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Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
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I have some conflicting thoughts on this.  It should be possible for a phone
to send the call itself direct to the PSAP URI.  The emergency call system
should be able to handle that.

On the other hand, carriers are generally thought of as useful partners by
the emergency call system, and a recommendation that they be bypassed won't
be appreciated by those folks even if there is an engineering argument that
it's more efficient.

Part of the reason is track and trace.  Part of the reason is subscriber
name and address.  The big one is identity.  Carriers provide reasonably
verified identity.

So, on balance, I think phonebcp should say, as it does now, that phones
pass emergency calls to wherever they normally pass outbound calls.
Frankly, the number of special processing steps we're asking phones to do
for emergency calls is getting too high.  I'd rather figure out ways to
remove some special processing, not add more.

Presently phones have to:
*Get location and do LoST lookups
*Watch for emergency dial strings
*Add Geolocation headers, PSAP URIs and service urns
*Disable some features
*Not send bye

If anything, that list is too long.  As with most things like it, the more
special stuff you do for emergencies, the less likely it is to actually work
in an emergency.

Generally, I think phones knowing where they are is a good thing in the big
picture, and having the phone do all dial plan interpretation is the right
thing also.  So the real special processing is the last three.  

Brian



> -----Original Message-----
> From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> Sent: Thursday, January 31, 2008 11:31 PM
> To: Brian Rosen; Hannes Tschofenig; speermint@ietf.org; ECRIT
> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> 
> We need to note the points where we are actually in agreement. Perhaps
> there was an inference that we weren't with respect to the following:
> 
> 1. The access network provider (the one that gets you onto the Internet)
> should not be responsible for providing any kind of emergency call
> processing.
> 
> 2. The access network provider does need to provide a location service
> that can be used for emergency call processing
> 
> 3. The LoST service with the translations for that point of access needs
> to be available from that point of access. Somebody is actually
> responsible for the physical infrastructure that supports it but it
> doesn't have to be the access provider - just so long as it's
> discoverable from the access.
> 
> I am in agreement with these principles - nothing I said below was
> inconsistent with them.
> 
> Direct to PSAP-URI calling is a supported mode of operation, yes? What
> distinguishes between the circumstance where a device does this versus
> trying to send the call through its VSP? It may be because the specific
> implementation of the VoIP client that the caller is using simply
> decides to do that. This includes the possibility that the client
> doesn't know its actually an emergency call and leaves it up to the VSP
> to do the digit analysis. It includes the case where the VoIP client
> doesn't know anything about ECRIT and just sends all calls to the VSP.
> 
> I don't think in this decision that's governed by the nature of the
> access is there? If not, discussion about NATs etc seem irrelevant to
> me. In fact, my standard "nomadic" VoIP service is Skype. In my office,
> our IS masters don't let Skype work. It's very difficult to guarantee
> that arbitrary VSP access is going to work from arbitrary access points.
> It's much simpler to guarantee that SIP calling directly to the relevant
> PSAP URIs for those points of access will always work. I think details
> such as whether an enterprise VoIP server is the thing that makes the
> PSAP call is just that - a detail. In such an environment, the clients
> and the server are part of the same package anyway. It's not at all the
> same situation as bringing Vonage, for example, into the picture when an
> emergency call is made.
> 
> When it comes to phone BCP, it should be quite possible to say that the
> phone client should always work to the local network and direct the call
> directly to the PSAP. This, in my view, is what pure-ECRIT should be
> about. It should also be possible to say that if the phone can only dial
> digits that are processed by the VSP, then subsequent behaviour is out
> of scope (but here's some suggestions). The NENA i2 architecture already
> has this covered and it's being picked up internationally anyway. It's
> an interim architecture; it'll go away as jurisdictions implement LoST
> and IP PSAP services and VoIP clients learn to use those functions. It
> would be preferable that, when the interim solution does disappear, that
> VSPs no longer need to have a role or responsibility for providing
> emergency access. It would be better for all concerned if they did not.
> 
> Here's the logic:
> 
> 1 Phone can't do genuine ECRIT (LoST discovery etc.) call VSP - invoke
> i2.
> 2.Phone does ECRIT but can't find LIS and/or LoST - invoke i2
> 3.Phone can find LIS and LoST - call PSAP directly
> 
> Cheers,
> Martin
> 
> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net]
> Sent: Friday, 1 February 2008 3:56 AM
> To: Dawson, Martin; 'Hannes Tschofenig'; speermint@ietf.org; 'ECRIT'
> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> 
> First my response to the original question:
> 
> IMS is special because the visited network is in the call path.  When
> the
> visited network has active elements in the call path, it makes sense to
> have
> the visited network handle emergency calls.  Note that there are lots
> and
> lots of complications with this.  Take for example, call back.  What you
> want is to have both a Contact URI that will route back to the device
> that
> placed the call, and you need a real AoR you can use to reach the caller
> later on.  The visited network would have to provide both.
> 
> When the visited network is not in the call path, you have to use the
> VSP.
> 
> 
> Then my response to Martin:
> > It's been said that the emergency service wants the subscriber details
> > from the VSP. A caller can have subscriptions to lots of Internet
> > services that also don't need to be involved in the call. Should we
> also
> > make emergency calling dependent on them?
> No.  However, involving the actual carrier that provides the service
> that is
> placing the call is essential.  Bypassing the carrier that provides the
> service that places the call is pretty tough, except in unusual
> circumstances.  IMS is an example of an unusual circumstance because
> they
> put the visited network in the call path.  Very few other systems do
> that.
> 
> I'm very conscious of what we must ask the access network to provide.
> This
> has lots of implications on engineering as well as cost and liability.
> Unless the device itself can locate itself without assistance of the
> access
> network, we ask the access network to provide location (or at least
> location
> assistance).  We ask them to provide access to a LoST server.  I want to
> stop there.  That's all they have to do.
> 
> >
> > I don't think it has to be the same as IMS. IMS requires a
> co-operative
> > E-CSCF that is actually part of the access network. With LoST and a
> > suitable emergency proxy, there doesn't have to be any dependency on
> the
> > access network to also provide call processing.
> IMS is different as discussed above.
> 
> >
> > Not involving the VSP in the call does not mean that the caller cannot
> > provide callback details (in principle, they could provide multiple
> > callback options). I'd also be interested in a discussion around
> > providing an automatic presence subscription from the emergency
> network
> > itself - with a temporary callback number. As long as the emergency
> > calling device is attached to the Internet, it should be able to stay
> > registered with that temporary subscription and also do regular
> location
> > updates to it.
> I think the notion that we can define the characteristics of an
> emergency
> proxy that works for all devices is not feasible.  Let me cite a few
> examples:
> 
> Emergency call from AOL/MSN/Yahoo.  We would have to require the device
> to
> implement SIP/SIMPLE.
> 
> Emergency call from OnStar.  Who is the emergency proxy provider?
> 
> Emergency call from "Help I've fallen and I can't get up" buttons with
> two
> way audio.  They would have to meet all of the phonebcp requirements.
> 
> Call backs are an issue.  You can get a contact URI, but you can't get
> an
> AoR you have any trust in for any call.
> 
> But the biggest problem is that I think any attempt to give the access
> network the obligation and the liability to totally handle emergency
> calls
> will not fly.
> 
> 
> >
> > Many of the most convoluted issues that have arisen in this forum have
> > been around the challenges that having the VSP involved in call
> routing
> > introduces (the need for forest guides etc). Since direct-calling is a
> > valid option, I know I'd prefer an emergency calling client that
> always
> > worked that way.
> 
> 
> 
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comments near end.

Brian Rosen wrote:
> I have some conflicting thoughts on this.  It should be possible for a phone
> to send the call itself direct to the PSAP URI.  The emergency call system
> should be able to handle that.
> 
> On the other hand, carriers are generally thought of as useful partners by
> the emergency call system, and a recommendation that they be bypassed won't
> be appreciated by those folks even if there is an engineering argument that
> it's more efficient.
> 
> Part of the reason is track and trace.  Part of the reason is subscriber
> name and address.  The big one is identity.  Carriers provide reasonably
> verified identity.
> 
> So, on balance, I think phonebcp should say, as it does now, that phones
> pass emergency calls to wherever they normally pass outbound calls.
> Frankly, the number of special processing steps we're asking phones to do
> for emergency calls is getting too high.  I'd rather figure out ways to
> remove some special processing, not add more.
> 
> Presently phones have to:
> *Get location and do LoST lookups
> *Watch for emergency dial strings
> *Add Geolocation headers, PSAP URIs and service urns
> *Disable some features
> *Not send bye
> 
> If anything, that list is too long.  As with most things like it, the more
> special stuff you do for emergencies, the less likely it is to actually work
> in an emergency.
> 
> Generally, I think phones knowing where they are is a good thing in the big
> picture, and having the phone do all dial plan interpretation is the right
> thing also.  So the real special processing is the last three.  

???

The last three are:

 > *Add Geolocation headers, PSAP URIs and service urns

If you expect the phone to be the one that knows where it is, then how 
can you delegate adding the geoloc headers to somebody else?

 > *Disable some features

Well, features that are provided by the SP can be disabled by the SP. 
But features provided by the phone have to be disabled there.

 > *Not send bye

I don't understand what you have in mind here. If you don't disable 
sending bye, then it will be sent on hangup. There is nothing you can do 
in the network to prevent it.

As I see it, the key things the phone must do itself are:
- obtain its location
- recognize the dial string(s) that identify emergency calls and insert 
service URN & geoloc info
- disable features that are implemented on the phone.

The rest could be done by some server on the path.

The recognition of emergency numbers could be a shared responsibility. 
If a number wasn't recognized by the phone, an upstream entity that is 
in a position to recognize it as such could return a 3xx containing the 
service URN. That would clue the phone in to trigger the special processing.

	Paul


> 
> Brian
> 
> 
> 
>> -----Original Message-----
>> From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
>> Sent: Thursday, January 31, 2008 11:31 PM
>> To: Brian Rosen; Hannes Tschofenig; speermint@ietf.org; ECRIT
>> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
>>
>> We need to note the points where we are actually in agreement. Perhaps
>> there was an inference that we weren't with respect to the following:
>>
>> 1. The access network provider (the one that gets you onto the Internet)
>> should not be responsible for providing any kind of emergency call
>> processing.
>>
>> 2. The access network provider does need to provide a location service
>> that can be used for emergency call processing
>>
>> 3. The LoST service with the translations for that point of access needs
>> to be available from that point of access. Somebody is actually
>> responsible for the physical infrastructure that supports it but it
>> doesn't have to be the access provider - just so long as it's
>> discoverable from the access.
>>
>> I am in agreement with these principles - nothing I said below was
>> inconsistent with them.
>>
>> Direct to PSAP-URI calling is a supported mode of operation, yes? What
>> distinguishes between the circumstance where a device does this versus
>> trying to send the call through its VSP? It may be because the specific
>> implementation of the VoIP client that the caller is using simply
>> decides to do that. This includes the possibility that the client
>> doesn't know its actually an emergency call and leaves it up to the VSP
>> to do the digit analysis. It includes the case where the VoIP client
>> doesn't know anything about ECRIT and just sends all calls to the VSP.
>>
>> I don't think in this decision that's governed by the nature of the
>> access is there? If not, discussion about NATs etc seem irrelevant to
>> me. In fact, my standard "nomadic" VoIP service is Skype. In my office,
>> our IS masters don't let Skype work. It's very difficult to guarantee
>> that arbitrary VSP access is going to work from arbitrary access points.
>> It's much simpler to guarantee that SIP calling directly to the relevant
>> PSAP URIs for those points of access will always work. I think details
>> such as whether an enterprise VoIP server is the thing that makes the
>> PSAP call is just that - a detail. In such an environment, the clients
>> and the server are part of the same package anyway. It's not at all the
>> same situation as bringing Vonage, for example, into the picture when an
>> emergency call is made.
>>
>> When it comes to phone BCP, it should be quite possible to say that the
>> phone client should always work to the local network and direct the call
>> directly to the PSAP. This, in my view, is what pure-ECRIT should be
>> about. It should also be possible to say that if the phone can only dial
>> digits that are processed by the VSP, then subsequent behaviour is out
>> of scope (but here's some suggestions). The NENA i2 architecture already
>> has this covered and it's being picked up internationally anyway. It's
>> an interim architecture; it'll go away as jurisdictions implement LoST
>> and IP PSAP services and VoIP clients learn to use those functions. It
>> would be preferable that, when the interim solution does disappear, that
>> VSPs no longer need to have a role or responsibility for providing
>> emergency access. It would be better for all concerned if they did not.
>>
>> Here's the logic:
>>
>> 1 Phone can't do genuine ECRIT (LoST discovery etc.) call VSP - invoke
>> i2.
>> 2.Phone does ECRIT but can't find LIS and/or LoST - invoke i2
>> 3.Phone can find LIS and LoST - call PSAP directly
>>
>> Cheers,
>> Martin
>>
>> -----Original Message-----
>> From: Brian Rosen [mailto:br@brianrosen.net]
>> Sent: Friday, 1 February 2008 3:56 AM
>> To: Dawson, Martin; 'Hannes Tschofenig'; speermint@ietf.org; 'ECRIT'
>> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
>>
>> First my response to the original question:
>>
>> IMS is special because the visited network is in the call path.  When
>> the
>> visited network has active elements in the call path, it makes sense to
>> have
>> the visited network handle emergency calls.  Note that there are lots
>> and
>> lots of complications with this.  Take for example, call back.  What you
>> want is to have both a Contact URI that will route back to the device
>> that
>> placed the call, and you need a real AoR you can use to reach the caller
>> later on.  The visited network would have to provide both.
>>
>> When the visited network is not in the call path, you have to use the
>> VSP.
>>
>>
>> Then my response to Martin:
>>> It's been said that the emergency service wants the subscriber details
>>> from the VSP. A caller can have subscriptions to lots of Internet
>>> services that also don't need to be involved in the call. Should we
>> also
>>> make emergency calling dependent on them?
>> No.  However, involving the actual carrier that provides the service
>> that is
>> placing the call is essential.  Bypassing the carrier that provides the
>> service that places the call is pretty tough, except in unusual
>> circumstances.  IMS is an example of an unusual circumstance because
>> they
>> put the visited network in the call path.  Very few other systems do
>> that.
>>
>> I'm very conscious of what we must ask the access network to provide.
>> This
>> has lots of implications on engineering as well as cost and liability.
>> Unless the device itself can locate itself without assistance of the
>> access
>> network, we ask the access network to provide location (or at least
>> location
>> assistance).  We ask them to provide access to a LoST server.  I want to
>> stop there.  That's all they have to do.
>>
>>> I don't think it has to be the same as IMS. IMS requires a
>> co-operative
>>> E-CSCF that is actually part of the access network. With LoST and a
>>> suitable emergency proxy, there doesn't have to be any dependency on
>> the
>>> access network to also provide call processing.
>> IMS is different as discussed above.
>>
>>> Not involving the VSP in the call does not mean that the caller cannot
>>> provide callback details (in principle, they could provide multiple
>>> callback options). I'd also be interested in a discussion around
>>> providing an automatic presence subscription from the emergency
>> network
>>> itself - with a temporary callback number. As long as the emergency
>>> calling device is attached to the Internet, it should be able to stay
>>> registered with that temporary subscription and also do regular
>> location
>>> updates to it.
>> I think the notion that we can define the characteristics of an
>> emergency
>> proxy that works for all devices is not feasible.  Let me cite a few
>> examples:
>>
>> Emergency call from AOL/MSN/Yahoo.  We would have to require the device
>> to
>> implement SIP/SIMPLE.
>>
>> Emergency call from OnStar.  Who is the emergency proxy provider?
>>
>> Emergency call from "Help I've fallen and I can't get up" buttons with
>> two
>> way audio.  They would have to meet all of the phonebcp requirements.
>>
>> Call backs are an issue.  You can get a contact URI, but you can't get
>> an
>> AoR you have any trust in for any call.
>>
>> But the biggest problem is that I think any attempt to give the access
>> network the obligation and the liability to totally handle emergency
>> calls
>> will not fly.
>>
>>
>>> Many of the most convoluted issues that have arisen in this forum have
>>> been around the challenges that having the VSP involved in call
>> routing
>>> introduces (the need for forest guides etc). Since direct-calling is a
>>> valid option, I know I'd prefer an emergency calling client that
>> always
>>> worked that way.
>>
>>
>> --------------------------------------------------------------------------
>> ----------------------
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> 
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John,

Thank you for the quick response.  Comments in-line...

-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com] 
Sent: Wednesday, January 30, 2008 12:47 AM
To: Daryl Malas; speermint@ietf.org
Subject: RE: [Speermint] RE: I-D
Action:draft-ietf-speermint-terminology-15.txt

Daryl,

Thanks, it's getting there. A few more things I noticed:

1. The document uses the words "application" and "session" in a number
of places, and I feel there may be some inconsistency. For example:
"A SIP Service Provider (or SSP) is an entity that provides 
   application services utilizing SIP signaling to its customers."
Should it really be "application services", or should it be "session
services". Consider a service provider that provides no more than SIP
proxy functionality. It can hardly be said to provide application
services to its users, so according to the definition it is not an SSP.
There is just one other instance of "application". I wonder if the two
instances should be changed to "session".

[DM] I think the confusion here is SIP is an application-layer control
protocol, which is used to create sessions.  I will check the uses of
application again, and will consider changing to "session".  I do not
see a problem with it.   

2. "In some cases, some entity (usually a 3rd party or federation) 
   provides peering assistance to the originating SSP by providing this 
   function.  The assisting entity may provide information relating to 
   direct (Section 4.2.1) or indirect (Section 4.2.2) peering as 
   necessary."
Two comments. First, should "peering assistance" be changed to
"assistance".
[DM] I think it means the same thing.
Second, this paragraph occurs within 4.3.2 on LRF. Should a similar
statement be added to 4.3.1 on LuF?
[DM] The reason why it is only in the LRF section, is because the LUF is
simply a mechanism.  It is used to query either an internal or an
"assisting" server.  No one can assist with the mechanism.

3. "In this case it is generally required that a 
   trust relationship is established between the originating SSP and the

   transit network on one side; and, the transit network representing 
   the termination network on the other side."
Two comments: First, in order to complement the first part, should the
latter part be rewritten as "; and the transit network and the
terminating SSP on the other side."?
[DM] It appears to me you just would like to change "network" to "SSP",
is this correct?!
Second, on the use of the term "transit network". Is this intentional,
or would it be better to talk about "transit SSP". In this context a
transit network must provide functionality at the session layer
(otherwise it would not be considered as indirect peering), so doesn't
this make it an SSP?
[DM] I agree.  The context of 4.2.2 is Layer 5 peering, so I think both
"network" instances should be corrected to "SSP's".

John

> -----Original Message-----
> From: Daryl Malas [mailto:D.Malas@cablelabs.com]
> Sent: 29 January 2008 21:41
> To: speermint@ietf.org
> Subject: [Speermint] RE: I-D
> Action:draft-ietf-speermint-terminology-15.txt
> 
> This version of the draft includes updates to the following sections:
> 
> Section 2. Context
> Section 4.2.3 Assisted Peering (REMOVED) Section 4.3.1 Location 
> Routing Function Section 4.3.2 Location Routing Function (Previously 
> Look-Up Function) Section 5 - Minor add to "Common dynamic policies"
> Section 9 & 10
> 
> --Daryl
> 
> 
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Tuesday, January 29, 2008 2:10 PM
> To: i-d-announce@ietf.org
> Cc: speermint@ietf.org
> Subject: [Speermint] I-D
> Action:draft-ietf-speermint-terminology-15.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 Terminology
> 	Author(s)       : D. Malas, D. Meyer
> 	Filename        : draft-ietf-speermint-terminology-15.txt
> 	Pages           : 12
> 	Date            : 2008-01-29
> 
> This document defines the terminology that is to be used in describing

> Session PEERing for Multimedia INTerconnect (SPEERMINT).
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-termi
> nology-15.
> txt
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Hi Daryl

I have one comment...

>
> 2. "In some cases, some entity (usually a 3rd party or federation)
>   provides peering assistance to the originating SSP by providing this
>   function.  The assisting entity may provide information relating to
>   direct (Section 4.2.1) or indirect (Section 4.2.2) peering as
>   necessary."
> Two comments. First, should "peering assistance" be changed to
> "assistance".
> [DM] I think it means the same thing.
> Second, this paragraph occurs within 4.3.2 on LRF. Should a similar
> statement be added to 4.3.1 on LuF?
> [DM] The reason why it is only in the LRF section, is because the  
> LUF is
> simply a mechanism.  It is used to query either an internal or an
> "assisting" server.  No one can assist with the mechanism.
>

DS: I agree with John on this one. In the same way that assistance can  
be offered for LRF it can be offered for LUF. You yourself state "It  
is used to query either an internal or an assisting server". To me -  
this is an external server providing "assistance" for the LUF.

D.
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Date: Mon, 04 Feb 2008 08:56:32 +0000
From: "Elwell, John" <john.elwell@siemens.com>
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Daryl,

See below.

John 

> -----Original Message-----
> From: Daryl Malas [mailto:D.Malas@cablelabs.com] 
> Sent: 01 February 2008 22:10
> To: Elwell, John; speermint@ietf.org
> Subject: RE: [Speermint] RE: I-D 
> Action:draft-ietf-speermint-terminology-15.txt
> 
> John,
> 
> Thank you for the quick response.  Comments in-line...
> 
> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Wednesday, January 30, 2008 12:47 AM
> To: Daryl Malas; speermint@ietf.org
> Subject: RE: [Speermint] RE: I-D
> Action:draft-ietf-speermint-terminology-15.txt
> 
> Daryl,
> 
> Thanks, it's getting there. A few more things I noticed:
> 
> 1. The document uses the words "application" and "session" in a number
> of places, and I feel there may be some inconsistency. For example:
> "A SIP Service Provider (or SSP) is an entity that provides 
>    application services utilizing SIP signaling to its customers."
> Should it really be "application services", or should it be "session
> services". Consider a service provider that provides no more than SIP
> proxy functionality. It can hardly be said to provide application
> services to its users, so according to the definition it is 
> not an SSP.
> There is just one other instance of "application". I wonder if the two
> instances should be changed to "session".
> 
> [DM] I think the confusion here is SIP is an application-layer control
> protocol, which is used to create sessions.  I will check the uses of
> application again, and will consider changing to "session".  I do not
> see a problem with it.   
> 
> 2. "In some cases, some entity (usually a 3rd party or federation) 
>    provides peering assistance to the originating SSP by 
> providing this 
>    function.  The assisting entity may provide information 
> relating to 
>    direct (Section 4.2.1) or indirect (Section 4.2.2) peering as 
>    necessary."
> Two comments. First, should "peering assistance" be changed to
> "assistance".
> [DM] I think it means the same thing.
> Second, this paragraph occurs within 4.3.2 on LRF. Should a similar
> statement be added to 4.3.1 on LuF?
> [DM] The reason why it is only in the LRF section, is because 
> the LUF is
> simply a mechanism.  It is used to query either an internal or an
> "assisting" server.  No one can assist with the mechanism.
> 
> 3. "In this case it is generally required that a 
>    trust relationship is established between the originating 
> SSP and the
> 
>    transit network on one side; and, the transit network representing 
>    the termination network on the other side."
> Two comments: First, in order to complement the first part, should the
> latter part be rewritten as "; and the transit network and the
> terminating SSP on the other side."?
> [DM] It appears to me you just would like to change "network" 
> to "SSP",
> is this correct?!
[JRE] My proposed wording change went further than that. The problem is
that for the "on one side" case the present words talk about the two
networks between which a trust relationship is established, but for the
"on the other side" case the present words mention only a single
network, which doesn't make sense.

> Second, on the use of the term "transit network". Is this intentional,
> or would it be better to talk about "transit SSP". In this context a
> transit network must provide functionality at the session layer
> (otherwise it would not be considered as indirect peering), so doesn't
> this make it an SSP?
> [DM] I agree.  The context of 4.2.2 is Layer 5 peering, so I 
> think both
> "network" instances should be corrected to "SSP's".
> 
> John
> 
> > -----Original Message-----
> > From: Daryl Malas [mailto:D.Malas@cablelabs.com]
> > Sent: 29 January 2008 21:41
> > To: speermint@ietf.org
> > Subject: [Speermint] RE: I-D
> > Action:draft-ietf-speermint-terminology-15.txt
> > 
> > This version of the draft includes updates to the following 
> sections:
> > 
> > Section 2. Context
> > Section 4.2.3 Assisted Peering (REMOVED) Section 4.3.1 Location 
> > Routing Function Section 4.3.2 Location Routing Function 
> (Previously 
> > Look-Up Function) Section 5 - Minor add to "Common dynamic policies"
> > Section 9 & 10
> > 
> > --Daryl
> > 
> > 
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > Sent: Tuesday, January 29, 2008 2:10 PM
> > To: i-d-announce@ietf.org
> > Cc: speermint@ietf.org
> > Subject: [Speermint] I-D
> > Action:draft-ietf-speermint-terminology-15.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 Terminology
> > 	Author(s)       : D. Malas, D. Meyer
> > 	Filename        : draft-ietf-speermint-terminology-15.txt
> > 	Pages           : 12
> > 	Date            : 2008-01-29
> > 
> > This document defines the terminology that is to be used in 
> describing
> 
> > Session PEERing for Multimedia INTerconnect (SPEERMINT).
> > 
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-speermint-termi
> > nology-15.
> > txt
> > 
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Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
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I think we will have to agree to disagree.  I noticed you ignored identity.

Anyone else think the only way emergency calls should (normally) work is to
have all devices place emergency calls direct to PSAPs with (normally) no
proxies?

Brian

> -----Original Message-----
> From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> Sent: Sunday, February 03, 2008 10:20 PM
> To: Brian Rosen; Hannes Tschofenig; speermint@ietf.org; ECRIT
> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> 
> Hi Brian,
> 
> This seems to suggest that there's a complexity argument for saying
> phones should pass emergency calls however they "normally" make calls.
> 
> If we have a precept that says it must always be possible to make a call
> directly to the PSAP URI from a given point of access, then also
> supporting call routing via the VSP can only add complexity. Issues such
> as how do VSPs that don't use SIP work, or recognizing dial strings when
> the VSP doesn't know what the intent and regional conventions are, or
> the VSP finding the right LoST server when it's not in a position to
> perform local discovery are all complexities that arise by not sending
> the call directly to the PSAP URI. They are all things that make the
> handling of an emergency call more brittle.
> 
> The question of what is "normal" can only become more fuzzy over time as
> well. Increasingly we just have IP-enabled multimedia devices that can
> have an arbitrary number of service clients/subscriptions hosted on
> them. I think this is really the environment that ECRIT should be
> focussed on. It's the device that should know how to initiate and
> support emergency calling - not the potential multitude of other service
> clients hosted on that device.
> 
> Ideally, all devices should have practically the same implementation
> that knows how to do local location and LoST service discovery and to
> subsequently contact the local emergency service. Dedicated VSP client
> devices could integrate such a reference implementation, provide a
> mechanism to call it, and only invoke VSP-specific emergency handling if
> (as I described below) the local network does not appear to support the
> necessary ECRIT functions. This should result in less complexity over
> time with more and more consistent implementations from the device
> perspective as well as the network perspective. As long as the reference
> client works, then the network is doing the right thing.
> 
> "Track and trace" become Internet access issues. We get to the point
> where we have optimal assurance that the access is local to the
> emergency jurisdiction and we have all the same wherewithal with respect
> to this function as we do with the PSTN - without needing to think that
> VSPs are important to it.
> 
> I think everything that ECRIT has is just about right. I just think that
> the mantra about "normal call processing" has things the wrong way
> around. i.e I have the opposite view to what you expressed, Brian.
> Ultimately, I think VSP involvement should go away - and not actually be
> the recommended approach in perpetuity. The VSP considerations should
> become recommendations that are ultimately deprecated.
> 
> Note - I understand very well that redoing documentation and recanting
> on policy is a demotivator. However, now is always the best time to
> change if we don't have things quite right.
> 
> Cheers,
> Martin
> 
> -----Original Message-----
> From: ecrit-bounces@ietf.org [mailto:ecrit-bounces@ietf.org] On Behalf
> Of Brian Rosen
> Sent: Saturday, 2 February 2008 12:28 AM
> To: Dawson, Martin; 'Hannes Tschofenig'; speermint@ietf.org; 'ECRIT'
> Subject: Re: [Ecrit] Emergency Calls & VoIP Peering
> 
> I have some conflicting thoughts on this.  It should be possible for a
> phone
> to send the call itself direct to the PSAP URI.  The emergency call
> system
> should be able to handle that.
> 
> On the other hand, carriers are generally thought of as useful partners
> by
> the emergency call system, and a recommendation that they be bypassed
> won't
> be appreciated by those folks even if there is an engineering argument
> that
> it's more efficient.
> 
> Part of the reason is track and trace.  Part of the reason is subscriber
> name and address.  The big one is identity.  Carriers provide reasonably
> verified identity.
> 
> So, on balance, I think phonebcp should say, as it does now, that phones
> pass emergency calls to wherever they normally pass outbound calls.
> Frankly, the number of special processing steps we're asking phones to
> do
> for emergency calls is getting too high.  I'd rather figure out ways to
> remove some special processing, not add more.
> 
> Presently phones have to:
> *Get location and do LoST lookups
> *Watch for emergency dial strings
> *Add Geolocation headers, PSAP URIs and service urns
> *Disable some features
> *Not send bye
> 
> If anything, that list is too long.  As with most things like it, the
> more
> special stuff you do for emergencies, the less likely it is to actually
> work
> in an emergency.
> 
> Generally, I think phones knowing where they are is a good thing in the
> big
> picture, and having the phone do all dial plan interpretation is the
> right
> thing also.  So the real special processing is the last three.
> 
> Brian
> 
> 
> 
> > -----Original Message-----
> > From: Dawson, Martin [mailto:Martin.Dawson@andrew.com]
> > Sent: Thursday, January 31, 2008 11:31 PM
> > To: Brian Rosen; Hannes Tschofenig; speermint@ietf.org; ECRIT
> > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> >
> > We need to note the points where we are actually in agreement. Perhaps
> > there was an inference that we weren't with respect to the following:
> >
> > 1. The access network provider (the one that gets you onto the
> Internet)
> > should not be responsible for providing any kind of emergency call
> > processing.
> >
> > 2. The access network provider does need to provide a location service
> > that can be used for emergency call processing
> >
> > 3. The LoST service with the translations for that point of access
> needs
> > to be available from that point of access. Somebody is actually
> > responsible for the physical infrastructure that supports it but it
> > doesn't have to be the access provider - just so long as it's
> > discoverable from the access.
> >
> > I am in agreement with these principles - nothing I said below was
> > inconsistent with them.
> >
> > Direct to PSAP-URI calling is a supported mode of operation, yes? What
> > distinguishes between the circumstance where a device does this versus
> > trying to send the call through its VSP? It may be because the
> specific
> > implementation of the VoIP client that the caller is using simply
> > decides to do that. This includes the possibility that the client
> > doesn't know its actually an emergency call and leaves it up to the
> VSP
> > to do the digit analysis. It includes the case where the VoIP client
> > doesn't know anything about ECRIT and just sends all calls to the VSP.
> >
> > I don't think in this decision that's governed by the nature of the
> > access is there? If not, discussion about NATs etc seem irrelevant to
> > me. In fact, my standard "nomadic" VoIP service is Skype. In my
> office,
> > our IS masters don't let Skype work. It's very difficult to guarantee
> > that arbitrary VSP access is going to work from arbitrary access
> points.
> > It's much simpler to guarantee that SIP calling directly to the
> relevant
> > PSAP URIs for those points of access will always work. I think details
> > such as whether an enterprise VoIP server is the thing that makes the
> > PSAP call is just that - a detail. In such an environment, the clients
> > and the server are part of the same package anyway. It's not at all
> the
> > same situation as bringing Vonage, for example, into the picture when
> an
> > emergency call is made.
> >
> > When it comes to phone BCP, it should be quite possible to say that
> the
> > phone client should always work to the local network and direct the
> call
> > directly to the PSAP. This, in my view, is what pure-ECRIT should be
> > about. It should also be possible to say that if the phone can only
> dial
> > digits that are processed by the VSP, then subsequent behaviour is out
> > of scope (but here's some suggestions). The NENA i2 architecture
> already
> > has this covered and it's being picked up internationally anyway. It's
> > an interim architecture; it'll go away as jurisdictions implement LoST
> > and IP PSAP services and VoIP clients learn to use those functions. It
> > would be preferable that, when the interim solution does disappear,
> that
> > VSPs no longer need to have a role or responsibility for providing
> > emergency access. It would be better for all concerned if they did
> not.
> >
> > Here's the logic:
> >
> > 1 Phone can't do genuine ECRIT (LoST discovery etc.) call VSP - invoke
> > i2.
> > 2.Phone does ECRIT but can't find LIS and/or LoST - invoke i2
> > 3.Phone can find LIS and LoST - call PSAP directly
> >
> > Cheers,
> > Martin
> >
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Friday, 1 February 2008 3:56 AM
> > To: Dawson, Martin; 'Hannes Tschofenig'; speermint@ietf.org; 'ECRIT'
> > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> >
> > First my response to the original question:
> >
> > IMS is special because the visited network is in the call path.  When
> > the
> > visited network has active elements in the call path, it makes sense
> to
> > have
> > the visited network handle emergency calls.  Note that there are lots
> > and
> > lots of complications with this.  Take for example, call back.  What
> you
> > want is to have both a Contact URI that will route back to the device
> > that
> > placed the call, and you need a real AoR you can use to reach the
> caller
> > later on.  The visited network would have to provide both.
> >
> > When the visited network is not in the call path, you have to use the
> > VSP.
> >
> >
> > Then my response to Martin:
> > > It's been said that the emergency service wants the subscriber
> details
> > > from the VSP. A caller can have subscriptions to lots of Internet
> > > services that also don't need to be involved in the call. Should we
> > also
> > > make emergency calling dependent on them?
> > No.  However, involving the actual carrier that provides the service
> > that is
> > placing the call is essential.  Bypassing the carrier that provides
> the
> > service that places the call is pretty tough, except in unusual
> > circumstances.  IMS is an example of an unusual circumstance because
> > they
> > put the visited network in the call path.  Very few other systems do
> > that.
> >
> > I'm very conscious of what we must ask the access network to provide.
> > This
> > has lots of implications on engineering as well as cost and liability.
> > Unless the device itself can locate itself without assistance of the
> > access
> > network, we ask the access network to provide location (or at least
> > location
> > assistance).  We ask them to provide access to a LoST server.  I want
> to
> > stop there.  That's all they have to do.
> >
> > >
> > > I don't think it has to be the same as IMS. IMS requires a
> > co-operative
> > > E-CSCF that is actually part of the access network. With LoST and a
> > > suitable emergency proxy, there doesn't have to be any dependency on
> > the
> > > access network to also provide call processing.
> > IMS is different as discussed above.
> >
> > >
> > > Not involving the VSP in the call does not mean that the caller
> cannot
> > > provide callback details (in principle, they could provide multiple
> > > callback options). I'd also be interested in a discussion around
> > > providing an automatic presence subscription from the emergency
> > network
> > > itself - with a temporary callback number. As long as the emergency
> > > calling device is attached to the Internet, it should be able to
> stay
> > > registered with that temporary subscription and also do regular
> > location
> > > updates to it.
> > I think the notion that we can define the characteristics of an
> > emergency
> > proxy that works for all devices is not feasible.  Let me cite a few
> > examples:
> >
> > Emergency call from AOL/MSN/Yahoo.  We would have to require the
> device
> > to
> > implement SIP/SIMPLE.
> >
> > Emergency call from OnStar.  Who is the emergency proxy provider?
> >
> > Emergency call from "Help I've fallen and I can't get up" buttons with
> > two
> > way audio.  They would have to meet all of the phonebcp requirements.
> >
> > Call backs are an issue.  You can get a contact URI, but you can't get
> > an
> > AoR you have any trust in for any call.
> >
> > But the biggest problem is that I think any attempt to give the access
> > network the obligation and the liability to totally handle emergency
> > calls
> > will not fly.
> >
> >
> > >
> > > Many of the most convoluted issues that have arisen in this forum
> have
> > > been around the challenges that having the VSP involved in call
> > routing
> > > introduces (the need for forest guides etc). Since direct-calling is
> a
> > > valid option, I know I'd prefer an emergency calling client that
> > always
> > > worked that way.
> >
> >
> >
> >
> ------------------------------------------------------------------------
> --
> > ----------------------
> > This message is for the designated recipient only and may
> > contain privileged, proprietary, or otherwise private information.
> > If you have received it in error, please notify the sender
> > immediately and delete the original.  Any unauthorized use of
> > this email is prohibited.
> >
> ------------------------------------------------------------------------
> --
> > ----------------------
> > [mf2]
> 
> _______________________________________________
> Ecrit mailing list
> Ecrit@ietf.org
> http://www.ietf.org/mailman/listinfo/ecrit
> 
> --------------------------------------------------------------------------
> ----------------------
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From: Duane <duane@e164.org>
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Brian Rosen wrote:
> I think we will have to agree to disagree.  I noticed you ignored identity.
> 
> Anyone else think the only way emergency calls should (normally) work is to
> have all devices place emergency calls direct to PSAPs with (normally) no
> proxies?

(sorry if I have duplicated what others have written, I'm trying to
catch up on emails/this thread)

This is a double edged sword, it reduces the risk or problems caused
when any of the hops in the middle fail especially if there is a lack of
redundancy, this would/should have better call quality if there is no
codec conversions occurring in the middle as well.

That said however setups I'd suggest putting the smarts at the emergency
side of things rather then the phone and you could deal with calls in
some kind of smart fashion ideally utilising geoip information so that
only people within x km distance can make those calls, otherwise extra
work would be generated for no ones benefit due to abuse.

You could go one step further and have calls go to a central switch that
redirects the SIP calls based on IP location etc.

Also I really like how all GSM carriers honour (or were forced to
honour) calls to 112, which can redirected to the local emergency number
and I really wish carriers the world over could come up with a similar
thing, not just for VoIP but all phone services in general, because as
the number of international travelers increase, how many keep up with
the local emergency number?

Then of course you could possible handle multilingual services this way
too, although for the local language you shouldn't have to wait for the
full prompt.

Even with the above limitations in place you would probably need some
ISP industry consensus on procedures with respect to localised abuse,
this would need to cover things like bot infested PCs generating calls.

No doubt as others have pointed out this would need some kind of
subscriber authentication/information and one way would be real time, or
near real time to subscriber information based on the IP address similar
to databases that store phone number and subscriber information used by
law enforcement agencies etc to cover IPs. Which is just another
slippery slope, especially for countries already in the grips of police
like states.

-- 

Best regards,
 Duane

http://www.freeauth.org - Enterprise Two Factor Authentication
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://e164.org - Because e164.arpa is a tax on VoIP

"In the long run the pessimist may be proved right,
    but the optimist has a better time on the trip."

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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Brian Rosen" <br@brianrosen.net>, "Stark, Barbara" <bs7652@att.com>,
	"Dawson, Martin" <Martin.Dawson@andrew.com>,
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The access network in 3GPP is the part of the network that provides the
radio access and the first level of mobility management. 

From there you have the access network and part of the core network
providing IP connectivity to the first element that appears at the SIP
layer. That IP connectivity can end you up in the local network or the
home network.

I would suggest you are probably looking for something like IMS local
network.

Regards

Keith 

> -----Original Message-----
> From: Brian Rosen [mailto:br@brianrosen.net] 
> Sent: Thursday, January 31, 2008 5:10 PM
> To: 'Stark, Barbara'; 'Dawson, Martin'; 'Hannes Tschofenig'; 
> speermint@ietf.org; 'ECRIT'
> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> 
> I made the same assumption, which I know to be wrong.  Thank 
> you for correcting us.  Is "3GPP Access Network" the right 
> terminology for the kind of network where the visited network 
> provides the E-CSCF?
> 
> Brian
> 
> > -----Original Message-----
> > From: Stark, Barbara [mailto:bs7652@att.com]
> > Sent: Thursday, January 31, 2008 12:05 PM
> > To: Dawson, Martin; Hannes Tschofenig; speermint@ietf.org; ECRIT
> > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> > 
> > snip...
> > > IMS requires a co-operative
> > > E-CSCF that is actually part of the access network.
> > 
> > This is not true. Please don't confuse "IMS" with "3GPP 
> Access Network"
> > and early 3GPP IMS definitions. They aren't the same. It's 
> absolutely 
> > possible to have an IMS core that exists only on the Internet, and 
> > that makes no assumptions about access networks.
> > Barbara
> > 
> > *****
> > 
> > The information transmitted is intended only for the person 
> or entity 
> > to which it is addressed and may contain confidential, proprietary, 
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> > information by persons or entities other than the intended 
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> > is prohibited. If you received this in error, please contact the 
> > sender and delete the material from all computers. GA622
> > 
> > 
> > 
> > _______________________________________________
> > Ecrit mailing list
> > Ecrit@ietf.org
> > https://www1.ietf.org/mailman/listinfo/ecrit
> 
> 
> _______________________________________________
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Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
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> This is a double edged sword, it reduces the risk or problems caused
> when any of the hops in the middle fail especially if there is a lack of
> redundancy, this would/should have better call quality if there is no
> codec conversions occurring in the middle as well.
Putting proxies in the middle doesn't imply something is in the middle of
the media path.  It often is, but the discussion we're having doesn't depend
on that.  Your double edged sword applies to codecs.  How many codecs can we
expect the PSAP to reasonably implement?  Are you forcing ALL devices to
implement at least one of those codecs?  History suggests this won't work.

> 
> That said however setups I'd suggest putting the smarts at the emergency
> side of things rather then the phone and you could deal with calls in
> some kind of smart fashion ideally utilising geoip information so that
> only people within x km distance can make those calls, otherwise extra
> work would be generated for no ones benefit due to abuse.
That makes no sense to me.  The PSAP can't enforce any form of admission
control unless it's under attack, and x km distance doesn't work at all
--service boundaries don't depend on distance from a caller, they depend on
jurisdictional boundaries that are absolute.

> 
> You could go one step further and have calls go to a central switch that
> redirects the SIP calls based on IP location etc.
There is no "central switch" in the Internet.  That is the point of the
Internet.

> 
> Also I really like how all GSM carriers honour (or were forced to
> honour) calls to 112, which can redirected to the local emergency number
> and I really wish carriers the world over could come up with a similar
> thing, not just for VoIP but all phone services in general, because as
> the number of international travelers increase, how many keep up with
> the local emergency number?
Please go back and look at the archives in this list for discussions on
local and home dial strings.  Despite the appeal of the entire world
accepting 1-1-2 as the sole emergency number, it won't happen in your
lifetime.

> 
> Then of course you could possible handle multilingual services this way
> too, although for the local language you shouldn't have to wait for the
> full prompt.
At the moment, that is outside the scope of the work in ecrit.  In NENA, we
plan to allow routing that depends on the language headers in the SIP
messages.

> 
> Even with the above limitations in place you would probably need some
> ISP industry consensus on procedures with respect to localised abuse,
> this would need to cover things like bot infested PCs generating calls.
As I said, identity is the big problem with a direct connection from caller
to PSAP.

> 
> No doubt as others have pointed out this would need some kind of
> subscriber authentication/information and one way would be real time, or
> near real time to subscriber information based on the IP address similar
> to databases that store phone number and subscriber information used by
> law enforcement agencies etc to cover IPs. Which is just another
> slippery slope, especially for countries already in the grips of police
> like states.
So you propose to have credential based authorization to get on the local
access network, map that to some kind of identity that can be put on a SIP
call, assume all bad devices will do that, and check it on entrance to a
PSAP? 

In the end, are you in favor of a direct connection from calling device to
PSAP (no proxy servers other than in the emergency services IP networks)?

Brian

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I'm looking for a term of art when a local access network is IMS aware, and
the local network provides the E-CSCF.  Not all IMS systems do that, right?

Is "IMS local network" a known term of art where I can expect an E-CSCF to
handle emergency calls rather than forwarding to the home domain?

Brian

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> Sent: Monday, February 04, 2008 1:20 PM
> To: Brian Rosen; Stark, Barbara; Dawson, Martin; Hannes Tschofenig;
> speermint@ietf.org; ECRIT
> Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> 
> The access network in 3GPP is the part of the network that provides the
> radio access and the first level of mobility management.
> 
> From there you have the access network and part of the core network
> providing IP connectivity to the first element that appears at the SIP
> layer. That IP connectivity can end you up in the local network or the
> home network.
> 
> I would suggest you are probably looking for something like IMS local
> network.
> 
> Regards
> 
> Keith
> 
> > -----Original Message-----
> > From: Brian Rosen [mailto:br@brianrosen.net]
> > Sent: Thursday, January 31, 2008 5:10 PM
> > To: 'Stark, Barbara'; 'Dawson, Martin'; 'Hannes Tschofenig';
> > speermint@ietf.org; 'ECRIT'
> > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> >
> > I made the same assumption, which I know to be wrong.  Thank
> > you for correcting us.  Is "3GPP Access Network" the right
> > terminology for the kind of network where the visited network
> > provides the E-CSCF?
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Stark, Barbara [mailto:bs7652@att.com]
> > > Sent: Thursday, January 31, 2008 12:05 PM
> > > To: Dawson, Martin; Hannes Tschofenig; speermint@ietf.org; ECRIT
> > > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> > >
> > > snip...
> > > > IMS requires a co-operative
> > > > E-CSCF that is actually part of the access network.
> > >
> > > This is not true. Please don't confuse "IMS" with "3GPP
> > Access Network"
> > > and early 3GPP IMS definitions. They aren't the same. It's
> > absolutely
> > > possible to have an IMS core that exists only on the Internet, and
> > > that makes no assumptions about access networks.
> > > Barbara
> > >
> > > *****
> > >
> > > The information transmitted is intended only for the person
> > or entity
> > > to which it is addressed and may contain confidential, proprietary,
> > > and/or privileged material. Any review, retransmission,
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> > > information by persons or entities other than the intended
> > recipient
> > > is prohibited. If you received this in error, please contact the
> > > sender and delete the material from all computers. GA622
> > >
> > >
> > >
> > > _______________________________________________
> > > Ecrit mailing list
> > > Ecrit@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/ecrit
> >
> >
> > _______________________________________________
> > Ecrit mailing list
> > Ecrit@ietf.org
> > https://www1.ietf.org/mailman/listinfo/ecrit
> >

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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
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Subject: Re: [Speermint] I-D Action:draft-ietf-speermint-terminology-15.txt
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Guys - 

I am at the end of my rope on this one.  John, I appreciate the feedback
but we're really iterating this greatly overdue draft for just you at
this point and I want to accelerate the feedback and close this draft
within one week.  Taking in the feedback via email, waiting for the I-D
editor to publish, then seeing more feedback and interating again is
looking like a 3 - 4 week process...

Daryl - Would you please conatct John privately and arrange a brief con
call to discuss and close any remaining concerns he has.  

John - I appreciate your making yourself available for ~30 mins to close
any remaining concerns you have.  

Thanks all.

Jason

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] 
> Sent: Wednesday, January 30, 2008 2:47 AM
> To: Daryl Malas; speermint@ietf.org
> Subject: RE: [Speermint] RE: I-D 
> Action:draft-ietf-speermint-terminology-15.txt
> 
> Daryl,
> 
> Thanks, it's getting there. A few more things I noticed:
> 
> 1. The document uses the words "application" and "session" in 
> a number of places, and I feel there may be some 
> inconsistency. For example:
> "A SIP Service Provider (or SSP) is an entity that provides 
>    application services utilizing SIP signaling to its customers."
> Should it really be "application services", or should it be 
> "session services". Consider a service provider that provides 
> no more than SIP proxy functionality. It can hardly be said 
> to provide application services to its users, so according to 
> the definition it is not an SSP.
> There is just one other instance of "application". I wonder 
> if the two instances should be changed to "session".
> 
> 2. "In some cases, some entity (usually a 3rd party or federation) 
>    provides peering assistance to the originating SSP by 
> providing this 
>    function.  The assisting entity may provide information 
> relating to 
>    direct (Section 4.2.1) or indirect (Section 4.2.2) peering as 
>    necessary."
> Two comments. First, should "peering assistance" be changed 
> to "assistance".
> Second, this paragraph occurs within 4.3.2 on LRF. Should a 
> similar statement be added to 4.3.1 on LuF?
> 
> 3. "In this case it is generally required that a 
>    trust relationship is established between the originating 
> SSP and the
> 
>    transit network on one side; and, the transit network representing 
>    the termination network on the other side."
> Two comments: First, in order to complement the first part, 
> should the latter part be rewritten as "; and the transit 
> network and the terminating SSP on the other side."?
> Second, on the use of the term "transit network". Is this 
> intentional, or would it be better to talk about "transit 
> SSP". In this context a transit network must provide 
> functionality at the session layer (otherwise it would not be 
> considered as indirect peering), so doesn't this make it an SSP?
> 
> John
> 
> > -----Original Message-----
> > From: Daryl Malas [mailto:D.Malas@cablelabs.com]
> > Sent: 29 January 2008 21:41
> > To: speermint@ietf.org
> > Subject: [Speermint] RE: I-D
> > Action:draft-ietf-speermint-terminology-15.txt
> > 
> > This version of the draft includes updates to the following 
> sections:
> > 
> > Section 2. Context
> > Section 4.2.3 Assisted Peering (REMOVED) Section 4.3.1 Location 
> > Routing Function Section 4.3.2 Location Routing Function 
> (Previously 
> > Look-Up Function) Section 5 - Minor add to "Common dynamic policies"
> > Section 9 & 10
> > 
> > --Daryl
> > 
> > 
> > -----Original Message-----
> > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > Sent: Tuesday, January 29, 2008 2:10 PM
> > To: i-d-announce@ietf.org
> > Cc: speermint@ietf.org
> > Subject: [Speermint] I-D
> > Action:draft-ietf-speermint-terminology-15.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 Terminology
> > 	Author(s)       : D. Malas, D. Meyer
> > 	Filename        : draft-ietf-speermint-terminology-15.txt
> > 	Pages           : 12
> > 	Date            : 2008-01-29
> > 
> > This document defines the terminology that is to be used in 
> describing 
> > Session PEERing for Multimedia INTerconnect (SPEERMINT).
> > 
> > A URL for this Internet-Draft is:
> > http://www.ietf.org/internet-drafts/draft-ietf-speermint-termi
> > nology-15.
> > txt
> > 
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From: "Livingood, Jason" <Jason_Livingood@cable.comcast.com>
To: "Alexander Mayrhofer" <alexander.mayrhofer@enum.at>, <speermint@ietf.org>,
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Sorry - but WGLC for this draft closed on 21 May 2007.  I sent you a
private note on the correction of your name, which I think we can get in
there in time.

Regards
Jason 

> -----Original Message-----
> From: Alexander Mayrhofer [mailto:alexander.mayrhofer@enum.at] 
> Sent: Monday, January 28, 2008 12:37 PM
> To: speermint@ietf.org; avshalom@il.ibm.com; aoki@aol.net; 
> sriram.parameswar@microsoft.com
> Subject: [Speermint] (extremely) late comments 
> ondraft-ietf-speermint-consolidated-presence-im-usecases-03
> 
> 
> Hi,
> 
> i found a couple of nits in the IM-usecases draft. I'm aware 
> that i'm _really_ late, but since i think they improve the 
> document, here you go:
> 
> - Abstract: please remove "(SP)" from the abstract. 
> Definition of acronyms must not happen in the abstract.
> 
> - 1. (Introduction): Please explain the acronym "PSTN" 
> acronym here - it's not on the well known list of abbreviations.
> 
> - 1. (Introduction): the second "(RTC)" (5 lines from end of 
> section) is
>   not needed (is explained in first line already). Please remove it.
> 
> - 2.1. It might be good to mention which user is associated 
> with with peering network - otherwise that becomes a riddle 
> :). So please add something like "Alice@example.com (with 
> example.com hosted by Peer A)..." to the text.
> 
> - 2.2. I think the "Ad-hock" group should be an "Ad-hoc" 
> group. Also, in that paragraph, you say "A public group may 
> be created by Alice..." 
> while i think it should read "An ad-hoc group may be created 
> by Alice..."
> 
> - 2.6. "Fully standardized (became RFCs)" is weak language - 
> as you know, not every RFC is a standard. Also, what is 
> "fully standardized"?
> 
> - 2.7. I think it should start with "A Federation" instead of 
> "Federation". Also, "Peer Networks may connect each other via 
> a peer-to-peer fashion" should probably read "Peer Networks 
> may connect to each other in a peer-to-peer fashion".
> 
> - 2.7. The text "the other Peering Network is who it claims to be" 
> sounds unclear to me. You could replace that text by "that 
> enables each Peering Network to clearly identify Peering 
> partners, and set up secured trust relationships" - but that 
> is just a personal suggestion.
> 
> - 4. I'd replace "We are seeing today users are concerned 
> with providing.." with "We are seeing today users are 
> concerned about providing .." and "domain" with "domain name".
> 
> - 4. The reference to section 2.3 would look better in 
> parentheses "... 
> (see Section 2.3)".
> 
> 
> - 4. Last sentence on page 6 - It would be good to change 
> that to "This document discusses ause cases for peering 
> between Peer Networks. It is out of scope of this document to 
> provide for..."
> 
> - 5. Acknowledgements: nit: My last name is misspelled :)
> 
> - 6.1. Normative References: You don't use RFC 2119 language 
> in the draft - please remove that reference, it's not neccessary.
> 
> 
> That's all, i hope it was helpful (though very late).
> 
> thanks,
> 
> Alex
> 
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From: "Dwight, Timothy M (Tim)" <timothy.dwight@verizonbusiness.com>
In-reply-to: <050b01c8675c$ab529f70$640fa8c0@cis.neustar.com>
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My assumption is that "local network" is fairly well understood to mean
"the network with which the IP-CAN to which the user is currently
attached, is associated".  In other words the local network is the
visited network if you're roaming, your home network otherwise.  

Do all IMS networks provide an E-CSCF?  I doubt any do at the moment,
because the standards are pretty new and not all details are resolved.

If there were an E-CSCF in the "IMS local network" I would expect it to
behave per TS 24.229, which indicates that the call is directed to the
PSAP by the E-CSCF and is NOT forwarded to the home network.  

Tim




> -----Original Message-----
> From: ecrit-bounces@ietf.org [mailto:ecrit-bounces@ietf.org] 
> On Behalf Of Brian Rosen
> Sent: Monday, February 04, 2008 12:35 PM
> To: 'DRAGE, Keith (Keith)'; 'Stark, Barbara'; 'Dawson, 
> Martin'; 'Hannes Tschofenig'; speermint@ietf.org; 'ECRIT'
> Subject: Re: [Ecrit] Emergency Calls & VoIP Peering
> 
> I'm looking for a term of art when a local access network is 
> IMS aware, and
> the local network provides the E-CSCF.  Not all IMS systems 
> do that, right?
> 
> Is "IMS local network" a known term of art where I can expect 
> an E-CSCF to
> handle emergency calls rather than forwarding to the home domain?
> 
> Brian
> 
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) [mailto:drage@alcatel-lucent.com]
> > Sent: Monday, February 04, 2008 1:20 PM
> > To: Brian Rosen; Stark, Barbara; Dawson, Martin; Hannes Tschofenig;
> > speermint@ietf.org; ECRIT
> > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> > 
> > The access network in 3GPP is the part of the network that 
> provides the
> > radio access and the first level of mobility management.
> > 
> > From there you have the access network and part of the core network
> > providing IP connectivity to the first element that appears 
> at the SIP
> > layer. That IP connectivity can end you up in the local 
> network or the
> > home network.
> > 
> > I would suggest you are probably looking for something like 
> IMS local
> > network.
> > 
> > Regards
> > 
> > Keith
> > 
> > > -----Original Message-----
> > > From: Brian Rosen [mailto:br@brianrosen.net]
> > > Sent: Thursday, January 31, 2008 5:10 PM
> > > To: 'Stark, Barbara'; 'Dawson, Martin'; 'Hannes Tschofenig';
> > > speermint@ietf.org; 'ECRIT'
> > > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> > >
> > > I made the same assumption, which I know to be wrong.  Thank
> > > you for correcting us.  Is "3GPP Access Network" the right
> > > terminology for the kind of network where the visited network
> > > provides the E-CSCF?
> > >
> > > Brian
> > >
> > > > -----Original Message-----
> > > > From: Stark, Barbara [mailto:bs7652@att.com]
> > > > Sent: Thursday, January 31, 2008 12:05 PM
> > > > To: Dawson, Martin; Hannes Tschofenig; speermint@ietf.org; ECRIT
> > > > Subject: RE: [Ecrit] Emergency Calls & VoIP Peering
> > > >
> > > > snip...
> > > > > IMS requires a co-operative
> > > > > E-CSCF that is actually part of the access network.
> > > >
> > > > This is not true. Please don't confuse "IMS" with "3GPP
> > > Access Network"
> > > > and early 3GPP IMS definitions. They aren't the same. It's
> > > absolutely
> > > > possible to have an IMS core that exists only on the 
> Internet, and
> > > > that makes no assumptions about access networks.
> > > > Barbara
> > > >
> > > > *****
> > > >
> > > > The information transmitted is intended only for the person
> > > or entity
> > > > to which it is addressed and may contain confidential, 
> proprietary,
> > > > and/or privileged material. Any review, retransmission,
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> > > > information by persons or entities other than the intended
> > > recipient
> > > > is prohibited. If you received this in error, please contact the
> > > > sender and delete the material from all computers. GA622
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Ecrit mailing list
> > > > Ecrit@ietf.org
> > > > https://www1.ietf.org/mailman/listinfo/ecrit
> > >
> > >
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> > >
> 
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I am planning to work on an update of this I-D soon.  Anyone have any
glaring issues that need to be fixed in or addressed by the draft that
they wish to point out?

http://www.ietf.org/internet-drafts/draft-ietf-speermint-srv-naptr-use-0
2.txt

Thanks
Jason
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Duane wrote:

> I am, but it's a very delicate situation that has to extend beyond
> boundaries to try and cover the globe, remember my little pipe dream
> about 112?

I don't buy into your pipe dream. If I want to have a dial plan in my 
office in the USA that uses office numbers as extension numbers, then I 
want to be able to call office 112, and telling me I can't have office 
number 112 because of emergency dialing conventions in Germany isn't 
going to fly with me.

Also, is it just a *complete* string of 112, or is it any number that is 
prefixed with 112? That makes it worse.

IMO my phone should have a dial plan that works the same no matter where 
I take it. If I am in Germany and want to dial somebody in Germany then 
I will do so using the same dialing conventions I use to dial that 
person in Germany when I am at home in the US. And if I have an 
emergency I will invoke that the same (911) no matter where I am, 
because that is the way I expect my phone to work. (Dialing 911 ought to 
get me the local German PSAP if I happen to be in Germany.)

But I am sure Brian has been thru this many times, and what *I* want 
doesn't matter.

	Paul
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Paul Kyzivat wrote:

> I don't buy into your pipe dream. If I want to have a dial plan in my

I'm not in Germany :P

However that small issue aside, any GSM mobile phone any where can dial
112 and it just works, although it doesn't have to be 112, do you also
setup 911 as an extension for kicks too? :)

-- 

Best regards,
 Duane


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Duane wrote:
> Paul Kyzivat wrote:
> 
>> I don't buy into your pipe dream. If I want to have a dial plan in my
> 
> I'm not in Germany :P

Sorry. I wasn't intending to suggest that you were. I only used germany 
as a place where I thought (I'm not certain) 112 was valid.

> However that small issue aside, any GSM mobile phone any where can dial
> 112 and it just works, although it doesn't have to be 112,

I hope it at least must be "112<send>", rather than it initiating the 
call after you press the "2" in "112" ??? At least then it only takes 
one sequence out of play, rather than all sequences with that as a prefix.

If you started imposing such a rule on land lines then I fear it would 
be restricted as a prefix, because many people don't expect to type a 
special <send> key on traditional phones.

> do you also
> setup 911 as an extension for kicks too? :)

I can see how a hotel where 911 isn't special might want to use 911 to 
dial room 11 on the 9th floor.

My primary point is that reserving particular numbers in one dial plan 
because of decisions made in an unrelated context is a very bad idea in 
general. I'm pretty sure that 112 and 911 aren't the only emergency 
numbers in the world. Should we reserve 666 everywhere because somebody 
somewhere decided to use that as their emergency number?

I do have some sympathy for this concept in places like hotels that 
provide phones for use by travelers. They already typically have a 
private dial plan, and can adjust it carefully to allow for all these 
reserved numbers that their guests might be inclined to try.

But I find it less than compelling that *my* cell phone needs to 
recognize 112 because somebody else might try to use it to make an 
emergency call.

	Thanks,
	Paul
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From: "Daryl Malas" <D.Malas@cablelabs.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Livingood, Jason" <Jason_Livingood@cable.comcast.com>,
	<speermint@ietf.org>
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Subject: Re: [Speermint] I-D Action:draft-ietf-speermint-terminology-15.txt
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Jason,

John and I had a call this afternoon to resolve all of the remaining
issues he had with the draft.  As I was just about to submit the rev
(and final)16, I noticed a couple of NIT issues.  I plan to resolve
these tomorrow and upload the version.

Thanks,

Daryl

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Elwell, John
Sent: Tuesday, February 05, 2008 1:22 AM
To: Livingood, Jason; Daryl Malas; speermint@ietf.org
Subject: Re: [Speermint] I-D
Action:draft-ietf-speermint-terminology-15.txt

Jason,

If comments are no longer solicited I will withdraw my comments - please
go ahead and request publication. However, I really don't see why it is
a 3-4 week cycle to bring out a new version - once comments have been
received and resolved, a new version can be published in less than half
an hour, now that we no longer go through the I-D editor.

John  

> -----Original Message-----
> From: Livingood, Jason [mailto:Jason_Livingood@cable.comcast.com]
> Sent: 04 February 2008 18:43
> To: Elwell, John; Daryl Malas; speermint@ietf.org
> Subject: RE: [Speermint] RE: I-D
> Action:draft-ietf-speermint-terminology-15.txt
> 
> Guys -
> 
> I am at the end of my rope on this one.  John, I appreciate the 
> feedback but we're really iterating this greatly overdue draft for 
> just you at this point and I want to accelerate the feedback and close

> this draft within one week.  Taking in the feedback via email, waiting

> for the I-D editor to publish, then seeing more feedback and 
> interating again is looking like a 3 - 4 week process...
> 
> Daryl - Would you please conatct John privately and arrange a brief 
> con call to discuss and close any remaining concerns he has.
> 
> John - I appreciate your making yourself available for ~30 mins to 
> close any remaining concerns you have.
> 
> Thanks all.
> 
> Jason
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: Wednesday, January 30, 2008 2:47 AM
> > To: Daryl Malas; speermint@ietf.org
> > Subject: RE: [Speermint] RE: I-D
> > Action:draft-ietf-speermint-terminology-15.txt
> > 
> > Daryl,
> > 
> > Thanks, it's getting there. A few more things I noticed:
> > 
> > 1. The document uses the words "application" and "session" in a 
> > number of places, and I feel there may be some inconsistency. For 
> > example:
> > "A SIP Service Provider (or SSP) is an entity that provides 
> >    application services utilizing SIP signaling to its customers."
> > Should it really be "application services", or should it be "session

> > services". Consider a service provider that provides no more than 
> > SIP proxy functionality. It can hardly be said to provide 
> > application services to its users, so according to the definition it

> > is not an SSP.
> > There is just one other instance of "application". I wonder if the 
> > two instances should be changed to "session".
> > 
> > 2. "In some cases, some entity (usually a 3rd party or federation) 
> >    provides peering assistance to the originating SSP by providing 
> > this
> >    function.  The assisting entity may provide information relating 
> > to
> >    direct (Section 4.2.1) or indirect (Section 4.2.2) peering as 
> >    necessary."
> > Two comments. First, should "peering assistance" be changed to 
> > "assistance".
> > Second, this paragraph occurs within 4.3.2 on LRF. Should a similar 
> > statement be added to 4.3.1 on LuF?
> > 
> > 3. "In this case it is generally required that a 
> >    trust relationship is established between the originating SSP and

> > the
> > 
> >    transit network on one side; and, the transit network
> representing
> >    the termination network on the other side."
> > Two comments: First, in order to complement the first part, should 
> > the latter part be rewritten as "; and the transit network and the 
> > terminating SSP on the other side."?
> > Second, on the use of the term "transit network". Is this 
> > intentional, or would it be better to talk about "transit SSP". In 
> > this context a transit network must provide functionality at the 
> > session layer (otherwise it would not be considered as indirect 
> > peering), so doesn't this make it an SSP?
> > 
> > John
> > 
> > > -----Original Message-----
> > > From: Daryl Malas [mailto:D.Malas@cablelabs.com]
> > > Sent: 29 January 2008 21:41
> > > To: speermint@ietf.org
> > > Subject: [Speermint] RE: I-D
> > > Action:draft-ietf-speermint-terminology-15.txt
> > > 
> > > This version of the draft includes updates to the following
> > sections:
> > > 
> > > Section 2. Context
> > > Section 4.2.3 Assisted Peering (REMOVED) Section 4.3.1 Location 
> > > Routing Function Section 4.3.2 Location Routing Function
> > (Previously
> > > Look-Up Function) Section 5 - Minor add to "Common
> dynamic policies"
> > > Section 9 & 10
> > > 
> > > --Daryl
> > > 
> > > 
> > > -----Original Message-----
> > > From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> > > Sent: Tuesday, January 29, 2008 2:10 PM
> > > To: i-d-announce@ietf.org
> > > Cc: speermint@ietf.org
> > > Subject: [Speermint] I-D
> > > Action:draft-ietf-speermint-terminology-15.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 Terminology
> > > 	Author(s)       : D. Malas, D. Meyer
> > > 	Filename        : draft-ietf-speermint-terminology-15.txt
> > > 	Pages           : 12
> > > 	Date            : 2008-01-29
> > > 
> > > This document defines the terminology that is to be used in
> > describing
> > > Session PEERing for Multimedia INTerconnect (SPEERMINT).
> > > 
> > > A URL for this Internet-Draft is:
> > > http://www.ietf.org/internet-drafts/draft-ietf-speermint-termi
> > > nology-15.
> > > txt
> > > 
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> > 
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> > 
> 
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Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
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Paul Kyzivat wrote:

> But I find it less than compelling that *my* cell phone needs to
> recognize 112 because somebody else might try to use it to make an
> emergency call.

That wasn't even my original point, I was pointing out travelers, ie
you, go somewhere different, ie Europe or Australia and take your GSM
mobile phone with you and you don't need to know or care what the local
emergency number is, you just have to remember 112.

-- 

Best regards,
 Duane

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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Duane" <duane@e164.org>, "Paul Kyzivat" <pkyzivat@cisco.com>
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GSM phones have a multilayered approach to this process, and the
distinction only works for numbers that are used in conjunction with a
public telecommunications number.

-	Firstly 112 and 911 are always recognised by the phone itself.

-	Secondly a batch of numbers can be configured on the SIM or
UICC. The principle here is that these are the local service numbers
that are used for emergency in foreign countries, but are not used for
other local service number purposes in the home network.

-	Thirdly if the SIM or UICC is not present, then the following
numbers also act the same as 112 and 911, that is 000, 08, 110, 999, 118
and 119.

-	Fourthly the protocol used between the network and the phone
also allows the downloading of emergency numbers, which are assumed to
be the valid emergency numbers for the local network.

However as soon as you get to private numbering plan numbers, then these
will frequently conflict, unless you have a means of distinguishing
numbers that appear in the private number plan from local service
numbers that appear in the public number plan. This can be done in ISDN,
and for tel URIs, but cannot be performed for an analogue phone sending
LD or DTMF.

For them not to conflict, any potential emergency number has to be
assigned as a valid number in that particular private numbering plan as
well as being a local service number in the public numbering plan. This
is something that owners of hotels, for example, are unlikely to do
without rebuilding the hotel minus room 112.

Finally, in response to Paul

>	I hope it at least must be "112<send>", rather than it
initiating the call after you 
>	press the "2" in "112" ??? At least then it only takes one
sequence out of play, rather 
>	than all sequences with that as a prefix.

Treating "112" and "1120" as two different valid numbers is something
that I have heard of only being used in the US. It has taken ages to
explain that people sometimes do it to colleagues outside the US.


Regards


Keith


> -----Original Message-----
> From: speermint-bounces@ietf.org 
> [mailto:speermint-bounces@ietf.org] On Behalf Of Duane
> Sent: Wednesday, February 06, 2008 3:10 AM
> To: Paul Kyzivat
> Cc: speermint@ietf.org
> Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
> 
> Paul Kyzivat wrote:
> 
> > But I find it less than compelling that *my* cell phone needs to 
> > recognize 112 because somebody else might try to use it to make an 
> > emergency call.
> 
> That wasn't even my original point, I was pointing out 
> travelers, ie you, go somewhere different, ie Europe or 
> Australia and take your GSM mobile phone with you and you 
> don't need to know or care what the local emergency number 
> is, you just have to remember 112.
> 
> -- 
> 
> Best regards,
>  Duane
> 
> --
> This message has been scanned for viruses and dangerous 
> content by MailScanner, and is believed to be clean.
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Duane wrote:
> Paul Kyzivat wrote:
> 
>> But I find it less than compelling that *my* cell phone needs to
>> recognize 112 because somebody else might try to use it to make an
>> emergency call.
> 
> That wasn't even my original point, I was pointing out travelers, ie
> you, go somewhere different, ie Europe or Australia and take your GSM
> mobile phone with you and you don't need to know or care what the local
> emergency number is, you just have to remember 112.

But us provincial Americans don't remember 112 as an emergency number.
[If God had meant 112 to be the emergency number then he wouldn't have 
given us 911.] :-)

I don't think of my self as provincial, but even though I travel out of 
the country a modest amount, I can't remember the last time I used a 
phone that was homed in another country and had a foreign dial plan, so 
I really have little interest in what other emergency numbers are except 
in discussions like this about implementing emergency services.

Emergency numbers are designed to work with a particular dial plan. If 
you import a number designed to work with one dial plan, into an 
unrelated dial plan, then you mostly get a mess.

In reality 112 is perhaps a fortuitous accident in that I don't think it 
conflicts with anything in the North American Numbering Plan.

	Thanks,
	Paul
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DRAGE, Keith (Keith) wrote:
> GSM phones have a multilayered approach to this process, and the
> distinction only works for numbers that are used in conjunction with a
> public telecommunications number.
> 
> -	Firstly 112 and 911 are always recognised by the phone itself.
> 
> -	Secondly a batch of numbers can be configured on the SIM or
> UICC. The principle here is that these are the local service numbers
> that are used for emergency in foreign countries, but are not used for
> other local service number purposes in the home network.
> 
> -	Thirdly if the SIM or UICC is not present, then the following
> numbers also act the same as 112 and 911, that is 000, 08, 110, 999, 118
> and 119.
> 
> -	Fourthly the protocol used between the network and the phone
> also allows the downloading of emergency numbers, which are assumed to
> be the valid emergency numbers for the local network.
> 
> However as soon as you get to private numbering plan numbers, then these
> will frequently conflict, unless you have a means of distinguishing
> numbers that appear in the private number plan from local service
> numbers that appear in the public number plan. This can be done in ISDN,
> and for tel URIs, but cannot be performed for an analogue phone sending
> LD or DTMF.
> 
> For them not to conflict, any potential emergency number has to be
> assigned as a valid number in that particular private numbering plan as
> well as being a local service number in the public numbering plan. This
> is something that owners of hotels, for example, are unlikely to do
> without rebuilding the hotel minus room 112.

They can do that at the same time they leave out room 666 and the 13th 
floor. :-)

> Finally, in response to Paul
> 
>> 	I hope it at least must be "112<send>", rather than it
> initiating the call after you 
>> 	press the "2" in "112" ??? At least then it only takes one
> sequence out of play, rather 
>> 	than all sequences with that as a prefix.
> 
> Treating "112" and "1120" as two different valid numbers is something
> that I have heard of only being used in the US. It has taken ages to
> explain that people sometimes do it to colleagues outside the US.

Funny, because the concept is alien in the US. I thought it was only in 
places like Germany and France where that sort of thing was in common 
use. I have just recently had a discussion of how to implement overlap 
dialing to support this somewhere. We keep hoping it will go away 
because of phones with <send> keys enabling en bloc dialing. But that 
argument doesn't seem to be winning yet.

I only bring it up because if you want to bring in a particular number 
that may otherwise conflict with the local dialing plan, then it has a 
larger potential impact if it preempts all numbers with that prefix than 
it has when it only preempts itself.

With en bloc dialing, 112 would only bother hotels with three digit room 
numbers. Hotels with with 4 digit room numbers wouldn't have to block 112*.

	Thanks,
	Paul

> Regards
> 
> 
> Keith
> 
> 
>> -----Original Message-----
>> From: speermint-bounces@ietf.org 
>> [mailto:speermint-bounces@ietf.org] On Behalf Of Duane
>> Sent: Wednesday, February 06, 2008 3:10 AM
>> To: Paul Kyzivat
>> Cc: speermint@ietf.org
>> Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
>>
>> Paul Kyzivat wrote:
>>
>>> But I find it less than compelling that *my* cell phone needs to 
>>> recognize 112 because somebody else might try to use it to make an 
>>> emergency call.
>> That wasn't even my original point, I was pointing out 
>> travelers, ie you, go somewhere different, ie Europe or 
>> Australia and take your GSM mobile phone with you and you 
>> don't need to know or care what the local emergency number 
>> is, you just have to remember 112.
>>
>> -- 
>>
>> Best regards,
>>  Duane
>>
>> --
>> This message has been scanned for viruses and dangerous 
>> content by MailScanner, and is believed to be clean.
>>
>> _______________________________________________
>> Speermint mailing list
>> Speermint@ietf.org
>> http://www.ietf.org/mailman/listinfo/speermint
>>
> 
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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> 
> Funny, because the concept is alien in the US. I thought it 
> was only in places like Germany and France where that sort of 
> thing was in common use. I have just recently had a 
> discussion of how to implement overlap dialing to support 
> this somewhere. We keep hoping it will go away because of 
> phones with <send> keys enabling en bloc dialing. But that 
> argument doesn't seem to be winning yet.

My only known example of the use of this is in the US, in local service numbers. If I can think back to about 1980 and a very hazy memory, it was used where xxx represented the local service number for operator assistance generally and xxx followed by an area code represented operator assistance specifically for that area. To distinguish required a shorter timeout between the two elements of the number in DTMF and LD phones.

Germany (and Austria) has variable length numbers but 112 and 1120 never coexist. Rather adjacent numbers in the numbering plan may be of different length, e.g.

1230
12310
123111
123112
123113

Regards

Keith

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Wednesday, February 06, 2008 5:32 PM
> To: DRAGE, Keith (Keith)
> Cc: Duane; speermint@ietf.org
> Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
> 
> 
> 
> DRAGE, Keith (Keith) wrote:
> > GSM phones have a multilayered approach to this process, and the 
> > distinction only works for numbers that are used in 
> conjunction with a 
> > public telecommunications number.
> > 
> > -	Firstly 112 and 911 are always recognised by the phone itself.
> > 
> > -	Secondly a batch of numbers can be configured on the SIM or
> > UICC. The principle here is that these are the local 
> service numbers 
> > that are used for emergency in foreign countries, but are 
> not used for 
> > other local service number purposes in the home network.
> > 
> > -	Thirdly if the SIM or UICC is not present, then the following
> > numbers also act the same as 112 and 911, that is 000, 08, 
> 110, 999, 
> > 118 and 119.
> > 
> > -	Fourthly the protocol used between the network and the phone
> > also allows the downloading of emergency numbers, which are 
> assumed to 
> > be the valid emergency numbers for the local network.
> > 
> > However as soon as you get to private numbering plan numbers, then 
> > these will frequently conflict, unless you have a means of 
> > distinguishing numbers that appear in the private number plan from 
> > local service numbers that appear in the public number 
> plan. This can 
> > be done in ISDN, and for tel URIs, but cannot be performed for an 
> > analogue phone sending LD or DTMF.
> > 
> > For them not to conflict, any potential emergency number has to be 
> > assigned as a valid number in that particular private 
> numbering plan 
> > as well as being a local service number in the public 
> numbering plan. 
> > This is something that owners of hotels, for example, are 
> unlikely to 
> > do without rebuilding the hotel minus room 112.
> 
> They can do that at the same time they leave out room 666 and 
> the 13th floor. :-)
> 
> > Finally, in response to Paul
> > 
> >> 	I hope it at least must be "112<send>", rather than it
> > initiating the call after you
> >> 	press the "2" in "112" ??? At least then it only takes one
> > sequence out of play, rather
> >> 	than all sequences with that as a prefix.
> > 
> > Treating "112" and "1120" as two different valid numbers is 
> something 
> > that I have heard of only being used in the US. It has 
> taken ages to 
> > explain that people sometimes do it to colleagues outside the US.
> 
> Funny, because the concept is alien in the US. I thought it 
> was only in places like Germany and France where that sort of 
> thing was in common use. I have just recently had a 
> discussion of how to implement overlap dialing to support 
> this somewhere. We keep hoping it will go away because of 
> phones with <send> keys enabling en bloc dialing. But that 
> argument doesn't seem to be winning yet.
> 
> I only bring it up because if you want to bring in a 
> particular number that may otherwise conflict with the local 
> dialing plan, then it has a larger potential impact if it 
> preempts all numbers with that prefix than it has when it 
> only preempts itself.
> 
> With en bloc dialing, 112 would only bother hotels with three 
> digit room numbers. Hotels with with 4 digit room numbers 
> wouldn't have to block 112*.
> 
> 	Thanks,
> 	Paul
> 
> > Regards
> > 
> > 
> > Keith
> > 
> > 
> >> -----Original Message-----
> >> From: speermint-bounces@ietf.org
> >> [mailto:speermint-bounces@ietf.org] On Behalf Of Duane
> >> Sent: Wednesday, February 06, 2008 3:10 AM
> >> To: Paul Kyzivat
> >> Cc: speermint@ietf.org
> >> Subject: Re: [Speermint] [Ecrit] Emergency Calls & VoIP Peering
> >>
> >> Paul Kyzivat wrote:
> >>
> >>> But I find it less than compelling that *my* cell phone needs to 
> >>> recognize 112 because somebody else might try to use it 
> to make an 
> >>> emergency call.
> >> That wasn't even my original point, I was pointing out 
> travelers, ie 
> >> you, go somewhere different, ie Europe or Australia and 
> take your GSM 
> >> mobile phone with you and you don't need to know or care what the 
> >> local emergency number is, you just have to remember 112.
> >>
> >> --
> >>
> >> Best regards,
> >>  Duane
> >>
> >> --
> >> This message has been scanned for viruses and dangerous content by 
> >> MailScanner, and is believed to be clean.
> >>
> >> _______________________________________________
> >> Speermint mailing list
> >> Speermint@ietf.org
> >> http://www.ietf.org/mailman/listinfo/speermint
> >>
> > 
> 
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A New Internet-Draft is available from the on-line Internet-Drafts 
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This draft is a work item of the Session PEERing for Multimedia INTerconnect Working Group of the IETF.

	Title		: SPEERMINT Terminology
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	Filename	: draft-ietf-speermint-terminology-16.txt
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Since we've had a few WGLCs on this document already, we are calling an
abbreviated WGLC from today, February 13, 2008, to February 21, 2008.

Thank you
Jason 

> -----Original Message-----
> From: speermint-bounces@ietf.org 
> [mailto:speermint-bounces@ietf.org] On Behalf Of 
> Internet-Drafts@ietf.org
> Sent: Tuesday, February 12, 2008 9:00 PM
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> Subject: [Speermint] I-D 
> ACTION:draft-ietf-speermint-terminology-16.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 Terminology
> 	Author(s)	: D. Malas, D. Meyer
> 	Filename	: draft-ietf-speermint-terminology-16.txt
> 	Pages		: 12
> 	Date		: 2008-2-12
> 	
> This document defines the terminology that is to be used in 
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> 
> A URL for this Internet-Draft is:
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We're finalizing the IETF 71 agenda now.  Do we want to add this
discussion topic to the meeting, and if so who wishes to speak and what
document will we review?

Jason
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Subject: I-D Action:draft-rosenberg-sipping-siptrunk-00.txt 

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

	Title           : What is a Session Initiation Protocol (SIP) Trunk
Anyway?
	Author(s)       : J. Rosenberg
	Filename        : draft-rosenberg-sipping-siptrunk-00.txt
	Pages           : 10
	Date            : 2008-02-15

The term "Session Initiation Protocol (SIP) Trunk" has become almost
commonplace amongst vendors and SIP providers.  Even though the notion of a
'trunk' has a well defined meaning in circuit switched systems, it has never
been defined for SIP.  This document provides a formal definition for a SIP
trunk, discusses its scope and applications, and establishes best practices
for identification and security of SIP trunks.

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

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

	Title           : A Provisioning Protocol for ENUM-SIP Addressing
Servers
	Author(s)       : K. Cartwright, et al.
	Filename        : draft-mule-peppermint-espp-protocol-00.txt
	Pages           : 117
	Date            : 2008-02-18

This document defines a provisioning protocol for ENUM-SIP addressing
servers.  This protocol has been recently published as part of
CableLabs(r) PacketCable(tm) specifications.  It allows SIP service
providers to provision and manage session establishment data used by SIP
network elements to route SIP sessions to the target destinations which may
be served by the operator's own internal network or by a session peering
partner.  The data provisioned into an ENUM-SIP addressing server is queried
by SIP entities using ENUM or SIP.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mule-peppermint-espp-protocol-00.t
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-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
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Subject: I-D Action:draft-mule-peppermint-espp-requirements-00.txt

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

	Title           : Provisioning Protocol Requirements for ENUM-SIP
Addressing Servers
	Author(s)       : T. Creighton, J. Mule
	Filename        : draft-mule-peppermint-espp-requirements-00.txt
	Pages           : 16
	Date            : 2008-02-18

This document presents use cases and protocol requirements for provisioning
ENUM-SIP addressing servers.  The provisioned data is used by an addressing
server to return part of the session establishment data to SIP entities so
that they can route SIP sessions to the target destinations.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mule-peppermint-espp-requirements-
00.txt

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the Session PEERing for Multimedia INTerconnect Working Group of the IETF.

	Title		: Presence & Instant Messaging Peering Use Cases
	Author(s)	: A. Houri
	Filename	: draft-ietf-speermint-consolidated-presence-im-usecases-04.txt
	Pages		: 9
	Date		: 2008-2-19
	
The document describes several use cases of peering of non-VoIP
   services between two or more Service Providers.  These Service
   Providers create a peering relationship between themselves thus
   enabling their users to collaborate with users on the other Service
   Provider network.  The target of the document is to drive
   requirements for peering between domains that provide the non-VoIP
   based collaboration services and presence and Instant Messaging (IM)
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I read the last version of the terminology document and it looks ok to
me. 

A few minor comments:
a) Reference sections
The terminology document has an informational status and no 2119 verbs.
Should the normative references be moved to the informative reference?
Also, please check the IDnits and warnings for unused references:
http://tools.ietf.org/tools/idnits/
Checking references for intended status: Informational
 
------------------------------------------------------------------------
----

  == Unused Reference: '9' is defined on line 480, but no explicit
reference
     was found in the text
     '[9]   Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA...'

  == Unused Reference: '10' is defined on line 484, but no explicit
reference
     was found in the text
     '[10]  Dierks, T. and E. Rescorla, "The TLS Protocol Version 1.1",
RF...'

  == Unused Reference: '11' is defined on line 487, but no explicit
reference
     was found in the text
     '[11]  Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
1...'

  == Unused Reference: '12' is defined on line 490, but no explicit
reference
     was found in the text
     '[12]  Postel, J., "DoD Standard Transmission Control Protocol",
RFC...'

  == Unused Reference: '13' is defined on line 493, but no explicit
reference
     was found in the text
     '[13]  Plummer, David C., "An Ethernet Address Resolution
Protocol",...'

  == Unused Reference: '14' is defined on line 496, but no explicit
reference
     was found in the text
     '[14]  Babiarz, J., Chan, K., and F. Baker, "Configuration
Guidelines...'


c) other typos:
The RFC editor will likely catch that but just in case:
In "7. Security Considerations": s/analyzes/analyses ?

Jean-Francois.

> -----Original Message-----
> From: speermint-bounces@ietf.org [mailto:speermint-
> bounces@ietf.org] On Behalf Of Livingood, Jason
> Sent: Wednesday, February 13, 2008 9:39 AM
> To: speermint@ietf.org
> Cc: Daryl Malas
> Subject: Re: [Speermint] I-D ACTION:draft-ietf-speermint-
> terminology-16.txt
> 
> Since we've had a few WGLCs on this document already, we are
> calling an
> abbreviated WGLC from today, February 13, 2008, to February 21,
> 2008.
> 
> Thank you
> Jason
> 
> > -----Original Message-----
> > From: speermint-bounces@ietf.org
> > [mailto:speermint-bounces@ietf.org] On Behalf Of
> > Internet-Drafts@ietf.org
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> > Cc: speermint@ietf.org
> > Subject: [Speermint] I-D
> > ACTION:draft-ietf-speermint-terminology-16.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 Terminology
> > 	Author(s)	: D. Malas, D. Meyer
> > 	Filename	: draft-ietf-speermint-terminology-16.txt
> > 	Pages		: 12
> > 	Date		: 2008-2-12
> >
> > This document defines the terminology that is to be used in
> >    describing Session PEERing for Multimedia INTerconnect
> (SPEERMINT).
> >
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From: "Saverio Niccolini" <Saverio.Niccolini@nw.neclab.eu>
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Hi,

a new version of the VoIP Threats draft is available.
Main changes are:
-- adapted to the terminology draft
-- added the BCPs

Comments are appreciated,
Saverio

> -----Original Message-----
> From: i-d-announce-bounces@ietf.org=20
> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of=20
> Internet-Drafts@ietf.org
> Sent: Friday, February 22, 2008 1:45 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-niccolini-speermint-voipthreats-03.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
> 	Title           : SPEERMINT Security BCPs
> 	Author(s)       : S. Niccolini, et al.
> 	Filename        : draft-niccolini-speermint-voipthreats-03.txt
> 	Pages           : 22
> 	Date            : 2008-02-22
>=20
> This memo presents the different security threats related to=20
> SPEERMINT classifying them into threats to the Location=20
> Function, to the Signaling Function and to the Media=20
> Function.  The different instances of the threats are briefly=20
> introduced inside the classification.  Finally the existing=20
> security solutions in SIP and RTP/RTCP are presented to=20
> describe the countermeasures currently available for such=20
> threats.  The objective of this document is to identify and=20
> enumerate the SPEERMINT-specific threat vectors in order to=20
> specify security-related requirements.  Once the requirements=20
> are identified, methods and solutions how to achieve such=20
> requirements can be selected.
>=20
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From speermint-bounces@ietf.org  Fri Feb 22 11:40:27 2008
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For general consumption in prep for Philly.  See y'all in the city of
brotherly love!  Feedback welcomed.

Changes:
- resync with the latest term draft.
- clarifications based on feedback from the last meeting - like
collapse/removal of assisted peering use cases into direct/indirect
models.

http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-consolidat
ed-usecases-05.txt

I am working on version 6, mostly weeding out nits, and will issue after
the WG session.  Thanks in advance for your feedback!!

Adam Uzelac
Director, Converged Network Arch and Planning
Global Crossing
585-563-4359
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Subject: [Speermint] FW: [CYBERTEL] The Future of IP Interconnection:
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FYI from our friends at the Commission..

********

The following report, produced by WIK-Consult on behalf of the 
European Commission, has just been published on the European 
Commission's web site. I had the pleasure and honor of working with 
some very exceptional colleagues. Those of you who are familiar with 
my previous work on the subject will see that this is a deeper and 
more comprehensive treatment.

===============

The Future of IP Interconnection: Technical, Economic and Public 
Policy Aspects - prepared by WIK-Consult.
http://ec.europa.eu/information_society/policy/ecomm/library/ext_studies/ind
ex_en.htm#2008

Executive Summary only:
http://ec.europa.eu/information_society/policy/ecomm/doc/library/ext_studies
/future_ip_intercon/ip_intercon_study_exec_sum.pdf

Full Study:
http://ec.europa.eu/information_society/policy/ecomm/doc/library/ext_studies
/future_ip_intercon/ip_intercon_study_final.pdf

Cheers,
= Scott

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Saverio,

The document looks quite good.  On behalf of VOIPSA, I'll say again  
thanks for the mention of the Threat Taxonomy. (I know I've said this  
to you personally in the past but also want to do so publicly.)

A couple of minor nits, mostly related to formatting:

- In sections 2.1.x and 2.2.1-2, you list out individual attacks as  
bullet items.  In section 2.2.3, you mention several attacks but do  
not list them as bullet items.  For the sake of consistency you might  
want to consider doing so.

- Sections 2.1.1 and 2.2.1 do not have introductory sentences but  
purely include bullets. I don't know if you care about being  
consistent with the other sections.

- In section 2.2.3, there is a missing "the" in the first sentence:  
"The LF can be THE object of DoS attacks."

- In section 2.3 there are a couple of places in the first sentence  
where you use "signaling function" but I think to be grammatically  
correct it should be "the signaling function".  (As I wonder if I'm  
going to unleash another grammar rathole... )

- In sections 2.3.2.1 and 2.3.2.2 the bullets suddenly disappeared  
(at least on http://www.ietf.org/internet-drafts/draft-niccolini- 
speermint-voipthreats-03.txt )

- Section 2.4.1 again has no bullets while 2.4.2-2.4.3 do

Like I said, really all just formatting issues.

The only real substantive comment I'll make is to ask this - have you  
considered if there is a threat of what I'll call a "self-inflicted  
DoS" within the SPEERMINT architecture?  By that I mean what happens  
when, for instance, there is a power outage and upon the return of  
power there is a large flood as all the SIP devices send INVITEs to  
the registrar.   Obviously that instance doesn't impact the SPEERMINT  
architecture, but are there similar issues?  For instance,  
misconfiguration of one SIP provider's servers sending a flood of  
bogus packets within the peering fabric?  If there are such  
situations it may make sense to at least mention them in this  
document.  I find that people often focus on the external attackers  
and forget to look at how to protect against internal configuration  
problems that can have the same ultimate effect as an external attacker.

Overall a great document,
Dan

On Feb 22, 2008, at 1:25 PM, Saverio Niccolini wrote:

> Hi,
>
> a new version of the VoIP Threats draft is available.
> Main changes are:
> -- adapted to the terminology draft
> -- added the BCPs
>
> Comments are appreciated,
> Saverio
>
>> -----Original Message-----
>> From: i-d-announce-bounces@ietf.org
>> [mailto:i-d-announce-bounces@ietf.org] On Behalf Of
>> Internet-Drafts@ietf.org
>> Sent: Friday, February 22, 2008 1:45 PM
>> To: i-d-announce@ietf.org
>> Subject: I-D Action:draft-niccolini-speermint-voipthreats-03.txt
>>
>> A New Internet-Draft is available from the on-line
>> Internet-Drafts directories.
>>
>> 	Title           : SPEERMINT Security BCPs
>> 	Author(s)       : S. Niccolini, et al.
>> 	Filename        : draft-niccolini-speermint-voipthreats-03.txt
>> 	Pages           : 22
>> 	Date            : 2008-02-22
>>
>> This memo presents the different security threats related to
>> SPEERMINT classifying them into threats to the Location
>> Function, to the Signaling Function and to the Media
>> Function.  The different instances of the threats are briefly
>> introduced inside the classification.  Finally the existing
>> security solutions in SIP and RTP/RTCP are presented to
>> describe the countermeasures currently available for such
>> threats.  The objective of this document is to identify and
>> enumerate the SPEERMINT-specific threat vectors in order to
>> specify security-related requirements.  Once the requirements
>> are identified, methods and solutions how to achieve such
>> requirements can be selected.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-niccolini-speermint-
>> voipthreats-03.txt
>>
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>> 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.
>>
> ============================================================
> Dr. Saverio Niccolini
> Senior Researcher
> NEC Laboratories Europe, Network Research Division	
> Kurfuerstenanlage 36, D-69115 Heidelberg
> Tel.     +49 (0)6221 4342-118
> Fax:     +49 (0)6221 4342-155
> e-mail:  saverio.niccolini@nw.neclab.eu <-- !!! NEW ADDRESS !!!
> ============================================================
> NEC Europe Limited Registered Office: NEC House, 1 Victoria
> Road, London W3 6BL Registered in England 2832014<draft-niccolini- 
> speermint- 
> voipthreats-03.URL>_______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> http://www.ietf.org/mailman/listinfo/speermint

-- 
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork@voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Bring your web applications to the phone.
Find out how at http://evolution.voxeo.com





--Apple-Mail-252-155346919
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=ISO-8859-1

<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">
Saverio,<div><br class=3D"webkit-block-placeholder"></div><div>The =
document looks quite good. =A0On behalf of VOIPSA, I'll say again thanks =
for the mention of the Threat Taxonomy. (I know I've said this to you =
personally in the past but also want to do so publicly.)</div><div><br =
class=3D"webkit-block-placeholder"></div><div>A couple of minor nits, =
mostly related to formatting:</div><div><br =
class=3D"webkit-block-placeholder"></div><div>- In sections 2.1.x and =
2.2.1-2, you list out individual attacks as bullet items. =A0In section =
2.2.3, you mention several attacks but do not list them as bullet items. =
=A0For the sake of consistency you might want to consider doing =
so.</div><div><br></div><div>- Sections 2.1.1 and 2.2.1 do not have =
introductory sentences but purely include bullets. I don't know if you =
care about being consistent with the other sections.</div><div><br =
class=3D"webkit-block-placeholder"></div><div>- In section 2.2.3, there =
is a missing "the" in the first sentence: "The LF can be THE object of =
DoS attacks."</div><div><br =
class=3D"webkit-block-placeholder"></div><div>- In section 2.3 there are =
a couple of places in the first sentence where you use "signaling =
function" but I think to be grammatically correct it should be "the =
signaling function". =A0(As I wonder if I'm going to unleash another =
grammar rathole... )</div><div><br =
class=3D"webkit-block-placeholder"></div><div>- In sections 2.3.2.1 and =
2.3.2.2 the bullets suddenly disappeared (at least on=A0<a =
href=3D"http://www.ietf.org/internet-drafts/draft-niccolini-speermint-voip=
threats-03.txt">http://www.ietf.org/internet-drafts/draft-niccolini-speerm=
int-voipthreats-03.txt</a> )</div><div><br =
class=3D"webkit-block-placeholder"></div><div>- Section 2.4.1 again has =
no bullets while 2.4.2-2.4.3 do</div><div><br =
class=3D"webkit-block-placeholder"></div><div>Like I said, really all =
just formatting issues.</div><div><br =
class=3D"webkit-block-placeholder"></div><div>The only real substantive =
comment I'll make is to ask this - have you considered if there is a =
threat of what I'll call a "self-inflicted DoS" within the SPEERMINT =
architecture? =A0By that I mean what happens when, for instance, there =
is a power outage and upon the return of power there is a large flood as =
all the SIP devices send INVITEs to the registrar. =A0 Obviously that =
instance doesn't impact the SPEERMINT architecture, but are there =
similar issues? =A0For instance, misconfiguration of one SIP provider's =
servers sending a flood of bogus packets within the peering fabric? =A0If =
there are such situations it may make sense to at least mention them in =
this document. =A0I find that people often focus on the external =
attackers and forget to look at how to protect against internal =
configuration problems that can have the same ultimate effect as an =
external attacker.</div><div><br =
class=3D"webkit-block-placeholder"></div><div>Overall a great =
document,</div><div>Dan</div><div><br =
class=3D"webkit-block-placeholder"></div><div><div><div>On Feb 22, 2008, =
at 1:25 PM, Saverio Niccolini wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Hi,</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">a new version of the VoIP =
Threats draft is available.</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Main changes =
are:</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">-- adapted to the terminology =
draft</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">-- added the BCPs</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Comments =
are appreciated,</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Saverio</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div> <blockquote =
type=3D"cite"><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">-----Original =
Message-----</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">From: <a =
href=3D"mailto:i-d-announce-bounces@ietf.org">i-d-announce-bounces@ietf.or=
g</a><span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">[<a =
href=3D"mailto:i-d-announce-bounces@ietf.org">mailto:i-d-announce-bounces@=
ietf.org</a>] On Behalf Of<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><a =
href=3D"mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a></div=
><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Sent: Friday, February 22, 2008 1:45 PM</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">To: <a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Subject: I-D =
Action:draft-niccolini-speermint-voipthreats-03.txt<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><br></div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">A New =
Internet-Draft is available from the on-line<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">Internet-Drafts directories.</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Title <span =
class=3D"Apple-converted-space">=A0 =A0 =A0 =A0 =A0 </span>: SPEERMINT =
Security BCPs</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Author(s) <span =
class=3D"Apple-converted-space">=A0 =A0 =A0 </span>: S. Niccolini, et =
al.</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Filename<span =
class=3D"Apple-converted-space">=A0 =A0 =A0 =A0 </span>: =
draft-niccolini-speermint-voipthreats-03.txt</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Pages <span =
class=3D"Apple-converted-space">=A0 =A0 =A0 =A0 =A0 </span>: =
22</div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Date<span =
class=3D"Apple-converted-space">=A0 =A0 =A0 =A0 =A0 =A0 </span>: =
2008-02-22</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><br></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">This memo presents the different security threats =
related to<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">SPEERMINT classifying them into threats to the =
Location<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Function, to the Signaling Function and to the =
Media<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Function.<span class=3D"Apple-converted-space">=A0 =
</span>The different instances of the threats are briefly<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">introduced inside the classification.<span =
class=3D"Apple-converted-space">=A0 </span>Finally the existing<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">security =
solutions in SIP and RTP/RTCP are presented to<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">describe =
the countermeasures currently available for such<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">threats.<span class=3D"Apple-converted-space">=A0 </span>The objective =
of this document is to identify and<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">enumerate the SPEERMINT-specific threat vectors in order to<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">specify =
security-related requirements.<span class=3D"Apple-converted-space">=A0 =
</span>Once the requirements<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">are =
identified, methods and solutions how to achieve such<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">requirements can be selected.</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">A URL for this Internet-Draft =
is:</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><a =
href=3D"http://www.ietf.org/internet-drafts/draft-niccolini-speermint-">ht=
tp://www.ietf.org/internet-drafts/draft-niccolini-speermint-</a></div><div=
 style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">voipthreats-03.txt</div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
min-height: 14px; "><br></div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">To remove =
yourself from the I-D Announcement list, send a<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">message =
to <a =
href=3D"mailto:i-d-announce-request@ietf.org">i-d-announce-request@ietf.or=
g</a> with the word<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; =
">unsubscribe in the body of the message.</div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">You can =
also visit <a =
href=3D"https://www1.ietf.org/mailman/listinfo/I-D-announce">https://www1.=
ietf.org/mailman/listinfo/I-D-announce</a></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">to =
change your subscription settings.</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; min-height: =
14px; "><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Internet-Drafts are also =
available by anonymous FTP. Login<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">with the =
username "anonymous" and a password of your e-mail<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">address. =
After logging in, type "cd internet-drafts" and then</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>"get =
draft-niccolini-speermint-voipthreats-03.txt".</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">A list =
of Internet-Drafts directories can be found in<span =
class=3D"Apple-converted-space">=A0</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><a =
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</=
a> or<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><a =
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf=
/1shadow-sites.txt</a></div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; =
"><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Internet-Drafts can also be =
obtained by e-mail.</div><div style=3D"margin-top: 0px; margin-right: =
0px; margin-bottom: 0px; margin-left: 0px; min-height: 14px; =
"><br></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Send a message to:</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span><a =
href=3D"mailto:mailserv@ietf.org">mailserv@ietf.org</a>.</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">In the body type:</div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>"FILE<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; =
">/internet-drafts/draft-niccolini-speermint-voipthreats-03.txt".</div><di=
v style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">NOTE: =
<span class=3D"Apple-converted-space">=A0 </span>The mail server at =
ietf.org can return the document in</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>MIME-encoded form by using the "mpack" utility.<span =
class=3D"Apple-converted-space">=A0 </span>To use this</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>feature, insert the command =
"ENCODING mime" before the "FILE"</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>command.<span class=3D"Apple-converted-space">=A0 </span>To =
decode the response(s), you will need "munpack" or</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>a MIME-compliant mail =
reader.<span class=3D"Apple-converted-space">=A0 </span>Different =
MIME-compliant<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">mail readers</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>exhibit =
different behavior, especially when dealing with</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>"multipart" MIME messages (i.e. =
documents which have been split</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>up into =
multiple messages), so check your local documentation on</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>how to manipulate these =
messages.</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; min-height: 14px; "><br></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Below is the data which will enable a MIME compliant =
mail<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">reader implementation to automatically retrieve the =
ASCII<span class=3D"Apple-converted-space">=A0</span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">version of the Internet-Draft.</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div> </blockquote><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; =
">=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</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Dr. Saverio =
Niccolini</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">Senior Researcher</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session 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-04.txt
	Pages           : 28
	Date            : 2008-02-25

A number of use cases have been described for session peering of
voice, presence, instant messaging and other types of multimedia
traffic.  This memo captures some of the requirements identified by
these use case scenarios.  It is intended to become an informational
document linking the use cases to potential protocol solutions.

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A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the Session PEERing for Multimedia INTerconnect Working Group of the IETF.

	Title		: SPEERMINT Peering Architecture
	Author(s)	: R. Penno
	Filename	: draft-ietf-speermint-architecture-05.txt
	Pages		: 22
	Date		: 2008-2-25
	
This document defines the SPEERMINT peering architecture, its  
  functional components and peering interface functions. It also 
  describes the steps taken to establish a session between two peering   
  domains in the context of the functions defined.

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From speermint-bounces@ietf.org  Mon Feb 25 16:48:56 2008
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
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Date: Mon, 25 Feb 2008 18:24:02 -0500
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Awesome draft. (I missed it the first time 'round)

I agree with your statement of "The obvious solution is to add an on-demand mapping step ahead of the routing protocol."  The trick will be how that works/scales, and how it interacts with a routing protocol.  The easy case is the non-transit bilateral peering - the very hard case is transit.

One very minor nit: I don't think TRIP failed due to being prefix-based and thus needing aggregation.  I think it failed because the market wasn't ready for it (the problem didn't need to be fixed for most people yet), the idea of implementing/using BGP scared off voip vendors, and it didn't cover some of the routing selection models people need.

-hadriel


> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Monday, February 25, 2008 4:15 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-lendl-speermint-background-01.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>         Title           : VoIP Peering: Background and Assumptions
>         Author(s)       : O. Lendl
>         Filename        : draft-lendl-speermint-background-01.txt
>         Pages           : 15
>         Date            : 2008-02-25
>
> This documents provides background for the work on VoIP peering and
> tries to provide guidance on what kind of work is needed to
> facilitate widespread SIP-based peering of telephony networks.  It is
> intended to spur discussion on the work about peering and should also
> serve as input to the ongoing discussions on reducing Spam for
> Internet Telephony (SPIT).
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-lendl-speermint-background-
> 01.txt
>
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From: "Daryl Malas" <D.Malas@cablelabs.com>
To: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>, <speermint@ietf.org>
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"VoIP consolidated use cases" author's,

First, let me say, I think the draft is off to a good start.  I would
like to suggest some things, I believe will allow the draft to provide
better clarity and more depth for a reader to understand the reasons and
situations relating to VoIP peering.

Here are a few encompassing comments and general nits of the document:

- I recommend looking through RFC 3285 before the next revision of the
draft.  There are a number of Word template issues, which can be
resolved fairly easy.

- The references need to be updated.

- Finally, as I review the use cases, I am looking for the following
basic structure within each use case: Goal, Summary, Actors,
Preconditions, Triggers, Basic course of events, Alternative paths (if
applicable), and any Policy considerations.

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

More detailed comments:

Section 4 - "Each grouping can further sub-divided to Direct Peering and
Indirect Peering [15]."  Should it be, "Each grouping can be further
sub-divided to Direct and Indirect Peering [15]"?

"Each use-case has to specify how a few core operations which are to be
performed by its members."  I would like to suggest re-wording to, "Each
use case must specify a basic set of required operations to be performed
by each member when peering."

Section 5.1 - I would expect this intro paragraph to describe the goal
of static peering.  For example, "Static peering  is used in cases, when
two peers want a consistent and "hardened" approach to peering.  In this
scenario, all of the variables are very well defined and known by each
peer....."

Section 5.1.1 - Who are the actors (e.g. Enterprise A wants to peer with
Enterprise B, SSP A wants to peer with SSP B, SSP A is providing static
peering for Enterprise A, B, and C, etc.)?  What are the preconditions
required for this use case to be successful (e.g. each peer must
exchange their allowable set of codecs, authentication method, no
RFC1918 addresses, etc.)?  What are the triggers (i.e. what causes SSPA
to think it can send it to a peer?)

The sentence beginning, "Two SSP negotiate and agree..." seems to begin
the course of events, but you have other events numbered below.  This is
confusing.  I would suggest moving these into the numbered events below.

Each of the events listed, such provide an example of what is occurring
with each.  The current ones are too high level of value in deriving
requirements.  For example:

1. UAC initiates a call via SIP INVITE to O-Proxy

(Snippets frm RFC 3665)	
   INVITE sip:+13031112222@denver.example.com SIP/2.0
   Via: SIP/2.0/TCP client.atlanta.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
   To: +13031112222 <sip:+13031112222@denver.example.com>
   Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
   
2. O-Proxy queries for SED information from a routing database (ENUM?? -
LUF)

(The result of the ENUM query is...)

$ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
   NAPTR 10 100 "u" "E2U+SIP" "!^.*$
             !sip:\\1@sbe1-denver.sspa.example.com!".

3. Routing database entity replies with SED to called party (NAPTR -
LRF)

sbe1-denver.sspa.example.com.
      order pref flgs service  regx    replacement
   IN NAPTR 100  50   "s"   "SIP+D2U" ""
_sip._udp.sbe1-denver.sspa.example.com

4. O-Proxy sends a DNS SRV query for
_sip._udp.sbe1-denver.sspa.example.com which results in:

   $ORIGIN _sip._udp.sbe1-denver.sspa.example.com.
    (Some content not included for clarity purposes.)
   IN SRV  0  1   5060   sbe1-denver.sspa.example.com

   ;; ANSWER SECTION
   sbe1-denver.sspa.example.com   A   192.0.2.17
   sbe2-denver.sspa.example.com   A   192.0.2.18

5.  Call sent to terminating domain proxy.

INVITE sip:+13031112222@sbe1-denver.sspa.example.com SIP/2.0
   Via: SIP/2.0/TCP client.atlanta.example.com:5060;branch=z9hG4bK74bf9
   Max-Forwards: 70
   From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
   To: +13031112222 <sip:+13031112222@denver.example.com>
   Contact: <sip:alice@client.atlanta.example.com;transport=tcp>

6.  T-Proxy determines called party location by querying the ENUM server

$ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
   NAPTR 10 100 "u" "E2U+SIP" "!^.*$
             !sip:\\bob@proxy1-denver.sspa.example.com!".

....and so on.  I think this detail will provide clarity for the
requirements and for readers to understand the necessary steps to
successfully peer.  I think this should be done for each of the use
cases.

Section 5.1.1 - These appear to be optional (or additional potential)
pre-conditions, which could occur relative to this use case.  If this is
case, I would suggest calling them Optional Pre-conditions and not
Options and Nuances.

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

I think making similar, relative changes to each of the use case
sections will get the very close to a finished product.

--Daryl

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Uzelac, Adam
Sent: Friday, February 22, 2008 12:37 PM
To: speermint@ietf.org
Subject: [Speermint]
draft-ietf-speermint-voip-consolidated-usecases-05.txt

For general consumption in prep for Philly.  See y'all in the city of
brotherly love!  Feedback welcomed.

Changes:
- resync with the latest term draft.
- clarifications based on feedback from the last meeting - like
collapse/removal of assisted peering use cases into direct/indirect
models.

http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-consolidat
ed-usecases-05.txt

I am working on version 6, mostly weeding out nits, and will issue after
the WG session.  Thanks in advance for your feedback!!

Adam Uzelac
Director, Converged Network Arch and Planning Global Crossing
585-563-4359
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From speermint-bounces@ietf.org  Mon Feb 25 17:46:52 2008
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From: "Richard Shockey" <richard@shockey.us>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>, <otmar.lendl@enum.at>
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>  Awesome draft. (I missed it the first time 'round)
>  
>  I agree with your statement of "The obvious solution is to add an on-
>  demand mapping step ahead of the routing protocol."  The trick will be
>  how that works/scales, and how it interacts with a routing protocol.
>  The easy case is the non-transit bilateral peering - the very hard
>  case is transit.
>  
>  One very minor nit: I don't think TRIP failed due to being prefix-
>  based and thus needing aggregation.  I think it failed because the
>  market wasn't ready for it (the problem didn't need to be fixed for
>  most people yet), the idea of implementing/using BGP scared off voip
>  vendors, and it didn't cover some of the routing selection models
>  people need.

IMHO TRIP failed because it was using a 120 mm howitzer to deal with a
problem better dealt with a BB gun. There were enough issues with BGP as it
was.

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Well... working for one of only two vendors who implemented it, afaict, it failed for multiple reasons.  None of them had to do with prefix aggregation or scalability, at least at the time.  BGP was definitely a poor choice, though more for marketing/perception reasons than technical, albeit there were some major technical issues with it too.  It was never tied to BGP for IP routing - it operated on a separate plane really, and even had a different protocol model for internal route distribution, but yes it inherited too much of BGP in terms of deployment model and some unnecessary baggage.  The big killer though, imho, was there was no real problem at the time that it solved.  It's moot though - I'm definitely not suggesting a TRIP down memory lane.
:)

-hadriel


> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Monday, February 25, 2008 8:26 PM
> To: Hadriel Kaplan; otmar.lendl@enum.at
> Cc: speermint@ietf.org; PEPPERMINT@ietf.org
> Subject: RE: [Speermint] I-D Action:draft-lendl-speermint-background-
> 01.txt
>
> >  Awesome draft. (I missed it the first time 'round)
> >
> >  I agree with your statement of "The obvious solution is to add an on-
> >  demand mapping step ahead of the routing protocol."  The trick will be
> >  how that works/scales, and how it interacts with a routing protocol.
> >  The easy case is the non-transit bilateral peering - the very hard
> >  case is transit.
> >
> >  One very minor nit: I don't think TRIP failed due to being prefix-
> >  based and thus needing aggregation.  I think it failed because the
> >  market wasn't ready for it (the problem didn't need to be fixed for
> >  most people yet), the idea of implementing/using BGP scared off voip
> >  vendors, and it didn't cover some of the routing selection models
> >  people need.
>
> IMHO TRIP failed because it was using a 120 mm howitzer to deal with a
> problem better dealt with a BB gun. There were enough issues with BGP as
> it
> was.

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From: "Saverio Niccolini" <Saverio.Niccolini@nw.neclab.eu>
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Dan,

thanks for the review, I will take care of your editorial comments
in the next revision of the draft.
  
> The only real substantive comment I'll make is to ask this - 
> have you considered if there is a threat of what I'll call a 
> "self-inflicted DoS" within the SPEERMINT architecture?  By 
> that I mean what happens when, for instance, there is a power 
> outage and upon the return of power there is a large flood as 
> all the SIP devices send INVITEs to the registrar.   
> Obviously that instance doesn't impact the SPEERMINT 
> architecture, but are there similar issues?  For instance, 
> misconfiguration of one SIP provider's servers sending a 
> flood of bogus packets within the peering fabric?  If there 
> are such situations it may make sense to at least mention 
> them in this document.  I find that people often focus on the 
> external attackers and forget to look at how to protect 
> against internal configuration problems that can have the 
> same ultimate effect as an external attacker.

Right, we explicitly did not look at these and if the group
has consensus on this we can add them in the next version of the
draft. Anyway we need to be sure that the threats we are addressing
are specific to SPEERMINT, we excluded domain-internal threats
from the classification (not interesting to SPEERMINT the REGISTER
after loss of power example you cited).
Can you suggest (maybe offline) more specifically which kind of
threats that are SPEERMINT specific did you have in mind?

> Overall a great document,

Thanks, looking forward to discussing it at the next IETF during
the SPEERMINt session (we should have time allocated for this on
the agenda).

Cheers,
Saverio

============================================================
Dr. Saverio Niccolini
Senior Researcher
NEC Laboratories Europe, Network Research Division	
Kurfuerstenanlage 36, D-69115 Heidelberg
Tel.     +49 (0)6221 4342-118
Fax:     +49 (0)6221 4342-155
e-mail:  saverio.niccolini@nw.neclab.eu <-- !!! NEW ADDRESS !!!
============================================================
NEC Europe Limited Registered Office: NEC House, 1 Victoria
Road, London W3 6BL Registered in England 2832014
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From: "Uzelac, Adam" <Adam.Uzelac@globalcrossing.com>
To: "Daryl Malas" <D.Malas@cablelabs.com>, <speermint@ietf.org>
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Daryl,

Excellent comments and thanks for the feedback.  We have version 6
queued up and are in the process of updating.  Regarding some of your
comments, starting with the "(Snippets frm RFC 3665)" - you state: 

> ...I think this detail will provide clarity for 
> the requirements and for readers to understand the necessary 
> steps to successfully peer.  I think this should be done for 
> each of the use cases.

I read that you are requesting more detail in the use cases, but I feel
it isn't necessary to get the pertinent points across.  Initially, and I
believe to a certain extent still, one goal of capturing current use
cases (what is "occurring in the wild today), is to capture
requirements.  At the same time, I think that a goal in capturing the
VoIP use cases at present is to identify the "ugliness", like
limited-to-no transparency of SIP/VoIP Peering as it exists today.
Also, with the advent of B2BUAs as the SBE/DBEs in the call flows, there
would be numerous iterations of SIP Header manipulation/mangling that is
done for various business, regulatory and other reasons.  It could be a
very slippery slope to start digging into the details, and would also
balloon the draft greatly.

To summarize, I am on the fence with this one.  Others have thoughts?
Will the details help clarify, or narrow the discussion too much?

-Adam

> -----Original Message-----
> From: speermint-bounces@ietf.org 
> [mailto:speermint-bounces@ietf.org] On Behalf Of Daryl Malas
> Sent: Monday, February 25, 2008 11:38 PM
> To: Uzelac, Adam; speermint@ietf.org
> Subject: Re: 
> [Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> "VoIP consolidated use cases" author's,
> 
> First, let me say, I think the draft is off to a good start.  
> I would like to suggest some things, I believe will allow the 
> draft to provide better clarity and more depth for a reader 
> to understand the reasons and situations relating to VoIP peering.
> 
> Here are a few encompassing comments and general nits of the document:
> 
> - I recommend looking through RFC 3285 before the next 
> revision of the draft.  There are a number of Word template 
> issues, which can be resolved fairly easy.
> 
> - The references need to be updated.
> 
> - Finally, as I review the use cases, I am looking for the 
> following basic structure within each use case: Goal, 
> Summary, Actors, Preconditions, Triggers, Basic course of 
> events, Alternative paths (if applicable), and any Policy 
> considerations.
> 
> --------------------------------
> 
> More detailed comments:
> 
> Section 4 - "Each grouping can further sub-divided to Direct 
> Peering and Indirect Peering [15]."  Should it be, "Each 
> grouping can be further sub-divided to Direct and Indirect 
> Peering [15]"?
> 
> "Each use-case has to specify how a few core operations which 
> are to be performed by its members."  I would like to suggest 
> re-wording to, "Each use case must specify a basic set of 
> required operations to be performed by each member when peering."
> 
> Section 5.1 - I would expect this intro paragraph to describe 
> the goal of static peering.  For example, "Static peering  is 
> used in cases, when two peers want a consistent and 
> "hardened" approach to peering.  In this scenario, all of the 
> variables are very well defined and known by each peer....."
> 
> Section 5.1.1 - Who are the actors (e.g. Enterprise A wants 
> to peer with Enterprise B, SSP A wants to peer with SSP B, 
> SSP A is providing static peering for Enterprise A, B, and C, 
> etc.)?  What are the preconditions required for this use case 
> to be successful (e.g. each peer must exchange their 
> allowable set of codecs, authentication method, no
> RFC1918 addresses, etc.)?  What are the triggers (i.e. what 
> causes SSPA to think it can send it to a peer?)
> 
> The sentence beginning, "Two SSP negotiate and agree..." 
> seems to begin the course of events, but you have other 
> events numbered below.  This is confusing.  I would suggest 
> moving these into the numbered events below.
> 
> Each of the events listed, such provide an example of what is 
> occurring with each.  The current ones are too high level of 
> value in deriving requirements.  For example:
> 
> 1. UAC initiates a call via SIP INVITE to O-Proxy
> 
> (Snippets frm RFC 3665)	
>    INVITE sip:+13031112222@denver.example.com SIP/2.0
>    Via: SIP/2.0/TCP 
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
>    
> 2. O-Proxy queries for SED information from a routing 
> database (ENUM?? -
> LUF)
> 
> (The result of the ENUM query is...)
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\1@sbe1-denver.sspa.example.com!".
> 
> 3. Routing database entity replies with SED to called party (NAPTR -
> LRF)
> 
> sbe1-denver.sspa.example.com.
>       order pref flgs service  regx    replacement
>    IN NAPTR 100  50   "s"   "SIP+D2U" ""
> _sip._udp.sbe1-denver.sspa.example.com
> 
> 4. O-Proxy sends a DNS SRV query for
> _sip._udp.sbe1-denver.sspa.example.com which results in:
> 
>    $ORIGIN _sip._udp.sbe1-denver.sspa.example.com.
>     (Some content not included for clarity purposes.)
>    IN SRV  0  1   5060   sbe1-denver.sspa.example.com
> 
>    ;; ANSWER SECTION
>    sbe1-denver.sspa.example.com   A   192.0.2.17
>    sbe2-denver.sspa.example.com   A   192.0.2.18
> 
> 5.  Call sent to terminating domain proxy.
> 
> INVITE sip:+13031112222@sbe1-denver.sspa.example.com SIP/2.0
>    Via: SIP/2.0/TCP 
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
> 
> 6.  T-Proxy determines called party location by querying the 
> ENUM server
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\bob@proxy1-denver.sspa.example.com!".
> 
> ....and so on.  I think this detail will provide clarity for 
> the requirements and for readers to understand the necessary 
> steps to successfully peer.  I think this should be done for 
> each of the use cases.
> 
> Section 5.1.1 - These appear to be optional (or additional 
> potential) pre-conditions, which could occur relative to this 
> use case.  If this is case, I would suggest calling them 
> Optional Pre-conditions and not Options and Nuances.
> 
> ---------------------------------
> 
> I think making similar, relative changes to each of the use 
> case sections will get the very close to a finished product.
> 
> --Daryl
> 
> -----Original Message-----
> From: speermint-bounces@ietf.org 
> [mailto:speermint-bounces@ietf.org] On Behalf Of Uzelac, Adam
> Sent: Friday, February 22, 2008 12:37 PM
> To: speermint@ietf.org
> Subject: [Speermint]
> draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> For general consumption in prep for Philly.  See y'all in the 
> city of brotherly love!  Feedback welcomed.
> 
> Changes:
> - resync with the latest term draft.
> - clarifications based on feedback from the last meeting - 
> like collapse/removal of assisted peering use cases into 
> direct/indirect models.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-
> consolidat
> ed-usecases-05.txt
> 
> I am working on version 6, mostly weeding out nits, and will 
> issue after the WG session.  Thanks in advance for your feedback!!
> 
> Adam Uzelac
> Director, Converged Network Arch and Planning Global Crossing
> 585-563-4359
> _______________________________________________
> Speermint mailing list
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> 
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Adam,

I am not suggesting a detailed signaling example of every possible
scenario, but I am suggesting a relative example of the scenario
described within the use case.  For example, static peering can use ENUM
or a redirect server.  I am suggesting a detailed view of the ENUM
process within the peering use case, to help the reader understand the
complexities of manipulating the signaling from the UAC to the UAS.  As
you probably already know, this draft is not intended to define the
solution, but rather describe the things that need to happen.  How it
should or must happen is up to later drafts.  This is not as simple as
it seems, and requires more than just a "oh by the way...".  Then, after
the use case, you suggest other possible conditions.  You can say, "The
described course of events is an example, and here are some other
possibilities, and here are a few things, which must be considered with
these other possibilities."

Also, you describe the necessity to identify the "ugliness".  This is my
point, wrt providing detail.  I see a fluffy high level overview in the
draft, not much ugliness.

By the way, I went through a similar situation recently...it's an easy
problem to talk about, until you start diving down into a step-by-step
analysis of what needs to happen and what needs to change for it to be
successful.  I think this is some of what Otmar has been trying to say
for awhile.  The final solution may require other protocol work, but I
do not get a sense of this possibility yet based on the use cases.

Thanks...

--Daryl 

-----Original Message-----
From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com] 
Sent: Tuesday, February 26, 2008 8:02 AM
To: Daryl Malas; speermint@ietf.org
Subject: RE:
[Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt

Daryl,

Excellent comments and thanks for the feedback.  We have version 6
queued up and are in the process of updating.  Regarding some of your
comments, starting with the "(Snippets frm RFC 3665)" - you state: 

> ...I think this detail will provide clarity for the requirements and 
> for readers to understand the necessary steps to successfully peer.  I

> think this should be done for each of the use cases.

I read that you are requesting more detail in the use cases, but I feel
it isn't necessary to get the pertinent points across.  Initially, and I
believe to a certain extent still, one goal of capturing current use
cases (what is "occurring in the wild today), is to capture
requirements.  At the same time, I think that a goal in capturing the
VoIP use cases at present is to identify the "ugliness", like
limited-to-no transparency of SIP/VoIP Peering as it exists today.
Also, with the advent of B2BUAs as the SBE/DBEs in the call flows, there
would be numerous iterations of SIP Header manipulation/mangling that is
done for various business, regulatory and other reasons.  It could be a
very slippery slope to start digging into the details, and would also
balloon the draft greatly.

To summarize, I am on the fence with this one.  Others have thoughts?
Will the details help clarify, or narrow the discussion too much?

-Adam

> -----Original Message-----
> From: speermint-bounces@ietf.org
> [mailto:speermint-bounces@ietf.org] On Behalf Of Daryl Malas
> Sent: Monday, February 25, 2008 11:38 PM
> To: Uzelac, Adam; speermint@ietf.org
> Subject: Re: 
> [Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> "VoIP consolidated use cases" author's,
> 
> First, let me say, I think the draft is off to a good start.  
> I would like to suggest some things, I believe will allow the draft to

> provide better clarity and more depth for a reader to understand the 
> reasons and situations relating to VoIP peering.
> 
> Here are a few encompassing comments and general nits of the document:
> 
> - I recommend looking through RFC 3285 before the next revision of the

> draft.  There are a number of Word template issues, which can be 
> resolved fairly easy.
> 
> - The references need to be updated.
> 
> - Finally, as I review the use cases, I am looking for the following 
> basic structure within each use case: Goal, Summary, Actors, 
> Preconditions, Triggers, Basic course of events, Alternative paths (if

> applicable), and any Policy considerations.
> 
> --------------------------------
> 
> More detailed comments:
> 
> Section 4 - "Each grouping can further sub-divided to Direct Peering 
> and Indirect Peering [15]."  Should it be, "Each grouping can be 
> further sub-divided to Direct and Indirect Peering [15]"?
> 
> "Each use-case has to specify how a few core operations which are to 
> be performed by its members."  I would like to suggest re-wording to, 
> "Each use case must specify a basic set of required operations to be 
> performed by each member when peering."
> 
> Section 5.1 - I would expect this intro paragraph to describe the goal

> of static peering.  For example, "Static peering  is used in cases, 
> when two peers want a consistent and "hardened" approach to peering.  
> In this scenario, all of the variables are very well defined and known

> by each peer....."
> 
> Section 5.1.1 - Who are the actors (e.g. Enterprise A wants to peer 
> with Enterprise B, SSP A wants to peer with SSP B, SSP A is providing 
> static peering for Enterprise A, B, and C, etc.)?  What are the 
> preconditions required for this use case to be successful (e.g. each 
> peer must exchange their allowable set of codecs, authentication 
> method, no
> RFC1918 addresses, etc.)?  What are the triggers (i.e. what causes 
> SSPA to think it can send it to a peer?)
> 
> The sentence beginning, "Two SSP negotiate and agree..." 
> seems to begin the course of events, but you have other events 
> numbered below.  This is confusing.  I would suggest moving these into

> the numbered events below.
> 
> Each of the events listed, such provide an example of what is 
> occurring with each.  The current ones are too high level of value in 
> deriving requirements.  For example:
> 
> 1. UAC initiates a call via SIP INVITE to O-Proxy
> 
> (Snippets frm RFC 3665)	
>    INVITE sip:+13031112222@denver.example.com SIP/2.0
>    Via: SIP/2.0/TCP
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
>    
> 2. O-Proxy queries for SED information from a routing database (ENUM??

> -
> LUF)
> 
> (The result of the ENUM query is...)
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\1@sbe1-denver.sspa.example.com!".
> 
> 3. Routing database entity replies with SED to called party (NAPTR -
> LRF)
> 
> sbe1-denver.sspa.example.com.
>       order pref flgs service  regx    replacement
>    IN NAPTR 100  50   "s"   "SIP+D2U" ""
> _sip._udp.sbe1-denver.sspa.example.com
> 
> 4. O-Proxy sends a DNS SRV query for
> _sip._udp.sbe1-denver.sspa.example.com which results in:
> 
>    $ORIGIN _sip._udp.sbe1-denver.sspa.example.com.
>     (Some content not included for clarity purposes.)
>    IN SRV  0  1   5060   sbe1-denver.sspa.example.com
> 
>    ;; ANSWER SECTION
>    sbe1-denver.sspa.example.com   A   192.0.2.17
>    sbe2-denver.sspa.example.com   A   192.0.2.18
> 
> 5.  Call sent to terminating domain proxy.
> 
> INVITE sip:+13031112222@sbe1-denver.sspa.example.com SIP/2.0
>    Via: SIP/2.0/TCP
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
> 
> 6.  T-Proxy determines called party location by querying the ENUM 
> server
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\bob@proxy1-denver.sspa.example.com!".
> 
> ....and so on.  I think this detail will provide clarity for the 
> requirements and for readers to understand the necessary steps to 
> successfully peer.  I think this should be done for each of the use 
> cases.
> 
> Section 5.1.1 - These appear to be optional (or additional
> potential) pre-conditions, which could occur relative to this use 
> case.  If this is case, I would suggest calling them Optional 
> Pre-conditions and not Options and Nuances.
> 
> ---------------------------------
> 
> I think making similar, relative changes to each of the use case 
> sections will get the very close to a finished product.
> 
> --Daryl
> 
> -----Original Message-----
> From: speermint-bounces@ietf.org
> [mailto:speermint-bounces@ietf.org] On Behalf Of Uzelac, Adam
> Sent: Friday, February 22, 2008 12:37 PM
> To: speermint@ietf.org
> Subject: [Speermint]
> draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> For general consumption in prep for Philly.  See y'all in the city of 
> brotherly love!  Feedback welcomed.
> 
> Changes:
> - resync with the latest term draft.
> - clarifications based on feedback from the last meeting - like 
> collapse/removal of assisted peering use cases into direct/indirect 
> models.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-
> consolidat
> ed-usecases-05.txt
> 
> I am working on version 6, mostly weeding out nits, and will issue 
> after the WG session.  Thanks in advance for your feedback!!
> 
> Adam Uzelac
> Director, Converged Network Arch and Planning Global Crossing
> 585-563-4359
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> http://www.ietf.org/mailman/listinfo/speermint
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> http://www.ietf.org/mailman/listinfo/speermint
> 
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One more thing...the detailed view should only describe known processes.
Other steps, should include what needs to happen regardless of an
existing process.  For example, SSPA must somehow get their subscriber
information into the LUF (as we know...to be potentially handled by
PEPPERMINT).  This is work, which can be identified by the use cases and
then a solution defined SPEERMINT or in other groups as necessary.

--Daryl

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Tuesday, February 26, 2008 8:37 AM
To: Uzelac, Adam; speermint@ietf.org
Subject: Re:
[Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt

Adam,

I am not suggesting a detailed signaling example of every possible
scenario, but I am suggesting a relative example of the scenario
described within the use case.  For example, static peering can use ENUM
or a redirect server.  I am suggesting a detailed view of the ENUM
process within the peering use case, to help the reader understand the
complexities of manipulating the signaling from the UAC to the UAS.  As
you probably already know, this draft is not intended to define the
solution, but rather describe the things that need to happen.  How it
should or must happen is up to later drafts.  This is not as simple as
it seems, and requires more than just a "oh by the way...".  Then, after
the use case, you suggest other possible conditions.  You can say, "The
described course of events is an example, and here are some other
possibilities, and here are a few things, which must be considered with
these other possibilities."

Also, you describe the necessity to identify the "ugliness".  This is my
point, wrt providing detail.  I see a fluffy high level overview in the
draft, not much ugliness.

By the way, I went through a similar situation recently...it's an easy
problem to talk about, until you start diving down into a step-by-step
analysis of what needs to happen and what needs to change for it to be
successful.  I think this is some of what Otmar has been trying to say
for awhile.  The final solution may require other protocol work, but I
do not get a sense of this possibility yet based on the use cases.

Thanks...

--Daryl 

-----Original Message-----
From: Uzelac, Adam [mailto:Adam.Uzelac@globalcrossing.com]
Sent: Tuesday, February 26, 2008 8:02 AM
To: Daryl Malas; speermint@ietf.org
Subject: RE:
[Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt

Daryl,

Excellent comments and thanks for the feedback.  We have version 6
queued up and are in the process of updating.  Regarding some of your
comments, starting with the "(Snippets frm RFC 3665)" - you state: 

> ...I think this detail will provide clarity for the requirements and 
> for readers to understand the necessary steps to successfully peer.  I

> think this should be done for each of the use cases.

I read that you are requesting more detail in the use cases, but I feel
it isn't necessary to get the pertinent points across.  Initially, and I
believe to a certain extent still, one goal of capturing current use
cases (what is "occurring in the wild today), is to capture
requirements.  At the same time, I think that a goal in capturing the
VoIP use cases at present is to identify the "ugliness", like
limited-to-no transparency of SIP/VoIP Peering as it exists today.
Also, with the advent of B2BUAs as the SBE/DBEs in the call flows, there
would be numerous iterations of SIP Header manipulation/mangling that is
done for various business, regulatory and other reasons.  It could be a
very slippery slope to start digging into the details, and would also
balloon the draft greatly.

To summarize, I am on the fence with this one.  Others have thoughts?
Will the details help clarify, or narrow the discussion too much?

-Adam

> -----Original Message-----
> From: speermint-bounces@ietf.org
> [mailto:speermint-bounces@ietf.org] On Behalf Of Daryl Malas
> Sent: Monday, February 25, 2008 11:38 PM
> To: Uzelac, Adam; speermint@ietf.org
> Subject: Re: 
> [Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> "VoIP consolidated use cases" author's,
> 
> First, let me say, I think the draft is off to a good start.  
> I would like to suggest some things, I believe will allow the draft to

> provide better clarity and more depth for a reader to understand the 
> reasons and situations relating to VoIP peering.
> 
> Here are a few encompassing comments and general nits of the document:
> 
> - I recommend looking through RFC 3285 before the next revision of the

> draft.  There are a number of Word template issues, which can be 
> resolved fairly easy.
> 
> - The references need to be updated.
> 
> - Finally, as I review the use cases, I am looking for the following 
> basic structure within each use case: Goal, Summary, Actors, 
> Preconditions, Triggers, Basic course of events, Alternative paths (if

> applicable), and any Policy considerations.
> 
> --------------------------------
> 
> More detailed comments:
> 
> Section 4 - "Each grouping can further sub-divided to Direct Peering 
> and Indirect Peering [15]."  Should it be, "Each grouping can be 
> further sub-divided to Direct and Indirect Peering [15]"?
> 
> "Each use-case has to specify how a few core operations which are to 
> be performed by its members."  I would like to suggest re-wording to, 
> "Each use case must specify a basic set of required operations to be 
> performed by each member when peering."
> 
> Section 5.1 - I would expect this intro paragraph to describe the goal

> of static peering.  For example, "Static peering  is used in cases, 
> when two peers want a consistent and "hardened" approach to peering.
> In this scenario, all of the variables are very well defined and known

> by each peer....."
> 
> Section 5.1.1 - Who are the actors (e.g. Enterprise A wants to peer 
> with Enterprise B, SSP A wants to peer with SSP B, SSP A is providing 
> static peering for Enterprise A, B, and C, etc.)?  What are the 
> preconditions required for this use case to be successful (e.g. each 
> peer must exchange their allowable set of codecs, authentication 
> method, no
> RFC1918 addresses, etc.)?  What are the triggers (i.e. what causes 
> SSPA to think it can send it to a peer?)
> 
> The sentence beginning, "Two SSP negotiate and agree..." 
> seems to begin the course of events, but you have other events 
> numbered below.  This is confusing.  I would suggest moving these into

> the numbered events below.
> 
> Each of the events listed, such provide an example of what is 
> occurring with each.  The current ones are too high level of value in 
> deriving requirements.  For example:
> 
> 1. UAC initiates a call via SIP INVITE to O-Proxy
> 
> (Snippets frm RFC 3665)	
>    INVITE sip:+13031112222@denver.example.com SIP/2.0
>    Via: SIP/2.0/TCP
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
>    
> 2. O-Proxy queries for SED information from a routing database (ENUM??

> -
> LUF)
> 
> (The result of the ENUM query is...)
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\1@sbe1-denver.sspa.example.com!".
> 
> 3. Routing database entity replies with SED to called party (NAPTR -
> LRF)
> 
> sbe1-denver.sspa.example.com.
>       order pref flgs service  regx    replacement
>    IN NAPTR 100  50   "s"   "SIP+D2U" ""
> _sip._udp.sbe1-denver.sspa.example.com
> 
> 4. O-Proxy sends a DNS SRV query for
> _sip._udp.sbe1-denver.sspa.example.com which results in:
> 
>    $ORIGIN _sip._udp.sbe1-denver.sspa.example.com.
>     (Some content not included for clarity purposes.)
>    IN SRV  0  1   5060   sbe1-denver.sspa.example.com
> 
>    ;; ANSWER SECTION
>    sbe1-denver.sspa.example.com   A   192.0.2.17
>    sbe2-denver.sspa.example.com   A   192.0.2.18
> 
> 5.  Call sent to terminating domain proxy.
> 
> INVITE sip:+13031112222@sbe1-denver.sspa.example.com SIP/2.0
>    Via: SIP/2.0/TCP
> client.atlanta.example.com:5060;branch=z9hG4bK74bf9
>    Max-Forwards: 70
>    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
>    To: +13031112222 <sip:+13031112222@denver.example.com>
>    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
> 
> 6.  T-Proxy determines called party location by querying the ENUM 
> server
> 
> $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
>    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
>              !sip:\\bob@proxy1-denver.sspa.example.com!".
> 
> ....and so on.  I think this detail will provide clarity for the 
> requirements and for readers to understand the necessary steps to 
> successfully peer.  I think this should be done for each of the use 
> cases.
> 
> Section 5.1.1 - These appear to be optional (or additional
> potential) pre-conditions, which could occur relative to this use 
> case.  If this is case, I would suggest calling them Optional 
> Pre-conditions and not Options and Nuances.
> 
> ---------------------------------
> 
> I think making similar, relative changes to each of the use case 
> sections will get the very close to a finished product.
> 
> --Daryl
> 
> -----Original Message-----
> From: speermint-bounces@ietf.org
> [mailto:speermint-bounces@ietf.org] On Behalf Of Uzelac, Adam
> Sent: Friday, February 22, 2008 12:37 PM
> To: speermint@ietf.org
> Subject: [Speermint]
> draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> For general consumption in prep for Philly.  See y'all in the city of 
> brotherly love!  Feedback welcomed.
> 
> Changes:
> - resync with the latest term draft.
> - clarifications based on feedback from the last meeting - like 
> collapse/removal of assisted peering use cases into direct/indirect 
> models.
> 
> http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-
> consolidat
> ed-usecases-05.txt
> 
> I am working on version 6, mostly weeding out nits, and will issue 
> after the WG session.  Thanks in advance for your feedback!!
> 
> Adam Uzelac
> Director, Converged Network Arch and Planning Global Crossing
> 585-563-4359
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From: "Daryl Malas" <D.Malas@cablelabs.com>
To: <otmar.lendl@enum.at>
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Otmar,

I also enjoyed reading the draft.  I think you bring up some very
interesting and valid discussion points.  I also think you re-enforce
there is much to do within SPEERMINT.

A comment to the use case authors...please read this draft, because I
think it has some important topics to touch on in your draft.

I encourage everyone to take the time to read this draft prior to
Philadelphia.

Thank you, Otmar, for taking the time to update this draft. 

--Daryl

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Hadriel Kaplan
Sent: Monday, February 25, 2008 4:24 PM
To: otmar.lendl@enum.at
Cc: speermint@ietf.org; PEPPERMINT@ietf.org
Subject: Re: [Speermint] I-D
Action:draft-lendl-speermint-background-01.txt


Awesome draft. (I missed it the first time 'round)

I agree with your statement of "The obvious solution is to add an
on-demand mapping step ahead of the routing protocol."  The trick will
be how that works/scales, and how it interacts with a routing protocol.
The easy case is the non-transit bilateral peering - the very hard case
is transit.

One very minor nit: I don't think TRIP failed due to being prefix-based
and thus needing aggregation.  I think it failed because the market
wasn't ready for it (the problem didn't need to be fixed for most people
yet), the idea of implementing/using BGP scared off voip vendors, and it
didn't cover some of the routing selection models people need.

-hadriel


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>
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
>
>         Title           : VoIP Peering: Background and Assumptions
>         Author(s)       : O. Lendl
>         Filename        : draft-lendl-speermint-background-01.txt
>         Pages           : 15
>         Date            : 2008-02-25
>
> This documents provides background for the work on VoIP peering and 
> tries to provide guidance on what kind of work is needed to facilitate

> widespread SIP-based peering of telephony networks.  It is intended to

> spur discussion on the work about peering and should also serve as 
> input to the ongoing discussions on reducing Spam for Internet 
> Telephony (SPIT).
>
> A URL for this Internet-Draft is:
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> 01.txt
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From: "Michael Hammer (mhammer)" <mhammer@cisco.com>
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Adam,

Only the idea I expressed before that since the main gist of this is
call routing, 
the main SIP header that should be mangled is the Route header.

Mike


> -----Original Message-----
> From: speermint-bounces@ietf.org 
> [mailto:speermint-bounces@ietf.org] On Behalf Of Uzelac, Adam
> Sent: Tuesday, February 26, 2008 10:02 AM
> To: Daryl Malas; speermint@ietf.org
> Subject: Re: 
> [Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt
> 
> Daryl,
> 
> Excellent comments and thanks for the feedback.  We have 
> version 6 queued up and are in the process of updating.  
> Regarding some of your comments, starting with the "(Snippets 
> frm RFC 3665)" - you state: 
> 
> > ...I think this detail will provide clarity for the 
> requirements and 
> > for readers to understand the necessary steps to 
> successfully peer.  I 
> > think this should be done for each of the use cases.
> 
> I read that you are requesting more detail in the use cases, 
> but I feel it isn't necessary to get the pertinent points 
> across.  Initially, and I believe to a certain extent still, 
> one goal of capturing current use cases (what is "occurring 
> in the wild today), is to capture requirements.  At the same 
> time, I think that a goal in capturing the VoIP use cases at 
> present is to identify the "ugliness", like limited-to-no 
> transparency of SIP/VoIP Peering as it exists today.
> Also, with the advent of B2BUAs as the SBE/DBEs in the call 
> flows, there would be numerous iterations of SIP Header 
> manipulation/mangling that is done for various business, 
> regulatory and other reasons.  It could be a very slippery 
> slope to start digging into the details, and would also 
> balloon the draft greatly.
> 
> To summarize, I am on the fence with this one.  Others have thoughts?
> Will the details help clarify, or narrow the discussion too much?
> 
> -Adam
> 
> > -----Original Message-----
> > From: speermint-bounces@ietf.org
> > [mailto:speermint-bounces@ietf.org] On Behalf Of Daryl Malas
> > Sent: Monday, February 25, 2008 11:38 PM
> > To: Uzelac, Adam; speermint@ietf.org
> > Subject: Re: 
> > [Speermint]draft-ietf-speermint-voip-consolidated-usecases-05.txt
> > 
> > "VoIP consolidated use cases" author's,
> > 
> > First, let me say, I think the draft is off to a good start.  
> > I would like to suggest some things, I believe will allow 
> the draft to 
> > provide better clarity and more depth for a reader to 
> understand the 
> > reasons and situations relating to VoIP peering.
> > 
> > Here are a few encompassing comments and general nits of 
> the document:
> > 
> > - I recommend looking through RFC 3285 before the next 
> revision of the 
> > draft.  There are a number of Word template issues, which can be 
> > resolved fairly easy.
> > 
> > - The references need to be updated.
> > 
> > - Finally, as I review the use cases, I am looking for the 
> following 
> > basic structure within each use case: Goal, Summary, Actors, 
> > Preconditions, Triggers, Basic course of events, 
> Alternative paths (if 
> > applicable), and any Policy considerations.
> > 
> > --------------------------------
> > 
> > More detailed comments:
> > 
> > Section 4 - "Each grouping can further sub-divided to 
> Direct Peering 
> > and Indirect Peering [15]."  Should it be, "Each grouping can be 
> > further sub-divided to Direct and Indirect Peering [15]"?
> > 
> > "Each use-case has to specify how a few core operations 
> which are to 
> > be performed by its members."  I would like to suggest 
> re-wording to, 
> > "Each use case must specify a basic set of required 
> operations to be 
> > performed by each member when peering."
> > 
> > Section 5.1 - I would expect this intro paragraph to 
> describe the goal 
> > of static peering.  For example, "Static peering  is used in cases, 
> > when two peers want a consistent and "hardened" approach to 
> peering.  
> > In this scenario, all of the variables are very well 
> defined and known 
> > by each peer....."
> > 
> > Section 5.1.1 - Who are the actors (e.g. Enterprise A wants to peer 
> > with Enterprise B, SSP A wants to peer with SSP B, SSP A is 
> providing 
> > static peering for Enterprise A, B, and C, etc.)?  What are the 
> > preconditions required for this use case to be successful 
> (e.g. each 
> > peer must exchange their allowable set of codecs, authentication 
> > method, no
> > RFC1918 addresses, etc.)?  What are the triggers (i.e. what causes 
> > SSPA to think it can send it to a peer?)
> > 
> > The sentence beginning, "Two SSP negotiate and agree..." 
> > seems to begin the course of events, but you have other events 
> > numbered below.  This is confusing.  I would suggest moving 
> these into 
> > the numbered events below.
> > 
> > Each of the events listed, such provide an example of what is 
> > occurring with each.  The current ones are too high level 
> of value in 
> > deriving requirements.  For example:
> > 
> > 1. UAC initiates a call via SIP INVITE to O-Proxy
> > 
> > (Snippets frm RFC 3665)	
> >    INVITE sip:+13031112222@denver.example.com SIP/2.0
> >    Via: SIP/2.0/TCP
> > client.atlanta.example.com:5060;branch=z9hG4bK74bf9
> >    Max-Forwards: 70
> >    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
> >    To: +13031112222 <sip:+13031112222@denver.example.com>
> >    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
> >    
> > 2. O-Proxy queries for SED information from a routing 
> database (ENUM?? 
> > -
> > LUF)
> > 
> > (The result of the ENUM query is...)
> > 
> > $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
> >    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
> >              !sip:\\1@sbe1-denver.sspa.example.com!".
> > 
> > 3. Routing database entity replies with SED to called party (NAPTR -
> > LRF)
> > 
> > sbe1-denver.sspa.example.com.
> >       order pref flgs service  regx    replacement
> >    IN NAPTR 100  50   "s"   "SIP+D2U" ""
> > _sip._udp.sbe1-denver.sspa.example.com
> > 
> > 4. O-Proxy sends a DNS SRV query for
> > _sip._udp.sbe1-denver.sspa.example.com which results in:
> > 
> >    $ORIGIN _sip._udp.sbe1-denver.sspa.example.com.
> >     (Some content not included for clarity purposes.)
> >    IN SRV  0  1   5060   sbe1-denver.sspa.example.com
> > 
> >    ;; ANSWER SECTION
> >    sbe1-denver.sspa.example.com   A   192.0.2.17
> >    sbe2-denver.sspa.example.com   A   192.0.2.18
> > 
> > 5.  Call sent to terminating domain proxy.
> > 
> > INVITE sip:+13031112222@sbe1-denver.sspa.example.com SIP/2.0
> >    Via: SIP/2.0/TCP
> > client.atlanta.example.com:5060;branch=z9hG4bK74bf9
> >    Max-Forwards: 70
> >    From: Alice <sip:alice@atlanta.example.com>;tag=9fxced76sl
> >    To: +13031112222 <sip:+13031112222@denver.example.com>
> >    Contact: <sip:alice@client.atlanta.example.com;transport=tcp>
> > 
> > 6.  T-Proxy determines called party location by querying the ENUM 
> > server
> > 
> > $ORIGIN 2.2.2.2.1.1.1.3.0.3.1.denver.example.com
> >    NAPTR 10 100 "u" "E2U+SIP" "!^.*$
> >              !sip:\\bob@proxy1-denver.sspa.example.com!".
> > 
> > ....and so on.  I think this detail will provide clarity for the 
> > requirements and for readers to understand the necessary steps to 
> > successfully peer.  I think this should be done for each of the use 
> > cases.
> > 
> > Section 5.1.1 - These appear to be optional (or additional
> > potential) pre-conditions, which could occur relative to this use 
> > case.  If this is case, I would suggest calling them Optional 
> > Pre-conditions and not Options and Nuances.
> > 
> > ---------------------------------
> > 
> > I think making similar, relative changes to each of the use case 
> > sections will get the very close to a finished product.
> > 
> > --Daryl
> > 
> > -----Original Message-----
> > From: speermint-bounces@ietf.org
> > [mailto:speermint-bounces@ietf.org] On Behalf Of Uzelac, Adam
> > Sent: Friday, February 22, 2008 12:37 PM
> > To: speermint@ietf.org
> > Subject: [Speermint]
> > draft-ietf-speermint-voip-consolidated-usecases-05.txt
> > 
> > For general consumption in prep for Philly.  See y'all in 
> the city of 
> > brotherly love!  Feedback welcomed.
> > 
> > Changes:
> > - resync with the latest term draft.
> > - clarifications based on feedback from the last meeting - like 
> > collapse/removal of assisted peering use cases into direct/indirect 
> > models.
> > 
> > http://www.ietf.org/internet-drafts/draft-ietf-speermint-voip-
> > consolidat
> > ed-usecases-05.txt
> > 
> > I am working on version 6, mostly weeding out nits, and will issue 
> > after the WG session.  Thanks in advance for your feedback!!
> > 
> > Adam Uzelac
> > Director, Converged Network Arch and Planning Global Crossing
> > 585-563-4359
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > http://www.ietf.org/mailman/listinfo/speermint
> > _______________________________________________
> > Speermint mailing list
> > Speermint@ietf.org
> > http://www.ietf.org/mailman/listinfo/speermint
> > 
> _______________________________________________
> Speermint mailing list
> Speermint@ietf.org
> http://www.ietf.org/mailman/listinfo/speermint
> 
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To: "Daryl Malas" <D.Malas@cablelabs.com>,
	<otmar.lendl@enum.at>
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I read the first 00 draft and enjoyed it. I will definitely read this
one. Thanks Otmar for taking the time to write it.

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Tuesday, February 26, 2008 12:59 PM
To: otmar.lendl@enum.at
Cc: speermint@ietf.org
Subject: Re: [Speermint] I-D
Action:draft-lendl-speermint-background-01.txt

Otmar,

I also enjoyed reading the draft.  I think you bring up some very
interesting and valid discussion points.  I also think you re-enforce
there is much to do within SPEERMINT.

A comment to the use case authors...please read this draft, because I
think it has some important topics to touch on in your draft.

I encourage everyone to take the time to read this draft prior to
Philadelphia.

Thank you, Otmar, for taking the time to update this draft. 

--Daryl

-----Original Message-----
From: speermint-bounces@ietf.org [mailto:speermint-bounces@ietf.org] On
Behalf Of Hadriel Kaplan
Sent: Monday, February 25, 2008 4:24 PM
To: otmar.lendl@enum.at
Cc: speermint@ietf.org; PEPPERMINT@ietf.org
Subject: Re: [Speermint] I-D
Action:draft-lendl-speermint-background-01.txt


Awesome draft. (I missed it the first time 'round)

I agree with your statement of "The obvious solution is to add an
on-demand mapping step ahead of the routing protocol."  The trick will
be how that works/scales, and how it interacts with a routing protocol.
The easy case is the non-transit bilateral peering - the very hard case
is transit.

One very minor nit: I don't think TRIP failed due to being prefix-based
and thus needing aggregation.  I think it failed because the market
wasn't ready for it (the problem didn't need to be fixed for most people
yet), the idea of implementing/using BGP scared off voip vendors, and it
didn't cover some of the routing selection models people need.

-hadriel


> -----Original Message-----
> From: i-d-announce-bounces@ietf.org
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> Sent: Monday, February 25, 2008 4:15 PM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-lendl-speermint-background-01.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
>
>         Title           : VoIP Peering: Background and Assumptions
>         Author(s)       : O. Lendl
>         Filename        : draft-lendl-speermint-background-01.txt
>         Pages           : 15
>         Date            : 2008-02-25
>
> This documents provides background for the work on VoIP peering and 
> tries to provide guidance on what kind of work is needed to facilitate

> widespread SIP-based peering of telephony networks.  It is intended to

> spur discussion on the work about peering and should also serve as 
> input to the ongoing discussions on reducing Spam for Internet 
> Telephony (SPIT).
>
> A URL for this Internet-Draft is:
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> 01.txt
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From: Dan York <dyork@voxeo.com>
Date: Tue, 26 Feb 2008 17:49:52 -0500
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Saverio,

On Feb 26, 2008, at 9:13 AM, Saverio Niccolini wrote:

> Right, we explicitly did not look at these and if the group
> has consensus on this we can add them in the next version of the
> draft. Anyway we need to be sure that the threats we are addressing
> are specific to SPEERMINT, we excluded domain-internal threats
> from the classification (not interesting to SPEERMINT the REGISTER
> after loss of power example you cited).

DY> Yes, I realized that my loss-of-power example was NOT applicable  
to SPEERMINT.

> Can you suggest (maybe offline) more specifically which kind of
> threats that are SPEERMINT specific did you have in mind?


DY> Unfortunately I haven't spent enough time yet analyzing the  
SPEERMINT architecture to be able to provide solid feedback on this  
point.  In my initial thinking, there aren't many cases that leap out  
at me.  Perhaps the misconfiguration of a SIP proxy at a provider's  
network that starts spewing a flood of packets out into the peering  
fabric?  I'm not honestly sure... but I will give it some thought.

Regards,
Dan

-- 
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork@voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Bring your web applications to the phone.
Find out how at http://evolution.voxeo.com





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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">
Saverio,<div><br><div><div>On Feb 26, 2008, at 9:13 AM, Saverio =
Niccolini wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; "><span class=3D"Apple-style-span" =
style=3D"-webkit-text-stroke-width: -1; ">Right, we explicitly did not =
look at these and if the group</span></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">has =
consensus on this we can add them in the next version of the</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">draft. Anyway we need to be sure that the threats we =
are addressing</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">are specific to SPEERMINT, we =
excluded domain-internal threats</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">from the =
classification (not interesting to SPEERMINT the REGISTER</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">after loss of power example you =
cited).</div></blockquote><div><br =
class=3D"webkit-block-placeholder"></div>DY&gt; Yes, I realized that my =
loss-of-power example was NOT applicable to =
SPEERMINT.</div><div><br><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Can you suggest (maybe offline) more specifically =
which kind of</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; ">threats that are SPEERMINT =
specific did you have in mind?</div> </blockquote></div><div><br =
class=3D"webkit-block-placeholder"></div>DY&gt; Unfortunately I haven't =
spent enough time yet analyzing the SPEERMINT architecture to be able to =
provide solid feedback on this point. =A0In my initial thinking, there =
aren't many cases that leap out at me. =A0Perhaps the misconfiguration =
of a SIP proxy at a provider's network that starts spewing a flood of =
packets out into the peering fabric? =A0I'm not honestly sure... but I =
will give it some thought.</div><div><br =
class=3D"webkit-block-placeholder"></div><div>Regards,</div><div>Dan</div>=
<div><br><div> <span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; border-spacing: 0px 0px; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
text-align: auto; -khtml-text-decorations-in-effect: none; text-indent: =
0px; -apple-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">--=A0</div><div style=3D"margin-top: 0px; =
margin-right: 0px; margin-bottom: 0px; margin-left: 0px; ">Dan York, =
CISSP, Director of Emerging Communication Technology</div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Office of the CTO=A0 =A0<span =
class=3D"Apple-converted-space">=A0</span>Voxeo Corporation<span =
class=3D"Apple-converted-space">=A0</span>=A0 =A0<span =
class=3D"Apple-converted-space">=A0</span><a =
href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; ">Phone: +1-407-455-5859=A0<span =
class=3D"Apple-converted-space">=A0</span>Skype: danyork=A0<span =
class=3D"Apple-converted-space">=A0</span><a =
href=3D"http://www.voxeo.com">http://www.voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
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Subject: [Speermint] Draft IETF 71 Agenda
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Please comment if you would like any changes or additions.

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