter-federation connects. The latter 
> are not covered by draft-lendl-speermint-federations-00.
> 
> A common understanding of this aspect will be important.
> 
> > 4.  Procedures
> 
> I like this section. It's very helpful for the discussion.
> 
> > 4.1.2  User ENUM lookup
> > 
> >    If an external E.164 address is the target, the initiating peer
> >    consults the public "User ENUM" rooted at e164.arpa, 
> according to the
> >    procedures described in RFC 3761.  The peer MUST query 
> for the "E2U+
> >    sip" enumservice as described in RFC 3674 [11], but MAY check for
> >    other enumservices.  The initiating peer MAY consult a cache or
> >    alternate representation of the ENUM data rather than actual DNS
> >    queries.  Also, the peer MAY skip actual DNS queries if the
> >    initiating peer is sure that the target address country 
> code is not
> >    represented in e164.arpa.  If a sip: or sips: URI is 
> chosen, the peer
> >    skips to Section 4.1.5.
> 
> IMHO the URIs as returned by User-ENUM do not require TLS 
> (except when they are sips: ones) as described in 4.1.6.
> 
> They could point to end-user controlled devices (an Asterisk 
> installation, other VoIP-enabled PBX's or simple SIP 
> services).  I see no need to insist on anything more than 
> pure RFC3263 for these cases.
> 
> > 4.1.6  Setup TLS connection
> > 
> >    Once a transport, port, and address are found, the 
> initiating peer
> >    will open or find a reusable TLS connection to the peer.  The
> >    initiating provider MUST verify the server certificate 
> which SHOULD
> >    be rooted in a well-known certificate authority.  
> 
> That is going to be contentious. What are the "well-known 
> certificate authorities"? The ones which have their keys 
> installed in MSIE and Firefox?
> 
> Do you expect people to run white-lists based on the keys 
> themselves or based on special-purpose CAs?
> 
> Or blacklists to keep baddies out?
> 
>  If you allow the same CAs as used for TLS certs for websites 
> then this  is a huge community of potential peering partners. 
> Not all of them  are well-meaning and serious operators.
> 
> Do you expect to use CRLs?
> 
> What if the destination SIP server hosts multiple domains? 
> Which cert is he supposed to present to an incoming SIP/TLS 
> connection?
> 
> (This is the same problem as hosting multiple https: servers 
> on a single server. Do you need one IP address per domain?)
> 
> >    The initiating
> >    provider MUST be prepared to provide a TLS client 
> certificate upon
> >    request during the TLS handshake.  The client certificate MUST
> >    contain a DNS or URI choice type in the subjectAltName which
> >    corresponds to the domain asserted in the host 
> production of the From
> >    header URI.  The certificate SHOULD be valid and rooted 
> in a well-
> >    known certificate authority.
> 
> Same question again as above.
> 
> What about TLS connection reuse? If both SIP servers serve 
> URIs in multiple domains, can they reuse TLS connections 
> which were established based on calls to a different domain 
> (but hosted on the same server)?
> 
> >    Note that the client certificate MAY contain a list of 
> entries in the
> >    subjectAltName, only one of which has to match the 
> domain in the From
> >    header URI.
> 
> What are the scalability limits to that?
> 
> Any moderately sized ISP mail server hosts mails for 
> thousands of domains. Can this TLS setup scale to similar 
> levels of domain hosting?
> 
> > 4.2  Receiving Peer
> > 4.2.4  Receive SIP requests
> 
> >    The receiving peer MAY reject incoming requests due to 
> local policy.
> >    When a request is rejected because the initiating peer is not
> >    authorized to peer, the receiving peer SHOULD respond with a 403
> >    response with the reason phrase "Unsupported Peer".
> 
> You have to distinguish rejects based on TLS negotiations and 
> those on the SIP layer.
> 
> What is the sending peer supposed to do in that case?
> Fall back to 4.1.4? Cache that result? If caching: per 
> destination URI or per SIP proxy? For how long?
> 
> How do you prevent unsuccessful TLS establishments to become 
> a kind of DoS attack? Firewalling could be an option on the 
> destination side, but dropped packets lead to timeouts on the 
> sender side which lead to unpredictable call-establishment delays.
> 
> --------
> 
> As I wrote above: this draft is a good starting point for discussions.
> 
> /ol
> --
> < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 

_______________________________________________
Speermint mailing list
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https://www1.ietf.org/mailman/listinfo/speermint


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From speermint-bounces@ietf.org Mon Jun 19 15:09:54 2006
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Date: Mon, 19 Jun 2006 21:09:45 +0200
From: Otmar Lendl <lendl@nic.at>
To: Brian Rosen <br@brianrosen.net>
Subject: Re: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
Message-ID: <20060619190945.GF2086@nic.at>
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On 2006/06/19 19:06, Brian Rosen <br@brianrosen.net> wrote:
> Does this strike a chord with others?

http://www.ietf.org/internet-drafts/draft-lendl-speermint-federations-00.txt
3., first example.

> A federation is DEFINED, at least partially, by a CA handing out certs to
> club members useful to restrict calls to members of the club?
> 
> I do think many federations will have more restrictive security policies
> than simply possession of a valid federation cert, but this idea could be
> one of those rare cases where a PKI can work.
> 
> Remember: this is TLS.  The NEXT HOP is the one that counts.  So transit
> networks would work here.  A transit network offering transit from A to B
> would have a cert in A and a cert in B.  Originator X, in federation A, can
> check the transit network's A cert, and Y, the terminating network in B, can
> check the transit network's B cert.
> 
> X -> ( A cloud ) -> Transit -> ( B cloud ) -> Y
> 
> It's simple enough.  If peering is by federation, it might work pretty well.

Yes.

My drafts aimed at getting the simple case within clouds/clubs (which
are called federations in my drafts) nailed down and then build some
sort of transit scheme on top of that. Some proposals on how to do this
transit routing are floating around lunch conversations and private
email and we hope to get some preliminary drafts ready for Montreal.

Please note that TLS is just one scheme of building such clubs
of SIP providers. You can do the same thing with IPsec, Firewalling
or VPN setups.

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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From speermint-bounces@ietf.org Mon Jun 19 15:44:52 2006
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Patrick,

I can see that there may be many reasons why the calling SP does not 
have, and cannot establish a peering relationship with the called SP, 
and would indeed need to use an intermediary.

But if SP-A lacks trust that SP-B will deliver the call to Bob, then 
adding an intermediary cannot fix that.

	Paul

Patrick Melampy wrote:
> Paul,
> 
> It's clear that most of us Engineers (including me) don't understand what's
> required to establish a business relationship for telephonic communications.
> I'm guessing there are issues with taxation, regulation, service level
> agreements, minimum usage, billing issues, currency issues, credit, and
> more. Although I've never seen a peering agreement for voice traffic, I'm
> guessing they are comprehensive. I personally am aware of IP Peering
> relationships for voice that even have damage clauses for periods of service
> interruption.
> 
> Maybe somebody who understands why these relationships are so difficult to
> establish and maintain could shed some light on this. If anybody could
> easily and quickly establish a new peering relationship, then your position
> is correct. Maybe you can help resolve all of the business issues to enable
> "easy and quick" peering with on contracts, no taxes or regulation, and no
> service level agreements.
> 
> I do believe that Adam is correct that trust will depend on the signaling
> infrastructure being open only to those networks that are trust worthy --
> with pre-established contractual guarantees. Any network that would break
> the SIPS chain of trust MUST be excluded, or we need to find a way to ensure
> that every party in the communication knows that the call is leaving the
> trusted world and entering into untrusted networks. As far as I know, there
> is no way currently to know that a SIP communication has followed a SIPS
> trusted set of proxies from end-to-end.
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Saturday, June 17, 2006 4:12 PM
> To: Livingood, Jason
> Cc: speermint@ietf.org; Uzelac, Adam
> Subject: Re: [Speermint] Updated Draft: SPEERMINT Peering Architecture
> 
>>> -----Original Message-----
>>> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com] 
> 
>>> A service provide would be best served to provide its 
>>> subscribers a way to contact anyone else.  The burden to 
>>> facilitate that session is on the SP.  Will a SP in Topeka 
>>> trust a SP in Oslo enough to establish a direct peering 
>>> arrangement?  Without that direct peer, how will the 
>>> subscribers of the Topeka SP access the folks in Oslo?  An 
>>> intermediary most likely will be involved with the value 
>>> being reachability and trust that should a call be sent to 
>>> the transit provider, it will be completed as intended by the 
>>> calling party. At the end of the day, it will be the one that 
>>> will complete the call that gets the buck.  Unfortunately 
>>> it's the PSTN that facilitates that Topeka to Oslo call today 
>>> more often than not.
> 
> Assuming the provider for the caller can determine the provider for the 
> callee, then I fail to see how introducing an intermediary can improve 
> trust. No matter how many intermediaries are involved, ultimately the 
> callee's provider will be involved in the call. Establishing a direct 
> peering relationship seems at least as good, if not better.
> 
> 	Paul
> 
> _______________________________________________
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> Speermint@ietf.org
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> 

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Hi Otmar,

Thanks for the review.  My comments are inline.

On 6/19/06, Otmar Lendl <lendl@nic.at> wrote:
> On 2006/06/19 10:06, Rohan Mahy <rohan.mahy@gmail.com> wrote:
> > Hi Folks,
> >
> > I just submitted a draft with a minimalist approach to peering.  I
> > thought this might be a useful way to have a productive discussion on
> > requirements.
>
> Thanks for the contribution.
>
> It's always helpful to have something concrete to talk about.
>
> Some comments:
>
> > 3.  Overview
> >
> >    The approach described here assumes that the initiating peer first
> >    discovers a SIP [2] target URI handled by the receiving peer, then
> >    sends SIP requests directly to the receiving peer.
>
> Perhaps you should also notice here that such ad-hoc peering implies
> settlement-free peering.

When the receiving peer is the carrier-of-record, settlement-free
certainly makes a lot of sense.  The approach does not prevent
out-of-band settlement, which still makes sense if the receiving peer
is providing PSTN termination.

> >    Note that in a federation of domains, the initiating peer role can be
> >    logically decomposed.  Domain A1 can forward requests to domain A2,
> >    so that A2 actually opens a TLS connection to the receiving peer.
> >    Likewise, the receiving peer function can be similarly decomposed.
> >    How functions are distributed between or among domains or nodes is
> >    not subject to this specification as long as the requirements
> >    described in this document are met.
>
> We have to be careful here to keep the terminology consistent between
> the various documents.
>
> I used "peering" for calls within a federation, you seem to use
> "peering" only for inter-federation connects. The latter are not
> covered by draft-lendl-speermint-federations-00.
>
> A common understanding of this aspect will be important.

Agreed we should get a consensus on what to call inter-federation and
intra-federation communications.

> > 4.  Procedures
>
> I like this section. It's very helpful for the discussion.
>
> > 4.1.2  User ENUM lookup
> >
> >    If an external E.164 address is the target, the initiating peer
> >    consults the public "User ENUM" rooted at e164.arpa, according to the
> >    procedures described in RFC 3761.  The peer MUST query for the "E2U+
> >    sip" enumservice as described in RFC 3674 [11], but MAY check for
> >    other enumservices.  The initiating peer MAY consult a cache or
> >    alternate representation of the ENUM data rather than actual DNS
> >    queries.  Also, the peer MAY skip actual DNS queries if the
> >    initiating peer is sure that the target address country code is not
> >    represented in e164.arpa.  If a sip: or sips: URI is chosen, the peer
> >    skips to Section 4.1.5.
>
> IMHO the URIs as returned by User-ENUM do not require TLS (except when
> they are sips: ones) as described in 4.1.6.
>
> They could point to end-user controlled devices (an Asterisk
> installation, other VoIP-enabled PBX's or simple SIP services).  I see
> no need to insist on anything more than pure RFC3263 for these cases.

good point.  I think this would be a good change.

> > 4.1.6  Setup TLS connection
> >
> >    Once a transport, port, and address are found, the initiating peer
> >    will open or find a reusable TLS connection to the peer.  The
> >    initiating provider MUST verify the server certificate which SHOULD
> >    be rooted in a well-known certificate authority.
>
> That is going to be contentious. What are the "well-known certificate
> authorities"? The ones which have their keys installed in MSIE and
> Firefox?
>
> Do you expect people to run white-lists based on the keys themselves
> or based on special-purpose CAs?
>
> Or blacklists to keep baddies out?
>
>  If you allow the same CAs as used for TLS certs for websites then this
>  is a huge community of potential peering partners. Not all of them
>  are well-meaning and serious operators.

I think well-known CAs (as used in web-browser for example) are
appropriate for authentication.  The reason I put a SHOULD above is to
allow certs signed by a club, or even self-signed certs if both peers
are OK with that. How the receiving peer decides to authorize the
initiating peer is subject to local policy.  Some peers could use an
explicit whitelist mechanism (similar to traditional telephone network
peering) and other peers could use a blacklist mechanism (similar to
email networks with spam filtering).  Either approach could use rating
and/or accreditation services to assist in the authorization decision
making.

> Do you expect to use CRLs?

I do not expect hosts to perform real-time CRL checks, but CRLs are
still potentially useful input to an authorization decision.

> What if the destination SIP server hosts multiple domains? Which
> cert is he supposed to present to an incoming SIP/TLS connection?

If the SIP server hosts multiple effectively equivalent domains (ex:
mozilla.org, mozilla.com, getfirefox.com), it could present a single
certificate with all of the domains in the subjectAltName (which is a
really a list of names).

For ordinary virtual hosting, the receiving peer needs to assign a
separate port for each virtual host, just like we do with HTTPS
virtual hosting, but because of SRV records, we can serve several
hosts on a single IP address.

Eventually we should migrate to using a TLS client extension (it
already exists), so the TLS client can state in the client hello which
domain its trying to contact.  The TLS server then responds with the
appropriate cert.

> (This is the same problem as hosting multiple https: servers on
> a single server. Do you need one IP address per domain?)
>
> >    The initiating
> >    provider MUST be prepared to provide a TLS client certificate upon
> >    request during the TLS handshake.  The client certificate MUST
> >    contain a DNS or URI choice type in the subjectAltName which
> >    corresponds to the domain asserted in the host production of the From
> >    header URI.  The certificate SHOULD be valid and rooted in a well-
> >    known certificate authority.
>
> Same question again as above.
>
> What about TLS connection reuse? If both SIP servers serve URIs
> in multiple domains, can they reuse TLS connections which were
> established based on calls to a different domain (but hosted
> on the same server)?

As long as the certificate used to open the connection still covers
the target domain.

> >    Note that the client certificate MAY contain a list of entries in the
> >    subjectAltName, only one of which has to match the domain in the From
> >    header URI.
>
> What are the scalability limits to that?

Currently getting a CA to sign the resulting certificate is the hard part. ;-)

> Any moderately sized ISP mail server hosts mails for thousands of
> domains. Can this TLS setup scale to similar levels of domain hosting?

If User ENUM does not need to use TLS, we only need to scale to the
number of domains one receiving peer needs to handle for its
carrier(s) of record.

> > 4.2  Receiving Peer
> > 4.2.4  Receive SIP requests
>
> >    The receiving peer MAY reject incoming requests due to local policy.
> >    When a request is rejected because the initiating peer is not
> >    authorized to peer, the receiving peer SHOULD respond with a 403
> >    response with the reason phrase "Unsupported Peer".
>
> You have to distinguish rejects based on TLS negotiations and those on
> the SIP layer.

good point.

> What is the sending peer supposed to do in that case?
> Fall back to 4.1.4? Cache that result? If caching: per destination
> URI or per SIP proxy? For how long?

I think the sending peer should continue trying other possibilities
until it can't any more.

I would like a message that conveys:  you are not authorized to peer
with this target domain name, so do not try again unless you have
administrative information suggesting that situation has changed.
This is different from the typical meaning of a 403 which means that
the *request* is forbidden.

> How do you prevent unsuccessful TLS establishments to become a kind of
> DoS attack? Firewalling could be an option on the destination side,
> but dropped packets lead to timeouts on the sender side which lead to
> unpredictable call-establishment delays.

there are already a variety of products that manage TLS establishment
DoS attacks for web commerce.

thanks,
-rohan

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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>
Subject: Re: [Speermint] use of RFC 3263 for locating peer's SIP server
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Hi Jean-Francois,

To use full NAPTR and SRV lookups in a peering context, we are only
really worried about a pair of SIP proxies (one for the initiating
peer and another for the receiving peer) implementing NAPTR.  Note
that most of these proxies will be consulting ENUM records and will
need to implement NAPTR anyway for ENUM.  I don't think RFC3263 is too
much to ask for.

thanks,
-rohan

On 6/19/06, Jean-Francois Mule <jf.mule@cablelabs.com> wrote:
> Hi,
>
>    We have discussed the use of the DNS procedures specified in RFC3263
> in speermint to resolve a SIP URI into a reachable host (IP address and
> port), and transport protocol.
>
>    To document the best current practices, could you contact me if you
> have any implementation or operational experience in field use of 3263
> mechanisms (full 3263 or partial NAPTR/SRV)?
>         - based on Robert Sparks' presentation in IETF#65 SIPPING on the
> status of the implementation in sip interoperability testing events:
>         40% of implementers showing up in SIPit do NAPTR,
>         50% do SRV.
>
>         - when implemented on the products, how many folks use those
> options? any feedback on the field use of 3263 for NAPTR lookup for SIP
> transport protocol selection?
> a) The RFC indicates that if the URI contains the transport protocol,
> then it should be used (no lookup): is this common? Should it be?
> b) In a few publicly-available product administration guides I have
> looked at, NAPTR lookup for transport selection can be turned on/off
> when it exists as anyone would expect. Do service providers use it? If
> no, what are the issues?
> ==> I'm told that some service providers mandate one transport protocol
> (say TCP) for SIP termination and therefore, the transport protocol to
> reach a peering partner is often statically configured along with the
> sip route. Is this common?
> ===> some folks believe it is costly to do on a per query basis. Is this
> a matter of recommending proper DNS caching and long enough TTL values
> for those things?
>
>         - what about SRV? Same questions.
>
> If I get enough feedback, my goal is to provide a brief summary on the
> above to the list, and possibly, draft one or two slides outline any
> issues at IETF#66.
>
> Thanks,
> Jean-Francois.
>
>
>
> _______________________________________________
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From speermint-bounces@ietf.org Mon Jun 19 19:15:52 2006
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Subject: RE: [speermint] minimum set of requirements
	forSIP-basedVoIPinterconnection and emergency calling
Date: Mon, 19 Jun 2006 19:15:16 -0400
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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
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Always interesting to see the WG pounce on a pretty innocent question.  =
;-)  To wit, should there be special emergency calling requirements for =
interdomain calls?  Nothing more and nothing less.  One answer is NO, =
all calls are treated the same.  The other answer is YES, we believe =
emergency calls should somehow be treated differently between two =
domains OR at least have minimially different requirements. =20

I believe there is a line between ECRIT and SPEERMINT on this as with =
ENUM and SPEERMINT.  We just want to know if there may be any additional =
requirements if we get a URI that clearly (in whatever manner) indicates =
an emergency call.  Not so controversial a question? ;-)

So perhaps now, we'll leave it as a placeholder question and Dave and I =
will chat up the ECRIT chairs in Montreal to get their thoughts.  We're =
clearly not at consensus yet I think. =20

Regards
Jason
=20

> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
> Sent: Friday, June 16, 2006 5:04 AM
> To: Stastny Richard
> Cc: speermint@ietf.org
> Subject: RE: [speermint] minimum set of requirements=20
> forSIP-basedVoIPinterconnection and emergency calling
>=20
> Stastny Richard wrote:
> > Sorry, but I doubt that emergency services is within the scope of=20
> > speermint
> >
> > For emergency services normally either the UA itself or the=20
> first hop=20
> > proxy is responsible, so it never reaches the peering point at all
>         ^^^^^
> There are examples of VSP providers that do offer sip-based=20
> VoIP interconnect services today including 9-1-1. So your=20
> assertion that emergency calls do not reach "peering points"=20
> between VSPs is simply not true (based on current practices e
> and this is by no means cable-specific).
>=20
> > For emergency services normally either the UA itself or the=20
> first hop=20
> > proxy is responsible
> I presume you mean responsible for "invoking the special=20
> routing procedures caused by an emergency dialed string or=20
> use of SOS URN".
> This is ok but it does not necessarily mean that, upon=20
> determination that the call is an emergency call, the=20
> originating VSP does have to handle the SIP call to the PSAP=20
> within its own domain.=20
> Some VSPs do have gateways connected to the PSTN and the=20
> whole selective routers thingy in the North American system=20
> for example to do this; some don't and hand off the call to=20
> another VSP. The question of who does the ECRIT mapping=20
> protocol resolution is also not relevant here.
>=20
> My first question was:
>    - should there be any minimal set of requirements (and=20
> ones applicable beyond national standards) for handling=20
> emergency calls that should be captured in a speermint=20
> requirement doc for SIP-based VoIP interconnect?
>=20
> Jean-Fran=E7ois
>=20
>=20
> _______________________________________________
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>=20

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Subject: RE: [Speermint] Just submitted draft-mahy-speermint-direct-peering-00
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> Some comments:
>=20
> > 3.  Overview
> >=20
> >    The approach described here assumes that the initiating=20
> peer first
> >    discovers a SIP [2] target URI handled by the receiving=20
> peer, then
> >    sends SIP requests directly to the receiving peer.
>=20
> Perhaps you should also notice here that such ad-hoc peering=20
> implies settlement-free peering.

I think it may be premature to speculate on settlement or cost models.
Let's focus on the underlying stuff first.

Jason

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Brian Rosen schrieb:

...
> I do think many federations will have more restrictive security policies
> than simply possession of a valid federation cert, but this idea could be
> one of those rare cases where a PKI can work.
this is why we worked on OpenSer to add support for multiple TLS 
connection, per-connection certificates and enable choice of TLS pipe 
based on the target federation as delivered by the domain policy 
matching method as per Otmar's I-D

for inquisitive minds:
see http://www.openser.org/docs/tls.html for the multiple TLS 
connection/cert support
see http://www.enum.at/index.php?id=dokumente (free signup required) for 
code and documentation on domain policy matching, and infrastructure 
ENUM resolver in OpenSer

-Michael
> It's simple enough.  If peering is by federation, it might work pretty well.
>
> Brian
>
> -----Original Message-----
> From: Michael Hammer (mhammer) [mailto:mhammer@cisco.com] 
> Sent: Monday, June 19, 2006 11:44 AM
> To: Otmar Lendl; speermint@ietf.org
> Subject: RE: [Speermint] Just submitted
> draft-mahy-speermint-direct-peering-00
>
> To add to the discussion.
>
> I think it might be useful that federations as part of their role in
> peering to establish certificate authorities.
>
> One aspect of the comments below that we have to keep very distinct is
> that of authentication from authorization.  You could potentially know,
> because a certificate is valid, exactly who the offending SP is.
> Whether you then white list or black list is a separate authorization
> issue.  In that case, if they subscribe to get your policy, you send
> them to the ingress proxy at:  0.0.0.0  :)
>
> Mike
>
>
>   
>> -----Original Message-----
>> From: Otmar Lendl [mailto:lendl@nic.at] 
>> Sent: Monday, June 19, 2006 11:12 AM
>> To: speermint@ietf.org
>> Subject: Re: [Speermint] Just submitted 
>> draft-mahy-speermint-direct-peering-00
>>
>> On 2006/06/19 10:06, Rohan Mahy <rohan.mahy@gmail.com> wrote:
>>     
>>> Hi Folks,
>>>
>>> I just submitted a draft with a minimalist approach to peering.  I 
>>> thought this might be a useful way to have a productive 
>>>       
>> discussion on 
>>     
>>> requirements.
>>>       
>> Thanks for the contribution.
>>
>> It's always helpful to have something concrete to talk about.
>>
>> Some comments:
>>
>>     
>>> 3.  Overview
>>>
>>>    The approach described here assumes that the initiating 
>>>       
>> peer first
>>     
>>>    discovers a SIP [2] target URI handled by the receiving 
>>>       
>> peer, then
>>     
>>>    sends SIP requests directly to the receiving peer.
>>>       
>> Perhaps you should also notice here that such ad-hoc peering 
>> implies settlement-free peering.
>>
>>     
>>>    Note that in a federation of domains, the initiating 
>>>       
>> peer role can be
>>     
>>>    logically decomposed.  Domain A1 can forward requests to 
>>>       
>> domain A2,
>>     
>>>    so that A2 actually opens a TLS connection to the receiving peer.
>>>    Likewise, the receiving peer function can be similarly 
>>>       
>> decomposed.
>>     
>>>    How functions are distributed between or among domains 
>>>       
>> or nodes is
>>     
>>>    not subject to this specification as long as the requirements
>>>    described in this document are met.
>>>       
>> We have to be careful here to keep the terminology consistent 
>> between the various documents.
>>
>> I used "peering" for calls within a federation, you seem to 
>> use "peering" only for inter-federation connects. The latter 
>> are not covered by draft-lendl-speermint-federations-00.
>>
>> A common understanding of this aspect will be important.
>>
>>     
>>> 4.  Procedures
>>>       
>> I like this section. It's very helpful for the discussion.
>>
>>     
>>> 4.1.2  User ENUM lookup
>>>
>>>    If an external E.164 address is the target, the initiating peer
>>>    consults the public "User ENUM" rooted at e164.arpa, 
>>>       
>> according to the
>>     
>>>    procedures described in RFC 3761.  The peer MUST query 
>>>       
>> for the "E2U+
>>     
>>>    sip" enumservice as described in RFC 3674 [11], but MAY check for
>>>    other enumservices.  The initiating peer MAY consult a cache or
>>>    alternate representation of the ENUM data rather than actual DNS
>>>    queries.  Also, the peer MAY skip actual DNS queries if the
>>>    initiating peer is sure that the target address country 
>>>       
>> code is not
>>     
>>>    represented in e164.arpa.  If a sip: or sips: URI is 
>>>       
>> chosen, the peer
>>     
>>>    skips to Section 4.1.5.
>>>       
>> IMHO the URIs as returned by User-ENUM do not require TLS 
>> (except when they are sips: ones) as described in 4.1.6.
>>
>> They could point to end-user controlled devices (an Asterisk 
>> installation, other VoIP-enabled PBX's or simple SIP 
>> services).  I see no need to insist on anything more than 
>> pure RFC3263 for these cases.
>>
>>     
>>> 4.1.6  Setup TLS connection
>>>
>>>    Once a transport, port, and address are found, the 
>>>       
>> initiating peer
>>     
>>>    will open or find a reusable TLS connection to the peer.  The
>>>    initiating provider MUST verify the server certificate 
>>>       
>> which SHOULD
>>     
>>>    be rooted in a well-known certificate authority.  
>>>       
>> That is going to be contentious. What are the "well-known 
>> certificate authorities"? The ones which have their keys 
>> installed in MSIE and Firefox?
>>
>> Do you expect people to run white-lists based on the keys 
>> themselves or based on special-purpose CAs?
>>
>> Or blacklists to keep baddies out?
>>
>>  If you allow the same CAs as used for TLS certs for websites 
>> then this  is a huge community of potential peering partners. 
>> Not all of them  are well-meaning and serious operators.
>>
>> Do you expect to use CRLs?
>>
>> What if the destination SIP server hosts multiple domains? 
>> Which cert is he supposed to present to an incoming SIP/TLS 
>> connection?
>>
>> (This is the same problem as hosting multiple https: servers 
>> on a single server. Do you need one IP address per domain?)
>>
>>     
>>>    The initiating
>>>    provider MUST be prepared to provide a TLS client 
>>>       
>> certificate upon
>>     
>>>    request during the TLS handshake.  The client certificate MUST
>>>    contain a DNS or URI choice type in the subjectAltName which
>>>    corresponds to the domain asserted in the host 
>>>       
>> production of the From
>>     
>>>    header URI.  The certificate SHOULD be valid and rooted 
>>>       
>> in a well-
>>     
>>>    known certificate authority.
>>>       
>> Same question again as above.
>>
>> What about TLS connection reuse? If both SIP servers serve 
>> URIs in multiple domains, can they reuse TLS connections 
>> which were established based on calls to a different domain 
>> (but hosted on the same server)?
>>
>>     
>>>    Note that the client certificate MAY contain a list of 
>>>       
>> entries in the
>>     
>>>    subjectAltName, only one of which has to match the 
>>>       
>> domain in the From
>>     
>>>    header URI.
>>>       
>> What are the scalability limits to that?
>>
>> Any moderately sized ISP mail server hosts mails for 
>> thousands of domains. Can this TLS setup scale to similar 
>> levels of domain hosting?
>>
>>     
>>> 4.2  Receiving Peer
>>> 4.2.4  Receive SIP requests
>>>       
>>>    The receiving peer MAY reject incoming requests due to 
>>>       
>> local policy.
>>     
>>>    When a request is rejected because the initiating peer is not
>>>    authorized to peer, the receiving peer SHOULD respond with a 403
>>>    response with the reason phrase "Unsupported Peer".
>>>       
>> You have to distinguish rejects based on TLS negotiations and 
>> those on the SIP layer.
>>
>> What is the sending peer supposed to do in that case?
>> Fall back to 4.1.4? Cache that result? If caching: per 
>> destination URI or per SIP proxy? For how long?
>>
>> How do you prevent unsuccessful TLS establishments to become 
>> a kind of DoS attack? Firewalling could be an option on the 
>> destination side, but dropped packets lead to timeouts on the 
>> sender side which lead to unpredictable call-establishment delays.
>>
>> --------
>>
>> As I wrote above: this draft is a good starting point for discussions.
>>
>> /ol
>> --
>> < Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>>
>>     
>
> _______________________________________________
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>   



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=20
The Version 3.0 of the Speermint Peering Architecture can be found at:
 http://www.ietf.org/internet-drafts/draft-khan-ip-serv-peer-arch-03.txt

Thank you.

Sohel Khan


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From: Avshalom Houri <AVSHALOM@il.ibm.com>
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Subject: [Speermint] Provisioning requirements
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http://www.ietf.org/internet-drafts/draft-houri-speermint-rtc-provisioning-reqs-00.txt

Real Time Communications (RTC) tools are becoming as prevalent and
essential for users on the Internet as email.  While RTC tools can,
like email, be implemented directly by users in a point-to-point
fashion, they are often provided for or on behalf of a community of
users within an administrative domain.  As the use of these tools
grows, users increasingly have the need to communicate with users not
only within their own community but with those in other communities
as well.  In practice, each community is controlled by some
authority, and so there is a need to provide for easier establishment
of connectivity between communities, and the management of the inter-
community link.  This document contains an initial list of
requirements for provisioning and managing connectivity between
communities.

--Avshalom


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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 PEERing for Multimedia INTerconnect Working Group of the IETF.

	Title		: SPEERMINT Requirements for SIP-based VoIP Interconnection
	Author(s)	: J. Mule
	Filename	: draft-ietf-speermint-requirements-00.txt
	Pages		: 10
	Date		: 2006-6-21
	
   This document describes general requirements for Session PEERing for
   Multimedia INTerconnect and defines the minimum set of requirements
   applicable to SIP session peering for VoIP interconnects.

   In its current form, the document is a first draft based on the
   SPEERMINT mailing list's discussions on requirements.  The main
   objectives are to generate consensus on what categories of
   requirements should be covered, and to start more discussions on the
   technical protocol requirements that apply to VoIP interconnects.


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From speermint-bounces@ietf.org Thu Jun 22 09:10:03 2006
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An initial draft of the speermint wg requirements is now available at:
http://www.ietf.org/internet-drafts/draft-ietf-speermint-requirements-00
.txt

Jean-Francois.
jfm@cablelabs.com


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----- Forwarded message from IETF Chair <chair@ietf.org> -----

> From: IETF Chair <chair@ietf.org>
> To: IETF Announcement list <ietf-announce@ietf.org>
> Cc:=20
> Subject: Montreal agenda still in flux=20
> Date: Thu, 22 Jun 2006 13:26:51 -0400
>=20
> Please be aware that there are still a few unresolved clashes,
> and the Secretariat is working on this, so (as always) please do
> not be surprised if there are some more agenda changes. Obviously=20
> we attempt to minimize late changes, but they are likely to=20
> continue for several more days.
>=20
> For the latest agenda, please check this URL:
>=20
> https://datatracker.ietf.org/public/meeting_agenda_html.cgi?meeting_num=
=3D66
>=20
> _______________________________________________
> IETF-Announce mailing list
> IETF-Announce@ietf.org
> https://www1.ietf.org/mailman/listinfo/ietf-announce

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Subject: Comments (RE: [speermint] requirements draft submitted)
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Hello Francois,

Comments on your draft

"2.3.  No blocked calls"

I think the title of this subsection should be changed to something else
since blocked calls (just like normal IP packets) can always happen due
to call limits, QoS, white/black lists, etc, etc

I think there is a distinction on what can be achieved in terms of
static vs. dynamic data available. The data is DNS is static and does
not reflect that state of the network. Actually since DNS is an out of
band mechanism, it does not even guarantee that your layer 3 peering
connections are up.

"Per-service provider opt-in
   without consultation of a centralized 'peering registry', but rather
   by publishing local configuration choices only is highly desirable."

Not sure I understand the connection between the first and second
phrases. Did you mean "Instead of a per-service...., providers should
publish"?=20

Or is the first option acceptable?  Personally I think that tackling the
(N^2) problem is very important but there will certainly be deployments
where providers prefer a centralized registry (intra federation calls
could be an example).


"2.5.  Independence of lower layers

   The system needs to be independent of details on what technologies
   are used route"

Which "system" do you refer here? I think independence is very important
but not sure it is totally achievable. I guess it depends how it is
worded. For example, see my comments to the no blocked calls above.

"2.7.  Minimal additional cost on call initiation

   Since each call setup implies execution of any proposed algorithm, it
   should incur minimal overhead and delay, and employ caching wherever
   possible to avoid extra protocol round trips."

We should mention that caching brings with it security problems. I can
foresee similar attacks like "SIP/URI" cache poisoning, etc

"2.8.  Look beyond SIP"

Good point. Although SMTP and SIP have different needs in terms of delay
and security.

"3.3.  Media-related requirements"

It seems to me media related requirements as you stated come into play
_if_ somebody is using a media relay. Otherwise the decision is between
the endpoints and we do not care/it is out of scope right?

For example,


"   o  the support of the VoIP metric block as defined in RTP Control
      Protocol Extended Reports [RFC3611] MAY be supported."

Between which entities? I'm assuming you mean between media relays since
if it is between the endpoints we really do not care/it is out of scope,
right?=20


I think media requirements strictly related to speermint could be QoS
across the fabric and discussed in the list and stated in the message
flows draft and maybe some requirements in terms of control vs data
routing.

"3.4.  Security requirements

   All SIP messages MUST be sent over TLS [RFC3546] to provide
   transport-layer security as defined in RFC 3261,"

Giving the fact that we can have security through physical segregation,
intra-federation/provider calls, etc, the "MUST" here seems too generic.

I think we should say something like "in the case of public on-demand
peering SIP messages...."


Good draft.

Thanks,

Reinaldo






> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Thursday, June 22, 2006 6:10 AM
> To: speermint@ietf.org
> Subject: [speermint] requirements draft submitted
>=20
>=20
> An initial draft of the speermint wg requirements is now available at:
>
http://www.ietf.org/internet-drafts/draft-ietf-speermint-requirements-00
> .txt
>=20
> Jean-Francois.
> jfm@cablelabs.com
>=20
>=20
> _______________________________________________
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> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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From speermint-bounces@ietf.org Fri Jun 23 09:56:27 2006
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From: Otmar Lendl <lendl@nic.at>
To: Reinaldo Penno <rpenno@juniper.net>
Subject: Re: Comments (RE: [speermint] requirements draft submitted)
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On 2006/06/22 20:06, Reinaldo Penno <rpenno@juniper.net> wrote:
> Hello Francois,
> 
> Comments on your draft

Some comments from me as well.

> 
> "2.3.  No blocked calls"
> 
> I think the title of this subsection should be changed to something else
> since blocked calls (just like normal IP packets) can always happen due
> to call limits, QoS, white/black lists, etc, etc

I agree with you on call limits and QoS as these are dynamic variables
which are hard to avoid. On the other hand white/black lists are
configuration issues and in that case I think we should try to avoid
sending INVITES to someone who per configuration does not want to talk
to us.

> I think there is a distinction on what can be achieved in terms of
> static vs. dynamic data available. The data is DNS is static and does
> not reflect that state of the network. Actually since DNS is an out of
> band mechanism, it does not even guarantee that your layer 3 peering
> connections are up.

Wait a second. We're talking about requirements here. The DNS may be 
part of the solution but need not be. The requirement just states
that you should not blindly try to send SIP invites without having
good reasons to think that the call will get through.

If you do your peering using special layer 3 connections, then
this requirement means that you should synchronize your SIP stack
with your layer 3 state. I.e.: don't bother generating SIP
packets if the IPSEC tunnels are down.

Or if you do some sort of SUBSCRIBE/NOTIFY to establish peerings, then
you shouldn't send INVITES unless the last notify you received 
gave you a green light.

> "Per-service provider opt-in
>    without consultation of a centralized 'peering registry', but rather
>    by publishing local configuration choices only is highly desirable."
> 
> Not sure I understand the connection between the first and second
> phrases. Did you mean "Instead of a per-service...., providers should
> publish"? 

Yes, that sentence is broken.

What about:

"A VSP opt-in by publishing local configuration choices is preferrable
to a centralized 'peering registry'."

> Or is the first option acceptable?  Personally I think that tackling the
> (N^2) problem is very important but there will certainly be deployments
> where providers prefer a centralized registry (intra federation calls
> could be an example).

Agreed, within federations such registries could be fine. I doubt that
such registries can scale to the inter-federation (that is: global
reachability) case.

Such a global registry might be possible on technical grounds, but
in terms of politics and economics this is not feasible. 

Who would every VSP trust to run that global registry?

> "2.5.  Independence of lower layers
> 
>    The system needs to be independent of details on what technologies
>    are used route"
> 
> Which "system" do you refer here? I think independence is very important
> but not sure it is totally achievable. I guess it depends how it is
> worded. For example, see my comments to the no blocked calls above.

"system" here stands for the world-wide peer discovery system.

The algorithm to find peers should be independent of how you
talk to that peer. 

> "2.7.  Minimal additional cost on call initiation
> 
>    Since each call setup implies execution of any proposed algorithm, it
>    should incur minimal overhead and delay, and employ caching wherever
>    possible to avoid extra protocol round trips."
> 
> We should mention that caching brings with it security problems. I can
> foresee similar attacks like "SIP/URI" cache poisoning, etc

What about some generic security requirement? (independently of caching)

> 
> "2.8.  Look beyond SIP"
> 
> Good point. Although SMTP and SIP have different needs in terms of delay
> and security.

Yes. If you look e.g. at MMS which are transmitted via SMTP you might
find similar requirements in terms of security.

[...]
> 
> "3.4.  Security requirements
> 
>    All SIP messages MUST be sent over TLS [RFC3546] to provide
>    transport-layer security as defined in RFC 3261,"
> 
> Giving the fact that we can have security through physical segregation,
> intra-federation/provider calls, etc, the "MUST" here seems too generic.
> 
> I think we should say something like "in the case of public on-demand
> peering SIP messages...."
> 

Agreed.

IMHO the Speermint work falls into two separate areas:

* Federation discovery & routing

  This needs to be a global standard.

* Intra-federation peering rules

  In theory each federation could write its own rules. 
  Speermint should provide a best practice document for that.

> Good draft.

yes.

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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From speermint-bounces@ietf.org Fri Jun 23 10:22:10 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
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Section 3.5 indicates:

   The minimum requirements to allow a successful VoIP interconnection
   include:

   o  the mandatory support of RTP *and* RTCP as defined in [RFC3550],

   o  the support of compatible codecs between communication peers, the
      G.711 MUST be supported, the IETF iLBC [RFC3951] codec and its RTP
      payload format [RFC3952] SHOULD be supported.

   o  the support of the VoIP metric block as defined in RTP Control
      Protocol Extended Reports [RFC3611] MAY be supported.

What exactly does this mean to the two bodies that are making the peering arrangement. Is this a requirement to control the codecs used by the UAs attached to the peering organisations equipment before peering can occur, or a requirement to support transcoding if these requirements are not met by the population of attached UAs.

Additional nit comment. RFC 2327 is obsolete and sdp-new is imminently about to be published and I assume we are going to reference that document (it will be RFC 4566).

regards

Keith

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: 21 June 2006 23:50
> To: i-d-announce@ietf.org
> Cc: speermint@ietf.org
> Subject: [Speermint] I-D 
> ACTION:draft-ietf-speermint-requirements-00.txt
> 
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> This draft is a work item of the Session PEERing for 
> Multimedia INTerconnect Working Group of the IETF.
> 
> 	Title		: SPEERMINT Requirements for SIP-based 
> VoIP Interconnection
> 	Author(s)	: J. Mule
> 	Filename	: draft-ietf-speermint-requirements-00.txt
> 	Pages		: 10
> 	Date		: 2006-6-21
> 	
>    This document describes general requirements for Session 
> PEERing for
>    Multimedia INTerconnect and defines the minimum set of requirements
>    applicable to SIP session peering for VoIP interconnects.
> 
>    In its current form, the document is a first draft based on the
>    SPEERMINT mailing list's discussions on requirements.  The main
>    objectives are to generate consensus on what categories of
>    requirements should be covered, and to start more 
> discussions on the
>    technical protocol requirements that apply to VoIP interconnects.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
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FYI,

an updated version of draft-lendl-speermint-federations has been
submitted to the RFC editors. This version adds a lot of explanatory
text and examples. I am glad that Mike Hammer and Michael Haberler
joined in as co-authors. Michael will do a presentation in Montreal.

The new version can be found at

http://www.enum.at/ietf/draft-lendl-speermint-federations-01.txt
and
http://www.enum.at/ietf/draft-lendl-speermint-federations-01.html

Comments are welcome!

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt 
Date: Fri, 23 Jun 2006 17:23:38 -0400
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Shouldn't the requirement be that the policy of a federation should
address the default codecs used by a federation to interconnect?

Mike


> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]=20
> Sent: Friday, June 23, 2006 10:22 AM
> To: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt=20
>=20
> Section 3.5 indicates:
>=20
>    The minimum requirements to allow a successful VoIP interconnection
>    include:
>=20
>    o  the mandatory support of RTP *and* RTCP as defined in [RFC3550],
>=20
>    o  the support of compatible codecs between communication=20
> peers, the
>       G.711 MUST be supported, the IETF iLBC [RFC3951] codec=20
> and its RTP
>       payload format [RFC3952] SHOULD be supported.
>=20
>    o  the support of the VoIP metric block as defined in RTP Control
>       Protocol Extended Reports [RFC3611] MAY be supported.
>=20
> What exactly does this mean to the two bodies that are making=20
> the peering arrangement. Is this a requirement to control the=20
> codecs used by the UAs attached to the peering organisations=20
> equipment before peering can occur, or a requirement to=20
> support transcoding if these requirements are not met by the=20
> population of attached UAs.
>=20
> Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> imminently about to be published and I assume we are going to=20
> reference that document (it will be RFC 4566).
>=20
> regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > Sent: 21 June 2006 23:50
> > To: i-d-announce@ietf.org
> > Cc: speermint@ietf.org
> > Subject: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.txt
> >=20
> >=20
> >=20
> > A New Internet-Draft is available from the on-line Internet-Drafts=20
> > directories.
> > This draft is a work item of the Session PEERing for Multimedia=20
> > INTerconnect Working Group of the IETF.
> >=20
> > 	Title		: SPEERMINT Requirements for SIP-based=20
> > VoIP Interconnection
> > 	Author(s)	: J. Mule
> > 	Filename	: draft-ietf-speermint-requirements-00.txt
> > 	Pages		: 10
> > 	Date		: 2006-6-21
> > =09
> >    This document describes general requirements for Session PEERing=20
> > for
> >    Multimedia INTerconnect and defines the minimum set of=20
> requirements
> >    applicable to SIP session peering for VoIP interconnects.
> >=20
> >    In its current form, the document is a first draft based on the
> >    SPEERMINT mailing list's discussions on requirements.  The main
> >    objectives are to generate consensus on what categories of
> >    requirements should be covered, and to start more discussions on=20
> > the
> >    technical protocol requirements that apply to VoIP interconnects.
> >=20
> >=20
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> rements-00.txt
>=20
> To remove yourself from the I-D Announcement list, send a=20
> message to i-d-announce-request@ietf.org with the word=20
> unsubscribe in the body of the message. =20
> You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce
> to change your subscription settings.
>=20
>=20
> Internet-Drafts are also available by anonymous FTP. Login=20
> with the username "anonymous" and a password of your e-mail=20
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> 	"get draft-ietf-speermint-requirements-00.txt".
>=20
> A list of Internet-Drafts directories can be found in=20
> http://www.ietf.org/shadow.html or=20
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>=20
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
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> 	mailserv@ietf.org.
> In the body type:
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> =09
> 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
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> Internet-Draft.
>=20
> _______________________________________________
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> https://www1.ietf.org/mailman/listinfo/speermint
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 >o  Should the minimum set of requirements for VoIP interconnect
>      include any media-related requirements at all?

 >  o  The speerming charter defines "VoIP" as in voice calls.  Does
 >     voice communication mean audio only or more? audio, DTMF tones,
  >    real-time fax, voiceband data?

IMHO not: Speermint is dealing with the interworking between proxies
media is basically an interworling between endpoints
=20
In addition, as said in the second bullet, why just defining the =
audio-codecs?
=20
=20
regards
-sta

=20

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From: "Rohan Mahy" <rohan.mahy@gmail.com>
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Mike,

There is no requirement for a federation to exist in speermint.  It
might or it might not, but the IETF won't be very effective trying to
put requirements on something which is effectively a business
arrangement.  We can define a set of protocols however, which needs to
be useful in the big-I Internet and hopefully for folks building
federations and other clubs as well.

Everyone,

I think codec selection should be out of scope.  Mandating G.711 will
have no positive effect now.  Everyone supports it today in some way
or another, but it could be detrimental in the future.  The point of
the SDP negotiation is to allow RTP *endpoints* to select the best
codec in common under the circumstances.  I don't think we can do any
better than that in a requirements document, especially since we are
describing an interface between SIP intermediaries that I hope will
not even be a participant in the RTP media most of the time.

thanks,
-rohan



On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> Shouldn't the requirement be that the policy of a federation should
> address the default codecs used by a federation to interconnect?
>
> Mike
>
>
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > Sent: Friday, June 23, 2006 10:22 AM
> > To: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> > Section 3.5 indicates:
> >
> >    The minimum requirements to allow a successful VoIP interconnection
> >    include:
> >
> >    o  the mandatory support of RTP *and* RTCP as defined in [RFC3550],
> >
> >    o  the support of compatible codecs between communication
> > peers, the
> >       G.711 MUST be supported, the IETF iLBC [RFC3951] codec
> > and its RTP
> >       payload format [RFC3952] SHOULD be supported.
> >
> >    o  the support of the VoIP metric block as defined in RTP Control
> >       Protocol Extended Reports [RFC3611] MAY be supported.
> >
> > What exactly does this mean to the two bodies that are making
> > the peering arrangement. Is this a requirement to control the
> > codecs used by the UAs attached to the peering organisations
> > equipment before peering can occur, or a requirement to
> > support transcoding if these requirements are not met by the
> > population of attached UAs.
> >
> > Additional nit comment. RFC 2327 is obsolete and sdp-new is
> > imminently about to be published and I assume we are going to
> > reference that document (it will be RFC 4566).
> >
> > regards
> >
> > Keith
> >
> > > -----Original Message-----
> > > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > > Sent: 21 June 2006 23:50
> > > To: i-d-announce@ietf.org
> > > Cc: speermint@ietf.org
> > > Subject: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.txt
> > >
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts
> > > directories.
> > > This draft is a work item of the Session PEERing for Multimedia
> > > INTerconnect Working Group of the IETF.
> > >
> > >     Title           : SPEERMINT Requirements for SIP-based
> > > VoIP Interconnection
> > >     Author(s)       : J. Mule
> > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > >     Pages           : 10
> > >     Date            : 2006-6-21
> > >
> > >    This document describes general requirements for Session PEERing
> > > for
> > >    Multimedia INTerconnect and defines the minimum set of
> > requirements
> > >    applicable to SIP session peering for VoIP interconnects.
> > >
> > >    In its current form, the document is a first draft based on the
> > >    SPEERMINT mailing list's discussions on requirements.  The main
> > >    objectives are to generate consensus on what categories of
> > >    requirements should be covered, and to start more discussions on
> > > the
> > >    technical protocol requirements that apply to VoIP interconnects.
> > >
> > >
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > rements-00.txt
> >
> > To remove yourself from the I-D Announcement list, send a
> > message to i-d-announce-request@ietf.org with the word
> > unsubscribe in the body of the message.
> > You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce
> > to change your subscription settings.
> >
> >
> > 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-speermint-requirements-00.txt".
> >
> > A list of Internet-Drafts directories can be found in
> > http://www.ietf.org/shadow.html or
> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> > Internet-Drafts can also be obtained by e-mail.
> >
> > Send a message to:
> >       mailserv@ietf.org.
> > In the body type:
> >       "FILE
> > /internet-drafts/draft-ietf-speermint-requirements-00.txt".
> >
> > NOTE: The mail server at ietf.org can return the document in
> >       MIME-encoded form by using the "mpack" utility.  To use this
> >       feature, insert the command "ENCODING mime" before the "FILE"
> >       command.  To decode the response(s), you will need "munpack" or
> >       a MIME-compliant mail reader.  Different MIME-compliant
> > mail readers
> >       exhibit different behavior, especially when dealing with
> >       "multipart" MIME messages (i.e. documents which have been split
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> >       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.
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>
> _______________________________________________
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
Date: Mon, 26 Jun 2006 14:19:44 -0400
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Rohan Mahy" <rohan.mahy@gmail.com>
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Rohan,

Agree with your codec point.  I was indirectly pushing that out as being
just another policy element.

You are missing what I am promoting.  I want "federation" to be a
protocol element, so you can map domains to federations.  Being able to
learn such associations in DNS will enable two parties to determine if
they "speak" the same federation dialect, i.e. can route SIP directly or
not.

Mike=20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
> Sent: Sunday, June 25, 2006 2:41 PM
> To: Michael Hammer (mhammer)
> Cc: Drage, Keith (Keith); speermint@ietf.org
> Subject: Re: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> Mike,
>=20
> There is no requirement for a federation to exist in=20
> speermint.  It might or it might not, but the IETF won't be=20
> very effective trying to put requirements on something which=20
> is effectively a business arrangement.  We can define a set=20
> of protocols however, which needs to be useful in the big-I=20
> Internet and hopefully for folks building federations and=20
> other clubs as well.
>=20
> Everyone,
>=20
> I think codec selection should be out of scope.  Mandating=20
> G.711 will have no positive effect now.  Everyone supports it=20
> today in some way or another, but it could be detrimental in=20
> the future.  The point of the SDP negotiation is to allow RTP=20
> *endpoints* to select the best codec in common under the=20
> circumstances.  I don't think we can do any better than that=20
> in a requirements document, especially since we are=20
> describing an interface between SIP intermediaries that I=20
> hope will not even be a participant in the RTP media most of the time.
>=20
> thanks,
> -rohan
>=20
>=20
>=20
> On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > Shouldn't the requirement be that the policy of a federation should=20
> > address the default codecs used by a federation to interconnect?
> >
> > Mike
> >
> >
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Friday, June 23, 2006 10:22 AM
> > > To: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > > Section 3.5 indicates:
> > >
> > >    The minimum requirements to allow a successful VoIP=20
> interconnection
> > >    include:
> > >
> > >    o  the mandatory support of RTP *and* RTCP as defined in=20
> > > [RFC3550],
> > >
> > >    o  the support of compatible codecs between=20
> communication peers,=20
> > > the
> > >       G.711 MUST be supported, the IETF iLBC [RFC3951]=20
> codec and its=20
> > > RTP
> > >       payload format [RFC3952] SHOULD be supported.
> > >
> > >    o  the support of the VoIP metric block as defined in=20
> RTP Control
> > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > >
> > > What exactly does this mean to the two bodies that are making the=20
> > > peering arrangement. Is this a requirement to control the codecs=20
> > > used by the UAs attached to the peering organisations equipment=20
> > > before peering can occur, or a requirement to support=20
> transcoding if=20
> > > these requirements are not met by the population of attached UAs.
> > >
> > > Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> > > imminently about to be published and I assume we are going to=20
> > > reference that document (it will be RFC 4566).
> > >
> > > regards
> > >
> > > Keith
> > >
> > > > -----Original Message-----
> > > > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > > > Sent: 21 June 2006 23:50
> > > > To: i-d-announce@ietf.org
> > > > Cc: speermint@ietf.org
> > > > Subject: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > >
> > > >
> > > >
> > > > A New Internet-Draft is available from the on-line=20
> Internet-Drafts=20
> > > > directories.
> > > > This draft is a work item of the Session PEERing for Multimedia=20
> > > > INTerconnect Working Group of the IETF.
> > > >
> > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > VoIP Interconnection
> > > >     Author(s)       : J. Mule
> > > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > > >     Pages           : 10
> > > >     Date            : 2006-6-21
> > > >
> > > >    This document describes general requirements for Session=20
> > > > PEERing for
> > > >    Multimedia INTerconnect and defines the minimum set of
> > > requirements
> > > >    applicable to SIP session peering for VoIP interconnects.
> > > >
> > > >    In its current form, the document is a first draft=20
> based on the
> > > >    SPEERMINT mailing list's discussions on=20
> requirements.  The main
> > > >    objectives are to generate consensus on what categories of
> > > >    requirements should be covered, and to start more=20
> discussions=20
> > > > on the
> > > >    technical protocol requirements that apply to VoIP=20
> interconnects.
> > > >
> > > >
> > > > A URL for this Internet-Draft is:
> > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > > rements-00.txt
> > >
> > > To remove yourself from the I-D Announcement list, send a=20
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> > >
> > > A list of Internet-Drafts directories can be found in=20
> > > http://www.ietf.org/shadow.html or=20
> > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > >
> > >
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> > >
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> > >       exhibit different behavior, especially when dealing with
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> > > _______________________________________________
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> > >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20

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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Rohan Mahy" <rohan.mahy@gmail.com>,
	"Michael Hammer (mhammer)" <mhammer@cisco.com>
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I don't see why a Federation is deemed exclusively a business
arrangement.  A Federation should be an optional parameter in the
protocol set that facilitates SIP peering.  IMO, the main driver is
scale. Achieving scale without some hierarchical tools and therefore
summation options makes for problematic reachability propagation.
Thinking of DNS -  Hierarchy in the domain names is NOT the reason for
DNS scalability - the namespace is essentially flat in practice.
Federations presents an another option to the direct P2P model.  The
attempt is not to ingrain an archaic business model that forces
transitory providers into the P2P world.  Federations can help with
reachability awareness through a further segregation of the data-set.
If every single endpoint is the data-set, and we aspire to have everyone
be able to directly communicate with everyone else (think pure P2P),
then WHOA, what a database I would have to connect-to in the sky -OR-
WHOA, what a database of reachabiliy I would have resident on my local
proxy! =20

Adam Uzelac

-----Original Message-----
From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
Sent: Sunday, June 25, 2006 2:41 PM
To: Michael Hammer (mhammer)
Cc: speermint@ietf.org
Subject: Re: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t xt

Mike,

There is no requirement for a federation to exist in speermint.  It
might or it might not, but the IETF won't be very effective trying to
put requirements on something which is effectively a business
arrangement.  We can define a set of protocols however, which needs to
be useful in the big-I Internet and hopefully for folks building
federations and other clubs as well.

Everyone,

I think codec selection should be out of scope.  Mandating G.711 will
have no positive effect now.  Everyone supports it today in some way or
another, but it could be detrimental in the future.  The point of the
SDP negotiation is to allow RTP *endpoints* to select the best codec in
common under the circumstances.  I don't think we can do any better than
that in a requirements document, especially since we are describing an
interface between SIP intermediaries that I hope will not even be a
participant in the RTP media most of the time.

thanks,
-rohan



On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> Shouldn't the requirement be that the policy of a federation should=20
> address the default codecs used by a federation to interconnect?
>
> Mike
>
>
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > Sent: Friday, June 23, 2006 10:22 AM
> > To: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> > Section 3.5 indicates:
> >
> >    The minimum requirements to allow a successful VoIP
interconnection
> >    include:
> >
> >    o  the mandatory support of RTP *and* RTCP as defined in=20
> > [RFC3550],
> >
> >    o  the support of compatible codecs between communication peers,=20
> > the
> >       G.711 MUST be supported, the IETF iLBC [RFC3951] codec and its

> > RTP
> >       payload format [RFC3952] SHOULD be supported.
> >
> >    o  the support of the VoIP metric block as defined in RTP Control
> >       Protocol Extended Reports [RFC3611] MAY be supported.
> >
> > What exactly does this mean to the two bodies that are making the=20
> > peering arrangement. Is this a requirement to control the codecs=20
> > used by the UAs attached to the peering organisations equipment=20
> > before peering can occur, or a requirement to support transcoding if

> > these requirements are not met by the population of attached UAs.
> >
> > Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> > imminently about to be published and I assume we are going to=20
> > reference that document (it will be RFC 4566).
> >
> > regards
> >
> > Keith
> >
> > > -----Original Message-----
> > > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > > Sent: 21 June 2006 23:50
> > > To: i-d-announce@ietf.org
> > > Cc: speermint@ietf.org
> > > Subject: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.txt
> > >
> > >
> > >
> > > A New Internet-Draft is available from the on-line Internet-Drafts

> > > directories.
> > > This draft is a work item of the Session PEERing for Multimedia=20
> > > INTerconnect Working Group of the IETF.
> > >
> > >     Title           : SPEERMINT Requirements for SIP-based
> > > VoIP Interconnection
> > >     Author(s)       : J. Mule
> > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > >     Pages           : 10
> > >     Date            : 2006-6-21
> > >
> > >    This document describes general requirements for Session=20
> > > PEERing for
> > >    Multimedia INTerconnect and defines the minimum set of
> > requirements
> > >    applicable to SIP session peering for VoIP interconnects.
> > >
> > >    In its current form, the document is a first draft based on the
> > >    SPEERMINT mailing list's discussions on requirements.  The main
> > >    objectives are to generate consensus on what categories of
> > >    requirements should be covered, and to start more discussions=20
> > > on the
> > >    technical protocol requirements that apply to VoIP
interconnects.
> > >
> > >
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > rements-00.txt
> >
> > To remove yourself from the I-D Announcement list, send a message to

> > i-d-announce-request@ietf.org with the word unsubscribe in the body=20
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> >
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> > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> > Internet-Drafts can also be obtained by e-mail.
> >
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> >       mailserv@ietf.org.
> > In the body type:
> >       "FILE
> > /internet-drafts/draft-ietf-speermint-requirements-00.txt".
> >
> > NOTE: The mail server at ietf.org can return the document in
> >       MIME-encoded form by using the "mpack" utility.  To use this
> >       feature, insert the command "ENCODING mime" before the "FILE"
> >       command.  To decode the response(s), you will need "munpack"
or
> >       a MIME-compliant mail reader.  Different MIME-compliant mail=20
> > 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=20
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> > Internet-Draft.
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>

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On 2006/06/23 17:06, Otmar Lendl <lendl@nic.at> wrote:
> 
> FYI,
> 
> an updated version of draft-lendl-speermint-federations has been
> submitted to the RFC editors. This version adds a lot of explanatory
> text and examples. I am glad that Mike Hammer and Michael Haberler
> joined in as co-authors. Michael will do a presentation in Montreal.
> 
> The new version can be found at

This draft is now online at the official repository:

http://www.ietf.org/internet-drafts/draft-lendl-speermint-federations-01.txt

A new version of the basic (SIP-independent) algorithm is available, too:

http://www.ietf.org/internet-drafts/draft-lendl-domain-policy-ddds-01.txt

/ol
-- 
< Otmar Lendl (lendl@nic.at) | nic.at Systems Engineer >

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FYI=20

The final agenda for the 66th IETF meeting can be found at
http://www.ietf.org/meetings/IETF-66.html

Not all requests for changes were able to be made due to additional
conflicts.

The WG will have a firm agenda completed in a couple of days.

Regards
Jason


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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
Date: Tue, 27 Jun 2006 11:23:36 -0400
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My initial thought is that you are accurate saying that federations are
NOT REQUIRED for speermint.  Federations MAY or MAY NOT be used, as you
noted, and to that extent will be considered in some fashion here.

As for codecs, I think we'll have to see based on further WG discussion.
I would surely agree that codec negotiation / selection is *well*
handled in the underlying SIP protocol itself.  However, we may well
find a need to specify (or not) a lowest common denominator codec but
we'll have to see how our requirements work progress.

Jason

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
> Sent: Sunday, June 25, 2006 2:41 PM
> To: Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> Mike,
>=20
> There is no requirement for a federation to exist in=20
> speermint.  It might or it might not, but the IETF won't be=20
> very effective trying to put requirements on something which=20
> is effectively a business arrangement.  We can define a set=20
> of protocols however, which needs to be useful in the big-I=20
> Internet and hopefully for folks building federations and=20
> other clubs as well.
>=20
> Everyone,
>=20
> I think codec selection should be out of scope.  Mandating=20
> G.711 will have no positive effect now.  Everyone supports it=20
> today in some way or another, but it could be detrimental in=20
> the future.  The point of the SDP negotiation is to allow RTP=20
> *endpoints* to select the best codec in common under the=20
> circumstances.  I don't think we can do any better than that=20
> in a requirements document, especially since we are=20
> describing an interface between SIP intermediaries that I=20
> hope will not even be a participant in the RTP media most of the time.
>=20
> thanks,
> -rohan
>=20
>=20
>=20
> On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > Shouldn't the requirement be that the policy of a federation should=20
> > address the default codecs used by a federation to interconnect?
> >
> > Mike
> >
> >
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Friday, June 23, 2006 10:22 AM
> > > To: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > > Section 3.5 indicates:
> > >
> > >    The minimum requirements to allow a successful VoIP=20
> interconnection
> > >    include:
> > >
> > >    o  the mandatory support of RTP *and* RTCP as defined in=20
> > > [RFC3550],
> > >
> > >    o  the support of compatible codecs between=20
> communication peers,=20
> > > the
> > >       G.711 MUST be supported, the IETF iLBC [RFC3951]=20
> codec and its=20
> > > RTP
> > >       payload format [RFC3952] SHOULD be supported.
> > >
> > >    o  the support of the VoIP metric block as defined in=20
> RTP Control
> > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > >
> > > What exactly does this mean to the two bodies that are making the=20
> > > peering arrangement. Is this a requirement to control the codecs=20
> > > used by the UAs attached to the peering organisations equipment=20
> > > before peering can occur, or a requirement to support=20
> transcoding if=20
> > > these requirements are not met by the population of attached UAs.
> > >
> > > Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> > > imminently about to be published and I assume we are going to=20
> > > reference that document (it will be RFC 4566).
> > >
> > > regards
> > >
> > > Keith
> > >
> > > > -----Original Message-----
> > > > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > > > Sent: 21 June 2006 23:50
> > > > To: i-d-announce@ietf.org
> > > > Cc: speermint@ietf.org
> > > > Subject: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > >
> > > >
> > > >
> > > > A New Internet-Draft is available from the on-line=20
> Internet-Drafts=20
> > > > directories.
> > > > This draft is a work item of the Session PEERing for Multimedia=20
> > > > INTerconnect Working Group of the IETF.
> > > >
> > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > VoIP Interconnection
> > > >     Author(s)       : J. Mule
> > > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > > >     Pages           : 10
> > > >     Date            : 2006-6-21
> > > >
> > > >    This document describes general requirements for Session=20
> > > > PEERing for
> > > >    Multimedia INTerconnect and defines the minimum set of
> > > requirements
> > > >    applicable to SIP session peering for VoIP interconnects.
> > > >
> > > >    In its current form, the document is a first draft=20
> based on the
> > > >    SPEERMINT mailing list's discussions on=20
> requirements.  The main
> > > >    objectives are to generate consensus on what categories of
> > > >    requirements should be covered, and to start more=20
> discussions=20
> > > > on the
> > > >    technical protocol requirements that apply to VoIP=20
> interconnects.
> > > >
> > > >
> > > > A URL for this Internet-Draft is:
> > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > > rements-00.txt
> > >
> > > To remove yourself from the I-D Announcement list, send a=20
> message to=20
> > > i-d-announce-request@ietf.org with the word unsubscribe=20
> in the body=20
> > > of the message.
> > > You can also visit=20
> > > https://www1.ietf.org/mailman/listinfo/I-D-announce
> > > to change your subscription settings.
> > >
> > >
> > > Internet-Drafts are also available by anonymous FTP.=20
> Login with the=20
> > > username "anonymous" and a password of your e-mail address. After=20
> > > logging in, type "cd internet-drafts" and then
> > >       "get draft-ietf-speermint-requirements-00.txt".
> > >
> > > A list of Internet-Drafts directories can be found in=20
> > > http://www.ietf.org/shadow.html or=20
> > > 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-speermint-requirements-00.txt".
> > >
> > > NOTE: The mail server at ietf.org can return the document in
> > >       MIME-encoded form by using the "mpack" utility.  To use this
> > >       feature, insert the command "ENCODING mime" before=20
> the "FILE"
> > >       command.  To decode the response(s), you will need=20
> "munpack" or
> > >       a MIME-compliant mail reader.  Different=20
> MIME-compliant mail=20
> > > 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.
> > >
> > >
> > > Below is the data which will enable a MIME compliant mail reader=20
> > > implementation to automatically retrieve the ASCII version of the=20
> > > Internet-Draft.
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

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We have a tentative agenda posted now.  It is still subject to change /
somewhat in flux.  We'll post to the list as it is updated.
=20
http://www3.ietf.org/proceedings/06jul/agenda/speermint.txt
=20
Session PEERing for Multimedia INTerconnect (speermint)

MONDAY, JULY 10, 2006, 1300 - 1500 (Afternoon Session I)=20
Room Name: 520DEF
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

CHAIRS:	    David Meyer      	<dmm@1-4-5.net>
	    Jason Livingood  	<jason_livingood@cable.comcast.com>

SECRETARY:  Alexander Mayrhofer <alexander.mayrhofer@nic.at>

AGENDA

 o Welcome & Introduction to speermint / Administrivia
5 minutes
     Jason Livingood

   - Mailing list: speermint-request@ietf.org
     subscribe

   - Scribe (jabber)
  =20
   - Blue Sheets

   - Review of Milestones

 o Agenda Bashing
5 minutes
     Dave Meyer

 o Review and Status of Open Work Items                     =20

   Primary / Official WG Drafts
   -----------------------------
    1 - SPEERMINT Terminology Draft
15 minutes
	draft-ietf-speermint-terminology-00
	  Dave Meyer

    2 - Minimum Set of Requirements for SIP-Based VoIP Interconnection
20 minutes
    	draft-ietf-speermint-requirements-00
	  Jean-Francois Mule

    New or Updated Individual Drafts
   ----------------------------------

    3 - SPEERMINT Routing Architecture Draft
15 minutes
	draft-khan-ip-serv-peer-arch-03
     	  Sohel Khan

    4 - Message Flows Associated with SPEERMINT Routing Architecture
15 minutes
    	draft-penno-message-flows-02=20
	  Reinaldo Penno

    5 - Session Peering Use Case for Cable
10 minutes
	draft-lee-speermint-use-case-cable-00
	  Yiu Lee

    6 - A Minimalist Approach to Direct Peering
10 minutes
	draft-mahy-speermint-direct-peering-00=20
	  Rohan Mahy

    7 - Real-Time Communications Provisioning Requirements
10 minutes
	draft-houri-speermint-rtc-provisioning-reqs-00
	  Avshalom Houri

    8 - A Federation based VoIP Peering Architecture
10 minutes
	draft-lendl-speermint-federations-01
	  Michael Haberler


 o Open Discussion
5 minutes
  =20



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<BODY>
<DIV><SPAN class=3D795472516-27062006><FONT face=3DArial size=3D2>We =
have a tentative=20
agenda posted now.&nbsp; It is still subject to change / somewhat in =
flux.&nbsp;=20
We'll post to the list as it is updated.</FONT></SPAN></DIV>
<DIV><SPAN class=3D795472516-27062006><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D795472516-27062006><FONT face=3DArial size=3D2><A=20
href=3D"http://www3.ietf.org/proceedings/06jul/agenda/speermint.txt">http=
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size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D795472516-27062006><PRE>Session PEERing for =
Multimedia INTerconnect (speermint)

MONDAY, JULY 10, 2006, 1300 - 1500 (Afternoon Session I)=20
Room Name: 520DEF
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

CHAIRS:	    David Meyer      	&lt;dmm@1-4-5.net&gt;
	    Jason Livingood  	&lt;jason_livingood@cable.comcast.com&gt;

SECRETARY:  Alexander Mayrhofer &lt;alexander.mayrhofer@nic.at&gt;

AGENDA

 o Welcome &amp; Introduction to speermint / Administrivia	  		5 minutes
     Jason Livingood

   - Mailing list: speermint-request@ietf.org
     subscribe

   - Scribe (jabber)
  =20
   - Blue Sheets

   - Review of Milestones

 o Agenda Bashing                                         		5 minutes
     Dave Meyer

 o Review and Status of Open Work Items                     =20

   Primary / Official WG Drafts
   -----------------------------
    1 - SPEERMINT Terminology Draft					15 minutes
	draft-ietf-speermint-terminology-00
	  Dave Meyer

    2 - Minimum Set of Requirements for SIP-Based VoIP Interconnection	=
20 minutes
    	draft-ietf-speermint-requirements-00
	  Jean-Francois Mule

    New or Updated Individual Drafts
   ----------------------------------

    3 - SPEERMINT Routing Architecture Draft				15 minutes
	draft-khan-ip-serv-peer-arch-03
     	  Sohel Khan

    4 - Message Flows Associated with SPEERMINT Routing Architecture	15 =
minutes
    	draft-penno-message-flows-02=20
	  Reinaldo Penno

    5 - Session Peering Use Case for Cable				10 minutes
	draft-lee-speermint-use-case-cable-00
	  Yiu Lee

    6 - A Minimalist Approach to Direct Peering				10 minutes
	draft-mahy-speermint-direct-peering-00=20
	  Rohan Mahy

    7 - Real-Time Communications Provisioning Requirements		10 minutes
	draft-houri-speermint-rtc-provisioning-reqs-00
	  Avshalom Houri

    8 - A Federation based VoIP Peering Architecture			10 minutes
	draft-lendl-speermint-federations-01
	  Michael Haberler


 o Open Discussion                                       		5 minutes
  =20

</PRE></SPAN></DIV></BODY></HTML>

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From speermint-bounces@ietf.org Tue Jun 27 13:31:01 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>, Rohan Mahy
	<rohan.mahy@gmail.com>, "Michael Hammer (mhammer)" <mhammer@cisco.com>
Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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But this still does not answer my original question.

How can you mandate a codec for what is essentially a network operator when that operator does not own the terminals (which contain the vast majority of the codecs). Are you making a requirement for transcoding support in the absence of whatever codec set is defined, or is it just an empty letter?

regards

Keith

> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> Sent: 27 June 2006 16:24
> To: Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D
> ACTION:draft-ietf-speermint-requirements-00.t xt
> 
> 
> My initial thought is that you are accurate saying that 
> federations are
> NOT REQUIRED for speermint.  Federations MAY or MAY NOT be 
> used, as you
> noted, and to that extent will be considered in some fashion here.
> 
> As for codecs, I think we'll have to see based on further WG 
> discussion.
> I would surely agree that codec negotiation / selection is *well*
> handled in the underlying SIP protocol itself.  However, we may well
> find a need to specify (or not) a lowest common denominator codec but
> we'll have to see how our requirements work progress.
> 
> Jason
> 
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan.mahy@gmail.com] 
> > Sent: Sunday, June 25, 2006 2:41 PM
> > To: Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: Re: [Speermint] I-D 
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> > 
> > Mike,
> > 
> > There is no requirement for a federation to exist in 
> > speermint.  It might or it might not, but the IETF won't be 
> > very effective trying to put requirements on something which 
> > is effectively a business arrangement.  We can define a set 
> > of protocols however, which needs to be useful in the big-I 
> > Internet and hopefully for folks building federations and 
> > other clubs as well.
> > 
> > Everyone,
> > 
> > I think codec selection should be out of scope.  Mandating 
> > G.711 will have no positive effect now.  Everyone supports it 
> > today in some way or another, but it could be detrimental in 
> > the future.  The point of the SDP negotiation is to allow RTP 
> > *endpoints* to select the best codec in common under the 
> > circumstances.  I don't think we can do any better than that 
> > in a requirements document, especially since we are 
> > describing an interface between SIP intermediaries that I 
> > hope will not even be a participant in the RTP media most 
> of the time.
> > 
> > thanks,
> > -rohan
> > 
> > 
> > 
> > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > Shouldn't the requirement be that the policy of a 
> federation should 
> > > address the default codecs used by a federation to interconnect?
> > >
> > > Mike
> > >
> > >
> > > > -----Original Message-----
> > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > To: speermint@ietf.org
> > > > Subject: RE: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > > Section 3.5 indicates:
> > > >
> > > >    The minimum requirements to allow a successful VoIP 
> > interconnection
> > > >    include:
> > > >
> > > >    o  the mandatory support of RTP *and* RTCP as defined in 
> > > > [RFC3550],
> > > >
> > > >    o  the support of compatible codecs between 
> > communication peers, 
> > > > the
> > > >       G.711 MUST be supported, the IETF iLBC [RFC3951] 
> > codec and its 
> > > > RTP
> > > >       payload format [RFC3952] SHOULD be supported.
> > > >
> > > >    o  the support of the VoIP metric block as defined in 
> > RTP Control
> > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > >
> > > > What exactly does this mean to the two bodies that are 
> making the 
> > > > peering arrangement. Is this a requirement to control 
> the codecs 
> > > > used by the UAs attached to the peering organisations equipment 
> > > > before peering can occur, or a requirement to support 
> > transcoding if 
> > > > these requirements are not met by the population of 
> attached UAs.
> > > >
> > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is 
> > > > imminently about to be published and I assume we are going to 
> > > > reference that document (it will be RFC 4566).
> > > >
> > > > regards
> > > >
> > > > Keith
> > > >
> > > > > -----Original Message-----
> > > > > From: Internet-Drafts@ietf.org 
> [mailto:Internet-Drafts@ietf.org]
> > > > > Sent: 21 June 2006 23:50
> > > > > To: i-d-announce@ietf.org
> > > > > Cc: speermint@ietf.org
> > > > > Subject: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > >
> > > > >
> > > > >
> > > > > A New Internet-Draft is available from the on-line 
> > Internet-Drafts 
> > > > > directories.
> > > > > This draft is a work item of the Session PEERing for 
> Multimedia 
> > > > > INTerconnect Working Group of the IETF.
> > > > >
> > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > VoIP Interconnection
> > > > >     Author(s)       : J. Mule
> > > > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > > > >     Pages           : 10
> > > > >     Date            : 2006-6-21
> > > > >
> > > > >    This document describes general requirements for Session 
> > > > > PEERing for
> > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > requirements
> > > > >    applicable to SIP session peering for VoIP interconnects.
> > > > >
> > > > >    In its current form, the document is a first draft 
> > based on the
> > > > >    SPEERMINT mailing list's discussions on 
> > requirements.  The main
> > > > >    objectives are to generate consensus on what categories of
> > > > >    requirements should be covered, and to start more 
> > discussions 
> > > > > on the
> > > > >    technical protocol requirements that apply to VoIP 
> > interconnects.
> > > > >
> > > > >
> > > > > A URL for this Internet-Draft is:
> > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > > > rements-00.txt
> > > >
> > > > To remove yourself from the I-D Announcement list, send a 
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> > > >
> > > >
> > > > Internet-Drafts can also be obtained by e-mail.
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> > 
> > _______________________________________________
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> > https://www1.ietf.org/mailman/listinfo/speermint
> > 
> 
> _______________________________________________
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From speermint-bounces@ietf.org Tue Jun 27 13:41:41 2006
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
Date: Tue, 27 Jun 2006 13:41:31 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
To: "Drage, Keith \(Keith\)" <drage@lucent.com>,
	"Livingood, Jason" <Jason_Livingood@cable.comcast.com>,
	"Rohan Mahy" <rohan.mahy@gmail.com>,
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Hello,

When I posted my comments I also thought that this part was weird. Maybe
Francois had something else in mind. I see codecs coming into play in a
media relay/transcoding scenario.=20

But even then, a media relay (or a pair of them) would negotiate the
necessary codecs as transparently as possible.=20

And in the case of SUBSCRIBE/NOTIFY policies and media relays, the
preferred codecs would be known in advance if transcoding was needed.=20

But I'm open to hear scenarios where mandating a codec would be
necessary.

Regards,

REinaldo

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: Tuesday, June 27, 2006 10:31 AM
> To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t
> xt
>=20
> But this still does not answer my original question.
>=20
> How can you mandate a codec for what is essentially a network operator
> when that operator does not own the terminals (which contain the vast
> majority of the codecs). Are you making a requirement for transcoding
> support in the absence of whatever codec set is defined, or is it just
an
> empty letter?
>=20
> regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> > Sent: 27 June 2006 16:24
> > To: Rohan Mahy; Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> >
> > My initial thought is that you are accurate saying that
> > federations are
> > NOT REQUIRED for speermint.  Federations MAY or MAY NOT be
> > used, as you
> > noted, and to that extent will be considered in some fashion here.
> >
> > As for codecs, I think we'll have to see based on further WG
> > discussion.
> > I would surely agree that codec negotiation / selection is *well*
> > handled in the underlying SIP protocol itself.  However, we may well
> > find a need to specify (or not) a lowest common denominator codec
but
> > we'll have to see how our requirements work progress.
> >
> > Jason
> >
> > > -----Original Message-----
> > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > Sent: Sunday, June 25, 2006 2:41 PM
> > > To: Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: Re: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > > Mike,
> > >
> > > There is no requirement for a federation to exist in
> > > speermint.  It might or it might not, but the IETF won't be
> > > very effective trying to put requirements on something which
> > > is effectively a business arrangement.  We can define a set
> > > of protocols however, which needs to be useful in the big-I
> > > Internet and hopefully for folks building federations and
> > > other clubs as well.
> > >
> > > Everyone,
> > >
> > > I think codec selection should be out of scope.  Mandating
> > > G.711 will have no positive effect now.  Everyone supports it
> > > today in some way or another, but it could be detrimental in
> > > the future.  The point of the SDP negotiation is to allow RTP
> > > *endpoints* to select the best codec in common under the
> > > circumstances.  I don't think we can do any better than that
> > > in a requirements document, especially since we are
> > > describing an interface between SIP intermediaries that I
> > > hope will not even be a participant in the RTP media most
> > of the time.
> > >
> > > thanks,
> > > -rohan
> > >
> > >
> > >
> > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > > Shouldn't the requirement be that the policy of a
> > federation should
> > > > address the default codecs used by a federation to interconnect?
> > > >
> > > > Mike
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > To: speermint@ietf.org
> > > > > Subject: RE: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Section 3.5 indicates:
> > > > >
> > > > >    The minimum requirements to allow a successful VoIP
> > > interconnection
> > > > >    include:
> > > > >
> > > > >    o  the mandatory support of RTP *and* RTCP as defined in
> > > > > [RFC3550],
> > > > >
> > > > >    o  the support of compatible codecs between
> > > communication peers,
> > > > > the
> > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > codec and its
> > > > > RTP
> > > > >       payload format [RFC3952] SHOULD be supported.
> > > > >
> > > > >    o  the support of the VoIP metric block as defined in
> > > RTP Control
> > > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > > >
> > > > > What exactly does this mean to the two bodies that are
> > making the
> > > > > peering arrangement. Is this a requirement to control
> > the codecs
> > > > > used by the UAs attached to the peering organisations
equipment
> > > > > before peering can occur, or a requirement to support
> > > transcoding if
> > > > > these requirements are not met by the population of
> > attached UAs.
> > > > >
> > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is
> > > > > imminently about to be published and I assume we are going to
> > > > > reference that document (it will be RFC 4566).
> > > > >
> > > > > regards
> > > > >
> > > > > Keith
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Internet-Drafts@ietf.org
> > [mailto:Internet-Drafts@ietf.org]
> > > > > > Sent: 21 June 2006 23:50
> > > > > > To: i-d-announce@ietf.org
> > > > > > Cc: speermint@ietf.org
> > > > > > Subject: [Speermint] I-D
> > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > >
> > > > > >
> > > > > >
> > > > > > A New Internet-Draft is available from the on-line
> > > Internet-Drafts
> > > > > > directories.
> > > > > > This draft is a work item of the Session PEERing for
> > Multimedia
> > > > > > INTerconnect Working Group of the IETF.
> > > > > >
> > > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > > VoIP Interconnection
> > > > > >     Author(s)       : J. Mule
> > > > > >     Filename        :
draft-ietf-speermint-requirements-00.txt
> > > > > >     Pages           : 10
> > > > > >     Date            : 2006-6-21
> > > > > >
> > > > > >    This document describes general requirements for Session
> > > > > > PEERing for
> > > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > > requirements
> > > > > >    applicable to SIP session peering for VoIP interconnects.
> > > > > >
> > > > > >    In its current form, the document is a first draft
> > > based on the
> > > > > >    SPEERMINT mailing list's discussions on
> > > requirements.  The main
> > > > > >    objectives are to generate consensus on what categories
of
> > > > > >    requirements should be covered, and to start more
> > > discussions
> > > > > > on the
> > > > > >    technical protocol requirements that apply to VoIP
> > > interconnects.
> > > > > >
> > > > > >
> > > > > > A URL for this Internet-Draft is:
> > > > > >
http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > > > > rements-00.txt
> > > > >
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> > > > >
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> > > > >
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> > > > >
> > > > > NOTE: The mail server at ietf.org can return the document in
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> > To use this
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> > version of the
> > > > > Internet-Draft.
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> > > > > _______________________________________________
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> > > > > Speermint@ietf.org
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> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
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> >
>=20
> _______________________________________________
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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From: "Michael Hammer \(mhammer\)" <mhammer@cisco.com>
To: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>,
	"Rohan Mahy" <rohan.mahy@gmail.com>
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IFF you are not interested in global rechability of users, then yes
small pockets of users can connect without federations, or whatever name
you want to give it.  However, if you want global scalability, something
other than a full mesh is needed, and that implies some degree of
hierarchy.  Call that grouping mechanism what you want, it is needed.

If the value of the PSTN is 4 billion squared, I would prefer the SIP
NGN network of the future to be at least as valuable.

Mike


> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]=20
> Sent: Tuesday, June 27, 2006 11:24 AM
> To: Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> My initial thought is that you are accurate saying that=20
> federations are NOT REQUIRED for speermint.  Federations MAY=20
> or MAY NOT be used, as you noted, and to that extent will be=20
> considered in some fashion here.
>=20
> As for codecs, I think we'll have to see based on further WG=20
> discussion.
> I would surely agree that codec negotiation / selection is=20
> *well* handled in the underlying SIP protocol itself. =20
> However, we may well find a need to specify (or not) a lowest=20
> common denominator codec but we'll have to see how our=20
> requirements work progress.
>=20
> Jason
>=20
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > Sent: Sunday, June 25, 2006 2:41 PM
> > To: Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: Re: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >=20
> > Mike,
> >=20
> > There is no requirement for a federation to exist in speermint.  It=20
> > might or it might not, but the IETF won't be very effective=20
> trying to=20
> > put requirements on something which is effectively a business=20
> > arrangement.  We can define a set of protocols however,=20
> which needs to=20
> > be useful in the big-I Internet and hopefully for folks building=20
> > federations and other clubs as well.
> >=20
> > Everyone,
> >=20
> > I think codec selection should be out of scope.  Mandating
> > G.711 will have no positive effect now.  Everyone supports=20
> it today in=20
> > some way or another, but it could be detrimental in the=20
> future.  The=20
> > point of the SDP negotiation is to allow RTP
> > *endpoints* to select the best codec in common under the=20
> > circumstances.  I don't think we can do any better than that in a=20
> > requirements document, especially since we are describing=20
> an interface=20
> > between SIP intermediaries that I hope will not even be a=20
> participant=20
> > in the RTP media most of the time.
> >=20
> > thanks,
> > -rohan
> >=20
> >=20
> >=20
> > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > Shouldn't the requirement be that the policy of a=20
> federation should=20
> > > address the default codecs used by a federation to interconnect?
> > >
> > > Mike
> > >
> > >
> > > > -----Original Message-----
> > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > To: speermint@ietf.org
> > > > Subject: RE: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > > Section 3.5 indicates:
> > > >
> > > >    The minimum requirements to allow a successful VoIP
> > interconnection
> > > >    include:
> > > >
> > > >    o  the mandatory support of RTP *and* RTCP as defined in=20
> > > > [RFC3550],
> > > >
> > > >    o  the support of compatible codecs between
> > communication peers,
> > > > the
> > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > codec and its
> > > > RTP
> > > >       payload format [RFC3952] SHOULD be supported.
> > > >
> > > >    o  the support of the VoIP metric block as defined in
> > RTP Control
> > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > >
> > > > What exactly does this mean to the two bodies that are=20
> making the=20
> > > > peering arrangement. Is this a requirement to control=20
> the codecs=20
> > > > used by the UAs attached to the peering organisations equipment=20
> > > > before peering can occur, or a requirement to support
> > transcoding if
> > > > these requirements are not met by the population of=20
> attached UAs.
> > > >
> > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> > > > imminently about to be published and I assume we are going to=20
> > > > reference that document (it will be RFC 4566).
> > > >
> > > > regards
> > > >
> > > > Keith
> > > >
> > > > > -----Original Message-----
> > > > > From: Internet-Drafts@ietf.org=20
> [mailto:Internet-Drafts@ietf.org]
> > > > > Sent: 21 June 2006 23:50
> > > > > To: i-d-announce@ietf.org
> > > > > Cc: speermint@ietf.org
> > > > > Subject: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > >
> > > > >
> > > > >
> > > > > A New Internet-Draft is available from the on-line
> > Internet-Drafts
> > > > > directories.
> > > > > This draft is a work item of the Session PEERing for=20
> Multimedia=20
> > > > > INTerconnect Working Group of the IETF.
> > > > >
> > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > VoIP Interconnection
> > > > >     Author(s)       : J. Mule
> > > > >     Filename        : draft-ietf-speermint-requirements-00.txt
> > > > >     Pages           : 10
> > > > >     Date            : 2006-6-21
> > > > >
> > > > >    This document describes general requirements for Session=20
> > > > > PEERing for
> > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > requirements
> > > > >    applicable to SIP session peering for VoIP interconnects.
> > > > >
> > > > >    In its current form, the document is a first draft
> > based on the
> > > > >    SPEERMINT mailing list's discussions on
> > requirements.  The main
> > > > >    objectives are to generate consensus on what categories of
> > > > >    requirements should be covered, and to start more
> > discussions
> > > > > on the
> > > > >    technical protocol requirements that apply to VoIP
> > interconnects.
> > > > >
> > > > >
> > > > > A URL for this Internet-Draft is:
> > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-requi
> > > > rements-00.txt
> > > >
> > > > To remove yourself from the I-D Announcement list, send a
> > message to
> > > > i-d-announce-request@ietf.org with the word unsubscribe
> > in the body
> > > > of the message.
> > > > You can also visit
> > > > https://www1.ietf.org/mailman/listinfo/I-D-announce
> > > > to change your subscription settings.
> > > >
> > > >
> > > > Internet-Drafts are also available by anonymous FTP.=20
> > Login with the
> > > > username "anonymous" and a password of your e-mail=20
> address. After=20
> > > > logging in, type "cd internet-drafts" and then
> > > >       "get draft-ietf-speermint-requirements-00.txt".
> > > >
> > > > A list of Internet-Drafts directories can be found in=20
> > > > http://www.ietf.org/shadow.html or=20
> > > > 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-speermint-requirements-00.txt".
> > > >
> > > > 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" 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=20
> mail reader=20
> > > > implementation to automatically retrieve the ASCII=20
> version of the=20
> > > > Internet-Draft.
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >=20
>=20

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Subject: RE: [Speermint] use of RFC 3263 for locating peer's SIP server
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Errors-To: speermint-bounces@ietf.org

Hi Rohan,

   Thank you for your input.=20

You wrote:
[snip]
> Note
> that most of these proxies will be consulting ENUM records and will
> need to implement NAPTR anyway for ENUM.  I don't think RFC3263 is too
> much to ask for.

I did not think full support for 3263 was too much to ask for "on the
paper". But my question was geared towards best current practices,
existing implementations (3263 has been lacking for transport protocol
selection) and initial operators deployments (not showing up in
numbers).

So far, zero positive response for full support from service providers
based on the question, one operator nack for 3263 NAPTR.

Jean-Francois.



> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: Monday, June 19, 2006 3:05 PM
> To: Jean-Francois Mule
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] use of RFC 3263 for locating peer's SIP
> server
>=20
> Hi Jean-Francois,
>=20
> To use full NAPTR and SRV lookups in a peering context, we are only
> really worried about a pair of SIP proxies (one for the initiating
> peer and another for the receiving peer) implementing NAPTR.  Note
> that most of these proxies will be consulting ENUM records and will
> need to implement NAPTR anyway for ENUM.  I don't think RFC3263 is too
> much to ask for.
>=20
> thanks,
> -rohan
>=20
> On 6/19/06, Jean-Francois Mule <jf.mule@cablelabs.com> wrote:
> > Hi,
> >
> >    We have discussed the use of the DNS procedures specified in
> RFC3263
> > in speermint to resolve a SIP URI into a reachable host (IP address
> and
> > port), and transport protocol.
> >
> >    To document the best current practices, could you contact me if
> you
> > have any implementation or operational experience in field use of
> 3263
> > mechanisms (full 3263 or partial NAPTR/SRV)?
> >         - based on Robert Sparks' presentation in IETF#65 SIPPING on
> the
> > status of the implementation in sip interoperability testing events:
> >         40% of implementers showing up in SIPit do NAPTR,
> >         50% do SRV.
> >
> >         - when implemented on the products, how many folks use those
> > options? any feedback on the field use of 3263 for NAPTR lookup for
> SIP
> > transport protocol selection?
> > a) The RFC indicates that if the URI contains the transport
protocol,
> > then it should be used (no lookup): is this common? Should it be?
> > b) In a few publicly-available product administration guides I have
> > looked at, NAPTR lookup for transport selection can be turned on/off
> > when it exists as anyone would expect. Do service providers use it?
> If
> > no, what are the issues?
> > =3D=3D> I'm told that some service providers mandate one transport
> protocol
> > (say TCP) for SIP termination and therefore, the transport protocol
> to
> > reach a peering partner is often statically configured along with
> the
> > sip route. Is this common?
> > =3D=3D=3D> some folks believe it is costly to do on a per query =
basis. Is
> this
> > a matter of recommending proper DNS caching and long enough TTL
> values
> > for those things?
> >
> >         - what about SRV? Same questions.
> >
> > If I get enough feedback, my goal is to provide a brief summary on
> the
> > above to the list, and possibly, draft one or two slides outline any
> > issues at IETF#66.
> >
> > Thanks,
> > Jean-Francois.
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >



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From speermint-bounces@ietf.org Tue Jun 27 16:05:34 2006
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Hi Reinaldo,

You wrote:
> Comments on your draft
>=20
> "2.3.  No blocked calls"
>=20
> I think the title of this subsection should be changed to something
> else
> since blocked calls (just like normal IP packets) can always happen
> due
> to call limits, QoS, white/black lists, etc, etc
I agree. How about URI reachability based on the current txt?

> I think there is a distinction on what can be achieved in terms of
> static vs. dynamic data available. The data is DNS is static and does
> not reflect that state of the network.
Not necessarily true.=20
While global DNS may have some lag, it is dynamic. It is part of common
DNS operations to actually reduce TTL in some DNS records just before
doing maintenance work on target servers to limit reachability issues
due to DNS caching, then do the maintenance and crank the TTLs back up
once the network is "more stable".

> Actually since DNS is an out of
> band mechanism, it does not even guarantee that your layer 3 peering
> connections are up.
>=20

> "Per-service provider opt-in
>    without consultation of a centralized 'peering registry', but
> rather
>    by publishing local configuration choices only is highly
> desirable."
> Not sure I understand the connection between the first and second
> phrases.=20
Me neither.

> Did you mean "Instead of a per-service...., providers should
> publish"?
>=20
> Or is the first option acceptable?  Personally I think that tackling
> the
> (N^2) problem is very important but there will certainly be
> deployments
> where providers prefer a centralized registry (intra federation calls
> could be an example).
This sentence in 2.4 is taken from the original draft and should be
clarified. I understood it as follows:
Publishing local configuration choices is highly desirable and
preferable over per-service provider opt-in without consultation of a
centralized 'peering registry'.

But even then, it is not clear to me what is meant... more work &
refinements are needed here.

> "2.5.  Independence of lower layers
>=20
>    The system needs to be independent of details on what technologies
>    are used route"
>=20
> Which "system" do you refer here? I think independence is very
> important
> but not sure it is totally achievable. I guess it depends how it is
> worded. For example, see my comments to the no blocked calls above.
Agreed.=20
Proposed new text:=20
s/system/session peering=20
   at least for now and until we have a function name from the
archictecture doc?
   Session peering SHOULD be independent of lower layers and the
technologies used to route the call...
=20

> "2.7.  Minimal additional cost on call initiation
>=20
>    Since each call setup implies execution of any proposed algorithm,
> it
>    should incur minimal overhead and delay, and employ caching
> wherever
>    possible to avoid extra protocol round trips."
>=20
> We should mention that caching brings with it security problems. I can
> foresee similar attacks like "SIP/URI" cache poisoning, etc
-> security consideration section: add txt on the implications of
caching?

> "3.3.  Media-related requirements"
>=20
> It seems to me media related requirements as you stated come into play
> _if_ somebody is using a media relay. Otherwise the decision is
> between
> the endpoints and we do not care/it is out of scope right?

It seems to me that VoIP interconnect is ultimately about exchanging rtp
packets (be it audio payloads or comfort noise).


>=20
> For example,
>=20
>=20
> "   o  the support of the VoIP metric block as defined in RTP Control
>       Protocol Extended Reports [RFC3611] MAY be supported."
>=20
> Between which entities? I'm assuming you mean between media relays
> since
> if it is between the endpoints we really do not care/it is out of
> scope,
> right?

This goes back to a problem in the current draft: requirements apply to
several types of entities.
=20
=20
> I think media requirements strictly related to speermint could be QoS
> across the fabric and discussed in the list and stated in the message
> flows draft and maybe some requirements in terms of control vs data
> routing.
>=20
> "3.4.  Security requirements
>=20
>    All SIP messages MUST be sent over TLS [RFC3546] to provide
>    transport-layer security as defined in RFC 3261,"
>=20
> Giving the fact that we can have security through physical
segregation,
> intra-federation/provider calls, etc, the "MUST" here seems too
> generic.

I tend to differ on this: it is not because you have lower layer
security (presumably hop by hop security associations) that you should
relax the end-to-end transport security requirements.=20
Rohan's well written draft is also in favor of mandatory support of TLS,
and so is the SIP Forum's IP PBX to VSP (here is a working link:
http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Inte
rop-sibley-final.pdf).
May be the requirement should be rewritten to put the MUST on SIP
entities involved in L5 peering.=20

How do others feel on this?

Thanks for your comments.
Jean-Francois.




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From speermint-bounces@ietf.org Tue Jun 27 16:16:36 2006
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Keith wrote:
> Section 3.5 indicates:
>=20
>    The minimum requirements to allow a successful VoIP interconnection
>    include:
>=20
>    o  the mandatory support of RTP *and* RTCP as defined in [RFC3550],
>=20
>    o  the support of compatible codecs between communication peers,
> the
>       G.711 MUST be supported, the IETF iLBC [RFC3951] codec and its
> RTP
>       payload format [RFC3952] SHOULD be supported.
>=20
>    o  the support of the VoIP metric block as defined in RTP Control
>       Protocol Extended Reports [RFC3611] MAY be supported.
>=20
> What exactly does this mean to the two bodies that are making the
> peering arrangement. Is this a requirement to control the codecs used
> by the UAs attached to the peering organisations equipment before
> peering can occur, or a requirement to support transcoding if these
> requirements are not met by the population of attached UAs.
Up for debate, but the later is more inline with a generic requirement
draft.=20


> Additional nit comment. RFC 2327 is obsolete and sdp-new is imminently
> about to be published and I assume we are going to reference that
> document (it will be RFC 4566).
Ok - will update the reference.

Thank you,
Jean-Francois.



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From speermint-bounces@ietf.org Tue Jun 27 16:24:23 2006
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Rohan wrote:
> Everyone,
>
> I think codec selection should be out of scope.  Mandating G.711 will
> have no positive effect now.  Everyone supports it today in some way
> or another, but it could be detrimental in the future.  The point of
> the SDP negotiation is to allow RTP *endpoints* to select the best
> codec in common under the circumstances.  I don't think we can do any
> better than that in a requirements document, especially since we are
> describing an interface between SIP intermediaries that I hope will
> not even be a participant in the RTP media most of the time.

This ID is for a set of *minimum* requirements for VoIP interconnect. I
have never walked in a meeting related to VoIP interop between 2 systems
or operators without having cleared the voice codec part (or assumed
G.711 is there).

I also note that it is in scope of the SIP Forum's use case for IP-PBX
to VSP VoIP interconnect.

Jean-Francois.


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From speermint-bounces@ietf.org Tue Jun 27 16:30:03 2006
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So is this kind of scenario in scope of speermint?
Take:
	- on one side, a 3GPP IMS network with the typical UE that does
not support G.711 on the terminal, and,
	- on the other, a VSP like free world dialup or alike that allow
any sip UA with any kind of codec.

Do we want to have some minimal requirements to ensure that negotiation
just does not fail?  I could care less where this transcoding function
is implemented.

Jean-Francois

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: Tuesday, June 27, 2006 11:31 AM
> To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-
> 00.t xt
>=20
> But this still does not answer my original question.
>=20
> How can you mandate a codec for what is essentially a network operator
> when that operator does not own the terminals (which contain the vast
> majority of the codecs). Are you making a requirement for transcoding
> support in the absence of whatever codec set is defined, or is it just
> an empty letter?
>=20
> regards
>=20
> Keith
>=20
> > -----Original Message-----
> > From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> > Sent: 27 June 2006 16:24
> > To: Rohan Mahy; Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> >
> > My initial thought is that you are accurate saying that
> > federations are
> > NOT REQUIRED for speermint.  Federations MAY or MAY NOT be
> > used, as you
> > noted, and to that extent will be considered in some fashion here.
> >
> > As for codecs, I think we'll have to see based on further WG
> > discussion.
> > I would surely agree that codec negotiation / selection is *well*
> > handled in the underlying SIP protocol itself.  However, we may well
> > find a need to specify (or not) a lowest common denominator codec
> but
> > we'll have to see how our requirements work progress.
> >
> > Jason
> >
> > > -----Original Message-----
> > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > Sent: Sunday, June 25, 2006 2:41 PM
> > > To: Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: Re: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > > Mike,
> > >
> > > There is no requirement for a federation to exist in
> > > speermint.  It might or it might not, but the IETF won't be
> > > very effective trying to put requirements on something which
> > > is effectively a business arrangement.  We can define a set
> > > of protocols however, which needs to be useful in the big-I
> > > Internet and hopefully for folks building federations and
> > > other clubs as well.
> > >
> > > Everyone,
> > >
> > > I think codec selection should be out of scope.  Mandating
> > > G.711 will have no positive effect now.  Everyone supports it
> > > today in some way or another, but it could be detrimental in
> > > the future.  The point of the SDP negotiation is to allow RTP
> > > *endpoints* to select the best codec in common under the
> > > circumstances.  I don't think we can do any better than that
> > > in a requirements document, especially since we are
> > > describing an interface between SIP intermediaries that I
> > > hope will not even be a participant in the RTP media most
> > of the time.
> > >
> > > thanks,
> > > -rohan
> > >
> > >
> > >
> > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > > Shouldn't the requirement be that the policy of a
> > federation should
> > > > address the default codecs used by a federation to interconnect?
> > > >
> > > > Mike
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > To: speermint@ietf.org
> > > > > Subject: RE: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Section 3.5 indicates:
> > > > >
> > > > >    The minimum requirements to allow a successful VoIP
> > > interconnection
> > > > >    include:
> > > > >
> > > > >    o  the mandatory support of RTP *and* RTCP as defined in
> > > > > [RFC3550],
> > > > >
> > > > >    o  the support of compatible codecs between
> > > communication peers,
> > > > > the
> > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > codec and its
> > > > > RTP
> > > > >       payload format [RFC3952] SHOULD be supported.
> > > > >
> > > > >    o  the support of the VoIP metric block as defined in
> > > RTP Control
> > > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > > >
> > > > > What exactly does this mean to the two bodies that are
> > making the
> > > > > peering arrangement. Is this a requirement to control
> > the codecs
> > > > > used by the UAs attached to the peering organisations
> equipment
> > > > > before peering can occur, or a requirement to support
> > > transcoding if
> > > > > these requirements are not met by the population of
> > attached UAs.
> > > > >
> > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is
> > > > > imminently about to be published and I assume we are going to
> > > > > reference that document (it will be RFC 4566).
> > > > >
> > > > > regards
> > > > >
> > > > > Keith
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Internet-Drafts@ietf.org
> > [mailto:Internet-Drafts@ietf.org]
> > > > > > Sent: 21 June 2006 23:50
> > > > > > To: i-d-announce@ietf.org
> > > > > > Cc: speermint@ietf.org
> > > > > > Subject: [Speermint] I-D
> > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > >
> > > > > >
> > > > > >
> > > > > > A New Internet-Draft is available from the on-line
> > > Internet-Drafts
> > > > > > directories.
> > > > > > This draft is a work item of the Session PEERing for
> > Multimedia
> > > > > > INTerconnect Working Group of the IETF.
> > > > > >
> > > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > > VoIP Interconnection
> > > > > >     Author(s)       : J. Mule
> > > > > >     Filename        : draft-ietf-speermint-requirements-
> 00.txt
> > > > > >     Pages           : 10
> > > > > >     Date            : 2006-6-21
> > > > > >
> > > > > >    This document describes general requirements for Session
> > > > > > PEERing for
> > > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > > requirements
> > > > > >    applicable to SIP session peering for VoIP interconnects.
> > > > > >
> > > > > >    In its current form, the document is a first draft
> > > based on the
> > > > > >    SPEERMINT mailing list's discussions on
> > > requirements.  The main
> > > > > >    objectives are to generate consensus on what categories
> of
> > > > > >    requirements should be covered, and to start more
> > > discussions
> > > > > > on the
> > > > > >    technical protocol requirements that apply to VoIP
> > > interconnects.
> > > > > >
> > > > > >
> > > > > > A URL for this Internet-Draft is:
> > > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-
> requi
> > > > > rements-00.txt
> > > > >
> > > > > To remove yourself from the I-D Announcement list, send a
> > > message to
> > > > > i-d-announce-request@ietf.org with the word unsubscribe
> > > in the body
> > > > > of the message.
> > > > > You can also visit
> > > > > https://www1.ietf.org/mailman/listinfo/I-D-announce
> > > > > to change your subscription settings.
> > > > >
> > > > >
> > > > > 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-speermint-requirements-00.txt".
> > > > >
> > > > > A list of Internet-Drafts directories can be found in
> > > > > http://www.ietf.org/shadow.html or
> > > > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > > >
> > > > >
> > > > > Internet-Drafts can also be obtained by e-mail.
> > > > >
> > > > > Send a message to:
> > > > >       mailserv@ietf.org.
> > > > > In the body type:
> > > > >       "FILE
> > > > > /internet-drafts/draft-ietf-speermint-requirements-00.txt".
> > > > >
> > > > > NOTE: The mail server at ietf.org can return the document in
> > > > >       MIME-encoded form by using the "mpack" utility.
> > To use this
> > > > >       feature, insert the command "ENCODING mime" before
> > > the "FILE"
> > > > >       command.  To decode the response(s), you will need
> > > "munpack" or
> > > > >       a MIME-compliant mail reader.  Different
> > > MIME-compliant mail
> > > > > readers
> > > > >       exhibit different behavior, especially when dealing with
> > > > >       "multipart" MIME messages (i.e. documents which
> > > have been split
> > > > >       up into multiple messages), so check your local
> > > documentation on
> > > > >       how to manipulate these messages.
> > > > >
> > > > >
> > > > > Below is the data which will enable a MIME compliant
> > mail reader
> > > > > implementation to automatically retrieve the ASCII
> > version of the
> > > > > Internet-Draft.
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint



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	"'Reinaldo Penno'" <rpenno@juniper.net>
Subject: RE: Comments (RE: [speermint] requirements draft submitted)
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> 
> "3.4.  Security requirements
> 
>    All SIP messages MUST be sent over TLS [RFC3546] to provide
>    transport-layer security as defined in RFC 3261,"
> 
> Giving the fact that we can have security through physical
segregation,
> intra-federation/provider calls, etc, the "MUST" here seems too
> generic.

I tend to differ on this: it is not because you have lower layer
security (presumably hop by hop security associations) that you should
relax the end-to-end transport security requirements. 
Rohan's well written draft is also in favor of mandatory support of TLS,
and so is the SIP Forum's IP PBX to VSP (here is a working link:
http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Inte
rop-sibley-final.pdf).
May be the requirement should be rewritten to put the MUST on SIP
entities involved in L5 peering. 

<brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.

Brian


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Subject: RE: Comments (RE: [speermint] requirements draft submitted)
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Just to set the record straight on this I-D:
> > Comments on your draft
This I-D is a wg draft so it is your draft too. I consider my role as an
editor on this one.

Regarding section 2.4 requirements, Omar wrote:
> What about:
>=20
> "A VSP opt-in by publishing local configuration choices is preferrable
> to a centralized 'peering registry'."
That is better, I could go with that.

> > "2.7.  Minimal additional cost on call initiation
> >
> >    Since each call setup implies execution of any proposed
algorithm,
> it
> >    should incur minimal overhead and delay, and employ caching
> wherever
> >    possible to avoid extra protocol round trips."
> >
> > We should mention that caching brings with it security problems. I
> can
> > foresee similar attacks like "SIP/URI" cache poisoning, etc
>=20
> What about some generic security requirement? (independently of
> caching)
Concrete proposal welcome for discussion as usual.


> > "3.4.  Security requirements
> >
> >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> >    transport-layer security as defined in RFC 3261,"
> >
> > Giving the fact that we can have security through physical
> segregation,
> > intra-federation/provider calls, etc, the "MUST" here seems too
> generic.
> >
> > I think we should say something like "in the case of public on-
> demand
> > peering SIP messages...."
> >
>=20
> Agreed.

Noted.

Jean-Francois.



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From: "Brian Rosen" <br@brianrosen.net>
To: "'Jean-Francois Mule'" <jf.mule@cablelabs.com>,
	"'Drage, Keith \(Keith\)'" <drage@lucent.com>,
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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Consider an all-IMS, all-GSM federation.  Do they have to have G.711?

Consider an all-video conference federation.  Do they have to have G.711?

Consider an all-IM federation.  Do they have to have G.711?


Brian

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com] 
Sent: Tuesday, June 27, 2006 4:30 PM
To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy; Michael Hammer
(mhammer)
Cc: speermint@ietf.org
Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t
xt

So is this kind of scenario in scope of speermint?
Take:
	- on one side, a 3GPP IMS network with the typical UE that does
not support G.711 on the terminal, and,
	- on the other, a VSP like free world dialup or alike that allow
any sip UA with any kind of codec.

Do we want to have some minimal requirements to ensure that negotiation
just does not fail?  I could care less where this transcoding function
is implemented.

Jean-Francois

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: Tuesday, June 27, 2006 11:31 AM
> To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-
> 00.t xt
> 
> But this still does not answer my original question.
> 
> How can you mandate a codec for what is essentially a network operator
> when that operator does not own the terminals (which contain the vast
> majority of the codecs). Are you making a requirement for transcoding
> support in the absence of whatever codec set is defined, or is it just
> an empty letter?
> 
> regards
> 
> Keith
> 
> > -----Original Message-----
> > From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> > Sent: 27 June 2006 16:24
> > To: Rohan Mahy; Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> >
> > My initial thought is that you are accurate saying that
> > federations are
> > NOT REQUIRED for speermint.  Federations MAY or MAY NOT be
> > used, as you
> > noted, and to that extent will be considered in some fashion here.
> >
> > As for codecs, I think we'll have to see based on further WG
> > discussion.
> > I would surely agree that codec negotiation / selection is *well*
> > handled in the underlying SIP protocol itself.  However, we may well
> > find a need to specify (or not) a lowest common denominator codec
> but
> > we'll have to see how our requirements work progress.
> >
> > Jason
> >
> > > -----Original Message-----
> > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > Sent: Sunday, June 25, 2006 2:41 PM
> > > To: Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: Re: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > > Mike,
> > >
> > > There is no requirement for a federation to exist in
> > > speermint.  It might or it might not, but the IETF won't be
> > > very effective trying to put requirements on something which
> > > is effectively a business arrangement.  We can define a set
> > > of protocols however, which needs to be useful in the big-I
> > > Internet and hopefully for folks building federations and
> > > other clubs as well.
> > >
> > > Everyone,
> > >
> > > I think codec selection should be out of scope.  Mandating
> > > G.711 will have no positive effect now.  Everyone supports it
> > > today in some way or another, but it could be detrimental in
> > > the future.  The point of the SDP negotiation is to allow RTP
> > > *endpoints* to select the best codec in common under the
> > > circumstances.  I don't think we can do any better than that
> > > in a requirements document, especially since we are
> > > describing an interface between SIP intermediaries that I
> > > hope will not even be a participant in the RTP media most
> > of the time.
> > >
> > > thanks,
> > > -rohan
> > >
> > >
> > >
> > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > > Shouldn't the requirement be that the policy of a
> > federation should
> > > > address the default codecs used by a federation to interconnect?
> > > >
> > > > Mike
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > To: speermint@ietf.org
> > > > > Subject: RE: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Section 3.5 indicates:
> > > > >
> > > > >    The minimum requirements to allow a successful VoIP
> > > interconnection
> > > > >    include:
> > > > >
> > > > >    o  the mandatory support of RTP *and* RTCP as defined in
> > > > > [RFC3550],
> > > > >
> > > > >    o  the support of compatible codecs between
> > > communication peers,
> > > > > the
> > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > codec and its
> > > > > RTP
> > > > >       payload format [RFC3952] SHOULD be supported.
> > > > >
> > > > >    o  the support of the VoIP metric block as defined in
> > > RTP Control
> > > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > > >
> > > > > What exactly does this mean to the two bodies that are
> > making the
> > > > > peering arrangement. Is this a requirement to control
> > the codecs
> > > > > used by the UAs attached to the peering organisations
> equipment
> > > > > before peering can occur, or a requirement to support
> > > transcoding if
> > > > > these requirements are not met by the population of
> > attached UAs.
> > > > >
> > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is
> > > > > imminently about to be published and I assume we are going to
> > > > > reference that document (it will be RFC 4566).
> > > > >
> > > > > regards
> > > > >
> > > > > Keith
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Internet-Drafts@ietf.org
> > [mailto:Internet-Drafts@ietf.org]
> > > > > > Sent: 21 June 2006 23:50
> > > > > > To: i-d-announce@ietf.org
> > > > > > Cc: speermint@ietf.org
> > > > > > Subject: [Speermint] I-D
> > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > >
> > > > > >
> > > > > >
> > > > > > A New Internet-Draft is available from the on-line
> > > Internet-Drafts
> > > > > > directories.
> > > > > > This draft is a work item of the Session PEERing for
> > Multimedia
> > > > > > INTerconnect Working Group of the IETF.
> > > > > >
> > > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > > VoIP Interconnection
> > > > > >     Author(s)       : J. Mule
> > > > > >     Filename        : draft-ietf-speermint-requirements-
> 00.txt
> > > > > >     Pages           : 10
> > > > > >     Date            : 2006-6-21
> > > > > >
> > > > > >    This document describes general requirements for Session
> > > > > > PEERing for
> > > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > > requirements
> > > > > >    applicable to SIP session peering for VoIP interconnects.
> > > > > >
> > > > > >    In its current form, the document is a first draft
> > > based on the
> > > > > >    SPEERMINT mailing list's discussions on
> > > requirements.  The main
> > > > > >    objectives are to generate consensus on what categories
> of
> > > > > >    requirements should be covered, and to start more
> > > discussions
> > > > > > on the
> > > > > >    technical protocol requirements that apply to VoIP
> > > interconnects.
> > > > > >
> > > > > >
> > > > > > A URL for this Internet-Draft is:
> > > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-
> requi
> > > > > rements-00.txt
> > > > >
> > > > > To remove yourself from the I-D Announcement list, send a
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> > > > > logging in, type "cd internet-drafts" and then
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> > > > >
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> > > > > http://www.ietf.org/shadow.html or
> > > > > ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > > >
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> > > > > /internet-drafts/draft-ietf-speermint-requirements-00.txt".
> > > > >
> > > > > NOTE: The mail server at ietf.org can return the document in
> > > > >       MIME-encoded form by using the "mpack" utility.
> > To use this
> > > > >       feature, insert the command "ENCODING mime" before
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> > > "munpack" or
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> > > > >
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From speermint-bounces@ietf.org Tue Jun 27 20:27:06 2006
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Subject: TLS for peering (was RE: Comments (RE: [speermint] requirements draft
	submitted))
Date: Tue, 27 Jun 2006 20:27:00 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
To: "Brian Rosen" <br@brianrosen.net>,
	"Jean-Francois Mule" <jf.mule@cablelabs.com>
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=20
TLS for _all_ scenarios? Even in private peerings, physically secure
peerings, etc?=20

Isn't that the same as requiring IPsec for all layer 3 peering? But
since Layer 5 peering might be more distributed, the number of TLS
sessions would be huge.

It would be good to hear the provider's opinion on this...

Thanks,

Reinaldo

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Tuesday, June 27, 2006 2:12 PM
> To: 'Jean-Francois Mule'; Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: RE: Comments (RE: [speermint] requirements draft submitted)
>=20
> >
> > "3.4.  Security requirements
> >
> >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> >    transport-layer security as defined in RFC 3261,"
> >
> > Giving the fact that we can have security through physical
> segregation,
> > intra-federation/provider calls, etc, the "MUST" here seems too
> > generic.
>=20
> I tend to differ on this: it is not because you have lower layer
> security (presumably hop by hop security associations) that you should
> relax the end-to-end transport security requirements.
> Rohan's well written draft is also in favor of mandatory support of
TLS,
> and so is the SIP Forum's IP PBX to VSP (here is a working link:
>
http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Inte
> rop-sibley-final.pdf).
> May be the requirement should be rewritten to put the MUST on SIP
> entities involved in L5 peering.
>=20
> <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
>=20
> Brian

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From speermint-bounces@ietf.org Tue Jun 27 20:44:36 2006
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Date: Tue, 27 Jun 2006 20:44:28 -0400
From: "Medhavi Bhatia" <mbhatia@3clogic.com>
To: "Reinaldo Penno" <rpenno@juniper.net>
Subject: Re: TLS for peering (was RE: Comments (RE: [speermint] requirements
	draft submitted))
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I am not a provider yet, but I think we dont need to mess with
TLS/TCP. As I understand one of the goals is to prevent the calls from
failing due to "peering only" reasons. This means that things like
codecs which are not negotiated by the peers need to be mandated at a
minimal level. TCP/TLS can be negotiated between the peers themselves
so it is a non-issue. Also as I still understand doing a high volume
calls with TLS is still expensive. Mandating TLS in cases where the
caller or callee dont require it doesnt seem to add any value.

-Medhavi.

On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
>
> TLS for _all_ scenarios? Even in private peerings, physically secure
> peerings, etc?
>
> Isn't that the same as requiring IPsec for all layer 3 peering? But
> since Layer 5 peering might be more distributed, the number of TLS
> sessions would be huge.
>
> It would be good to hear the provider's opinion on this...
>
> Thanks,
>
> Reinaldo
>
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 2:12 PM
> > To: 'Jean-Francois Mule'; Reinaldo Penno
> > Cc: speermint@ietf.org
> > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> >
> > >
> > > "3.4.  Security requirements
> > >
> > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > >    transport-layer security as defined in RFC 3261,"
> > >
> > > Giving the fact that we can have security through physical
> > segregation,
> > > intra-federation/provider calls, etc, the "MUST" here seems too
> > > generic.
> >
> > I tend to differ on this: it is not because you have lower layer
> > security (presumably hop by hop security associations) that you should
> > relax the end-to-end transport security requirements.
> > Rohan's well written draft is also in favor of mandatory support of
> TLS,
> > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> >
> http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Inte
> > rop-sibley-final.pdf).
> > May be the requirement should be rewritten to put the MUST on SIP
> > entities involved in L5 peering.
> >
> > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> >
> > Brian
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>

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From speermint-bounces@ietf.org Tue Jun 27 21:47:47 2006
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	requirementsdraft submitted))
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Medhavi Bhatia" <mbhatia@3clogic.com>,
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TLS is becoming more and more a subject when the topic of SIP
interconnections arises for me professionally.  This shows growing
momentum indeed.  With that said, I can not agree with TLS as a
steadfast requirement.  Economics and scale are my major concerns.  TLS
adds cost, server load, and there are fail-over issues to consider as
well.  For the person that designs an interconnect, I believe weighing
the cost against the benefit of the different models a useful exercise.
Mandating TLS, even if all parties involved in a peering arrangement
feel somehow "safe and secure" due to underlying security mechanisms
(L2/L3VPNs, etc), seems a little harsh. =20

Adam

-----Original Message-----
From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]=20
Sent: Tuesday, June 27, 2006 8:44 PM
To: Reinaldo Penno
Cc: speermint@ietf.org
Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
requirementsdraft submitted))

I am not a provider yet, but I think we dont need to mess with TLS/TCP.
As I understand one of the goals is to prevent the calls from failing
due to "peering only" reasons. This means that things like codecs which
are not negotiated by the peers need to be mandated at a minimal level.
TCP/TLS can be negotiated between the peers themselves so it is a
non-issue. Also as I still understand doing a high volume calls with TLS
is still expensive. Mandating TLS in cases where the caller or callee
dont require it doesnt seem to add any value.

-Medhavi.

On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
>
> TLS for _all_ scenarios? Even in private peerings, physically secure=20
> peerings, etc?
>
> Isn't that the same as requiring IPsec for all layer 3 peering? But=20
> since Layer 5 peering might be more distributed, the number of TLS=20
> sessions would be huge.
>
> It would be good to hear the provider's opinion on this...
>
> Thanks,
>
> Reinaldo
>
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 2:12 PM
> > To: 'Jean-Francois Mule'; Reinaldo Penno
> > Cc: speermint@ietf.org
> > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> >
> > >
> > > "3.4.  Security requirements
> > >
> > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > >    transport-layer security as defined in RFC 3261,"
> > >
> > > Giving the fact that we can have security through physical
> > segregation,
> > > intra-federation/provider calls, etc, the "MUST" here seems too=20
> > > generic.
> >
> > I tend to differ on this: it is not because you have lower layer=20
> > security (presumably hop by hop security associations) that you=20
> > should relax the end-to-end transport security requirements.
> > Rohan's well written draft is also in favor of mandatory support of
> TLS,
> > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> >
> http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> te
> > rop-sibley-final.pdf).
> > May be the requirement should be rewritten to put the MUST on SIP=20
> > entities involved in L5 peering.
> >
> > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> >
> > Brian
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>

_______________________________________________
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
To: "Brian Rosen" <br@brianrosen.net>,
	"Jean-Francois Mule" <jf.mule@cablelabs.com>,
	"Drage, Keith \(Keith\)" <drage@lucent.com>,
	"Rohan Mahy" <rohan.mahy@gmail.com>,
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Let's drop the IM and video examples -- for now.  Let's first focus on
getting the basic one working that we're committed to: voice. =20

So in the case of voice, the question at hand is - is there a lowest
common denominator voice codec which should be required to do peering?
If not, is there much utility in peering if certain provider types
cannot agree on a common codec?

Jason

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]=20
> Sent: Tuesday, June 27, 2006 5:40 PM
> To: 'Jean-Francois Mule'; 'Drage, Keith (Keith)'; Livingood,=20
> Jason; 'Rohan Mahy'; 'Michael Hammer (mhammer)'
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> Consider an all-IMS, all-GSM federation.  Do they have to have G.711?
>=20
> Consider an all-video conference federation.  Do they have to=20
> have G.711?
>=20
> Consider an all-IM federation.  Do they have to have G.711?
>=20
>=20
> Brian
>=20
> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Tuesday, June 27, 2006 4:30 PM
> To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy; Michael Hammer
> (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t
> xt
>=20
> So is this kind of scenario in scope of speermint?
> Take:
> 	- on one side, a 3GPP IMS network with the typical UE=20
> that does not support G.711 on the terminal, and,
> 	- on the other, a VSP like free world dialup or alike=20
> that allow any sip UA with any kind of codec.
>=20
> Do we want to have some minimal requirements to ensure that=20
> negotiation just does not fail?  I could care less where this=20
> transcoding function is implemented.
>=20
> Jean-Francois
>=20
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > Sent: Tuesday, June 27, 2006 11:31 AM
> > To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-
> > 00.t xt
> >=20
> > But this still does not answer my original question.
> >=20
> > How can you mandate a codec for what is essentially a=20
> network operator=20
> > when that operator does not own the terminals (which=20
> contain the vast=20
> > majority of the codecs). Are you making a requirement for=20
> transcoding=20
> > support in the absence of whatever codec set is defined, or=20
> is it just=20
> > an empty letter?
> >=20
> > regards
> >=20
> > Keith
> >=20
> > > -----Original Message-----
> > > From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> > > Sent: 27 June 2006 16:24
> > > To: Rohan Mahy; Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > >
> > > My initial thought is that you are accurate saying that=20
> federations=20
> > > are NOT REQUIRED for speermint.  Federations MAY or MAY=20
> NOT be used,=20
> > > as you noted, and to that extent will be considered in=20
> some fashion=20
> > > here.
> > >
> > > As for codecs, I think we'll have to see based on further WG=20
> > > discussion.
> > > I would surely agree that codec negotiation / selection is *well*=20
> > > handled in the underlying SIP protocol itself.  However,=20
> we may well=20
> > > find a need to specify (or not) a lowest common denominator codec
> > but
> > > we'll have to see how our requirements work progress.
> > >
> > > Jason
> > >
> > > > -----Original Message-----
> > > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > > Sent: Sunday, June 25, 2006 2:41 PM
> > > > To: Michael Hammer (mhammer)
> > > > Cc: speermint@ietf.org
> > > > Subject: Re: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > > Mike,
> > > >
> > > > There is no requirement for a federation to exist in=20
> speermint. =20
> > > > It might or it might not, but the IETF won't be very effective=20
> > > > trying to put requirements on something which is effectively a=20
> > > > business arrangement.  We can define a set of protocols=20
> however,=20
> > > > which needs to be useful in the big-I Internet and=20
> hopefully for=20
> > > > folks building federations and other clubs as well.
> > > >
> > > > Everyone,
> > > >
> > > > I think codec selection should be out of scope.  Mandating
> > > > G.711 will have no positive effect now.  Everyone supports it=20
> > > > today in some way or another, but it could be=20
> detrimental in the=20
> > > > future.  The point of the SDP negotiation is to allow RTP
> > > > *endpoints* to select the best codec in common under the=20
> > > > circumstances.  I don't think we can do any better than=20
> that in a=20
> > > > requirements document, especially since we are describing an=20
> > > > interface between SIP intermediaries that I hope will=20
> not even be=20
> > > > a participant in the RTP media most
> > > of the time.
> > > >
> > > > thanks,
> > > > -rohan
> > > >
> > > >
> > > >
> > > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > > > Shouldn't the requirement be that the policy of a
> > > federation should
> > > > > address the default codecs used by a federation to=20
> interconnect?
> > > > >
> > > > > Mike
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > > To: speermint@ietf.org
> > > > > > Subject: RE: [Speermint] I-D
> > > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > > >
> > > > > > Section 3.5 indicates:
> > > > > >
> > > > > >    The minimum requirements to allow a successful VoIP
> > > > interconnection
> > > > > >    include:
> > > > > >
> > > > > >    o  the mandatory support of RTP *and* RTCP as defined in=20
> > > > > > [RFC3550],
> > > > > >
> > > > > >    o  the support of compatible codecs between
> > > > communication peers,
> > > > > > the
> > > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > > codec and its
> > > > > > RTP
> > > > > >       payload format [RFC3952] SHOULD be supported.
> > > > > >
> > > > > >    o  the support of the VoIP metric block as defined in
> > > > RTP Control
> > > > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > > > >
> > > > > > What exactly does this mean to the two bodies that are
> > > making the
> > > > > > peering arrangement. Is this a requirement to control
> > > the codecs
> > > > > > used by the UAs attached to the peering organisations
> > equipment
> > > > > > before peering can occur, or a requirement to support
> > > > transcoding if
> > > > > > these requirements are not met by the population of
> > > attached UAs.
> > > > > >
> > > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is=20
> > > > > > imminently about to be published and I assume we=20
> are going to=20
> > > > > > reference that document (it will be RFC 4566).
> > > > > >
> > > > > > regards
> > > > > >
> > > > > > Keith
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Internet-Drafts@ietf.org
> > > [mailto:Internet-Drafts@ietf.org]
> > > > > > > Sent: 21 June 2006 23:50
> > > > > > > To: i-d-announce@ietf.org
> > > > > > > Cc: speermint@ietf.org
> > > > > > > Subject: [Speermint] I-D
> > > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > A New Internet-Draft is available from the on-line
> > > > Internet-Drafts
> > > > > > > directories.
> > > > > > > This draft is a work item of the Session PEERing for
> > > Multimedia
> > > > > > > INTerconnect Working Group of the IETF.
> > > > > > >
> > > > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > > > VoIP Interconnection
> > > > > > >     Author(s)       : J. Mule
> > > > > > >     Filename        : draft-ietf-speermint-requirements-
> > 00.txt
> > > > > > >     Pages           : 10
> > > > > > >     Date            : 2006-6-21
> > > > > > >
> > > > > > >    This document describes general requirements=20
> for Session=20
> > > > > > > PEERing for
> > > > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > > > requirements
> > > > > > >    applicable to SIP session peering for VoIP=20
> interconnects.
> > > > > > >
> > > > > > >    In its current form, the document is a first draft
> > > > based on the
> > > > > > >    SPEERMINT mailing list's discussions on
> > > > requirements.  The main
> > > > > > >    objectives are to generate consensus on what categories
> > of
> > > > > > >    requirements should be covered, and to start more
> > > > discussions
> > > > > > > on the
> > > > > > >    technical protocol requirements that apply to VoIP
> > > > interconnects.
> > > > > > >
> > > > > > >
> > > > > > > A URL for this Internet-Draft is:
> > > > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-
> > requi
> > > > > > rements-00.txt
> > > > > >
> > > > > > To remove yourself from the I-D Announcement list, send a
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> > > > > > Internet-Drafts are also available by anonymous FTP.
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> > > > > >
> > > > > > A list of Internet-Drafts directories can be found in=20
> > > > > > http://www.ietf.org/shadow.html or=20
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> > > > > >
> > > > > >
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> > > > > >
> > > > > > 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
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> > > > > >
> > > > > > Below is the data which will enable a MIME compliant
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> > > > > > implementation to automatically retrieve the ASCII
> > > version of the
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> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20

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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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Jason wrote:
>=20
> So in the case of voice, the question at hand is - is there a lowest
> common denominator voice codec which should be required to do peering?
> If not, is there much utility in peering if certain provider types
> cannot agree on a common codec?
>=20
The codecs are a terminal or end-user device issue. Peering between
service=20
providers have no influence on these. So if there is no match in
available=20
codecs, the only way out is to insert a transcoding function on one or
on both sides.

Since I am not so in protocol details, so I have a question: is there a
call
flow available to do so?

And if, the codec selected is part of the peering agreement itself.
Which=20
codec is up to the parties.

Richard
>=20

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Folks:

The bottom line of the support of codec is this: Support of QOS for the
VoIP application.

For example, one can do transcoding, but will they meet QOS
requirements? 

It is not simply the bandwidth. It is also about end-to-end delay,
jitter, and packet loss at the VoIP presentation layer (beyond IP
network layer).

Best regards,
Radhika R. Roy

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Stastny Richard
Sent: Wednesday, June 28, 2006 7:06 AM
To: Livingood, Jason
Cc: speermint@ietf.org
Subject: RE: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t xt

Jason wrote:
> 
> So in the case of voice, the question at hand is - is there a lowest
> common denominator voice codec which should be required to do peering?
> If not, is there much utility in peering if certain provider types
> cannot agree on a common codec?
> 
The codecs are a terminal or end-user device issue. Peering between
service 
providers have no influence on these. So if there is no match in
available 
codecs, the only way out is to insert a transcoding function on one or
on both sides.

Since I am not so in protocol details, so I have a question: is there a
call
flow available to do so?

And if, the codec selected is part of the peering agreement itself.
Which 
codec is up to the parties.

Richard
> 

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Codec can be implemented in the peering interface.
It can be internal to a provider network.
=20

-----Original Message-----
From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]=20
Sent: Wednesday, June 28, 2006 6:06 AM
To: Livingood, Jason
Cc: speermint@ietf.org
Subject: RE: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t xt

Jason wrote:
>=20
> So in the case of voice, the question at hand is - is there a lowest=20
> common denominator voice codec which should be required to do peering?
> If not, is there much utility in peering if certain provider types=20
> cannot agree on a common codec?
>=20
The codecs are a terminal or end-user device issue. Peering between
service providers have no influence on these. So if there is no match in
available codecs, the only way out is to insert a transcoding function
on one or on both sides.

Since I am not so in protocol details, so I have a question: is there a
call flow available to do so?

And if, the codec selected is part of the peering agreement itself.
Which
codec is up to the parties.

Richard
>=20

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



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From speermint-bounces@ietf.org Wed Jun 28 10:10:52 2006
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> Codec can be implemented in the peering interface.
On L5?

Richard

> -----Original Message-----
> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> Sent: Wednesday, June 28, 2006 3:47 PM
> To: Stastny Richard; Livingood, Jason
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t
> xt
>=20
> Codec can be implemented in the peering interface.
> It can be internal to a provider network.
>=20
>=20
> -----Original Message-----
> From: Stastny Richard [mailto:Richard.Stastny@oefeg.at]
> Sent: Wednesday, June 28, 2006 6:06 AM
> To: Livingood, Jason
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> Jason wrote:
> >
> > So in the case of voice, the question at hand is - is there a lowest
> > common denominator voice codec which should be required to do
peering?
> > If not, is there much utility in peering if certain provider types
> > cannot agree on a common codec?
> >
> The codecs are a terminal or end-user device issue. Peering between
> service providers have no influence on these. So if there is no match
in
> available codecs, the only way out is to insert a transcoding function
> on one or on both sides.
>=20
> Since I am not so in protocol details, so I have a question: is there
a
> call flow available to do so?
>=20
> And if, the codec selected is part of the peering agreement itself.
> Which
> codec is up to the parties.
>=20
> Richard
> >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20


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From speermint-bounces@ietf.org Wed Jun 28 10:44:14 2006
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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If there are 3 carriers...A, B, and C

A may only support AMR (wireless network)
B may support AMR and G.711 (wireless interconnect network)
C may only support G.711 (terminating PSNT network)

So now we have a problem...

A-----B  (negotiates to AMR) WORKS

B-----C  (negotiates to G.711) WORKS

But A----via B-----C  FAILS [has no agreeable CODEC's]

Thus B finds that on a call by call basis, if they support A and C, they may
not be able to provide universal connectivity from A to C and vica-versa.

I don't see how this problem can be avoided when there are 3 or more
carriers interconnected. I would suggest that this problem is outside the
scope of spearmint, and just a general problem.

I don't think we should require a codec (this doesn't always make sense),
nor do I think we should require transcoding which is offensive in its own
rite. I just think this is part of the internal problem B has with the
services B wishes to offer. B may have to provide either segmented service
offerings, and or transcoding internal to its network to support A and C
peering relationships.

Said another way, I do not believe incompatible CODEC's is a service
provider peering issue, but rather a general technology issue that exists
with or without service provider borders.


-----Original Message-----
From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com] 
Sent: Wednesday, June 28, 2006 6:46 AM
To: Brian Rosen; Jean-Francois Mule; Drage, Keith (Keith); Rohan Mahy;
Michael Hammer (mhammer)
Cc: speermint@ietf.org
Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t
xt

Let's drop the IM and video examples -- for now.  Let's first focus on
getting the basic one working that we're committed to: voice.  

So in the case of voice, the question at hand is - is there a lowest
common denominator voice codec which should be required to do peering?
If not, is there much utility in peering if certain provider types
cannot agree on a common codec?

Jason

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net] 
> Sent: Tuesday, June 27, 2006 5:40 PM
> To: 'Jean-Francois Mule'; 'Drage, Keith (Keith)'; Livingood, 
> Jason; 'Rohan Mahy'; 'Michael Hammer (mhammer)'
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D 
> ACTION:draft-ietf-speermint-requirements-00.t xt
> 
> Consider an all-IMS, all-GSM federation.  Do they have to have G.711?
> 
> Consider an all-video conference federation.  Do they have to 
> have G.711?
> 
> Consider an all-IM federation.  Do they have to have G.711?
> 
> 
> Brian
> 
> -----Original Message-----
> From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> Sent: Tuesday, June 27, 2006 4:30 PM
> To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy; Michael Hammer
> (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D 
> ACTION:draft-ietf-speermint-requirements-00.t
> xt
> 
> So is this kind of scenario in scope of speermint?
> Take:
> 	- on one side, a 3GPP IMS network with the typical UE 
> that does not support G.711 on the terminal, and,
> 	- on the other, a VSP like free world dialup or alike 
> that allow any sip UA with any kind of codec.
> 
> Do we want to have some minimal requirements to ensure that 
> negotiation just does not fail?  I could care less where this 
> transcoding function is implemented.
> 
> Jean-Francois
> 
> > -----Original Message-----
> > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > Sent: Tuesday, June 27, 2006 11:31 AM
> > To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D 
> ACTION:draft-ietf-speermint-requirements-
> > 00.t xt
> > 
> > But this still does not answer my original question.
> > 
> > How can you mandate a codec for what is essentially a 
> network operator 
> > when that operator does not own the terminals (which 
> contain the vast 
> > majority of the codecs). Are you making a requirement for 
> transcoding 
> > support in the absence of whatever codec set is defined, or 
> is it just 
> > an empty letter?
> > 
> > regards
> > 
> > Keith
> > 
> > > -----Original Message-----
> > > From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> > > Sent: 27 June 2006 16:24
> > > To: Rohan Mahy; Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > >
> > >
> > > My initial thought is that you are accurate saying that 
> federations 
> > > are NOT REQUIRED for speermint.  Federations MAY or MAY 
> NOT be used, 
> > > as you noted, and to that extent will be considered in 
> some fashion 
> > > here.
> > >
> > > As for codecs, I think we'll have to see based on further WG 
> > > discussion.
> > > I would surely agree that codec negotiation / selection is *well* 
> > > handled in the underlying SIP protocol itself.  However, 
> we may well 
> > > find a need to specify (or not) a lowest common denominator codec
> > but
> > > we'll have to see how our requirements work progress.
> > >
> > > Jason
> > >
> > > > -----Original Message-----
> > > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > > Sent: Sunday, June 25, 2006 2:41 PM
> > > > To: Michael Hammer (mhammer)
> > > > Cc: speermint@ietf.org
> > > > Subject: Re: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > > Mike,
> > > >
> > > > There is no requirement for a federation to exist in 
> speermint.  
> > > > It might or it might not, but the IETF won't be very effective 
> > > > trying to put requirements on something which is effectively a 
> > > > business arrangement.  We can define a set of protocols 
> however, 
> > > > which needs to be useful in the big-I Internet and 
> hopefully for 
> > > > folks building federations and other clubs as well.
> > > >
> > > > Everyone,
> > > >
> > > > I think codec selection should be out of scope.  Mandating
> > > > G.711 will have no positive effect now.  Everyone supports it 
> > > > today in some way or another, but it could be 
> detrimental in the 
> > > > future.  The point of the SDP negotiation is to allow RTP
> > > > *endpoints* to select the best codec in common under the 
> > > > circumstances.  I don't think we can do any better than 
> that in a 
> > > > requirements document, especially since we are describing an 
> > > > interface between SIP intermediaries that I hope will 
> not even be 
> > > > a participant in the RTP media most
> > > of the time.
> > > >
> > > > thanks,
> > > > -rohan
> > > >
> > > >
> > > >
> > > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com> wrote:
> > > > > Shouldn't the requirement be that the policy of a
> > > federation should
> > > > > address the default codecs used by a federation to 
> interconnect?
> > > > >
> > > > > Mike
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > > To: speermint@ietf.org
> > > > > > Subject: RE: [Speermint] I-D
> > > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > > >
> > > > > > Section 3.5 indicates:
> > > > > >
> > > > > >    The minimum requirements to allow a successful VoIP
> > > > interconnection
> > > > > >    include:
> > > > > >
> > > > > >    o  the mandatory support of RTP *and* RTCP as defined in 
> > > > > > [RFC3550],
> > > > > >
> > > > > >    o  the support of compatible codecs between
> > > > communication peers,
> > > > > > the
> > > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > > codec and its
> > > > > > RTP
> > > > > >       payload format [RFC3952] SHOULD be supported.
> > > > > >
> > > > > >    o  the support of the VoIP metric block as defined in
> > > > RTP Control
> > > > > >       Protocol Extended Reports [RFC3611] MAY be supported.
> > > > > >
> > > > > > What exactly does this mean to the two bodies that are
> > > making the
> > > > > > peering arrangement. Is this a requirement to control
> > > the codecs
> > > > > > used by the UAs attached to the peering organisations
> > equipment
> > > > > > before peering can occur, or a requirement to support
> > > > transcoding if
> > > > > > these requirements are not met by the population of
> > > attached UAs.
> > > > > >
> > > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new is 
> > > > > > imminently about to be published and I assume we 
> are going to 
> > > > > > reference that document (it will be RFC 4566).
> > > > > >
> > > > > > regards
> > > > > >
> > > > > > Keith
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Internet-Drafts@ietf.org
> > > [mailto:Internet-Drafts@ietf.org]
> > > > > > > Sent: 21 June 2006 23:50
> > > > > > > To: i-d-announce@ietf.org
> > > > > > > Cc: speermint@ietf.org
> > > > > > > Subject: [Speermint] I-D
> > > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > A New Internet-Draft is available from the on-line
> > > > Internet-Drafts
> > > > > > > directories.
> > > > > > > This draft is a work item of the Session PEERing for
> > > Multimedia
> > > > > > > INTerconnect Working Group of the IETF.
> > > > > > >
> > > > > > >     Title           : SPEERMINT Requirements for SIP-based
> > > > > > > VoIP Interconnection
> > > > > > >     Author(s)       : J. Mule
> > > > > > >     Filename        : draft-ietf-speermint-requirements-
> > 00.txt
> > > > > > >     Pages           : 10
> > > > > > >     Date            : 2006-6-21
> > > > > > >
> > > > > > >    This document describes general requirements 
> for Session 
> > > > > > > PEERing for
> > > > > > >    Multimedia INTerconnect and defines the minimum set of
> > > > > > requirements
> > > > > > >    applicable to SIP session peering for VoIP 
> interconnects.
> > > > > > >
> > > > > > >    In its current form, the document is a first draft
> > > > based on the
> > > > > > >    SPEERMINT mailing list's discussions on
> > > > requirements.  The main
> > > > > > >    objectives are to generate consensus on what categories
> > of
> > > > > > >    requirements should be covered, and to start more
> > > > discussions
> > > > > > > on the
> > > > > > >    technical protocol requirements that apply to VoIP
> > > > interconnects.
> > > > > > >
> > > > > > >
> > > > > > > A URL for this Internet-Draft is:
> > > > > > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-
> > requi
> > > > > > rements-00.txt
> > > > > >
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From speermint-bounces@ietf.org Wed Jun 28 11:15:34 2006
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
To: "Patrick Melampy" <PMelampy@acmepacket.com>
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Errors-To: speermint-bounces@ietf.org

I do not disagree with the fact that codec mismatches are a general
technology issue. =20

However, I am not yet convinced (I'd like some operators to weigh in)
that continuing to *discuss* whether or not there *may* be a need for a
lowest common demoninator codec requirement is necessarily bad.=20

Yes, endpoints or other elements will negotiate codecs, and no we're not
trying to tread into that area.  But I do think it's rational to
consider whether many peered domains may find it in their interest to
have a common codec, or language, to fall back on failing other, perhaps
more preferred or efficient codecs.  Whether someone had to transcode in
their network to meet such a requirement, were it even adopted, would be
out of scope as much as how the internals of IMS, PacketCable, and
washing machines are to us.  ;-)

Again, looking to have some operators weigh in with their thoughts.

Jason

> -----Original Message-----
> From: Patrick Melampy [mailto:PMelampy@acmepacket.com]=20
> Sent: Wednesday, June 28, 2006 10:44 AM
> To: Livingood, Jason
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t xt
>=20
> If there are 3 carriers...A, B, and C
>=20
> A may only support AMR (wireless network) B may support AMR=20
> and G.711 (wireless interconnect network) C may only support=20
> G.711 (terminating PSNT network)
>=20
> So now we have a problem...
>=20
> A-----B  (negotiates to AMR) WORKS
>=20
> B-----C  (negotiates to G.711) WORKS
>=20
> But A----via B-----C  FAILS [has no agreeable CODEC's]
>=20
> Thus B finds that on a call by call basis, if they support A=20
> and C, they may not be able to provide universal connectivity=20
> from A to C and vica-versa.
>=20
> I don't see how this problem can be avoided when there are 3=20
> or more carriers interconnected. I would suggest that this=20
> problem is outside the scope of spearmint, and just a general problem.
>=20
> I don't think we should require a codec (this doesn't always=20
> make sense), nor do I think we should require transcoding=20
> which is offensive in its own rite. I just think this is part=20
> of the internal problem B has with the services B wishes to=20
> offer. B may have to provide either segmented service=20
> offerings, and or transcoding internal to its network to=20
> support A and C peering relationships.
>=20
> Said another way, I do not believe incompatible CODEC's is a=20
> service provider peering issue, but rather a general=20
> technology issue that exists with or without service provider borders.
>=20
>=20
> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> Sent: Wednesday, June 28, 2006 6:46 AM
> To: Brian Rosen; Jean-Francois Mule; Drage, Keith (Keith);=20
> Rohan Mahy; Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D=20
> ACTION:draft-ietf-speermint-requirements-00.t
> xt
>=20
> Let's drop the IM and video examples -- for now.  Let's first=20
> focus on getting the basic one working that we're committed=20
> to: voice. =20
>=20
> So in the case of voice, the question at hand is - is there a=20
> lowest common denominator voice codec which should be=20
> required to do peering?
> If not, is there much utility in peering if certain provider=20
> types cannot agree on a common codec?
>=20
> Jason
>=20
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 5:40 PM
> > To: 'Jean-Francois Mule'; 'Drage, Keith (Keith)'; Livingood, Jason;=20
> > 'Rohan Mahy'; 'Michael Hammer (mhammer)'
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >=20
> > Consider an all-IMS, all-GSM federation.  Do they have to=20
> have G.711?
> >=20
> > Consider an all-video conference federation.  Do they have to have=20
> > G.711?
> >=20
> > Consider an all-IM federation.  Do they have to have G.711?
> >=20
> >=20
> > Brian
> >=20
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Tuesday, June 27, 2006 4:30 PM
> > To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy;=20
> Michael Hammer
> > (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t
> > xt
> >=20
> > So is this kind of scenario in scope of speermint?
> > Take:
> > 	- on one side, a 3GPP IMS network with the typical UE=20
> that does not=20
> > support G.711 on the terminal, and,
> > 	- on the other, a VSP like free world dialup or alike=20
> that allow any=20
> > sip UA with any kind of codec.
> >=20
> > Do we want to have some minimal requirements to ensure that=20
> > negotiation just does not fail?  I could care less where this=20
> > transcoding function is implemented.
> >=20
> > Jean-Francois
> >=20
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Tuesday, June 27, 2006 11:31 AM
> > > To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-
> > > 00.t xt
> > >=20
> > > But this still does not answer my original question.
> > >=20
> > > How can you mandate a codec for what is essentially a
> > network operator
> > > when that operator does not own the terminals (which
> > contain the vast
> > > majority of the codecs). Are you making a requirement for
> > transcoding
> > > support in the absence of whatever codec set is defined, or
> > is it just
> > > an empty letter?
> > >=20
> > > regards
> > >=20
> > > Keith
> > >=20
> > > > -----Original Message-----
> > > > From: Livingood, Jason=20
> [mailto:Jason_Livingood@cable.comcast.com]
> > > > Sent: 27 June 2006 16:24
> > > > To: Rohan Mahy; Michael Hammer (mhammer)
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > >
> > > > My initial thought is that you are accurate saying that
> > federations
> > > > are NOT REQUIRED for speermint.  Federations MAY or MAY
> > NOT be used,
> > > > as you noted, and to that extent will be considered in
> > some fashion
> > > > here.
> > > >
> > > > As for codecs, I think we'll have to see based on further WG=20
> > > > discussion.
> > > > I would surely agree that codec negotiation / selection=20
> is *well*=20
> > > > handled in the underlying SIP protocol itself.  However,
> > we may well
> > > > find a need to specify (or not) a lowest common=20
> denominator codec
> > > but
> > > > we'll have to see how our requirements work progress.
> > > >
> > > > Jason
> > > >
> > > > > -----Original Message-----
> > > > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > > > Sent: Sunday, June 25, 2006 2:41 PM
> > > > > To: Michael Hammer (mhammer)
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Mike,
> > > > >
> > > > > There is no requirement for a federation to exist in
> > speermint. =20
> > > > > It might or it might not, but the IETF won't be very=20
> effective=20
> > > > > trying to put requirements on something which is=20
> effectively a=20
> > > > > business arrangement.  We can define a set of protocols
> > however,
> > > > > which needs to be useful in the big-I Internet and
> > hopefully for
> > > > > folks building federations and other clubs as well.
> > > > >
> > > > > Everyone,
> > > > >
> > > > > I think codec selection should be out of scope.  Mandating
> > > > > G.711 will have no positive effect now.  Everyone supports it=20
> > > > > today in some way or another, but it could be
> > detrimental in the
> > > > > future.  The point of the SDP negotiation is to allow RTP
> > > > > *endpoints* to select the best codec in common under the=20
> > > > > circumstances.  I don't think we can do any better than
> > that in a
> > > > > requirements document, especially since we are describing an=20
> > > > > interface between SIP intermediaries that I hope will
> > not even be
> > > > > a participant in the RTP media most
> > > > of the time.
> > > > >
> > > > > thanks,
> > > > > -rohan
> > > > >
> > > > >
> > > > >
> > > > > On 6/23/06, Michael Hammer (mhammer)=20
> <mhammer@cisco.com> wrote:
> > > > > > Shouldn't the requirement be that the policy of a
> > > > federation should
> > > > > > address the default codecs used by a federation to
> > interconnect?
> > > > > >
> > > > > > Mike
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > > > To: speermint@ietf.org
> > > > > > > Subject: RE: [Speermint] I-D=20
> > > > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > > > >
> > > > > > > Section 3.5 indicates:
> > > > > > >
> > > > > > >    The minimum requirements to allow a successful VoIP
> > > > > interconnection
> > > > > > >    include:
> > > > > > >
> > > > > > >    o  the mandatory support of RTP *and* RTCP as=20
> defined in=20
> > > > > > > [RFC3550],
> > > > > > >
> > > > > > >    o  the support of compatible codecs between
> > > > > communication peers,
> > > > > > > the
> > > > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > > > codec and its
> > > > > > > RTP
> > > > > > >       payload format [RFC3952] SHOULD be supported.
> > > > > > >
> > > > > > >    o  the support of the VoIP metric block as defined in
> > > > > RTP Control
> > > > > > >       Protocol Extended Reports [RFC3611] MAY be=20
> supported.
> > > > > > >
> > > > > > > What exactly does this mean to the two bodies that are
> > > > making the
> > > > > > > peering arrangement. Is this a requirement to control
> > > > the codecs
> > > > > > > used by the UAs attached to the peering organisations
> > > equipment
> > > > > > > before peering can occur, or a requirement to support
> > > > > transcoding if
> > > > > > > these requirements are not met by the population of
> > > > attached UAs.
> > > > > > >
> > > > > > > Additional nit comment. RFC 2327 is obsolete and=20
> sdp-new is=20
> > > > > > > imminently about to be published and I assume we
> > are going to
> > > > > > > reference that document (it will be RFC 4566).
> > > > > > >
> > > > > > > regards
> > > > > > >
> > > > > > > Keith
> > > > > > >
> > > > > > > > -----Original Message-----
> > > > > > > > From: Internet-Drafts@ietf.org
> > > > [mailto:Internet-Drafts@ietf.org]
> > > > > > > > Sent: 21 June 2006 23:50
> > > > > > > > To: i-d-announce@ietf.org
> > > > > > > > Cc: speermint@ietf.org
> > > > > > > > Subject: [Speermint] I-D
> > > > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > > A New Internet-Draft is available from the on-line
> > > > > Internet-Drafts
> > > > > > > > directories.
> > > > > > > > This draft is a work item of the Session PEERing for
> > > > Multimedia
> > > > > > > > INTerconnect Working Group of the IETF.
> > > > > > > >
> > > > > > > >     Title           : SPEERMINT Requirements=20
> for SIP-based
> > > > > > > > VoIP Interconnection
> > > > > > > >     Author(s)       : J. Mule
> > > > > > > >     Filename        : draft-ietf-speermint-requirements-
> > > 00.txt
> > > > > > > >     Pages           : 10
> > > > > > > >     Date            : 2006-6-21
> > > > > > > >
> > > > > > > >    This document describes general requirements
> > for Session
> > > > > > > > PEERing for
> > > > > > > >    Multimedia INTerconnect and defines the=20
> minimum set of
> > > > > > > requirements
> > > > > > > >    applicable to SIP session peering for VoIP
> > interconnects.
> > > > > > > >
> > > > > > > >    In its current form, the document is a first draft
> > > > > based on the
> > > > > > > >    SPEERMINT mailing list's discussions on
> > > > > requirements.  The main
> > > > > > > >    objectives are to generate consensus on what=20
> categories
> > > of
> > > > > > > >    requirements should be covered, and to start more
> > > > > discussions
> > > > > > > > on the
> > > > > > > >    technical protocol requirements that apply to VoIP
> > > > > interconnects.
> > > > > > > >
> > > > > > > >
> > > > > > > > A URL for this Internet-Draft is:
> > > > > > > >=20
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-
> > > requi
> > > > > > > rements-00.txt
> > > > > > >
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> > > > > > >
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> > > > > > >       "FILE
> > > > > > >=20
> /internet-drafts/draft-ietf-speermint-requirements-00.txt".
> > > > > > >
> > > > > > > NOTE: The mail server at ietf.org can return the=20
> 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.
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > Speermint mailing list
> > > > > > > Speermint@ietf.org
> > > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >=20
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> >=20
> >=20
> >=20
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >=20
> >=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

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From speermint-bounces@ietf.org Wed Jun 28 11:28:50 2006
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Subject: RE: TLS for peering (was RE: Comments (RE: [speermint]
	requirementsdraft submitted))
From: Daryl Malas <daryl@level3.net>
To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
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I agree with Adam's assessment.  Requiring TLS is simply not realistic
at this point.  I believe it has the potential of becoming a more common
practice in the future, but not right now.

--Daryl

On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> TLS is becoming more and more a subject when the topic of SIP
> interconnections arises for me professionally.  This shows growing
> momentum indeed.  With that said, I can not agree with TLS as a
> steadfast requirement.  Economics and scale are my major concerns.  TLS
> adds cost, server load, and there are fail-over issues to consider as
> well.  For the person that designs an interconnect, I believe weighing
> the cost against the benefit of the different models a useful exercise.
> Mandating TLS, even if all parties involved in a peering arrangement
> feel somehow "safe and secure" due to underlying security mechanisms
> (L2/L3VPNs, etc), seems a little harsh.  
> 
> Adam
> 
> -----Original Message-----
> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com] 
> Sent: Tuesday, June 27, 2006 8:44 PM
> To: Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> requirementsdraft submitted))
> 
> I am not a provider yet, but I think we dont need to mess with TLS/TCP.
> As I understand one of the goals is to prevent the calls from failing
> due to "peering only" reasons. This means that things like codecs which
> are not negotiated by the peers need to be mandated at a minimal level.
> TCP/TLS can be negotiated between the peers themselves so it is a
> non-issue. Also as I still understand doing a high volume calls with TLS
> is still expensive. Mandating TLS in cases where the caller or callee
> dont require it doesnt seem to add any value.
> 
> -Medhavi.
> 
> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> >
> > TLS for _all_ scenarios? Even in private peerings, physically secure 
> > peerings, etc?
> >
> > Isn't that the same as requiring IPsec for all layer 3 peering? But 
> > since Layer 5 peering might be more distributed, the number of TLS 
> > sessions would be huge.
> >
> > It would be good to hear the provider's opinion on this...
> >
> > Thanks,
> >
> > Reinaldo
> >
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > Cc: speermint@ietf.org
> > > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> > >
> > > >
> > > > "3.4.  Security requirements
> > > >
> > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > >    transport-layer security as defined in RFC 3261,"
> > > >
> > > > Giving the fact that we can have security through physical
> > > segregation,
> > > > intra-federation/provider calls, etc, the "MUST" here seems too 
> > > > generic.
> > >
> > > I tend to differ on this: it is not because you have lower layer 
> > > security (presumably hop by hop security associations) that you 
> > > should relax the end-to-end transport security requirements.
> > > Rohan's well written draft is also in favor of mandatory support of
> > TLS,
> > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > >
> > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> > te
> > > rop-sibley-final.pdf).
> > > May be the requirement should be rewritten to put the MUST on SIP 
> > > entities involved in L5 peering.
> > >
> > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> > >
> > > Brian
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 
> _______________________________________________
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> https://www1.ietf.org/mailman/listinfo/speermint


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From speermint-bounces@ietf.org Wed Jun 28 11:41:21 2006
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Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Patrick Melampy" <PMelampy@acmepacket.com>,
	"Livingood, Jason" <Jason_Livingood@cable.comcast.com>
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Patrick,

See inline: this is a typical case of using a list formed from a union
of groups but the groups' intersection results in an empty list.

Patrick wrote:
> If there are 3 carriers...A, B, and C
>=20
> A may only support AMR (wireless network)
> B may support AMR and G.711 (wireless interconnect network)
> C may only support G.711 (terminating PSNT network)
>=20
> So now we have a problem...
>=20
> A-----B  (negotiates to AMR) WORKS
            ^^^^^^^^^^
If they "negotiate" codec, it means they do perform session peering
(A,B).

>=20
> B-----C  (negotiates to G.711) WORKS
Same here (B,C)

> But A----via B-----C  FAILS [has no agreeable CODEC's]
           ^^^
A and C do not perform session peering directly.
This is transit case: why should we care about this case?
If B agrees to transit calls from A to C (i.e. B can be a resolved
target for users of C in A's land), then B should provide codec
transcoding.=20
=20
> Thus B finds that on a call by call basis, if they support A and C,
> they may
> not be able to provide universal connectivity from A to C and vica-
> versa.
Blame B.

> I don't see how this problem can be avoided when there are 3 or more
> carriers interconnected.=20
The transit case is just a curved ball here, nothing new is added. A and
C are not L5 peers, period.

> I would suggest that this problem is outside
> the
> scope of spearmint, and just a general problem.
>=20
> I don't think we should require a codec (this doesn't always make
> sense),
> nor do I think we should require transcoding which is offensive in its
> own
> rite.=20

The direct peers (A,B) and (B,C) do have a codec in common.

> I just think this is part of the internal problem B has with the
> services B wishes to offer. B may have to provide either segmented
> service
> offerings, and or transcoding internal to its network to support A and
> C
> peering relationships.
>=20
> Said another way, I do not believe incompatible CODEC's is a service
> provider peering issue, but rather a general technology issue that
> exists
> with or without service provider borders.

I'd rather speak in terms of requirements rather than issues. Let's
depart from the provocative requirements of mandating G.711 everywhere
to refocus on more open questions: should any codec recommendations be
made in the specific WG deliverable for "the minimum set of requirements
for VoIP interconnects"?

Jean-Francois.

> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> Sent: Wednesday, June 28, 2006 6:46 AM
> To: Brian Rosen; Jean-Francois Mule; Drage, Keith (Keith); Rohan Mahy;
> Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-
> 00.t
> xt
>=20
> Let's drop the IM and video examples -- for now.  Let's first focus on
> getting the basic one working that we're committed to: voice.
>=20
> So in the case of voice, the question at hand is - is there a lowest
> common denominator voice codec which should be required to do peering?
> If not, is there much utility in peering if certain provider types
> cannot agree on a common codec?
>=20
> Jason
>=20
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 5:40 PM
> > To: 'Jean-Francois Mule'; 'Drage, Keith (Keith)'; Livingood,
> > Jason; 'Rohan Mahy'; 'Michael Hammer (mhammer)'
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> > Consider an all-IMS, all-GSM federation.  Do they have to have
> G.711?
> >
> > Consider an all-video conference federation.  Do they have to
> > have G.711?
> >
> > Consider an all-IM federation.  Do they have to have G.711?
> >
> >
> > Brian
> >
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Tuesday, June 27, 2006 4:30 PM
> > To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy; Michael
> Hammer
> > (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t
> > xt
> >
> > So is this kind of scenario in scope of speermint?
> > Take:
> > 	- on one side, a 3GPP IMS network with the typical UE
> > that does not support G.711 on the terminal, and,
> > 	- on the other, a VSP like free world dialup or alike
> > that allow any sip UA with any kind of codec.
> >
> > Do we want to have some minimal requirements to ensure that
> > negotiation just does not fail?  I could care less where this
> > transcoding function is implemented.
> >
> > Jean-Francois
> >
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Tuesday, June 27, 2006 11:31 AM
> > > To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-
> > > 00.t xt
> > >
> > > But this still does not answer my original question.
> > >
> > > How can you mandate a codec for what is essentially a
> > network operator
> > > when that operator does not own the terminals (which
> > contain the vast
> > > majority of the codecs). Are you making a requirement for
> > transcoding
> > > support in the absence of whatever codec set is defined, or
> > is it just
> > > an empty letter?
> > >
> > > regards
> > >
> > > Keith
> > >
> > > > -----Original Message-----
> > > > From: Livingood, Jason
> [mailto:Jason_Livingood@cable.comcast.com]
> > > > Sent: 27 June 2006 16:24
> > > > To: Rohan Mahy; Michael Hammer (mhammer)
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > >
> > > > My initial thought is that you are accurate saying that
> > federations
> > > > are NOT REQUIRED for speermint.  Federations MAY or MAY
> > NOT be used,
> > > > as you noted, and to that extent will be considered in
> > some fashion
> > > > here.
> > > >
> > > > As for codecs, I think we'll have to see based on further WG
> > > > discussion.
> > > > I would surely agree that codec negotiation / selection is
> *well*
> > > > handled in the underlying SIP protocol itself.  However,
> > we may well
> > > > find a need to specify (or not) a lowest common denominator
> codec
> > > but
> > > > we'll have to see how our requirements work progress.
> > > >
> > > > Jason
> > > >
> > > > > -----Original Message-----
> > > > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > > > Sent: Sunday, June 25, 2006 2:41 PM
> > > > > To: Michael Hammer (mhammer)
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Mike,
> > > > >
> > > > > There is no requirement for a federation to exist in
> > speermint.
> > > > > It might or it might not, but the IETF won't be very effective
> > > > > trying to put requirements on something which is effectively a
> > > > > business arrangement.  We can define a set of protocols
> > however,
> > > > > which needs to be useful in the big-I Internet and
> > hopefully for
> > > > > folks building federations and other clubs as well.
> > > > >
> > > > > Everyone,
> > > > >
> > > > > I think codec selection should be out of scope.  Mandating
> > > > > G.711 will have no positive effect now.  Everyone supports it
> > > > > today in some way or another, but it could be
> > detrimental in the
> > > > > future.  The point of the SDP negotiation is to allow RTP
> > > > > *endpoints* to select the best codec in common under the
> > > > > circumstances.  I don't think we can do any better than
> > that in a
> > > > > requirements document, especially since we are describing an
> > > > > interface between SIP intermediaries that I hope will
> > not even be
> > > > > a participant in the RTP media most
> > > > of the time.
> > > > >
> > > > > thanks,
> > > > > -rohan
> > > > >
> > > > >
> > > > >
> > > > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com>
> wrote:
> > > > > > Shouldn't the requirement be that the policy of a
> > > > federation should
> > > > > > address the default codecs used by a federation to
> > interconnect?
> > > > > >
> > > > > > Mike
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > > > To: speermint@ietf.org
> > > > > > > Subject: RE: [Speermint] I-D
> > > > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > > > >
> > > > > > > Section 3.5 indicates:
> > > > > > >
> > > > > > >    The minimum requirements to allow a successful VoIP
> > > > > interconnection
> > > > > > >    include:
> > > > > > >
> > > > > > >    o  the mandatory support of RTP *and* RTCP as defined
> in
> > > > > > > [RFC3550],
> > > > > > >
> > > > > > >    o  the support of compatible codecs between
> > > > > communication peers,
> > > > > > > the
> > > > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > > > codec and its
> > > > > > > RTP
> > > > > > >       payload format [RFC3952] SHOULD be supported.
> > > > > > >
> > > > > > >    o  the support of the VoIP metric block as defined in
> > > > > RTP Control
> > > > > > >       Protocol Extended Reports [RFC3611] MAY be
supported.
> > > > > > >
> > > > > > > What exactly does this mean to the two bodies that are
> > > > making the
> > > > > > > peering arrangement. Is this a requirement to control
> > > > the codecs
> > > > > > > used by the UAs attached to the peering organisations
> > > equipment
> > > > > > > before peering can occur, or a requirement to support
> > > > > transcoding if
> > > > > > > these requirements are not met by the population of
> > > > attached UAs.
> > > > > > >
> > > > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new
> is
> > > > > > > imminently about to be published and I assume we
> > are going to
> > > > > > > reference that document (it will be RFC 4566).
> > > > > > >
> > > > > > > regards
> > > > > > >
> > > > > > > Keith
> > > > > > >
> > > > > > > > -----Original Message-----
> > > > > > > > From: Internet-Drafts@ietf.org
> > > > [mailto:Internet-Drafts@ietf.org]
> > > > > > > > Sent: 21 June 2006 23:50
> > > > > > > > To: i-d-announce@ietf.org
> > > > > > > > Cc: speermint@ietf.org
> > > > > > > > Subject: [Speermint] I-D
> > > > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > > A New Internet-Draft is available from the on-line
> > > > > Internet-Drafts
> > > > > > > > directories.
> > > > > > > > This draft is a work item of the Session PEERing for
> > > > Multimedia
> > > > > > > > INTerconnect Working Group of the IETF.
> > > > > > > >
> > > > > > > >     Title           : SPEERMINT Requirements for SIP-
> based
> > > > > > > > VoIP Interconnection
> > > > > > > >     Author(s)       : J. Mule
> > > > > > > >     Filename        : draft-ietf-speermint-requirements-
> > > 00.txt
> > > > > > > >     Pages           : 10
> > > > > > > >     Date            : 2006-6-21
> > > > > > > >
> > > > > > > >    This document describes general requirements
> > for Session
> > > > > > > > PEERing for
> > > > > > > >    Multimedia INTerconnect and defines the minimum set
> of
> > > > > > > requirements
> > > > > > > >    applicable to SIP session peering for VoIP
> > interconnects.
> > > > > > > >
> > > > > > > >    In its current form, the document is a first draft
> > > > > based on the
> > > > > > > >    SPEERMINT mailing list's discussions on
> > > > > requirements.  The main
> > > > > > > >    objectives are to generate consensus on what
> categories
> > > of
> > > > > > > >    requirements should be covered, and to start more
> > > > > discussions
> > > > > > > > on the
> > > > > > > >    technical protocol requirements that apply to VoIP
> > > > > interconnects.
> > > > > > > >
> > > > > > > >
> > > > > > > > A URL for this Internet-Draft is:
> > > > > > > > http://www.ietf.org/internet-drafts/draft-ietf-
> speermint-
> > > requi
> > > > > > > rements-00.txt
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> >
>=20
> _______________________________________________
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>=20
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So, what do u recommended ?
Keep TLS optional?
Any other method?

Sohel=20


-----Original Message-----
From: "Daryl Malas" <daryl@level3.net>
To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
Cc: "speermint@ietf.org" <speermint@ietf.org>
Sent: 6/28/06 10:29 AM
Subject: RE: TLS for peering (was RE: Comments (RE: =
[speermint]requirementsdraft submitted))

I agree with Adam's assessment.  Requiring TLS is simply not realistic
at this point.  I believe it has the potential of becoming a more common
practice in the future, but not right now.

--Daryl

On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> TLS is becoming more and more a subject when the topic of SIP
> interconnections arises for me professionally.  This shows growing
> momentum indeed.  With that said, I can not agree with TLS as a
> steadfast requirement.  Economics and scale are my major concerns.  =
TLS
> adds cost, server load, and there are fail-over issues to consider as
> well.  For the person that designs an interconnect, I believe weighing
> the cost against the benefit of the different models a useful =
exercise.
> Mandating TLS, even if all parties involved in a peering arrangement
> feel somehow "safe and secure" due to underlying security mechanisms
> (L2/L3VPNs, etc), seems a little harsh. =20
>=20
> Adam
>=20
> -----Original Message-----
> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]=20
> Sent: Tuesday, June 27, 2006 8:44 PM
> To: Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> requirementsdraft submitted))
>=20
> I am not a provider yet, but I think we dont need to mess with =
TLS/TCP.
> As I understand one of the goals is to prevent the calls from failing
> due to "peering only" reasons. This means that things like codecs =
which
> are not negotiated by the peers need to be mandated at a minimal =
level.
> TCP/TLS can be negotiated between the peers themselves so it is a
> non-issue. Also as I still understand doing a high volume calls with =
TLS
> is still expensive. Mandating TLS in cases where the caller or callee
> dont require it doesnt seem to add any value.
>=20
> -Medhavi.
>=20
> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> >
> > TLS for _all_ scenarios? Even in private peerings, physically secure =

> > peerings, etc?
> >
> > Isn't that the same as requiring IPsec for all layer 3 peering? But=20
> > since Layer 5 peering might be more distributed, the number of TLS=20
> > sessions would be huge.
> >
> > It would be good to hear the provider's opinion on this...
> >
> > Thanks,
> >
> > Reinaldo
> >
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > Cc: speermint@ietf.org
> > > Subject: RE: Comments (RE: [speermint] requirements draft =
submitted)
> > >
> > > >
> > > > "3.4.  Security requirements
> > > >
> > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > >    transport-layer security as defined in RFC 3261,"
> > > >
> > > > Giving the fact that we can have security through physical
> > > segregation,
> > > > intra-federation/provider calls, etc, the "MUST" here seems too=20
> > > > generic.
> > >
> > > I tend to differ on this: it is not because you have lower layer=20
> > > security (presumably hop by hop security associations) that you=20
> > > should relax the end-to-end transport security requirements.
> > > Rohan's well written draft is also in favor of mandatory support =
of
> > TLS,
> > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > >
> > =
http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> > te
> > > rop-sibley-final.pdf).
> > > May be the requirement should be rewritten to put the MUST on SIP=20
> > > entities involved in L5 peering.
> > >
> > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a =
heartbeat.
> > >
> > > Brian
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
> _______________________________________________
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> Speermint@ietf.org
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>=20
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Jean-Francois Mule" <jf.mule@cablelabs.com>,
	"Patrick Melampy" <PMelampy@acmepacket.com>,
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A couple thoughts here.  I completely agree with Jean-Francois that we
should focus on the open question: should any codec recommendations be
made in the specific WG deliverable for "the minimum set of requirements
for VoIP interconnects"? - Here are my thoughts.  Transcoding sucks (ok
- it's not-preferred I mean), but a current reality because, as an
operator, we have the obligation to complete the call due to the
expectations of the peering relationship. If the call is to/fro the
PSTN, then G711 is most likely in the mix.  If the call is going to an
enterprise that has an affinity to G729, well then G729 it is for them.
I would _love_ to do anything that eliminates all the bad things (like
adding additional jitter and play-out buffers, adding superfluous
elements into call flows, degrading call quality due to a poor-quality
codec "hops" in the mix) that media transcoding introduces.  But I think
that by restricting the applicability by way of mandating codecs, etc is
not the right thing to do AT THIS STAGE.  I think we should take the
tactic of suggesting an IETF codec (iLBC or SPEEX) and indicate that a
common denominator is in everyone's best interest. (Well, everyone
except the transcoding vendors!  You know who you are!!)  But leave it
as a suggestion for now.  Did I mention that I didn't like
transcoding?!?!

-AU

-----Original Message-----
From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]=20
Sent: Wednesday, June 28, 2006 11:41 AM
To: Patrick Melampy; Livingood, Jason
Cc: speermint@ietf.org
Subject: RE: [Speermint] I-D
ACTION:draft-ietf-speermint-requirements-00.t xt

Patrick,

See inline: this is a typical case of using a list formed from a union
of groups but the groups' intersection results in an empty list.

Patrick wrote:
> If there are 3 carriers...A, B, and C
>=20
> A may only support AMR (wireless network) B may support AMR and G.711=20
> (wireless interconnect network) C may only support G.711 (terminating=20
> PSNT network)
>=20
> So now we have a problem...
>=20
> A-----B  (negotiates to AMR) WORKS
            ^^^^^^^^^^
If they "negotiate" codec, it means they do perform session peering
(A,B).

>=20
> B-----C  (negotiates to G.711) WORKS
Same here (B,C)

> But A----via B-----C  FAILS [has no agreeable CODEC's]
           ^^^
A and C do not perform session peering directly.
This is transit case: why should we care about this case?
If B agrees to transit calls from A to C (i.e. B can be a resolved
target for users of C in A's land), then B should provide codec
transcoding.=20
=20
> Thus B finds that on a call by call basis, if they support A and C,=20
> they may not be able to provide universal connectivity from A to C and

> vica- versa.
Blame B.

> I don't see how this problem can be avoided when there are 3 or more=20
> carriers interconnected.
The transit case is just a curved ball here, nothing new is added. A and
C are not L5 peers, period.

> I would suggest that this problem is outside the scope of spearmint,=20
> and just a general problem.
>=20
> I don't think we should require a codec (this doesn't always make=20
> sense), nor do I think we should require transcoding which is=20
> offensive in its own rite.

The direct peers (A,B) and (B,C) do have a codec in common.

> I just think this is part of the internal problem B has with the=20
> services B wishes to offer. B may have to provide either segmented=20
> service offerings, and or transcoding internal to its network to=20
> support A and C peering relationships.
>=20
> Said another way, I do not believe incompatible CODEC's is a service=20
> provider peering issue, but rather a general technology issue that=20
> exists with or without service provider borders.

I'd rather speak in terms of requirements rather than issues. Let's
depart from the provocative requirements of mandating G.711 everywhere
to refocus on more open questions: should any codec recommendations be
made in the specific WG deliverable for "the minimum set of requirements
for VoIP interconnects"?

Jean-Francois.

> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> Sent: Wednesday, June 28, 2006 6:46 AM
> To: Brian Rosen; Jean-Francois Mule; Drage, Keith (Keith); Rohan Mahy;

> Michael Hammer (mhammer)
> Cc: speermint@ietf.org
> Subject: RE: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-
> 00.t
> xt
>=20
> Let's drop the IM and video examples -- for now.  Let's first focus on

> getting the basic one working that we're committed to: voice.
>=20
> So in the case of voice, the question at hand is - is there a lowest=20
> common denominator voice codec which should be required to do peering?
> If not, is there much utility in peering if certain provider types=20
> cannot agree on a common codec?
>=20
> Jason
>=20
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 5:40 PM
> > To: 'Jean-Francois Mule'; 'Drage, Keith (Keith)'; Livingood, Jason;=20
> > 'Rohan Mahy'; 'Michael Hammer (mhammer)'
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t xt
> >
> > Consider an all-IMS, all-GSM federation.  Do they have to have
> G.711?
> >
> > Consider an all-video conference federation.  Do they have to have=20
> > G.711?
> >
> > Consider an all-IM federation.  Do they have to have G.711?
> >
> >
> > Brian
> >
> > -----Original Message-----
> > From: Jean-Francois Mule [mailto:jf.mule@cablelabs.com]
> > Sent: Tuesday, June 27, 2006 4:30 PM
> > To: Drage, Keith (Keith); Livingood, Jason; Rohan Mahy; Michael
> Hammer
> > (mhammer)
> > Cc: speermint@ietf.org
> > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-00.t
> > xt
> >
> > So is this kind of scenario in scope of speermint?
> > Take:
> > 	- on one side, a 3GPP IMS network with the typical UE that does
not=20
> > support G.711 on the terminal, and,
> > 	- on the other, a VSP like free world dialup or alike that allow

> > any sip UA with any kind of codec.
> >
> > Do we want to have some minimal requirements to ensure that=20
> > negotiation just does not fail?  I could care less where this=20
> > transcoding function is implemented.
> >
> > Jean-Francois
> >
> > > -----Original Message-----
> > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > Sent: Tuesday, June 27, 2006 11:31 AM
> > > To: Livingood, Jason; Rohan Mahy; Michael Hammer (mhammer)
> > > Cc: speermint@ietf.org
> > > Subject: RE: [Speermint] I-D
> > ACTION:draft-ietf-speermint-requirements-
> > > 00.t xt
> > >
> > > But this still does not answer my original question.
> > >
> > > How can you mandate a codec for what is essentially a
> > network operator
> > > when that operator does not own the terminals (which
> > contain the vast
> > > majority of the codecs). Are you making a requirement for
> > transcoding
> > > support in the absence of whatever codec set is defined, or
> > is it just
> > > an empty letter?
> > >
> > > regards
> > >
> > > Keith
> > >
> > > > -----Original Message-----
> > > > From: Livingood, Jason
> [mailto:Jason_Livingood@cable.comcast.com]
> > > > Sent: 27 June 2006 16:24
> > > > To: Rohan Mahy; Michael Hammer (mhammer)
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: [Speermint] I-D
> > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > >
> > > >
> > > > My initial thought is that you are accurate saying that
> > federations
> > > > are NOT REQUIRED for speermint.  Federations MAY or MAY
> > NOT be used,
> > > > as you noted, and to that extent will be considered in
> > some fashion
> > > > here.
> > > >
> > > > As for codecs, I think we'll have to see based on further WG=20
> > > > discussion.
> > > > I would surely agree that codec negotiation / selection is
> *well*
> > > > handled in the underlying SIP protocol itself.  However,
> > we may well
> > > > find a need to specify (or not) a lowest common denominator
> codec
> > > but
> > > > we'll have to see how our requirements work progress.
> > > >
> > > > Jason
> > > >
> > > > > -----Original Message-----
> > > > > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > > > > Sent: Sunday, June 25, 2006 2:41 PM
> > > > > To: Michael Hammer (mhammer)
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: [Speermint] I-D
> > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > >
> > > > > Mike,
> > > > >
> > > > > There is no requirement for a federation to exist in
> > speermint.
> > > > > It might or it might not, but the IETF won't be very effective

> > > > > trying to put requirements on something which is effectively a

> > > > > business arrangement.  We can define a set of protocols
> > however,
> > > > > which needs to be useful in the big-I Internet and
> > hopefully for
> > > > > folks building federations and other clubs as well.
> > > > >
> > > > > Everyone,
> > > > >
> > > > > I think codec selection should be out of scope.  Mandating
> > > > > G.711 will have no positive effect now.  Everyone supports it=20
> > > > > today in some way or another, but it could be
> > detrimental in the
> > > > > future.  The point of the SDP negotiation is to allow RTP
> > > > > *endpoints* to select the best codec in common under the=20
> > > > > circumstances.  I don't think we can do any better than
> > that in a
> > > > > requirements document, especially since we are describing an=20
> > > > > interface between SIP intermediaries that I hope will
> > not even be
> > > > > a participant in the RTP media most
> > > > of the time.
> > > > >
> > > > > thanks,
> > > > > -rohan
> > > > >
> > > > >
> > > > >
> > > > > On 6/23/06, Michael Hammer (mhammer) <mhammer@cisco.com>
> wrote:
> > > > > > Shouldn't the requirement be that the policy of a
> > > > federation should
> > > > > > address the default codecs used by a federation to
> > interconnect?
> > > > > >
> > > > > > Mike
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> > > > > > > Sent: Friday, June 23, 2006 10:22 AM
> > > > > > > To: speermint@ietf.org
> > > > > > > Subject: RE: [Speermint] I-D=20
> > > > > > > ACTION:draft-ietf-speermint-requirements-00.t xt
> > > > > > >
> > > > > > > Section 3.5 indicates:
> > > > > > >
> > > > > > >    The minimum requirements to allow a successful VoIP
> > > > > interconnection
> > > > > > >    include:
> > > > > > >
> > > > > > >    o  the mandatory support of RTP *and* RTCP as defined
> in
> > > > > > > [RFC3550],
> > > > > > >
> > > > > > >    o  the support of compatible codecs between
> > > > > communication peers,
> > > > > > > the
> > > > > > >       G.711 MUST be supported, the IETF iLBC [RFC3951]
> > > > > codec and its
> > > > > > > RTP
> > > > > > >       payload format [RFC3952] SHOULD be supported.
> > > > > > >
> > > > > > >    o  the support of the VoIP metric block as defined in
> > > > > RTP Control
> > > > > > >       Protocol Extended Reports [RFC3611] MAY be
supported.
> > > > > > >
> > > > > > > What exactly does this mean to the two bodies that are
> > > > making the
> > > > > > > peering arrangement. Is this a requirement to control
> > > > the codecs
> > > > > > > used by the UAs attached to the peering organisations
> > > equipment
> > > > > > > before peering can occur, or a requirement to support
> > > > > transcoding if
> > > > > > > these requirements are not met by the population of
> > > > attached UAs.
> > > > > > >
> > > > > > > Additional nit comment. RFC 2327 is obsolete and sdp-new
> is
> > > > > > > imminently about to be published and I assume we
> > are going to
> > > > > > > reference that document (it will be RFC 4566).
> > > > > > >
> > > > > > > regards
> > > > > > >
> > > > > > > Keith
> > > > > > >
> > > > > > > > -----Original Message-----
> > > > > > > > From: Internet-Drafts@ietf.org
> > > > [mailto:Internet-Drafts@ietf.org]
> > > > > > > > Sent: 21 June 2006 23:50
> > > > > > > > To: i-d-announce@ietf.org
> > > > > > > > Cc: speermint@ietf.org
> > > > > > > > Subject: [Speermint] I-D=20
> > > > > > > > ACTION:draft-ietf-speermint-requirements-00.txt
> > > > > > > >
> > > > > > > >
> > > > > > > >
> > > > > > > > A New Internet-Draft is available from the on-line
> > > > > Internet-Drafts
> > > > > > > > directories.
> > > > > > > > This draft is a work item of the Session PEERing for
> > > > Multimedia
> > > > > > > > INTerconnect Working Group of the IETF.
> > > > > > > >
> > > > > > > >     Title           : SPEERMINT Requirements for SIP-
> based
> > > > > > > > VoIP Interconnection
> > > > > > > >     Author(s)       : J. Mule
> > > > > > > >     Filename        : draft-ietf-speermint-requirements-
> > > 00.txt
> > > > > > > >     Pages           : 10
> > > > > > > >     Date            : 2006-6-21
> > > > > > > >
> > > > > > > >    This document describes general requirements
> > for Session
> > > > > > > > PEERing for
> > > > > > > >    Multimedia INTerconnect and defines the minimum set
> of
> > > > > > > requirements
> > > > > > > >    applicable to SIP session peering for VoIP
> > interconnects.
> > > > > > > >
> > > > > > > >    In its current form, the document is a first draft
> > > > > based on the
> > > > > > > >    SPEERMINT mailing list's discussions on
> > > > > requirements.  The main
> > > > > > > >    objectives are to generate consensus on what
> categories
> > > of
> > > > > > > >    requirements should be covered, and to start more
> > > > > discussions
> > > > > > > > on the
> > > > > > > >    technical protocol requirements that apply to VoIP
> > > > > interconnects.
> > > > > > > >
> > > > > > > >
> > > > > > > > A URL for this Internet-Draft is:
> > > > > > > > http://www.ietf.org/internet-drafts/draft-ietf-
> speermint-
> > > requi
> > > > > > > rements-00.txt
> > > > > > >
> > > > > > > To remove yourself from the I-D Announcement list, send a
> > > > > message to
> > > > > > > i-d-announce-request@ietf.org with the word unsubscribe
> > > > > in the body
> > > > > > > of the message.
> > > > > > > You can also visit
> > > > > > > https://www1.ietf.org/mailman/listinfo/I-D-announce
> > > > > > > to change your subscription settings.
> > > > > > >
> > > > > > >
> > > > > > > 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-speermint-requirements-00.txt".
> > > > > > >
> > > > > > > A list of Internet-Drafts directories can be found in=20
> > > > > > > http://www.ietf.org/shadow.html or=20
> > > > > > > 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-speermint-requirements-00.txt".
> > > > > > >
> > > > > > > NOTE: The mail server at ietf.org can return the document
> in
> > > > > > >       MIME-encoded form by using the "mpack" utility.
> > > > To use this
> > > > > > >       feature, insert the command "ENCODING mime" before
> > > > > the "FILE"
> > > > > > >       command.  To decode the response(s), you will need
> > > > > "munpack" or
> > > > > > >       a MIME-compliant mail reader.  Different
> > > > > MIME-compliant mail
> > > > > > > readers
> > > > > > >       exhibit different behavior, especially when
> > dealing with
> > > > > > >       "multipart" MIME messages (i.e. documents which
> > > > > have been split
> > > > > > >       up into multiple messages), so check your local
> > > > > documentation on
> > > > > > >       how to manipulate these messages.
> > > > > > >
> > > > > > >
> > > > > > > Below is the data which will enable a MIME compliant
> > > > mail reader
> > > > > > > implementation to automatically retrieve the ASCII
> > > > version of the
> > > > > > > Internet-Draft.
> > > > > > >
> > > > > > > _______________________________________________
> > > > > > > Speermint mailing list
> > > > > > > Speermint@ietf.org
> > > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > > >
> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
> _______________________________________________
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From: "Richard Shockey" <richard@shockey.us>
To: "'Khan, Sohel Q [CTO]'" <Sohel.Q.Khan@sprint.com>,
	"'Daryl Malas'" <daryl@level3.net>,
	"'Uzelac, Adam'" <Adam.Uzelac@globalcrossing.com>
Subject: RE: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
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TLS is a core part of the SIP specification. Saying its not realistic is
just plain bovine effluent.

You are not going to interconnect at the most basic level unless there is a
level of trust between the domains. So how do you plan on authenticating the
domain?

This was the absolute conclusion of the SIP Forum when we came up with the
SIP Connect specification for PBX to SP interconnection.

SP to SP interconnection is not that much different.

http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Interop-
sibley-final.pdf 

<sigh> 

> -----Original Message-----
> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> Sent: Wednesday, June 28, 2006 12:06 PM
> To: Daryl Malas; Uzelac, Adam
> Cc: speermint@ietf.org
> Subject: RE: TLS for peering (was RE: Comments
> (RE:[speermint]requirementsdraft submitted))
> 
> So, what do u recommended ?
> Keep TLS optional?
> Any other method?
> 
> Sohel
> 
> 
> -----Original Message-----
> From: "Daryl Malas" <daryl@level3.net>
> To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> Cc: "speermint@ietf.org" <speermint@ietf.org>
> Sent: 6/28/06 10:29 AM
> Subject: RE: TLS for peering (was RE: Comments (RE:
> [speermint]requirementsdraft submitted))
> 
> I agree with Adam's assessment.  Requiring TLS is simply not realistic
> at this point.  I believe it has the potential of becoming a more common
> practice in the future, but not right now.
> 
> --Daryl
> 
> On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > TLS is becoming more and more a subject when the topic of SIP
> > interconnections arises for me professionally.  This shows growing
> > momentum indeed.  With that said, I can not agree with TLS as a
> > steadfast requirement.  Economics and scale are my major concerns.  TLS
> > adds cost, server load, and there are fail-over issues to consider as
> > well.  For the person that designs an interconnect, I believe weighing
> > the cost against the benefit of the different models a useful exercise.
> > Mandating TLS, even if all parties involved in a peering arrangement
> > feel somehow "safe and secure" due to underlying security mechanisms
> > (L2/L3VPNs, etc), seems a little harsh.
> >
> > Adam
> >
> > -----Original Message-----
> > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > Sent: Tuesday, June 27, 2006 8:44 PM
> > To: Reinaldo Penno
> > Cc: speermint@ietf.org
> > Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> > requirementsdraft submitted))
> >
> > I am not a provider yet, but I think we dont need to mess with TLS/TCP.
> > As I understand one of the goals is to prevent the calls from failing
> > due to "peering only" reasons. This means that things like codecs which
> > are not negotiated by the peers need to be mandated at a minimal level.
> > TCP/TLS can be negotiated between the peers themselves so it is a
> > non-issue. Also as I still understand doing a high volume calls with TLS
> > is still expensive. Mandating TLS in cases where the caller or callee
> > dont require it doesnt seem to add any value.
> >
> > -Medhavi.
> >
> > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > >
> > > TLS for _all_ scenarios? Even in private peerings, physically secure
> > > peerings, etc?
> > >
> > > Isn't that the same as requiring IPsec for all layer 3 peering? But
> > > since Layer 5 peering might be more distributed, the number of TLS
> > > sessions would be huge.
> > >
> > > It would be good to hear the provider's opinion on this...
> > >
> > > Thanks,
> > >
> > > Reinaldo
> > >
> > > > -----Original Message-----
> > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> > > >
> > > > >
> > > > > "3.4.  Security requirements
> > > > >
> > > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > > >    transport-layer security as defined in RFC 3261,"
> > > > >
> > > > > Giving the fact that we can have security through physical
> > > > segregation,
> > > > > intra-federation/provider calls, etc, the "MUST" here seems too
> > > > > generic.
> > > >
> > > > I tend to differ on this: it is not because you have lower layer
> > > > security (presumably hop by hop security associations) that you
> > > > should relax the end-to-end transport security requirements.
> > > > Rohan's well written draft is also in favor of mandatory support of
> > > TLS,
> > > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > > >
> > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> > > te
> > > > rop-sibley-final.pdf).
> > > > May be the requirement should be rewritten to put the MUST on SIP
> > > > entities involved in L5 peering.
> > > >
> > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> > > >
> > > > Brian
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> 
> 
> _______________________________________________
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> Speermint@ietf.org
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> 
> 
> 
> _______________________________________________
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From: Andrew Newton <andy@hxr.us>
Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
Date: Wed, 28 Jun 2006 14:43:08 -0400
To: richard@shockey.us
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	"'Uzelac, Adam'" <Adam.Uzelac@globalcrossing.com>
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While we are at it, why don't we make signed DNS zones and DNSSEC  
TSIG mandatory too.  That way we can have many things in an RFC that  
are flat out ignored.

I agree with Adam and Daryl.

-andy

On Jun 28, 2006, at 2:28 PM, Richard Shockey wrote:

>
>
> TLS is a core part of the SIP specification. Saying its not  
> realistic is
> just plain bovine effluent.
>
> You are not going to interconnect at the most basic level unless  
> there is a
> level of trust between the domains. So how do you plan on  
> authenticating the
> domain?
>
> This was the absolute conclusion of the SIP Forum when we came up  
> with the
> SIP Connect specification for PBX to SP interconnection.
>
> SP to SP interconnection is not that much different.
>
> http://data.memberclicks.com/bbattach/130808/sf-draft-twg- 
> IP_PBX_SP_Interop-
> sibley-final.pdf
>
> <sigh>
>
>> -----Original Message-----
>> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
>> Sent: Wednesday, June 28, 2006 12:06 PM
>> To: Daryl Malas; Uzelac, Adam
>> Cc: speermint@ietf.org
>> Subject: RE: TLS for peering (was RE: Comments
>> (RE:[speermint]requirementsdraft submitted))
>>
>> So, what do u recommended ?
>> Keep TLS optional?
>> Any other method?
>>
>> Sohel
>>
>>
>> -----Original Message-----
>> From: "Daryl Malas" <daryl@level3.net>
>> To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
>> Cc: "speermint@ietf.org" <speermint@ietf.org>
>> Sent: 6/28/06 10:29 AM
>> Subject: RE: TLS for peering (was RE: Comments (RE:
>> [speermint]requirementsdraft submitted))
>>
>> I agree with Adam's assessment.  Requiring TLS is simply not  
>> realistic
>> at this point.  I believe it has the potential of becoming a more  
>> common
>> practice in the future, but not right now.
>>
>> --Daryl
>>
>> On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
>>> TLS is becoming more and more a subject when the topic of SIP
>>> interconnections arises for me professionally.  This shows growing
>>> momentum indeed.  With that said, I can not agree with TLS as a
>>> steadfast requirement.  Economics and scale are my major  
>>> concerns.  TLS
>>> adds cost, server load, and there are fail-over issues to  
>>> consider as
>>> well.  For the person that designs an interconnect, I believe  
>>> weighing
>>> the cost against the benefit of the different models a useful  
>>> exercise.
>>> Mandating TLS, even if all parties involved in a peering arrangement
>>> feel somehow "safe and secure" due to underlying security mechanisms
>>> (L2/L3VPNs, etc), seems a little harsh.
>>>
>>> Adam
>>>
>>> -----Original Message-----
>>> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
>>> Sent: Tuesday, June 27, 2006 8:44 PM
>>> To: Reinaldo Penno
>>> Cc: speermint@ietf.org
>>> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
>>> requirementsdraft submitted))
>>>
>>> I am not a provider yet, but I think we dont need to mess with  
>>> TLS/TCP.
>>> As I understand one of the goals is to prevent the calls from  
>>> failing
>>> due to "peering only" reasons. This means that things like codecs  
>>> which
>>> are not negotiated by the peers need to be mandated at a minimal  
>>> level.
>>> TCP/TLS can be negotiated between the peers themselves so it is a
>>> non-issue. Also as I still understand doing a high volume calls  
>>> with TLS
>>> is still expensive. Mandating TLS in cases where the caller or  
>>> callee
>>> dont require it doesnt seem to add any value.
>>>
>>> -Medhavi.
>>>
>>> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
>>>>
>>>> TLS for _all_ scenarios? Even in private peerings, physically  
>>>> secure
>>>> peerings, etc?
>>>>
>>>> Isn't that the same as requiring IPsec for all layer 3 peering? But
>>>> since Layer 5 peering might be more distributed, the number of TLS
>>>> sessions would be huge.
>>>>
>>>> It would be good to hear the provider's opinion on this...
>>>>
>>>> Thanks,
>>>>
>>>> Reinaldo
>>>>
>>>>> -----Original Message-----
>>>>> From: Brian Rosen [mailto:br@brianrosen.net]
>>>>> Sent: Tuesday, June 27, 2006 2:12 PM
>>>>> To: 'Jean-Francois Mule'; Reinaldo Penno
>>>>> Cc: speermint@ietf.org
>>>>> Subject: RE: Comments (RE: [speermint] requirements draft  
>>>>> submitted)
>>>>>
>>>>>>
>>>>>> "3.4.  Security requirements
>>>>>>
>>>>>>    All SIP messages MUST be sent over TLS [RFC3546] to provide
>>>>>>    transport-layer security as defined in RFC 3261,"
>>>>>>
>>>>>> Giving the fact that we can have security through physical
>>>>> segregation,
>>>>>> intra-federation/provider calls, etc, the "MUST" here seems too
>>>>>> generic.
>>>>>
>>>>> I tend to differ on this: it is not because you have lower layer
>>>>> security (presumably hop by hop security associations) that you
>>>>> should relax the end-to-end transport security requirements.
>>>>> Rohan's well written draft is also in favor of mandatory  
>>>>> support of
>>>> TLS,
>>>>> and so is the SIP Forum's IP PBX to VSP (here is a working link:
>>>>>
>>>> http://data.memberclicks.com/bbattach/130808/sf-draft-twg- 
>>>> IP_PBX_SP_In
>>>> te
>>>>> rop-sibley-final.pdf).
>>>>> May be the requirement should be rewritten to put the MUST on SIP
>>>>> entities involved in L5 peering.
>>>>>
>>>>> <brian>I'm in favor of requiring TLS.  I'd say MUST in a  
>>>>> heartbeat.
>>>>>
>>>>> Brian
>>>>
>>>> _______________________________________________
>>>> Speermint mailing list
>>>> Speermint@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/speermint
>>>>
>>>
>>> _______________________________________________
>>> Speermint mailing list
>>> Speermint@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/speermint
>>>
>>> _______________________________________________
>>> Speermint mailing list
>>> Speermint@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/speermint
>>
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>>
>>
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint


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From: "Richard Shockey" <richard@shockey.us>
To: "'Andrew Newton'" <andy@hxr.us>
Subject: RE: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
Date: Wed, 28 Jun 2006 14:48:25 -0400
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Really Andy ...  

how do you propose to establish a trust relationship between SIP domain
boundaries without TLS? 

If you cannot establish a trust relationship between interconnecting
partners then what is the purpose of this whole exercise?

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Wednesday, June 28, 2006 2:43 PM
> To: richard@shockey.us
> Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> speermint@ietf.org
> Subject: Re: TLS for peering (was RE: Comments
> (RE:[speermint]requirementsdraft submitted))
> 
> While we are at it, why don't we make signed DNS zones and DNSSEC
> TSIG mandatory too.  That way we can have many things in an RFC that
> are flat out ignored.
> 
> I agree with Adam and Daryl.
> 
> -andy
> 
> On Jun 28, 2006, at 2:28 PM, Richard Shockey wrote:
> 
> >
> >
> > TLS is a core part of the SIP specification. Saying its not
> > realistic is
> > just plain bovine effluent.
> >
> > You are not going to interconnect at the most basic level unless
> > there is a
> > level of trust between the domains. So how do you plan on
> > authenticating the
> > domain?
> >
> > This was the absolute conclusion of the SIP Forum when we came up
> > with the
> > SIP Connect specification for PBX to SP interconnection.
> >
> > SP to SP interconnection is not that much different.
> >
> > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_Interop-
> > sibley-final.pdf
> >
> > <sigh>
> >
> >> -----Original Message-----
> >> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> >> Sent: Wednesday, June 28, 2006 12:06 PM
> >> To: Daryl Malas; Uzelac, Adam
> >> Cc: speermint@ietf.org
> >> Subject: RE: TLS for peering (was RE: Comments
> >> (RE:[speermint]requirementsdraft submitted))
> >>
> >> So, what do u recommended ?
> >> Keep TLS optional?
> >> Any other method?
> >>
> >> Sohel
> >>
> >>
> >> -----Original Message-----
> >> From: "Daryl Malas" <daryl@level3.net>
> >> To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> >> Cc: "speermint@ietf.org" <speermint@ietf.org>
> >> Sent: 6/28/06 10:29 AM
> >> Subject: RE: TLS for peering (was RE: Comments (RE:
> >> [speermint]requirementsdraft submitted))
> >>
> >> I agree with Adam's assessment.  Requiring TLS is simply not
> >> realistic
> >> at this point.  I believe it has the potential of becoming a more
> >> common
> >> practice in the future, but not right now.
> >>
> >> --Daryl
> >>
> >> On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> >>> TLS is becoming more and more a subject when the topic of SIP
> >>> interconnections arises for me professionally.  This shows growing
> >>> momentum indeed.  With that said, I can not agree with TLS as a
> >>> steadfast requirement.  Economics and scale are my major
> >>> concerns.  TLS
> >>> adds cost, server load, and there are fail-over issues to
> >>> consider as
> >>> well.  For the person that designs an interconnect, I believe
> >>> weighing
> >>> the cost against the benefit of the different models a useful
> >>> exercise.
> >>> Mandating TLS, even if all parties involved in a peering arrangement
> >>> feel somehow "safe and secure" due to underlying security mechanisms
> >>> (L2/L3VPNs, etc), seems a little harsh.
> >>>
> >>> Adam
> >>>
> >>> -----Original Message-----
> >>> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> >>> Sent: Tuesday, June 27, 2006 8:44 PM
> >>> To: Reinaldo Penno
> >>> Cc: speermint@ietf.org
> >>> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> >>> requirementsdraft submitted))
> >>>
> >>> I am not a provider yet, but I think we dont need to mess with
> >>> TLS/TCP.
> >>> As I understand one of the goals is to prevent the calls from
> >>> failing
> >>> due to "peering only" reasons. This means that things like codecs
> >>> which
> >>> are not negotiated by the peers need to be mandated at a minimal
> >>> level.
> >>> TCP/TLS can be negotiated between the peers themselves so it is a
> >>> non-issue. Also as I still understand doing a high volume calls
> >>> with TLS
> >>> is still expensive. Mandating TLS in cases where the caller or
> >>> callee
> >>> dont require it doesnt seem to add any value.
> >>>
> >>> -Medhavi.
> >>>
> >>> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> >>>>
> >>>> TLS for _all_ scenarios? Even in private peerings, physically
> >>>> secure
> >>>> peerings, etc?
> >>>>
> >>>> Isn't that the same as requiring IPsec for all layer 3 peering? But
> >>>> since Layer 5 peering might be more distributed, the number of TLS
> >>>> sessions would be huge.
> >>>>
> >>>> It would be good to hear the provider's opinion on this...
> >>>>
> >>>> Thanks,
> >>>>
> >>>> Reinaldo
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: Brian Rosen [mailto:br@brianrosen.net]
> >>>>> Sent: Tuesday, June 27, 2006 2:12 PM
> >>>>> To: 'Jean-Francois Mule'; Reinaldo Penno
> >>>>> Cc: speermint@ietf.org
> >>>>> Subject: RE: Comments (RE: [speermint] requirements draft
> >>>>> submitted)
> >>>>>
> >>>>>>
> >>>>>> "3.4.  Security requirements
> >>>>>>
> >>>>>>    All SIP messages MUST be sent over TLS [RFC3546] to provide
> >>>>>>    transport-layer security as defined in RFC 3261,"
> >>>>>>
> >>>>>> Giving the fact that we can have security through physical
> >>>>> segregation,
> >>>>>> intra-federation/provider calls, etc, the "MUST" here seems too
> >>>>>> generic.
> >>>>>
> >>>>> I tend to differ on this: it is not because you have lower layer
> >>>>> security (presumably hop by hop security associations) that you
> >>>>> should relax the end-to-end transport security requirements.
> >>>>> Rohan's well written draft is also in favor of mandatory
> >>>>> support of
> >>>> TLS,
> >>>>> and so is the SIP Forum's IP PBX to VSP (here is a working link:
> >>>>>
> >>>> http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> >>>> IP_PBX_SP_In
> >>>> te
> >>>>> rop-sibley-final.pdf).
> >>>>> May be the requirement should be rewritten to put the MUST on SIP
> >>>>> entities involved in L5 peering.
> >>>>>
> >>>>> <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> >>>>> heartbeat.
> >>>>>
> >>>>> Brian
> >>>>
> >>>> _______________________________________________
> >>>> Speermint mailing list
> >>>> Speermint@ietf.org
> >>>> https://www1.ietf.org/mailman/listinfo/speermint
> >>>>
> >>>
> >>> _______________________________________________
> >>> Speermint mailing list
> >>> Speermint@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/speermint
> >>>
> >>> _______________________________________________
> >>> Speermint mailing list
> >>> Speermint@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/speermint
> >>
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/speermint
> >>
> >>
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint


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Subject: Re: TLS for peering (was RE: Comments
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On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> how do you propose to establish a trust relationship between SIP  
> domain
> boundaries without TLS?

There are other methods: security lower in the stack, private  
connections, address ACLs...  the things that providers use today.

> If you cannot establish a trust relationship between interconnecting
> partners then what is the purpose of this whole exercise?

To answer the second part of that sentence: I don't know; I've been  
asking that question for quite some time.

-andy

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Andrew Newton'" <andy@hxr.us>, richard@shockey.us
Subject: RE: TLS for peering (was RE: Comments (RE:[speermint]requirements
	draft submitted))
Date: Wed, 28 Jun 2006 20:08:51 +0100
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Given that writing a requirement in RFC 3261 (standards track) that all SIP implementations must support TLS has not resulted in its universal deployment and use, why should writing it into a SPEERMINT BCP result in any wider deployment and use (BCP is what the milestone in the charter currently indicates)? 

[And if there is a rational answer to the above question - it would be worth identifying if this means we should be taking that back into SIP rather than documenting it in SPEERMINT]

I guess that this raises a more general issue. Is this BCP (and other SPEERMINT work) meant to be focussed on trying to change the world, or should it be focussed on solving immediate problems with getting SIP networks to connect. If the former, then by all means specify TLS. If the latter, then it should probably identify that security solutions are required, and then make sure that sufficient guidance is given on the mechanisms currently available, including TLS, but probably also addressing IPsec.
	
regards

Keith

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: 28 June 2006 19:43
> To: richard@shockey.us
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: Re: TLS for peering (was RE: Comments
> (RE:[speermint]requirementsdraft submitted))
> 
> 
> While we are at it, why don't we make signed DNS zones and DNSSEC  
> TSIG mandatory too.  That way we can have many things in an RFC that  
> are flat out ignored.
> 
> I agree with Adam and Daryl.
> 
> -andy
> 
> On Jun 28, 2006, at 2:28 PM, Richard Shockey wrote:
> 
> >
> >
> > TLS is a core part of the SIP specification. Saying its not  
> > realistic is
> > just plain bovine effluent.
> >
> > You are not going to interconnect at the most basic level unless  
> > there is a
> > level of trust between the domains. So how do you plan on  
> > authenticating the
> > domain?
> >
> > This was the absolute conclusion of the SIP Forum when we came up  
> > with the
> > SIP Connect specification for PBX to SP interconnection.
> >
> > SP to SP interconnection is not that much different.
> >
> > http://data.memberclicks.com/bbattach/130808/sf-draft-twg- 
> > IP_PBX_SP_Interop-
> > sibley-final.pdf
> >
> > <sigh>
> >
> >> -----Original Message-----
> >> From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> >> Sent: Wednesday, June 28, 2006 12:06 PM
> >> To: Daryl Malas; Uzelac, Adam
> >> Cc: speermint@ietf.org
> >> Subject: RE: TLS for peering (was RE: Comments
> >> (RE:[speermint]requirementsdraft submitted))
> >>
> >> So, what do u recommended ?
> >> Keep TLS optional?
> >> Any other method?
> >>
> >> Sohel
> >>
> >>
> >> -----Original Message-----
> >> From: "Daryl Malas" <daryl@level3.net>
> >> To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> >> Cc: "speermint@ietf.org" <speermint@ietf.org>
> >> Sent: 6/28/06 10:29 AM
> >> Subject: RE: TLS for peering (was RE: Comments (RE:
> >> [speermint]requirementsdraft submitted))
> >>
> >> I agree with Adam's assessment.  Requiring TLS is simply not  
> >> realistic
> >> at this point.  I believe it has the potential of becoming a more  
> >> common
> >> practice in the future, but not right now.
> >>
> >> --Daryl
> >>
> >> On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> >>> TLS is becoming more and more a subject when the topic of SIP
> >>> interconnections arises for me professionally.  This shows growing
> >>> momentum indeed.  With that said, I can not agree with TLS as a
> >>> steadfast requirement.  Economics and scale are my major  
> >>> concerns.  TLS
> >>> adds cost, server load, and there are fail-over issues to  
> >>> consider as
> >>> well.  For the person that designs an interconnect, I believe  
> >>> weighing
> >>> the cost against the benefit of the different models a useful  
> >>> exercise.
> >>> Mandating TLS, even if all parties involved in a peering 
> arrangement
> >>> feel somehow "safe and secure" due to underlying security 
> mechanisms
> >>> (L2/L3VPNs, etc), seems a little harsh.
> >>>
> >>> Adam
> >>>
> >>> -----Original Message-----
> >>> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> >>> Sent: Tuesday, June 27, 2006 8:44 PM
> >>> To: Reinaldo Penno
> >>> Cc: speermint@ietf.org
> >>> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> >>> requirementsdraft submitted))
> >>>
> >>> I am not a provider yet, but I think we dont need to mess with  
> >>> TLS/TCP.
> >>> As I understand one of the goals is to prevent the calls from  
> >>> failing
> >>> due to "peering only" reasons. This means that things 
> like codecs  
> >>> which
> >>> are not negotiated by the peers need to be mandated at a minimal  
> >>> level.
> >>> TCP/TLS can be negotiated between the peers themselves so it is a
> >>> non-issue. Also as I still understand doing a high volume calls  
> >>> with TLS
> >>> is still expensive. Mandating TLS in cases where the caller or  
> >>> callee
> >>> dont require it doesnt seem to add any value.
> >>>
> >>> -Medhavi.
> >>>
> >>> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> >>>>
> >>>> TLS for _all_ scenarios? Even in private peerings, physically  
> >>>> secure
> >>>> peerings, etc?
> >>>>
> >>>> Isn't that the same as requiring IPsec for all layer 3 
> peering? But
> >>>> since Layer 5 peering might be more distributed, the 
> number of TLS
> >>>> sessions would be huge.
> >>>>
> >>>> It would be good to hear the provider's opinion on this...
> >>>>
> >>>> Thanks,
> >>>>
> >>>> Reinaldo
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: Brian Rosen [mailto:br@brianrosen.net]
> >>>>> Sent: Tuesday, June 27, 2006 2:12 PM
> >>>>> To: 'Jean-Francois Mule'; Reinaldo Penno
> >>>>> Cc: speermint@ietf.org
> >>>>> Subject: RE: Comments (RE: [speermint] requirements draft  
> >>>>> submitted)
> >>>>>
> >>>>>>
> >>>>>> "3.4.  Security requirements
> >>>>>>
> >>>>>>    All SIP messages MUST be sent over TLS [RFC3546] to provide
> >>>>>>    transport-layer security as defined in RFC 3261,"
> >>>>>>
> >>>>>> Giving the fact that we can have security through physical
> >>>>> segregation,
> >>>>>> intra-federation/provider calls, etc, the "MUST" here seems too
> >>>>>> generic.
> >>>>>
> >>>>> I tend to differ on this: it is not because you have lower layer
> >>>>> security (presumably hop by hop security associations) that you
> >>>>> should relax the end-to-end transport security requirements.
> >>>>> Rohan's well written draft is also in favor of mandatory  
> >>>>> support of
> >>>> TLS,
> >>>>> and so is the SIP Forum's IP PBX to VSP (here is a working link:
> >>>>>
> >>>> http://data.memberclicks.com/bbattach/130808/sf-draft-twg- 
> >>>> IP_PBX_SP_In
> >>>> te
> >>>>> rop-sibley-final.pdf).
> >>>>> May be the requirement should be rewritten to put the 
> MUST on SIP
> >>>>> entities involved in L5 peering.
> >>>>>
> >>>>> <brian>I'm in favor of requiring TLS.  I'd say MUST in a  
> >>>>> heartbeat.
> >>>>>
> >>>>> Brian
> >>>>
> >>>> _______________________________________________
> >>>> Speermint mailing list
> >>>> Speermint@ietf.org
> >>>> https://www1.ietf.org/mailman/listinfo/speermint
> >>>>
> >>>
> >>> _______________________________________________
> >>> Speermint mailing list
> >>> Speermint@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/speermint
> >>>
> >>> _______________________________________________
> >>> Speermint mailing list
> >>> Speermint@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/speermint
> >>
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/speermint
> >>
> >>
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/speermint
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> 
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 

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From: "Richard Shockey" <richard@shockey.us>
To: "'Andrew Newton'" <andy@hxr.us>
Subject: RE: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
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Well the one thing we all seem to agree on is that without some trust
relationship between interconnecting partners at their domain boundaries
this whole exercise is a vast waste of time and electrons.

Now then I think Keith has a point.

"If the former, then by all means specify TLS. If the latter, then it should
probably identify that security solutions are required, and then make sure
that sufficient guidance is given on the mechanisms currently available,
including TLS, but probably also addressing IPsec."

Required ..I's say security solutions are mandated as in MUST.

There _are_ certainly more than one mechanism, besides TLS that can be used
to establish security & trust at the boundaries. So ..those techniques MUST
be deliberately specified along with the risks associated with them.

That said there are many of us that are getting very very tired of the
constant whining about this or that aspect of 3261 that for one reason or
another some developer or service provider doesn't like. Some of us just
went through 6 months of this discussion to reach consensus on the
SIPConnect with TLS support since it was the obvious choice.

TLS support IS mandated in 3261 so IMHO get with the program.
	
I've yet to see real concrete data on why its such a problem to implement or
support in high volume SIP transaction systems. Lots of http based Ecommerce
seems to rely on per transaction secure connections so I just don't get what
the problem is here.	

> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Wednesday, June 28, 2006 3:06 PM
> To: richard@shockey.us
> Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> speermint@ietf.org
> Subject: Re: TLS for peering (was RE: Comments
> (RE:[speermint]requirementsdraft submitted))
> 
> 
> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > how do you propose to establish a trust relationship between SIP
> > domain
> > boundaries without TLS?
> 
> There are other methods: security lower in the stack, private
> connections, address ACLs...  the things that providers use today.
> 
> > If you cannot establish a trust relationship between interconnecting
> > partners then what is the purpose of this whole exercise?
> 
> To answer the second part of that sentence: I don't know; I've been
> asking that question for quite some time.
> 
> -andy


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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'richard@shockey.us'" <richard@shockey.us>, "'Andrew Newton'"
	<andy@hxr.us>
Subject: RE: TLS for peering (was RE: Comments (RE:[speermint]requirements
	draft submitted))
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I was using REQUIRED as in the RFC 2119 sense - I just forgot to capitalise it.

regards

Keith

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: 28 June 2006 20:41
> To: 'Andrew Newton'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE: Comments
> (RE:[speermint]requirementsdraft submitted))
> 
> 
> 
> 
> Well the one thing we all seem to agree on is that without some trust
> relationship between interconnecting partners at their domain 
> boundaries
> this whole exercise is a vast waste of time and electrons.
> 
> Now then I think Keith has a point.
> 
> "If the former, then by all means specify TLS. If the latter, 
> then it should
> probably identify that security solutions are required, and 
> then make sure
> that sufficient guidance is given on the mechanisms currently 
> available,
> including TLS, but probably also addressing IPsec."
> 
> Required ..I's say security solutions are mandated as in MUST.
> 
> There _are_ certainly more than one mechanism, besides TLS 
> that can be used
> to establish security & trust at the boundaries. So ..those 
> techniques MUST
> be deliberately specified along with the risks associated with them.
> 
> That said there are many of us that are getting very very tired of the
> constant whining about this or that aspect of 3261 that for 
> one reason or
> another some developer or service provider doesn't like. Some 
> of us just
> went through 6 months of this discussion to reach consensus on the
> SIPConnect with TLS support since it was the obvious choice.
> 
> TLS support IS mandated in 3261 so IMHO get with the program.
> 	
> I've yet to see real concrete data on why its such a problem 
> to implement or
> support in high volume SIP transaction systems. Lots of http 
> based Ecommerce
> seems to rely on per transaction secure connections so I just 
> don't get what
> the problem is here.	
> 
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Wednesday, June 28, 2006 3:06 PM
> > To: richard@shockey.us
> > Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> > speermint@ietf.org
> > Subject: Re: TLS for peering (was RE: Comments
> > (RE:[speermint]requirementsdraft submitted))
> > 
> > 
> > On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > > how do you propose to establish a trust relationship between SIP
> > > domain
> > > boundaries without TLS?
> > 
> > There are other methods: security lower in the stack, private
> > connections, address ACLs...  the things that providers use today.
> > 
> > > If you cannot establish a trust relationship between 
> interconnecting
> > > partners then what is the purpose of this whole exercise?
> > 
> > To answer the second part of that sentence: I don't know; I've been
> > asking that question for quite some time.
> > 
> > -andy
> 
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 

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Hello Richard,

I guess you put the answer in the email itself. Adam and others talked
about cost vs. benefit.=20

Speaking out of experience, these 'e-commerce' sites you mention have
stacks and stacks of SSL accelerators and/or offpath SSL decryption
engines. It definitely increases the equipment cost, management, real
state and operation of the solution.

Also, if you have looked long enough on RFCs, many, many things that are
mandated never see the light of the day because they lack consensus, are
complex, never factored service provider's opinion, etc, etc.=20

Layer 3 peering has been going on for decades without IPsec.=20

I believe we are not saying TLS is evil. We are saying 'MUST' is not the
right word.=20

Regards,

Reinaldo


> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Wednesday, June 28, 2006 12:41 PM
> To: 'Andrew Newton'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
>=20
>=20
> Well the one thing we all seem to agree on is that without some trust
> relationship between interconnecting partners at their domain
boundaries
> this whole exercise is a vast waste of time and electrons.
>=20
> Now then I think Keith has a point.
>=20
> "If the former, then by all means specify TLS. If the latter, then it
> should
> probably identify that security solutions are required, and then make
sure
> that sufficient guidance is given on the mechanisms currently
available,
> including TLS, but probably also addressing IPsec."
>=20
> Required ..I's say security solutions are mandated as in MUST.
>=20
> There _are_ certainly more than one mechanism, besides TLS that can be
> used
> to establish security & trust at the boundaries. So ..those techniques
> MUST
> be deliberately specified along with the risks associated with them.
>=20
> That said there are many of us that are getting very very tired of the
> constant whining about this or that aspect of 3261 that for one reason
or
> another some developer or service provider doesn't like. Some of us
just
> went through 6 months of this discussion to reach consensus on the
> SIPConnect with TLS support since it was the obvious choice.
>=20
> TLS support IS mandated in 3261 so IMHO get with the program.
>=20
> I've yet to see real concrete data on why its such a problem to
implement
> or
> support in high volume SIP transaction systems. Lots of http based
> Ecommerce
> seems to rely on per transaction secure connections so I just don't
get
> what
> the problem is here.
>=20
> > -----Original Message-----
> > From: Andrew Newton [mailto:andy@hxr.us]
> > Sent: Wednesday, June 28, 2006 3:06 PM
> > To: richard@shockey.us
> > Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> > speermint@ietf.org
> > Subject: Re: TLS for peering (was RE: Comments
> > (RE:[speermint]requirementsdraft submitted))
> >
> >
> > On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > > how do you propose to establish a trust relationship between SIP
> > > domain
> > > boundaries without TLS?
> >
> > There are other methods: security lower in the stack, private
> > connections, address ACLs...  the things that providers use today.
> >
> > > If you cannot establish a trust relationship between
interconnecting
> > > partners then what is the purpose of this whole exercise?
> >
> > To answer the second part of that sentence: I don't know; I've been
> > asking that question for quite some time.
> >
> > -andy
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
From: Daryl Malas <daryl@level3.net>
To: richard@shockey.us
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...I simply said REQUIRING it would be unrealistic...I didn't say
including it as an optional method is unrealistic.  There are multiple
ways to authenticate peers as described in Penno's message flows draft.

On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> 
> TLS is a core part of the SIP specification. Saying its not realistic is
> just plain bovine effluent.
> 
> You are not going to interconnect at the most basic level unless there is a
> level of trust between the domains. So how do you plan on authenticating the
> domain?
> 
> This was the absolute conclusion of the SIP Forum when we came up with the
> SIP Connect specification for PBX to SP interconnection.
> 
> SP to SP interconnection is not that much different.
> 
> http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_Interop-
> sibley-final.pdf 
> 
> <sigh> 
> 
> > -----Original Message-----
> > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > Sent: Wednesday, June 28, 2006 12:06 PM
> > To: Daryl Malas; Uzelac, Adam
> > Cc: speermint@ietf.org
> > Subject: RE: TLS for peering (was RE: Comments
> > (RE:[speermint]requirementsdraft submitted))
> > 
> > So, what do u recommended ?
> > Keep TLS optional?
> > Any other method?
> > 
> > Sohel
> > 
> > 
> > -----Original Message-----
> > From: "Daryl Malas" <daryl@level3.net>
> > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > Sent: 6/28/06 10:29 AM
> > Subject: RE: TLS for peering (was RE: Comments (RE:
> > [speermint]requirementsdraft submitted))
> > 
> > I agree with Adam's assessment.  Requiring TLS is simply not realistic
> > at this point.  I believe it has the potential of becoming a more common
> > practice in the future, but not right now.
> > 
> > --Daryl
> > 
> > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > TLS is becoming more and more a subject when the topic of SIP
> > > interconnections arises for me professionally.  This shows growing
> > > momentum indeed.  With that said, I can not agree with TLS as a
> > > steadfast requirement.  Economics and scale are my major concerns.  TLS
> > > adds cost, server load, and there are fail-over issues to consider as
> > > well.  For the person that designs an interconnect, I believe weighing
> > > the cost against the benefit of the different models a useful exercise.
> > > Mandating TLS, even if all parties involved in a peering arrangement
> > > feel somehow "safe and secure" due to underlying security mechanisms
> > > (L2/L3VPNs, etc), seems a little harsh.
> > >
> > > Adam
> > >
> > > -----Original Message-----
> > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > To: Reinaldo Penno
> > > Cc: speermint@ietf.org
> > > Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> > > requirementsdraft submitted))
> > >
> > > I am not a provider yet, but I think we dont need to mess with TLS/TCP.
> > > As I understand one of the goals is to prevent the calls from failing
> > > due to "peering only" reasons. This means that things like codecs which
> > > are not negotiated by the peers need to be mandated at a minimal level.
> > > TCP/TLS can be negotiated between the peers themselves so it is a
> > > non-issue. Also as I still understand doing a high volume calls with TLS
> > > is still expensive. Mandating TLS in cases where the caller or callee
> > > dont require it doesnt seem to add any value.
> > >
> > > -Medhavi.
> > >
> > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > >
> > > > TLS for _all_ scenarios? Even in private peerings, physically secure
> > > > peerings, etc?
> > > >
> > > > Isn't that the same as requiring IPsec for all layer 3 peering? But
> > > > since Layer 5 peering might be more distributed, the number of TLS
> > > > sessions would be huge.
> > > >
> > > > It would be good to hear the provider's opinion on this...
> > > >
> > > > Thanks,
> > > >
> > > > Reinaldo
> > > >
> > > > > -----Original Message-----
> > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > Cc: speermint@ietf.org
> > > > > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> > > > >
> > > > > >
> > > > > > "3.4.  Security requirements
> > > > > >
> > > > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > >
> > > > > > Giving the fact that we can have security through physical
> > > > > segregation,
> > > > > > intra-federation/provider calls, etc, the "MUST" here seems too
> > > > > > generic.
> > > > >
> > > > > I tend to differ on this: it is not because you have lower layer
> > > > > security (presumably hop by hop security associations) that you
> > > > > should relax the end-to-end transport security requirements.
> > > > > Rohan's well written draft is also in favor of mandatory support of
> > > > TLS,
> > > > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > > > >
> > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> > > > te
> > > > > rop-sibley-final.pdf).
> > > > > May be the requirement should be rewritten to put the MUST on SIP
> > > > > entities involved in L5 peering.
> > > > >
> > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> > > > >
> > > > > Brian
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > 
> > 
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> > 
> > 
> > 
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint


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> -----Original Message-----
> From: Reinaldo Penno [mailto:rpenno@juniper.net]
> Sent: Wednesday, June 28, 2006 4:17 PM
> To: richard@shockey.us; Andrew Newton
> Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> Hello Richard,
> 
> I guess you put the answer in the email itself. Adam and others talked
> about cost vs. benefit.
> 
> Speaking out of experience, these 'e-commerce' sites you mention have
> stacks and stacks of SSL accelerators and/or offpath SSL decryption
> engines. It definitely increases the equipment cost, management, real
> state and operation of the solution.

Like duh? So what?

There is real money involved here. Are you saying in inter carrier SIP
communications, transaction security, fraud avoidance and UA identity
management/control across domain boundaries are not worth the security cost?

Voice communications alone is a 300 Billion US business world wide? What's a
few CPU cycles compared to that.

 
> Also, if you have looked long enough on RFCs, many, many things that are
> mandated never see the light of the day because they lack consensus, are
> complex, never factored service provider's opinion, etc, etc.

Been there done that. You could say that for about 60% of the Security Area
RFC's, IPv6, DNSSEC, IDN, PKI, Public ENUM in e164.arpa, etal 

> 
> Layer 3 peering has been going on for decades without IPsec.


Yes but that has been for best efforts IP traffic. 

The entire premise of speermint at its core is the supposition that though
all bits are created equal on the Internet it seems that some SIP bits are
more equal than others.

If you don't believe that, this WG has no reason for existence. 

Since some traffic between SIP interconnection partners ( notice I never say
peering ) may involve PSTN termination charges or use bandwidth in high
volume priority access mode, there must be a reasonable level of trust
between interconnecting partners. There has to be the capability of the
terminating service provider trusting the first who the originating provider
is and second trusting the calling party identity provided to them for
billing purposes if nothing else. Therefore the security methodology
mandated is critical.


> 
> I believe we are not saying TLS is evil. We are saying 'MUST' is not the
> right word.

Well again 3261 said MUST for a very very good reason. I would still require
MUST support TLS in any SPEERMENT standard or BCP. However I accept that
other trust/security methodologies can work but they had better be well
documented along with the possible security threats associated with each
proposed method.

> 
> Regards,
> 
> Reinaldo
> 
> 
> > -----Original Message-----
> > From: Richard Shockey [mailto:richard@shockey.us]
> > Sent: Wednesday, June 28, 2006 12:41 PM
> > To: 'Andrew Newton'
> > Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> >
> >
> > Well the one thing we all seem to agree on is that without some trust
> > relationship between interconnecting partners at their domain
> boundaries
> > this whole exercise is a vast waste of time and electrons.
> >
> > Now then I think Keith has a point.
> >
> > "If the former, then by all means specify TLS. If the latter, then it
> > should
> > probably identify that security solutions are required, and then make
> sure
> > that sufficient guidance is given on the mechanisms currently
> available,
> > including TLS, but probably also addressing IPsec."
> >
> > Required ..I's say security solutions are mandated as in MUST.
> >
> > There _are_ certainly more than one mechanism, besides TLS that can be
> > used
> > to establish security & trust at the boundaries. So ..those techniques
> > MUST
> > be deliberately specified along with the risks associated with them.
> >
> > That said there are many of us that are getting very very tired of the
> > constant whining about this or that aspect of 3261 that for one reason
> or
> > another some developer or service provider doesn't like. Some of us
> just
> > went through 6 months of this discussion to reach consensus on the
> > SIPConnect with TLS support since it was the obvious choice.
> >
> > TLS support IS mandated in 3261 so IMHO get with the program.
> >
> > I've yet to see real concrete data on why its such a problem to
> implement
> > or
> > support in high volume SIP transaction systems. Lots of http based
> > Ecommerce
> > seems to rely on per transaction secure connections so I just don't
> get
> > what
> > the problem is here.
> >
> > > -----Original Message-----
> > > From: Andrew Newton [mailto:andy@hxr.us]
> > > Sent: Wednesday, June 28, 2006 3:06 PM
> > > To: richard@shockey.us
> > > Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> > > speermint@ietf.org
> > > Subject: Re: TLS for peering (was RE: Comments
> > > (RE:[speermint]requirementsdraft submitted))
> > >
> > >
> > > On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > > > how do you propose to establish a trust relationship between SIP
> > > > domain
> > > > boundaries without TLS?
> > >
> > > There are other methods: security lower in the stack, private
> > > connections, address ACLs...  the things that providers use today.
> > >
> > > > If you cannot establish a trust relationship between
> interconnecting
> > > > partners then what is the purpose of this whole exercise?
> > >
> > > To answer the second part of that sentence: I don't know; I've been
> > > asking that question for quite some time.
> > >
> > > -andy
> >
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint


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Richard,

If you are so confident that TLS will become the standard for peering,
there is nothing to fear, just say that it should be used in the absence
of other methods.=20

If your prediction is correct, providers will actually require and use
it, vendors will implement TLS, and it will become ubiquitous. =20

Shoving down the throats of people an operational requirement that will
increase the cost of a deployment is not the way to go. IETF is about
running code and consensus.

More comments inline...

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Wednesday, June 28, 2006 2:36 PM
> To: Reinaldo Penno; 'Andrew Newton'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Reinaldo Penno [mailto:rpenno@juniper.net]
> > Sent: Wednesday, June 28, 2006 4:17 PM
> > To: richard@shockey.us; Andrew Newton
> > Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > Hello Richard,
> >
> > I guess you put the answer in the email itself. Adam and others
talked
> > about cost vs. benefit.
> >
> > Speaking out of experience, these 'e-commerce' sites you mention
have
> > stacks and stacks of SSL accelerators and/or offpath SSL decryption
> > engines. It definitely increases the equipment cost, management,
real
> > state and operation of the solution.
>=20
> Like duh? So what?
>=20
[[Reinaldo]]=20

I'm sorry, but you first argument was quite simple and my answer was
pointing out that e-commerce sites use special hardware.=20


> There is real money involved here. Are you saying in inter carrier SIP
> communications, transaction security, fraud avoidance and UA identity
> management/control across domain boundaries are not worth the security
> cost?

It is not for you and me to decide. We should provide the necessary
alternatives and the market will lean one way or another.=20

>=20
> Voice communications alone is a 300 Billion US business world wide?
What's
> a
> few CPU cycles compared to that.
>=20
>=20
> > Also, if you have looked long enough on RFCs, many, many things that
are
> > mandated never see the light of the day because they lack consensus,
are
> > complex, never factored service provider's opinion, etc, etc.
>=20
> Been there done that. You could say that for about 60% of the Security
> Area
> RFC's, IPv6, DNSSEC, IDN, PKI, Public ENUM in e164.arpa, etal
>=20
> >
> > Layer 3 peering has been going on for decades without IPsec.
>=20
>=20
> Yes but that has been for best efforts IP traffic.
>=20
> The entire premise of speermint at its core is the supposition that
though
> all bits are created equal on the Internet it seems that some SIP bits
are
> more equal than others.
>=20
> If you don't believe that, this WG has no reason for existence.
>=20
[[Reinaldo]]=20

That's a logical fallacy. In the end SIP is just a protocol like HTTP,
DNS and others that run over IP. And what _actually_ needs strict QoS is
_media_ and not SIP signaling.=20

Are you going to suggest secure RTP as well between every single call on
the planet?=20

> Since some traffic between SIP interconnection partners ( notice I
never
> say
> peering ) may involve PSTN termination charges or use bandwidth in
high
> volume priority access mode, there must be a reasonable level of trust
> between interconnecting partners. There has to be the capability of
the
> terminating service provider trusting the first who the originating
> provider
> is and second trusting the calling party identity provided to them for
> billing purposes if nothing else. Therefore the security methodology
> mandated is critical.
>=20
>=20
> >
> > I believe we are not saying TLS is evil. We are saying 'MUST' is not
the
> > right word.
>=20
> Well again 3261 said MUST for a very very good reason. I would still
> require
> MUST support TLS in any SPEERMENT standard or BCP. However I accept
that
> other trust/security methodologies can work but they had better be
well
> documented along with the possible security threats associated with
each
> proposed method.
>=20
> >
> > Regards,
> >
> > Reinaldo
> >
> >
> > > -----Original Message-----
> > > From: Richard Shockey [mailto:richard@shockey.us]
> > > Sent: Wednesday, June 28, 2006 12:41 PM
> > > To: 'Andrew Newton'
> > > Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> > > Subject: RE: TLS for peering (was RE:
> > > Comments(RE:[speermint]requirementsdraft submitted))
> > >
> > >
> > >
> > > Well the one thing we all seem to agree on is that without some
trust
> > > relationship between interconnecting partners at their domain
> > boundaries
> > > this whole exercise is a vast waste of time and electrons.
> > >
> > > Now then I think Keith has a point.
> > >
> > > "If the former, then by all means specify TLS. If the latter, then
it
> > > should
> > > probably identify that security solutions are required, and then
make
> > sure
> > > that sufficient guidance is given on the mechanisms currently
> > available,
> > > including TLS, but probably also addressing IPsec."
> > >
> > > Required ..I's say security solutions are mandated as in MUST.
> > >
> > > There _are_ certainly more than one mechanism, besides TLS that
can be
> > > used
> > > to establish security & trust at the boundaries. So ..those
techniques
> > > MUST
> > > be deliberately specified along with the risks associated with
them.
> > >
> > > That said there are many of us that are getting very very tired of
the
> > > constant whining about this or that aspect of 3261 that for one
reason
> > or
> > > another some developer or service provider doesn't like. Some of
us
> > just
> > > went through 6 months of this discussion to reach consensus on the
> > > SIPConnect with TLS support since it was the obvious choice.
> > >
> > > TLS support IS mandated in 3261 so IMHO get with the program.
> > >
> > > I've yet to see real concrete data on why its such a problem to
> > implement
> > > or
> > > support in high volume SIP transaction systems. Lots of http based
> > > Ecommerce
> > > seems to rely on per transaction secure connections so I just
don't
> > get
> > > what
> > > the problem is here.
> > >
> > > > -----Original Message-----
> > > > From: Andrew Newton [mailto:andy@hxr.us]
> > > > Sent: Wednesday, June 28, 2006 3:06 PM
> > > > To: richard@shockey.us
> > > > Cc: 'Khan, Sohel Q [CTO]'; 'Daryl Malas'; 'Uzelac, Adam';
> > > > speermint@ietf.org
> > > > Subject: Re: TLS for peering (was RE: Comments
> > > > (RE:[speermint]requirementsdraft submitted))
> > > >
> > > >
> > > > On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > > > > how do you propose to establish a trust relationship between
SIP
> > > > > domain
> > > > > boundaries without TLS?
> > > >
> > > > There are other methods: security lower in the stack, private
> > > > connections, address ACLs...  the things that providers use
today.
> > > >
> > > > > If you cannot establish a trust relationship between
> > interconnecting
> > > > > partners then what is the purpose of this whole exercise?
> > > >
> > > > To answer the second part of that sentence: I don't know; I've
been
> > > > asking that question for quite some time.
> > > >
> > > > -andy
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint

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Errors-To: speermint-bounces@ietf.org

I agree with Richard's wording below.=20

"...any proposed BCP in SPEERMINT for intercarrier interconnection using
SIP MUST support a trust/security model....you MUST support TLS as one
of the supported models." =20

I also believe that a trust/security model must be in place, and I have
written as much in this forum. =20

Adam

-----Original Message-----
From: Richard Shockey [mailto:richard@shockey.us]=20
Sent: Wednesday, June 28, 2006 5:49 PM
To: 'Daryl Malas'
Cc: 'Khan, Sohel Q [CTO]'; Uzelac, Adam; speermint@ietf.org
Subject: RE: TLS for peering (was RE:
Comments(RE:[speermint]requirementsdraft submitted))



> -----Original Message-----
> From: Daryl Malas [mailto:daryl@level3.net]
> Sent: Wednesday, June 28, 2006 5:24 PM
> To: richard@shockey.us
> Cc: 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'; speermint@ietf.org
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
> ...I simply said REQUIRING it would be unrealistic...I didn't say=20
> including it as an optional method is unrealistic.  There are multiple

> ways to authenticate peers as described in Penno's message flows
draft.


Ah .. you are heading into a semantic trap here.=20

My position is any proposed BCP in SPEERMINT for intercarrier
interconnection ( did I say peering here :-) using SIP MUST support a
trust/security model.  I'm not saying you should be REQUIRED to use TLS
at all times but only that you MUST support TLS as one of the supported
models.

It is my opinion based on many years of personal experience in the IETF
that you that you will not get anything through the IESG to an RFC
unless something like the above position is incorporated into the
drafts.



>=20
> On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> >
> > TLS is a core part of the SIP specification. Saying its not=20
> > realistic is just plain bovine effluent.
> >
> > You are not going to interconnect at the most basic level unless=20
> > there
> is a
> > level of trust between the domains. So how do you plan on=20
> > authenticating
> the
> > domain?
> >
> > This was the absolute conclusion of the SIP Forum when we came up=20
> > with
> the
> > SIP Connect specification for PBX to SP interconnection.
> >
> > SP to SP interconnection is not that much different.
> >
> > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> IP_PBX_SP_Interop-
> > sibley-final.pdf
> >
> > <sigh>
> >
> > > -----Original Message-----
> > > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > > Sent: Wednesday, June 28, 2006 12:06 PM
> > > To: Daryl Malas; Uzelac, Adam
> > > Cc: speermint@ietf.org
> > > Subject: RE: TLS for peering (was RE: Comments=20
> > > (RE:[speermint]requirementsdraft submitted))
> > >
> > > So, what do u recommended ?
> > > Keep TLS optional?
> > > Any other method?
> > >
> > > Sohel
> > >
> > >
> > > -----Original Message-----
> > > From: "Daryl Malas" <daryl@level3.net>
> > > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > > Sent: 6/28/06 10:29 AM
> > > Subject: RE: TLS for peering (was RE: Comments (RE:
> > > [speermint]requirementsdraft submitted))
> > >
> > > I agree with Adam's assessment.  Requiring TLS is simply not=20
> > > realistic at this point.  I believe it has the potential of=20
> > > becoming a more
> common
> > > practice in the future, but not right now.
> > >
> > > --Daryl
> > >
> > > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > > TLS is becoming more and more a subject when the topic of SIP=20
> > > > interconnections arises for me professionally.  This shows=20
> > > > growing momentum indeed.  With that said, I can not agree with=20
> > > > TLS as a steadfast requirement.  Economics and scale are my
major concerns.
> TLS
> > > > adds cost, server load, and there are fail-over issues to=20
> > > > consider
> as
> > > > well.  For the person that designs an interconnect, I believe
> weighing
> > > > the cost against the benefit of the different models a useful
> exercise.
> > > > Mandating TLS, even if all parties involved in a peering=20
> > > > arrangement feel somehow "safe and secure" due to underlying=20
> > > > security mechanisms (L2/L3VPNs, etc), seems a little harsh.
> > > >
> > > > Adam
> > > >
> > > > -----Original Message-----
> > > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > > To: Reinaldo Penno
> > > > Cc: speermint@ietf.org
> > > > Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]=20
> > > > requirementsdraft submitted))
> > > >
> > > > I am not a provider yet, but I think we dont need to mess with
> TLS/TCP.
> > > > As I understand one of the goals is to prevent the calls from
> failing
> > > > due to "peering only" reasons. This means that things like=20
> > > > codecs
> which
> > > > are not negotiated by the peers need to be mandated at a minimal
> level.
> > > > TCP/TLS can be negotiated between the peers themselves so it is=20
> > > > a non-issue. Also as I still understand doing a high volume=20
> > > > calls with
> TLS
> > > > is still expensive. Mandating TLS in cases where the caller or
> callee
> > > > dont require it doesnt seem to add any value.
> > > >
> > > > -Medhavi.
> > > >
> > > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > > >
> > > > > TLS for _all_ scenarios? Even in private peerings, physically
> secure
> > > > > peerings, etc?
> > > > >
> > > > > Isn't that the same as requiring IPsec for all layer 3
peering?
> But
> > > > > since Layer 5 peering might be more distributed, the number of

> > > > > TLS sessions would be huge.
> > > > >
> > > > > It would be good to hear the provider's opinion on this...
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Reinaldo
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > > Cc: speermint@ietf.org
> > > > > > Subject: RE: Comments (RE: [speermint] requirements draft
> submitted)
> > > > > >
> > > > > > >
> > > > > > > "3.4.  Security requirements
> > > > > > >
> > > > > > >    All SIP messages MUST be sent over TLS [RFC3546] to
provide
> > > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > > >
> > > > > > > Giving the fact that we can have security through physical
> > > > > > segregation,
> > > > > > > intra-federation/provider calls, etc, the "MUST" here=20
> > > > > > > seems
> too
> > > > > > > generic.
> > > > > >
> > > > > > I tend to differ on this: it is not because you have lower=20
> > > > > > layer security (presumably hop by hop security associations)

> > > > > > that you should relax the end-to-end transport security
requirements.
> > > > > > Rohan's well written draft is also in favor of mandatory=20
> > > > > > support
> of
> > > > > TLS,
> > > > > > and so is the SIP Forum's IP PBX to VSP (here is a working
link:
> > > > > >
> > > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> IP_PBX_SP_In
> > > > > te
> > > > > > rop-sibley-final.pdf).
> > > > > > May be the requirement should be rewritten to put the MUST=20
> > > > > > on
> SIP
> > > > > > entities involved in L5 peering.
> > > > > >
> > > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> heartbeat.
> > > > > >
> > > > > > Brian
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint


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On Jun 28, 2006, at 5:35 PM, Richard Shockey wrote:

> Like duh? So what?
>
> There is real money involved here.

There's real money involved in many aspects of Internet operations.   
Life would certainly be better if the gTLD providers did the right  
thing by offering secure protocols too, but you don't see them  
stampeding to do so.

The other thing I don't like about the magical TLS pixie dust is that  
it does not take into account the fact that there is no root trust  
anchor.  Some CAs will sell anybody a cert for $25, some give them  
away for FREE.  What's the point of all that additional TLS  
infrastructure if it can be easily circumvented by a trust path that  
leads back to a money order purchased at a gas station.

-andy
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<HTML><BODY style=3D"word-wrap: break-word; -khtml-nbsp-mode: space; -kht=
ml-line-break: after-white-space; "><BR><DIV><DIV>On Jun 28, 2006, at 5:3=
5 PM, Richard Shockey wrote:</DIV><BR class=3D"Apple-interchange-newline"=
><BLOCKQUOTE type=3D"cite"><P style=3D"margin: 0.0px 0.0px 0.0px 0.0px"><=
FONT face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica">Like =
duh? So what?</FONT></P> <P style=3D"margin: 0.0px 0.0px 0.0px 0.0px; fon=
t: 12.0px Helvetica; min-height: 14.0px"><BR></P> <P style=3D"margin: 0.0=
px 0.0px 0.0px 0.0px"><FONT face=3D"Helvetica" size=3D"3" style=3D"font: =
12.0px Helvetica">There is real money involved here.<SPAN class=3D"Apple-=
converted-space">=A0</SPAN></FONT></P> </BLOCKQUOTE></DIV><BR><DIV>There'=
s real money involved in many aspects of Internet operations.=A0 Life wou=
ld certainly be better if the gTLD providers did the right thing by offer=
ing secure protocols too, but you don't see them stampeding to do so.</DI=
V><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>The other thing I=
 don't like about the magical TLS pixie dust is that it does not take int=
o account the fact that there is no root trust anchor.=A0 Some CAs will s=
ell anybody a cert for $25, some give them away for FREE.=A0 What's the p=
oint of all that additional TLS infrastructure if it can be easily circum=
vented by a trust path that leads back to a money order purchased at a ga=
s station.</DIV><DIV><BR class=3D"khtml-block-placeholder"></DIV><DIV>-an=
dy</DIV></BODY></HTML>
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From: "Richard Shockey" <richard@shockey.us>
To: "'Daryl Malas'" <daryl@level3.net>
Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
	submitted))
Date: Wed, 28 Jun 2006 17:49:01 -0400
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> -----Original Message-----
> From: Daryl Malas [mailto:daryl@level3.net]
> Sent: Wednesday, June 28, 2006 5:24 PM
> To: richard@shockey.us
> Cc: 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'; speermint@ietf.org
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> ...I simply said REQUIRING it would be unrealistic...I didn't say
> including it as an optional method is unrealistic.  There are multiple
> ways to authenticate peers as described in Penno's message flows draft.


Ah .. you are heading into a semantic trap here. 

My position is any proposed BCP in SPEERMINT for intercarrier
interconnection ( did I say peering here :-) using SIP MUST support a
trust/security model.  I'm not saying you should be REQUIRED to use TLS at
all times but only that you MUST support TLS as one of the supported models.

It is my opinion based on many years of personal experience in the IETF that
you that you will not get anything through the IESG to an RFC unless
something like the above position is incorporated into the drafts.



> 
> On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> >
> > TLS is a core part of the SIP specification. Saying its not realistic is
> > just plain bovine effluent.
> >
> > You are not going to interconnect at the most basic level unless there
> is a
> > level of trust between the domains. So how do you plan on authenticating
> the
> > domain?
> >
> > This was the absolute conclusion of the SIP Forum when we came up with
> the
> > SIP Connect specification for PBX to SP interconnection.
> >
> > SP to SP interconnection is not that much different.
> >
> > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> IP_PBX_SP_Interop-
> > sibley-final.pdf
> >
> > <sigh>
> >
> > > -----Original Message-----
> > > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > > Sent: Wednesday, June 28, 2006 12:06 PM
> > > To: Daryl Malas; Uzelac, Adam
> > > Cc: speermint@ietf.org
> > > Subject: RE: TLS for peering (was RE: Comments
> > > (RE:[speermint]requirementsdraft submitted))
> > >
> > > So, what do u recommended ?
> > > Keep TLS optional?
> > > Any other method?
> > >
> > > Sohel
> > >
> > >
> > > -----Original Message-----
> > > From: "Daryl Malas" <daryl@level3.net>
> > > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > > Sent: 6/28/06 10:29 AM
> > > Subject: RE: TLS for peering (was RE: Comments (RE:
> > > [speermint]requirementsdraft submitted))
> > >
> > > I agree with Adam's assessment.  Requiring TLS is simply not realistic
> > > at this point.  I believe it has the potential of becoming a more
> common
> > > practice in the future, but not right now.
> > >
> > > --Daryl
> > >
> > > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > > TLS is becoming more and more a subject when the topic of SIP
> > > > interconnections arises for me professionally.  This shows growing
> > > > momentum indeed.  With that said, I can not agree with TLS as a
> > > > steadfast requirement.  Economics and scale are my major concerns.
> TLS
> > > > adds cost, server load, and there are fail-over issues to consider
> as
> > > > well.  For the person that designs an interconnect, I believe
> weighing
> > > > the cost against the benefit of the different models a useful
> exercise.
> > > > Mandating TLS, even if all parties involved in a peering arrangement
> > > > feel somehow "safe and secure" due to underlying security mechanisms
> > > > (L2/L3VPNs, etc), seems a little harsh.
> > > >
> > > > Adam
> > > >
> > > > -----Original Message-----
> > > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > > To: Reinaldo Penno
> > > > Cc: speermint@ietf.org
> > > > Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> > > > requirementsdraft submitted))
> > > >
> > > > I am not a provider yet, but I think we dont need to mess with
> TLS/TCP.
> > > > As I understand one of the goals is to prevent the calls from
> failing
> > > > due to "peering only" reasons. This means that things like codecs
> which
> > > > are not negotiated by the peers need to be mandated at a minimal
> level.
> > > > TCP/TLS can be negotiated between the peers themselves so it is a
> > > > non-issue. Also as I still understand doing a high volume calls with
> TLS
> > > > is still expensive. Mandating TLS in cases where the caller or
> callee
> > > > dont require it doesnt seem to add any value.
> > > >
> > > > -Medhavi.
> > > >
> > > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > > >
> > > > > TLS for _all_ scenarios? Even in private peerings, physically
> secure
> > > > > peerings, etc?
> > > > >
> > > > > Isn't that the same as requiring IPsec for all layer 3 peering?
> But
> > > > > since Layer 5 peering might be more distributed, the number of TLS
> > > > > sessions would be huge.
> > > > >
> > > > > It would be good to hear the provider's opinion on this...
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Reinaldo
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > > Cc: speermint@ietf.org
> > > > > > Subject: RE: Comments (RE: [speermint] requirements draft
> submitted)
> > > > > >
> > > > > > >
> > > > > > > "3.4.  Security requirements
> > > > > > >
> > > > > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > > >
> > > > > > > Giving the fact that we can have security through physical
> > > > > > segregation,
> > > > > > > intra-federation/provider calls, etc, the "MUST" here seems
> too
> > > > > > > generic.
> > > > > >
> > > > > > I tend to differ on this: it is not because you have lower layer
> > > > > > security (presumably hop by hop security associations) that you
> > > > > > should relax the end-to-end transport security requirements.
> > > > > > Rohan's well written draft is also in favor of mandatory support
> of
> > > > > TLS,
> > > > > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > > > > >
> > > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> IP_PBX_SP_In
> > > > > te
> > > > > > rop-sibley-final.pdf).
> > > > > > May be the requirement should be rewritten to put the MUST on
> SIP
> > > > > > entities involved in L5 peering.
> > > > > >
> > > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> heartbeat.
> > > > > >
> > > > > > Brian
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> > >
> > >
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint


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From speermint-bounces@ietf.org Wed Jun 28 18:19:07 2006
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Subject: note: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
To: <richard@shockey.us>,
	"Daryl Malas" <daryl@level3.net>
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Thanks everyone for coming to the fair today.  Please step away from
your keyboards and go home for the evening and have a nice, cold beer.
:-)  I'm in Belgium at the moment, so there's no shortage of that.

Let's resume again tomorrow and reduce the tone just a few notches
please.

There are valid points on **both** sides of the issue that can be
discussed on the list and, perhaps best in person in Montreal.  Both
sides ought to duly consider the dispassionate arguements.

We will need to keep in mind that for now, we are focusing on the bare
minimum set of requirements to get VoIP peering working (as a first use
case).  There will surely be issues that come up, perhaps like this one,
in a BCP or later/other document that says something like "oh, by the
way, if you want to protect X/Y/Z or whatever, you really out to
implement A/B/C."  So I think there are ways to satisfy both sides of
the issue here and this is not an all or nothing scenario.  We can all
be right and all feel the love. ;-)

It'd be helpful to discuss this more but maybe for now we can table this
for 24 hours or something like that.

Jason

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]=20
> Sent: Wednesday, June 28, 2006 5:49 PM
> To: 'Daryl Malas'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:=20
> Comments(RE:[speermint]requirementsdraftsubmitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Daryl Malas [mailto:daryl@level3.net]
> > Sent: Wednesday, June 28, 2006 5:24 PM
> > To: richard@shockey.us
> > Cc: 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'; speermint@ietf.org
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >=20
> > ...I simply said REQUIRING it would be unrealistic...I didn't say=20
> > including it as an optional method is unrealistic.  There=20
> are multiple=20
> > ways to authenticate peers as described in Penno's message=20
> flows draft.
>=20
>=20
> Ah .. you are heading into a semantic trap here.=20
>=20
> My position is any proposed BCP in SPEERMINT for intercarrier=20
> interconnection ( did I say peering here :-) using SIP MUST=20
> support a trust/security model.  I'm not saying you should be=20
> REQUIRED to use TLS at all times but only that you MUST=20
> support TLS as one of the supported models.
>=20
> It is my opinion based on many years of personal experience=20
> in the IETF that you that you will not get anything through=20
> the IESG to an RFC unless something like the above position=20
> is incorporated into the drafts.
>=20
>=20
>=20
> >=20
> > On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> > >
> > > TLS is a core part of the SIP specification. Saying its not=20
> > > realistic is just plain bovine effluent.
> > >
> > > You are not going to interconnect at the most basic level unless=20
> > > there
> > is a
> > > level of trust between the domains. So how do you plan on=20
> > > authenticating
> > the
> > > domain?
> > >
> > > This was the absolute conclusion of the SIP Forum when we came up=20
> > > with
> > the
> > > SIP Connect specification for PBX to SP interconnection.
> > >
> > > SP to SP interconnection is not that much different.
> > >
> > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_Interop-
> > > sibley-final.pdf
> > >
> > > <sigh>
> > >
> > > > -----Original Message-----
> > > > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > > > Sent: Wednesday, June 28, 2006 12:06 PM
> > > > To: Daryl Malas; Uzelac, Adam
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: TLS for peering (was RE: Comments=20
> > > > (RE:[speermint]requirementsdraft submitted))
> > > >
> > > > So, what do u recommended ?
> > > > Keep TLS optional?
> > > > Any other method?
> > > >
> > > > Sohel
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: "Daryl Malas" <daryl@level3.net>
> > > > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > > > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > > > Sent: 6/28/06 10:29 AM
> > > > Subject: RE: TLS for peering (was RE: Comments (RE:
> > > > [speermint]requirementsdraft submitted))
> > > >
> > > > I agree with Adam's assessment.  Requiring TLS is simply not=20
> > > > realistic at this point.  I believe it has the potential of=20
> > > > becoming a more
> > common
> > > > practice in the future, but not right now.
> > > >
> > > > --Daryl
> > > >
> > > > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > > > TLS is becoming more and more a subject when the topic of SIP=20
> > > > > interconnections arises for me professionally.  This shows=20
> > > > > growing momentum indeed.  With that said, I can not=20
> agree with=20
> > > > > TLS as a steadfast requirement.  Economics and scale=20
> are my major concerns.
> > TLS
> > > > > adds cost, server load, and there are fail-over issues to=20
> > > > > consider
> > as
> > > > > well.  For the person that designs an interconnect, I believe
> > weighing
> > > > > the cost against the benefit of the different models a useful
> > exercise.
> > > > > Mandating TLS, even if all parties involved in a peering=20
> > > > > arrangement feel somehow "safe and secure" due to underlying=20
> > > > > security mechanisms (L2/L3VPNs, etc), seems a little harsh.
> > > > >
> > > > > Adam
> > > > >
> > > > > -----Original Message-----
> > > > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > > > To: Reinaldo Penno
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: TLS for peering (was RE: Comments (RE:=20
> [speermint]=20
> > > > > requirementsdraft submitted))
> > > > >
> > > > > I am not a provider yet, but I think we dont need to mess with
> > TLS/TCP.
> > > > > As I understand one of the goals is to prevent the calls from
> > failing
> > > > > due to "peering only" reasons. This means that things like=20
> > > > > codecs
> > which
> > > > > are not negotiated by the peers need to be mandated=20
> at a minimal
> > level.
> > > > > TCP/TLS can be negotiated between the peers=20
> themselves so it is=20
> > > > > a non-issue. Also as I still understand doing a high volume=20
> > > > > calls with
> > TLS
> > > > > is still expensive. Mandating TLS in cases where the caller or
> > callee
> > > > > dont require it doesnt seem to add any value.
> > > > >
> > > > > -Medhavi.
> > > > >
> > > > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > > > >
> > > > > > TLS for _all_ scenarios? Even in private peerings,=20
> physically
> > secure
> > > > > > peerings, etc?
> > > > > >
> > > > > > Isn't that the same as requiring IPsec for all=20
> layer 3 peering?
> > But
> > > > > > since Layer 5 peering might be more distributed,=20
> the number of=20
> > > > > > TLS sessions would be huge.
> > > > > >
> > > > > > It would be good to hear the provider's opinion on this...
> > > > > >
> > > > > > Thanks,
> > > > > >
> > > > > > Reinaldo
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > > > Cc: speermint@ietf.org
> > > > > > > Subject: RE: Comments (RE: [speermint] requirements draft
> > submitted)
> > > > > > >
> > > > > > > >
> > > > > > > > "3.4.  Security requirements
> > > > > > > >
> > > > > > > >    All SIP messages MUST be sent over TLS=20
> [RFC3546] to provide
> > > > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > > > >
> > > > > > > > Giving the fact that we can have security=20
> through physical
> > > > > > > segregation,
> > > > > > > > intra-federation/provider calls, etc, the "MUST" here=20
> > > > > > > > seems
> > too
> > > > > > > > generic.
> > > > > > >
> > > > > > > I tend to differ on this: it is not because you=20
> have lower=20
> > > > > > > layer security (presumably hop by hop security=20
> associations)=20
> > > > > > > that you should relax the end-to-end transport=20
> security requirements.
> > > > > > > Rohan's well written draft is also in favor of mandatory=20
> > > > > > > support
> > of
> > > > > > TLS,
> > > > > > > and so is the SIP Forum's IP PBX to VSP (here is=20
> a working link:
> > > > > > >
> > > > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_In
> > > > > > te
> > > > > > > rop-sibley-final.pdf).
> > > > > > > May be the requirement should be rewritten to put=20
> the MUST=20
> > > > > > > on
> > SIP
> > > > > > > entities involved in L5 peering.
> > > > > > >
> > > > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> > heartbeat.
> > > > > > >
> > > > > > > Brian
> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>=20

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Subject: RE: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
Date: Wed, 28 Jun 2006 18:20:40 -0400
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Thread-Topic: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
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From: "Reinaldo Penno" <rpenno@juniper.net>
To: <richard@shockey.us>,
	"Daryl Malas" <daryl@level3.net>
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	"Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
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The following security 'models' come to mind:

1 - TLS
2 - IPSec
3 - Physical security
4 - Access lists
5 - Registrations=20
6 - others?=20

TLS just like the other methods has pros and cons that should be
weighted by the persons that will actually use it.

I hope we can agree with that.

Regards,

Reinaldo

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Wednesday, June 28, 2006 2:49 PM
> To: 'Daryl Malas'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Daryl Malas [mailto:daryl@level3.net]
> > Sent: Wednesday, June 28, 2006 5:24 PM
> > To: richard@shockey.us
> > Cc: 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'; speermint@ietf.org
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > ...I simply said REQUIRING it would be unrealistic...I didn't say
> > including it as an optional method is unrealistic.  There are
multiple
> > ways to authenticate peers as described in Penno's message flows
draft.
>=20
>=20
> Ah .. you are heading into a semantic trap here.
>=20
> My position is any proposed BCP in SPEERMINT for intercarrier
> interconnection ( did I say peering here :-) using SIP MUST support a
> trust/security model.  I'm not saying you should be REQUIRED to use
TLS at
> all times but only that you MUST support TLS as one of the supported
> models.
>=20
> It is my opinion based on many years of personal experience in the
IETF
> that
> you that you will not get anything through the IESG to an RFC unless
> something like the above position is incorporated into the drafts.
>=20
>=20
>=20
> >
> > On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> > >
> > > TLS is a core part of the SIP specification. Saying its not
realistic
> is
> > > just plain bovine effluent.
> > >
> > > You are not going to interconnect at the most basic level unless
there
> > is a
> > > level of trust between the domains. So how do you plan on
> authenticating
> > the
> > > domain?
> > >
> > > This was the absolute conclusion of the SIP Forum when we came up
with
> > the
> > > SIP Connect specification for PBX to SP interconnection.
> > >
> > > SP to SP interconnection is not that much different.
> > >
> > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_Interop-
> > > sibley-final.pdf
> > >
> > > <sigh>
> > >
> > > > -----Original Message-----
> > > > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > > > Sent: Wednesday, June 28, 2006 12:06 PM
> > > > To: Daryl Malas; Uzelac, Adam
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: TLS for peering (was RE: Comments
> > > > (RE:[speermint]requirementsdraft submitted))
> > > >
> > > > So, what do u recommended ?
> > > > Keep TLS optional?
> > > > Any other method?
> > > >
> > > > Sohel
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: "Daryl Malas" <daryl@level3.net>
> > > > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > > > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > > > Sent: 6/28/06 10:29 AM
> > > > Subject: RE: TLS for peering (was RE: Comments (RE:
> > > > [speermint]requirementsdraft submitted))
> > > >
> > > > I agree with Adam's assessment.  Requiring TLS is simply not
> realistic
> > > > at this point.  I believe it has the potential of becoming a
more
> > common
> > > > practice in the future, but not right now.
> > > >
> > > > --Daryl
> > > >
> > > > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > > > TLS is becoming more and more a subject when the topic of SIP
> > > > > interconnections arises for me professionally.  This shows
growing
> > > > > momentum indeed.  With that said, I can not agree with TLS as
a
> > > > > steadfast requirement.  Economics and scale are my major
concerns.
> > TLS
> > > > > adds cost, server load, and there are fail-over issues to
consider
> > as
> > > > > well.  For the person that designs an interconnect, I believe
> > weighing
> > > > > the cost against the benefit of the different models a useful
> > exercise.
> > > > > Mandating TLS, even if all parties involved in a peering
> arrangement
> > > > > feel somehow "safe and secure" due to underlying security
> mechanisms
> > > > > (L2/L3VPNs, etc), seems a little harsh.
> > > > >
> > > > > Adam
> > > > >
> > > > > -----Original Message-----
> > > > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > > > To: Reinaldo Penno
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: TLS for peering (was RE: Comments (RE:
[speermint]
> > > > > requirementsdraft submitted))
> > > > >
> > > > > I am not a provider yet, but I think we dont need to mess with
> > TLS/TCP.
> > > > > As I understand one of the goals is to prevent the calls from
> > failing
> > > > > due to "peering only" reasons. This means that things like
codecs
> > which
> > > > > are not negotiated by the peers need to be mandated at a
minimal
> > level.
> > > > > TCP/TLS can be negotiated between the peers themselves so it
is a
> > > > > non-issue. Also as I still understand doing a high volume
calls
> with
> > TLS
> > > > > is still expensive. Mandating TLS in cases where the caller or
> > callee
> > > > > dont require it doesnt seem to add any value.
> > > > >
> > > > > -Medhavi.
> > > > >
> > > > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > > > >
> > > > > > TLS for _all_ scenarios? Even in private peerings,
physically
> > secure
> > > > > > peerings, etc?
> > > > > >
> > > > > > Isn't that the same as requiring IPsec for all layer 3
peering?
> > But
> > > > > > since Layer 5 peering might be more distributed, the number
of
> TLS
> > > > > > sessions would be huge.
> > > > > >
> > > > > > It would be good to hear the provider's opinion on this...
> > > > > >
> > > > > > Thanks,
> > > > > >
> > > > > > Reinaldo
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > > > Cc: speermint@ietf.org
> > > > > > > Subject: RE: Comments (RE: [speermint] requirements draft
> > submitted)
> > > > > > >
> > > > > > > >
> > > > > > > > "3.4.  Security requirements
> > > > > > > >
> > > > > > > >    All SIP messages MUST be sent over TLS [RFC3546] to
> provide
> > > > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > > > >
> > > > > > > > Giving the fact that we can have security through
physical
> > > > > > > segregation,
> > > > > > > > intra-federation/provider calls, etc, the "MUST" here
seems
> > too
> > > > > > > > generic.
> > > > > > >
> > > > > > > I tend to differ on this: it is not because you have lower
> layer
> > > > > > > security (presumably hop by hop security associations)
that
> you
> > > > > > > should relax the end-to-end transport security
requirements.
> > > > > > > Rohan's well written draft is also in favor of mandatory
> support
> > of
> > > > > > TLS,
> > > > > > > and so is the SIP Forum's IP PBX to VSP (here is a working
> link:
> > > > > > >
> > > > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_In
> > > > > > te
> > > > > > > rop-sibley-final.pdf).
> > > > > > > May be the requirement should be rewritten to put the MUST
on
> > SIP
> > > > > > > entities involved in L5 peering.
> > > > > > >
> > > > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> > heartbeat.
> > > > > > >
> > > > > > > Brian
> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraft submitted))
From: Daryl Malas <daryl@level3.net>
To: richard@shockey.us
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I 100% agree with your statement below.  Thanks for clarifying...I
apparently stepped in the semantic trap...it's sticky I might add. ;-)

Thanks...--Daryl

On Wed, 2006-06-28 at 17:49 -0400, Richard Shockey wrote:
> 
> > -----Original Message-----
> > From: Daryl Malas [mailto:daryl@level3.net]
> > Sent: Wednesday, June 28, 2006 5:24 PM
> > To: richard@shockey.us
> > Cc: 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'; speermint@ietf.org
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> > 
> > ...I simply said REQUIRING it would be unrealistic...I didn't say
> > including it as an optional method is unrealistic.  There are multiple
> > ways to authenticate peers as described in Penno's message flows draft.
> 
> 
> Ah .. you are heading into a semantic trap here. 
> 
> My position is any proposed BCP in SPEERMINT for intercarrier
> interconnection ( did I say peering here :-) using SIP MUST support a
> trust/security model.  I'm not saying you should be REQUIRED to use TLS at
> all times but only that you MUST support TLS as one of the supported models.
> 
> It is my opinion based on many years of personal experience in the IETF that
> you that you will not get anything through the IESG to an RFC unless
> something like the above position is incorporated into the drafts.
> 
> 
> 
> > 
> > On Wed, 2006-06-28 at 14:28 -0400, Richard Shockey wrote:
> > >
> > > TLS is a core part of the SIP specification. Saying its not realistic is
> > > just plain bovine effluent.
> > >
> > > You are not going to interconnect at the most basic level unless there
> > is a
> > > level of trust between the domains. So how do you plan on authenticating
> > the
> > > domain?
> > >
> > > This was the absolute conclusion of the SIP Forum when we came up with
> > the
> > > SIP Connect specification for PBX to SP interconnection.
> > >
> > > SP to SP interconnection is not that much different.
> > >
> > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_Interop-
> > > sibley-final.pdf
> > >
> > > <sigh>
> > >
> > > > -----Original Message-----
> > > > From: Khan, Sohel Q [CTO] [mailto:Sohel.Q.Khan@sprint.com]
> > > > Sent: Wednesday, June 28, 2006 12:06 PM
> > > > To: Daryl Malas; Uzelac, Adam
> > > > Cc: speermint@ietf.org
> > > > Subject: RE: TLS for peering (was RE: Comments
> > > > (RE:[speermint]requirementsdraft submitted))
> > > >
> > > > So, what do u recommended ?
> > > > Keep TLS optional?
> > > > Any other method?
> > > >
> > > > Sohel
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: "Daryl Malas" <daryl@level3.net>
> > > > To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
> > > > Cc: "speermint@ietf.org" <speermint@ietf.org>
> > > > Sent: 6/28/06 10:29 AM
> > > > Subject: RE: TLS for peering (was RE: Comments (RE:
> > > > [speermint]requirementsdraft submitted))
> > > >
> > > > I agree with Adam's assessment.  Requiring TLS is simply not realistic
> > > > at this point.  I believe it has the potential of becoming a more
> > common
> > > > practice in the future, but not right now.
> > > >
> > > > --Daryl
> > > >
> > > > On Tue, 2006-06-27 at 21:47 -0400, Uzelac, Adam wrote:
> > > > > TLS is becoming more and more a subject when the topic of SIP
> > > > > interconnections arises for me professionally.  This shows growing
> > > > > momentum indeed.  With that said, I can not agree with TLS as a
> > > > > steadfast requirement.  Economics and scale are my major concerns.
> > TLS
> > > > > adds cost, server load, and there are fail-over issues to consider
> > as
> > > > > well.  For the person that designs an interconnect, I believe
> > weighing
> > > > > the cost against the benefit of the different models a useful
> > exercise.
> > > > > Mandating TLS, even if all parties involved in a peering arrangement
> > > > > feel somehow "safe and secure" due to underlying security mechanisms
> > > > > (L2/L3VPNs, etc), seems a little harsh.
> > > > >
> > > > > Adam
> > > > >
> > > > > -----Original Message-----
> > > > > From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> > > > > Sent: Tuesday, June 27, 2006 8:44 PM
> > > > > To: Reinaldo Penno
> > > > > Cc: speermint@ietf.org
> > > > > Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> > > > > requirementsdraft submitted))
> > > > >
> > > > > I am not a provider yet, but I think we dont need to mess with
> > TLS/TCP.
> > > > > As I understand one of the goals is to prevent the calls from
> > failing
> > > > > due to "peering only" reasons. This means that things like codecs
> > which
> > > > > are not negotiated by the peers need to be mandated at a minimal
> > level.
> > > > > TCP/TLS can be negotiated between the peers themselves so it is a
> > > > > non-issue. Also as I still understand doing a high volume calls with
> > TLS
> > > > > is still expensive. Mandating TLS in cases where the caller or
> > callee
> > > > > dont require it doesnt seem to add any value.
> > > > >
> > > > > -Medhavi.
> > > > >
> > > > > On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > > > >
> > > > > > TLS for _all_ scenarios? Even in private peerings, physically
> > secure
> > > > > > peerings, etc?
> > > > > >
> > > > > > Isn't that the same as requiring IPsec for all layer 3 peering?
> > But
> > > > > > since Layer 5 peering might be more distributed, the number of TLS
> > > > > > sessions would be huge.
> > > > > >
> > > > > > It would be good to hear the provider's opinion on this...
> > > > > >
> > > > > > Thanks,
> > > > > >
> > > > > > Reinaldo
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > > > > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > > > > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > > > > > Cc: speermint@ietf.org
> > > > > > > Subject: RE: Comments (RE: [speermint] requirements draft
> > submitted)
> > > > > > >
> > > > > > > >
> > > > > > > > "3.4.  Security requirements
> > > > > > > >
> > > > > > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > > > > > >    transport-layer security as defined in RFC 3261,"
> > > > > > > >
> > > > > > > > Giving the fact that we can have security through physical
> > > > > > > segregation,
> > > > > > > > intra-federation/provider calls, etc, the "MUST" here seems
> > too
> > > > > > > > generic.
> > > > > > >
> > > > > > > I tend to differ on this: it is not because you have lower layer
> > > > > > > security (presumably hop by hop security associations) that you
> > > > > > > should relax the end-to-end transport security requirements.
> > > > > > > Rohan's well written draft is also in favor of mandatory support
> > of
> > > > > > TLS,
> > > > > > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > > > > > >
> > > > > > http://data.memberclicks.com/bbattach/130808/sf-draft-twg-
> > IP_PBX_SP_In
> > > > > > te
> > > > > > > rop-sibley-final.pdf).
> > > > > > > May be the requirement should be rewritten to put the MUST on
> > SIP
> > > > > > > entities involved in L5 peering.
> > > > > > >
> > > > > > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
> > heartbeat.
> > > > > > >
> > > > > > > Brian
> > > > > >
> > > > > > _______________________________________________
> > > > > > Speermint mailing list
> > > > > > Speermint@ietf.org
> > > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint


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> -----Original Message-----
> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]
> Sent: Wednesday, June 28, 2006 6:00 PM
> To: richard@shockey.us; Daryl Malas
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> I agree with Richard's wording below.
> 
> "...any proposed BCP in SPEERMINT for intercarrier interconnection using
> SIP MUST support a trust/security model....you MUST support TLS as one
> of the supported models."
> 
> I also believe that a trust/security model must be in place, and I have
> written as much in this forum.


Thank you. This is my point exactly.


> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint


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> -----Original Message-----
> From: Reinaldo Penno [mailto:rpenno@juniper.net]
> Sent: Wednesday, June 28, 2006 6:02 PM
> To: richard@shockey.us; Andrew Newton
> Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> Richard,
> 
> If you are so confident that TLS will become the standard for peering,
> there is nothing to fear, just say that it should be used in the absence
> of other methods.


My point is that in SIP communications TLS IS a standard directly required
in 3261. That cannot be avoided..if you say you support SIP by definition
you MUST support TLS.

> 
> If your prediction is correct, providers will actually require and use
> it, vendors will implement TLS, and it will become ubiquitous.

I did not say there were not other trust models only that you cannot side
step the requirement to support TLS.

> 
> Shoving down the throats of people an operational requirement that will
> increase the cost of a deployment is not the way to go. IETF is about
> running code and consensus.

So you choose to pick and choose what points in the SIP standard you want?
How do you propose to implement interoperability? You are the violation of
consensus. Follow the standard as agreed to by the SIP WG and approved by
the IETF otherwise you are in noncompliance.



> 
> More comments inline...


> > > >
> > > > _______________________________________________
> > > > Speermint mailing list
> > > > Speermint@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/speermint


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An excellent point Andy ..now we are making progress here.

 

But is this a discussion over what is the root CA authority between service
providers should it be?

 

"To be a CA or not to be a CA that is the question"  Bill Shakespeare 

 

That is a reasonable point of discussion on how or to whom anchors the trust
between the domain boundaries. Which is my ultimate point.

 

All other discussions are moot points unless the trust model is clear.

 

My personal view is that so long as two consenting service providers can
agree on the CA the boundary trust relationship can be established.

 

Why should SIP interconnection be any different from traditional network
interconnection which is a matrix of bi-lateral agreements?  Look at the
GSM? 600 members and 32000 bilateral agreements covering each and every
detail including neutral settlement clearing houses.

 

My point is that TLS MUST be supported and it is up to service providers to
decide who they mutually agree to be their trusted CA.

 

 

 

 

 

  _____  

From: Andrew Newton [mailto:andy@hxr.us] 
Sent: Wednesday, June 28, 2006 6:02 PM
To: richard@shockey.us
Cc: 'Reinaldo Penno'; speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac,
Adam'
Subject: Re: TLS for peering (was RE:
Comments(RE:[speermint]requirementsdraft submitted))

 

 

On Jun 28, 2006, at 5:35 PM, Richard Shockey wrote:





Like duh? So what?

 

There is real money involved here. 

 

There's real money involved in many aspects of Internet operations. Life
would certainly be better if the gTLD providers did the right thing by
offering secure protocols too, but you don't see them stampeding to do so.

 

The other thing I don't like about the magical TLS pixie dust is that it
does not take into account the fact that there is no root trust anchor. Some
CAs will sell anybody a cert for $25, some give them away for FREE. What's
the point of all that additional TLS infrastructure if it can be easily
circumvented by a trust path that leads back to a money order purchased at a
gas station.

 

-andy


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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>But is this a discussion over what =
is the
root CA authority between service providers should it =
be?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
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10.0pt;font-family:Arial;color:navy'>&#8220;To be a CA or not to be a CA =
that
is the question&#8221; &nbsp;Bill Shakespeare =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>That is a reasonable point of =
discussion
on how or to whom anchors the trust between the domain boundaries. Which =
is my
ultimate point.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>All other discussions are moot =
points
unless the trust model is clear.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>My personal view is that so long as =
two
consenting service providers can agree on the CA the boundary trust
relationship can be established.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Why should SIP interconnection be =
any
different from traditional network interconnection which is a matrix of
bi-lateral agreements? &nbsp;Look at the GSM? 600 members and 32000 =
bilateral
agreements covering each and every detail including neutral settlement =
clearing
houses.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>My point is that TLS MUST be =
supported and
it is up to service providers to decide who they mutually agree to be =
their
trusted CA.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
Andrew Newton
[mailto:andy@hxr.us] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, June 28, =
2006
6:02 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> <st1:PersonName =
w:st=3D"on">richard@shockey.us</st1:PersonName><br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> 'Reinaldo Penno';
speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: TLS for =
peering (was
RE: Comments(RE:[speermint]requirementsdraft =
submitted))</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>On Jun 28, 2006, at 5:35 PM, Richard Shockey =
wrote:<o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><br>
<br>
<o:p></o:p></span></font></p>

<p style=3D'margin:0in;margin-bottom:.0001pt'><font size=3D1 =
face=3DHelvetica><span
style=3D'font-size:9.0pt;font-family:Helvetica'>Like duh? So =
what?</span></font><o:p></o:p></p>

<p style=3D'margin:0in;margin-bottom:.0001pt;min-height: 14.0px'><font =
size=3D1
face=3DHelvetica><span =
style=3D'font-size:9.0pt;font-family:Helvetica'><o:p>&nbsp;</o:p></span><=
/font></p>

<p style=3D'margin:0in;margin-bottom:.0001pt'><font size=3D1 =
face=3DHelvetica><span
style=3D'font-size:9.0pt;font-family:Helvetica'>There is real money =
involved
here.<span class=3Dapple-converted-space> =
</span></span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>There's real money involved in many aspects of Internet =
operations.
Life would certainly be better if the gTLD providers did the right thing =
by
offering secure protocols too, but you don't see them stampeding to do =
so.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>The other thing I don't like about the magical TLS pixie dust is =
that
it does not take into account the fact that there is no root trust =
anchor. Some
CAs will sell anybody a cert for $25, some give them away for FREE. =
What's the
point of all that additional TLS infrastructure if it can be easily
circumvented by a trust path that leads back to a money order purchased =
at a
gas station.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>-andy<o:p></o:p></span></font></p>

</div>

</div>

</div>

</body>

</html>

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From speermint-bounces@ietf.org Wed Jun 28 22:08:43 2006
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From: "Richard Shockey" <richard@shockey.us>
To: "'Reinaldo Penno'" <rpenno@juniper.net>, "'Daryl Malas'" <daryl@level3.net>
Subject: RE: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
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> -----Original Message-----
> From: Reinaldo Penno [mailto:rpenno@juniper.net]
> Sent: Wednesday, June 28, 2006 6:21 PM
> To: richard@shockey.us; Daryl Malas
> Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
> 
> The following security 'models' come to mind:
> 
> 1 - TLS
> 2 - IPSec
> 3 - Physical security
> 4 - Access lists
> 5 - Registrations
> 6 - others?
> 
> TLS just like the other methods has pros and cons that should be
> weighted by the persons that will actually use it.

But it cannot be deprecated. It MUST be supported. Its part of the core SIP
standard you cannot side step this issue. 

> 
> I hope we can agree with that.


This is exactly where we begin to make progress. Define the reasonable
security/trust models including TLS that we can find consensus on, the
smaller the list the better.

This is really the core problem in spearmint. What is the ultimate
trust/security model that service providers can rely on to accept traffic
across domain boundaries. 

Are you really BT or Sprint or FT or Cablevision etc. Or are you really IBM
or Ford or Bombardier or Sony for that matter. Can I bill you for your
traffic? How do I know you are who you say you are.



> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint


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From speermint-bounces@ietf.org Thu Jun 29 03:53:51 2006
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Subject: Re: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
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> I agree with Richard's wording below.
>
> "...any proposed BCP in SPEERMINT for intercarrier interconnection =
using
> SIP MUST support a trust/security model....you MUST support TLS as one
> of the supported models."
>
> I also believe that a trust/security model must be in place, and I =
have
> written as much in this forum.

Count me in too.
=20
I also want to add Rich's statements:
=20
>So you choose to pick and choose what points in the SIP standard you =
want?
>How do you propose to implement interoperability? You are the violation =
of
>consensus. Follow the standard as agreed to by the SIP WG and approved =
by
>the IETF otherwise you are in noncompliance.
=20
and

>My point is that TLS MUST be supported and it is up to service =
providers to decide who they mutually agree to be >their trusted CA.
=20
-sta


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From speermint-bounces@ietf.org Thu Jun 29 08:13:38 2006
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'richard@shockey.us'" <richard@shockey.us>, "'Reinaldo Penno'"
	<rpenno@juniper.net>, "'Daryl Malas'" <daryl@level3.net>
Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsd
	raftsubmitted))
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Just to be clear. RFC 3261 requires implementations to support TLS. To be conformant it must also response to a SIPS URI with the use of TLS, but presumably it can also reject a request for any reason it likes. 

That does not require it to use a SIPS URI on any outgoing request, and therefore use TLS on outgoing requests.

Therefore substantially the RFC 3261 requirement is to support TLS, not to use it. You seem to be verging on implying that RFC 3261 says it must also be used.

regards

Keith

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: 29 June 2006 03:06
> To: 'Reinaldo Penno'; 'Daryl Malas'
> Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
> 
> 
> 
> 
> > -----Original Message-----
> > From: Reinaldo Penno [mailto:rpenno@juniper.net]
> > Sent: Wednesday, June 28, 2006 6:21 PM
> > To: richard@shockey.us; Daryl Malas
> > Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraftsubmitted))
> > 
> > The following security 'models' come to mind:
> > 
> > 1 - TLS
> > 2 - IPSec
> > 3 - Physical security
> > 4 - Access lists
> > 5 - Registrations
> > 6 - others?
> > 
> > TLS just like the other methods has pros and cons that should be
> > weighted by the persons that will actually use it.
> 
> But it cannot be deprecated. It MUST be supported. Its part 
> of the core SIP
> standard you cannot side step this issue. 
> 
> > 
> > I hope we can agree with that.
> 
> 
> This is exactly where we begin to make progress. Define the reasonable
> security/trust models including TLS that we can find consensus on, the
> smaller the list the better.
> 
> This is really the core problem in spearmint. What is the ultimate
> trust/security model that service providers can rely on to 
> accept traffic
> across domain boundaries. 
> 
> Are you really BT or Sprint or FT or Cablevision etc. Or are 
> you really IBM
> or Ford or Bombardier or Sony for that matter. Can I bill you for your
> traffic? How do I know you are who you say you are.
> 
> 
> 
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint
> 
> 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
> 

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From: "Brian Rosen" <br@brianrosen.net>
To: "'Uzelac, Adam'" <Adam.Uzelac@globalcrossing.com>,
	"'Medhavi Bhatia'" <mbhatia@3clogic.com>,
	"'Reinaldo Penno'" <rpenno@juniper.net>
Subject: RE: TLS for peering (was RE: Comments (RE:
	[speermint]requirementsdraft submitted))
Date: Thu, 29 Jun 2006 10:09:15 -0400
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<apologies for sporadic replies, I'm traveling with intermittent access>
This us a PEERING system with federations.  The likelihood of a persistent
TLS connection being maintained is high, and the cost of establishing a new
one is relatively low, especially if, as on another thread, one of the
things a federation means is a CA signing certs for use by the federation.
The number of peers is going to be relatively low, improving the probability
of the persistent TLS connection being available.

Brian

-----Original Message-----
From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com] 
Sent: Tuesday, June 27, 2006 9:48 PM
To: Medhavi Bhatia; Reinaldo Penno
Cc: speermint@ietf.org
Subject: RE: TLS for peering (was RE: Comments (RE:
[speermint]requirementsdraft submitted))

TLS is becoming more and more a subject when the topic of SIP
interconnections arises for me professionally.  This shows growing
momentum indeed.  With that said, I can not agree with TLS as a
steadfast requirement.  Economics and scale are my major concerns.  TLS
adds cost, server load, and there are fail-over issues to consider as
well.  For the person that designs an interconnect, I believe weighing
the cost against the benefit of the different models a useful exercise.
Mandating TLS, even if all parties involved in a peering arrangement
feel somehow "safe and secure" due to underlying security mechanisms
(L2/L3VPNs, etc), seems a little harsh.  

Adam

-----Original Message-----
From: Medhavi Bhatia [mailto:mbhatia@3clogic.com] 
Sent: Tuesday, June 27, 2006 8:44 PM
To: Reinaldo Penno
Cc: speermint@ietf.org
Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
requirementsdraft submitted))

I am not a provider yet, but I think we dont need to mess with TLS/TCP.
As I understand one of the goals is to prevent the calls from failing
due to "peering only" reasons. This means that things like codecs which
are not negotiated by the peers need to be mandated at a minimal level.
TCP/TLS can be negotiated between the peers themselves so it is a
non-issue. Also as I still understand doing a high volume calls with TLS
is still expensive. Mandating TLS in cases where the caller or callee
dont require it doesnt seem to add any value.

-Medhavi.

On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
>
> TLS for _all_ scenarios? Even in private peerings, physically secure 
> peerings, etc?
>
> Isn't that the same as requiring IPsec for all layer 3 peering? But 
> since Layer 5 peering might be more distributed, the number of TLS 
> sessions would be huge.
>
> It would be good to hear the provider's opinion on this...
>
> Thanks,
>
> Reinaldo
>
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Tuesday, June 27, 2006 2:12 PM
> > To: 'Jean-Francois Mule'; Reinaldo Penno
> > Cc: speermint@ietf.org
> > Subject: RE: Comments (RE: [speermint] requirements draft submitted)
> >
> > >
> > > "3.4.  Security requirements
> > >
> > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > >    transport-layer security as defined in RFC 3261,"
> > >
> > > Giving the fact that we can have security through physical
> > segregation,
> > > intra-federation/provider calls, etc, the "MUST" here seems too 
> > > generic.
> >
> > I tend to differ on this: it is not because you have lower layer 
> > security (presumably hop by hop security associations) that you 
> > should relax the end-to-end transport security requirements.
> > Rohan's well written draft is also in favor of mandatory support of
> TLS,
> > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> >
> http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> te
> > rop-sibley-final.pdf).
> > May be the requirement should be rewritten to put the MUST on SIP 
> > entities involved in L5 peering.
> >
> > <brian>I'm in favor of requiring TLS.  I'd say MUST in a heartbeat.
> >
> > Brian
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>

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	"'Reinaldo Penno'" <rpenno@juniper.net>, "'Daryl Malas'" <daryl@level3.net>
Subject: RE: TLS for peering (was RE:
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> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: Thursday, June 29, 2006 8:13 AM
> To: 'richard@shockey.us'; 'Reinaldo Penno'; 'Daryl Malas'
> Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
> 
> Just to be clear. RFC 3261 requires implementations to support TLS. To be
> conformant it must also response to a SIPS URI with the use of TLS, but
> presumably it can also reject a request for any reason it likes.

OF course.

> 
> That does not require it to use a SIPS URI on any outgoing request, and
> therefore use TLS on outgoing requests.
> 
> Therefore substantially the RFC 3261 requirement is to support TLS, not to
> use it. You seem to be verging on implying that RFC 3261 says it must also
> be used.
> 
> 

Apologies is I was not clear. You MUST support TLS as one of multiple
security/trust options.


> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >


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This whole discussion leads back to the observation that many of these
type of agreements are aspects that ALL SPs may not come to agreement,
but some subset might have agreement.
=20
In other words, it is a "federation" policy issue.
=20
Mike
=20


________________________________

	From: Andrew Newton [mailto:andy@hxr.us]=20
	Sent: Wednesday, June 28, 2006 6:02 PM
	To: richard@shockey.us
	Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org; 'Uzelac, Adam'
	Subject: Re: TLS for peering (was RE:
Comments(RE:[speermint]requirementsdraftsubmitted))
=09
=09

	On Jun 28, 2006, at 5:35 PM, Richard Shockey wrote:


		Like duh? So what?

	=09
	=09

		There is real money involved here.=20


	There's real money involved in many aspects of Internet
operations.  Life would certainly be better if the gTLD providers did
the right thing by offering secure protocols too, but you don't see them
stampeding to do so.

	The other thing I don't like about the magical TLS pixie dust is
that it does not take into account the fact that there is no root trust
anchor.  Some CAs will sell anybody a cert for $25, some give them away
for FREE.  What's the point of all that additional TLS infrastructure if
it can be easily circumvented by a trust path that leads back to a money
order purchased at a gas station.

	-andy


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2914" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; khtml-nbsp-mode: space; =
khtml-line-break: after-white-space">
<DIV dir=3Dltr align=3Dleft><SPAN class=3D382574314-29062006><FONT =
face=3DCourier=20
color=3D#0000ff>This whole discussion leads back to the observation that =
many of=20
these type of agreements are aspects that ALL SPs may not come to =
agreement, but=20
some subset might have agreement.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D382574314-29062006><FONT =
face=3DCourier=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D382574314-29062006><FONT =
face=3DCourier=20
color=3D#0000ff>In other words, it is a "federation" policy=20
issue.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D382574314-29062006><FONT =
face=3DCourier=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D382574314-29062006><FONT =
face=3DCourier=20
color=3D#0000ff>Mike</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D382574314-29062006></SPAN>&nbsp;</DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Andrew Newton =
[mailto:andy@hxr.us]=20
  <BR><B>Sent:</B> Wednesday, June 28, 2006 6:02 PM<BR><B>To:</B>=20
  richard@shockey.us<BR><B>Cc:</B> 'Khan, Sohel Q [CTO]'; =
speermint@ietf.org;=20
  'Uzelac, Adam'<BR><B>Subject:</B> Re: TLS for peering (was RE:=20
  =
Comments(RE:[speermint]requirementsdraftsubmitted))<BR></FONT><BR></DIV>
  <DIV></DIV><BR>
  <DIV>
  <DIV>On Jun 28, 2006, at 5:35 PM, Richard Shockey wrote:</DIV><BR=20
  class=3DApple-interchange-newline>
  <BLOCKQUOTE type=3D"cite">
    <P style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica" =
face=3DHelvetica=20
    size=3D3>Like duh? So what?</FONT></P>
    <P style=3D"MIN-HEIGHT: 14px; MARGIN: 0px; FONT: 12px =
Helvetica"><BR></P>
    <P style=3D"MARGIN: 0px"><FONT style=3D"FONT: 12px Helvetica" =
face=3DHelvetica=20
    size=3D3>There is real money involved here.<SPAN=20
    =
class=3DApple-converted-space>&nbsp;</SPAN></FONT></P></BLOCKQUOTE></DIV>=
<BR>
  <DIV>There's real money involved in many aspects of Internet =
operations.&nbsp;=20
  Life would certainly be better if the gTLD providers did the right =
thing by=20
  offering secure protocols too, but you don't see them stampeding to do =

  so.</DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <DIV>The other thing I don't like about the magical TLS pixie dust is =
that it=20
  does not take into account the fact that there is no root trust =
anchor.&nbsp;=20
  Some CAs will sell anybody a cert for $25, some give them away for =
FREE.&nbsp;=20
  What's the point of all that additional TLS infrastructure if it can =
be easily=20
  circumvented by a trust path that leads back to a money order =
purchased at a=20
  gas station.</DIV>
  <DIV><BR class=3Dkhtml-block-placeholder></DIV>
  <DIV>-andy</DIV></BLOCKQUOTE></BODY></HTML>

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Richard Shockey wrote:
> My point is that TLS MUST be supported and it is up to service providers to
> decide who they mutually agree to be their trusted CA.

So a mechanism MUST be supported even though the bits to get it to work 
will not be specified?  Haven't you ever wondered why SMTPS has never 
succeeded in solving the spam problem?

What's wrong with just saying that service providers MUST mutually agree 
upon security and authorization mechanisms?

-andy


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From speermint-bounces@ietf.org Thu Jun 29 12:26:18 2006
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Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
	submitted))
Date: Thu, 29 Jun 2006 12:26:08 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
To: <richard@shockey.us>,
	"Andrew Newton" <andy@hxr.us>
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	"Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
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Richard,

Until very recently there was consensus around TLS: nobody implemented
it.

- "But hey, isn't it a MUST on the RFC?" one would ask. Yes, but there
was no interoperability problem at all. People just did not use it.

It took years for TLS to gain momentum and I can guarantee that it is
_not_ because the RFC 3261says it MUST, but because users saw security
advantages on it in certain deployments.

All I'm saying is that we must provide alternatives so that small ISPs
that do not have tons of money as you it, ISPs that do not care about
TLS and others can interoperate.=20

Regards,

Reinaldo


> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Wednesday, June 28, 2006 6:37 PM
> To: Reinaldo Penno; 'Andrew Newton'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Reinaldo Penno [mailto:rpenno@juniper.net]
> > Sent: Wednesday, June 28, 2006 6:02 PM
> > To: richard@shockey.us; Andrew Newton
> > Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > Richard,
> >
> > If you are so confident that TLS will become the standard for
peering,
> > there is nothing to fear, just say that it should be used in the
absence
> > of other methods.
>=20
>=20
> My point is that in SIP communications TLS IS a standard directly
required
> in 3261. That cannot be avoided..if you say you support SIP by
definition
> you MUST support TLS.
>=20
> >
> > If your prediction is correct, providers will actually require and
use
> > it, vendors will implement TLS, and it will become ubiquitous.
>=20
> I did not say there were not other trust models only that you cannot
side
> step the requirement to support TLS.
>=20
> >
> > Shoving down the throats of people an operational requirement that
will
> > increase the cost of a deployment is not the way to go. IETF is
about
> > running code and consensus.
>=20
> So you choose to pick and choose what points in the SIP standard you
want?
> How do you propose to implement interoperability? You are the
violation of
> consensus. Follow the standard as agreed to by the SIP WG and approved
by
> the IETF otherwise you are in noncompliance.
>=20
>=20
>=20
> >
> > More comments inline...
>=20
>=20
> > > > >
> > > > > _______________________________________________
> > > > > Speermint mailing list
> > > > > Speermint@ietf.org
> > > > > https://www1.ietf.org/mailman/listinfo/speermint

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From speermint-bounces@ietf.org Thu Jun 29 12:34:52 2006
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Subject: RE: TLS for peering (was RE: Comments (RE:
	[speermint]requirementsdraft submitted))
Date: Thu, 29 Jun 2006 12:34:48 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
To: "Brian Rosen" <br@brianrosen.net>,
	"Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>,
	"Medhavi Bhatia" <mbhatia@3clogic.com>
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Hummm....I'm not sure about that. It depends on many factors.

First and foremost it depends on which model you embrace: p2p/email or
federations. In the email model the number of TLS connection will be
considerable.=20

In the federations model it depends on the scale of the federations.
I've been told that there are some 3000 carriers in the US and 8000
public traded companies alone

This does not count the million of end users that might get their own
domain and run a Sip Proxy.=20

Regards,

Reinaldo

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Thursday, June 29, 2006 7:09 AM
> To: 'Uzelac, Adam'; 'Medhavi Bhatia'; Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: RE: TLS for peering (was RE: Comments (RE:
> [speermint]requirementsdraft submitted))
>=20
> <apologies for sporadic replies, I'm traveling with intermittent
access>
> This us a PEERING system with federations.  The likelihood of a
persistent
> TLS connection being maintained is high, and the cost of establishing
a
> new
> one is relatively low, especially if, as on another thread, one of the
> things a federation means is a CA signing certs for use by the
federation.
> The number of peers is going to be relatively low, improving the
> probability
> of the persistent TLS connection being available.
>=20
> Brian
>=20
> -----Original Message-----
> From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]
> Sent: Tuesday, June 27, 2006 9:48 PM
> To: Medhavi Bhatia; Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: RE: TLS for peering (was RE: Comments (RE:
> [speermint]requirementsdraft submitted))
>=20
> TLS is becoming more and more a subject when the topic of SIP
> interconnections arises for me professionally.  This shows growing
> momentum indeed.  With that said, I can not agree with TLS as a
> steadfast requirement.  Economics and scale are my major concerns.
TLS
> adds cost, server load, and there are fail-over issues to consider as
> well.  For the person that designs an interconnect, I believe weighing
> the cost against the benefit of the different models a useful
exercise.
> Mandating TLS, even if all parties involved in a peering arrangement
> feel somehow "safe and secure" due to underlying security mechanisms
> (L2/L3VPNs, etc), seems a little harsh.
>=20
> Adam
>=20
> -----Original Message-----
> From: Medhavi Bhatia [mailto:mbhatia@3clogic.com]
> Sent: Tuesday, June 27, 2006 8:44 PM
> To: Reinaldo Penno
> Cc: speermint@ietf.org
> Subject: Re: TLS for peering (was RE: Comments (RE: [speermint]
> requirementsdraft submitted))
>=20
> I am not a provider yet, but I think we dont need to mess with
TLS/TCP.
> As I understand one of the goals is to prevent the calls from failing
> due to "peering only" reasons. This means that things like codecs
which
> are not negotiated by the peers need to be mandated at a minimal
level.
> TCP/TLS can be negotiated between the peers themselves so it is a
> non-issue. Also as I still understand doing a high volume calls with
TLS
> is still expensive. Mandating TLS in cases where the caller or callee
> dont require it doesnt seem to add any value.
>=20
> -Medhavi.
>=20
> On 6/27/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> >
> > TLS for _all_ scenarios? Even in private peerings, physically secure
> > peerings, etc?
> >
> > Isn't that the same as requiring IPsec for all layer 3 peering? But
> > since Layer 5 peering might be more distributed, the number of TLS
> > sessions would be huge.
> >
> > It would be good to hear the provider's opinion on this...
> >
> > Thanks,
> >
> > Reinaldo
> >
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Tuesday, June 27, 2006 2:12 PM
> > > To: 'Jean-Francois Mule'; Reinaldo Penno
> > > Cc: speermint@ietf.org
> > > Subject: RE: Comments (RE: [speermint] requirements draft
submitted)
> > >
> > > >
> > > > "3.4.  Security requirements
> > > >
> > > >    All SIP messages MUST be sent over TLS [RFC3546] to provide
> > > >    transport-layer security as defined in RFC 3261,"
> > > >
> > > > Giving the fact that we can have security through physical
> > > segregation,
> > > > intra-federation/provider calls, etc, the "MUST" here seems too
> > > > generic.
> > >
> > > I tend to differ on this: it is not because you have lower layer
> > > security (presumably hop by hop security associations) that you
> > > should relax the end-to-end transport security requirements.
> > > Rohan's well written draft is also in favor of mandatory support
of
> > TLS,
> > > and so is the SIP Forum's IP PBX to VSP (here is a working link:
> > >
> >
http://data.memberclicks.com/bbattach/130808/sf-draft-twg-IP_PBX_SP_In
> > te
> > > rop-sibley-final.pdf).
> > > May be the requirement should be rewritten to put the MUST on SIP
> > > entities involved in L5 peering.
> > >
> > > <brian>I'm in favor of requiring TLS.  I'd say MUST in a
heartbeat.
> > >
> > > Brian
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
> _______________________________________________
> Speermint mailing list
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>=20
> _______________________________________________
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From speermint-bounces@ietf.org Thu Jun 29 12:38:20 2006
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Subject: RE: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
Date: Thu, 29 Jun 2006 12:38:15 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
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No, it will not be deprecated. TLS must be supported as one of the
options, but mandating its use is another issue.

There are many ways to establish trust.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Wednesday, June 28, 2006 7:06 PM
> To: Reinaldo Penno; 'Daryl Malas'
> Cc: speermint@ietf.org; 'Khan, Sohel Q [CTO]'; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Reinaldo Penno [mailto:rpenno@juniper.net]
> > Sent: Wednesday, June 28, 2006 6:21 PM
> > To: richard@shockey.us; Daryl Malas
> > Cc: speermint@ietf.org; Khan, Sohel Q [CTO]; Uzelac, Adam
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraftsubmitted))
> >
> > The following security 'models' come to mind:
> >
> > 1 - TLS
> > 2 - IPSec
> > 3 - Physical security
> > 4 - Access lists
> > 5 - Registrations
> > 6 - others?
> >
> > TLS just like the other methods has pros and cons that should be
> > weighted by the persons that will actually use it.
>=20
> But it cannot be deprecated. It MUST be supported. Its part of the
core
> SIP
> standard you cannot side step this issue.
>=20
> >
> > I hope we can agree with that.
>=20
>=20
> This is exactly where we begin to make progress. Define the reasonable
> security/trust models including TLS that we can find consensus on, the
> smaller the list the better.
>=20
> This is really the core problem in spearmint. What is the ultimate
> trust/security model that service providers can rely on to accept
traffic
> across domain boundaries.
>=20
> Are you really BT or Sprint or FT or Cablevision etc. Or are you
really
> IBM
> or Ford or Bombardier or Sony for that matter. Can I bill you for your
> traffic? How do I know you are who you say you are.
>=20
>=20
>=20
> > >
> > > _______________________________________________
> > > Speermint mailing list
> > > Speermint@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/speermint

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From speermint-bounces@ietf.org Thu Jun 29 20:05:30 2006
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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Andrew Newton" <andy@hxr.us>
Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
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Andy,

What you've described are a bunch of hacks that service providers use
because they don't peer over the Internet or because they can't rely
on the other guy to support TLS.  The combination of hacks is much,
much more expensive in terms of both capital equipment and
operationally than just using TLS.

A $2000 rack mount server can encrypt and decrypt a phenominal volume
of TLS traffic.  Hardware crypto accellerators are available that
increase this volume and decrease initial TLS handshake latencies by
another couple of orders of magnitude.

I'm willing to bet a few beers that the amount of data that is
encrypted using TLS exceeds the total volume of PSTN signaling traffic
on a daily basis.

To those who have a problem with TLS, I ask them to provide some
quantitative data or real requirements rather than rhetoric and FUD.

thanks,
-rohan





On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>
> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > how do you propose to establish a trust relationship between SIP
> > domain
> > boundaries without TLS?
>
> There are other methods: security lower in the stack, private
> connections, address ACLs...  the things that providers use today.
>
> > If you cannot establish a trust relationship between interconnecting
> > partners then what is the purpose of this whole exercise?
>
> To answer the second part of that sentence: I don't know; I've been
> asking that question for quite some time.
>
> -andy
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>

_______________________________________________
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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Reinaldo Penno" <rpenno@juniper.net>
Subject: Re: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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On 6/28/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> Shoving down the throats of people an operational requirement that will
> increase the cost of a deployment is not the way to go. IETF is about
> running code and consensus.

Can you prove to this group that TLS would cost more than using IPsec?
 than using a private walled garden network?  Can you describe a model
where this begins to look expensive compared to all the other various
boxes SPs are installing.  Where are your requirements?

> Are you going to suggest secure RTP as well between every single call on
> the planet?

I certainly won't.  The attack isn't as relevant as the attack on the
signaling channel.

thanks,
-rohan

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From: Benjamin Black <ben@layer8.net>
Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
Date: Thu, 29 Jun 2006 17:29:32 -0700
To: Rohan Mahy <rohan.mahy@gmail.com>
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rohan,

i don't think it is sufficient to simply make tls a requirement.   
what is the problem you are trying to solve using tls?  given that,  
what are the specific features/options of tls that minimally solve  
that problem?  for example, eNULL encryption may not be appropriate,  
but is available if the entirety of the specification is "must use tls".

all this talk of what the performance impact of tls may or may not be  
presumes specifics of requirements, design, and implementation that i  
have not seen clearly articulated in this thread.


thanks,
ben

On Jun 29, 2006, at 5:05 PM, Rohan Mahy wrote:

> Andy,
>
> What you've described are a bunch of hacks that service providers use
> because they don't peer over the Internet or because they can't rely
> on the other guy to support TLS.  The combination of hacks is much,
> much more expensive in terms of both capital equipment and
> operationally than just using TLS.
>
> A $2000 rack mount server can encrypt and decrypt a phenominal volume
> of TLS traffic.  Hardware crypto accellerators are available that
> increase this volume and decrease initial TLS handshake latencies by
> another couple of orders of magnitude.
>
> I'm willing to bet a few beers that the amount of data that is
> encrypted using TLS exceeds the total volume of PSTN signaling traffic
> on a daily basis.
>
> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
>
> thanks,
> -rohan
>
>
>
>
>
> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>>
>> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
>> > how do you propose to establish a trust relationship between SIP
>> > domain
>> > boundaries without TLS?
>>
>> There are other methods: security lower in the stack, private
>> connections, address ACLs...  the things that providers use today.
>>
>> > If you cannot establish a trust relationship between  
>> interconnecting
>> > partners then what is the purpose of this whole exercise?
>>
>> To answer the second part of that sentence: I don't know; I've been
>> asking that question for quite some time.
>>
>> -andy
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>


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Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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Rohan,

I'm okay with TLS as long as there are alternatives, but you email seems
contradictory. You complain of rhetoric, but

(1) " much, much more expensive in terms of both capital equipment and
operationally than just using TLS ",=20

(2) " A $2000 rack mount server can encrypt and decrypt a phenominal
volume
of TLS traffic",=20

(3) " by another couple of orders of magnitude",=20

Can you provide quantitative data instead of, you know...rhetoric?

I'll be very interested on reading some serious data on (1) above and
the number of TLS _connections per second_ on a $2000 rack server.

As for the beers, are you saying the number of TLS _connections per
second_ on the Internet today is greater than the number of calls in the
PSTN? Worldwide? That would be an interesting bet. Have any numbers on
that?

Thanks,

Reinaldo

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: Thursday, June 29, 2006 5:05 PM
> To: Andrew Newton
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> Subject: Re: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
> Andy,
>=20
> What you've described are a bunch of hacks that service providers use
> because they don't peer over the Internet or because they can't rely
> on the other guy to support TLS.  The combination of hacks is much,
> much more expensive in terms of both capital equipment and
> operationally than just using TLS.
>=20
> A $2000 rack mount server can encrypt and decrypt a phenominal volume
> of TLS traffic.  Hardware crypto accellerators are available that
> increase this volume and decrease initial TLS handshake latencies by
> another couple of orders of magnitude.
>=20
> I'm willing to bet a few beers that the amount of data that is
> encrypted using TLS exceeds the total volume of PSTN signaling traffic
> on a daily basis.
>=20
> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
>=20
> thanks,
> -rohan
>=20
>=20
>=20
>=20
>=20
> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
> >
> > On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > > how do you propose to establish a trust relationship between SIP
> > > domain
> > > boundaries without TLS?
> >
> > There are other methods: security lower in the stack, private
> > connections, address ACLs...  the things that providers use today.
> >
> > > If you cannot establish a trust relationship between
interconnecting
> > > partners then what is the purpose of this whole exercise?
> >
> > To answer the second part of that sentence: I don't know; I've been
> > asking that question for quite some time.
> >
> > -andy
> >
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > https://www1.ietf.org/mailman/listinfo/speermint
> >
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

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Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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Rohan,

There is no need to resort to these kinds of personal statements. The
burden of the proof does not need to be on me.=20

As with any scientific discussion, I'm validating the other side. What
I'm hearing is that people say TLS is faster, securer, cheaper...than
_any_ alternative.

What is wrong in trying to validate that? Or asking where is the data to
prove that?=20

Thanks,

Reinaldo=20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: Thursday, June 29, 2006 5:21 PM
> To: Reinaldo Penno
> Cc: richard@shockey.us; Andrew Newton; Khan, Sohel Q [CTO];
> speermint@ietf.org; Uzelac, Adam
> Subject: Re: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
> On 6/28/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > Shoving down the throats of people an operational requirement that
will
> > increase the cost of a deployment is not the way to go. IETF is
about
> > running code and consensus.
>=20
> Can you prove to this group that TLS would cost more than using IPsec?
>  than using a private walled garden network?  Can you describe a model
> where this begins to look expensive compared to all the other various
> boxes SPs are installing.  Where are your requirements?
>=20
> > Are you going to suggest secure RTP as well between every single
call on
> > the planet?
>=20
> I certainly won't.  The attack isn't as relevant as the attack on the
> signaling channel.
>=20
> thanks,
> -rohan

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> -----Original Message-----
> From: Benjamin Black [mailto:ben@layer8.net]
> Sent: Thursday, June 29, 2006 8:30 PM
> To: Rohan Mahy
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> Subject: Re: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> rohan,
> 
> i don't think it is sufficient to simply make tls a requirement.
> what is the problem you are trying to solve using tls? 

Face the music  you will support TLS whether you like it or not. This is SIP
101. It will not be a requirement that you use it but I can tell you I'd
never give my voice traffic to any service provider that did not accept a
TLS connection at their point of interconnection.

Hello here... SIP signaling security, preventing MIM attacks, trust
relationships between domain bountries do you want a longer list of why you
need SIP security?

This whole thread is highlighting the central problem this list seems to
have has which is unwillingness to confront the central issues of security
and trust relationships between service providers and what protocols are
REQUIRED to create those trust relationships.

OK if you are a GSM network you figured this out already ..create a totally
private internet with completely private points of interconnection and
within a the great wall of GSM you have security.

It works ..is that the model you want?  

Brother Rohan is spot on here. CPU cycles are cheep considering what the
requirements for security here are.

I just don't get it why does everyone so dislikes TLS..its simple proven to
scale and it works.

Frankly I think the chairs need to step in here and drive a little consensus
that without a robust security/trust model defined ASAP for speermint all
other discussions are futile and a waste of time.




 given that,
> what are the specific features/options of tls that minimally solve
> that problem?  for example, eNULL encryption may not be appropriate,
> but is available if the entirety of the specification is "must use tls".
> 
> all this talk of what the performance impact of tls may or may not be
> presumes specifics of requirements, design, and implementation that i
> have not seen clearly articulated in this thread.
> 
> 
> thanks,
> ben
> 
> On Jun 29, 2006, at 5:05 PM, Rohan Mahy wrote:
> 
> > Andy,
> >
> > What you've described are a bunch of hacks that service providers use
> > because they don't peer over the Internet or because they can't rely
> > on the other guy to support TLS.  The combination of hacks is much,
> > much more expensive in terms of both capital equipment and
> > operationally than just using TLS.
> >
> > A $2000 rack mount server can encrypt and decrypt a phenominal volume
> > of TLS traffic.  Hardware crypto accellerators are available that
> > increase this volume and decrease initial TLS handshake latencies by
> > another couple of orders of magnitude.
> >
> > I'm willing to bet a few beers that the amount of data that is
> > encrypted using TLS exceeds the total volume of PSTN signaling traffic
> > on a daily basis.
> >
> > To those who have a problem with TLS, I ask them to provide some
> > quantitative data or real requirements rather than rhetoric and FUD.
> >
> > thanks,
> > -rohan
> >
> >
> >
> >
> >
> > On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
> >>
> >> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> >> > how do you propose to establish a trust relationship between SIP
> >> > domain
> >> > boundaries without TLS?
> >>
> >> There are other methods: security lower in the stack, private
> >> connections, address ACLs...  the things that providers use today.
> >>
> >> > If you cannot establish a trust relationship between
> >> interconnecting
> >> > partners then what is the purpose of this whole exercise?
> >>
> >> To answer the second part of that sentence: I don't know; I've been
> >> asking that question for quite some time.
> >>
> >> -andy


_______________________________________________
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From speermint-bounces@ietf.org Thu Jun 29 21:14:20 2006
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Subject: RE: TLS for peering (was RE:
	Comments(RE:[speermint]requirementsdraftsubmitted))
Date: Thu, 29 Jun 2006 21:14:13 -0400
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From: "Reinaldo Penno" <rpenno@juniper.net>
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IMO TLS must be supported but we should spell out the alternatives.

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Thursday, June 29, 2006 6:02 PM
> To: 'Benjamin Black'; 'Rohan Mahy'
> Cc: 'Khan, Sohel Q [CTO]'; speermint@ietf.org; 'Uzelac, Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraftsubmitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Benjamin Black [mailto:ben@layer8.net]
> > Sent: Thursday, June 29, 2006 8:30 PM
> > To: Rohan Mahy
> > Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> > Subject: Re: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > rohan,
> >
> > i don't think it is sufficient to simply make tls a requirement.
> > what is the problem you are trying to solve using tls?
>=20
> Face the music  you will support TLS whether you like it or not. This
is
> SIP
> 101. It will not be a requirement that you use it but I can tell you
I'd
> never give my voice traffic to any service provider that did not
accept a
> TLS connection at their point of interconnection.
>=20
> Hello here... SIP signaling security, preventing MIM attacks, trust
> relationships between domain bountries do you want a longer list of
why
> you
> need SIP security?
>=20
> This whole thread is highlighting the central problem this list seems
to
> have has which is unwillingness to confront the central issues of
security
> and trust relationships between service providers and what protocols
are
> REQUIRED to create those trust relationships.
>=20
> OK if you are a GSM network you figured this out already ..create a
> totally
> private internet with completely private points of interconnection and
> within a the great wall of GSM you have security.
>=20
> It works ..is that the model you want?
>=20
> Brother Rohan is spot on here. CPU cycles are cheep considering what
the
> requirements for security here are.
>=20
> I just don't get it why does everyone so dislikes TLS..its simple
proven
> to
> scale and it works.
>=20
> Frankly I think the chairs need to step in here and drive a little
> consensus
> that without a robust security/trust model defined ASAP for speermint
all
> other discussions are futile and a waste of time.
>=20
>=20
>=20
>=20
>  given that,
> > what are the specific features/options of tls that minimally solve
> > that problem?  for example, eNULL encryption may not be appropriate,
> > but is available if the entirety of the specification is "must use
tls".
> >
> > all this talk of what the performance impact of tls may or may not
be
> > presumes specifics of requirements, design, and implementation that
i
> > have not seen clearly articulated in this thread.
> >
> >
> > thanks,
> > ben
> >
> > On Jun 29, 2006, at 5:05 PM, Rohan Mahy wrote:
> >
> > > Andy,
> > >
> > > What you've described are a bunch of hacks that service providers
use
> > > because they don't peer over the Internet or because they can't
rely
> > > on the other guy to support TLS.  The combination of hacks is
much,
> > > much more expensive in terms of both capital equipment and
> > > operationally than just using TLS.
> > >
> > > A $2000 rack mount server can encrypt and decrypt a phenominal
volume
> > > of TLS traffic.  Hardware crypto accellerators are available that
> > > increase this volume and decrease initial TLS handshake latencies
by
> > > another couple of orders of magnitude.
> > >
> > > I'm willing to bet a few beers that the amount of data that is
> > > encrypted using TLS exceeds the total volume of PSTN signaling
traffic
> > > on a daily basis.
> > >
> > > To those who have a problem with TLS, I ask them to provide some
> > > quantitative data or real requirements rather than rhetoric and
FUD.
> > >
> > > thanks,
> > > -rohan
> > >
> > >
> > >
> > >
> > >
> > > On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
> > >>
> > >> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > >> > how do you propose to establish a trust relationship between
SIP
> > >> > domain
> > >> > boundaries without TLS?
> > >>
> > >> There are other methods: security lower in the stack, private
> > >> connections, address ACLs...  the things that providers use
today.
> > >>
> > >> > If you cannot establish a trust relationship between
> > >> interconnecting
> > >> > partners then what is the purpose of this whole exercise?
> > >>
> > >> To answer the second part of that sentence: I don't know; I've
been
> > >> asking that question for quite some time.
> > >>
> > >> -andy
>=20
>=20
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint

_______________________________________________
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	"'Reinaldo Penno'" <rpenno@juniper.net>
Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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Right on Brother Rohan .. protection of the 5060 signaling channel is much
more important than the RTP media stream. 



> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: Thursday, June 29, 2006 8:21 PM
> To: Reinaldo Penno
> Cc: richard@shockey.us; Andrew Newton; Khan, Sohel Q [CTO];
> speermint@ietf.org; Uzelac, Adam
> Subject: Re: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> On 6/28/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > Shoving down the throats of people an operational requirement that will
> > increase the cost of a deployment is not the way to go. IETF is about
> > running code and consensus.
> 
> Can you prove to this group that TLS would cost more than using IPsec?
>  than using a private walled garden network?  Can you describe a model
> where this begins to look expensive compared to all the other various
> boxes SPs are installing.  Where are your requirements?


Well here is my requirement you cannot have interconnection between service
providers without securing and authenticating the source of the SIP 5060
signaling data.

We know TLS works. That is why it is in the SIP standard. That is why it
MUST be supported.

The burden of proof is now on you and others to describe what other
security/trust models you want supported and the risks associated with them.
And those methods should either match or exceed the capabilities of TLS.


> 
> > Are you going to suggest secure RTP as well between every single call on
> > the planet?
> 
> I certainly won't.  The attack isn't as relevant as the attack on the
> signaling channel.
> 
> thanks,
> -rohan


_______________________________________________
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From speermint-bounces@ietf.org Thu Jun 29 21:39:36 2006
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Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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Date: Thu, 29 Jun 2006 21:39:26 -0400
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I did not know there was some kind strange brotherhood going on here
but...

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Thursday, June 29, 2006 6:15 PM
> To: 'Rohan Mahy'; Reinaldo Penno
> Cc: 'Andrew Newton'; 'Khan, Sohel Q [CTO]'; speermint@ietf.org;
'Uzelac,
> Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
>=20
> Right on Brother Rohan .. protection of the 5060 signaling channel is
much
> more important than the RTP media stream.
>=20
>=20
>=20
> > -----Original Message-----
> > From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > Sent: Thursday, June 29, 2006 8:21 PM
> > To: Reinaldo Penno
> > Cc: richard@shockey.us; Andrew Newton; Khan, Sohel Q [CTO];
> > speermint@ietf.org; Uzelac, Adam
> > Subject: Re: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > On 6/28/06, Reinaldo Penno <rpenno@juniper.net> wrote:
> > > Shoving down the throats of people an operational requirement that
> will
> > > increase the cost of a deployment is not the way to go. IETF is
about
> > > running code and consensus.
> >
> > Can you prove to this group that TLS would cost more than using
IPsec?
> >  than using a private walled garden network?  Can you describe a
model
> > where this begins to look expensive compared to all the other
various
> > boxes SPs are installing.  Where are your requirements?
>=20
>=20
> Well here is my requirement you cannot have interconnection between
> service
> providers without securing and authenticating the source of the SIP
5060
> signaling data.
[[Reinaldo]]=20

huh? There are plenty of VoIP peering today without TLS. They seem to be
working. Are they not?

The concept of securing and trust is the one that matters between
service providers. If they think physical security is okay, then that is
good enough for them.=20

>=20
> We know TLS works. That is why it is in the SIP standard. That is why
it
> MUST be supported.
>=20
> The burden of proof is now on you and others to describe what other
> security/trust models you want supported and the risks associated with
> them.
> And those methods should either match or exceed the capabilities of
TLS.
>=20
>=20
[[Reinaldo]]=20

Not really. They only need to be alternatives for people that when peer
do not need TLS. There are many small enterprises; small carriers that
do not want, cannot afford Tor do not have the expertise to deal with
LS. This is true especially outside the US. There is life outside the
rich and powerful providers.


> >
> > > Are you going to suggest secure RTP as well between every single
call
> on
> > > the planet?
> >
> > I certainly won't.  The attack isn't as relevant as the attack on
the
> > signaling channel.
> >
> > thanks,
> > -rohan

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> -----Original Message-----
> From: Reinaldo Penno [mailto:rpenno@juniper.net]
> Sent: Thursday, June 29, 2006 9:39 PM
> To: richard@shockey.us; Rohan Mahy
> Cc: Andrew Newton; Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
> 
> I did not know there was some kind strange brotherhood going on here
> but...


You never heard of the SIP Mafia?  





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Sorry Don Shockley....I will have you TLS ready by next week...I promise
(:-)


> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Thursday, June 29, 2006 6:53 PM
> To: Reinaldo Penno; 'Rohan Mahy'
> Cc: 'Andrew Newton'; 'Khan, Sohel Q [CTO]'; speermint@ietf.org;
'Uzelac,
> Adam'
> Subject: RE: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>=20
>=20
>=20
> > -----Original Message-----
> > From: Reinaldo Penno [mailto:rpenno@juniper.net]
> > Sent: Thursday, June 29, 2006 9:39 PM
> > To: richard@shockey.us; Rohan Mahy
> > Cc: Andrew Newton; Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac,
Adam
> > Subject: RE: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >
> > I did not know there was some kind strange brotherhood going on here
> > but...
>=20
>=20
> You never heard of the SIP Mafia?
>=20
>=20

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From speermint-bounces@ietf.org Thu Jun 29 22:17:07 2006
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richard,

i agree 100%.  this is really important.  so important, in fact, that  
the requirements should be clearly stated so the problem is actually  
solved.  tls _can_ be an extremely powerful tool, but simply  
requiring its use is insufficient to achieve the goals of this  
working group, imo.


thanks,
ben

On Jun 29, 2006, at 6:02 PM, Richard Shockey wrote:

>
>
>> -----Original Message-----
>> From: Benjamin Black [mailto:ben@layer8.net]
>> Sent: Thursday, June 29, 2006 8:30 PM
>> To: Rohan Mahy
>> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
>> Subject: Re: TLS for peering (was RE:
>> Comments(RE:[speermint]requirementsdraft submitted))
>>
>> rohan,
>>
>> i don't think it is sufficient to simply make tls a requirement.
>> what is the problem you are trying to solve using tls?
>
> Face the music  you will support TLS whether you like it or not.  
> This is SIP
> 101. It will not be a requirement that you use it but I can tell  
> you I'd
> never give my voice traffic to any service provider that did not  
> accept a
> TLS connection at their point of interconnection.
>
> Hello here... SIP signaling security, preventing MIM attacks, trust
> relationships between domain bountries do you want a longer list of  
> why you
> need SIP security?
>
> This whole thread is highlighting the central problem this list  
> seems to
> have has which is unwillingness to confront the central issues of  
> security
> and trust relationships between service providers and what  
> protocols are
> REQUIRED to create those trust relationships.
>
> OK if you are a GSM network you figured this out already ..create a  
> totally
> private internet with completely private points of interconnection and
> within a the great wall of GSM you have security.
>
> It works ..is that the model you want?
>
> Brother Rohan is spot on here. CPU cycles are cheep considering  
> what the
> requirements for security here are.
>
> I just don't get it why does everyone so dislikes TLS..its simple  
> proven to
> scale and it works.
>
> Frankly I think the chairs need to step in here and drive a little  
> consensus
> that without a robust security/trust model defined ASAP for  
> speermint all
> other discussions are futile and a waste of time.
>
>
>
>
>  given that,
>> what are the specific features/options of tls that minimally solve
>> that problem?  for example, eNULL encryption may not be appropriate,
>> but is available if the entirety of the specification is "must use  
>> tls".
>>
>> all this talk of what the performance impact of tls may or may not be
>> presumes specifics of requirements, design, and implementation that i
>> have not seen clearly articulated in this thread.
>>
>>
>> thanks,
>> ben
>>
>> On Jun 29, 2006, at 5:05 PM, Rohan Mahy wrote:
>>
>>> Andy,
>>>
>>> What you've described are a bunch of hacks that service providers  
>>> use
>>> because they don't peer over the Internet or because they can't rely
>>> on the other guy to support TLS.  The combination of hacks is much,
>>> much more expensive in terms of both capital equipment and
>>> operationally than just using TLS.
>>>
>>> A $2000 rack mount server can encrypt and decrypt a phenominal  
>>> volume
>>> of TLS traffic.  Hardware crypto accellerators are available that
>>> increase this volume and decrease initial TLS handshake latencies by
>>> another couple of orders of magnitude.
>>>
>>> I'm willing to bet a few beers that the amount of data that is
>>> encrypted using TLS exceeds the total volume of PSTN signaling  
>>> traffic
>>> on a daily basis.
>>>
>>> To those who have a problem with TLS, I ask them to provide some
>>> quantitative data or real requirements rather than rhetoric and FUD.
>>>
>>> thanks,
>>> -rohan
>>>
>>>
>>>
>>>
>>>
>>> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>>>>
>>>> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
>>>>> how do you propose to establish a trust relationship between SIP
>>>>> domain
>>>>> boundaries without TLS?
>>>>
>>>> There are other methods: security lower in the stack, private
>>>> connections, address ACLs...  the things that providers use today.
>>>>
>>>>> If you cannot establish a trust relationship between
>>>> interconnecting
>>>>> partners then what is the purpose of this whole exercise?
>>>>
>>>> To answer the second part of that sentence: I don't know; I've been
>>>> asking that question for quite some time.
>>>>
>>>> -andy
>
>


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	From a customer point of view, I'm sure they would rather we kept our 
signaling protected, AND kept their calls private. Fraud charge handling 
and billing problems (sadly) are almost expected by customers these 
days, this is aggravation for sure; but not a customer loss for the most 
part. Having their intimate chats with kids/mom/dog (or other bits) 
recorded and punted around the internet is a lost customer, loss of face 
for the provider, and probably ends up with lawyers involved.

	So protecting signaling certainly helps protect the bearer(s), but I 
don't think I would put it MUCH higher then the data, maybe just a wee 
bit higher ;). Most customers pay for the service after all, not the 
implementation.

	As far as my input into the requirement for TLS as a method of 
establishing trust between peers, I agree with it as long as there are 
other allowable methods.

	-pee


Richard Shockey wrote:
> Right on Brother Rohan .. protection of the 5060 signaling channel is much
> more important than the RTP media stream. 
> 
> 
> 
>> -----Original Message-----
>

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Rohan Mahy wrote:

> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
> 

	Here's a single point of data for you.

	Number of devices in our voip network that handle TLS connections with 
their current hardware and software configurations:

	 => 0.

	I'm not defending anyone so don't bother with the "pick a different 
vendor(s)" replies =). There has been limited demand for TCP/TLS to the 
equipment vendors, so consequently there has been limited development 
into supporting it (everyone is *working on it*). It could be a 
reasonable BCP from this group helps accelerate that process.


	-pee


> thanks,
> -rohan
> 

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> A $2000 rack mount server can encrypt and decrypt a phenominal volume
> of TLS traffic.  Hardware crypto accellerators are available that
> increase this volume and decrease initial TLS handshake latencies by
> another couple of orders of magnitude.
> 
> I'm willing to bet a few beers that the amount of data that is
> encrypted using TLS exceeds the total volume of PSTN signaling traffic
> on a daily basis.

A pretty good bet if I do say so. Its tough to break out per transactions
but the underlying supposition is that there is a heck of a lot of TLS going
on.

Well the numbers are huge no matter how you count them. Basic stats are that
on the basis of 162M US access lines in 2004 generated 422.4 Billion access
minutes of which 81 B of those were toll calls. Due to LNP and other network
issues everyone of those requires a form of SS7 dip. ( FCC )

Ecommerce stats are now 

Total Worldwide eCommerce Revenues, 2004 (B2B & B2C)
Region 	Total
North America 	$3.5 trillion
Asia Pacific 	$1.6 trillion
Western Europe 	$1.5 trillion
Latin America 	$81.8 billion
Rest of World 	$68.6 billion

Source: Forrester Research

Payment Types In The US, 2005 - 2010

Volume ($ trillion) 	2005 	2010
Paper 	$3.4 (50%) 	$3.6 (36%)
Cards 	$2.7 (39%) 	$4.3 (49%)
Electronic 	$0.7 (11%) 	$1.3 (15%)
Total 	$6.8 (100%) 	$8.7 (100%)

Source: The Nilson Report


> 
> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
> 
> thanks,
> -rohan
> 


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From speermint-bounces@ietf.org Fri Jun 30 10:37:20 2006
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From: Andrew Newton <andy@hxr.us>
Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
Date: Fri, 30 Jun 2006 10:30:42 -0400
To: Rohan Mahy <rohan.mahy@gmail.com>
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Rohan,

I am not a liberty to give specific numbers, but I can assure you  
that your assumptions are wrong.  We have a lot of very expensive  
equipment that would fall over if it did all signaling over TLS or  
even just TCP.  And we have done the tests and measurements, and a  
$2000 rack mount server cannot handle the load.

And since you are asking for "real requirements", I'd like to know  
what your "real requirements" are for specifying TLS for signaling.   
Specifically, I'd like to know the following:

1) Why is it you are advocating security of the signaling but not the  
media?  What requirement determines that signaling is important to  
secure but media is not important to secure?
2) Why do you believe TLS offers better authentication than any other  
method, especially given that there is no global trust anchor?
3) What requirement makes TLS more secure than a dedicated link?
4) What requirement makes TLS more secure than IPSec?
5) Given that providers interconnect today without TLS, what  
requirement is not met by the current interconnection methods that  
TLS will provide?

Don't get me wrong.  I think all vendors should support TLS, and I  
think providers and carriers should be able to have it at their  
disposal when doing interconnection, but it is only one of many options.

-andy

On Jun 29, 2006, at 8:05 PM, Rohan Mahy wrote:

> Andy,
>
> What you've described are a bunch of hacks that service providers use
> because they don't peer over the Internet or because they can't rely
> on the other guy to support TLS.  The combination of hacks is much,
> much more expensive in terms of both capital equipment and
> operationally than just using TLS.
>
> A $2000 rack mount server can encrypt and decrypt a phenominal volume
> of TLS traffic.  Hardware crypto accellerators are available that
> increase this volume and decrease initial TLS handshake latencies by
> another couple of orders of magnitude.
>
> I'm willing to bet a few beers that the amount of data that is
> encrypted using TLS exceeds the total volume of PSTN signaling traffic
> on a daily basis.
>
> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
>
> thanks,
> -rohan
>
>
>
>
>
> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>>
>> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
>> > how do you propose to establish a trust relationship between SIP
>> > domain
>> > boundaries without TLS?
>>
>> There are other methods: security lower in the stack, private
>> connections, address ACLs...  the things that providers use today.
>>
>> > If you cannot establish a trust relationship between  
>> interconnecting
>> > partners then what is the purpose of this whole exercise?
>>
>> To answer the second part of that sentence: I don't know; I've been
>> asking that question for quite some time.
>>
>> -andy
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>>


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On Thu, Jun 29, 2006 at 09:02:18PM -0400, Richard Shockey wrote:
>=20
>=20
> > -----Original Message-----
> > From: Benjamin Black [mailto:ben@layer8.net]
> > Sent: Thursday, June 29, 2006 8:30 PM
> > To: Rohan Mahy
> > Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> > Subject: Re: TLS for peering (was RE:
> > Comments(RE:[speermint]requirementsdraft submitted))
> >=20
> > rohan,
> >=20
> > i don't think it is sufficient to simply make tls a requirement.
> > what is the problem you are trying to solve using tls?=20
>=20
> Face the music  you will support TLS whether you like it or not. This is =
SIP
> 101. It will not be a requirement that you use it but I can tell you I'd
> never give my voice traffic to any service provider that did not accept a
> TLS connection at their point of interconnection.
>=20
> Hello here... SIP signaling security, preventing MIM attacks, trust
> relationships between domain bountries do you want a longer list of why y=
ou
> need SIP security?
>=20
> This whole thread is highlighting the central problem this list seems to
> have has which is unwillingness to confront the central issues of security
> and trust relationships between service providers and what protocols are
> REQUIRED to create those trust relationships.
>=20
> OK if you are a GSM network you figured this out already ..create a total=
ly
> private internet with completely private points of interconnection and
> within a the great wall of GSM you have security.
>=20
> It works ..is that the model you want? =20
>=20
> Brother Rohan is spot on here. CPU cycles are cheep considering what the
> requirements for security here are.
>=20
> I just don't get it why does everyone so dislikes TLS..its simple proven =
to
> scale and it works.
>=20
> Frankly I think the chairs need to step in here and drive a little consen=
sus
> that without a robust security/trust model defined ASAP for speermint all
> other discussions are futile and a waste of time.

	Thanks for the prompting Richard. FYI, one of the things
	we've been discussing to address this is a "speermint
	security considerations" document, on the theory that
	rather that thrashing around as we've been doing, we'll
	have a concrete piece of work that we can discuss.=20


	--dmm

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From speermint-bounces@ietf.org Fri Jun 30 11:27:05 2006
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Subject: RE: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
	submitted))
Date: Fri, 30 Jun 2006 10:26:55 -0500
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A voice of reason among the group. =20

We went through this very discussion, in Sigtran, and came out that
IPSEC works just as well and left the choice to the implementers between
IPSEC and TLS.  Why not leverage this work?  See RFC
http://www.ietf.org/rfc/rfc3788.txt, especially sections 7 & 8.




-----Original Message-----
From: Andrew Newton [mailto:andy@hxr.us]=20
Sent: Friday, June 30, 2006 9:31 AM
To: Rohan Mahy
Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
Subject: Re: TLS for peering (was RE:
Comments(RE:[speermint]requirementsdraft submitted))

Rohan,

I am not a liberty to give specific numbers, but I can assure you =20
that your assumptions are wrong.  We have a lot of very expensive =20
equipment that would fall over if it did all signaling over TLS or =20
even just TCP.  And we have done the tests and measurements, and a =20
$2000 rack mount server cannot handle the load.

And since you are asking for "real requirements", I'd like to know =20
what your "real requirements" are for specifying TLS for signaling.  =20
Specifically, I'd like to know the following:

1) Why is it you are advocating security of the signaling but not the =20
media?  What requirement determines that signaling is important to =20
secure but media is not important to secure?
2) Why do you believe TLS offers better authentication than any other =20
method, especially given that there is no global trust anchor?
3) What requirement makes TLS more secure than a dedicated link?
4) What requirement makes TLS more secure than IPSec?
5) Given that providers interconnect today without TLS, what =20
requirement is not met by the current interconnection methods that =20
TLS will provide?

Don't get me wrong.  I think all vendors should support TLS, and I =20
think providers and carriers should be able to have it at their =20
disposal when doing interconnection, but it is only one of many options.

-andy

On Jun 29, 2006, at 8:05 PM, Rohan Mahy wrote:

> Andy,
>
> What you've described are a bunch of hacks that service providers use
> because they don't peer over the Internet or because they can't rely
> on the other guy to support TLS.  The combination of hacks is much,
> much more expensive in terms of both capital equipment and
> operationally than just using TLS.
>
> A $2000 rack mount server can encrypt and decrypt a phenominal volume
> of TLS traffic.  Hardware crypto accellerators are available that
> increase this volume and decrease initial TLS handshake latencies by
> another couple of orders of magnitude.
>
> I'm willing to bet a few beers that the amount of data that is
> encrypted using TLS exceeds the total volume of PSTN signaling traffic
> on a daily basis.
>
> To those who have a problem with TLS, I ask them to provide some
> quantitative data or real requirements rather than rhetoric and FUD.
>
> thanks,
> -rohan
>
>
>
>
>
> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>>
>> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
>> > how do you propose to establish a trust relationship between SIP
>> > domain
>> > boundaries without TLS?
>>
>> There are other methods: security lower in the stack, private
>> connections, address ACLs...  the things that providers use today.
>>
>> > If you cannot establish a trust relationship between =20
>> interconnecting
>> > partners then what is the purpose of this whole exercise?
>>
>> To answer the second part of that sentence: I don't know; I've been
>> asking that question for quite some time.
>>
>> -andy
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> https://www1.ietf.org/mailman/listinfo/speermint
>>


_______________________________________________
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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Andrew Newton" <andy@hxr.us>
Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
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Andy,

I am just leaving for a short vacation, but I will commit to answering
your questions before the meeting in Montreal.  I think these are good
questions and I want to take the time to answer them properly.

So that we can make apples to apples comparisons, it would be useful
to start with an order of magnitude number of calls per day.  This
does not need to be Sunrocket's (or any other carrier's) actual
current or expected peering traffic, but it should be plausible.

thanks,
-rohan


On 6/30/06, Andrew Newton <andy@hxr.us> wrote:
> Rohan,
>
> I am not a liberty to give specific numbers, but I can assure you
> that your assumptions are wrong.  We have a lot of very expensive
> equipment that would fall over if it did all signaling over TLS or
> even just TCP.  And we have done the tests and measurements, and a
> $2000 rack mount server cannot handle the load.
>
> And since you are asking for "real requirements", I'd like to know
> what your "real requirements" are for specifying TLS for signaling.
> Specifically, I'd like to know the following:
>
> 1) Why is it you are advocating security of the signaling but not the
> media?  What requirement determines that signaling is important to
> secure but media is not important to secure?
> 2) Why do you believe TLS offers better authentication than any other
> method, especially given that there is no global trust anchor?
> 3) What requirement makes TLS more secure than a dedicated link?
> 4) What requirement makes TLS more secure than IPSec?
> 5) Given that providers interconnect today without TLS, what
> requirement is not met by the current interconnection methods that
> TLS will provide?
>
> Don't get me wrong.  I think all vendors should support TLS, and I
> think providers and carriers should be able to have it at their
> disposal when doing interconnection, but it is only one of many options.
>
> -andy
>
> On Jun 29, 2006, at 8:05 PM, Rohan Mahy wrote:
>
> > Andy,
> >
> > What you've described are a bunch of hacks that service providers use
> > because they don't peer over the Internet or because they can't rely
> > on the other guy to support TLS.  The combination of hacks is much,
> > much more expensive in terms of both capital equipment and
> > operationally than just using TLS.
> >
> > A $2000 rack mount server can encrypt and decrypt a phenominal volume
> > of TLS traffic.  Hardware crypto accellerators are available that
> > increase this volume and decrease initial TLS handshake latencies by
> > another couple of orders of magnitude.
> >
> > I'm willing to bet a few beers that the amount of data that is
> > encrypted using TLS exceeds the total volume of PSTN signaling traffic
> > on a daily basis.
> >
> > To those who have a problem with TLS, I ask them to provide some
> > quantitative data or real requirements rather than rhetoric and FUD.
> >
> > thanks,
> > -rohan
> >
> >
> >
> >
> >
> > On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
> >>
> >> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> >> > how do you propose to establish a trust relationship between SIP
> >> > domain
> >> > boundaries without TLS?
> >>
> >> There are other methods: security lower in the stack, private
> >> connections, address ACLs...  the things that providers use today.
> >>
> >> > If you cannot establish a trust relationship between
> >> interconnecting
> >> > partners then what is the purpose of this whole exercise?
> >>
> >> To answer the second part of that sentence: I don't know; I've been
> >> asking that question for quite some time.
> >>
> >> -andy
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/speermint
> >>
>
>

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Subject: [Speermint] TLS for peering - Just the facts please :-)
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I have been involved with several TLS implementations for SIP  
including writing most the code for one and also some deployments.  
Sorry, I am not fully up to speed on this discussion of if I say  
something that has already been well covered, my apologies - there  
have been a few transaction on the topic :-)  I'm don't know what  
this WG should choose but let me provide some information that might  
help....


Do vendors implement it?
I don't have the exact numbers but I think at the last SIPit there  
was something like over 50 vendors that had TLS - we can get more  
details if needed. Every major vendor I am aware of has it on a proxy  
that would be used in this type of situations. It's actually pretty  
easy to support - don't confuse it with sips which is somewhat harder  
to support. Most proxies use some SSL library such as openssl or  
BSAFE. Given a library like that, it tends to add less than a 1000  
lines of code to the proxy.

Do people use it?
Microsoft has a popular presence server known as LCS. LCS  uses TLS  
for it's peering.  If you do a bit of googling around you will find  
there are a lot of things that choose to interoperate with LCS.  
Enterprise that deploys LCS can peer with AOL, Yahoo, and MSN. Again  
over TLS. It's a pretty good example of some running code. Don't  
underestimate the size of this deployment and keep in mind it is  
already peering with three of the very largest IM providers. I forget  
when this all first deployed but it's been years.

Is it Expensive?
Your average proxy running on some intel or sun box will do something  
like a few thousand sip transactions per second then it hits the CPU  
limit. Every proxy I have seen, the performance bottleneck has been  
CPU. Now if you compute the bandwidth for this amount of SIP traffic,  
then look at how much CPU is required to AES encrypt that much  
traffic, you will realize - the encryption time is going to make  
pretty much no difference to the number of transaction per second  
this box can do. If you talk to SIP implementers, the hard part of  
TLS is keeping open tens of thousands of connections form a proxy to  
a UA. This is not what you are talking about here - Here we are  
talking about keeping open 100s to 1000s of connections from one  
proxy to another. These are long lived connections - they can be set  
up quickly and stay up for very long time. A small amount of data  
flows over them (by small, I mean less than few Mbps). I expect the  
expense to be very low in terms of implementation effort, hardware  
required to run this, and operation effort to run them. Don't confuse  
the issues of proxy doing TLS to many UA connected to it with the  
proxy doing TLS to the other proxies it connects to.

What does it provide?
It provides some integrity and confidentiality protection, yawn,  
yawn, yawn - not clear that this helps your threat models much but is  
definitely of some use. But let's get on to the next thing. TLS lets  
you authenticate that you connected to the proxy you meant to connect  
to. I expect this might be very important for the threat models you  
are interested in. The reason in MUST implement in proxies is that  
this is the *only* way SIP provides for one proxy to know that it  
connected to the correct next proxy. If you need to know who you are  
connecting to, you probably want to think seriously about TLS.

It also provides a way of  using certificates to delegate certain  
authorization decisions. Let me give an example. Lets say a bunch of  
organization A, B, C agree to peer and form so peering consortium,  
call it PeersAreUs. Everyone that joins PeersAreUs has to sing some  
sort of contract about behavior. They do enum look ups to some enum  
root at PeersAreUs. Now  when a new organization D joins PeersAreUs,  
how do A,B,C find out they are authorized. One possibility is having  
PeersAreUs issue some sort of cert to D. Now A looks up a number in  
enum, gets routed to D, and D presents a certificate that lets A know  
that D is a valid member of PeersAreUs. This sounds sort of  
interesting to me from a operation expense point of view. Note that  
A, B, and C did not need to do anything when D joined and PeersAreUs  
did not have to run some large proxy that could deal with all the  
traffic of all it's members. I imagine you can find other ways to  
achieve this too, but hey, this one is already MUST implement in 3261  
and widely implemented in deployed devices.

Is IPsec an alternative?
Possibly for some threat models and not for others. The inter-sting  
thing about IPsec is how will you key it? Given the scope of work in  
this WG, I would expect this group to describe that if IPsec is  
chosen. Setting up manual key associations between from one proxy to  
thousands of others sounds like an very expensive solution from an  
operations point of view so I would guess that something other than  
manual configuration would be chosen. Once you go down that path you  
run into questions of how the proxy knows it connected to the correct  
domain. I'm willing to believe that there might be solutions to this  
but not clear to me what they are or that anyone has implemented them.

Are there other alternatives?
The answer to this is always yes but I encourage the group to figure  
out what the threats, list the alternative solutions, and focus on  
saying which concrete proposal looks best instead of saying I don't  
like proposal X. This is going to be an exercise in picking the thing  
you hate least not having some silver bullet that is perfect in every  
way.

Should this group mandate TLS use?
Heck if I know :-) I think you are a ways from saying all the things  
you need to accomplish in securing a peer so it makes it a little  
hard to say if any given solution would meet all the needs.




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Subject: Re: TLS for peering (was RE: Comments(RE:[speermint]requirementsdraft
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kevin,

3788 is a very useful reference for this discussion.  i think  
speermint needs both better specified requirements and a richer set  
of options, however.


bb

On Jun 30, 2006, at 8:26 AM, McCandless, Kevin wrote:

> A voice of reason among the group.
>
> We went through this very discussion, in Sigtran, and came out that
> IPSEC works just as well and left the choice to the implementers  
> between
> IPSEC and TLS.  Why not leverage this work?  See RFC
> http://www.ietf.org/rfc/rfc3788.txt, especially sections 7 & 8.
>
>
>
>
> -----Original Message-----
> From: Andrew Newton [mailto:andy@hxr.us]
> Sent: Friday, June 30, 2006 9:31 AM
> To: Rohan Mahy
> Cc: Khan, Sohel Q [CTO]; speermint@ietf.org; Uzelac, Adam
> Subject: Re: TLS for peering (was RE:
> Comments(RE:[speermint]requirementsdraft submitted))
>
> Rohan,
>
> I am not a liberty to give specific numbers, but I can assure you
> that your assumptions are wrong.  We have a lot of very expensive
> equipment that would fall over if it did all signaling over TLS or
> even just TCP.  And we have done the tests and measurements, and a
> $2000 rack mount server cannot handle the load.
>
> And since you are asking for "real requirements", I'd like to know
> what your "real requirements" are for specifying TLS for signaling.
> Specifically, I'd like to know the following:
>
> 1) Why is it you are advocating security of the signaling but not the
> media?  What requirement determines that signaling is important to
> secure but media is not important to secure?
> 2) Why do you believe TLS offers better authentication than any other
> method, especially given that there is no global trust anchor?
> 3) What requirement makes TLS more secure than a dedicated link?
> 4) What requirement makes TLS more secure than IPSec?
> 5) Given that providers interconnect today without TLS, what
> requirement is not met by the current interconnection methods that
> TLS will provide?
>
> Don't get me wrong.  I think all vendors should support TLS, and I
> think providers and carriers should be able to have it at their
> disposal when doing interconnection, but it is only one of many  
> options.
>
> -andy
>
> On Jun 29, 2006, at 8:05 PM, Rohan Mahy wrote:
>
>> Andy,
>>
>> What you've described are a bunch of hacks that service providers use
>> because they don't peer over the Internet or because they can't rely
>> on the other guy to support TLS.  The combination of hacks is much,
>> much more expensive in terms of both capital equipment and
>> operationally than just using TLS.
>>
>> A $2000 rack mount server can encrypt and decrypt a phenominal volume
>> of TLS traffic.  Hardware crypto accellerators are available that
>> increase this volume and decrease initial TLS handshake latencies by
>> another couple of orders of magnitude.
>>
>> I'm willing to bet a few beers that the amount of data that is
>> encrypted using TLS exceeds the total volume of PSTN signaling  
>> traffic
>> on a daily basis.
>>
>> To those who have a problem with TLS, I ask them to provide some
>> quantitative data or real requirements rather than rhetoric and FUD.
>>
>> thanks,
>> -rohan
>>
>>
>>
>>
>>
>> On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
>>>
>>> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
>>>> how do you propose to establish a trust relationship between SIP
>>>> domain
>>>> boundaries without TLS?
>>>
>>> There are other methods: security lower in the stack, private
>>> connections, address ACLs...  the things that providers use today.
>>>
>>>> If you cannot establish a trust relationship between
>>> interconnecting
>>>> partners then what is the purpose of this whole exercise?
>>>
>>> To answer the second part of that sentence: I don't know; I've been
>>> asking that question for quite some time.
>>>
>>> -andy
>>>
>>> _______________________________________________
>>> Speermint mailing list
>>> Speermint@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/speermint
>>>
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>
>
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> https://www1.ietf.org/mailman/listinfo/speermint
>


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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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On Jun 28, 2006, at 3:46 AM, Livingood, Jason wrote:

> So in the case of voice, the question at hand is - is there a lowest
> common denominator voice codec which should be required to do peering?

Don't know what you should do but ... pretty much every voice device  
does G.711 so the "cost" of mandating at least that is pretty low.

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Subject: Re: [Speermint] I-D ACTION:draft-ietf-speermint-requirements-00.t xt
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On Jun 30, 2006, at 9:16 AM, Cullen Jennings wrote:

>
> On Jun 28, 2006, at 3:46 AM, Livingood, Jason wrote:
>
>> So in the case of voice, the question at hand is - is there a lowest
>> common denominator voice codec which should be required to do  
>> peering?
>
> Don't know what you should do but ... pretty much every voice  
> device does G.711 so the "cost" of mandating at least that is  
> pretty low.
>

I clearly had not had enough coffee before hitting send on this  
message. Please ignore this this complete wrong statement :-) Sorry.

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On Jun 30, 2006, at 12:00 PM, Cullen Jennings wrote:
> My real concern is IETF is acting as if we will move to TCP,  
> deployments are acting as if this is not possible. This mismatch  
> seems like a growing problem for us and I am concerned about it. I  
> think that this speermint WG has operators that are connected with  
> the real issues so this seems as good a place as any to discuss this.

For us the UDP vs. TCP issue is much more important on the registrar  
part of the network thanks to the ever cursed NAT.  But even with  
peering, UDP is better performance wise.  Persistent TCP connections  
also tend to cause problems with load balancing and fault tolerance  
schemes, which is a big issue.

I do agree that TCP is important and strategies will need to be  
developed to better handle it.

-andy


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On Jun 30, 2006, at 7:30 AM, Andrew Newton wrote:

>
> I am not a liberty to give specific numbers, but I can assure you  
> that your assumptions are wrong.  We have a lot of very expensive  
> equipment that would fall over if it did all signaling over TLS or  
> even just TCP.  And we have done the tests and measurements, and a  
> $2000 rack mount server cannot handle the load.

Andy, this gets to another topic which is possible even more or  
important than this TLS or not questions. the questions is TCP or UDP  
for proxy to proxy.

I have a growing concern of a theory vs. practice mismatch here. I  
don't know if you saw the recent thread on MMUSIC but Magnus  
constructed a very plausible SDP message that could occur in real  
world that is over 10k long (that would be 10k bytes in practice but  
10,000 octets in theory :-).  Many of the deployments I am aware of  
use UDP and say there are some serious problems with TCP. I believe  
this but as much as possible would like to understand what they are.  
On the other hand, the IETF is more and more defining stuff in SIP  
that is pushing the size of SIP packets to the point where it is  
questionable if they will work on UDP given the fragmentation issues.  
Some of the concerns I have see about TCP have been based on the  
ideas each transaction would form a new TCP connection - this clearly  
won't work so let's base the conversation on long lived TCP  
connections between peering proxies.

My real concern is IETF is acting as if we will move to TCP,  
deployments are acting as if this is not possible. This mismatch  
seems like a growing problem for us and I am concerned about it. I  
think that this speermint WG has operators that are connected with  
the real issues so this seems as good a place as any to discuss this.

Cullen (BTW, all these emails sent with my individual contributor hat  
on)

PS - note TLS can be run on TCP or over UDP with DTLS so the TLS or  
not question is separate from the UDP/TCP question

PPS - just to head off the inevitable rat hole ahead ... Yes, I know  
about SCTP. On this thread I'm interested in what are the problems  
with deploying TCP and not really interested in all the great things  
SCTP can provide - we can get to that later. 

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I've been reading the thread about TLS and speermint with
interest and thought I'd add my $.02.

1. In Web-based TLS settings, the performance is dominated by the
   public key crypto in the handshake (typically RSA). The 
   canonical reference on this is [CDW02], which obtained
   about 250cps on a 933 MHz machine with semi-realistic
   simulated workloads. Extrapolating, a 3.8 GHz single-core
   machine would give you order 1000 cps.

2. As I understand it, this setting is mostly long-lived 
   connections so the dominating factor isn't connection setup 
   but rather throughput. Microbenchmarks show performance
   something like:

   RC4   -- 125 MB/s
   SHA-1 -- 10-116 MB/s (dependent on packet size)


3. We've run HTTPS benchmarks on large files with high rates of session
   resumption (which removes the effect of the handshake) and get
   throughputs of about 33 MB/s on low-end server class machines
   (3.0 GHz P4s running FreeBSD). For comparison, we get about
   60 MB/s running pure HTTP traffic with otherwise equivalent
   parameters. Note that this isn't perfectly realistic because
   the performance of HTTPS partly depends on record size, so if
   you're using a lot of small (< 1k, say) messages, you'd see
   inferior performance.

4. I would expect TLS and IPsec performance to be roughly 
   equivalent, provided that you used the same algorithms.
   TLS handshake performance is somewhat better because it doesn't
   require a DH exchange, but as indicated above, that cost 
   probably isn't an issue here.

   There has been some work on comparing SSL and IPsec bulk 
   performance (E.g., [AS05]), which seems to indicate that
   SSL is faster but I haven't studied the methodology
   enough to be comfortable making that claim.

Hope this helps,
-Ekr

   

[CDW02] Cristian Coarfa, Peter Druschel, and Dan S. Wallach, "Performance
Analysis of TLS Web Servers", Network and Distributed Systems Security
Symposium (San Diego, California), February 2002.
http://www.cs.rice.edu/~dwallach/pub/ndss2002.html

[AS05] Abdel Nasir Alshamsi and Takamichi Saito, "A Technical Comparison
of IPSec and SSL", AINA 2005. 

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Subject: Re: TLS for peering (was RE: Comments
	(RE:[speermint]requirementsdraft submitted))
From: Daryl Malas <daryl@level3.net>
To: Rohan Mahy <rohan.mahy@gmail.com>
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Date: Fri, 30 Jun 2006 11:36:09 -0600
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Okay, we have gone round and round on this topic....I have become lost
in the discussion and want to make sure we are still arguing the same
question.  Is the question we are arguing, still:

--Should TCP/TLS be a MUST requirement for peering authentication, or
can it be a SHOULD requirement (made up of multiple potential methods)
for peering?

If this is the right question being argued, what is the problem in
compromising on allowing it to be an acceptable alternative for securing
a peering relationship?

Why does it have to be the only option?

I still assert, it should be included in the requirements document as a
one of several methods (one of the methods is required -- but not all)
for providing security in a peering topology.  This being said...it
would be to the detriment of a provider not willing to peer with another
provider if that provider only supports TLS (yet they would be compliant
with the requirements).  I can say we (Level 3) would seriously consider
implementing it (if we hadn't already) and supporting it based on the
opportunity.  But, before an opportunity presents itself, TCP/TLS
creates a lot of (arguably) unnecessary overhead.

--Daryl


On Fri, 2006-06-30 at 08:40 -0700, Rohan Mahy wrote:
> Andy,
> 
> I am just leaving for a short vacation, but I will commit to answering
> your questions before the meeting in Montreal.  I think these are good
> questions and I want to take the time to answer them properly.
> 
> So that we can make apples to apples comparisons, it would be useful
> to start with an order of magnitude number of calls per day.  This
> does not need to be Sunrocket's (or any other carrier's) actual
> current or expected peering traffic, but it should be plausible.
> 
> thanks,
> -rohan
> 
> 
> On 6/30/06, Andrew Newton <andy@hxr.us> wrote:
> > Rohan,
> >
> > I am not a liberty to give specific numbers, but I can assure you
> > that your assumptions are wrong.  We have a lot of very expensive
> > equipment that would fall over if it did all signaling over TLS or
> > even just TCP.  And we have done the tests and measurements, and a
> > $2000 rack mount server cannot handle the load.
> >
> > And since you are asking for "real requirements", I'd like to know
> > what your "real requirements" are for specifying TLS for signaling.
> > Specifically, I'd like to know the following:
> >
> > 1) Why is it you are advocating security of the signaling but not the
> > media?  What requirement determines that signaling is important to
> > secure but media is not important to secure?
> > 2) Why do you believe TLS offers better authentication than any other
> > method, especially given that there is no global trust anchor?
> > 3) What requirement makes TLS more secure than a dedicated link?
> > 4) What requirement makes TLS more secure than IPSec?
> > 5) Given that providers interconnect today without TLS, what
> > requirement is not met by the current interconnection methods that
> > TLS will provide?
> >
> > Don't get me wrong.  I think all vendors should support TLS, and I
> > think providers and carriers should be able to have it at their
> > disposal when doing interconnection, but it is only one of many options.
> >
> > -andy
> >
> > On Jun 29, 2006, at 8:05 PM, Rohan Mahy wrote:
> >
> > > Andy,
> > >
> > > What you've described are a bunch of hacks that service providers use
> > > because they don't peer over the Internet or because they can't rely
> > > on the other guy to support TLS.  The combination of hacks is much,
> > > much more expensive in terms of both capital equipment and
> > > operationally than just using TLS.
> > >
> > > A $2000 rack mount server can encrypt and decrypt a phenominal volume
> > > of TLS traffic.  Hardware crypto accellerators are available that
> > > increase this volume and decrease initial TLS handshake latencies by
> > > another couple of orders of magnitude.
> > >
> > > I'm willing to bet a few beers that the amount of data that is
> > > encrypted using TLS exceeds the total volume of PSTN signaling traffic
> > > on a daily basis.
> > >
> > > To those who have a problem with TLS, I ask them to provide some
> > > quantitative data or real requirements rather than rhetoric and FUD.
> > >
> > > thanks,
> > > -rohan
> > >
> > >
> > >
> > >
> > >
> > > On 6/28/06, Andrew Newton <andy@hxr.us> wrote:
> > >>
> > >> On Jun 28, 2006, at 2:48 PM, Richard Shockey wrote:
> > >> > how do you propose to establish a trust relationship between SIP
> > >> > domain
> > >> > boundaries without TLS?
> > >>
> > >> There are other methods: security lower in the stack, private
> > >> connections, address ACLs...  the things that providers use today.
> > >>
> > >> > If you cannot establish a trust relationship between
> > >> interconnecting
> > >> > partners then what is the purpose of this whole exercise?
> > >>
> > >> To answer the second part of that sentence: I don't know; I've been
> > >> asking that question for quite some time.
> > >>
> > >> -andy
> > >>
> > >> _______________________________________________
> > >> Speermint mailing list
> > >> Speermint@ietf.org
> > >> https://www1.ietf.org/mailman/listinfo/speermint
> > >>
> >
> >
> 
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