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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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inline...

Avasarala Ranjit-A20990 wrote:

> I'm still a little confused here.
> If phone is on and subscription is active, then in general notification
> should be possible, regardless of registration. I realize IMS doesn't
> permit that, but from a general standard perspective we shouldn't assume
> it. The manifestation in this case is that the NOTIFY would fail and the
> subscription would eventually be torn down.
> 
> So I think the cases are:
> - there is a subscription and notification works
> - there is a subscription but notification fails
> - there is no subscription.
> 
> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as Notification
> fails, we can say - not sending notification due to (1) no match of
> filter criteria (2) phone not registered or outside coverage area due to
> which there is no connectivity and hence notification cannot be sent 
> Anyway if no subscription, no notification. 

Are you suggesting that you might have a subscription, and have the 
filter criteria met, and yet not attempt to send a NOTIFY because the 
phone is not registered or has no network connectivity???

That seems wrong. ISTM that either you would remove the subscription, or 
else you would try to send the NOTIFY and then discover that it fails.

OTOH, this probably ought not to be within scope of what is discussed in 
this draft.

>> [3] in the example u suggested where sip:alice@atlanta.org with two
>> contacts: sip:line1@alice.office.atlanta.com and 
>> sip:line2@alice.home.atlanta.com and
>>     both the phones ring, then it would be a case of parallel forking.
>> Now say one or both the phones have set diversion rules, to divert the
> 
> In this case, what does it mean for "one or both phones have set
> diversion rules"? Does that somehow imply distinct rules for each phone?
> 
> I don't see how that could work. Aren't the diversion rules for the
> *AOR*?
> 
> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there is a
> diversion rule set for the AoR and when diversions occur, the
> notifications are
> Sent to the AoR.

Not sent to the AoR - sent to the remote contact of the dialog for each 
subscription to the AoR. Same point as below that you agreed with.

> I see they could be set/changed from either phone. But its a single set
> of rules isn't it? So I might set the rules from the office, and then
> cancel them from home.
> 
> <Ranjit-Feb26-2> yes single set and could be set from either of the
> phones. True.
> 
>> call to say 
>>     sip:voicemail@alice.atlanta.com, then the notifications would get 
>> generated with respect to both the contacts.
> 
> Again I'm confused by the terminology you use, and how it relates
> operationally. The diversion happens on the AOR, and the subscriptions
> are on the AOR, and the notifications are sent to the subscribers. I
> don't see where the registered contacts enter into the picture at all.
> 
> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
> 
> If you have something else in mind, I hope you can explain it better in
> the draft.
> 
>> [4] Yes the value of N is configurable and depends on how many 
>> diversions the subscribe wants to keep or could be dependent on server
> 
>> policy (some maximum
>>     number). If the server policy is to send notifications as and when
> 
>> they occur and not keep any divsersions information, then the value of
> 
>> last N
>>     diversions will be zero. But if the server chooses to retain the 
>> last diversion always, then the value of N will be 1 and so on.
> 
> All is straightforward for N>=1.
> 
> For N=0 things are messy. If we are modeling this as state, and
> notifications are about state changes, then the new diversion is added
> to the (previously empty) state and that triggers notifications to all
> current subscriptions. Then if you don't want to retain even this much
> state (because N=0) you immediately remove that version from the state. 
> But that is again a state change, so you end up with another
> notification of that state change.
> 
> <Ranjit-Feb26-2> True. Actually the notifications are for informing the
> subscriber of the call diversions that happen. The notifications are not
> for informing the subscriber of state changes in the notifier or the
> number of diversions that occurred. As I mentioned in the abstract as
> well as 24.604, the main intent of CDIVN service is to inform the
> subscribers of communication diversions that occur on their behalf in
> the network and this information will help them manage their rules
> better. 

I am not certain, but think we may still be talking past one another to 
some extent.

The state can be considered just a formalism for defining the service 
clearly, regardless of whether it has intrinsic value to the CDIV 
service. But it *is* a formalism. So if it is introduced, then its 
behavior can't be ignored when inconvenient. Defining the state with N=0 
leads to some undesired behavior. The simplest way around that is to 
insist that N be >=1.

If its important to you that no state be retained once a notification 
has been sent about the most recent diversion, then you will have to 
find some other way to specify the service, that still meets the 
expectations of 3265. I don't know if there is anything there that Adam 
would consider valid. We will have to get his opinion on that.

	Thanks,
	Paul

> <Ranjit-Feb26-2> But if we want to include the information related to
> how many diversions have happened till time (say from 12 AM ) till the
> time the notification is being sent, then as you say we can store the
> diversions. This would be useful when the subscriber specifies in the
> filter criteria to send notifications only during certain time like
> between 5 and 6 PM and not whenever a diversion occurs. Then N is grater
> than 1.

> I presume you don't want that behavior. If you can live with N>=1 then
> we don't have the problem at all. If you need to support N=0 and don't
> want the extra notify, then we have to figure out if there is some trick
> to how things are modeled to get the desired effect. I've suggested
> something previously, but Adam didn't like it.
> 
> This needs sorting out.
> 
> 	Thanks,
> 	Paul

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Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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On 3/1/10 14:45, Mar 1, Paul Kyzivat wrote:
> Are you suggesting that you might have a subscription, and have the 
> filter criteria met, and yet not attempt to send a NOTIFY because the 
> phone is not registered or has no network connectivity???
>
> That seems wrong.

It seems wrong because it is wrong. If the terminal cannot be reached, 
the subscription goes away.

/a

From D.Malas@cablelabs.com  Mon Mar  1 15:50:53 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'dispatch@ietf.org'" <dispatch@ietf.org>
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Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt <mailto:adam@nostrum.com>

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<BODY>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D500354723-01032010>All,</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D500354723-01032010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial>A New Internet-Draft is available from the=
 on-line=20
Internet-Drafts directories. <BR>&nbsp;<BR>Title : SIP Egress Route Use in =
ENUM=20
<BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
<A href=3D"mailto:adam@nostrum.com"></A></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D500354723-01032010>Regards,</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D500354723-01032010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D500354723-01032010>Daryl</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial></FONT>&nbsp;</DIV>
<DIV align=3Dleft>
<DIV><SPAN class=3D715351122-14012010><FONT size=3D2=20
face=3DArial>---------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=3D715351122-14012010><FONT size=3D2 face=3DArial>Daryl=20
Malas</FONT></SPAN></DIV>
<DIV><SPAN class=3D715351122-14012010><FONT size=3D2 face=3DArial>Project D=
irector, IP=20
Interconnects</FONT></SPAN></DIV>
<DIV><SPAN class=3D715351122-14012010><FONT size=3D2 face=3DArial>w - +1 30=
3 661=20
3302</FONT></SPAN></DIV>
<DIV><SPAN class=3D715351122-14012010><FONT size=3D2 face=3DArial><A=20
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com=
</A></FONT></SPAN></DIV></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Inline

<Ranjit-Mar2>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, March 02, 2010 2:16 AM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

inline...

Avasarala Ranjit-A20990 wrote:

> I'm still a little confused here.
> If phone is on and subscription is active, then in general=20
> notification should be possible, regardless of registration. I realize

> IMS doesn't permit that, but from a general standard perspective we=20
> shouldn't assume it. The manifestation in this case is that the NOTIFY

> would fail and the subscription would eventually be torn down.
>=20
> So I think the cases are:
> - there is a subscription and notification works
> - there is a subscription but notification fails
> - there is no subscription.
>=20
> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as=20
> Notification fails, we can say - not sending notification due to (1)=20
> no match of filter criteria (2) phone not registered or outside=20
> coverage area due to which there is no connectivity and hence=20
> notification cannot be sent Anyway if no subscription, no
notification.

Are you suggesting that you might have a subscription, and have the
filter criteria met, and yet not attempt to send a NOTIFY because the
phone is not registered or has no network connectivity???

That seems wrong. ISTM that either you would remove the subscription, or
else you would try to send the NOTIFY and then discover that it fails.

<Ranjit-Mar2> yes the subscription could be removed when the device is
not reachable.=20

OTOH, this probably ought not to be within scope of what is discussed in
this draft.

<Ranjit-Mar2> yes true not within the scope of this draft.

>> [3] in the example u suggested where sip:alice@atlanta.org with two
>> contacts: sip:line1@alice.office.atlanta.com and=20
>> sip:line2@alice.home.atlanta.com and
>>     both the phones ring, then it would be a case of parallel
forking.
>> Now say one or both the phones have set diversion rules, to divert=20
>> the
>=20
> In this case, what does it mean for "one or both phones have set=20
> diversion rules"? Does that somehow imply distinct rules for each
phone?
>=20
> I don't see how that could work. Aren't the diversion rules for the=20
> *AOR*?
>=20
> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there is

> a diversion rule set for the AoR and when diversions occur, the=20
> notifications are Sent to the AoR.

Not sent to the AoR - sent to the remote contact of the dialog for each
subscription to the AoR. Same point as below that you agreed with.

<Ranjit-Mar2> Yes agree with u.=20

> I see they could be set/changed from either phone. But its a single=20
> set of rules isn't it? So I might set the rules from the office, and=20
> then cancel them from home.
>=20
> <Ranjit-Feb26-2> yes single set and could be set from either of the=20
> phones. True.
>=20
>> call to say=20
>>     sip:voicemail@alice.atlanta.com, then the notifications would get

>> generated with respect to both the contacts.
>=20
> Again I'm confused by the terminology you use, and how it relates=20
> operationally. The diversion happens on the AOR, and the subscriptions

> are on the AOR, and the notifications are sent to the subscribers. I=20
> don't see where the registered contacts enter into the picture at all.
>=20
> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>=20
> If you have something else in mind, I hope you can explain it better=20
> in the draft.
>=20
>> [4] Yes the value of N is configurable and depends on how many=20
>> diversions the subscribe wants to keep or could be dependent on=20
>> server
>=20
>> policy (some maximum
>>     number). If the server policy is to send notifications as and=20
>> when
>=20
>> they occur and not keep any divsersions information, then the value=20
>> of
>=20
>> last N
>>     diversions will be zero. But if the server chooses to retain the=20
>> last diversion always, then the value of N will be 1 and so on.
>=20
> All is straightforward for N>=3D1.
>=20
> For N=3D0 things are messy. If we are modeling this as state, and=20
> notifications are about state changes, then the new diversion is added

> to the (previously empty) state and that triggers notifications to all

> current subscriptions. Then if you don't want to retain even this much

> state (because N=3D0) you immediately remove that version from the
state.
> But that is again a state change, so you end up with another=20
> notification of that state change.
>=20
> <Ranjit-Feb26-2> True. Actually the notifications are for informing=20
> the subscriber of the call diversions that happen. The notifications=20
> are not for informing the subscriber of state changes in the notifier=20
> or the number of diversions that occurred. As I mentioned in the=20
> abstract as well as 24.604, the main intent of CDIVN service is to=20
> inform the subscribers of communication diversions that occur on their

> behalf in the network and this information will help them manage their

> rules better.

I am not certain, but think we may still be talking past one another to
some extent.

The state can be considered just a formalism for defining the service
clearly, regardless of whether it has intrinsic value to the CDIV
service. But it *is* a formalism. So if it is introduced, then its
behavior can't be ignored when inconvenient. Defining the state with =
N=3D0
leads to some undesired behavior. The simplest way around that is to
insist that N be >=3D1.

<Ranjit-Mar2> Ok in that case we can say that value of N >=3D1 and the
notifier has always maintains history of last diversion at a minimu. It
can maintain more than 1 too depending on the policy.=20

If its important to you that no state be retained once a notification
has been sent about the most recent diversion, then you will have to
find some other way to specify the service, that still meets the
expectations of 3265. I don't know if there is anything there that Adam
would consider valid. We will have to get his opinion on that.

<Ranjit-Mar2> In that case I am OK maintaining the hostory of diversions
that occurred as part of state information and retaining that state even
after the notification for that diversion is sent - so that in case I do
not get a confirmation, I would be able to re-send the notification.
Also maintaining state of diversions would help when the notifications
need to sent only at a particular time (if fikter criteria mentions it).


	Thanks,
	Paul

> <Ranjit-Feb26-2> But if we want to include the information related to=20
> how many diversions have happened till time (say from 12 AM ) till the

> time the notification is being sent, then as you say we can store the=20
> diversions. This would be useful when the subscriber specifies in the=20
> filter criteria to send notifications only during certain time like=20
> between 5 and 6 PM and not whenever a diversion occurs. Then N is=20
> grater than 1.

> I presume you don't want that behavior. If you can live with N>=3D1 =
then

> we don't have the problem at all. If you need to support N=3D0 and =
don't

> want the extra notify, then we have to figure out if there is some=20
> trick to how things are modeled to get the desired effect. I've=20
> suggested something previously, but Adam didn't like it.
>=20
> This needs sorting out.
>=20
> 	Thanks,
> 	Paul

From pkyzivat@cisco.com  Tue Mar  2 05:48:46 2010
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To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
References: <4B7EF1DA.60002@cisco.com> <4B7F1368.6040603@cisco.com><4B7F223D.3080008@cisco.com> <4B7F2CFD.8070901@cisco.com> <4B7FFA2B.5080303@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A31824D@ZMY16EXM66.ds.mot.com> <4B832169.8090100@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A318476@ZMY16EXM66.ds.mot.com> <4B83E1D4.7040108@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A318504@ZMY16EXM66.ds.mot.com> <4B8422E6.1060709@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A318794@ZMY16EXM66.ds.mot.com> <4B86E6AE.6010308@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A31899F@ZMY16EXM66.ds.mot.com> <4B87DC79.6010104@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A318AAF@ZMY16EXM66.ds.mot.com> <4B8C277E.4000701@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A318C2D@ZMY16EXM66.ds.mot.com>
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I think this all sounds good now.

	Thanks,
	Paul

Avasarala Ranjit-A20990 wrote:
> Inline
> 
> <Ranjit-Mar2> 
> 
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Tuesday, March 02, 2010 2:16 AM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> inline...
> 
> Avasarala Ranjit-A20990 wrote:
> 
>> I'm still a little confused here.
>> If phone is on and subscription is active, then in general 
>> notification should be possible, regardless of registration. I realize
> 
>> IMS doesn't permit that, but from a general standard perspective we 
>> shouldn't assume it. The manifestation in this case is that the NOTIFY
> 
>> would fail and the subscription would eventually be torn down.
>>
>> So I think the cases are:
>> - there is a subscription and notification works
>> - there is a subscription but notification fails
>> - there is no subscription.
>>
>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as 
>> Notification fails, we can say - not sending notification due to (1) 
>> no match of filter criteria (2) phone not registered or outside 
>> coverage area due to which there is no connectivity and hence 
>> notification cannot be sent Anyway if no subscription, no
> notification.
> 
> Are you suggesting that you might have a subscription, and have the
> filter criteria met, and yet not attempt to send a NOTIFY because the
> phone is not registered or has no network connectivity???
> 
> That seems wrong. ISTM that either you would remove the subscription, or
> else you would try to send the NOTIFY and then discover that it fails.
> 
> <Ranjit-Mar2> yes the subscription could be removed when the device is
> not reachable. 
> 
> OTOH, this probably ought not to be within scope of what is discussed in
> this draft.
> 
> <Ranjit-Mar2> yes true not within the scope of this draft.
> 
>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>> contacts: sip:line1@alice.office.atlanta.com and 
>>> sip:line2@alice.home.atlanta.com and
>>>     both the phones ring, then it would be a case of parallel
> forking.
>>> Now say one or both the phones have set diversion rules, to divert 
>>> the
>> In this case, what does it mean for "one or both phones have set 
>> diversion rules"? Does that somehow imply distinct rules for each
> phone?
>> I don't see how that could work. Aren't the diversion rules for the 
>> *AOR*?
>>
>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there is
> 
>> a diversion rule set for the AoR and when diversions occur, the 
>> notifications are Sent to the AoR.
> 
> Not sent to the AoR - sent to the remote contact of the dialog for each
> subscription to the AoR. Same point as below that you agreed with.
> 
> <Ranjit-Mar2> Yes agree with u. 
> 
>> I see they could be set/changed from either phone. But its a single 
>> set of rules isn't it? So I might set the rules from the office, and 
>> then cancel them from home.
>>
>> <Ranjit-Feb26-2> yes single set and could be set from either of the 
>> phones. True.
>>
>>> call to say 
>>>     sip:voicemail@alice.atlanta.com, then the notifications would get
> 
>>> generated with respect to both the contacts.
>> Again I'm confused by the terminology you use, and how it relates 
>> operationally. The diversion happens on the AOR, and the subscriptions
> 
>> are on the AOR, and the notifications are sent to the subscribers. I 
>> don't see where the registered contacts enter into the picture at all.
>>
>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>
>> If you have something else in mind, I hope you can explain it better 
>> in the draft.
>>
>>> [4] Yes the value of N is configurable and depends on how many 
>>> diversions the subscribe wants to keep or could be dependent on 
>>> server
>>> policy (some maximum
>>>     number). If the server policy is to send notifications as and 
>>> when
>>> they occur and not keep any divsersions information, then the value 
>>> of
>>> last N
>>>     diversions will be zero. But if the server chooses to retain the 
>>> last diversion always, then the value of N will be 1 and so on.
>> All is straightforward for N>=1.
>>
>> For N=0 things are messy. If we are modeling this as state, and 
>> notifications are about state changes, then the new diversion is added
> 
>> to the (previously empty) state and that triggers notifications to all
> 
>> current subscriptions. Then if you don't want to retain even this much
> 
>> state (because N=0) you immediately remove that version from the
> state.
>> But that is again a state change, so you end up with another 
>> notification of that state change.
>>
>> <Ranjit-Feb26-2> True. Actually the notifications are for informing 
>> the subscriber of the call diversions that happen. The notifications 
>> are not for informing the subscriber of state changes in the notifier 
>> or the number of diversions that occurred. As I mentioned in the 
>> abstract as well as 24.604, the main intent of CDIVN service is to 
>> inform the subscribers of communication diversions that occur on their
> 
>> behalf in the network and this information will help them manage their
> 
>> rules better.
> 
> I am not certain, but think we may still be talking past one another to
> some extent.
> 
> The state can be considered just a formalism for defining the service
> clearly, regardless of whether it has intrinsic value to the CDIV
> service. But it *is* a formalism. So if it is introduced, then its
> behavior can't be ignored when inconvenient. Defining the state with N=0
> leads to some undesired behavior. The simplest way around that is to
> insist that N be >=1.
> 
> <Ranjit-Mar2> Ok in that case we can say that value of N >=1 and the
> notifier has always maintains history of last diversion at a minimu. It
> can maintain more than 1 too depending on the policy. 
> 
> If its important to you that no state be retained once a notification
> has been sent about the most recent diversion, then you will have to
> find some other way to specify the service, that still meets the
> expectations of 3265. I don't know if there is anything there that Adam
> would consider valid. We will have to get his opinion on that.
> 
> <Ranjit-Mar2> In that case I am OK maintaining the hostory of diversions
> that occurred as part of state information and retaining that state even
> after the notification for that diversion is sent - so that in case I do
> not get a confirmation, I would be able to re-send the notification.
> Also maintaining state of diversions would help when the notifications
> need to sent only at a particular time (if fikter criteria mentions it).
> 
> 
> 	Thanks,
> 	Paul
> 
>> <Ranjit-Feb26-2> But if we want to include the information related to 
>> how many diversions have happened till time (say from 12 AM ) till the
> 
>> time the notification is being sent, then as you say we can store the 
>> diversions. This would be useful when the subscriber specifies in the 
>> filter criteria to send notifications only during certain time like 
>> between 5 and 6 PM and not whenever a diversion occurs. Then N is 
>> grater than 1.
> 
>> I presume you don't want that behavior. If you can live with N>=1 then
> 
>> we don't have the problem at all. If you need to support N=0 and don't
> 
>> want the extra notify, then we have to figure out if there is some 
>> trick to how things are modeled to get the desired effect. I've 
>> suggested something previously, but Adam didn't like it.
>>
>> This needs sorting out.
>>
>> 	Thanks,
>> 	Paul
> 

From ranjit@motorola.com  Tue Mar  2 08:56:26 2010
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Ok will incorporate all the comments we discussed so far in the updated
version (-04)=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, March 02, 2010 7:19 PM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

I think this all sounds good now.

	Thanks,
	Paul

Avasarala Ranjit-A20990 wrote:
> Inline
>=20
> <Ranjit-Mar2>
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 02, 2010 2:16 AM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
> inline...
>=20
> Avasarala Ranjit-A20990 wrote:
>=20
>> I'm still a little confused here.
>> If phone is on and subscription is active, then in general=20
>> notification should be possible, regardless of registration. I=20
>> realize
>=20
>> IMS doesn't permit that, but from a general standard perspective we=20
>> shouldn't assume it. The manifestation in this case is that the=20
>> NOTIFY
>=20
>> would fail and the subscription would eventually be torn down.
>>
>> So I think the cases are:
>> - there is a subscription and notification works
>> - there is a subscription but notification fails
>> - there is no subscription.
>>
>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as=20
>> Notification fails, we can say - not sending notification due to (1)=20
>> no match of filter criteria (2) phone not registered or outside=20
>> coverage area due to which there is no connectivity and hence=20
>> notification cannot be sent Anyway if no subscription, no
> notification.
>=20
> Are you suggesting that you might have a subscription, and have the=20
> filter criteria met, and yet not attempt to send a NOTIFY because the=20
> phone is not registered or has no network connectivity???
>=20
> That seems wrong. ISTM that either you would remove the subscription,=20
> or else you would try to send the NOTIFY and then discover that it
fails.
>=20
> <Ranjit-Mar2> yes the subscription could be removed when the device is

> not reachable.
>=20
> OTOH, this probably ought not to be within scope of what is discussed=20
> in this draft.
>=20
> <Ranjit-Mar2> yes true not within the scope of this draft.
>=20
>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>> contacts: sip:line1@alice.office.atlanta.com and=20
>>> sip:line2@alice.home.atlanta.com and
>>>     both the phones ring, then it would be a case of parallel
> forking.
>>> Now say one or both the phones have set diversion rules, to divert=20
>>> the
>> In this case, what does it mean for "one or both phones have set=20
>> diversion rules"? Does that somehow imply distinct rules for each
> phone?
>> I don't see how that could work. Aren't the diversion rules for the=20
>> *AOR*?
>>
>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there=20
>> is
>=20
>> a diversion rule set for the AoR and when diversions occur, the=20
>> notifications are Sent to the AoR.
>=20
> Not sent to the AoR - sent to the remote contact of the dialog for=20
> each subscription to the AoR. Same point as below that you agreed
with.
>=20
> <Ranjit-Mar2> Yes agree with u.=20
>=20
>> I see they could be set/changed from either phone. But its a single=20
>> set of rules isn't it? So I might set the rules from the office, and=20
>> then cancel them from home.
>>
>> <Ranjit-Feb26-2> yes single set and could be set from either of the=20
>> phones. True.
>>
>>> call to say=20
>>>     sip:voicemail@alice.atlanta.com, then the notifications would=20
>>> get
>=20
>>> generated with respect to both the contacts.
>> Again I'm confused by the terminology you use, and how it relates=20
>> operationally. The diversion happens on the AOR, and the=20
>> subscriptions
>=20
>> are on the AOR, and the notifications are sent to the subscribers. I=20
>> don't see where the registered contacts enter into the picture at
all.
>>
>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>
>> If you have something else in mind, I hope you can explain it better=20
>> in the draft.
>>
>>> [4] Yes the value of N is configurable and depends on how many=20
>>> diversions the subscribe wants to keep or could be dependent on=20
>>> server policy (some maximum
>>>     number). If the server policy is to send notifications as and=20
>>> when they occur and not keep any divsersions information, then the=20
>>> value of last N
>>>     diversions will be zero. But if the server chooses to retain the

>>> last diversion always, then the value of N will be 1 and so on.
>> All is straightforward for N>=3D1.
>>
>> For N=3D0 things are messy. If we are modeling this as state, and=20
>> notifications are about state changes, then the new diversion is=20
>> added
>=20
>> to the (previously empty) state and that triggers notifications to=20
>> all
>=20
>> current subscriptions. Then if you don't want to retain even this=20
>> much
>=20
>> state (because N=3D0) you immediately remove that version from the
> state.
>> But that is again a state change, so you end up with another=20
>> notification of that state change.
>>
>> <Ranjit-Feb26-2> True. Actually the notifications are for informing=20
>> the subscriber of the call diversions that happen. The notifications=20
>> are not for informing the subscriber of state changes in the notifier

>> or the number of diversions that occurred. As I mentioned in the=20
>> abstract as well as 24.604, the main intent of CDIVN service is to=20
>> inform the subscribers of communication diversions that occur on=20
>> their
>=20
>> behalf in the network and this information will help them manage=20
>> their
>=20
>> rules better.
>=20
> I am not certain, but think we may still be talking past one another=20
> to some extent.
>=20
> The state can be considered just a formalism for defining the service=20
> clearly, regardless of whether it has intrinsic value to the CDIV=20
> service. But it *is* a formalism. So if it is introduced, then its=20
> behavior can't be ignored when inconvenient. Defining the state with=20
> N=3D0 leads to some undesired behavior. The simplest way around that =
is=20
> to insist that N be >=3D1.
>=20
> <Ranjit-Mar2> Ok in that case we can say that value of N >=3D1 and the =

> notifier has always maintains history of last diversion at a minimu.=20
> It can maintain more than 1 too depending on the policy.
>=20
> If its important to you that no state be retained once a notification=20
> has been sent about the most recent diversion, then you will have to=20
> find some other way to specify the service, that still meets the=20
> expectations of 3265. I don't know if there is anything there that=20
> Adam would consider valid. We will have to get his opinion on that.
>=20
> <Ranjit-Mar2> In that case I am OK maintaining the hostory of=20
> diversions that occurred as part of state information and retaining=20
> that state even after the notification for that diversion is sent - so

> that in case I do not get a confirmation, I would be able to re-send
the notification.
> Also maintaining state of diversions would help when the notifications

> need to sent only at a particular time (if fikter criteria mentions
it).
>=20
>=20
> 	Thanks,
> 	Paul
>=20
>> <Ranjit-Feb26-2> But if we want to include the information related to

>> how many diversions have happened till time (say from 12 AM ) till=20
>> the
>=20
>> time the notification is being sent, then as you say we can store the

>> diversions. This would be useful when the subscriber specifies in the

>> filter criteria to send notifications only during certain time like=20
>> between 5 and 6 PM and not whenever a diversion occurs. Then N is=20
>> grater than 1.
>=20
>> I presume you don't want that behavior. If you can live with N>=3D1=20
>> then
>=20
>> we don't have the problem at all. If you need to support N=3D0 and=20
>> don't
>=20
>> want the extra notify, then we have to figure out if there is some=20
>> trick to how things are modeled to get the desired effect. I've=20
>> suggested something previously, but Adam didn't like it.
>>
>> This needs sorting out.
>>
>> 	Thanks,
>> 	Paul
>=20

From dean.willis@softarmor.com  Tue Mar  2 10:36:16 2010
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On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:

> All,
>
> A New Internet-Draft is available from the on-line Internet-Drafts  
> directories.
>
> Title : SIP Egress Route Use in ENUM
> Author(s) : D. Malas
> Filename : draft-malas-dispatch-sip-egress-route-00.txt
> Pages : 11
> Date : 2010-03-01
>

The universe has apparently shifted beneath my feet again. When did  
DNS (and ENUM is just DNS, or maybe a private-root DNS depending on  
what you believe) become a legitimate repository for intermediary  
routing information?


Let's take the scenario from the draft:

    +=====================++                 ++====================+
                          ||                 ||
        SSP1 Network  +--------+             ||       SSP2 Network
    ssp1.example.com  |        |             ||   ssp2.example.com
                 +--->|  SBE1  |---+         ||
                 |    |        |   |         ||
                 |    +--------+   |         ||
                 |        ||       |         ||
                 |        ||       |         ||
     +-------+   |        ||       |     +--------+     +-------+
     |       |---+        ||       +---->|        |---->|       |
     | Proxy |            ||             |  SBE3  |     | Proxy |
     |       |---+        ||       +---->|        |---->|       |
     +-------+   |        ||       |     +--------+     +-------+
                 |        ||       |         ||
                 |        ||       |         ||
                 |    +--------+   |         ||
                 |    |        |   |         ||
                 +--->|  SBE2  |---+         ||
                      |        |             ||
                      +--------+             ||
                          ||                 ||
    +=====================++                 ++====================+


Here's the scenario recast in a "more traditional" terminology:

SSP2 has externally advertised (at least to SSP1) that SBE3 is a (or  
"the") route to the address space of SSP2. Both SBE1 and SBE2 have  
been informed of this advertisement. Presumably there's a route in the  
other direction as well, although that's out of scope for the example.

SSP1 wishes to advertise, within the administrative domain of SSP1,  
two routes to the address space of SSP2. The preferred route is  
through SBE2, with a secondary route through SBE1, although nothing  
has been said about the load-metric so we assume the policy use "Use  
SBE1 only if SBE2 isn't responding".

Explain to me why ANY of this routing/reachability information should  
be in DNS or a DNS-like static map? That's just absurd! We appear to   
going through some extreme convolutions to twist a dynamic routing  
problem (that should be dealt with by protocols more like BGP and  
OSPF) into a name-service problem.

I think it's time for a complete re-think of what's going on here.

ENUM is NOT a routing protocol. It's a distributed database for  
mapping names to addresses and for looking up related information  
about names, where the associations involved are relatively long-lived  
(and for the most part effectively static).

There's a reason we don't have every Internet router do a DNS lookup  
on a request-destination and use that lookup result to select a next- 
hop route. We COULD do this, it just requires having lots and lots of  
DNS servers configured with alt-root structures that reflect local  
routing tables, and that get continuously reprovisioned to reflect  
changes in network topology, load, and policy.  We COULD do this --  
why don't we?  The answer is that doing this would be deeply,  
profoundly, and astoundingly inefficient. It's just the wrong way to  
solve the problem. It's sort of like trying to print highway maps  
that, rather than showing the road to follow to reach a destination,  
only show the next exit or turn and therefore one needs an entirely  
different map for each possible place that one might currently be  
standing. Sure, it makes for a big business in maintaining and  
printing maps, but it makes actually getting anywhere profoundly  
painful and expensive.

So why on earth are we attempting to use the same pattern for routing  
phone calls?  Is it just a matter of ENUM being the only hammer we  
were given to play with? Folks, we can get a different tool if we need  
one; we don't need to keep beating on the plumbing with the ENUM  
hammer when what we really need is a pipe-threader or bender.

--
Dean
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 1, 2010, =
at 5:50 PM, Daryl Malas wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"> <div> =
<div><font size=3D"2" face=3D"Arial"><span =
class=3D"500354723-01032010">All,</span></font></div> <div><font =
size=3D"2" face=3D"Arial"><span =
class=3D"500354723-01032010"></span></font>&nbsp;</div> <div><font =
size=3D"2" face=3D"Arial">A New Internet-Draft is available from the =
on-line Internet-Drafts directories. <br>&nbsp;<br>Title : SIP Egress =
Route Use in ENUM <br>Author(s) : D. Malas <br>Filename : =
draft-malas-dispatch-sip-egress-route-00.txt <br>Pages : 11 <br>Date : =
2010-03-01 =
<br>&nbsp;<br></font></div></div></blockquote></div><br><div>The =
universe has apparently shifted beneath my feet again. When did DNS (and =
ENUM is just DNS, or maybe a private-root DNS depending on what you =
believe) become a legitimate repository for intermediary routing =
information?&nbsp;</div><div><br></div><div><br></div><div>Let's take =
the scenario from the draft:</div><div><br></div><div><span =
class=3D"Apple-style-span" style=3D"font-family: Times; "><pre =
style=3D"word-wrap: break-word; white-space: pre-wrap; ">   =
+=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+
                         ||                 ||
       SSP1 Network  +--------+             ||       SSP2 Network
   ssp1.example.com  |        |             ||   ssp2.example.com
                +---&gt;|  SBE1  |---+         ||
                |    |        |   |         ||
                |    +--------+   |         ||
                |        ||       |         ||
                |        ||       |         ||
    +-------+   |        ||       |     +--------+     +-------+
    |       |---+        ||       +----&gt;|        |----&gt;|       |
    | Proxy |            ||             |  SBE3  |     | Proxy |
    |       |---+        ||       +----&gt;|        |----&gt;|       |
    +-------+   |        ||       |     +--------+     +-------+
                |        ||       |         ||
                |        ||       |         ||
                |    +--------+   |         ||
                |    |        |   |         ||
                +---&gt;|  SBE2  |---+         ||
                     |        |             ||
                     +--------+             ||
                         ||                 ||
   +=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+</pre></spa=
n></div><div><br></div><div><br></div><div>Here's the scenario recast in =
a "more traditional" terminology:</div><div><br></div><div>SSP2 has =
externally advertised (at least to SSP1) that SBE3 is a (or "the") route =
to the address space of SSP2. Both SBE1 and SBE2 have been informed of =
this advertisement. Presumably there's a route in the other direction as =
well, although that's out of scope for the =
example.</div><div><br></div><div>SSP1 wishes to advertise, within the =
administrative domain of SSP1, two routes to the address space of SSP2. =
The preferred route is through SBE2, with a secondary route through =
SBE1, although nothing has been said about the load-metric so we assume =
the policy use "Use SBE1 only if SBE2 isn't =
responding".</div><div><br></div><div>Explain to me why ANY of this =
routing/reachability information should be in DNS or a DNS-like static =
map? That's just absurd! We appear to &nbsp;going through some extreme =
convolutions to twist a dynamic routing problem (that should be dealt =
with by protocols more like BGP and OSPF) into a name-service =
problem.</div><div><br></div><div>I think it's time for a complete =
re-think of what's going on here.</div><div><br></div><div>ENUM is NOT a =
routing protocol. It's a distributed database for mapping names to =
addresses and for looking up related information about names, where the =
associations involved are relatively long-lived (and for the most part =
effectively static).</div><div><br></div><div>There's a reason we don't =
have every Internet router do a DNS lookup on a request-destination and =
use that lookup result to select a next-hop route. We COULD do this, it =
just requires having lots and lots of DNS servers configured with =
alt-root structures that reflect local routing tables, and that get =
continuously reprovisioned to reflect changes in network topology, load, =
and policy. &nbsp;We COULD do this -- why don't we?&nbsp;&nbsp;The =
answer is that doing this would be deeply, profoundly, and astoundingly =
inefficient. It's just the wrong way to solve the problem. It's sort of =
like trying to print highway maps that, rather than showing the road to =
follow to reach a destination, only show the next exit or turn and =
therefore one needs an entirely different map for each possible place =
that one might currently be standing. Sure, it makes for a big business =
in maintaining and printing maps, but it makes actually getting anywhere =
profoundly painful and expensive.</div><div><br></div><div>So why on =
earth are we attempting to use the same pattern for routing phone calls? =
&nbsp;Is it just a matter of ENUM being the only hammer we were given to =
play with? Folks, we can get a different tool if we need one; we don't =
need to keep beating on the plumbing with the ENUM hammer when what we =
really need is a pipe-threader or =
bender.</div><div><br></div><div>--</div><div>Dean</div></body></html>=

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From: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
To: "Daryl Malas" <D.Malas@cablelabs.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Daryl:

I want to be sure I understand what you're proposing.  In what name
server would the modified records be? One local to SSP1 or some general
industry DNS? I'm trying to understand how SSP1 modifies an RR for an
SSP2 number.

=20

Thanks,

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

=20

All,

=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.=20
=20
Title : SIP Egress Route Use in ENUM=20
Author(s) : D. Malas=20
Filename : draft-malas-dispatch-sip-egress-route-00.txt=20
Pages : 11=20
Date : 2010-03-01=20
=20
In some cases a UA (or proxy) within a Session Initiation Protocol=20
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy=20
options to reach an ingress proxy within another SSP to reach the=20
target UA. If the originating SSP uses ENUM to identify the ingress=20
proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
response to identify a specific preferred egress proxy from the set=20
of multiple parallel proxies. This draft solves this problem by=20
defining a method for returning deterministic routing information=20
within an ENUM NAPTR response enabling the UA (or proxy) to choose a=20
preferred egress proxy. Using this information, the originating UA=20
now sends the SIP request through the predetermined preferred egress=20
proxy to reach the target SSP's proxy. This document describes=20
several methods to make this determination by incorporating variables=20
into a SIP service URI contained in the E.164 Number Mapping (ENUM)=20
response.=20
=20
A URL for this Internet-Draft is:=20
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-rout
e-00.txt=20

=20

Regards,

=20

Daryl

=20

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

Daryl Malas

Project Director, IP Interconnects

w - +1 303 661 3302

mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>=20

=20


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Daryl:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I want to be sure I understand what you&#8217;re
proposing.&nbsp; In what name server would the modified records be? One =
local
to SSP1 or some general industry DNS? I&#8217;m trying to understand how =
SSP1
modifies an RR for an SSP2 number.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>Daryl
Malas<br>
<b>Sent:</b> Monday, March 01, 2010 6:51 PM<br>
<b>To:</b> 'dispatch@ietf.org'<br>
<b>Subject:</b> [dispatch] I-D Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>All,</span><o=
:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A
New Internet-Draft is available from the on-line Internet-Drafts =
directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR =
<br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a =
<br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress =
<br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables =
<br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch=
-sip-egress-route-00.txt</a>
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Regards,</spa=
n><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl</span><=
o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>-------------=
--------------</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl
Malas</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Project
Director, IP Interconnects</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>w
- +1 303 661 3302</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><a
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.c=
om</a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

</body>

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From adam@nostrum.com  Tue Mar  2 11:26:05 2010
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On 3/2/10 12:36, Mar 2, Dean Willis wrote:
>
> On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:
>
>> All,
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>>
>> Title : SIP Egress Route Use in ENUM
>> Author(s) : D. Malas
>> Filename : draft-malas-dispatch-sip-egress-route-00.txt
>> Pages : 11
>> Date : 2010-03-01
>>
>
> The universe has apparently shifted beneath my feet again. When did 
> DNS (and ENUM is just DNS, or maybe a private-root DNS depending on 
> what you believe) become a legitimate repository for intermediary 
> routing information?

1992?

http://www.rfc-editor.org/rfc/rfc1383.txt

/a

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On 3/2/10 12:36, Mar 2, Dean Willis wrote:
<blockquote
 cite="mid:564499D5-6303-4727-AD8C-996D182D9726@softarmor.com"
 type="cite"><br>
  <div>
  <div>On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:</div>
  <br class="Apple-interchange-newline">
  <blockquote type="cite">
    <div>
    <div><font face="Arial" size="2"><span class="500354723-01032010">All,</span></font></div>
    <div><font face="Arial" size="2"><span class="500354723-01032010"></span></font>&nbsp;</div>
    <div><font face="Arial" size="2">A New Internet-Draft is available
from the on-line Internet-Drafts directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
    </font></div>
    </div>
  </blockquote>
  </div>
  <br>
  <div>The universe has apparently shifted beneath my feet again. When
did DNS (and ENUM is just DNS, or maybe a private-root DNS depending on
what you believe) become a legitimate repository for intermediary
routing information?</div>
</blockquote>
<br>
1992?<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.rfc-editor.org/rfc/rfc1383.txt">http://www.rfc-editor.org/rfc/rfc1383.txt</a><br>
<br>
/a<br>
</body>
</html>

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On 3/2/10 13:26, Mar 2, Adam Roach wrote:
> On 3/2/10 12:36, Mar 2, Dean Willis wrote:
>>
>> On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:
>>
>>> All,
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories.
>>>
>>> Title : SIP Egress Route Use in ENUM
>>> Author(s) : D. Malas
>>> Filename : draft-malas-dispatch-sip-egress-route-00.txt
>>> Pages : 11
>>> Date : 2010-03-01
>>>
>>
>> The universe has apparently shifted beneath my feet again. When did 
>> DNS (and ENUM is just DNS, or maybe a private-root DNS depending on 
>> what you believe) become a legitimate repository for intermediary 
>> routing information?
>
> 1992?
>
> http://www.rfc-editor.org/rfc/rfc1383.txt

And, really, if you look at the original DNS spec (RFC 882, published in 
1983), the MF record type (as distinct from the MD record type, both 
since subsumed by MX records) talk about the same kinds of information 
as Daryl is proposing here. Semantically: "To get to host B, deliver to 
host A, and indicate the domain of host B in the application data itself."

There's also the use of the "RT" (Route Through) record defined by RFC 
1183, which does exactly this kind of thing. In fact, a careful dusting 
off of RT (accompanied by a BCP for this specific application) would 
probably serve the purpose intended by this draft.

I really see very little historical basis for objecting to storing this 
kind of information in DNS at all.

/a

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On 3/2/10 13:26, Mar 2, Adam Roach wrote:
<blockquote cite="mid:4B8D6649.5080602@nostrum.com" type="cite">
  <meta content="text/html; charset=ISO-8859-1"
 http-equiv="Content-Type">
On 3/2/10 12:36, Mar 2, Dean Willis wrote:
  <blockquote
 cite="mid:564499D5-6303-4727-AD8C-996D182D9726@softarmor.com"
 type="cite"><br>
    <div>
    <div>On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:</div>
    <br class="Apple-interchange-newline">
    <blockquote type="cite">
      <div>
      <div><font face="Arial" size="2"><span class="500354723-01032010">All,</span></font></div>
      <div><font face="Arial" size="2"><span class="500354723-01032010"></span></font>&nbsp;</div>
      <div><font face="Arial" size="2">A New Internet-Draft is
available
from the on-line Internet-Drafts directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
      </font></div>
      </div>
    </blockquote>
    </div>
    <br>
    <div>The universe has apparently shifted beneath my feet again.
When
did DNS (and ENUM is just DNS, or maybe a private-root DNS depending on
what you believe) become a legitimate repository for intermediary
routing information?</div>
  </blockquote>
  <br>
1992?<br>
  <br>
  <a moz-do-not-send="true" class="moz-txt-link-freetext"
 href="http://www.rfc-editor.org/rfc/rfc1383.txt">http://www.rfc-editor.org/rfc/rfc1383.txt</a><br>
</blockquote>
<br>
And, really, if you look at the original DNS spec (RFC 882, published
in 1983), the MF record type (as distinct from the MD record type, both
since subsumed by MX records) talk about the same kinds of information
as Daryl is proposing here. Semantically: "To get to host B, deliver to
host A, and indicate the domain of host B in the application data
itself."<br>
<br>
There's also the use of the "RT" (Route Through) record defined by RFC
1183, which does exactly this kind of thing. In fact, a careful dusting
off of RT (accompanied by a BCP for this specific application) would
probably serve the purpose intended by this draft.<br>
<br>
I really see very little historical basis for objecting to storing this
kind of information in DNS at all.<br>
<br>
/a<br>
</body>
</html>

--------------030907040806070309060703--

From dean.willis@softarmor.com  Tue Mar  2 22:59:24 2010
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On Mar 2, 2010, at 1:26 PM, Adam Roach wrote:

> On 3/2/10 12:36, Mar 2, Dean Willis wrote:
>>
>>
>> On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:
>>
>>> All,
>>>
>>> A New Internet-Draft is available from the on-line Internet-Drafts  
>>> directories.
>>>
>>> Title : SIP Egress Route Use in ENUM
>>> Author(s) : D. Malas
>>> Filename : draft-malas-dispatch-sip-egress-route-00.txt
>>> Pages : 11
>>> Date : 2010-03-01
>>>
>>
>> The universe has apparently shifted beneath my feet again. When did  
>> DNS (and ENUM is just DNS, or maybe a private-root DNS depending on  
>> what you believe) become a legitimate repository for intermediary  
>> routing information?
>
> 1992?
>
> http://www.rfc-editor.org/rfc/rfc1383.txt

That was what we might colloquially  call a "valid experiment that  
successfully produced a result invalidating the hypothesis". Bravely  
done, but perhaps not so functional in the real world.  An  
experimental approach resulting in failure does not qualify the failed  
approach as a legitimate solution.

Do we learn from it, or just blindly do it all over again?

--
Dean


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 2, 2010, =
at 1:26 PM, Adam Roach wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"> <div =
bgcolor=3D"#ffffff" text=3D"#000000"> On 3/2/10 12:36, Mar 2, Dean =
Willis wrote: <blockquote =
cite=3D"mid:564499D5-6303-4727-AD8C-996D182D9726@softarmor.com" =
type=3D"cite"><br>  <div>  <div>On Mar 1, 2010, at 5:50 PM, Daryl Malas =
wrote:</div>  <br class=3D"Apple-interchange-newline">  <blockquote =
type=3D"cite">    <div>    <div><font face=3D"Arial" size=3D"2"><span =
class=3D"500354723-01032010">All,</span></font></div>    <div><font =
face=3D"Arial" size=3D"2"><span =
class=3D"500354723-01032010"></span></font>&nbsp;</div>    <div><font =
face=3D"Arial" size=3D"2">A New Internet-Draft is available from the =
on-line Internet-Drafts directories. <br> &nbsp;<br> Title : SIP Egress =
Route Use in ENUM <br> Author(s) : D. Malas <br> Filename : =
draft-malas-dispatch-sip-egress-route-00.txt <br> Pages : 11 <br> Date : =
2010-03-01 <br> &nbsp;<br>    </font></div>    </div>  </blockquote>  =
</div>  <br>  <div>The universe has apparently shifted beneath my feet =
again. When did DNS (and ENUM is just DNS, or maybe a private-root DNS =
depending on what you believe) become a legitimate repository for =
intermediary routing information?</div> </blockquote> <br> 1992?<br> =
<br> <a class=3D"moz-txt-link-freetext" =
href=3D"http://www.rfc-editor.org/rfc/rfc1383.txt">http://www.rfc-editor.o=
rg/rfc/rfc1383.txt</a><br></div></blockquote></div><br><div>That was =
what we might colloquially &nbsp;call a "valid experiment that =
successfully produced a result invalidating the hypothesis". Bravely =
done, but perhaps not so functional in the real world. &nbsp;An =
experimental approach resulting in failure does not qualify the failed =
approach as a legitimate solution.</div><div><br></div><div>Do we learn =
from it, or just blindly do it all over =
again?</div><div><br></div><div>--</div><div>Dean</div><div><br></div></bo=
dy></html>=

--Apple-Mail-1--617936864--

From dean.willis@softarmor.com  Tue Mar  2 23:05:50 2010
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On Mar 2, 2010, at 1:52 PM, Adam Roach wrote:

>
> And, really, if you look at the original DNS spec (RFC 882,  
> published in 1983), the MF record type (as distinct from the MD  
> record type, both since subsumed by MX records) talk about the same  
> kinds of information as Daryl is proposing here. Semantically: "To  
> get to host B, deliver to host A, and indicate the domain of host B  
> in the application data itself."
>
> There's also the use of the "RT" (Route Through) record defined by  
> RFC 1183, which does exactly this kind of thing. In fact, a careful  
> dusting off of RT (accompanied by a BCP for this specific  
> application) would probably serve the purpose intended by this draft.
>
> I really see very little historical basis for objecting to storing  
> this kind of information in DNS at all.

The unused routing cruft in the DNS is unused for a reason; it doesn't  
make for an operational network or a scaleable DNS. This is also why  
we don't use UUCP-style addressing for phone calls and web traffic.

The applicability of the approach can be summed up in the non-response  
to:

http://www.ops.ietf.org/lists/namedroppers/namedroppers.199x/msg00915.html

But you are right;  RT does almost exactly what the draft in question  
proposes. I wouldn't stop anybody from doing an experimental-track  
draft warming over RT for phone number routing. But I'd probably laugh  
at them.

--
Dean


From gregert@teigre.com  Tue Mar  2 23:16:13 2010
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On 02.03.2010 20:24, PFAUTZ, PENN L (ATTCORP) wrote:
>
> Daryl:
>
> I want to be sure I understand what you're proposing.  In what name 
> server would the modified records be? One local to SSP1 or some 
> general industry DNS? I'm trying to understand how SSP1 modifies an RR 
> for an SSP2 number.
>

A few follow-up questions to that. I'm a bit confused about the problem 
that you are trying to solve.

 From the draft it looks like the name server is local to SSP1. If that 
is correct, why would you like to standardize how you acquire routing 
information? If both the proxy and the name server are local to SSP1, I 
don't see why you need to standardize the data retrieval interface in a 
SIP context?  We already have a few of those.  And there must surely be 
more efficient ways of storing and retrieving such information?!
However, if the intention is primarily to be a standard for UAs to 
implement to avoid a ("static") proxy for determining routing path (i.e. 
a per session outbound proxy), we should surely look at a more generic 
approach to determining the outbound proxy?!  I thought dynamic routing 
was one of the reasons for having an outbound proxy and choosing an 
egress proxy is hardly a problem specific to e.164 addresses.
g-)

> Thanks,
>
> Penn Pfautz
>
> AT&T Access Management
>
> +1-732-420-4962
>
> *From:* dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] 
> *On Behalf Of *Daryl Malas
> *Sent:* Monday, March 01, 2010 6:51 PM
> *To:* 'dispatch@ietf.org'
> *Subject:* [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
>
> All,
>
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
>
> Title : SIP Egress Route Use in ENUM
> Author(s) : D. Malas
> Filename : draft-malas-dispatch-sip-egress-route-00.txt
> Pages : 11
> Date : 2010-03-01
>
> In some cases a UA (or proxy) within a Session Initiation Protocol
> [RFC3261] Service Provider (SSP) has multiple parallel egress proxy
> options to reach an ingress proxy within another SSP to reach the
> target UA. If the originating SSP uses ENUM to identify the ingress
> proxy of the target SSP, there is currently no way for the ENUM NAPTR
> response to identify a specific preferred egress proxy from the set
> of multiple parallel proxies. This draft solves this problem by
> defining a method for returning deterministic routing information
> within an ENUM NAPTR response enabling the UA (or proxy) to choose a
> preferred egress proxy. Using this information, the originating UA
> now sends the SIP request through the predetermined preferred egress
> proxy to reach the target SSP's proxy. This document describes
> several methods to make this determination by incorporating variables
> into a SIP service URI contained in the E.164 Number Mapping (ENUM)
> response.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-00.txt 
>
>
> Regards,
>
> Daryl
>
> ---------------------------
>
> Daryl Malas
>
> Project Director, IP Interconnects
>
> w - +1 303 661 3302
>
> mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


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On 02.03.2010 20:24, PFAUTZ, PENN L (ATTCORP) wrote:
<blockquote
 cite="mid:35FE871E2B085542A35726420E29DA6B0374F1AE@gaalpa1msgusr7a.ugd.att.com"
 type="cite">
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  <o:idmap v:ext="edit" data="1" />
 </o:shapelayout></xml><![endif]-->
  <div class="Section1">
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Daryl:<o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">I
want to be sure I understand what you&#8217;re
proposing.&nbsp; In what name server would the modified records be? One
local
to SSP1 or some general industry DNS? I&#8217;m trying to understand how SSP1
modifies an RR for an SSP2 number.</span></p>
  </div>
</blockquote>
<br>
A few follow-up questions to that. I'm a bit confused about the problem
that you are trying to solve.<br>
<br>
>From the draft it looks like the name server is local to SSP1. If that
is correct, why would you like to standardize how you acquire routing
information? If both the proxy and the name server are local to SSP1, I
don't see why you need to standardize the data retrieval interface in a
SIP context?&nbsp; We already have a few of those.&nbsp; And there must surely be
more efficient ways of storing and retrieving such information?!<br>
However, if the intention is primarily to be a standard for UAs to
implement to avoid a ("static") proxy for determining routing path
(i.e. a per session outbound proxy), we should surely look at a more
generic approach to determining the outbound proxy?!&nbsp; I thought dynamic
routing was one of the reasons for having an outbound proxy and
choosing an egress proxy is hardly a problem specific to e.164
addresses.<br>
g-)<br>
<br>
<blockquote
 cite="mid:35FE871E2B085542A35726420E29DA6B0374F1AE@gaalpa1msgusr7a.ugd.att.com"
 type="cite">
  <div class="Section1">
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Thanks,<o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Penn
Pfautz</span><span style="color: rgb(31, 73, 125);"><o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">AT&amp;T
Access Management</span><span style="color: rgb(31, 73, 125);"><o:p></o:p></span></p>
  </div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">+1-732-420-4962</span><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p></o:p></span></p>
  <div>
  <div
 style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
  <p class="MsoNormal"><b><span
 style="font-size: 10pt; font-family: &quot;Tahoma&quot;,&quot;sans-serif&quot;;">From:</span></b><span
 style="font-size: 10pt; font-family: &quot;Tahoma&quot;,&quot;sans-serif&quot;;">
<a class="moz-txt-link-abbreviated" href="mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.org</a>] <b>On
Behalf Of </b>Daryl
Malas<br>
  <b>Sent:</b> Monday, March 01, 2010 6:51 PM<br>
  <b>To:</b> '<a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a>'<br>
  <b>Subject:</b> [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>
  </div>
  </div>
  <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">All,</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">A
New Internet-Draft is available from the on-line Internet-Drafts
directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR <br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a <br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress <br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables <br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
  <a moz-do-not-send="true"
 href="http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-00.txt</a>
  </span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">Regards,</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">Daryl</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">---------------------------</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">Daryl
Malas</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">Project
Director, IP Interconnects</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;">w
- +1 303 661 3302</span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal"><span
 style="font-size: 10pt; font-family: &quot;Arial&quot;,&quot;sans-serif&quot;;"><a
 moz-do-not-send="true" href="blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com</a></span><o:p></o:p></p>
  </div>
  <div>
  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
  </div>
  </div>
  <pre wrap="">
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</blockquote>
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To: "'PFAUTZ, PENN L (ATTCORP)'" <pp3129@att.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 3 Mar 2010 09:21:52 -0700
Thread-Topic: [dispatch]  I-D Action: draft-malas-dispatch-sip-egress-route-00
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Penn,

The use case for this application is SSP1 (or someone on their behalf) prov=
isions their numbers and relative routes into the LUF associated with their=
 federation.  SSP2 reviews SSP1's "routes" available to them in the LUF bas=
ed on what SSP1 provisioned.  They then add an egress route and associate i=
t with the ingress route SSP1 provisioned.

Let me know if this makes sense.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

Thanks,

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<BODY lang=3DEN-US link=3Dblue vLink=3Dpurple>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Penn,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The use case for this application is SSP1 (or someone=
 on their=20
behalf) provisions their numbers and relative&nbsp;routes into the LUF=20
associated with their federation.&nbsp; SSP2 reviews&nbsp;SSP1's "routes"=20
available to them in the LUF based on what SSP1 provisioned.&nbsp; They the=
n add=20
an egress route and associate it with the ingress route SSP1=20
provisioned.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Let me know if this makes sense.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> PFAUTZ, PENN L (ATTCORP)=20
[mailto:pp3129@att.com] <BR><B>Sent:</B> Tuesday, March 02, 2010 12:24=20
PM<BR><B>To:</B> Daryl Malas; dispatch@ietf.org<BR><B>Subject:</B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Daryl:<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">I=20
want to be sure I understand what you&#8217;re proposing.&nbsp; In what nam=
e server=20
would the modified records be? One local to SSP1 or some general industry D=
NS?=20
I&#8217;m trying to understand how SSP1 modifies an RR for an SSP2=20
number.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Thanks,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">Penn=20
Pfautz</SPAN><SPAN style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">AT&amp;T=20
Access Management</SPAN><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P></DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">+1-732-420-4962</SPAN><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p></o:p></SPAN></P>
<DIV>
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<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B>On Behalf O=
f=20
</B>Daryl Malas<BR><B>Sent:</B> Monday, March 01, 2010 6:51 PM<BR><B>To:</B=
>=20
'dispatch@ietf.org'<BR><B>Subject:</B> [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">All,</SPAN><o:=
p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">A New Internet=
-Draft=20
is available from the on-line Internet-Drafts directories. <BR>&nbsp;<BR>Ti=
tle :=20
SIP Egress Route Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Regards,</SPAN=
><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl</SPAN><o=
:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">--------------=
-------------</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl=20
Malas</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Project Direct=
or, IP=20
Interconnects</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">w - +1 303 661=
=20
3302</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt"><A=20
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com=
</A></SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV></DIV></BODY></HTML>

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From D.Malas@cablelabs.com  Wed Mar  3 08:26:58 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Greger Viken Teigre'" <gregert@teigre.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 3 Mar 2010 09:26:51 -0700
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Greger,

Please see my last email in response to Penn and let me know if it helps.

Regards,

Daryl

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Greger Viken Teigre
Sent: Wednesday, March 03, 2010 12:16 AM
To: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

On 02.03.2010 20:24, PFAUTZ, PENN L (ATTCORP) wrote:
Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

A few follow-up questions to that. I'm a bit confused about the problem tha=
t you are trying to solve.

>From the draft it looks like the name server is local to SSP1. If that is c=
orrect, why would you like to standardize how you acquire routing informati=
on? If both the proxy and the name server are local to SSP1, I don't see wh=
y you need to standardize the data retrieval interface in a SIP context?  W=
e already have a few of those.  And there must surely be more efficient way=
s of storing and retrieving such information?!
However, if the intention is primarily to be a standard for UAs to implemen=
t to avoid a ("static") proxy for determining routing path (i.e. a per sess=
ion outbound proxy), we should surely look at a more generic approach to de=
termining the outbound proxy?!  I thought dynamic routing was one of the re=
asons for having an outbound proxy and choosing an egress proxy is hardly a=
 problem specific to e.164 addresses.
g-)

Thanks,
Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org> [mailto:d=
ispatch-bounces@ietf.org] On Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org<mailto:dispatch@ietf.org>'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
All,
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt
Regards,
Daryl
---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Greger,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Please see my last email in response to Penn and let =
me know=20
if it helps.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D622232616-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Greger Viken=20
Teigre<BR><B>Sent:</B> Wednesday, March 03, 2010 12:16 AM<BR><B>To:</B>=20
dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>On 02.03.2010 20:24, PFAUTZ, PENN L (ATTCORP) wrote:=20
<BLOCKQUOTE=20
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</xml><![endif]--><!--[if gte mso 9]><xml>
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  <o:idmap v:ext=3D"edit" data=3D"1" />
 </o:shapelayout></xml><![endif]-->
  <DIV class=3DSection1>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt">Daryl:<O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt">I=20
  want to be sure I understand what you&#8217;re proposing.&nbsp; In what n=
ame server=20
  would the modified records be? One local to SSP1 or some general industry=
 DNS?=20
  I&#8217;m trying to understand how SSP1 modifies an RR for an SSP2=20
  number.</SPAN></P></DIV></BLOCKQUOTE><BR>A few follow-up questions to tha=
t. I'm=20
a bit confused about the problem that you are trying to solve.<BR><BR>From =
the=20
draft it looks like the name server is local to SSP1. If that is correct, w=
hy=20
would you like to standardize how you acquire routing information? If both =
the=20
proxy and the name server are local to SSP1, I don't see why you need to=20
standardize the data retrieval interface in a SIP context?&nbsp; We already=
 have=20
a few of those.&nbsp; And there must surely be more efficient ways of stori=
ng=20
and retrieving such information?!<BR>However, if the intention is primarily=
 to=20
be a standard for UAs to implement to avoid a ("static") proxy for determin=
ing=20
routing path (i.e. a per session outbound proxy), we should surely look at =
a=20
more generic approach to determining the outbound proxy?!&nbsp; I thought=20
dynamic routing was one of the reasons for having an outbound proxy and cho=
osing=20
an egress proxy is hardly a problem specific to e.164 addresses.<BR>g-)<BR>=
<BR>
<BLOCKQUOTE=20
cite=3Dmid:35FE871E2B085542A35726420E29DA6B0374F1AE@gaalpa1msgusr7a.ugd.att=
.com=20
type=3D"cite">
  <DIV class=3DSection1>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt"><O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt"><O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt">Thanks,<O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt"><O:P></O:P></SPAN></P>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: rgb(31,73,125); FONT-S=
IZE: 10pt">Penn=20
  Pfautz</SPAN><SPAN style=3D"COLOR: rgb(31,73,125)"><O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: rgb(31,73,125); FONT-S=
IZE: 10pt">AT&amp;T=20
  Access Management</SPAN><SPAN=20
  style=3D"COLOR: rgb(31,73,125)"><O:P></O:P></SPAN></P></DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: rgb(31,73,125); FONT-S=
IZE: 10pt">+1-732-420-4962</SPAN><SPAN=20
  style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: rgb(31,73,125); FONT=
-SIZE: 11pt"><O:P></O:P></SPAN></P>
  <DIV>
  <DIV=20
  style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BO=
TTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: rgb(181,196,2=
23) 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
  <P class=3DMsoNormal><B><SPAN=20
  style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN=
></B><SPAN=20
  style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt"> <A=20
  class=3Dmoz-txt-link-abbreviated=20
  href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounces@ietf.org</A> [=
<A=20
  class=3Dmoz-txt-link-freetext=20
  href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounces@ietf.or=
g</A>]=20
  <B>On Behalf Of </B>Daryl Malas<BR><B>Sent:</B> Monday, March 01, 2010 6:=
51=20
  PM<BR><B>To:</B> '<A class=3Dmoz-txt-link-abbreviated=20
  href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</A>'<BR><B>Subject:</=
B>=20
  [dispatch] I-D Action:=20
  draft-malas-dispatch-sip-egress-route-00<O:P></O:P></SPAN></P></DIV></DIV=
>
  <P class=3DMsoNormal><O:P></O:P></P>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">All,</SPAN><=
O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">A New=20
  Internet-Draft is available from the on-line Internet-Drafts directories.=
=20
  <BR>&nbsp;<BR>Title : SIP Egress Route Use in ENUM <BR>Author(s) : D. Mal=
as=20
  <BR>Filename : draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 1=
1=20
  <BR>Date : 2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within =
a=20
  Session Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has mult=
iple=20
  parallel egress proxy <BR>options to reach an ingress proxy within anothe=
r SSP=20
  to reach the <BR>target UA. If the originating SSP uses ENUM to identify =
the=20
  ingress <BR>proxy of the target SSP, there is currently no way for the EN=
UM=20
  NAPTR <BR>response to identify a specific preferred egress proxy from the=
 set=20
  <BR>of multiple parallel proxies. This draft solves this problem by=20
  <BR>defining a method for returning deterministic routing information=20
  <BR>within an ENUM NAPTR response enabling the UA (or proxy) to choose a=
=20
  <BR>preferred egress proxy. Using this information, the originating UA <B=
R>now=20
  sends the SIP request through the predetermined preferred egress <BR>prox=
y to=20
  reach the target SSP's proxy. This document describes <BR>several methods=
 to=20
  make this determination by incorporating variables <BR>into a SIP service=
 URI=20
  contained in the E.164 Number Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>=
A URL=20
  for this Internet-Draft is: <BR><A=20
  href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt"=20
  moz-do-not-send=3D"true">http://www.ietf.org/internet-drafts/draft-malas-=
dispatch-sip-egress-route-00.txt</A>=20
  </SPAN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Regards,</SP=
AN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl</SPAN>=
<O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">------------=
---------------</SPAN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl=20
  Malas</SPAN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Project Dire=
ctor,=20
  IP Interconnects</SPAN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">w - +1 303 6=
61=20
  3302</SPAN><O:P></O:P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt"><A=20
  href=3D"blocked::mailto:d.malas@cablelabs.com"=20
  moz-do-not-send=3D"true">mailto:d.malas@cablelabs.com</A></SPAN><O:P></O:=
P></P></DIV>
  <DIV>
  <P class=3DMsoNormal><O:P></O:P></P></DIV></DIV><PRE wrap=3D""><FIELDSET =
class=3DmimeAttachmentHeader></FIELDSET>
_______________________________________________
dispatch mailing list
<A class=3Dmoz-txt-link-abbreviated href=3D"mailto:dispatch@ietf.org">dispa=
tch@ietf.org</A>
<A class=3Dmoz-txt-link-freetext href=3D"https://www.ietf.org/mailman/listi=
nfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</A>
  </PRE></BLOCKQUOTE><BR></BODY></HTML>

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From D.Malas@cablelabs.com  Wed Mar  3 08:31:42 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'PFAUTZ, PENN L (ATTCORP)'" <pp3129@att.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 3 Mar 2010 09:31:34 -0700
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Penn,

Just to continue my thought a bit further, the UI associated with adding an=
d managing the "egress route" is out of scope of this draft.  The purpose o=
f this draft is to define a standard approach for first, defining what the =
egress route should look like and then how it should be conveyed in the NAP=
TR.

Regards,

Daryl

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Wednesday, March 03, 2010 9:22 AM
To: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

The use case for this application is SSP1 (or someone on their behalf) prov=
isions their numbers and relative routes into the LUF associated with their=
 federation.  SSP2 reviews SSP1's "routes" available to them in the LUF bas=
ed on what SSP1 provisioned.  They then add an egress route and associate i=
t with the ingress route SSP1 provisioned.

Let me know if this makes sense.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

Thanks,

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
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class=3D765003016-03032010>Penn,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010>Just to continue my thought a bit further, the U=
I=20
associated with adding and managing the "egress route" is out of scope of t=
his=20
draft.&nbsp; The purpose of this draft is to define a standard approach for=
=20
first, defining what the egress route should look like and then how it shou=
ld be=20
conveyed in the NAPTR.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D765003016-03032010>Daryl</SPAN></FONT></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Daryl=20
Malas<BR><B>Sent:</B> Wednesday, March 03, 2010 9:22 AM<BR><B>To:</B> 'PFAU=
TZ,=20
PENN L (ATTCORP)'; dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] I-D=
=20
Action: draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Penn,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The use case for this application is SSP1 (or someone=
 on their=20
behalf) provisions their numbers and relative&nbsp;routes into the LUF=20
associated with their federation.&nbsp; SSP2 reviews&nbsp;SSP1's "routes"=20
available to them in the LUF based on what SSP1 provisioned.&nbsp; They the=
n add=20
an egress route and associate it with the ingress route SSP1=20
provisioned.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Let me know if this makes sense.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D405150316-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> PFAUTZ, PENN L (ATTCORP)=20
[mailto:pp3129@att.com] <BR><B>Sent:</B> Tuesday, March 02, 2010 12:24=20
PM<BR><B>To:</B> Daryl Malas; dispatch@ietf.org<BR><B>Subject:</B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Daryl:<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">I=20
want to be sure I understand what you&#8217;re proposing.&nbsp; In what nam=
e server=20
would the modified records be? One local to SSP1 or some general industry D=
NS?=20
I&#8217;m trying to understand how SSP1 modifies an RR for an SSP2=20
number.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Thanks,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">Penn=20
Pfautz</SPAN><SPAN style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">AT&amp;T=20
Access Management</SPAN><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P></DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">+1-732-420-4962</SPAN><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p></o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B>On Behalf O=
f=20
</B>Daryl Malas<BR><B>Sent:</B> Monday, March 01, 2010 6:51 PM<BR><B>To:</B=
>=20
'dispatch@ietf.org'<BR><B>Subject:</B> [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">All,</SPAN><o:=
p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">A New Internet=
-Draft=20
is available from the on-line Internet-Drafts directories. <BR>&nbsp;<BR>Ti=
tle :=20
SIP Egress Route Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Regards,</SPAN=
><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl</SPAN><o=
:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">--------------=
-------------</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl=20
Malas</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Project Direct=
or, IP=20
Interconnects</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">w - +1 303 661=
=20
3302</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt"><A=20
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com=
</A></SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV></DIV></BODY></HTML>

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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On 3/3/10 10:31 AM, Daryl Malas wrote:
> Just to continue my thought a bit further, the UI associated with 
> adding and managing the "egress route" is out of scope of this draft.  
> The purpose of this draft is to define a standard approach for first, 
> defining what the egress route should look like and then how it should 
> be conveyed in the NAPTR.
>

And, to be clear, what I read in this document (if we discard the third 
alternative that it discusses but clearly does not favor) is of the 
nature of a BCP, not a standard. The first two alternatives don't 
actually change existing protocols or propose any new protocols -- they 
show how existing protocols can be used to achieve the stated goal.

It looks like more of a clever, "hey, guys, I figured out a way to 
scratch an itch I had, and here's how I did it" kind of thing. I believe 
it would be useful to publish in that context.

/a

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On 3/3/10 10:31 AM, Daryl Malas wrote:
<blockquote
 cite="mid:114DAD31379DFA438C0A2E39B3B8AF5D01213F66B5@srvxchg"
 type="cite">
  <div dir="ltr" align="left"><font color="#0000ff" face="Arial"
 size="2"><span class="765003016-03032010">Just to continue my thought
a bit further, the UI associated with adding and managing the "egress
route" is out of scope of this draft.&nbsp; The purpose of this draft is to
define a standard approach for first, defining what the egress route
should look like and then how it should be conveyed in the NAPTR.</span></font></div>
  <br>
</blockquote>
<br>
And, to be clear, what I read in this document (if we discard the third
alternative that it discusses but clearly does not favor) is of the
nature of a BCP, not a standard. The first two alternatives don't
actually change existing protocols or propose any new protocols -- they
show how existing protocols can be used to achieve the stated goal.<br>
<br>
It looks like more of a clever, "hey, guys, I figured out a way to
scratch an itch I had, and here's how I did it" kind of thing. I
believe it would be useful to publish in that context.<br>
<br>
/a<br>
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From D.Malas@cablelabs.com  Wed Mar  3 08:48:32 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Date: Wed, 3 Mar 2010 09:48:30 -0700
Thread-Topic: [dispatch]  I-D Action: draft-malas-dispatch-sip-egress-route-00
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Dean,

I don't agree with your map analogy.  Every hop along the way in the draft =
example does not require a new look-up.  The single ENUM response provides =
all of the information necessary to reach the target.  It just includes "ne=
xt hop" information to provide more intelligent routing.  The final target =
information is still contained in the single response. Perhaps the reason E=
NUM may be a little different is based on the application associated with t=
he NAPTR.  SIP is an intelligent application with the ability to receive de=
tailed information to reach the target UA.  The draft just leverages the SI=
P service NAPTR to provide the SIP application with more of this detailed i=
nformation.  This way information does not need to be stored and managed in=
 many different locations.

Regards,

Daryl

________________________________
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: Tuesday, March 02, 2010 11:36 AM
To: Daryl Malas
Cc: 'dispatch@ietf.org'
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01


The universe has apparently shifted beneath my feet again. When did DNS (an=
d ENUM is just DNS, or maybe a private-root DNS depending on what you belie=
ve) become a legitimate repository for intermediary routing information?


Let's take the scenario from the draft:


   +=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+
                         ||                 ||
       SSP1 Network  +--------+             ||       SSP2 Network
   ssp1.example.com  |        |             ||   ssp2.example.com
                +--->|  SBE1  |---+         ||
                |    |        |   |         ||
                |    +--------+   |         ||
                |        ||       |         ||
                |        ||       |         ||
    +-------+   |        ||       |     +--------+     +-------+
    |       |---+        ||       +---->|        |---->|       |
    | Proxy |            ||             |  SBE3  |     | Proxy |
    |       |---+        ||       +---->|        |---->|       |
    +-------+   |        ||       |     +--------+     +-------+
                |        ||       |         ||
                |        ||       |         ||
                |    +--------+   |         ||
                |    |        |   |         ||
                +--->|  SBE2  |---+         ||
                     |        |             ||
                     +--------+             ||
                         ||                 ||
   +=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+


Here's the scenario recast in a "more traditional" terminology:

SSP2 has externally advertised (at least to SSP1) that SBE3 is a (or "the")=
 route to the address space of SSP2. Both SBE1 and SBE2 have been informed =
of this advertisement. Presumably there's a route in the other direction as=
 well, although that's out of scope for the example.

SSP1 wishes to advertise, within the administrative domain of SSP1, two rou=
tes to the address space of SSP2. The preferred route is through SBE2, with=
 a secondary route through SBE1, although nothing has been said about the l=
oad-metric so we assume the policy use "Use SBE1 only if SBE2 isn't respond=
ing".

Explain to me why ANY of this routing/reachability information should be in=
 DNS or a DNS-like static map? That's just absurd! We appear to  going thro=
ugh some extreme convolutions to twist a dynamic routing problem (that shou=
ld be dealt with by protocols more like BGP and OSPF) into a name-service p=
roblem.

I think it's time for a complete re-think of what's going on here.

ENUM is NOT a routing protocol. It's a distributed database for mapping nam=
es to addresses and for looking up related information about names, where t=
he associations involved are relatively long-lived (and for the most part e=
ffectively static).

There's a reason we don't have every Internet router do a DNS lookup on a r=
equest-destination and use that lookup result to select a next-hop route. W=
e COULD do this, it just requires having lots and lots of DNS servers confi=
gured with alt-root structures that reflect local routing tables, and that =
get continuously reprovisioned to reflect changes in network topology, load=
, and policy.  We COULD do this -- why don't we?  The answer is that doing =
this would be deeply, profoundly, and astoundingly inefficient. It's just t=
he wrong way to solve the problem. It's sort of like trying to print highwa=
y maps that, rather than showing the road to follow to reach a destination,=
 only show the next exit or turn and therefore one needs an entirely differ=
ent map for each possible place that one might currently be standing. Sure,=
 it makes for a big business in maintaining and printing maps, but it makes=
 actually getting anywhere profoundly painful and expensive.

So why on earth are we attempting to use the same pattern for routing phone=
 calls?  Is it just a matter of ENUM being the only hammer we were given to=
 play with? Folks, we can get a different tool if we need one; we don't nee=
d to keep beating on the plumbing with the ENUM hammer when what we really =
need is a pipe-threader or bender.

--
Dean

--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66B8srvxchg_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18876"></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; -webkit-nbsp-mode: space; -webkit-line-brea=
k: after-white-space">
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010>Dean,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010>I don't agree with your map analogy.&nbsp; Every=
 hop=20
along the way in the draft example does not require a new look-up.&nbsp; Th=
e=20
single ENUM response provides all of the information necessary to reach the=
=20
target.&nbsp; It just includes "next hop" information to provide more=20
intelligent routing.&nbsp; The final target information is still contained =
in=20
the single response. Perhaps the reason ENUM may be a little different is b=
ased=20
on the application associated with the NAPTR.&nbsp; SIP is an intelligent=20
application with the ability to receive detailed information to reach the t=
arget=20
UA.&nbsp; The draft just leverages the SIP service NAPTR to provide the SIP=
=20
application with more of this detailed information.&nbsp; This way informat=
ion=20
does not need to be stored and managed in many different=20
locations.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D331113416-03032010>Daryl</SPAN></FONT></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Dean Willis=20
[mailto:dean.willis@softarmor.com] <BR><B>Sent:</B> Tuesday, March 02, 2010=
=20
11:36 AM<BR><B>To:</B> Daryl Malas<BR><B>Cc:</B>=20
'dispatch@ietf.org'<BR><B>Subject:</B> Re: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV><BR>
<DIV>
<DIV>On Mar 1, 2010, at 5:50 PM, Daryl Malas wrote:</DIV><BR=20
class=3DApple-interchange-newline>
<BLOCKQUOTE type=3D"cite">
  <DIV>
  <DIV><FONT size=3D2 face=3DArial><SPAN=20
  class=3D500354723-01032010>All,</SPAN></FONT></DIV>
  <DIV><FONT size=3D2 face=3DArial><SPAN=20
  class=3D500354723-01032010></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2 face=3DArial>A New Internet-Draft is available from t=
he=20
  on-line Internet-Drafts directories. <BR>&nbsp;<BR>Title : SIP Egress Rou=
te=20
  Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
  draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
  2010-03-01 <BR>&nbsp;<BR></FONT></DIV></DIV></BLOCKQUOTE></DIV><BR>
<DIV>The universe has apparently shifted beneath my feet again. When did DN=
S=20
(and ENUM is just DNS, or maybe a private-root DNS depending on what you=20
believe) become a legitimate repository for intermediary routing=20
information?&nbsp;</DIV>
<DIV><BR></DIV>
<DIV><BR></DIV>
<DIV>Let's take the scenario from the draft:</DIV>
<DIV><BR></DIV>
<DIV><SPAN style=3D"FONT-FAMILY: Times" class=3DApple-style-span><PRE style=
=3D"WORD-WRAP: break-word; WHITE-SPACE: pre-wrap">   +=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+
                         ||                 ||
       SSP1 Network  +--------+             ||       SSP2 Network
   ssp1.example.com  |        |             ||   ssp2.example.com
                +---&gt;|  SBE1  |---+         ||
                |    |        |   |         ||
                |    +--------+   |         ||
                |        ||       |         ||
                |        ||       |         ||
    +-------+   |        ||       |     +--------+     +-------+
    |       |---+        ||       +----&gt;|        |----&gt;|       |
    | Proxy |            ||             |  SBE3  |     | Proxy |
    |       |---+        ||       +----&gt;|        |----&gt;|       |
    +-------+   |        ||       |     +--------+     +-------+
                |        ||       |         ||
                |        ||       |         ||
                |    +--------+   |         ||
                |    |        |   |         ||
                +---&gt;|  SBE2  |---+         ||
                     |        |             ||
                     +--------+             ||
                         ||                 ||
   +=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+<=
/PRE></SPAN></DIV>
<DIV><BR></DIV>
<DIV><BR></DIV>
<DIV>Here's the scenario recast in a "more traditional" terminology:</DIV>
<DIV><BR></DIV>
<DIV>SSP2 has externally advertised (at least to SSP1) that SBE3 is a (or "=
the")=20
route to the address space of SSP2. Both SBE1 and SBE2 have been informed o=
f=20
this advertisement. Presumably there's a route in the other direction as we=
ll,=20
although that's out of scope for the example.</DIV>
<DIV><BR></DIV>
<DIV>SSP1 wishes to advertise, within the administrative domain of SSP1, tw=
o=20
routes to the address space of SSP2. The preferred route is through SBE2, w=
ith a=20
secondary route through SBE1, although nothing has been said about the=20
load-metric so we assume the policy use "Use SBE1 only if SBE2 isn't=20
responding".</DIV>
<DIV><BR></DIV>
<DIV>Explain to me why ANY of this routing/reachability information should =
be in=20
DNS or a DNS-like static map? That's just absurd! We appear to &nbsp;going=
=20
through some extreme convolutions to twist a dynamic routing problem (that=
=20
should be dealt with by protocols more like BGP and OSPF) into a name-servi=
ce=20
problem.</DIV>
<DIV><BR></DIV>
<DIV>I think it's time for a complete re-think of what's going on here.</DI=
V>
<DIV><BR></DIV>
<DIV>ENUM is NOT a routing protocol. It's a distributed database for mappin=
g=20
names to addresses and for looking up related information about names, wher=
e the=20
associations involved are relatively long-lived (and for the most part=20
effectively static).</DIV>
<DIV><BR></DIV>
<DIV>There's a reason we don't have every Internet router do a DNS lookup o=
n a=20
request-destination and use that lookup result to select a next-hop route. =
We=20
COULD do this, it just requires having lots and lots of DNS servers configu=
red=20
with alt-root structures that reflect local routing tables, and that get=20
continuously reprovisioned to reflect changes in network topology, load, an=
d=20
policy. &nbsp;We COULD do this -- why don't we?&nbsp;&nbsp;The answer is th=
at=20
doing this would be deeply, profoundly, and astoundingly inefficient. It's =
just=20
the wrong way to solve the problem. It's sort of like trying to print highw=
ay=20
maps that, rather than showing the road to follow to reach a destination, o=
nly=20
show the next exit or turn and therefore one needs an entirely different ma=
p for=20
each possible place that one might currently be standing. Sure, it makes fo=
r a=20
big business in maintaining and printing maps, but it makes actually gettin=
g=20
anywhere profoundly painful and expensive.</DIV>
<DIV><BR></DIV>
<DIV>So why on earth are we attempting to use the same pattern for routing =
phone=20
calls? &nbsp;Is it just a matter of ENUM being the only hammer we were give=
n to=20
play with? Folks, we can get a different tool if we need one; we don't need=
 to=20
keep beating on the plumbing with the ENUM hammer when what we really need =
is a=20
pipe-threader or bender.</DIV>
<DIV><BR></DIV>
<DIV>--</DIV>
<DIV>Dean</DIV></BODY></HTML>

--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66B8srvxchg_--

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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Adam Roach'" <adam@nostrum.com>
Date: Wed, 3 Mar 2010 09:51:11 -0700
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Adam,

You "hit the nail on the head."  This is the exact intension of this draft.

--Daryl

________________________________
From: Adam Roach [mailto:adam@nostrum.com]
Sent: Wednesday, March 03, 2010 9:41 AM
To: Daryl Malas
Cc: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

On 3/3/10 10:31 AM, Daryl Malas wrote:
Just to continue my thought a bit further, the UI associated with adding an=
d managing the "egress route" is out of scope of this draft.  The purpose o=
f this draft is to define a standard approach for first, defining what the =
egress route should look like and then how it should be conveyed in the NAP=
TR.


And, to be clear, what I read in this document (if we discard the third alt=
ernative that it discusses but clearly does not favor) is of the nature of =
a BCP, not a standard. The first two alternatives don't actually change exi=
sting protocols or propose any new protocols -- they show how existing prot=
ocols can be used to achieve the stated goal.

It looks like more of a clever, "hey, guys, I figured out a way to scratch =
an itch I had, and here's how I did it" kind of thing. I believe it would b=
e useful to publish in that context.

/a

--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66B9srvxchg_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18876"></HEAD>
<BODY bgColor=3D#ffffff text=3D#000000>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523175016-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Adam,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523175016-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523175016-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>You "hit the nail on the head."&nbsp; This is the exa=
ct=20
intension of this draft.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523175016-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523175016-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>--Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Adam Roach [mailto:adam@nostrum.c=
om]=20
<BR><B>Sent:</B> Wednesday, March 03, 2010 9:41 AM<BR><B>To:</B> Daryl=20
Malas<BR><B>Cc:</B> 'PFAUTZ, PENN L (ATTCORP)';=20
dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>On 3/3/10 10:31 AM, Daryl Malas wrote:=20
<BLOCKQUOTE cite=3Dmid:114DAD31379DFA438C0A2E39B3B8AF5D01213F66B5@srvxchg=20
type=3D"cite">
  <DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><=
SPAN=20
  class=3D765003016-03032010>Just to continue my thought a bit further, the=
 UI=20
  associated with adding and managing the "egress route" is out of scope of=
 this=20
  draft.&nbsp; The purpose of this draft is to define a standard approach f=
or=20
  first, defining what the egress route should look like and then how it sh=
ould=20
  be conveyed in the NAPTR.</SPAN></FONT></DIV><BR></BLOCKQUOTE><BR>And, to=
 be=20
clear, what I read in this document (if we discard the third alternative th=
at it=20
discusses but clearly does not favor) is of the nature of a BCP, not a stan=
dard.=20
The first two alternatives don't actually change existing protocols or prop=
ose=20
any new protocols -- they show how existing protocols can be used to achiev=
e the=20
stated goal.<BR><BR>It looks like more of a clever, "hey, guys, I figured o=
ut a=20
way to scratch an itch I had, and here's how I did it" kind of thing. I bel=
ieve=20
it would be useful to publish in that context.<BR><BR>/a<BR></BODY></HTML>

--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66B9srvxchg_--

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Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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From: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
To: "Daryl Malas" <D.Malas@cablelabs.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Daryl=20

I was a little confused by this since I thought the call was from SSP1
to SSP2 and that SSP2 provisioned its numbers into the registry. If SSP1
get the SSP2 data and wants to add their egress routes in their local
copy I can see that but I wouldn't expect the resulting elaborated
NAPTRs to be in the Registry.=20

Have I got this right now?

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

The use case for this application is SSP1 (or someone on their behalf)
provisions their numbers and relative routes into the LUF associated
with their federation.  SSP2 reviews SSP1's "routes" available to them
in the LUF based on what SSP1 provisioned.  They then add an egress
route and associate it with the ingress route SSP1 provisioned.

=20

Let me know if this makes sense.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl:

I want to be sure I understand what you're proposing.  In what name
server would the modified records be? One local to SSP1 or some general
industry DNS? I'm trying to understand how SSP1 modifies an RR for an
SSP2 number.

=20

Thanks,

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

=20

All,

=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.=20
=20
Title : SIP Egress Route Use in ENUM=20
Author(s) : D. Malas=20
Filename : draft-malas-dispatch-sip-egress-route-00.txt=20
Pages : 11=20
Date : 2010-03-01=20
=20
In some cases a UA (or proxy) within a Session Initiation Protocol=20
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy=20
options to reach an ingress proxy within another SSP to reach the=20
target UA. If the originating SSP uses ENUM to identify the ingress=20
proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
response to identify a specific preferred egress proxy from the set=20
of multiple parallel proxies. This draft solves this problem by=20
defining a method for returning deterministic routing information=20
within an ENUM NAPTR response enabling the UA (or proxy) to choose a=20
preferred egress proxy. Using this information, the originating UA=20
now sends the SIP request through the predetermined preferred egress=20
proxy to reach the target SSP's proxy. This document describes=20
several methods to make this determination by incorporating variables=20
into a SIP service URI contained in the E.164 Number Mapping (ENUM)=20
response.=20
=20
A URL for this Internet-Draft is:=20
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-rout
e-00.txt=20

=20

Regards,

=20

Daryl

=20

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

Daryl Malas

Project Director, IP Interconnects

w - +1 303 661 3302

mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>=20

=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Daryl <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I was a little confused by this since I thought the call =
was
from SSP1 to SSP2 and that SSP2 provisioned its numbers into the =
registry. If
SSP1 get the SSP2 data and wants to add their egress routes in their =
local copy
I can see that but I wouldn&#8217;t expect the resulting elaborated =
NAPTRs to
be in the Registry. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Have I got this right now?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b>Sent:</b> Wednesday, March 03, 2010 11:22 AM<br>
<b>To:</b> PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Penn,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The use case for this application is SSP1 (or someone on =
their
behalf) provisions their numbers and relative&nbsp;routes into the LUF
associated with their federation.&nbsp; SSP2 reviews&nbsp;SSP1's =
&quot;routes&quot;
available to them in the LUF based on what SSP1 provisioned.&nbsp; They =
then
add an egress route and associate it with the ingress route SSP1 =
provisioned.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Let me know if this makes sense.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Daryl</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> PFAUTZ, PENN L (ATTCORP)
[mailto:pp3129@att.com] <br>
<b>Sent:</b> Tuesday, March 02, 2010 12:24 PM<br>
<b>To:</b> Daryl Malas; dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Daryl:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I want to be sure I understand what you&#8217;re =
proposing.&nbsp;
In what name server would the modified records be? One local to SSP1 or =
some
general industry DNS? I&#8217;m trying to understand how SSP1 modifies =
an RR
for an SSP2 number.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b>On Behalf Of </b>Daryl Malas<br>
<b>Sent:</b> Monday, March 01, 2010 6:51 PM<br>
<b>To:</b> 'dispatch@ietf.org'<br>
<b>Subject:</b> [dispatch] I-D Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>All,</span><o=
:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A
New Internet-Draft is available from the on-line Internet-Drafts =
directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR =
<br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a =
<br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress =
<br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables =
<br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch=
-sip-egress-route-00.txt</a>
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Regards,</spa=
n><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl</span><=
o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>-------------=
--------------</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl
Malas</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Project
Director, IP Interconnects</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>w
- +1 303 661 3302</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><a
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.c=
om</a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

</body>

</html>

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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'PFAUTZ, PENN L (ATTCORP)'" <pp3129@att.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 3 Mar 2010 11:01:04 -0700
Thread-Topic: [dispatch]  I-D Action: draft-malas-dispatch-sip-egress-route-00
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Penn,

I think what you're describing is leaning towards implementation specific q=
uestions.  I apologize, my example below is backwards, but the principle is=
 the same.  SSP2 provisions their numbers and "routes" into the ENUM server=
.  SSP1 can see the ingress routes for SSP2 numbers.  SSP1 can then specify=
 their own egress routes in the same ENUM server.

The method for SSP1's egress routes for SSP2 to be received into a "local c=
opy" is implementation specific, imo.  This being said, obviously, SSP3 and=
 SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 whe=
n they query the ENUM server.  If this is resolved in some "for queries fro=
m here give this" or "here is a local copy of peer numbers and routes plus =
*your* egress routes" is implementation specific.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Daryl
I was a little confused by this since I thought the call was from SSP1 to S=
SP2 and that SSP2 provisioned its numbers into the registry. If SSP1 get th=
e SSP2 data and wants to add their egress routes in their local copy I can =
see that but I wouldn't expect the resulting elaborated NAPTRs to be in the=
 Registry.
Have I got this right now?

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

The use case for this application is SSP1 (or someone on their behalf) prov=
isions their numbers and relative routes into the LUF associated with their=
 federation.  SSP2 reviews SSP1's "routes" available to them in the LUF bas=
ed on what SSP1 provisioned.  They then add an egress route and associate i=
t with the ingress route SSP1 provisioned.

Let me know if this makes sense.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

Thanks,

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Penn,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>I think what you're describing is leaning towards=20
implementation specific questions.&nbsp; I apologize, my example below is=20
backwards, but the principle is the same.&nbsp; SSP2 provisions their numbe=
rs=20
and "routes" into the ENUM server.&nbsp; SSP1 can see the ingress routes fo=
r=20
SSP2 numbers.&nbsp; SSP1 can then specify their own egress routes in the sa=
me=20
ENUM server.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The method for&nbsp;SSP1's egress routes for SSP2&nbs=
p;to=20
be&nbsp;received into a "local copy" is implementation specific, imo.&nbsp;=
 This=20
being said, obviously, SSP3 and SSP4 would not (want to) see SSP1's provisi=
oned=20
egress routes for SSP2 when they query the ENUM server.&nbsp; If this is=20
resolved in some "for queries from here give this" or "here is a local copy=
 of=20
peer numbers and routes plus *your* egress routes" is implementation=20
specific.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D483004417-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> PFAUTZ, PENN L (ATTCORP)=20
[mailto:pp3129@att.com] <BR><B>Sent:</B> Wednesday, March 03, 2010 10:38=20
AM<BR><B>To:</B> Daryl Malas; dispatch@ietf.org<BR><B>Subject:</B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Daryl=20
<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">I=20
was a little confused by this since I thought the call was from SSP1 to SSP=
2 and=20
that SSP2 provisioned its numbers into the registry. If SSP1 get the SSP2 d=
ata=20
and wants to add their egress routes in their local copy I can see that but=
 I=20
wouldn&#8217;t expect the resulting elaborated NAPTRs to be in the Registry=
.=20
<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Have=20
I got this right now?<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">Penn=20
Pfautz</SPAN><SPAN style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">AT&amp;T=20
Access Management</SPAN><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P></DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">+1-732-420-4962</SPAN><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p></o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt"> Daryl Malas=
=20
[mailto:D.Malas@cablelabs.com] <BR><B>Sent:</B> Wednesday, March 03, 2010 1=
1:22=20
AM<BR><B>To:</B> PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org<BR><B>Subject:=
</B>=20
RE: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: blue; FONT-SIZE: 10pt">P=
enn,</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: blue; FONT-SIZE: 10pt">T=
he use=20
case for this application is SSP1 (or someone on their behalf) provisions t=
heir=20
numbers and relative&nbsp;routes into the LUF associated with their=20
federation.&nbsp; SSP2 reviews&nbsp;SSP1's "routes" available to them in th=
e LUF=20
based on what SSP1 provisioned.&nbsp; They then add an egress route and=20
associate it with the ingress route SSP1 provisioned.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: blue; FONT-SIZE: 10pt">L=
et me=20
know if this makes sense.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: blue; FONT-SIZE: 10pt">R=
egards,</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: blue; FONT-SIZE: 10pt">D=
aryl</SPAN><o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter>
<HR align=3Dcenter SIZE=3D2 width=3D"100%">
</DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt"> PFAUTZ, PENN=
 L=20
(ATTCORP) [mailto:pp3129@att.com] <BR><B>Sent:</B> Tuesday, March 02, 2010 =
12:24=20
PM<BR><B>To:</B> Daryl Malas; dispatch@ietf.org<BR><B>Subject:</B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN><o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Daryl:<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">I=20
want to be sure I understand what you&#8217;re proposing.&nbsp; In what nam=
e server=20
would the modified records be? One local to SSP1 or some general industry D=
NS?=20
I&#8217;m trying to understand how SSP1 modifies an RR for an SSP2=20
number.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt">Thanks,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Calibri','sans-serif'; COLOR: #1f497d; FONT-SIZE: 11=
pt"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">Penn=20
Pfautz</SPAN><SPAN style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">AT&amp;T=20
Access Management</SPAN><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P></DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; COLOR: #1f497d; FONT-SIZE: 10pt=
">+1-732-420-4962</SPAN><SPAN=20
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<DIV>
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<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-FAMILY: 'Tahoma','sans-serif'; FONT-SIZE: 10pt">=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B>On Behalf O=
f=20
</B>Daryl Malas<BR><B>Sent:</B> Monday, March 01, 2010 6:51 PM<BR><B>To:</B=
>=20
'dispatch@ietf.org'<BR><B>Subject:</B> [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">All,</SPAN><o:=
p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">A New Internet=
-Draft=20
is available from the on-line Internet-Drafts directories. <BR>&nbsp;<BR>Ti=
tle :=20
SIP Egress Route Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Regards,</SPAN=
><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl</SPAN><o=
:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">--------------=
-------------</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Daryl=20
Malas</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">Project Direct=
or, IP=20
Interconnects</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt">w - +1 303 661=
=20
3302</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-FAMILY: 'Arial','sans-serif'; FONT-SIZE: 10pt"><A=20
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com=
</A></SPAN><o:p></o:p></P></DIV>
<DIV>
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From: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
To: "Daryl Malas" <D.Malas@cablelabs.com>, <dispatch@ietf.org>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Daryl:

Thanks. I now understand what you're after and it is just the
implementation questions that concern me.  It might be useful to make
clear you're not proposing any specific implementation.

For example, in a different context (drinks) if it were proposed to
support building these kinds of records in the registry as opposed to
the SSP1 network  I'd probably be agin it.

=20

Thanks for the clarification

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 1:01 PM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

I think what you're describing is leaning towards implementation
specific questions.  I apologize, my example below is backwards, but the
principle is the same.  SSP2 provisions their numbers and "routes" into
the ENUM server.  SSP1 can see the ingress routes for SSP2 numbers.
SSP1 can then specify their own egress routes in the same ENUM server.

=20

The method for SSP1's egress routes for SSP2 to be received into a
"local copy" is implementation specific, imo.  This being said,
obviously, SSP3 and SSP4 would not (want to) see SSP1's provisioned
egress routes for SSP2 when they query the ENUM server.  If this is
resolved in some "for queries from here give this" or "here is a local
copy of peer numbers and routes plus *your* egress routes" is
implementation specific.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl=20

I was a little confused by this since I thought the call was from SSP1
to SSP2 and that SSP2 provisioned its numbers into the registry. If SSP1
get the SSP2 data and wants to add their egress routes in their local
copy I can see that but I wouldn't expect the resulting elaborated
NAPTRs to be in the Registry.=20

Have I got this right now?

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

The use case for this application is SSP1 (or someone on their behalf)
provisions their numbers and relative routes into the LUF associated
with their federation.  SSP2 reviews SSP1's "routes" available to them
in the LUF based on what SSP1 provisioned.  They then add an egress
route and associate it with the ingress route SSP1 provisioned.

=20

Let me know if this makes sense.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl:

I want to be sure I understand what you're proposing.  In what name
server would the modified records be? One local to SSP1 or some general
industry DNS? I'm trying to understand how SSP1 modifies an RR for an
SSP2 number.

=20

Thanks,

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

=20

All,

=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.=20
=20
Title : SIP Egress Route Use in ENUM=20
Author(s) : D. Malas=20
Filename : draft-malas-dispatch-sip-egress-route-00.txt=20
Pages : 11=20
Date : 2010-03-01=20
=20
In some cases a UA (or proxy) within a Session Initiation Protocol=20
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy=20
options to reach an ingress proxy within another SSP to reach the=20
target UA. If the originating SSP uses ENUM to identify the ingress=20
proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
response to identify a specific preferred egress proxy from the set=20
of multiple parallel proxies. This draft solves this problem by=20
defining a method for returning deterministic routing information=20
within an ENUM NAPTR response enabling the UA (or proxy) to choose a=20
preferred egress proxy. Using this information, the originating UA=20
now sends the SIP request through the predetermined preferred egress=20
proxy to reach the target SSP's proxy. This document describes=20
several methods to make this determination by incorporating variables=20
into a SIP service URI contained in the E.164 Number Mapping (ENUM)=20
response.=20
=20
A URL for this Internet-Draft is:=20
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-rout
e-00.txt=20

=20

Regards,

=20

Daryl

=20

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

Daryl Malas

Project Director, IP Interconnects

w - +1 303 661 3302

mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>=20

=20


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<p class=3DMsoNormal><span =
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color:#1F497D'>Daryl:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks. I now understand what you&#8217;re after and it =
is just the
implementation questions that concern me. &nbsp;It might be useful to =
make
clear you&#8217;re not proposing any specific =
implementation.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>For example, in a different context (drinks) if it were =
proposed
to support building these kinds of records in the registry as opposed to =
the
SSP1 network &nbsp;I&#8217;d probably be agin it.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks for the clarification<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b>Sent:</b> Wednesday, March 03, 2010 1:01 PM<br>
<b>To:</b> PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Penn,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>I think what you're describing is leaning towards =
implementation
specific questions.&nbsp; I apologize, my example below is backwards, =
but the
principle is the same.&nbsp; SSP2 provisions their numbers and =
&quot;routes&quot;
into the ENUM server.&nbsp; SSP1 can see the ingress routes for SSP2
numbers.&nbsp; SSP1 can then specify their own egress routes in the same =
ENUM
server.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The method for&nbsp;SSP1's egress routes for SSP2&nbsp;to
be&nbsp;received into a &quot;local copy&quot; is implementation =
specific,
imo.&nbsp; This being said, obviously, SSP3 and SSP4 would not (want to) =
see
SSP1's provisioned egress routes for SSP2 when they query the ENUM
server.&nbsp; If this is resolved in some &quot;for queries from here =
give
this&quot; or &quot;here is a local copy of peer numbers and routes plus =
*your*
egress routes&quot; is implementation specific.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Daryl</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> PFAUTZ, PENN L (ATTCORP)
[mailto:pp3129@att.com] <br>
<b>Sent:</b> Wednesday, March 03, 2010 10:38 AM<br>
<b>To:</b> Daryl Malas; dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Daryl <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I was a little confused by this since I thought the call =
was
from SSP1 to SSP2 and that SSP2 provisioned its numbers into the =
registry. If SSP1
get the SSP2 data and wants to add their egress routes in their local =
copy I
can see that but I wouldn&#8217;t expect the resulting elaborated NAPTRs =
to be
in the Registry. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Have I got this right now?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b>Sent:</b> Wednesday, March 03, 2010 11:22 AM<br>
<b>To:</b> PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Penn,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The use case for this application is SSP1 (or someone on =
their
behalf) provisions their numbers and relative&nbsp;routes into the LUF
associated with their federation.&nbsp; SSP2 reviews&nbsp;SSP1's
&quot;routes&quot; available to them in the LUF based on what SSP1
provisioned.&nbsp; They then add an egress route and associate it with =
the
ingress route SSP1 provisioned.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Let me know if this makes sense.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Daryl</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> PFAUTZ, PENN L (ATTCORP)
[mailto:pp3129@att.com] <br>
<b>Sent:</b> Tuesday, March 02, 2010 12:24 PM<br>
<b>To:</b> Daryl Malas; dispatch@ietf.org<br>
<b>Subject:</b> RE: [dispatch] I-D Action: =
draft-malas-dispatch-sip-egress-route-00</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Daryl:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I want to be sure I understand what you&#8217;re
proposing.&nbsp; In what name server would the modified records be? One =
local
to SSP1 or some general industry DNS? I&#8217;m trying to understand how =
SSP1
modifies an RR for an SSP2 number.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>Daryl
Malas<br>
<b>Sent:</b> Monday, March 01, 2010 6:51 PM<br>
<b>To:</b> 'dispatch@ietf.org'<br>
<b>Subject:</b> [dispatch] I-D Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>All,</span><o=
:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>A
New Internet-Draft is available from the on-line Internet-Drafts =
directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR =
<br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a =
<br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress =
<br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables =
<br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch=
-sip-egress-route-00.txt</a>
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Regards,</spa=
n><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl</span><=
o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>-------------=
--------------</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Daryl
Malas</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Project
Director, IP Interconnects</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>w
- +1 303 661 3302</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><a
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.c=
om</a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

</body>

</html>

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From dean.willis@softarmor.com  Wed Mar  3 11:26:33 2010
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On Mar 3, 2010, at 10:48 AM, Daryl Malas wrote:

> Dean,
>
> I don't agree with your map analogy.  Every hop along the way in the  
> draft example does not require a new look-up.  The single ENUM  
> response provides all of the information necessary to reach the  
> target.  It just includes "next hop" information to provide more  
> intelligent routing.  The final target information is still  
> contained in the single response. Perhaps the reason ENUM may be a  
> little different is based on the application associated with the  
> NAPTR.  SIP is an intelligent application with the ability to  
> receive detailed information to reach the target UA.  The draft just  
> leverages the SIP service NAPTR to provide the SIP application with  
> more of this detailed information.  This way information does not  
> need to be stored and managed in many different locations.
>


But the "egress route" returned by the ENUM lookup is dependent on  
where the lookup is happening, right? This means lots of different  
ENUM databases that return different values for queries on the same  
key. This is the exact antithesis of the scalability model of DNS,  
which is predicated on universally-valid results that allow  
dissemination through recursive lookups and caching.

If each of SSP1 and SSP2 interconnects with SSP3, the "egress route"  
from SSP1 to SSP3 is NOT the same as the route from SSP2 to SSP3.

So the ENUM database seen by nodes at SSP1 returns different values  
from the ENUM database seen by nodes at SSP2, which is different yet  
again for nodes at SSP3. This means that for the "SSP3" key, at least  
two different context-dependent values are being stored in the  
databases. It's just not how DNS was designed to distribute. Yeah, it  
sort of works if you assume lots of independently provisioned  
databases, but it's still an operational nightmare and all you're  
getting out of ENUM is a standardized database format.

You have here a very real problem that is exactly analogous to the IP- 
level route-peering problem. Many years of working this problem at the  
IP level have shown that statically provisioned databases (aka DNS)  
are NOT an effective way to solve the problem at a large scale. DNS  
provides a nice mechanism for turning a name (or phone number) into a  
target IP address, but a lousy mechanism for telling us each of the  
intermediate hops needed to get from a source IP address to a target  
IP address (which is a vector of the router in between the two  
addresses).

We really need a SIP-targeted analog of the BGP/OSPF route-exchange  
protocols to solve the general problem.

--
Dean






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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 3, 2010, =
at 10:48 AM, Daryl Malas wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"> <div =
style=3D"WORD-WRAP: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space"> <div dir=3D"ltr" =
align=3D"left"><font color=3D"#0000ff" size=3D"2" face=3D"Arial"><span =
class=3D"331113416-03032010">Dean,</span></font></div> <div dir=3D"ltr" =
align=3D"left"><font color=3D"#0000ff" size=3D"2" face=3D"Arial"><span =
class=3D"331113416-03032010"></span></font>&nbsp;</div> <div dir=3D"ltr" =
align=3D"left"><font color=3D"#0000ff" size=3D"2" face=3D"Arial"><span =
class=3D"331113416-03032010">I don't agree with your map analogy.&nbsp; =
Every hop along the way in the draft example does not require a new =
look-up.&nbsp; The single ENUM response provides all of the information =
necessary to reach the target.&nbsp; It just includes "next hop" =
information to provide more intelligent routing.&nbsp; The final target =
information is still contained in the single response. Perhaps the =
reason ENUM may be a little different is based on the application =
associated with the NAPTR.&nbsp; SIP is an intelligent application with =
the ability to receive detailed information to reach the target =
UA.&nbsp; The draft just leverages the SIP service NAPTR to provide the =
SIP application with more of this detailed information.&nbsp; This way =
information does not need to be stored and managed in many different =
locations.</span></font></div> <div dir=3D"ltr" align=3D"left"><font =
color=3D"#0000ff" size=3D"2" face=3D"Arial"><span =
class=3D"331113416-03032010"></span></font>&nbsp;</div> =
</div></blockquote></div><div><br></div><div>But the "egress route" =
returned by the ENUM lookup is dependent on where the lookup is =
happening, right? This means lots of different ENUM databases that =
return different values for queries on the same key. This is the exact =
antithesis of the scalability model of DNS, which is predicated on =
universally-valid results that allow dissemination through recursive =
lookups and caching.</div><div><br></div><div>If each of SSP1 and SSP2 =
interconnects with SSP3, the "egress route" from SSP1 to SSP3 is NOT the =
same as the route from SSP2 to SSP3.</div><div><br></div><div>So the =
ENUM database seen by nodes at SSP1 returns different values from the =
ENUM database seen by nodes at SSP2, which is different yet again for =
nodes at SSP3. This means that for the "SSP3" key, at least two =
different context-dependent values are being stored in the databases. =
It's just not how DNS was designed to distribute. Yeah, it sort of works =
if you assume lots of independently provisioned databases, but it's =
still an operational nightmare and all you're getting out of ENUM is a =
standardized database format.</div><div><br></div><div>You have here a =
very real problem that is exactly analogous to the IP-level =
route-peering problem. Many years of working this problem at the IP =
level have shown that statically provisioned databases (aka DNS) are NOT =
an effective way to solve the problem at a large scale. DNS provides a =
nice mechanism for turning a name (or phone number) into a target IP =
address, but a lousy mechanism for telling us each of the intermediate =
hops needed to get from a source IP address to a target IP address =
(which is a vector of the router in between the two =
addresses).</div><div><br></div><div>We really need a SIP-targeted =
analog of the BGP/OSPF route-exchange protocols to solve the general =
problem.</div><div><br></div><div>--</div><div>Dean</div><div><br></div><d=
iv><br></div><div><br></div><div><br></div><div><br></div></body></html>=

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On Mar 3, 2010, at 12:01 PM, Daryl Malas wrote:

>
> The method for SSP1's egress routes for SSP2 to be received into a  
> "local copy" is implementation specific, imo.  This being said,  
> obviously, SSP3 and SSP4 would not (want to) see SSP1's provisioned  
> egress routes for SSP2 when they query the ENUM server.  If this is  
> resolved in some "for queries from here give this" or "here is a  
> local copy of peer numbers and routes plus *your* egress routes" is  
> implementation specific.

And this part right here is exactly why the DNS infrastructure of ENUM  
is NOT a good solution for the problem. Unlike Einstein, DNS does not  
embrace relativity (AKA location-context sensitivity). There are ugly  
hacks to make relativity happen in certain limited cases, but they  
defeat the fundamental scalability mechanisms of the DNS.

--
Dean
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 3, 2010, =
at 12:01 PM, Daryl Malas wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple"><div dir=3D"ltr" align=3D"left"><font =
class=3D"Apple-style-span" color=3D"#0000FF" face=3D"Arial"><span =
class=3D"Apple-style-span" style=3D"font-size: =
small;"><br></span></font></div><div dir=3D"ltr" align=3D"left"><span =
class=3D"483004417-03032010"><font color=3D"#0000ff" size=3D"2" =
face=3D"Arial">The method for&nbsp;SSP1's egress routes for SSP2&nbsp;to =
be&nbsp;received into a "local copy" is implementation specific, =
imo.&nbsp; This being said, obviously, SSP3 and SSP4 would not (want to) =
see SSP1's provisioned egress routes for SSP2 when they query the ENUM =
server.&nbsp; If this is resolved in some "for queries from here give =
this" or "here is a local copy of peer numbers and routes plus *your* =
egress routes" is implementation =
specific.</font></span></div></div></span></blockquote></div><br><div>And =
this part right here is exactly why the DNS infrastructure of ENUM is =
NOT a good solution for the problem. Unlike Einstein, DNS does not =
embrace relativity (AKA location-context sensitivity). There are ugly =
hacks to make relativity happen in certain limited cases, but they =
defeat the fundamental scalability mechanisms of the =
DNS.</div><div><br></div><div>--</div><div>Dean</div></body></html>=

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From: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
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Ah, this is exactly why the GSMA chose to handle the problem by moving
it to layer 3:  The IPX carriers implement a private IP network through
which connecting service providers advertise the IP addresses of their
border elements via BGP. An ENUM lookup resolves to a common IP address
for all querying parties but each SP associates the IP address with a
route learned from their chosen IPX provider. This is, of course, just
the basic Internet approach moved off the "public" Internet. It also
works for bilateral direct connections and, in principle, should prefer
such connections over a route through an IPX provider where both are
available. =20

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Dean Willis
Sent: Wednesday, March 03, 2010 2:26 PM
To: Daryl Malas
Cc: 'dispatch@ietf.org'
Subject: Re: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

=20

On Mar 3, 2010, at 10:48 AM, Daryl Malas wrote:





Dean,

=20

I don't agree with your map analogy.  Every hop along the way in the
draft example does not require a new look-up.  The single ENUM response
provides all of the information necessary to reach the target.  It just
includes "next hop" information to provide more intelligent routing.
The final target information is still contained in the single response.
Perhaps the reason ENUM may be a little different is based on the
application associated with the NAPTR.  SIP is an intelligent
application with the ability to receive detailed information to reach
the target UA.  The draft just leverages the SIP service NAPTR to
provide the SIP application with more of this detailed information.
This way information does not need to be stored and managed in many
different locations.

=20

=20

But the "egress route" returned by the ENUM lookup is dependent on where
the lookup is happening, right? This means lots of different ENUM
databases that return different values for queries on the same key. This
is the exact antithesis of the scalability model of DNS, which is
predicated on universally-valid results that allow dissemination through
recursive lookups and caching.

=20

If each of SSP1 and SSP2 interconnects with SSP3, the "egress route"
from SSP1 to SSP3 is NOT the same as the route from SSP2 to SSP3.

=20

So the ENUM database seen by nodes at SSP1 returns different values from
the ENUM database seen by nodes at SSP2, which is different yet again
for nodes at SSP3. This means that for the "SSP3" key, at least two
different context-dependent values are being stored in the databases.
It's just not how DNS was designed to distribute. Yeah, it sort of works
if you assume lots of independently provisioned databases, but it's
still an operational nightmare and all you're getting out of ENUM is a
standardized database format.

=20

You have here a very real problem that is exactly analogous to the
IP-level route-peering problem. Many years of working this problem at
the IP level have shown that statically provisioned databases (aka DNS)
are NOT an effective way to solve the problem at a large scale. DNS
provides a nice mechanism for turning a name (or phone number) into a
target IP address, but a lousy mechanism for telling us each of the
intermediate hops needed to get from a source IP address to a target IP
address (which is a vector of the router in between the two addresses).

=20

We really need a SIP-targeted analog of the BGP/OSPF route-exchange
protocols to solve the general problem.

=20

--

Dean

=20

=20

=20

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ah, this is exactly why the GSMA chose to handle the =
problem by
moving it to layer 3: &nbsp;The IPX carriers implement a private IP =
network
through which connecting service providers advertise the IP addresses of =
their
border elements via BGP. An ENUM lookup resolves to a common IP address =
for all
querying parties but each SP associates the IP address with a route =
learned
from their chosen IPX provider. This is, of course, just the basic =
Internet
approach moved off the &#8220;public&#8221; Internet. It also works for
bilateral direct connections and, in principle, should prefer such =
connections
over a route through an IPX provider where both are available.&nbsp; =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>Dean
Willis<br>
<b>Sent:</b> Wednesday, March 03, 2010 2:26 PM<br>
<b>To:</b> Daryl Malas<br>
<b>Cc:</b> 'dispatch@ietf.org'<br>
<b>Subject:</b> Re: [dispatch] I-D Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal>On Mar 3, 2010, at 10:48 AM, Daryl Malas =
wrote:<o:p></o:p></p>

</div>

<p class=3DMsoNormal><br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Dean,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>I don't agree with your map analogy.&nbsp; Every hop along =
the way
in the draft example does not require a new look-up.&nbsp; The single =
ENUM
response provides all of the information necessary to reach the =
target.&nbsp;
It just includes &quot;next hop&quot; information to provide more =
intelligent
routing.&nbsp; The final target information is still contained in the =
single
response. Perhaps the reason ENUM may be a little different is based on =
the
application associated with the NAPTR.&nbsp; SIP is an intelligent =
application
with the ability to receive detailed information to reach the target =
UA.&nbsp;
The draft just leverages the SIP service NAPTR to provide the SIP =
application
with more of this detailed information.&nbsp; This way information does =
not
need to be stored and managed in many different =
locations.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>But the &quot;egress route&quot; returned by the =
ENUM lookup
is dependent on where the lookup is happening, right? This means lots of
different ENUM databases that return different values for queries on the =
same
key. This is the exact antithesis of the scalability model of DNS, which =
is predicated
on universally-valid results that allow dissemination through recursive =
lookups
and caching.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>If each of SSP1 and SSP2 interconnects with SSP3, =
the
&quot;egress route&quot; from SSP1 to SSP3 is NOT the same as the route =
from
SSP2 to SSP3.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>So the ENUM database seen by nodes at SSP1 returns =
different
values from the ENUM database seen by nodes at SSP2, which is different =
yet
again for nodes at SSP3. This means that for the &quot;SSP3&quot; key, =
at least
two different context-dependent values are being stored in the =
databases. It's
just not how DNS was designed to distribute. Yeah, it sort of works if =
you
assume lots of independently provisioned databases, but it's still an
operational nightmare and all you're getting out of ENUM is a =
standardized
database format.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>You have here a very real problem that is exactly =
analogous
to the IP-level route-peering problem. Many years of working this =
problem at
the IP level have shown that statically provisioned databases (aka DNS) =
are NOT
an effective way to solve the problem at a large scale. DNS provides a =
nice
mechanism for turning a name (or phone number) into a target IP address, =
but a
lousy mechanism for telling us each of the intermediate hops needed to =
get from
a source IP address to a target IP address (which is a vector of the =
router in
between the two addresses).<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>We really need a SIP-targeted analog of the =
BGP/OSPF
route-exchange protocols to solve the general problem.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>--<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>Dean<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</body>

</html>

------_=_NextPart_001_01CABB09.685975B7--

From adam@nostrum.com  Wed Mar  3 13:42:56 2010
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On 3/3/10 1:26 PM, Dean Willis wrote:
>
> We really need a SIP-targeted analog of the BGP/OSPF route-exchange 
> protocols to solve the general problem.

So, what, like TRIP, but for things other than phone numbers?

/a

From dschwartz@xconnect.net  Wed Mar  3 15:57:06 2010
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From: David Schwartz <dschwartz@xconnect.net>
To: Adam Roach <adam@nostrum.com>, Dean Willis <dean.willis@softarmor.com>
Date: Thu, 4 Mar 2010 01:57:02 +0200
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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In the now expired draft-kaplan-drinks-lrf-requirements-00, Hadriel seemed =
to be pushing for a "Session Location Routing Protocol" (SLRP) to deal with=
 this issue...
________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Ad=
am Roach [adam@nostrum.com]
Sent: Wednesday, March 03, 2010 11:42 PM
To: Dean Willis
Cc: 'dispatch@ietf.org'
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

On 3/3/10 1:26 PM, Dean Willis wrote:
>
> We really need a SIP-targeted analog of the BGP/OSPF route-exchange
> protocols to solve the general problem.

So, what, like TRIP, but for things other than phone numbers?

/a
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From D.Malas@cablelabs.com  Wed Mar  3 16:01:52 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dwight, Timothy M (Tim)'" <timothy.dwight@verizon.com>, "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
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--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66CEsrvxchg_
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Tim,

Thank you for the additional context.  I think we are in sync regarding the=
 purposes of the draft as you indicate below.  Regarding placing a route he=
ader into a local non-ENUM database is obviously outside the scope of the d=
raft.  If you use a local private ENUM server to add your own route header =
(or egress route) to the NAPTR URI, is implementation specific and up to yo=
u.  Even in this implementation specific case, I believe defining a best pr=
actice for defining and including egress information in the NAPTR URI will =
provide better industry adoption and implementation consistency.  Do you ag=
ree?

Regards,

Daryl

________________________________
From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]
Sent: Wednesday, March 03, 2010 4:34 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Daryl,

I guess our use cases differ.

The draft illustrates a NAPTR producing a URI whose host part is the ingres=
s point to SSP2's network through which SSP2 prefers to receive session req=
uests targeted at the example TN.  It then talks about mechanisms to includ=
e in the ENUM response, information that "steer" the session request messag=
e through the "egress point" from SSP1's network, desired by SSP1.

I get that.  I've done it, in a private ENUM, so I know it's possible.  It'=
s a little trickier to manage in the context of a shared registry, but stil=
l seems within the capabilities of the commercial ENUM servers with which I=
'm familiar.  Thinking of the "route header" solution, SSP1 could maintain =
in the registry a table that instructs the registry provider to append a ro=
ute header identifying SBE2 to any URI whose host part was "SBE3.ssp2.examp=
le.com".  Although this sort of detail isn't subject to standardization I a=
ssume you have something like that in mind.

This seems workable in large part because SSP1 and SSP2 will in practice kn=
ow quite a lot about their interconnection.  If SSP1 had no prior knowledge=
 of the "ingress points" (e.g., SBE3) to SSP2's networks, it would be impos=
sible to maintain such a table.  But in practice it will know these ingress=
 points; moreover it will know how its own "egress points" are connected to=
 them (e.g., via a partial mesh of IPsec tunnels).  Again I'm assuming you =
have something like this in mind.


What I meant to propose, was a different use case.  Suppose SSP2 doesn't id=
entify its ingress point in the URI resulting from the NAPTR it returns.  S=
uppose instead it returns a URI whose host part is something like "call-ser=
ver-1.ssp2.example.com".  This is the FQDN of the call server (e.g., softsw=
itch, application server) in SSP2 network that will serve calls to the exam=
ple TN.  Returning such a value avoids SSP2 having to make another routing =
decision after receiving the call from SSP1.

But if you do that, you may also want to "steer" the session request to an =
ingress point (SBE3 in the draft).  This could be done using e.g., a ROUTE =
header imbedded in the URI in the NAPTR.  The logic to determine the conten=
t of this ROUTE header is simple since it depends only on where the call is=
 going not where it came from; and easily administered since it would be sp=
ecified by the same entity that serves the example TN.

SSP1 might also want to steer the call through SBE2, which could be done vi=
a any of the mechanisms in the draft, or via some table maintained outside =
of ENUM.

The 2 approaches are AFAIK both possible with existing standards and suppor=
table by currently available products.  Were it up to me, based on my curre=
nt (probably flawed and incomplete) understanding, I'd probably do the form=
er (identification of the "ingress point" to the serving domain) in the ENU=
M database or registry, and the latter (identification of the "egress point=
" from the querying domain, as a function of the URI returned in an ENUM re=
sponse) either via some non-ENUM database, or via some "private data" in my=
 local ENUM servers (private meaning not managed by the registry service).

Anyway I appreciated and enjoyed the draft, and appreciate the technical ex=
change.

Best regards,

Tim


________________________________
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 3:45 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Tim,

SSP2 does provide their preferred "ingress point."  This information is sti=
ll in the ENUM response regardless of whether an additional egress route va=
riable has been added or not.

I am not sure I understand your second paragraph.

Regards,

Daryl

________________________________
From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]
Sent: Wednesday, March 03, 2010 12:25 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Why not let SSP2 provide their preferred "ingress point" for session reques=
ts addressed to the queried TN, in the ENUM response?  That would presumabl=
y be the same for all querying networks, and it is data for which the SSP s=
erving the queried TN (SSP2) is authoritative.

In the common case of "mated peering points" (e.g., a pair of SBCs in SSP1 =
connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a local tabl=
e mapping the remote-network ingress points received in ENUM responses, to =
its own egress points (i.e., its SBCs that are connected via IPsec to the i=
ngress point in the ENUM response).

tim

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Wednesday, March 03, 2010 12:01 PM
To: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

I think what you're describing is leaning towards implementation specific q=
uestions.  I apologize, my example below is backwards, but the principle is=
 the same.  SSP2 provisions their numbers and "routes" into the ENUM server=
.  SSP1 can see the ingress routes for SSP2 numbers.  SSP1 can then specify=
 their own egress routes in the same ENUM server.

The method for SSP1's egress routes for SSP2 to be received into a "local c=
opy" is implementation specific, imo.  This being said, obviously, SSP3 and=
 SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 whe=
n they query the ENUM server.  If this is resolved in some "for queries fro=
m here give this" or "here is a local copy of peer numbers and routes plus =
*your* egress routes" is implementation specific.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl
I was a little confused by this since I thought the call was from SSP1 to S=
SP2 and that SSP2 provisioned its numbers into the registry. If SSP1 get th=
e SSP2 data and wants to add their egress routes in their local copy I can =
see that but I wouldn't expect the resulting elaborated NAPTRs to be in the=
 Registry.
Have I got this right now?

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

The use case for this application is SSP1 (or someone on their behalf) prov=
isions their numbers and relative routes into the LUF associated with their=
 federation.  SSP2 reviews SSP1's "routes" available to them in the LUF bas=
ed on what SSP1 provisioned.  They then add an egress route and associate i=
t with the ingress route SSP1 provisioned.

Let me know if this makes sense.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

Thanks,

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<BODY lang=3DEN-US link=3Dblue vLink=3Dpurple>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Tim,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Thank you for the additional context.&nbsp; I think&n=
bsp;we=20
are in sync regarding&nbsp;the purposes of the draft as you indicate=20
below.&nbsp; Regarding placing a route header into a local non-ENUM databas=
e is=20
obviously outside the scope of the draft.&nbsp; If you use a local private =
ENUM=20
server to add your own route header (or egress route) to the NAPTR URI, is=
=20
implementation specific and up to you.&nbsp; Even in this implementation=20
specific case, I believe defining a best practice for defining and includin=
g=20
egress information in the NAPTR URI will provide better industry adoption a=
nd=20
implementation consistency.&nbsp; Do you agree?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D967333723-03032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Dwight, Timothy M (Tim)=20
[mailto:timothy.dwight@verizon.com] <BR><B>Sent:</B> Wednesday, March 03, 2=
010=20
4:34 PM<BR><B>To:</B> Daryl Malas; PFAUTZ, PENN L (ATTCORP)<BR><B>Subject:<=
/B>=20
RE: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Daryl,<o=
:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">I guess =
our use=20
cases differ. &nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">The draf=
t=20
illustrates a NAPTR producing a URI whose host part is the ingress point to=
=20
SSP2&#8217;s network through which SSP2 prefers to receive session requests=
 targeted=20
at the example TN. &nbsp;It then talks about mechanisms to include in the E=
NUM=20
response, information that &#8220;steer&#8221; the session request message =
through the=20
&#8220;egress point&#8221; from SSP1&#8217;s network, desired by=20
SSP1.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">I get=20
that.&nbsp; I&#8217;ve done it, in a private ENUM, so I know it&#8217;s pos=
sible. &nbsp;It&#8217;s=20
a little trickier to manage in the context of a shared registry, but still =
seems=20
within the capabilities of the commercial ENUM servers with which I&#8217;m=
=20
familiar.&nbsp; Thinking of the &#8220;route header&#8221; solution, SSP1 c=
ould maintain in=20
the registry a table that instructs the registry provider to append a route=
=20
header identifying SBE2 to any URI whose host part was &#8220;SBE3.ssp2.exa=
mple.com&#8221;.=20
&nbsp;Although this sort of detail isn&#8217;t subject to standardization I=
 assume you=20
have something like that in mind.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">This see=
ms=20
workable in large part because SSP1 and SSP2 will in practice know quite a =
lot=20
about their interconnection. &nbsp;If SSP1 had no prior knowledge of the=20
&#8220;ingress points&#8221; (e.g., SBE3) to SSP2&#8217;s networks, it woul=
d be impossible to=20
maintain such a table. &nbsp;But in practice it will know these ingress poi=
nts;=20
moreover it will know how its own &#8220;egress points&#8221; are connected=
 to them (e.g.,=20
via a partial mesh of IPsec tunnels).&nbsp; Again I&#8217;m assuming you ha=
ve=20
something like this in mind.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">What I m=
eant to=20
propose, was a different use case.&nbsp; Suppose SSP2 doesn&#8217;t identif=
y its=20
ingress point in the URI resulting from the NAPTR it returns. &nbsp;Suppose=
=20
instead it returns a URI whose host part is something like=20
&#8220;call-server-1.ssp2.example.com&#8221;. &nbsp;This is the FQDN of the=
 call server=20
(e.g., softswitch, application server) in SSP2 network that will serve call=
s to=20
the example TN. &nbsp;Returning such a value avoids SSP2 having to make ano=
ther=20
routing decision after receiving the call from=20
SSP1.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">But if y=
ou do=20
that, you may also want to &#8220;steer&#8221; the session request to an in=
gress point (SBE3=20
in the draft). &nbsp;This could be done using e.g., a ROUTE header imbedded=
 in=20
the URI in the NAPTR.&nbsp; The logic to determine the content of this ROUT=
E=20
header is simple since it depends only on where the call is going not where=
 it=20
came from; and easily administered since it would be specified by the same=
=20
entity that serves the example TN.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">SSP1 mig=
ht also=20
want to steer the call through SBE2, which could be done via any of the=20
mechanisms in the draft, or via some table maintained outside of ENUM.=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">The 2=20
approaches are AFAIK both possible with existing standards and supportable =
by=20
currently available products. &nbsp;Were it up to me, based on my current=20
(probably flawed and incomplete) understanding, I&#8217;d probably do the f=
ormer=20
(identification of the &#8220;ingress point&#8221; to the serving domain) i=
n the ENUM=20
database or registry, and the latter (identification of the &#8220;egress p=
oint&#8221; from=20
the querying domain, as a function of the URI returned in an ENUM response)=
=20
either via some non-ENUM database, or via some &#8220;private data&#8221; i=
n my local ENUM=20
servers (private meaning not managed by the registry=20
service).<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Anyway I=
=20
appreciated and enjoyed the draft, and appreciate the technical=20
exchange.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Best=20
regards,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Tim<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: blue 1.5pt solid; PADDING=
-BOTTOM: 0in; PADDING-LEFT: 4pt; PADDING-RIGHT: 0in; BORDER-TOP: medium non=
e; BORDER-RIGHT: medium none; PADDING-TOP: 0in">
<DIV>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Daryl=20
Malas [mailto:D.Malas@cablelabs.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 3:45=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Dwight, Timothy M=
 (Tim);=20
PFAUTZ, PENN L (ATTCORP)<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Tim,</SPAN></FON=
T><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">SSP2 does provid=
e their=20
preferred "ingress point."&nbsp; This information is still in the ENUM resp=
onse=20
regardless of whether an additional egress route variable has been added or=
=20
not.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">I am not sure I=
=20
understand your second paragraph.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Dwight,=20
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 12:2=
5=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas; PFAU=
TZ,=20
PENN L (ATTCORP)<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B=
> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Why not =
let=20
SSP2 provide their preferred &#8220;ingress point&#8221; for session reques=
ts addressed to=20
the queried TN, in the ENUM response? &nbsp;That would presumably be the sa=
me=20
for all querying networks, and it is data for which the SSP serving the que=
ried=20
TN (SSP2) is authoritative.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">In the c=
ommon=20
case of &#8220;mated peering points&#8221; (e.g., a pair of SBCs in SSP1 co=
nnected to a pair=20
of SBCs in SSP2 via IPsec) SSP2 could keep a local table mapping the=20
remote-network ingress points received in ENUM responses, to its own egress=
=20
points (i.e., its SBCs that are connected via IPsec to the ingress point in=
 the=20
ENUM response).<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">tim<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: blue 1.5pt solid; PADDING=
-BOTTOM: 0in; PADDING-LEFT: 4pt; PADDING-RIGHT: 0in; BORDER-TOP: medium non=
e; BORDER-RIGHT: medium none; PADDING-TOP: 0in">
<DIV>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
>=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Daryl Malas<BR><B><SPAN=
=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 12:0=
1=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> 'PFAUTZ, PENN L=20
(ATTCORP)'; dispatch@ietf.org<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Penn,</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">I think what you=
're=20
describing is leaning towards implementation specific questions.&nbsp; I=20
apologize, my example below is backwards, but the principle is the same.&nb=
sp;=20
SSP2 provisions their numbers and "routes" into the ENUM server.&nbsp; SSP1=
 can=20
see the ingress routes for SSP2 numbers.&nbsp; SSP1 can then specify their =
own=20
egress routes in the same ENUM server.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">The method=20
for&nbsp;SSP1's egress routes for SSP2&nbsp;to be&nbsp;received into a "loc=
al=20
copy" is implementation specific, imo.&nbsp; This being said, obviously, SS=
P3=20
and SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 =
when=20
they query the ENUM server.&nbsp; If this is resolved in some "for queries =
from=20
here give this" or "here is a local copy of peer numbers and routes plus *y=
our*=20
egress routes" is implementation specific.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> PFAUTZ,=20
PENN L (ATTCORP) [mailto:pp3129@att.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 10:3=
8=20
AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas;=20
dispatch@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></=
B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Daryl=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">I was a lit=
tle=20
confused by this since I thought the call was from SSP1 to SSP2 and that SS=
P2=20
provisioned its numbers into the registry. If SSP1 get the SSP2 data and wa=
nts=20
to add their egress routes in their local copy I can see that but I wouldn&=
#8217;t=20
expect the resulting elaborated NAPTRs to be in the Registry.=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Have I got =
this=20
right now?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">Penn=20
Pfautz</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">AT&amp;T Acce=
ss=20
Management</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P></DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">+1-732-420-49=
62</SPAN></FONT><FONT=20
color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p></o:p>=
</SPAN></FONT></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Daryl=20
Malas [mailto:D.Malas@cablelabs.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 11:2=
2=20
AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> PFAUTZ, PENN L=20
(ATTCORP); dispatch@ietf.org<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></FONT></P></DIV>=
</DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Penn,</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">The use case for=
 this=20
application is SSP1 (or someone on their behalf) provisions their numbers a=
nd=20
relative&nbsp;routes into the LUF associated with their federation.&nbsp; S=
SP2=20
reviews&nbsp;SSP1's "routes" available to them in the LUF based on what SSP=
1=20
provisioned.&nbsp; They then add an egress route and associate it with the=
=20
ingress route SSP1 provisioned.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Let me know if t=
his=20
makes sense.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> PFAUTZ,=20
PENN L (ATTCORP) [mailto:pp3129@att.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, March 02, 2010 12:24=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas;=20
dispatch@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></=
B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Daryl:<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">I want to b=
e sure=20
I understand what you&#8217;re proposing.&nbsp; In what name server would t=
he modified=20
records be? One local to SSP1 or some general industry DNS? I&#8217;m tryin=
g to=20
understand how SSP1 modifies an RR for an SSP2=20
number.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Thanks,<o:p=
></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">Penn=20
Pfautz</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">AT&amp;T Acce=
ss=20
Management</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P></DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">+1-732-420-49=
62</SPAN></FONT><FONT=20
color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p></o:p>=
</SPAN></FONT></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
>=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Daryl Malas<BR><B><SPAN=
=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Monday, March 01, 2010 6:51=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B>=20
'dispatch@ietf.org'<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN>=
</B>=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></FONT></P></DIV>=
</DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">All,</SPAN></FONT><o:p></o:p>=
</P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">A New Internet-Draft is avail=
able=20
from the on-line Internet-Drafts directories. <BR>&nbsp;<BR>Title : SIP Egr=
ess=20
Route Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Regards,</SPAN></FONT><o:p></=
o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Daryl</SPAN></FONT><o:p></o:p=
></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">---------------------------</=
SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Daryl=20
Malas</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Project Director, IP=20
Interconnects</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">w - +1 303 661=20
3302</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt"><A=20
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.com=
</A></SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV></DIV></=
DIV></DIV></BODY></HTML>

--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66CEsrvxchg_--

From richard@shockey.us  Wed Mar  3 17:38:36 2010
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From: "Richard Shockey" <richard@shockey.us>
To: "'Adam Roach'" <adam@nostrum.com>, "'Dean Willis'" <dean.willis@softarmor.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Dean ..again this is not about _THE_ DNS only the use of DNS _technology_ to
provide a specific and necessary function for inter carrier routing as Penn
points out in the GSMA case. ENUM has difficulty in actually performing some
of these functions since the ENUM query has no way to signal what is the
source of the query since in many cases you need a differentiated response
which you correctly point out is not THE DNS as we commonly understand it.

Hadriel Kaplan put out a draft some time ago on source URI that would fix
that issue,  one I fully support.

http://ietfreport.isoc.org/all-ids/draft-kaplan-enum-source-uri-00.txt 

In some cases SIP redirect actually works better since the From: field can
allow the localized directory servers ( aka IP SCP's) to formulate a
response based on the source.

The meta issue all of these drafts are all of the lovely number translation
tasks that inspired my favorite NO TCAP NO SS7 baseball caps I made upyears
ago along with NO PRI T-shirts.

If you want to replace the PSTN you have to deal with TCAP replecemenmt and
the tools we have are ENUM and SIP and they both work. This also relates to
E.164 metadata like SPID's etal that will be addressed in the E2M BOF that
you and Bernie are running and BTW yes SIP needs a means to express non
routable E.164 metadata as well.

Any ideas on a new header for that problem?  Could you do that in a 302 MOVE
as well? 

And as for TRIP ..been there heard that argument before ..no one is going to
deploy a BGP like function for this. Not going to happen no more stacks in
the proxy's we need to deal with the protocols we have not try and invent
new ones.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Adam Roach
Sent: Wednesday, March 03, 2010 4:43 PM
To: Dean Willis
Cc: 'dispatch@ietf.org'
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

On 3/3/10 1:26 PM, Dean Willis wrote:
>
> We really need a SIP-targeted analog of the BGP/OSPF route-exchange 
> protocols to solve the general problem.

So, what, like TRIP, but for things other than phone numbers?

/a
_______________________________________________
dispatch mailing list
dispatch@ietf.org
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From kcartwright@tnsi.com  Thu Mar  4 06:38:57 2010
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From: "Cartwright, Kenneth" <kcartwright@tnsi.com>
To: Dean Willis <dean.willis@softarmor.com>, "d.malas@cablelabs.com" <d.malas@cablelabs.com>
Date: Thu, 4 Mar 2010 09:38:39 -0500
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--_000_754963199212404AB8E9CFCA6C3D0CDA0D890E145FTNSMAILNAwin2_
Content-Type: text/plain; charset="us-ascii"
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I think that these discussions that focus on some macro level principles th=
at underlay the worldwide DNS system, and that seem to presume that the und=
erlying implementation of this draft will live in plain vanilla DNS/bind im=
plementations, out on the world-wide domain name system are not necessarily=
 relevant.  It's certainly true that vanilla DNS and bind, as they live in =
the world-wide DNS system, were designed primarily to return the same answe=
r to all queriers, and to use TTL to improve performance via caching, and g=
enerally have comparatively simple internal data structures to house their =
data, etc, etc.  But I do not think that is the context within which this d=
raft is required to be used or evaluated, nor is that the context within wh=
ich many many ENUM capable implementations are designed/built/used.

When you get down to it, ENUM is really just a query/response protocol for =
the resolution activities, and implementing an application that can give st=
andards based ENUM responses does not require that that implementation be B=
ind-like in its additional capabilities and design, or that the usage conte=
xt be the world-wide DNS system, or that the internal data structure just b=
e a list of domain names with a link to a set of RRs, or that TTL/caching m=
ust always play a significant role in scalability or performance, or that r=
esponses must always be the same regardless of the querier.

Ken

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Dean Willis
Sent: Wednesday, March 03, 2010 2:30 PM
To: d.malas@cablelabs.com
Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 3, 2010, at 12:01 PM, Daryl Malas wrote:



The method for SSP1's egress routes for SSP2 to be received into a "local c=
opy" is implementation specific, imo.  This being said, obviously, SSP3 and=
 SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 whe=
n they query the ENUM server.  If this is resolved in some "for queries fro=
m here give this" or "here is a local copy of peer numbers and routes plus =
*your* egress routes" is implementation specific.

And this part right here is exactly why the DNS infrastructure of ENUM is N=
OT a good solution for the problem. Unlike Einstein, DNS does not embrace r=
elativity (AKA location-context sensitivity). There are ugly hacks to make =
relativity happen in certain limited cases, but they defeat the fundamental=
 scalability mechanisms of the DNS.

--
Dean

________________________________
This e-mail message is for the sole use of the intended recipient(s)and may
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ices.
Any unauthorised review, use, disclosure or distribution is prohibited. If =
you
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<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">I think that these discussions that fo=
cus on some macro level principles that underlay the worldwide DNS system, =
and that seem to presume
 that the underlying implementation of this draft will live in plain vanill=
a DNS/bind implementations, out on the world-wide domain name system are no=
t necessarily relevant.&nbsp; It&#8217;s certainly true that vanilla DNS an=
d bind, as they live in the world-wide DNS
 system, were designed primarily to return the same answer to all queriers,=
 and to use TTL to improve performance via caching, and generally have comp=
aratively simple internal data structures to house their data, etc, etc.&nb=
sp; But I do not think that is the context
 within which this draft is required to be used or evaluated, nor is that t=
he context within which many many ENUM capable implementations are designed=
/built/used.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">When you get down to it, ENUM is reall=
y just a query/response protocol for the resolution activities, and impleme=
nting an application
 that can give standards based ENUM responses does not require that that im=
plementation be Bind-like in its additional capabilities and design, or tha=
t the usage context be the world-wide DNS system, or that the internal data=
 structure just be a list of domain
 names with a link to a set of RRs, or that TTL/caching must always play a =
significant role in scalability or performance, or that responses must alwa=
ys be the same regardless of the querier.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Ken<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
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<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span style=3D"font-size:12.0pt">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> disp=
atch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
<b><span style=3D"font-weight:bold">On Behalf Of </span></b>Dean Willis<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Wednesday, March 03, 2=
010 2:30 PM<br>
<b><span style=3D"font-weight:bold">To:</span></b> d.malas@cablelabs.com<br=
>
<b><span style=3D"font-weight:bold">Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [dispatch] I-D =
Action: draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></=
p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">On Mar 3, 2010, at 12:01 PM, Daryl Malas wrote:<o:p></o:p></span></=
font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><br>
<br>
<o:p></o:p></span></font></p>
<span style=3D"orphans: 2;text-align:auto;widows: 2;-webkit-border-horizont=
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<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:blue">The method for&nbsp;SSP1's egress rout=
es for SSP2&nbsp;to be&nbsp;received into a &quot;local copy&quot; is imple=
mentation specific, imo.&nbsp; This being said, obviously,
 SSP3 and SSP4 would not (want to) see SSP1's provisioned egress routes for=
 SSP2 when they query the ENUM server.&nbsp; If this is resolved in some &q=
uot;for queries from here give this&quot; or &quot;here is a local copy of =
peer numbers and routes plus *your* egress routes&quot; is
 implementation specific.</span></font><font size=3D"4" color=3D"black" fac=
e=3D"Helvetica"><span style=3D"font-size:13.5pt;font-family:
Helvetica;color:black"><o:p></o:p></span></font></p>
</div>
</div>
</span>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">And this part right here is exactly why the DNS infrastructure of E=
NUM is NOT a good solution for the problem. Unlike Einstein, DNS does not e=
mbrace relativity (AKA
 location-context sensitivity). There are ugly hacks to make relativity hap=
pen in certain limited cases, but they defeat the fundamental scalability m=
echanisms of the DNS.<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">--<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">Dean<o:p></o:p></span></font></p>
</div>
</div>
<br>
<hr>
<font face=3D"Arial" color=3D"Gray" size=3D"1">This e-mail message is for t=
he sole use of the intended recipient(s)and may<br>
contain confidential and privileged information of Transaction Network Serv=
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Any unauthorised review, use, disclosure or distribution is prohibited. If =
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--_000_754963199212404AB8E9CFCA6C3D0CDA0D890E145FTNSMAILNAwin2_--

From dean.willis@softarmor.com  Thu Mar  4 08:15:51 2010
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On Mar 3, 2010, at 3:42 PM, Adam Roach wrote:

> On 3/3/10 1:26 PM, Dean Willis wrote:
>>
>> We really need a SIP-targeted analog of the BGP/OSPF route-exchange  
>> protocols to solve the general problem.
>
> So, what, like TRIP, but for things other than phone numbers?
>

Yep. Imagine a TRIP extended to say "I have a SIP peering agreement  
with domain X". Or better yet, imagine something that actually works  
and is widely implemented that allows sharing such route information.

A caveat: we also have simple phone-number-peering issues, and TRIP  
hasn't caught on there. Therefore, I suspect that TRIP isn't quite  
right. A thorough post-mortem analysis might be interesting.

--
Dean

From dean.willis@softarmor.com  Thu Mar  4 08:36:06 2010
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Subject: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Richard Shockey said:

> And as for TRIP ..been there heard that argument before ..no one is  
> going to
> deploy a BGP like function for this. Not going to happen no more  
> stacks in
> the proxy's we need to deal with the protocols we have not try and  
> invent
> new ones.

I'm starting another thread from this comment, because it raises a  
really good question.

TRIP was designed to exchange phone-number reachability information  
between SIP nodes. Nobody uses it. Why?

Is the problem with TRIP is that it's yet another protocol (as Richard  
suggests above), or is the problem that it doesn't convey the right  
information, just doesn't work, or something else?

What if the route-exchange function were done in SIP, perhaps  
something like an event package for routing?

Here's a possible model using an event package for routing. One might  
subscribe to the "routing" event package at a node, resulting in being  
NOTIFY-ed with a list of route vectors being advertised by that node.

This allows some interesting things:

1) Since SUBSCRIBE provides the information to specify "who is  
asking", the NOTIFY responses could contain route-tables customized  
for the recipient. This provides the location context sensitivity  
missing from ENUM queries.

2) Route-table entries could be vectors, something like "I, proxy B,  
have a route to proxy A, which offered me a route to example.com",  
allowing for selective route-redistribution.

3) Having gotten a preliminary routing table from subscribing to a  
peer, one could then query the peer's peers by subscribing directly to  
their routing updates. This establishes SIP reachability to those  
nodes (if the SUBSCRIBE works, you can reach them with SIP, and if  
their NOTIFY requests work, they can reach you). Further, it allows  
the peer's peer to make a determination of which routes they might  
want to offer you directly, as opposed to transitively via the first  
peer.

4) Filters in the  SUBSCRIBE could be used to indicate an interest in  
a subset of routes.

5) The routing vector syntax could easily allow for both phone numbers  
and domain-names.


--
Dean


From mohamed.boucadair@orange-ftgroup.com  Thu Mar  4 08:59:43 2010
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From: <mohamed.boucadair@orange-ftgroup.com>
To: Dean Willis <dean.willis@softarmor.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Mar 2010 17:59:40 +0100
Thread-Topic: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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Dear all,

I'm wondering whether it is true to say that TRIP does not provide an answe=
r to the inter-ITAD reachability issue. Whitout going into technical detail=
s, why do we have continuous requests for TRIP ITAD identifiers?

See http://www.iana.org/assignments/trip-parameters/trip-parameters.xhtml o=
r the IANA reports.

Cheers,
Med
=20

-----Message d'origine-----
De : dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] De la par=
t de Dean Willis
Envoy=E9 : jeudi 4 mars 2010 17:36
=C0 : dispatch@ietf.org
Objet : [dispatch] Why are we TRIPped out? Is there an sip-event alternativ=
e?

Richard Shockey said:

> And as for TRIP ..been there heard that argument before ..no one is=20=20
> going to
> deploy a BGP like function for this. Not going to happen no more=20=20
> stacks in
> the proxy's we need to deal with the protocols we have not try and=20=20
> invent
> new ones.

I'm starting another thread from this comment, because it raises a=20=20
really good question.

TRIP was designed to exchange phone-number reachability information=20=20
between SIP nodes. Nobody uses it. Why?

Is the problem with TRIP is that it's yet another protocol (as Richard=20=
=20
suggests above), or is the problem that it doesn't convey the right=20=20
information, just doesn't work, or something else?

What if the route-exchange function were done in SIP, perhaps=20=20
something like an event package for routing?

Here's a possible model using an event package for routing. One might=20=20
subscribe to the "routing" event package at a node, resulting in being=20=
=20
NOTIFY-ed with a list of route vectors being advertised by that node.

This allows some interesting things:

1) Since SUBSCRIBE provides the information to specify "who is=20=20
asking", the NOTIFY responses could contain route-tables customized=20=20
for the recipient. This provides the location context sensitivity=20=20
missing from ENUM queries.

2) Route-table entries could be vectors, something like "I, proxy B,=20=20
have a route to proxy A, which offered me a route to example.com",=20=20
allowing for selective route-redistribution.

3) Having gotten a preliminary routing table from subscribing to a=20=20
peer, one could then query the peer's peers by subscribing directly to=20=
=20
their routing updates. This establishes SIP reachability to those=20=20
nodes (if the SUBSCRIBE works, you can reach them with SIP, and if=20=20
their NOTIFY requests work, they can reach you). Further, it allows=20=20
the peer's peer to make a determination of which routes they might=20=20
want to offer you directly, as opposed to transitively via the first=20=20
peer.

4) Filters in the  SUBSCRIBE could be used to indicate an interest in=20=20
a subset of routes.

5) The routing vector syntax could easily allow for both phone numbers=20=
=20
and domain-names.


--
Dean

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From: Daryl Malas <D.Malas@CableLabs.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Mar 2010 10:05:39 -0700
Thread-Topic: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Dean,

It has been a long time, since I read the RFC.  If I recall correctly, the =
biggest problem with TRIP was the issue of Number Portability.  Unlike BGP =
where there is a common rule of summarizing IP address blocks into class A'=
s or B's, you cannot do the same with E.164 addresses.  Before number porta=
bility, TRIP would work good, because you can say I have NPA-NX1 through NP=
A-NX3.  If you want to reach them go here.  Now, there are rarely any conti=
guous blocks for any service provider.  If SSPs used TRIP, they would have =
to put in every single E.164 address (e.g. NPA-NXX-XXXX) into a route table=
.  This would make route tables contain millions of lines, which would slow=
 down routing considerably...not to mention the maintenance nightmare...not=
 to mention the route convergence would be ugly.  Perhaps there could be a =
3rd party application that searches for contiguous ranges, but this too wou=
ld be maintenance heavy as TNs are ported from one SSP to another; and, the=
re is no guarantee it would significantly improve the routing database issu=
e.

Perhaps I am just way off on this, but this is my recollection.

Regards,

Daryl=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Dean Willis
Sent: Thursday, March 04, 2010 9:36 AM
To: dispatch@ietf.org
Subject: [dispatch] Why are we TRIPped out? Is there an sip-event alternati=
ve?

Richard Shockey said:

> And as for TRIP ..been there heard that argument before ..no one is=20
> going to deploy a BGP like function for this. Not going to happen no=20
> more stacks in the proxy's we need to deal with the protocols we have=20
> not try and invent new ones.

I'm starting another thread from this comment, because it raises a really g=
ood question.

TRIP was designed to exchange phone-number reachability information between=
 SIP nodes. Nobody uses it. Why?

Is the problem with TRIP is that it's yet another protocol (as Richard sugg=
ests above), or is the problem that it doesn't convey the right information=
, just doesn't work, or something else?

What if the route-exchange function were done in SIP, perhaps something lik=
e an event package for routing?

Here's a possible model using an event package for routing. One might subsc=
ribe to the "routing" event package at a node, resulting in being NOTIFY-ed=
 with a list of route vectors being advertised by that node.

This allows some interesting things:

1) Since SUBSCRIBE provides the information to specify "who is asking", the=
 NOTIFY responses could contain route-tables customized for the recipient. =
This provides the location context sensitivity missing from ENUM queries.

2) Route-table entries could be vectors, something like "I, proxy B, have a=
 route to proxy A, which offered me a route to example.com", allowing for s=
elective route-redistribution.

3) Having gotten a preliminary routing table from subscribing to a peer, on=
e could then query the peer's peers by subscribing directly to their routin=
g updates. This establishes SIP reachability to those nodes (if the SUBSCRI=
BE works, you can reach them with SIP, and if their NOTIFY requests work, t=
hey can reach you). Further, it allows the peer's peer to make a determinat=
ion of which routes they might want to offer you directly, as opposed to tr=
ansitively via the first peer.

4) Filters in the  SUBSCRIBE could be used to indicate an interest in a sub=
set of routes.

5) The routing vector syntax could easily allow for both phone numbers and =
domain-names.


--
Dean

_______________________________________________
dispatch mailing list
dispatch@ietf.org
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dwight, Timothy M (Tim)'" <timothy.dwight@verizon.com>, "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
Date: Thu, 4 Mar 2010 10:12:07 -0700
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Tim,

Personally, I think the question of how the information if uniquely conveye=
d to each individual SSP is an implementation issue.  I have seen it work i=
n different methods.  My may concern with the draft is to create uniformity=
 in how the information is conveyed to the SIP/ENUM agent.

Regards,

Daryl

________________________________
From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]
Sent: Wednesday, March 03, 2010 9:08 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Daryl,

I agree that defining a best practice often has the impact you suggest.

IMHO the technique you describe has merit.  What's not clear is how a diffe=
rent response is generated per querying domain.  The draft shows response f=
ormats that would be appropriate for SSP1, but not appropriate for any othe=
r domain.  This implies that the response would be somehow tailored for eac=
h querying domain.  I think some elaboration on how that works (e.g., what =
has to be provisioned, by whom, to make it happen) would be helpful.

tim

________________________________
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 6:02 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Tim,

Thank you for the additional context.  I think we are in sync regarding the=
 purposes of the draft as you indicate below.  Regarding placing a route he=
ader into a local non-ENUM database is obviously outside the scope of the d=
raft.  If you use a local private ENUM server to add your own route header =
(or egress route) to the NAPTR URI, is implementation specific and up to yo=
u.  Even in this implementation specific case, I believe defining a best pr=
actice for defining and including egress information in the NAPTR URI will =
provide better industry adoption and implementation consistency.  Do you ag=
ree.

Regards,

Daryl

________________________________
From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]
Sent: Wednesday, March 03, 2010 4:34 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl,

I guess our use cases differ.

The draft illustrates a NAPTR producing a URI whose host part is the ingres=
s point to SSP2's network through which SSP2 prefers to receive session req=
uests targeted at the example TN.  It then talks about mechanisms to includ=
e in the ENUM response, information that "steer" the session request messag=
e through the "egress point" from SSP1's network, desired by SSP1.

I get that.  I've done it, in a private ENUM, so I know it's possible.  It'=
s a little trickier to manage in the context of a shared registry, but stil=
l seems within the capabilities of the commercial ENUM servers with which I=
'm familiar.  Thinking of the "route header" solution, SSP1 could maintain =
in the registry a table that instructs the registry provider to append a ro=
ute header identifying SBE2 to any URI whose host part was "SBE3.ssp2.examp=
le.com".  Although this sort of detail isn't subject to standardization I a=
ssume you have something like that in mind.

This seems workable in large part because SSP1 and SSP2 will in practice kn=
ow quite a lot about their interconnection.  If SSP1 had no prior knowledge=
 of the "ingress points" (e.g., SBE3) to SSP2's networks, it would be impos=
sible to maintain such a table.  But in practice it will know these ingress=
 points; moreover it will know how its own "egress points" are connected to=
 them (e.g., via a partial mesh of IPsec tunnels).  Again I'm assuming you =
have something like this in mind.


What I meant to propose, was a different use case.  Suppose SSP2 doesn't id=
entify its ingress point in the URI resulting from the NAPTR it returns.  S=
uppose instead it returns a URI whose host part is something like "call-ser=
ver-1.ssp2.example.com".  This is the FQDN of the call server (e.g., softsw=
itch, application server) in SSP2 network that will serve calls to the exam=
ple TN.  Returning such a value avoids SSP2 having to make another routing =
decision after receiving the call from SSP1.

But if you do that, you may also want to "steer" the session request to an =
ingress point (SBE3 in the draft).  This could be done using e.g., a ROUTE =
header imbedded in the URI in the NAPTR.  The logic to determine the conten=
t of this ROUTE header is simple since it depends only on where the call is=
 going not where it came from; and easily administered since it would be sp=
ecified by the same entity that serves the example TN.

SSP1 might also want to steer the call through SBE2, which could be done vi=
a any of the mechanisms in the draft, or via some table maintained outside =
of ENUM.

The 2 approaches are AFAIK both possible with existing standards and suppor=
table by currently available products.  Were it up to me, based on my curre=
nt (probably flawed and incomplete) understanding, I'd probably do the form=
er (identification of the "ingress point" to the serving domain) in the ENU=
M database or registry, and the latter (identification of the "egress point=
" from the querying domain, as a function of the URI returned in an ENUM re=
sponse) either via some non-ENUM database, or via some "private data" in my=
 local ENUM servers (private meaning not managed by the registry service).

Anyway I appreciated and enjoyed the draft, and appreciate the technical ex=
change.

Best regards,

Tim


________________________________
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 3:45 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Tim,

SSP2 does provide their preferred "ingress point."  This information is sti=
ll in the ENUM response regardless of whether an additional egress route va=
riable has been added or not.

I am not sure I understand your second paragraph.

Regards,

Daryl

________________________________
From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]
Sent: Wednesday, March 03, 2010 12:25 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Why not let SSP2 provide their preferred "ingress point" for session reques=
ts addressed to the queried TN, in the ENUM response?  That would presumabl=
y be the same for all querying networks, and it is data for which the SSP s=
erving the queried TN (SSP2) is authoritative.

In the common case of "mated peering points" (e.g., a pair of SBCs in SSP1 =
connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a local tabl=
e mapping the remote-network ingress points received in ENUM responses, to =
its own egress points (i.e., its SBCs that are connected via IPsec to the i=
ngress point in the ENUM response).

tim

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Wednesday, March 03, 2010 12:01 PM
To: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

I think what you're describing is leaning towards implementation specific q=
uestions.  I apologize, my example below is backwards, but the principle is=
 the same.  SSP2 provisions their numbers and "routes" into the ENUM server=
.  SSP1 can see the ingress routes for SSP2 numbers.  SSP1 can then specify=
 their own egress routes in the same ENUM server.

The method for SSP1's egress routes for SSP2 to be received into a "local c=
opy" is implementation specific, imo.  This being said, obviously, SSP3 and=
 SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 whe=
n they query the ENUM server.  If this is resolved in some "for queries fro=
m here give this" or "here is a local copy of peer numbers and routes plus =
*your* egress routes" is implementation specific.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl
I was a little confused by this since I thought the call was from SSP1 to S=
SP2 and that SSP2 provisioned its numbers into the registry. If SSP1 get th=
e SSP2 data and wants to add their egress routes in their local copy I can =
see that but I wouldn't expect the resulting elaborated NAPTRs to be in the=
 Registry.
Have I got this right now?

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: Daryl Malas [mailto:D.Malas@cablelabs.com]
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0

Penn,

The use case for this application is SSP1 (or someone on their behalf) prov=
isions their numbers and relative routes into the LUF associated with their=
 federation.  SSP2 reviews SSP1's "routes" available to them in the LUF bas=
ed on what SSP1 provisioned.  They then add an egress route and associate i=
t with the ingress route SSP1 provisioned.

Let me know if this makes sense.

Regards,

Daryl

________________________________
From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0
Daryl:
I want to be sure I understand what you're proposing.  In what name server =
would the modified records be? One local to SSP1 or some general industry D=
NS? I'm trying to understand how SSP1 modifies an RR for an SSP2 number.

Thanks,

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

All,

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

Title : SIP Egress Route Use in ENUM
Author(s) : D. Malas
Filename : draft-malas-dispatch-sip-egress-route-00.txt
Pages : 11
Date : 2010-03-01

In some cases a UA (or proxy) within a Session Initiation Protocol
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy
options to reach an ingress proxy within another SSP to reach the
target UA. If the originating SSP uses ENUM to identify the ingress
proxy of the target SSP, there is currently no way for the ENUM NAPTR
response to identify a specific preferred egress proxy from the set
of multiple parallel proxies. This draft solves this problem by
defining a method for returning deterministic routing information
within an ENUM NAPTR response enabling the UA (or proxy) to choose a
preferred egress proxy. Using this information, the originating UA
now sends the SIP request through the predetermined preferred egress
proxy to reach the target SSP's proxy. This document describes
several methods to make this determination by incorporating variables
into a SIP service URI contained in the E.164 Number Mapping (ENUM)
response.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-route-0=
0.txt

Regards,

Daryl

---------------------------
Daryl Malas
Project Director, IP Interconnects
w - +1 303 661 3302
mailto:d.malas@cablelabs.com<blocked::mailto:d.malas@cablelabs.com>


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
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size=3D2 face=3DArial>Tim,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Personally, I think the question of how the informati=
on if=20
uniquely conveyed to each individual SSP is an implementation issue.&nbsp; =
I=20
have seen it work in different methods.&nbsp; My may concern with the draft=
 is=20
to create uniformity in how the information is conveyed to the SIP/ENUM=20
agent.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D482461017-04032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Dwight, Timothy M (Tim)=20
[mailto:timothy.dwight@verizon.com] <BR><B>Sent:</B> Wednesday, March 03, 2=
010=20
9:08 PM<BR><B>To:</B> Daryl Malas; PFAUTZ, PENN L (ATTCORP)<BR><B>Cc:</B>=20
dispatch@ietf.org<BR><B>Subject:</B> RE: [dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Daryl,<o=
:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">I agree =
that=20
defining a best practice often has the impact you=20
suggest.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">IMHO the=
=20
technique you describe has merit. &nbsp;What&#8217;s not clear is how a dif=
ferent=20
response is generated per querying domain. &nbsp;The draft shows response=20
formats that would be appropriate for SSP1, but not appropriate for any oth=
er=20
domain.&nbsp; This implies that the response would be somehow tailored for =
each=20
querying domain. &nbsp;I think some elaboration on how that works (e.g., wh=
at=20
has to be provisioned, by whom, to make it happen) would be=20
helpful.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">tim<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: blue 1.5pt solid; PADDING=
-BOTTOM: 0in; PADDING-LEFT: 4pt; PADDING-RIGHT: 0in; BORDER-TOP: medium non=
e; BORDER-RIGHT: medium none; PADDING-TOP: 0in">
<DIV>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Daryl=20
Malas [mailto:D.Malas@cablelabs.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 6:02=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Dwight, Timothy M=
 (Tim);=20
PFAUTZ, PENN L (ATTCORP)<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN>=
</B>=20
dispatch@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></=
B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Tim,</SPAN></FON=
T><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Thank you for th=
e=20
additional context.&nbsp; I think&nbsp;we are in sync regarding&nbsp;the=20
purposes of the draft as you indicate below.&nbsp; Regarding placing a rout=
e=20
header into a local non-ENUM database is obviously outside the scope of the=
=20
draft.&nbsp; If you use a local private ENUM server to add your own route h=
eader=20
(or egress route) to the NAPTR URI, is implementation specific and up to=20
you.&nbsp; Even in this implementation specific case, I believe defining a =
best=20
practice for defining and including egress information in the NAPTR URI wil=
l=20
provide better industry adoption and implementation consistency.&nbsp; Do y=
ou=20
agree.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Dwight,=20
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 4:34=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas; PFAU=
TZ,=20
PENN L (ATTCORP)<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B=
> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Daryl,<o=
:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">I guess =
our use=20
cases differ. &nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">The draf=
t=20
illustrates a NAPTR producing a URI whose host part is the ingress point to=
=20
SSP2&#8217;s network through which SSP2 prefers to receive session requests=
 targeted=20
at the example TN. &nbsp;It then talks about mechanisms to include in the E=
NUM=20
response, information that &#8220;steer&#8221; the session request message =
through the=20
&#8220;egress point&#8221; from SSP1&#8217;s network, desired by=20
SSP1.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">I get=20
that.&nbsp; I&#8217;ve done it, in a private ENUM, so I know it&#8217;s pos=
sible. &nbsp;It&#8217;s=20
a little trickier to manage in the context of a shared registry, but still =
seems=20
within the capabilities of the commercial ENUM servers with which I&#8217;m=
=20
familiar.&nbsp; Thinking of the &#8220;route header&#8221; solution, SSP1 c=
ould maintain in=20
the registry a table that instructs the registry provider to append a route=
=20
header identifying SBE2 to any URI whose host part was &#8220;SBE3.ssp2.exa=
mple.com&#8221;.=20
&nbsp;Although this sort of detail isn&#8217;t subject to standardization I=
 assume you=20
have something like that in mind.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">This see=
ms=20
workable in large part because SSP1 and SSP2 will in practice know quite a =
lot=20
about their interconnection. &nbsp;If SSP1 had no prior knowledge of the=20
&#8220;ingress points&#8221; (e.g., SBE3) to SSP2&#8217;s networks, it woul=
d be impossible to=20
maintain such a table. &nbsp;But in practice it will know these ingress poi=
nts;=20
moreover it will know how its own &#8220;egress points&#8221; are connected=
 to them (e.g.,=20
via a partial mesh of IPsec tunnels).&nbsp; Again I&#8217;m assuming you ha=
ve=20
something like this in mind.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">What I m=
eant to=20
propose, was a different use case.&nbsp; Suppose SSP2 doesn&#8217;t identif=
y its=20
ingress point in the URI resulting from the NAPTR it returns. &nbsp;Suppose=
=20
instead it returns a URI whose host part is something like=20
&#8220;call-server-1.ssp2.example.com&#8221;. &nbsp;This is the FQDN of the=
 call server=20
(e.g., softswitch, application server) in SSP2 network that will serve call=
s to=20
the example TN. &nbsp;Returning such a value avoids SSP2 having to make ano=
ther=20
routing decision after receiving the call from=20
SSP1.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">But if y=
ou do=20
that, you may also want to &#8220;steer&#8221; the session request to an in=
gress point (SBE3=20
in the draft). &nbsp;This could be done using e.g., a ROUTE header imbedded=
 in=20
the URI in the NAPTR.&nbsp; The logic to determine the content of this ROUT=
E=20
header is simple since it depends only on where the call is going not where=
 it=20
came from; and easily administered since it would be specified by the same=
=20
entity that serves the example TN.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">SSP1 mig=
ht also=20
want to steer the call through SBE2, which could be done via any of the=20
mechanisms in the draft, or via some table maintained outside of ENUM.=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">The 2=20
approaches are AFAIK both possible with existing standards and supportable =
by=20
currently available products. &nbsp;Were it up to me, based on my current=20
(probably flawed and incomplete) understanding, I&#8217;d probably do the f=
ormer=20
(identification of the &#8220;ingress point&#8221; to the serving domain) i=
n the ENUM=20
database or registry, and the latter (identification of the &#8220;egress p=
oint&#8221; from=20
the querying domain, as a function of the URI returned in an ENUM response)=
=20
either via some non-ENUM database, or via some &#8220;private data&#8221; i=
n my local ENUM=20
servers (private meaning not managed by the registry=20
service).<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Anyway I=
=20
appreciated and enjoyed the draft, and appreciate the technical=20
exchange.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Best=20
regards,<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Tim<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: blue 1.5pt solid; PADDING=
-BOTTOM: 0in; PADDING-LEFT: 4pt; PADDING-RIGHT: 0in; BORDER-TOP: medium non=
e; BORDER-RIGHT: medium none; PADDING-TOP: 0in">
<DIV>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Daryl=20
Malas [mailto:D.Malas@cablelabs.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 3:45=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Dwight, Timothy M=
 (Tim);=20
PFAUTZ, PENN L (ATTCORP)<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Tim,</SPAN></FON=
T><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">SSP2 does provid=
e their=20
preferred "ingress point."&nbsp; This information is still in the ENUM resp=
onse=20
regardless of whether an additional egress route variable has been added or=
=20
not.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">I am not sure I=
=20
understand your second paragraph.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Dwight,=20
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 12:2=
5=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas; PFAU=
TZ,=20
PENN L (ATTCORP)<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B=
> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">Why not =
let=20
SSP2 provide their preferred &#8220;ingress point&#8221; for session reques=
ts addressed to=20
the queried TN, in the ENUM response? &nbsp;That would presumably be the sa=
me=20
for all querying networks, and it is data for which the SSP serving the que=
ried=20
TN (SSP2) is authoritative.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">In the c=
ommon=20
case of &#8220;mated peering points&#8221; (e.g., a pair of SBCs in SSP1 co=
nnected to a pair=20
of SBCs in SSP2 via IPsec) SSP2 could keep a local table mapping the=20
remote-network ingress points received in ENUM responses, to its own egress=
=20
points (i.e., its SBCs that are connected via IPsec to the ingress point in=
 the=20
ENUM response).<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt">tim<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3D"Courier New"><SPAN=
=20
style=3D"FONT-FAMILY: 'Courier New'; COLOR: blue; FONT-SIZE: 10pt"><o:p>&nb=
sp;</o:p></SPAN></FONT></P>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: blue 1.5pt solid; PADDING=
-BOTTOM: 0in; PADDING-LEFT: 4pt; PADDING-RIGHT: 0in; BORDER-TOP: medium non=
e; BORDER-RIGHT: medium none; PADDING-TOP: 0in">
<DIV>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
>=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Daryl Malas<BR><B><SPAN=
=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 12:0=
1=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> 'PFAUTZ, PENN L=20
(ATTCORP)'; dispatch@ietf.org<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> Re: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P></DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Penn,</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">I think what you=
're=20
describing is leaning towards implementation specific questions.&nbsp; I=20
apologize, my example below is backwards, but the principle is the same.&nb=
sp;=20
SSP2 provisions their numbers and "routes" into the ENUM server.&nbsp; SSP1=
 can=20
see the ingress routes for SSP2 numbers.&nbsp; SSP1 can then specify their =
own=20
egress routes in the same ENUM server.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">The method=20
for&nbsp;SSP1's egress routes for SSP2&nbsp;to be&nbsp;received into a "loc=
al=20
copy" is implementation specific, imo.&nbsp; This being said, obviously, SS=
P3=20
and SSP4 would not (want to) see SSP1's provisioned egress routes for SSP2 =
when=20
they query the ENUM server.&nbsp; If this is resolved in some "for queries =
from=20
here give this" or "here is a local copy of peer numbers and routes plus *y=
our*=20
egress routes" is implementation specific.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR tabIndex=3D-1 align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> PFAUTZ,=20
PENN L (ATTCORP) [mailto:pp3129@att.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 10:3=
8=20
AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas;=20
dispatch@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></=
B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Daryl=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">I was a lit=
tle=20
confused by this since I thought the call was from SSP1 to SSP2 and that SS=
P2=20
provisioned its numbers into the registry. If SSP1 get the SSP2 data and wa=
nts=20
to add their egress routes in their local copy I can see that but I wouldn&=
#8217;t=20
expect the resulting elaborated NAPTRs to be in the Registry.=20
<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Have I got =
this=20
right now?<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">Penn=20
Pfautz</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">AT&amp;T Acce=
ss=20
Management</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P></DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">+1-732-420-49=
62</SPAN></FONT><FONT=20
color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p></o:p>=
</SPAN></FONT></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> Daryl=20
Malas [mailto:D.Malas@cablelabs.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Wednesday, March 03, 2010 11:2=
2=20
AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> PFAUTZ, PENN L=20
(ATTCORP); dispatch@ietf.org<BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> RE: [dispatch] I-D Action:=
=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></FONT></P></DIV>=
</DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Penn,</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">The use case for=
 this=20
application is SSP1 (or someone on their behalf) provisions their numbers a=
nd=20
relative&nbsp;routes into the LUF associated with their federation.&nbsp; S=
SP2=20
reviews&nbsp;SSP1's "routes" available to them in the LUF based on what SSP=
1=20
provisioned.&nbsp; They then add an egress route and associate it with the=
=20
ingress route SSP1 provisioned.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Let me know if t=
his=20
makes sense.</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Regards,</SPAN><=
/FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3Dblue size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: blue; FONT-SIZE: 10pt">Daryl</SPAN></FO=
NT><o:p></o:p></P>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV style=3D"TEXT-ALIGN: center" class=3DMsoNormal align=3Dcenter><FONT si=
ze=3D3=20
face=3D"Times New Roman"><SPAN style=3D"FONT-SIZE: 12pt">
<HR align=3Dcenter SIZE=3D2 width=3D"100%">
</SPAN></FONT></DIV>
<P style=3D"MARGIN-BOTTOM: 12pt" class=3DMsoNormal><B><FONT size=3D2 face=
=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
> PFAUTZ,=20
PENN L (ATTCORP) [mailto:pp3129@att.com] <BR><B><SPAN=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, March 02, 2010 12:24=
=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> Daryl Malas;=20
dispatch@ietf.org<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></=
B> RE:=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00</SPAN></FONT><o:p></o:p></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Daryl:<o:p>=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">I want to b=
e sure=20
I understand what you&#8217;re proposing.&nbsp; In what name server would t=
he modified=20
records be? One local to SSP1 or some general industry DNS? I&#8217;m tryin=
g to=20
understand how SSP1 modifies an RR for an SSP2=20
number.<o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt">Thanks,<o:p=
></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p>&nbsp;=
</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">Penn=20
Pfautz</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">AT&amp;T Acce=
ss=20
Management</SPAN></FONT><FONT color=3D#1f497d><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></FONT></P></DIV>
<P class=3DMsoNormal><FONT color=3D#1f497d size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; COLOR: #1f497d; FONT-SIZE: 10pt">+1-732-420-49=
62</SPAN></FONT><FONT=20
color=3D#1f497d size=3D2 face=3DCalibri><SPAN=20
style=3D"FONT-FAMILY: Calibri; COLOR: #1f497d; FONT-SIZE: 11pt"><o:p></o:p>=
</SPAN></FONT></P>
<DIV>
<DIV=20
style=3D"BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTT=
OM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt sol=
id; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<P class=3DMsoNormal><B><FONT size=3D2 face=3DTahoma><SPAN=20
style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt; FONT-WEIGHT: bold">From:</SP=
AN></FONT></B><FONT=20
size=3D2 face=3DTahoma><SPAN style=3D"FONT-FAMILY: Tahoma; FONT-SIZE: 10pt"=
>=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B><SPAN=20
style=3D"FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Daryl Malas<BR><B><SPAN=
=20
style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Monday, March 01, 2010 6:51=20
PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B>=20
'dispatch@ietf.org'<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN>=
</B>=20
[dispatch] I-D Action:=20
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></SPAN></FONT></P></DIV>=
</DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">All,</SPAN></FONT><o:p></o:p>=
</P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">A New Internet-Draft is avail=
able=20
from the on-line Internet-Drafts directories. <BR>&nbsp;<BR>Title : SIP Egr=
ess=20
Route Use in ENUM <BR>Author(s) : D. Malas <BR>Filename :=20
draft-malas-dispatch-sip-egress-route-00.txt <BR>Pages : 11 <BR>Date :=20
2010-03-01 <BR>&nbsp;<BR>In some cases a UA (or proxy) within a Session=20
Initiation Protocol <BR>[RFC3261] Service Provider (SSP) has multiple paral=
lel=20
egress proxy <BR>options to reach an ingress proxy within another SSP to re=
ach=20
the <BR>target UA. If the originating SSP uses ENUM to identify the ingress=
=20
<BR>proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
<BR>response to identify a specific preferred egress proxy from the set <BR=
>of=20
multiple parallel proxies. This draft solves this problem by <BR>defining a=
=20
method for returning deterministic routing information <BR>within an ENUM N=
APTR=20
response enabling the UA (or proxy) to choose a <BR>preferred egress proxy.=
=20
Using this information, the originating UA <BR>now sends the SIP request th=
rough=20
the predetermined preferred egress <BR>proxy to reach the target SSP's prox=
y.=20
This document describes <BR>several methods to make this determination by=20
incorporating variables <BR>into a SIP service URI contained in the E.164 N=
umber=20
Mapping (ENUM) <BR>response. <BR>&nbsp;<BR>A URL for this Internet-Draft is=
:=20
<BR><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress=
-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip=
-egress-route-00.txt</A>=20
</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
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<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Regards,</SPAN></FONT><o:p></=
o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Daryl</SPAN></FONT><o:p></o:p=
></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D3 face=3D"Times New Roman"><SPAN=20
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<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">---------------------------</=
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<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Daryl=20
Malas</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
style=3D"FONT-FAMILY: Arial; FONT-SIZE: 10pt">Project Director, IP=20
Interconnects</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><FONT size=3D2 face=3DArial><SPAN=20
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--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F66D1srvxchg_--

From dean.willis@softarmor.com  Thu Mar  4 09:21:07 2010
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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On Mar 4, 2010, at 11:05 AM, Daryl Malas wrote:

> Dean,
>
> It has been a long time, since I read the RFC.  If I recall  
> correctly, the biggest problem with TRIP was the issue of Number  
> Portability.  Unlike BGP where there is a common rule of summarizing  
> IP address blocks into class A's or B's, you cannot do the same with  
> E.164 addresses.  Before number portability, TRIP would work good,  
> because you can say I have NPA-NX1 through NPA-NX3.  If you want to  
> reach them go here.  Now, there are rarely any contiguous blocks for  
> any service provider.  If SSPs used TRIP, they would have to put in  
> every single E.164 address (e.g. NPA-NXX-XXXX) into a route table.   
> This would make route tables contain millions of lines, which would  
> slow down routing considerably...not to mention the maintenance  
> nightmare...not to mention the route convergence would be ugly.   
> Perhaps there could be a 3rd party application that searches for  
> contiguous ranges, but this too would be maintenance heavy as TNs  
> are ported from one SSP to another; and, there is no guarantee it  
> would significantly improve the routing database issue.
>

Ok, that's good input. Really good.

I attempt a rephrase:

Due to the sparse characteristic of number ranges resulting from LNP,  
route aggregation just doesn't work for phone numbers. TRIP is based  
on effective route aggregation; without it, the route-sets grow  
excessively large. Therefore, an ENUM style model of mapping phone- 
number to provider (effectively global, relatively static associations  
that are cacheable) works much better.

We expect to do at least one ENUM dip per phone-number in order to  
find out who the provider for that phone number is. I argue that doing  
further ENUM dips to find the routes to that provider are a Bad Idea,  
as this sort of information is arguably not well-suited to a DNS style  
database. Yes, one can hack together lots of heavily customized- 
locally-provisioned ENUM servers, but it's still a major operational  
suckage event.

Given this, there would appear to be no reasonable requirement for  
dynamic routing (BGP/OSPF/TRIP) based on telephone numbers. However,  
given the inter-provider peering model, there remains a requirement  
for dynamic routing based on peer identifiers (which I presume to be  
domain names).

So what we need is something that expresses the dynamic and shifting  
associations of inter-provider peerings and their associated weights  
and costs, and provides for disseminating this information to our SIP  
routing nodes. Further, the routing vectors involved are not always  
single-hops, but need to be able to express multiple-hop vectors.  
Conversely, the routing vectors need to be able to express aggregate  
weighting vectors while optionally suppressing remote hops, as such  
might be needed for masking remote peering agreements.

--
Dean



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Subject: Re: [dispatch] Why are we TRIPped out? Is there an	sip-event	alternative?
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Asterisk DUNDI ... they use it extensively.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of mohamed.boucadair@orange-ftgroup.com
Sent: Thursday, March 04, 2010 12:00 PM
To: Dean Willis; dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event
alternative?


Dear all,

I'm wondering whether it is true to say that TRIP does not provide an answer
to the inter-ITAD reachability issue. Whitout going into technical details,
why do we have continuous requests for TRIP ITAD identifiers?

See http://www.iana.org/assignments/trip-parameters/trip-parameters.xhtml or
the IANA reports.

Cheers,
Med


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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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+1 .. you cannot route on number blocks due to LNP. 90% of the NPA NXX 1000
blocks are already broken.

How many times to I have to remind people of that? And I don't even work for
NSR anymore. 

And BGP is too heavy weight

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Daryl Malas
Sent: Thursday, March 04, 2010 12:06 PM
To: 'Dean Willis'; dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event
alternative?

Dean,

It has been a long time, since I read the RFC.  If I recall correctly, the
biggest problem with TRIP was the issue of Number Portability.  Unlike BGP
where there is a common rule of summarizing IP address blocks into class A's
or B's, you cannot do the same with E.164 addresses.  Before number
portability, TRIP would work good, because you can say I have NPA-NX1
through NPA-NX3.  If you want to reach them go here.  Now, there are rarely
any contiguous blocks for any service provider.  If SSPs used TRIP, they
would have to put in every single E.164 address (e.g. NPA-NXX-XXXX) into a
route table.  This would make route tables contain millions of lines, which
would slow down routing considerably...not to mention the maintenance
nightmare...not to mention the route convergence would be ugly.  Perhaps
there could be a 3rd party application that searches for contiguous ranges,
but this too would be maintenance heavy as TNs are ported from one SSP to
another; and, there is no guaran
 tee it would significantly improve the routing database issue.

Perhaps I am just way off on this, but this is my recollection.

Regards,

Daryl 

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Dean Willis
Sent: Thursday, March 04, 2010 9:36 AM
To: dispatch@ietf.org
Subject: [dispatch] Why are we TRIPped out? Is there an sip-event
alternative?

Richard Shockey said:

> And as for TRIP ..been there heard that argument before ..no one is 
> going to deploy a BGP like function for this. Not going to happen no 
> more stacks in the proxy's we need to deal with the protocols we have 
> not try and invent new ones.

I'm starting another thread from this comment, because it raises a really
good question.

TRIP was designed to exchange phone-number reachability information between
SIP nodes. Nobody uses it. Why?

Is the problem with TRIP is that it's yet another protocol (as Richard
suggests above), or is the problem that it doesn't convey the right
information, just doesn't work, or something else?

What if the route-exchange function were done in SIP, perhaps something like
an event package for routing?

Here's a possible model using an event package for routing. One might
subscribe to the "routing" event package at a node, resulting in being
NOTIFY-ed with a list of route vectors being advertised by that node.

This allows some interesting things:

1) Since SUBSCRIBE provides the information to specify "who is asking", the
NOTIFY responses could contain route-tables customized for the recipient.
This provides the location context sensitivity missing from ENUM queries.

2) Route-table entries could be vectors, something like "I, proxy B, have a
route to proxy A, which offered me a route to example.com", allowing for
selective route-redistribution.

3) Having gotten a preliminary routing table from subscribing to a peer, one
could then query the peer's peers by subscribing directly to their routing
updates. This establishes SIP reachability to those nodes (if the SUBSCRIBE
works, you can reach them with SIP, and if their NOTIFY requests work, they
can reach you). Further, it allows the peer's peer to make a determination
of which routes they might want to offer you directly, as opposed to
transitively via the first peer.

4) Filters in the  SUBSCRIBE could be used to indicate an interest in a
subset of routes.

5) The routing vector syntax could easily allow for both phone numbers and
domain-names.


--
Dean

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From: "Richard Shockey" <richard@shockey.us>
To: "'Dean Willis'" <dean.willis@softarmor.com>, <dispatch@ietf.org>
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Date: Thu, 4 Mar 2010 13:05:17 -0500
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Dean Willis
Sent: Thursday, March 04, 2010 11:36 AM
To: dispatch@ietf.org
Subject: [dispatch] Why are we TRIPped out? Is there an sip-event
alternative?

Richard Shockey said:

> And as for TRIP ..been there heard that argument before ..no one is  
> going to
> deploy a BGP like function for this. Not going to happen no more  
> stacks in
> the proxy's we need to deal with the protocols we have not try and  
> invent
> new ones.

I'm starting another thread from this comment, because it raises a  
really good question.

TRIP was designed to exchange phone-number reachability information  
between SIP nodes. Nobody uses it. Why?

RS> Heard of LNP? 

Is the problem with TRIP is that it's yet another protocol (as Richard  
suggests above), or is the problem that it doesn't convey the right  
information, just doesn't work, or something else?

RS> IMHO more the former rather than the latter.

What if the route-exchange function were done in SIP, perhaps  
something like an event package for routing?

RS> Nahh.. the routing paths change you want a query mechanism that can
dynamically change targets based on both internal network topology as well
as source identification.

Here's a possible model using an event package for routing. One might  
subscribe to the "routing" event package at a node, resulting in being  
NOTIFY-ed with a list of route vectors being advertised by that node.

This allows some interesting things:

1) Since SUBSCRIBE provides the information to specify "who is  
asking", the NOTIFY responses could contain route-tables customized  
for the recipient. This provides the location context sensitivity  
missing from ENUM queries.

RS> You are on to something here which is the dynamic "who is asking" but
the response could be lots of things including a SPID or network identifier
if the target is truly PSTN and the target URI is a matter of bilateral
agreement. This is really about the function of the local proxy directory in
a carrier network. I want to simply send an invite but not get back a 302
..something else which may be some form of routing meta data. 

2) Route-table entries could be vectors, something like "I, proxy B,  
have a route to proxy A, which offered me a route to example.com",  
allowing for selective route-redistribution.

3) Having gotten a preliminary routing table from subscribing to a  
peer, one could then query the peer's peers by subscribing directly to  
their routing updates. This establishes SIP reachability to those  
nodes (if the SUBSCRIBE works, you can reach them with SIP, and if  
their NOTIFY requests work, they can reach you). Further, it allows  
the peer's peer to make a determination of which routes they might  
want to offer you directly, as opposed to transitively via the first  
peer.

4) Filters in the  SUBSCRIBE could be used to indicate an interest in  
a subset of routes.

5) The routing vector syntax could easily allow for both phone numbers  
and domain-names.

RS> maybe but gee .. ITS ALL ABOUT PHONE NUMBERS. :-) 


--
Dean

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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From: "Dwight, Timothy M (Tim)" <timothy.dwight@verizon.com>
To: Daryl Malas <D.Malas@cablelabs.com>, "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
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Daryl,

=20

I agree that defining a best practice often has the impact you suggest.

=20

IMHO the technique you describe has merit.  What's not clear is how a
different response is generated per querying domain.  The draft shows
response formats that would be appropriate for SSP1, but not appropriate
for any other domain.  This implies that the response would be somehow
tailored for each querying domain.  I think some elaboration on how that
works (e.g., what has to be provisioned, by whom, to make it happen)
would be helpful.

=20

tim

=20

________________________________

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 6:02 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Tim,

=20

Thank you for the additional context.  I think we are in sync regarding
the purposes of the draft as you indicate below.  Regarding placing a
route header into a local non-ENUM database is obviously outside the
scope of the draft.  If you use a local private ENUM server to add your
own route header (or egress route) to the NAPTR URI, is implementation
specific and up to you.  Even in this implementation specific case, I
believe defining a best practice for defining and including egress
information in the NAPTR URI will provide better industry adoption and
implementation consistency.  Do you agree.

=20

Regards,

=20

Daryl

=20

________________________________

From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]=20
Sent: Wednesday, March 03, 2010 4:34 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl,

=20

I guess our use cases differ. =20

=20

The draft illustrates a NAPTR producing a URI whose host part is the
ingress point to SSP2's network through which SSP2 prefers to receive
session requests targeted at the example TN.  It then talks about
mechanisms to include in the ENUM response, information that "steer" the
session request message through the "egress point" from SSP1's network,
desired by SSP1.

=20

I get that.  I've done it, in a private ENUM, so I know it's possible.
It's a little trickier to manage in the context of a shared registry,
but still seems within the capabilities of the commercial ENUM servers
with which I'm familiar.  Thinking of the "route header" solution, SSP1
could maintain in the registry a table that instructs the registry
provider to append a route header identifying SBE2 to any URI whose host
part was "SBE3.ssp2.example.com".  Although this sort of detail isn't
subject to standardization I assume you have something like that in
mind.

=20

This seems workable in large part because SSP1 and SSP2 will in practice
know quite a lot about their interconnection.  If SSP1 had no prior
knowledge of the "ingress points" (e.g., SBE3) to SSP2's networks, it
would be impossible to maintain such a table.  But in practice it will
know these ingress points; moreover it will know how its own "egress
points" are connected to them (e.g., via a partial mesh of IPsec
tunnels).  Again I'm assuming you have something like this in mind.

=20

=20

What I meant to propose, was a different use case.  Suppose SSP2 doesn't
identify its ingress point in the URI resulting from the NAPTR it
returns.  Suppose instead it returns a URI whose host part is something
like "call-server-1.ssp2.example.com".  This is the FQDN of the call
server (e.g., softswitch, application server) in SSP2 network that will
serve calls to the example TN.  Returning such a value avoids SSP2
having to make another routing decision after receiving the call from
SSP1.

=20

But if you do that, you may also want to "steer" the session request to
an ingress point (SBE3 in the draft).  This could be done using e.g., a
ROUTE header imbedded in the URI in the NAPTR.  The logic to determine
the content of this ROUTE header is simple since it depends only on
where the call is going not where it came from; and easily administered
since it would be specified by the same entity that serves the example
TN.

=20

SSP1 might also want to steer the call through SBE2, which could be done
via any of the mechanisms in the draft, or via some table maintained
outside of ENUM.=20

=20

The 2 approaches are AFAIK both possible with existing standards and
supportable by currently available products.  Were it up to me, based on
my current (probably flawed and incomplete) understanding, I'd probably
do the former (identification of the "ingress point" to the serving
domain) in the ENUM database or registry, and the latter (identification
of the "egress point" from the querying domain, as a function of the URI
returned in an ENUM response) either via some non-ENUM database, or via
some "private data" in my local ENUM servers (private meaning not
managed by the registry service).

=20

Anyway I appreciated and enjoyed the draft, and appreciate the technical
exchange.

=20

Best regards,

=20

Tim

=20

=20

________________________________

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 3:45 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Tim,

=20

SSP2 does provide their preferred "ingress point."  This information is
still in the ENUM response regardless of whether an additional egress
route variable has been added or not.

=20

I am not sure I understand your second paragraph.

=20

Regards,

=20

Daryl

=20

________________________________

From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]=20
Sent: Wednesday, March 03, 2010 12:25 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Why not let SSP2 provide their preferred "ingress point" for session
requests addressed to the queried TN, in the ENUM response?  That would
presumably be the same for all querying networks, and it is data for
which the SSP serving the queried TN (SSP2) is authoritative.

=20

In the common case of "mated peering points" (e.g., a pair of SBCs in
SSP1 connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a
local table mapping the remote-network ingress points received in ENUM
responses, to its own egress points (i.e., its SBCs that are connected
via IPsec to the ingress point in the ENUM response).

=20

tim

=20

________________________________

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Wednesday, March 03, 2010 12:01 PM
To: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

I think what you're describing is leaning towards implementation
specific questions.  I apologize, my example below is backwards, but the
principle is the same.  SSP2 provisions their numbers and "routes" into
the ENUM server.  SSP1 can see the ingress routes for SSP2 numbers.
SSP1 can then specify their own egress routes in the same ENUM server.

=20

The method for SSP1's egress routes for SSP2 to be received into a
"local copy" is implementation specific, imo.  This being said,
obviously, SSP3 and SSP4 would not (want to) see SSP1's provisioned
egress routes for SSP2 when they query the ENUM server.  If this is
resolved in some "for queries from here give this" or "here is a local
copy of peer numbers and routes plus *your* egress routes" is
implementation specific.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl=20

I was a little confused by this since I thought the call was from SSP1
to SSP2 and that SSP2 provisioned its numbers into the registry. If SSP1
get the SSP2 data and wants to add their egress routes in their local
copy I can see that but I wouldn't expect the resulting elaborated
NAPTRs to be in the Registry.=20

Have I got this right now?

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

The use case for this application is SSP1 (or someone on their behalf)
provisions their numbers and relative routes into the LUF associated
with their federation.  SSP2 reviews SSP1's "routes" available to them
in the LUF based on what SSP1 provisioned.  They then add an egress
route and associate it with the ingress route SSP1 provisioned.

=20

Let me know if this makes sense.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl:

I want to be sure I understand what you're proposing.  In what name
server would the modified records be? One local to SSP1 or some general
industry DNS? I'm trying to understand how SSP1 modifies an RR for an
SSP2 number.

=20

Thanks,

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

=20

All,

=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.=20
=20
Title : SIP Egress Route Use in ENUM=20
Author(s) : D. Malas=20
Filename : draft-malas-dispatch-sip-egress-route-00.txt=20
Pages : 11=20
Date : 2010-03-01=20
=20
In some cases a UA (or proxy) within a Session Initiation Protocol=20
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy=20
options to reach an ingress proxy within another SSP to reach the=20
target UA. If the originating SSP uses ENUM to identify the ingress=20
proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
response to identify a specific preferred egress proxy from the set=20
of multiple parallel proxies. This draft solves this problem by=20
defining a method for returning deterministic routing information=20
within an ENUM NAPTR response enabling the UA (or proxy) to choose a=20
preferred egress proxy. Using this information, the originating UA=20
now sends the SIP request through the predetermined preferred egress=20
proxy to reach the target SSP's proxy. This document describes=20
several methods to make this determination by incorporating variables=20
into a SIP service URI contained in the E.164 Number Mapping (ENUM)=20
response.=20
=20
A URL for this Internet-Draft is:=20
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-rout
e-00.txt=20

=20

Regards,

=20

Daryl

=20

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

Daryl Malas

Project Director, IP Interconnects

w - +1 303 661 3302

mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>=20

=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Daryl,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I agree =
that
defining a best practice often has the impact you =
suggest.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>IMHO the
technique you describe has merit. &nbsp;What&#8217;s not clear is how a =
different
response is generated per querying domain. &nbsp;The draft shows =
response formats
that would be appropriate for SSP1, but not appropriate for any other =
domain.&nbsp; This
implies that the response would be somehow tailored for each querying =
domain. &nbsp;I
think some elaboration on how that works (e.g., what has to be =
provisioned, by
whom, to make it happen) would be helpful.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
6:02 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dwight, Timothy M =
(Tim);
PFAUTZ, PENN L (ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Tim,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Thank you for the additional
context.&nbsp; I think&nbsp;we are in sync regarding&nbsp;the purposes =
of the
draft as you indicate below.&nbsp; Regarding placing a route header into =
a
local non-ENUM database is obviously outside the scope of the =
draft.&nbsp; If
you use a local private ENUM server to add your own route header (or =
egress
route) to the NAPTR URI, is implementation specific and up to you.&nbsp; =
Even
in this implementation specific case, I believe defining a best practice =
for
defining and including egress information in the NAPTR URI will provide =
better
industry adoption and implementation consistency.&nbsp; Do you =
agree.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Dwight,
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
4:34 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; PFAUTZ, =
PENN L
(ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Daryl,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I guess =
our use
cases differ. &nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>The =
draft
illustrates a NAPTR producing a URI whose host part is the ingress point =
to
SSP2&#8217;s network through which SSP2 prefers to receive session =
requests
targeted at the example TN. &nbsp;It then talks about mechanisms to =
include in
the ENUM response, information that &#8220;steer&#8221; the session =
request
message through the &#8220;egress point&#8221; from SSP1&#8217;s =
network,
desired by SSP1.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I get =
that.&nbsp;
I&#8217;ve done it, in a private ENUM, so I know it&#8217;s possible.
&nbsp;It&#8217;s a little trickier to manage in the context of a shared
registry, but still seems within the capabilities of the commercial ENUM
servers with which I&#8217;m familiar.&nbsp; Thinking of the =
&#8220;route
header&#8221; solution, SSP1 could maintain in the registry a table that
instructs the registry provider to append a route header identifying =
SBE2 to
any URI whose host part was &#8220;SBE3.ssp2.example.com&#8221;. =
&nbsp;Although
this sort of detail isn&#8217;t subject to standardization I assume you =
have
something like that in mind.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>This =
seems
workable in large part because SSP1 and SSP2 will in practice know quite =
a lot
about their interconnection. &nbsp;If SSP1 had no prior knowledge of the
&#8220;ingress points&#8221; (e.g., SBE3) to SSP2&#8217;s networks, it =
would be
impossible to maintain such a table. &nbsp;But in practice it will know =
these
ingress points; moreover it will know how its own &#8220;egress =
points&#8221;
are connected to them (e.g., via a partial mesh of IPsec tunnels).&nbsp; =
Again
I&#8217;m assuming you have something like this in =
mind.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>What I =
meant to
propose, was a different use case.&nbsp; Suppose SSP2 doesn&#8217;t =
identify
its ingress point in the URI resulting from the NAPTR it returns. =
&nbsp;Suppose
instead it returns a URI whose host part is something like
&#8220;call-server-1.ssp2.example.com&#8221;. &nbsp;This is the FQDN of =
the
call server (e.g., softswitch, application server) in SSP2 network that =
will
serve calls to the example TN. &nbsp;Returning such a value avoids SSP2 =
having
to make another routing decision after receiving the call from =
SSP1.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>But if =
you do
that, you may also want to &#8220;steer&#8221; the session request to an
ingress point (SBE3 in the draft). &nbsp;This could be done using e.g., =
a ROUTE
header imbedded in the URI in the NAPTR.&nbsp; The logic to determine =
the
content of this ROUTE header is simple since it depends only on where =
the call
is going not where it came from; and easily administered since it would =
be
specified by the same entity that serves the example =
TN.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>SSP1 =
might also
want to steer the call through SBE2, which could be done via any of the
mechanisms in the draft, or via some table maintained outside of ENUM. =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>The 2 =
approaches
are AFAIK both possible with existing standards and supportable by =
currently
available products. &nbsp;Were it up to me, based on my current =
(probably
flawed and incomplete) understanding, I&#8217;d probably do the former
(identification of the &#8220;ingress point&#8221; to the serving =
domain) in
the ENUM database or registry, and the latter (identification of the
&#8220;egress point&#8221; from the querying domain, as a function of =
the URI
returned in an ENUM response) either via some non-ENUM database, or via =
some
&#8220;private data&#8221; in my local ENUM servers (private meaning not
managed by the registry service).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Anyway I
appreciated and enjoyed the draft, and appreciate the technical =
exchange.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Best =
regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
3:45 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dwight, Timothy M =
(Tim);
PFAUTZ, PENN L (ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Tim,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>SSP2 does provide their preferred
&quot;ingress point.&quot;&nbsp; This information is still in the ENUM =
response
regardless of whether an additional egress route variable has been added =
or
not.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>I am not sure I understand your =
second
paragraph.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Dwight,
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
12:25 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; PFAUTZ, =
PENN L
(ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Why not =
let SSP2
provide their preferred &#8220;ingress point&#8221; for session requests
addressed to the queried TN, in the ENUM response? &nbsp;That would =
presumably
be the same for all querying networks, and it is data for which the SSP =
serving
the queried TN (SSP2) is authoritative.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>In the =
common
case of &#8220;mated peering points&#8221; (e.g., a pair of SBCs in SSP1
connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a local =
table
mapping the remote-network ingress points received in ENUM responses, to =
its
own egress points (i.e., its SBCs that are connected via IPsec to the =
ingress
point in the ENUM response).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Daryl Malas<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
12:01 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> 'PFAUTZ, PENN L =
(ATTCORP)';
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Penn,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>I think what you're describing is =
leaning
towards implementation specific questions.&nbsp; I apologize, my example =
below
is backwards, but the principle is the same.&nbsp; SSP2 provisions their
numbers and &quot;routes&quot; into the ENUM server.&nbsp; SSP1 can see =
the
ingress routes for SSP2 numbers.&nbsp; SSP1 can then specify their own =
egress
routes in the same ENUM server.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The method for&nbsp;SSP1's egress =
routes
for SSP2&nbsp;to be&nbsp;received into a &quot;local copy&quot; is
implementation specific, imo.&nbsp; This being said, obviously, SSP3 and =
SSP4
would not (want to) see SSP1's provisioned egress routes for SSP2 when =
they
query the ENUM server.&nbsp; If this is resolved in some &quot;for =
queries from
here give this&quot; or &quot;here is a local copy of peer numbers and =
routes
plus *your* egress routes&quot; is implementation =
specific.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> PFAUTZ,
PENN L (ATTCORP) [mailto:pp3129@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
10:38 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; =
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Daryl =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>I was a =
little
confused by this since I thought the call was from SSP1 to SSP2 and that =
SSP2
provisioned its numbers into the registry. If SSP1 get the SSP2 data and =
wants
to add their egress routes in their local copy I can see that but I
wouldn&#8217;t expect the resulting elaborated NAPTRs to be in the =
Registry. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Have I got =
this
right now?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>Penn =
Pfautz</span></font><font
color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>AT&amp;T =
Access
Management</span></font><font color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>+1-732-420-496=
2</span></font><font
size=3D2 color=3D"#1f497d" face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:
Calibri;color:#1F497D'><o:p></o:p></span></font></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
11:22 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> PFAUTZ, PENN L =
(ATTCORP);
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></font></p>

</div>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Penn,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The use case for this application =
is SSP1
(or someone on their behalf) provisions their numbers and =
relative&nbsp;routes
into the LUF associated with their federation.&nbsp; SSP2 =
reviews&nbsp;SSP1's
&quot;routes&quot; available to them in the LUF based on what SSP1
provisioned.&nbsp; They then add an egress route and associate it with =
the
ingress route SSP1 provisioned.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Let me know if this makes =
sense.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> PFAUTZ,
PENN L (ATTCORP) [mailto:pp3129@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, March 02, =
2010
12:24 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; =
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Daryl:<o:p><=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>I want to =
be sure I
understand what you&#8217;re proposing.&nbsp; In what name server would =
the
modified records be? One local to SSP1 or some general industry DNS? =
I&#8217;m
trying to understand how SSP1 modifies an RR for an SSP2 =
number.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Thanks,<o:p>=
</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>Penn =
Pfautz</span></font><font
color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>AT&amp;T =
Access
Management</span></font><font color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>+1-732-420-496=
2</span></font><font
size=3D2 color=3D"#1f497d" face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:
Calibri;color:#1F497D'><o:p></o:p></span></font></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b><span =
style=3D'font-weight:bold'>On Behalf
Of </span></b>Daryl Malas<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, March 01, =
2010 6:51
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> =
'dispatch@ietf.org'<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [dispatch] I-D =
Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></font></p>

</div>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>All,</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>A New Internet-Draft is available from the on-line
Internet-Drafts directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR =
<br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a =
<br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress =
<br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables =
<br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch=
-sip-egress-route-00.txt</a>
</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Regards,</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Daryl</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>---------------------------</span></font><o:p></o:p></=
p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Daryl Malas</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Project Director, IP =
Interconnects</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>w - +1 303 661 3302</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><a =
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.c=
om</a></span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

</div>

</div>

</div>

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From: "Dwight, Timothy M (Tim)" <timothy.dwight@verizon.com>
To: Daryl Malas <D.Malas@cablelabs.com>, "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
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Daryl,

=20

Maybe it would be sufficient to know that there is at least one way it
could be done.  That way need not be recommended let alone mandatory.
Just a demonstration that there is some feasible way to construct the
black box that would produce the externally visible behavior the draft
describes.  That would at least address skepticism about whether it
could be built.

=20

Ideally though I think we'd want to know more about the applicability of
the mechanism; which could either be discussed explicitly or revealed
implicitly by the nature of the "straw man" implementation.

=20

Tim

=20

=20

________________________________

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Thursday, March 04, 2010 11:12 AM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Tim,

=20

Personally, I think the question of how the information if uniquely
conveyed to each individual SSP is an implementation issue.  I have seen
it work in different methods.  My may concern with the draft is to
create uniformity in how the information is conveyed to the SIP/ENUM
agent.

=20

Regards,

=20

Daryl

=20

________________________________

From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]=20
Sent: Wednesday, March 03, 2010 9:08 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl,

=20

I agree that defining a best practice often has the impact you suggest.

=20

IMHO the technique you describe has merit.  What's not clear is how a
different response is generated per querying domain.  The draft shows
response formats that would be appropriate for SSP1, but not appropriate
for any other domain.  This implies that the response would be somehow
tailored for each querying domain.  I think some elaboration on how that
works (e.g., what has to be provisioned, by whom, to make it happen)
would be helpful.

=20

tim

=20

________________________________

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 6:02 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Cc: dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Tim,

=20

Thank you for the additional context.  I think we are in sync regarding
the purposes of the draft as you indicate below.  Regarding placing a
route header into a local non-ENUM database is obviously outside the
scope of the draft.  If you use a local private ENUM server to add your
own route header (or egress route) to the NAPTR URI, is implementation
specific and up to you.  Even in this implementation specific case, I
believe defining a best practice for defining and including egress
information in the NAPTR URI will provide better industry adoption and
implementation consistency.  Do you agree.

=20

Regards,

=20

Daryl

=20

________________________________

From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]=20
Sent: Wednesday, March 03, 2010 4:34 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl,

=20

I guess our use cases differ. =20

=20

The draft illustrates a NAPTR producing a URI whose host part is the
ingress point to SSP2's network through which SSP2 prefers to receive
session requests targeted at the example TN.  It then talks about
mechanisms to include in the ENUM response, information that "steer" the
session request message through the "egress point" from SSP1's network,
desired by SSP1.

=20

I get that.  I've done it, in a private ENUM, so I know it's possible.
It's a little trickier to manage in the context of a shared registry,
but still seems within the capabilities of the commercial ENUM servers
with which I'm familiar.  Thinking of the "route header" solution, SSP1
could maintain in the registry a table that instructs the registry
provider to append a route header identifying SBE2 to any URI whose host
part was "SBE3.ssp2.example.com".  Although this sort of detail isn't
subject to standardization I assume you have something like that in
mind.

=20

This seems workable in large part because SSP1 and SSP2 will in practice
know quite a lot about their interconnection.  If SSP1 had no prior
knowledge of the "ingress points" (e.g., SBE3) to SSP2's networks, it
would be impossible to maintain such a table.  But in practice it will
know these ingress points; moreover it will know how its own "egress
points" are connected to them (e.g., via a partial mesh of IPsec
tunnels).  Again I'm assuming you have something like this in mind.

=20

=20

What I meant to propose, was a different use case.  Suppose SSP2 doesn't
identify its ingress point in the URI resulting from the NAPTR it
returns.  Suppose instead it returns a URI whose host part is something
like "call-server-1.ssp2.example.com".  This is the FQDN of the call
server (e.g., softswitch, application server) in SSP2 network that will
serve calls to the example TN.  Returning such a value avoids SSP2
having to make another routing decision after receiving the call from
SSP1.

=20

But if you do that, you may also want to "steer" the session request to
an ingress point (SBE3 in the draft).  This could be done using e.g., a
ROUTE header imbedded in the URI in the NAPTR.  The logic to determine
the content of this ROUTE header is simple since it depends only on
where the call is going not where it came from; and easily administered
since it would be specified by the same entity that serves the example
TN.

=20

SSP1 might also want to steer the call through SBE2, which could be done
via any of the mechanisms in the draft, or via some table maintained
outside of ENUM.=20

=20

The 2 approaches are AFAIK both possible with existing standards and
supportable by currently available products.  Were it up to me, based on
my current (probably flawed and incomplete) understanding, I'd probably
do the former (identification of the "ingress point" to the serving
domain) in the ENUM database or registry, and the latter (identification
of the "egress point" from the querying domain, as a function of the URI
returned in an ENUM response) either via some non-ENUM database, or via
some "private data" in my local ENUM servers (private meaning not
managed by the registry service).

=20

Anyway I appreciated and enjoyed the draft, and appreciate the technical
exchange.

=20

Best regards,

=20

Tim

=20

=20

________________________________

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 3:45 PM
To: Dwight, Timothy M (Tim); PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Tim,

=20

SSP2 does provide their preferred "ingress point."  This information is
still in the ENUM response regardless of whether an additional egress
route variable has been added or not.

=20

I am not sure I understand your second paragraph.

=20

Regards,

=20

Daryl

=20

________________________________

From: Dwight, Timothy M (Tim) [mailto:timothy.dwight@verizon.com]=20
Sent: Wednesday, March 03, 2010 12:25 PM
To: Daryl Malas; PFAUTZ, PENN L (ATTCORP)
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Why not let SSP2 provide their preferred "ingress point" for session
requests addressed to the queried TN, in the ENUM response?  That would
presumably be the same for all querying networks, and it is data for
which the SSP serving the queried TN (SSP2) is authoritative.

=20

In the common case of "mated peering points" (e.g., a pair of SBCs in
SSP1 connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a
local table mapping the remote-network ingress points received in ENUM
responses, to its own egress points (i.e., its SBCs that are connected
via IPsec to the ingress point in the ENUM response).

=20

tim

=20

________________________________

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Wednesday, March 03, 2010 12:01 PM
To: 'PFAUTZ, PENN L (ATTCORP)'; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

I think what you're describing is leaning towards implementation
specific questions.  I apologize, my example below is backwards, but the
principle is the same.  SSP2 provisions their numbers and "routes" into
the ENUM server.  SSP1 can see the ingress routes for SSP2 numbers.
SSP1 can then specify their own egress routes in the same ENUM server.

=20

The method for SSP1's egress routes for SSP2 to be received into a
"local copy" is implementation specific, imo.  This being said,
obviously, SSP3 and SSP4 would not (want to) see SSP1's provisioned
egress routes for SSP2 when they query the ENUM server.  If this is
resolved in some "for queries from here give this" or "here is a local
copy of peer numbers and routes plus *your* egress routes" is
implementation specific.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Wednesday, March 03, 2010 10:38 AM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl=20

I was a little confused by this since I thought the call was from SSP1
to SSP2 and that SSP2 provisioned its numbers into the registry. If SSP1
get the SSP2 data and wants to add their egress routes in their local
copy I can see that but I wouldn't expect the resulting elaborated
NAPTRs to be in the Registry.=20

Have I got this right now?

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: Daryl Malas [mailto:D.Malas@cablelabs.com]=20
Sent: Wednesday, March 03, 2010 11:22 AM
To: PFAUTZ, PENN L (ATTCORP); dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

=20

Penn,

=20

The use case for this application is SSP1 (or someone on their behalf)
provisions their numbers and relative routes into the LUF associated
with their federation.  SSP2 reviews SSP1's "routes" available to them
in the LUF based on what SSP1 provisioned.  They then add an egress
route and associate it with the ingress route SSP1 provisioned.

=20

Let me know if this makes sense.

=20

Regards,

=20

Daryl

=20

________________________________

From: PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]=20
Sent: Tuesday, March 02, 2010 12:24 PM
To: Daryl Malas; dispatch@ietf.org
Subject: RE: [dispatch] I-D Action:
draft-malas-dispatch-sip-egress-route-00

Daryl:

I want to be sure I understand what you're proposing.  In what name
server would the modified records be? One local to SSP1 or some general
industry DNS? I'm trying to understand how SSP1 modifies an RR for an
SSP2 number.

=20

Thanks,

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Daryl Malas
Sent: Monday, March 01, 2010 6:51 PM
To: 'dispatch@ietf.org'
Subject: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

=20

All,

=20

A New Internet-Draft is available from the on-line Internet-Drafts
directories.=20
=20
Title : SIP Egress Route Use in ENUM=20
Author(s) : D. Malas=20
Filename : draft-malas-dispatch-sip-egress-route-00.txt=20
Pages : 11=20
Date : 2010-03-01=20
=20
In some cases a UA (or proxy) within a Session Initiation Protocol=20
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy=20
options to reach an ingress proxy within another SSP to reach the=20
target UA. If the originating SSP uses ENUM to identify the ingress=20
proxy of the target SSP, there is currently no way for the ENUM NAPTR=20
response to identify a specific preferred egress proxy from the set=20
of multiple parallel proxies. This draft solves this problem by=20
defining a method for returning deterministic routing information=20
within an ENUM NAPTR response enabling the UA (or proxy) to choose a=20
preferred egress proxy. Using this information, the originating UA=20
now sends the SIP request through the predetermined preferred egress=20
proxy to reach the target SSP's proxy. This document describes=20
several methods to make this determination by incorporating variables=20
into a SIP service URI contained in the E.164 Number Mapping (ENUM)=20
response.=20
=20
A URL for this Internet-Draft is:=20
http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egress-rout
e-00.txt=20

=20

Regards,

=20

Daryl

=20

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

Daryl Malas

Project Director, IP Interconnects

w - +1 303 661 3302

mailto:d.malas@cablelabs.com <blocked::mailto:d.malas@cablelabs.com>=20

=20


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New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Maybe it =
would be
sufficient to know that there is at least one way it could be =
done.&nbsp; That way need
not be recommended let alone mandatory.&nbsp; Just a demonstration that =
there is
some feasible way to construct the black box that would produce the =
externally
visible behavior the draft describes.&nbsp; That would at least address =
skepticism about
whether it could be built.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Ideally =
though I
think we&#8217;d want to know more about the applicability of the =
mechanism; which
could either be discussed explicitly or revealed implicitly by the =
nature of
the &#8220;straw man&#8221; implementation.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Thursday, March 04, =
2010
11:12 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dwight, Timothy M =
(Tim);
PFAUTZ, PENN L (ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Tim,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Personally, I think the question of =
how
the information if uniquely conveyed to each individual SSP is an
implementation issue.&nbsp; I have seen it work in different =
methods.&nbsp; My
may concern with the draft is to create uniformity in how the =
information is
conveyed to the SIP/ENUM agent.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Dwight,
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
9:08 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; PFAUTZ, =
PENN L
(ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Daryl,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I agree =
that
defining a best practice often has the impact you =
suggest.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>IMHO the
technique you describe has merit. &nbsp;What&#8217;s not clear is how a
different response is generated per querying domain. &nbsp;The draft =
shows
response formats that would be appropriate for SSP1, but not appropriate =
for
any other domain.&nbsp; This implies that the response would be somehow
tailored for each querying domain. &nbsp;I think some elaboration on how =
that
works (e.g., what has to be provisioned, by whom, to make it happen) =
would be
helpful.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
6:02 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dwight, Timothy M =
(Tim);
PFAUTZ, PENN L (ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Tim,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Thank you for the additional
context.&nbsp; I think&nbsp;we are in sync regarding&nbsp;the purposes =
of the
draft as you indicate below.&nbsp; Regarding placing a route header into =
a
local non-ENUM database is obviously outside the scope of the =
draft.&nbsp; If
you use a local private ENUM server to add your own route header (or =
egress
route) to the NAPTR URI, is implementation specific and up to you.&nbsp; =
Even
in this implementation specific case, I believe defining a best practice =
for
defining and including egress information in the NAPTR URI will provide =
better
industry adoption and implementation consistency.&nbsp; Do you =
agree.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Dwight, Timothy
M (Tim) [mailto:timothy.dwight@verizon.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
4:34 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; PFAUTZ, =
PENN L
(ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Daryl,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I guess =
our use
cases differ. &nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>The =
draft
illustrates a NAPTR producing a URI whose host part is the ingress point =
to
SSP2&#8217;s network through which SSP2 prefers to receive session =
requests
targeted at the example TN. &nbsp;It then talks about mechanisms to =
include in
the ENUM response, information that &#8220;steer&#8221; the session =
request
message through the &#8220;egress point&#8221; from SSP1&#8217;s =
network,
desired by SSP1.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>I get =
that.&nbsp;
I&#8217;ve done it, in a private ENUM, so I know it&#8217;s possible.
&nbsp;It&#8217;s a little trickier to manage in the context of a shared
registry, but still seems within the capabilities of the commercial ENUM
servers with which I&#8217;m familiar.&nbsp; Thinking of the =
&#8220;route
header&#8221; solution, SSP1 could maintain in the registry a table that
instructs the registry provider to append a route header identifying =
SBE2 to
any URI whose host part was &#8220;SBE3.ssp2.example.com&#8221;. =
&nbsp;Although
this sort of detail isn&#8217;t subject to standardization I assume you =
have
something like that in mind.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>This =
seems
workable in large part because SSP1 and SSP2 will in practice know quite =
a lot
about their interconnection. &nbsp;If SSP1 had no prior knowledge of the
&#8220;ingress points&#8221; (e.g., SBE3) to SSP2&#8217;s networks, it =
would be
impossible to maintain such a table. &nbsp;But in practice it will know =
these
ingress points; moreover it will know how its own &#8220;egress =
points&#8221;
are connected to them (e.g., via a partial mesh of IPsec tunnels).&nbsp; =
Again
I&#8217;m assuming you have something like this in =
mind.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>What I =
meant to
propose, was a different use case.&nbsp; Suppose SSP2 doesn&#8217;t =
identify
its ingress point in the URI resulting from the NAPTR it returns. =
&nbsp;Suppose
instead it returns a URI whose host part is something like
&#8220;call-server-1.ssp2.example.com&#8221;. &nbsp;This is the FQDN of =
the
call server (e.g., softswitch, application server) in SSP2 network that =
will
serve calls to the example TN. &nbsp;Returning such a value avoids SSP2 =
having
to make another routing decision after receiving the call from =
SSP1.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>But if =
you do
that, you may also want to &#8220;steer&#8221; the session request to an
ingress point (SBE3 in the draft). &nbsp;This could be done using e.g., =
a ROUTE
header imbedded in the URI in the NAPTR.&nbsp; The logic to determine =
the
content of this ROUTE header is simple since it depends only on where =
the call
is going not where it came from; and easily administered since it would =
be
specified by the same entity that serves the example =
TN.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>SSP1 =
might also
want to steer the call through SBE2, which could be done via any of the
mechanisms in the draft, or via some table maintained outside of ENUM. =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>The 2 =
approaches
are AFAIK both possible with existing standards and supportable by =
currently
available products. &nbsp;Were it up to me, based on my current =
(probably
flawed and incomplete) understanding, I&#8217;d probably do the former
(identification of the &#8220;ingress point&#8221; to the serving =
domain) in
the ENUM database or registry, and the latter (identification of the
&#8220;egress point&#8221; from the querying domain, as a function of =
the URI
returned in an ENUM response) either via some non-ENUM database, or via =
some
&#8220;private data&#8221; in my local ENUM servers (private meaning not
managed by the registry service).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Anyway I
appreciated and enjoyed the draft, and appreciate the technical =
exchange.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Best =
regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>Tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
3:45 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dwight, Timothy M =
(Tim);
PFAUTZ, PENN L (ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Tim,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>SSP2 does provide their preferred
&quot;ingress point.&quot;&nbsp; This information is still in the ENUM =
response
regardless of whether an additional egress route variable has been added =
or
not.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>I am not sure I understand your =
second
paragraph.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> Dwight,
Timothy M (Tim) [mailto:timothy.dwight@verizon.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
12:25 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; PFAUTZ, =
PENN L
(ATTCORP)<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>Why not =
let SSP2
provide their preferred &#8220;ingress point&#8221; for session requests
addressed to the queried TN, in the ENUM response? &nbsp;That would =
presumably
be the same for all querying networks, and it is data for which the SSP =
serving
the queried TN (SSP2) is authoritative.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier New";color:blue'>In the =
common
case of &#8220;mated peering points&#8221; (e.g., a pair of SBCs in SSP1
connected to a pair of SBCs in SSP2 via IPsec) SSP2 could keep a local =
table
mapping the remote-network ingress points received in ENUM responses, to =
its
own egress points (i.e., its SBCs that are connected via IPsec to the =
ingress
point in the ENUM response).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'>tim<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3D"Courier =
New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New";color:blue'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b><span =
style=3D'font-weight:bold'>On Behalf
Of </span></b>Daryl Malas<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
12:01 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> 'PFAUTZ, PENN L =
(ATTCORP)';
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Penn,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>I think what you're describing is =
leaning
towards implementation specific questions.&nbsp; I apologize, my example =
below
is backwards, but the principle is the same.&nbsp; SSP2 provisions their
numbers and &quot;routes&quot; into the ENUM server.&nbsp; SSP1 can see =
the
ingress routes for SSP2 numbers.&nbsp; SSP1 can then specify their own =
egress
routes in the same ENUM server.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The method for&nbsp;SSP1's egress =
routes
for SSP2&nbsp;to be&nbsp;received into a &quot;local copy&quot; is
implementation specific, imo.&nbsp; This being said, obviously, SSP3 and =
SSP4
would not (want to) see SSP1's provisioned egress routes for SSP2 when =
they
query the ENUM server.&nbsp; If this is resolved in some &quot;for =
queries from
here give this&quot; or &quot;here is a local copy of peer numbers and =
routes
plus *your* egress routes&quot; is implementation =
specific.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabIndex=3D-1>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> PFAUTZ,
PENN L (ATTCORP) [mailto:pp3129@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
10:38 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; =
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Daryl =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>I was a =
little
confused by this since I thought the call was from SSP1 to SSP2 and that =
SSP2
provisioned its numbers into the registry. If SSP1 get the SSP2 data and =
wants
to add their egress routes in their local copy I can see that but I
wouldn&#8217;t expect the resulting elaborated NAPTRs to be in the =
Registry. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Have I got =
this
right now?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>Penn =
Pfautz</span></font><font
color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>AT&amp;T =
Access
Management</span></font><font color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>+1-732-420-496=
2</span></font><font
size=3D2 color=3D"#1f497d" face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:
Calibri;color:#1F497D'><o:p></o:p></span></font></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Daryl =
Malas
[mailto:D.Malas@cablelabs.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March =
03, 2010
11:22 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> PFAUTZ, PENN L =
(ATTCORP);
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></font></p>

</div>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Penn,</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>The use case for this application =
is SSP1
(or someone on their behalf) provisions their numbers and =
relative&nbsp;routes
into the LUF associated with their federation.&nbsp; SSP2 =
reviews&nbsp;SSP1's
&quot;routes&quot; available to them in the LUF based on what SSP1
provisioned.&nbsp; They then add an egress route and associate it with =
the
ingress route SSP1 provisioned.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Let me know if this makes =
sense.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Regards,</span></font><o:p></o:p></p=
>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblue face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:blue'>Daryl</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</spa=
n></font></b><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'> PFAUTZ,
PENN L (ATTCORP) [mailto:pp3129@att.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, March 02, =
2010
12:24 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Daryl Malas; =
dispatch@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> RE: [dispatch] =
I-D
Action: =
draft-malas-dispatch-sip-egress-route-00</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Daryl:<o:p><=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>I want to =
be sure I
understand what you&#8217;re proposing.&nbsp; In what name server would =
the
modified records be? One local to SSP1 or some general industry DNS? =
I&#8217;m
trying to understand how SSP1 modifies an RR for an SSP2 =
number.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>Thanks,<o:p>=
</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>Penn =
Pfautz</span></font><font
color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>AT&amp;T =
Access
Management</span></font><font color=3D"#1f497d"><span =
style=3D'color:#1F497D'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#1F497D'>+1-732-420-496=
2</span></font><font
size=3D2 color=3D"#1f497d" face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:
Calibri;color:#1F497D'><o:p></o:p></span></font></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b><span =
style=3D'font-weight:bold'>On Behalf
Of </span></b>Daryl Malas<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, March 01, =
2010 6:51
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> =
'dispatch@ietf.org'<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [dispatch] I-D =
Action:
draft-malas-dispatch-sip-egress-route-00<o:p></o:p></span></font></p>

</div>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>All,</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>A New Internet-Draft is available from the on-line
Internet-Drafts directories. <br>
&nbsp;<br>
Title : SIP Egress Route Use in ENUM <br>
Author(s) : D. Malas <br>
Filename : draft-malas-dispatch-sip-egress-route-00.txt <br>
Pages : 11 <br>
Date : 2010-03-01 <br>
&nbsp;<br>
In some cases a UA (or proxy) within a Session Initiation Protocol <br>
[RFC3261] Service Provider (SSP) has multiple parallel egress proxy <br>
options to reach an ingress proxy within another SSP to reach the <br>
target UA. If the originating SSP uses ENUM to identify the ingress <br>
proxy of the target SSP, there is currently no way for the ENUM NAPTR =
<br>
response to identify a specific preferred egress proxy from the set <br>
of multiple parallel proxies. This draft solves this problem by <br>
defining a method for returning deterministic routing information <br>
within an ENUM NAPTR response enabling the UA (or proxy) to choose a =
<br>
preferred egress proxy. Using this information, the originating UA <br>
now sends the SIP request through the predetermined preferred egress =
<br>
proxy to reach the target SSP's proxy. This document describes <br>
several methods to make this determination by incorporating variables =
<br>
into a SIP service URI contained in the E.164 Number Mapping (ENUM) <br>
response. <br>
&nbsp;<br>
A URL for this Internet-Draft is: <br>
<a
href=3D"http://www.ietf.org/internet-drafts/draft-malas-dispatch-sip-egre=
ss-route-00.txt">http://www.ietf.org/internet-drafts/draft-malas-dispatch=
-sip-egress-route-00.txt</a>
</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Regards,</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Daryl</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>---------------------------</span></font><o:p></o:p></=
p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Daryl Malas</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Project Director, IP =
Interconnects</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>w - +1 303 661 3302</span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><a =
href=3D"blocked::mailto:d.malas@cablelabs.com">mailto:d.malas@cablelabs.c=
om</a></span></font><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

</div>

</div>

</div>

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</div>

</body>

</html>

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From kpfleming@digium.com  Thu Mar  4 14:21:29 2010
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Date: Thu, 04 Mar 2010 16:21:26 -0600
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Organization: Digium, Inc.
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] [sipcore] I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Paul E. Jones wrote:

> As an informational RFC, the objective would be nothing more than to
> document the issues that have been identified.  It does not try to address a
> problem, but merely raise awareness to issues in a manner similar to many
> other Informational RFCs over the years.

It's been a few weeks, but we've finally also published the problem
statement that focuses on SIP/SDP issues related to FAX over IP, and it
is located here:

http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,412/Itemid,261/

There has been some discussion amongst our SIP Forum working group about
possibly combining the initial problem statement and this new one as
step towards making a document that is more likely to be accepted (and
relevant) as an informational RFC.

Are there others here who feel that would be a worthwhile exercise?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

From dean.willis@softarmor.com  Thu Mar  4 21:19:14 2010
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Date: Thu, 04 Mar 2010 23:19:04 -0600
From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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Richard Shockey wrote:
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Dean Willis
> Sent: Thursday, March 04, 2010 11:36 AM
> To: dispatch@ietf.org
> Subject: [dispatch] Why are we TRIPped out? Is there an sip-event
> alternative?
>
> Richard Shockey said:
>
>   
>> And as for TRIP ..been there heard that argument before ..no one is  
>> going to
>> deploy a BGP like function for this. Not going to happen no more  
>> stacks in
>> the proxy's we need to deal with the protocols we have not try and  
>> invent
>> new ones.
>>     
>
> I'm starting another thread from this comment, because it raises a  
> really good question.
>
> TRIP was designed to exchange phone-number reachability information  
> between SIP nodes. Nobody uses it. Why?
>
> RS> Heard of LNP? 
>
> Is the problem with TRIP is that it's yet another protocol (as Richard  
> suggests above), or is the problem that it doesn't convey the right  
> information, just doesn't work, or something else?
>
> RS> IMHO more the former rather than the latter.
>
> What if the route-exchange function were done in SIP, perhaps  
> something like an event package for routing?
>
> RS> Nahh.. the routing paths change you want a query mechanism that can
> dynamically change targets based on both internal network topology as well
> as source identification.
an event package could satisfy both of those.

>
> RS> You are on to something here which is the dynamic "who is asking" but
> the response could be lots of things including a SPID or network identifier
> if the target is truly PSTN and the target URI is a matter of bilateral
> agreement. This is really about the function of the local proxy directory in
> a carrier network. I want to simply send an invite but not get back a 302
> ..something else which may be some form of routing meta data. 
>
>   
But hoe does the local proxy learn this stuff? This still calls for a
dynamically modeled routing topology.

> RS> maybe but gee .. ITS ALL ABOUT PHONE NUMBERS. :-) 
>   
but you just said route aggregationon phone numbers doesn't work due to lnp!

--
dean

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From: <mohamed.boucadair@orange-ftgroup.com>
To: Dean Willis <dean.willis@softarmor.com>, Daryl Malas <d.malas@cablelabs.com>
Date: Fri, 5 Mar 2010 08:38:19 +0100
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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=20
Dear Dean, all,

In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interc=
onnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp). Partners in=
volved in that study all agreed that a more distributed mode and dynamic mo=
de is suitable to meet the global reachability requirement and also to allo=
w for a more flexible interconnection model. A consensus has been reached a=
mong participant, that placing communications would involve several ITAD in=
 the path and thus this characteristic should be known before selecting the=
 next ITAD domain for the delivery of a call (e.g., avoid domains of compet=
ing SPs, avoid black listed ITADs, avoid congested links, allow multiple IP=
 telephony paths, minimize interconnection cost, and in general allow for m=
ore traffic engineering at the telephony level, etc.)=20

Below is listed a set of recommendations that has been proposed but the P18=
53 study when specifying and designing an interconnection model for session=
-based services. These recommendations have been drawn from the perspective=
 of IP telephony services but are valid for presence, Instant Message, etc:

1.	Adopt the same interconnection model for various services relying on loo=
kup function. Particularly: IP telephony, video, IM and presence.
2.	Implement a cascaded model (i.e., several ITAD hop for placing a call):
.	For flexibility and scalability reasons;
.	Offers both peering and transit modes;
.	Does not lead to a frozen service architecture;
3.	Functional separation between the lookup service and session management =
(i.e., initiation, terminating, modify, etc.). In particular, SIP is a sess=
ion management protocol and ** not ** a telephony routing protocol;
4.	Activate dynamic means to exchange telephony routing information:
.	Abolish the usage of static practices and "file" based model;
.	The protocol can be used to interconnect internal service platforms and a=
lso with external ones;
.	Convey information to assess the telephony routes.
5.	Encourage aggregation-based advertisement model:
.	Optimise the telephony routing tables/ENUM records, etc.;
.	Check how number portability affects aggregation;
.	How to support roaming.
6.	Adopt a model supporting both closed and open communities which supports=
 the federation scheme and compatible with IP telephony Exchange points pre=
sence;
7.	Investigate means to assess the stability and the convergence of the ove=
rall IP telephony interconnection model;
8.	Promote interconnection practices which are ** agnostic ** to the used s=
ignaling and media protocol;
9.	Use "Media trunks" instead of individual streams: reduce messages overhe=
ad, bandwidth optimisation, etc.;
10.	Standardize/share/promote SIA (Service Interconnection Agreement) templ=
ate and its technical clauses;
11.	Investigate means/procedures to negotiate SIAs;
12.	Adapt the billing/charging model for inter-domain IP telephony services;
13.	Elaborate an inter-domain monitoring model including a template, method=
ology, etc.;
14.	Specify means to enable sophisticated service engineering: load balanci=
ng, SIA optimisation, cost optimisation, avoid congestion and loops, enhanc=
e QoS, etc.;
15.	Activate automatic failure/degradation detection and recovery means at =
the service level;
16.	Promote means to ensure synchronization between the service and network=
 layers;
17.	Provisioning considerations: routing information, access control, polic=
ies, SIA to network nodes;
18.	Consider legal recommendations (e.g. legal intercept);
19.	Encourage a "Service Provider SIP/H.323 profile": this reduces the comp=
lexity of required functions in SBC nodes, protocol repair functions may be=
 obsolete, etc.;
20.	Interconnection model/solution should support planned maintenance opera=
tions (e.g. update physical card, link, add new node, etc.);
21.	Extended TRIP protocol is suggested to be used as base for IP Telephony=
 routing;
22.	Use simple coherent SIA profiling for homogeneous traffic and to avoid =
added complexity in routing optimisation. Interconnection Agreements should=
 constantly be reviewed and a minimum utilization threshold defined. Interc=
onnections with low usage should be cancelled to reduce cost and complexity;
23.	Employ active, non-intrusive measurements on live traffic with probes t=
o ensure that interconnections meet the agreed requirements, maintain high =
service quality, increase reliability, detect and avoid failures even befor=
e they occur, avoid misuse (e.g. SPIT) in the network, etc. A monitoring te=
mplate should be determined before deployment;
24.	In the SIAs to ensure quality, with a minimum MOS score of 3.6, prefera=
bly above 4.0, one-way delay less than 250 ms (even for transatlantic links=
) and packet loss below 2%. Active measurements should be employed to updat=
e and confirm SIAs.


Cheers,
Med

-----Message d'origine-----
De : dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] De la par=
t de Dean Willis
Envoy=E9 : jeudi 4 mars 2010 18:21
=C0 : Daryl Malas
Cc : dispatch@ietf.org
Objet : Re: [dispatch] Why are we TRIPped out? Is there an sip-event altern=
ative?


On Mar 4, 2010, at 11:05 AM, Daryl Malas wrote:

> Dean,
>
> It has been a long time, since I read the RFC.  If I recall=20=20
> correctly, the biggest problem with TRIP was the issue of Number=20=20
> Portability.  Unlike BGP where there is a common rule of summarizing=20=
=20
> IP address blocks into class A's or B's, you cannot do the same with=20=
=20
> E.164 addresses.  Before number portability, TRIP would work good,=20=20
> because you can say I have NPA-NX1 through NPA-NX3.  If you want to=20=20
> reach them go here.  Now, there are rarely any contiguous blocks for=20=
=20
> any service provider.  If SSPs used TRIP, they would have to put in=20=20
> every single E.164 address (e.g. NPA-NXX-XXXX) into a route table.=20=20=
=20
> This would make route tables contain millions of lines, which would=20=20
> slow down routing considerably...not to mention the maintenance=20=20
> nightmare...not to mention the route convergence would be ugly.=20=20=20
> Perhaps there could be a 3rd party application that searches for=20=20
> contiguous ranges, but this too would be maintenance heavy as TNs=20=20
> are ported from one SSP to another; and, there is no guarantee it=20=20
> would significantly improve the routing database issue.
>

Ok, that's good input. Really good.

I attempt a rephrase:

Due to the sparse characteristic of number ranges resulting from LNP,=20=20
route aggregation just doesn't work for phone numbers. TRIP is based=20=20
on effective route aggregation; without it, the route-sets grow=20=20
excessively large. Therefore, an ENUM style model of mapping phone-=20
number to provider (effectively global, relatively static associations=20=
=20
that are cacheable) works much better.

We expect to do at least one ENUM dip per phone-number in order to=20=20
find out who the provider for that phone number is. I argue that doing=20=
=20
further ENUM dips to find the routes to that provider are a Bad Idea,=20=20
as this sort of information is arguably not well-suited to a DNS style=20=
=20
database. Yes, one can hack together lots of heavily customized-=20
locally-provisioned ENUM servers, but it's still a major operational=20=20
suckage event.

Given this, there would appear to be no reasonable requirement for=20=20
dynamic routing (BGP/OSPF/TRIP) based on telephone numbers. However,=20=20
given the inter-provider peering model, there remains a requirement=20=20
for dynamic routing based on peer identifiers (which I presume to be=20=20
domain names).

So what we need is something that expresses the dynamic and shifting=20=20
associations of inter-provider peerings and their associated weights=20=20
and costs, and provides for disseminating this information to our SIP=20=20
routing nodes. Further, the routing vectors involved are not always=20=20
single-hops, but need to be able to express multiple-hop vectors.=20=20
Conversely, the routing vectors need to be able to express aggregate=20=20
weighting vectors while optionally suppressing remote hops, as such=20=20
might be needed for masking remote peering agreements.

--
Dean


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From laura.liess.dt@googlemail.com  Fri Mar  5 04:37:37 2010
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Subject: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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Hi,

We submitted a new version of the Alert-Info URN draft.

The new version incorporates the ML discussions where we think we
already reached a clear agreement. There are still open points.

The most important changes are:
- Requirements added/changed to incorporate the "semantic vs.
non-semantic" discussion and the new non-semantic examples from the
list.
- New alert-identifier "ringback" added.
- Chapter 4.2 "Public telephone network tones" deleted. "Busy",
"intrusion" and "record" were not correct anyway. Country list for
ringback tones deleted, reference to ISO 3166-1.
- Section 4.3.6 on crisis deleted
- "tone" and "ringback" combination deleted from chapter "Combinations of URN".
- Chapter "Combination Rules for Alert URN Indications" upgraded. We
hope the rules are  much more simple now when we distinguish between
ring tones and ringback tones.


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

	Title           : Alert-Info URNs for the Session Initiation Protocol (SIP)
	Author(s)       : D. Alexeitsev, et al.
	Filename        : draft-liess-dispatch-alert-info-urns-01.txt
	Pages           : 20
	Date            : 2010-02-26

The Session Initiation Protocol (SIP) supports the capability to
provide a reference to the alternative ringback tone (RBT) for
caller, or ring tone (RT) for callee using the Alert-Info header.
However, the reference addresses only the network resources with
specific rendering properties.  There is currently no support for
predefined standard identifiers for ringback tones or semantic
indications without being tied to a particular rendering.  To
overcome this limitation and support new applications, a new family
of URNs for use in SIP Alert-Info header fields is defined in this
specification.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-liess-dispatch-alert-info-urns-01.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.


Kind regards,
Laura

From pkyzivat@cisco.com  Fri Mar  5 07:37:57 2010
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Laura,

Comments inline and at the end.

Laura Liess wrote:
> Hi,
> 
> We submitted a new version of the Alert-Info URN draft.
> 
> The new version incorporates the ML discussions where we think we
> already reached a clear agreement. There are still open points.
> 
> The most important changes are:
> - Requirements added/changed to incorporate the "semantic vs.
> non-semantic" discussion and the new non-semantic examples from the
> list.
> - New alert-identifier "ringback" added.
> - Chapter 4.2 "Public telephone network tones" deleted. "Busy",
> "intrusion" and "record" were not correct anyway. Country list for
> ringback tones deleted, reference to ISO 3166-1.
> - Section 4.3.6 on crisis deleted

Then why is it still mentioned in REQ-2?

> - "tone" and "ringback" combination deleted from chapter "Combinations of URN".
> - Chapter "Combination Rules for Alert URN Indications" upgraded. We
> hope the rules are  much more simple now when we distinguish between
> ring tones and ringback tones.

----------

Section 1:

    This specification does not change the usage of the SIP Alert-Info-
    header defined in the RFC3261.  The Alert-Info-header can be used in
    INVITE requests and 180 Ringing responses.

IMO it would make sense to make an extension to 3261 to permit this 
header in any 18x response. This could help resolve issues about what 
ought to be rendered to the caller in case of 182 or 183.

Section 3:

       ...  The left-most
       label is called "alert-category" and is separated from right-side
       of the alert-identifier, the alert-indication, by a semicolon...

          urn:alert:tone:{tone-indication}

The text says "semicolon" but the example uses colon, and the ABNF also 
shows colon.

Section 4:

The way the word "tone" is used in this section could cause confusion, 
after having gone to the trouble of explaining that this mechanism is 
for indicating semantic intent that might not be rendered as tones. Its 
further confused because section 4.1 PBX Tones are encoded using 
urn:alert:tone, but that 4.2 Service Tones are encoded using 
urn:alert:service.

I suggest using some other word than "tone" in cases when audio 
rendering may not be used. For example "indication". Perhaps "caller 
indication" could be used instead of ringback, and "callee indication" 
could be used instead of "ring". (Thats in descriptive text. Perhaps the 
alert-category of "tone" should be changed too, but I'm not ready to 
propose that yet.)

Sections 4.2.6, 4.2.7, 4.2.8:

These talk about "this sub-level". But sub-level isn't defined anywhere. 
There is "sub-indication" in the ABNF, but the examples given in these 
sections don't use that.

These are shown as urn:alert:tone, which means they can only be used for 
callee indication, not caller indication. Is that intended?

I know this might open old wounds, but perhaps it would be better to 
make these values simply be sub-indications on those values where it 
makes sense. But if you don't want to go there I understand.

Section 4.3:

Aren't there also country-specific ring tones? (I know the British ones 
in the movies are different from the American ones.) Should there be 
provision for specifying these? See more on this below.

Why no provision for non-country-specific ringbacks? This is potentially 
a case for customization - perhaps the callee's presence status could be 
used by the callee to select an appropriate ringback to the caller.

Section 4.4:

Are all combinations potentially valid? Or can there be at most one per 
category?

How about having alert:modifier as a category? Then all the other 
categories could be mutually exclusive, and zero or more modifiers would 
adjust whatever one was supplied. In that case, priority, zip, and 
delayed would be modifiers. (Maybe normal also ought to be a modifier, 
and maybe country code should also be a modifier - 
urn:alert:modifier:country:za.)

Section 5:

I think it would be a good idea to mention something about what a UA 
should do with these in more complex cases:
- alert-info received but in-band media also received
- an invite was forked and the UAC is getting responses from
   multiple early dialogs, containing conflicting alert-info
   and/or in-band media.

It occurs to me that in some cases early media should preempt the 
rendering of alert-info, but in other cases it should not. Maybe that 
should be left to implementor discretion, but maybe there are cases 
where this should be deterministic - I haven't thought about it enough 
to have a firm opinion.

Rolling all that up into some overall suggestions:

You have some things that only apply to the "ring", some that only apply 
to the "ringback" and some that apply to both. And then there are things 
that can act as modifiers to any of the others. But the distinction 
between ring and ringback is redundant with the message it travels in. 
So how about:

urn:alert:service:xxx has all the things you have as services plus all 
the things you have as "tones". In an INVITE external/internal describe 
the caller, and in a 18x they describe the callee. Maybe some additional 
values that are useful for describing the callee status.

urn:alert:modifier:xxx where xxx is priority/zip/delayed or country.cc
where cc is a country code.

Combinations allowed are one service, and one or more distinct 
modifiers. (Must be distinct based on top-level.)

From bernard_aboba@hotmail.com  Fri Mar  5 08:29:51 2010
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Subject: Re: [dispatch] [sipcore] I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Kevin Fleming said:

 

"It's been a few weeks, but we've finally also published the problem
statement that focuses on SIP/SDP issues related to FAX over IP, and it is
located here:

 

http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,41
2/Itemid,261/

 

There has been some discussion amongst our SIP Forum working group about
possibly combining the initial problem statement and this new one as step
towards making a document that is more likely to be accepted (and relevant)
as an informational RFC.

 

Are there others here who feel that would be a worthwhile exercise?"

 

 

[BA] The original document was largely focused on issues outside the IETF
(e.g. problems with T.38).  That can't be said about this document.  Given
that, it might make more sense to focus on the new document rather than
combining it with the previous statement. 


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</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Kevin =
Fleming said:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>&#8220;It's been a few weeks, but we've finally =
also published the problem statement that focuses on SIP/SDP issues =
related to FAX over IP, and it is located here:<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText><a =
href=3D"http://www.sipforum.org/component/option,com_docman/task,doc_down=
load/gid,412/Itemid,261/">http://www.sipforum.org/component/option,com_do=
cman/task,doc_download/gid,412/Itemid,261/</a><o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>There =
has been some discussion amongst our SIP Forum working group about =
possibly combining the initial problem statement and this new one as =
step towards making a document that is more likely to be accepted (and =
relevant) as an informational RFC.<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Are =
there others here who feel that would be a worthwhile =
exercise?&#8221;<o:p></o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>[BA] =
The original document was largely focused on issues outside the IETF =
(e.g. problems with T.38).&nbsp; That can&#8217;t be said about this =
document.&nbsp; Given that, it might make more sense to focus on the new =
document rather than combining it with the previous statement. =
<o:p></o:p></p></div></body></html>
------=_NextPart_000_0085_01CABC3E.06563E30--

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To: "Kevin P. Fleming" <kpfleming@digium.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] [sipcore] I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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In my previous email about this documents needs, I put=20

>>> The more important part to publish IMHO is
>>> documenting some issues, problems, deficiencies and/or bugs in some
>>> IETF specifications that are leading to lack of interoperability in
>>> deployments. That type of problem statement would be great =
information
>>> to get written down in a concrete form that allows the appropriate =
WG
>>> to go fix the problems.

The stuff you have in=20
> =
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,=
412/Itemid,261/
is exactly the sort of level of detail I was hoping for.=20

Some of these items are about issues which ITU-T protocol which are =
probably not appropriate for an IETF document but many of them are =
issues with IETF protocols, or limitations where the some other protocol =
may require new parameters in SDP to resolve the issues. All these seem =
very appropriate for an IETF document.=20

So I certainly think combining some of this would result in a draft that =
was more useful for the IETF to help fix fax problems.=20

Cullen=20

On Mar 4, 2010, at 3:21 PM, Kevin P. Fleming wrote:

> Paul E. Jones wrote:
>=20
>> As an informational RFC, the objective would be nothing more than to
>> document the issues that have been identified.  It does not try to =
address a
>> problem, but merely raise awareness to issues in a manner similar to =
many
>> other Informational RFCs over the years.
>=20
> It's been a few weeks, but we've finally also published the problem
> statement that focuses on SIP/SDP issues related to FAX over IP, and =
it
> is located here:
>=20
> =
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,=
412/Itemid,261/
>=20
> There has been some discussion amongst our SIP Forum working group =
about
> possibly combining the initial problem statement and this new one as
> step towards making a document that is more likely to be accepted (and
> relevant) as an informational RFC.
>=20
> Are there others here who feel that would be a worthwhile exercise?
>=20
> --=20
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming@digium.com
> Check us out at www.digium.com & www.asterisk.org


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From dean.willis@softarmor.com  Fri Mar  5 10:16:03 2010
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com> <mohamed.boucadair@orange-ftgroup.com 
 > wrote:

>
> Dear Dean, all,
>
> In the last year, we have undertaken a study about IP telephony  
> interconnect at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp) 
> .

I'm having a hard time resolving this link, as it appears to be 404.  
Perhaps there's a typo?

With respect to:

>
> 21.	Extended TRIP protocol is suggested to be used as base for IP  
> Telephony routing;

We've heard here that TRIP, as currently modeled, doesn't work due to  
the LNP problem. What sorts of extensions id you have in mind? Were  
they intended to address this problem?

--
Dean


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 5, 2010, =
at 1:38 AM, &lt;<a =
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@ora=
nge-ftgroup.com</a>&gt; &lt;<a =
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@ora=
nge-ftgroup.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div><br>Dear Dean, all,<br><br>In the last year, we have =
undertaken a study about IP telephony interconnect at large scale (URL: =
<a =
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_chal=
lenges_of_IP_telephony_Eurescom_study_P1853.asp)">http://www.eurescom.eu/m=
essage/messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_=
study_P1853.asp)</a>.</div></blockquote><div><br></div><div>I'm having a =
hard time resolving this link, as it appears to be 404. Perhaps there's =
a typo?</div><div><br></div><div>With respect =
to:</div><div><br></div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" =
color=3D"#000000"><br></font></div><div>21.<span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>Extended TRIP protocol is =
suggested to be used as base for IP Telephony =
routing;<br></div></blockquote></div><br><div>We've heard here that =
TRIP, as currently modeled, doesn't work due to the LNP problem. What =
sorts of extensions id you have in mind? Were they intended to address =
this =
problem?</div><div><br></div><div>--</div><div>Dean</div><div><br></div></=
body></html>=

--Apple-Mail-4--404538706--

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To: Dean Willis <dean.willis@softarmor.com>, "<mohamed.boucadair@orange-ftgroup.com>" <mohamed.boucadair@orange-ftgroup.com>
Date: Fri, 5 Mar 2010 13:25:22 -0500
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It does not like the trailing ")", this one worked for me:

http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_of=
_IP_telephony_Eurescom_study_P1853.asp

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Dean Willis
Sent: Friday, March 05, 2010 1:16 PM
To: <mohamed.boucadair@orange-ftgroup.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alter=
native?


On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com<mailto:mo=
hamed.boucadair@orange-ftgroup.com>> <mohamed.boucadair@orange-ftgroup.com<=
mailto:mohamed.boucadair@orange-ftgroup.com>> wrote:



Dear Dean, all,

In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interc=
onnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp).

I'm having a hard time resolving this link, as it appears to be 404. Perhap=
s there's a typo?

With respect to:


21.       Extended TRIP protocol is suggested to be used as base for IP Tel=
ephony routing;

We've heard here that TRIP, as currently modeled, doesn't work due to the L=
NP problem. What sorts of extensions id you have in mind? Were they intende=
d to address this problem?

--
Dean


--_000_429446390BA91242867940DBE798A06B7B26BBBB5Fmail_
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xmlns:x=3D"urn:schemas-microsoft-com:office:excel" xmlns:p=3D"urn:schemas-m=
icrosoft-com:office:powerpoint" xmlns:a=3D"urn:schemas-microsoft-com:office=
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d;
-webkit-nbsp-mode: space;-webkit-line-break: after-white-space'>

<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier N=
ew";
color:#1F497D'>It does not like the trailing &#8220;)&#8221;, this one work=
ed
for me:<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier N=
ew";
color:#1F497D'>http://www.eurescom.eu/message/messageMay2009/Interconnectio=
n_challenges_of_IP_telephony_Eurescom_study_P1853.asp<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Courier N=
ew";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal style=3D'margin-left:.5in'><b><span style=3D'font-size=
:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style=3D'font-size=
:10.0pt;
font-family:"Tahoma","sans-serif"'> dispatch-bounces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b>On Behalf Of </b>Dean Willis<br>
<b>Sent:</b> Friday, March 05, 2010 1:16 PM<br>
<b>To:</b> &lt;mohamed.boucadair@orange-ftgroup.com&gt;<br>
<b>Cc:</b> dispatch@ietf.org<br>
<b>Subject:</b> Re: [dispatch] Why are we TRIPped out? Is there an sip-even=
t
alternative?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>On Mar 5, 2010, at 1:38 AM,=
 &lt;<a
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@oran=
ge-ftgroup.com</a>&gt;
&lt;<a href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucada=
ir@orange-ftgroup.com</a>&gt;
wrote:<o:p></o:p></p>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><br>
Dear Dean, all,<br>
<br>
In the last year, we have undertaken a study about IP telephony interconnec=
t at
large scale (URL: <a
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_chall=
enges_of_IP_telephony_Eurescom_study_P1853.asp)">http://www.eurescom.eu/mes=
sage/messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_stu=
dy_P1853.asp)</a>.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>I'm having a hard time reso=
lving
this link, as it appears to be 404. Perhaps there's a typo?<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>With respect to:<o:p></o:p>=
</p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>21.<span class=3Dapple-tab-=
span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Extended
TRIP protocol is suggested to be used as base for IP Telephony routing;<o:p=
></o:p></p>

</div>

</blockquote>

</div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>We've heard here that TRIP,=
 as
currently modeled, doesn't work due to the LNP problem. What sorts of
extensions id you have in mind? Were they intended to address this problem?=
<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>--<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'>Dean<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal style=3D'margin-left:.5in'><o:p>&nbsp;</o:p></p>

</div>

</div>

</body>

</html>

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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Dean Willis said:

 

"I'm having a hard time resolving this link, as it appears to be 404.
Perhaps there's a typo?"

 

[BA] Try this
<http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_of
_IP_telephony_Eurescom_study_P1853.asp> . 

 

The study seems to be available on a password-protected link here:

http://www.eurescom.eu/Public/Projects/P1800-series/P1853/

 

 


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Dean =
Willis said:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoPlainText>&#8220;I'm having a hard time resolving this link, =
as it appears to be 404.&nbsp; Perhaps there's a =
typo?&#8221;<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>[BA] Try <a =
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_cha=
llenges_of_IP_telephony_Eurescom_study_P1853.asp">this</a>. =
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>The study seems to be available on a =
password-protected link here:<o:p></o:p></p><p class=3DMsoNormal><a =
href=3D"http://www.eurescom.eu/Public/Projects/P1800-series/P1853/">http:=
//www.eurescom.eu/Public/Projects/P1800-series/P1853/</a><o:p></o:p></p><=
p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
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To: "'Cullen Jennings'" <fluffy@cisco.com>, "'Kevin P. Fleming'" <kpfleming@digium.com>
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Subject: Re: [dispatch] [sipcore]	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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In line...

In my previous email about this documents needs, I put 

>>> The more important part to publish IMHO is
>>> documenting some issues, problems, deficiencies and/or bugs in some
>>> IETF specifications that are leading to lack of interoperability in
>>> deployments. That type of problem statement would be great information
>>> to get written down in a concrete form that allows the appropriate WG
>>> to go fix the problems.

The stuff you have in 
>
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,41
2/Itemid,261/
is exactly the sort of level of detail I was hoping for. 

Some of these items are about issues which ITU-T protocol which are probably
not appropriate for an IETF document but many of them are issues with IETF
protocols, or limitations where the some other protocol may require new
parameters in SDP to resolve the issues. All these seem very appropriate for
an IETF document. 


RS> Cullen, respectfully, I'm not sure I completely agree with that
sentiment. Since this is purely a Informational document that describes
issues with Real-time Communications in SIP I don't understand why it would
be inappropriate to discuss possible issues and interactions with ITU
protocols. I don't understand why things like that are "out of bounds". We
have published documents in the past that discuss issues orthogonal to IETF
protocols.

http://www.ietf.org/rfc/rfc3482.txt

being a good example.

RS> The problems we are seeing in fax are directly related to the
interaction of SIP and T.38 how can we propose solutions without directly
addressing the problem? 


So I certainly think combining some of this would result in a draft that was
more useful for the IETF to help fix fax problems. 

Cullen 

On Mar 4, 2010, at 3:21 PM, Kevin P. Fleming wrote:

> Paul E. Jones wrote:
> 
>> As an informational RFC, the objective would be nothing more than to
>> document the issues that have been identified.  It does not try to
address a
>> problem, but merely raise awareness to issues in a manner similar to many
>> other Informational RFCs over the years.
> 
> It's been a few weeks, but we've finally also published the problem
> statement that focuses on SIP/SDP issues related to FAX over IP, and it
> is located here:
> 
>
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,41
2/Itemid,261/
> 
> There has been some discussion amongst our SIP Forum working group about
> possibly combining the initial problem statement and this new one as
> step towards making a document that is more likely to be accepted (and
> relevant) as an informational RFC.
> 
> Are there others here who feel that would be a worthwhile exercise?
> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming@digium.com
> Check us out at www.digium.com & www.asterisk.org


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



_______________________________________________
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From dean.willis@softarmor.com  Fri Mar  5 12:05:05 2010
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Subject: [dispatch] Thoughts on SIP routing, TRIP, and ENUM (Long!)
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Fundamental routing problem:

The fundamental SIP routing problem is this: I've been given a message  
that is addressed either to a telephone number or to a user@domain  
name. Where do I send this SIP request to next?

For "really stupid nodes", the question is easily answered: if the  
destination isn't "me", I send the request to my default proxy and let  
it deal with the problem. The default proxy might route the message  
itself, or it might send back a 302 informing the me as to where to  
send the request. It is important to note that in general, this "next  
hop" decision represents a singular hop. We have no way to 302-respond  
with a vector that specifies a sequence of relays to be used.

For a smarter host on "the open Internet", the answer is also easy. If  
the target identifier is a "user@domain" URI, then I look up the  
domain part in DNS and send the request there.  If the target  
identifier is a phone number, I use ENUM to look up an associated SIP  
URI, and I send the request to that URI. Local routing tables that  
have much the same effect may also be used, but are not specified in  
the governing RFCs.

But the "real world" (at least as we have allowed it to become  
perverted) doesn't work that way. In this unhappy place, we don't have  
direct end-to-end SIP routability. Instead, for many reasons good,  
bad, and mostly practical, we must transit a set of relay nodes to get  
from point A to B.  Further complicating matters, we may not even know  
what the whole set of nodes are, as various portions of the vector may  
be "topology hidden."

So in practice, a "really stupid node" like a phone-gateway or PBX  
sends its request to a routing proxy, which then uses a combination of  
ENUM, DNS, and local policy to select a next-hop relay. Note that this  
decision-making process tends to repeat at each relay node, since 1)  
we don't have away to represent a multi-hop vector in a 302 response,  
and 2) each node has limited view of the end-to-end visibility, in  
part due to topology hiding. There's no good exchange mechanism for  
this information, and no standard way to inform SIP nodes about how to  
use it. We've seen proposals for source-context-sensitive private ENUM  
solutions, mutagenic SIP proxy servers that return 302s, various web- 
service approaches, and others.



TRIP: A solution?

TRIP proposes one model, wherein the "smart" nodes (called LS, or  
Location Servers) exchange reachability information with each other  
using a variant of BGP. This provides us a rich fabric for advertising  
routes, control peering policy, deciding what parts of stuff to  
"topology hide", and so on. Where BGP uses IP addresses (or the  
prefixes thereof) as routing targets, TRIP uses telephone numbers (or  
the prefixes thereof). Like BGP, it allows for "internal" routing  
using a flood model, and "external" routing using more controlled  
policy.

In shortest essence, TRIP allows a first LS to say to an external LS   
"For the following phone numbers and phone number ranges, I suggest  
you use this other node or set of nodes (by IP or DNS name) as your  
next hop(s). And  I heard this from node X, who heard it from node Y,  
who heard it from node Z."  It allows allows a first LS to say to its  
internal peer "Here are all the routes I know; feel free to use them  
and please tell me about any you know that I don't."

But TRIP has not caught on. I've heard mixed answers, which boil down  
to 1) its aggregation model for routes is broken, meaning Too Many  
Routes (as a product of LNP which makes phone numbers messy), 2) it's  
too complicated, and 3) Cisco owns the IPR, and I'm not sure they want  
to share. I'm not sure I believe this is all there is to it.

The REAL problem with TRIP may be its fundamental routing model. It  
assumes that the routing of a phone number is a property of a given  
phone number (or aggregation range), and doesn't capitalize on the  
actual peering relationships. Further, it doesn't really allows us to  
talk about "routes to domains". just "routes to phone numbers", so it  
doesn't help with a call to "sip:user@example.com" at all. And as we  
know,  given the peering models, RFC 3263 is also inadequate for "sip@user@example.com 
". We can't go from here to there using "example.com"'s SIP SRV  
record. Instead, we need to find the right peering partner who can get  
us there, and we don't have a good way to do that.



Another model: Inter-ITAD routing:

If we assume that each SIP Service Provider (SSP)  is essentially the  
same thing as an ITAD (Internet Telephony Administrative Domain), and  
that ITADs peer with each other along well-defined channels (using  
Session Border Elements, SBEs, known by other acronyms including  
Little Spawns of Satan etc.), then this gives us another routing  
topology to consider.

Assume a routing protocol that tells ITAD A that to reach ITAD Z, the  
request should be sent to ITAD Q (or perhaps to a vector of Q and T)  
who will act as an intermediary. If ITAD A can directly reach Q, then  
A knows how to route the request. A can send the request to A's SBE  
that is adjacent to Q, which then hands it to Q's SBE (adjacent to A),  
which then consults its own routing tables to figure out how to move  
the request on, either to Z, to the previously declared secondary  
route T, or to some other ITAD known to Q that will serve as a relay.

With this model, we don't need to know the route for a given E.164  
number (or @domain part). All we need to know is which ITAD that  
target is associated with. We then consult the ITAD routing table to  
send the request. This means that the largest dynamic routing table  
required is the inter-ITAD map, which is exponentially smaller than  
the number of telephone routes.

So, given a dynamic inter/intra ITAD routing protocol that can tell us  
what SBE to use to move towards a given ITAD, all we need to know is a  
way to translate a phone number or a domain-part into an ITAD.

This is the easy part. ENUM gives us an excellent mechanism for  
turning phone numbers into either numeric or domain-style ITAD  
identifiers. The mapping of phone number (or phone number range) to  
ITAD is relatively static, globally valid, and well-suited to the  
recursive-lookup and cache model that ENUM inherited from DNS.

This gives us a new simple model for call routing at the LS for phone- 
number targeted calls

1) If the number isn't locally known, look up its ITAD in ENUM. This  
might be range-matched or fully qualified; it's just a standard ENUM  
lookup.
2) Consult the ITAD routing table to select an SBE (and possibly  
additional transit ITADs) to find the routing vector consisting of one  
or more SBEs and zero or more transit ITADs.
3) Canonicalize the request URI and add the SBEs/ITADs as loose route  
hops
4) Send the request.

For domain-targeted calls, we might also get to add in "Check to see  
if the destination is reachable via RFC 3263, and if so, use that"  
somewhere in the process.

The short form of this model is 1) Find the ITAD of the destination.  
2) Find a route to that ITAD (which might be one or more  
intermediaries. 3) Send the request there.



Conclusion:

So, other than a dynamic routing protocol that meets inter/intra-ITAD  
routing requirements and could be derived from TRIP (or built using  
some other model, such as a SIP event package), what are we missing?


--
Dean



















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To: Richard Shockey <richard@shockey.us>
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Subject: Re: [dispatch] [sipcore]	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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On Mar 5, 2010, at 12:11 PM, Richard Shockey wrote:

> In line...
>=20
> In my previous email about this documents needs, I put=20
>=20
>>>> The more important part to publish IMHO is
>>>> documenting some issues, problems, deficiencies and/or bugs in some
>>>> IETF specifications that are leading to lack of interoperability in
>>>> deployments. That type of problem statement would be great =
information
>>>> to get written down in a concrete form that allows the appropriate =
WG
>>>> to go fix the problems.
>=20
> The stuff you have in=20
>>=20
> =
http://www.sipforum.org/component/option,com_docman/task,doc_download/gid,=
41
> 2/Itemid,261/
> is exactly the sort of level of detail I was hoping for.=20
>=20
> Some of these items are about issues which ITU-T protocol which are =
probably
> not appropriate for an IETF document but many of them are issues with =
IETF
> protocols, or limitations where the some other protocol may require =
new
> parameters in SDP to resolve the issues. All these seem very =
appropriate for
> an IETF document.=20
>=20
>=20
> RS> Cullen, respectfully, I'm not sure I completely agree with that
> sentiment. Since this is purely a Informational document that =
describes
> issues with Real-time Communications in SIP I don't understand why it =
would
> be inappropriate to discuss possible issues and interactions with ITU
> protocols.
> I don't understand why things like that are "out of bounds". We
> have published documents in the past that discuss issues orthogonal to =
IETF
> protocols.
>=20
> http://www.ietf.org/rfc/rfc3482.txt

That document was to document knowledge needed about PSTN such that =
future people updating the IETF PS documents in this area would have =
enough information to not mess it up. As you of all people remember, =
there has been a lot of energy spend educating people about phones =
numbers.  I certainly did not say I thought discussion "possible issues =
and interactions with ITU protocols is out of scope". I think I actually =
said the opposite of that. However, to try and make the point clearer. =
The IETF is usually caution in saying "SDO X really did a bad job of =
protocol X".  That often has no real value but, at times, it is the =
right thing  the IETF does do roughly that. There has been more than one =
IETF document that shared that sentiment and often have a title like "X =
considered harmful".=20

I have been trying to encourage a document that is useful in that it =
leads  to helping the IETF know what the IETF needs to do to make fax =
work better.  If you think this is the wrong direction to be going, I'm =
totally open to suggestions. What are is it you are actually trying to =
accomplish with a publication here?=20


>=20
> being a good example.
>=20
> RS> The problems we are seeing in fax are directly related to the
> interaction of SIP and T.38 how can we propose solutions without =
directly
> addressing the problem?=20
>=20
>=20
> So I certainly think combining some of this would result in a draft =
that was
> more useful for the IETF to help fix fax problems.=20
>=20
> Cullen=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




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Dean Willis wrote:

> This is the easy part. ENUM gives us an excellent mechanism for turning
> phone numbers into either numeric or domain-style ITAD identifiers. The
> mapping of phone number (or phone number range) to ITAD is relatively
> static, globally valid, and well-suited to the recursive-lookup and
> cache model that ENUM inherited from DNS.

Define 'relatively static' in this context... With LNP in North America
at least, there are thousands upon thousands of phone numbers being
moved from one provider to another every week day.

> This gives us a new simple model for call routing at the LS for
> phone-number targeted calls
> 
> 1) If the number isn't locally known, look up its ITAD in ENUM. This
> might be range-matched or fully qualified; it's just a standard ENUM
> lookup.
> 2) Consult the ITAD routing table to select an SBE (and possibly
> additional transit ITADs) to find the routing vector consisting of one
> or more SBEs and zero or more transit ITADs.
> 3) Canonicalize the request URI and add the SBEs/ITADs as loose route hops
> 4) Send the request.

It is my understanding that at the SS7 level in areas where LNP has been
deployed, the initial route for any number is still the provider who was
assigned the block containing that number; if the number has since been
ported, that provider's switch will reply with what is effectively a 302
to the new provider servicing that number. This was done because of the
same reasons you are talking about here... a routing table that could
route to the phone-number level would be enormous.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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Subject: Re: [dispatch] Thoughts on SIP routing, TRIP, and ENUM (Long!)
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On Mar 5, 2010, at 2:30 PM, Kevin P. Fleming wrote:

> Dean Willis wrote:
>
>> This is the easy part. ENUM gives us an excellent mechanism for  
>> turning
>> phone numbers into either numeric or domain-style ITAD identifiers.  
>> The
>> mapping of phone number (or phone number range) to ITAD is relatively
>> static, globally valid, and well-suited to the recursive-lookup and
>> cache model that ENUM inherited from DNS.
>
> Define 'relatively static' in this context... With LNP in North  
> America
> at least, there are thousands upon thousands of phone numbers being
> moved from one provider to another every week day.

Yep, and there are hundred of thousands of DNS records changing every  
day. This is still handily in the design-model for DNS.

>
>> This gives us a new simple model for call routing at the LS for
>> phone-number targeted calls
>>
>> 1) If the number isn't locally known, look up its ITAD in ENUM. This
>> might be range-matched or fully qualified; it's just a standard ENUM
>> lookup.
>> 2) Consult the ITAD routing table to select an SBE (and possibly
>> additional transit ITADs) to find the routing vector consisting of  
>> one
>> or more SBEs and zero or more transit ITADs.
>> 3) Canonicalize the request URI and add the SBEs/ITADs as loose  
>> route hops
>> 4) Send the request.
>
> It is my understanding that at the SS7 level in areas where LNP has  
> been
> deployed, the initial route for any number is still the provider who  
> was
> assigned the block containing that number; if the number has since  
> been
> ported, that provider's switch will reply with what is effectively a  
> 302
> to the new provider servicing that number. This was done because of  
> the
> same reasons you are talking about here... a routing table that could
> route to the phone-number level would be enormous.
>

Hmm. I thought we were doing LNP dips for every phone number on call  
setup. Since the proposed ENUM dip handles both ported and non-ported  
cases, it seems like this would work well. Although it should be  
pointed out that porting MAY still send the call to the original ITAD  
for service until the TTL has exhausted, so the original ITAD has a  
little more work to do when effecting a port than just update one ENUM  
record.

--
Dean


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Dean said:

"So, other than a dynamic routing protocol that meets inter/intra-ITAD
routing requirements and could be derived from TRIP (or built using some
other model, such as a SIP event package), what are we missing?"

 

[BA] I think there is quite a bit missing, actually.  In practice, a video
call may need to be routed differently from a pure voice call, even in the
case of an INVITE with a Request-URI that represents a phone number.  For
example, a PBX might typically be set up to route calls to a SIP trunk based
on the phone number in the Request-URI.  However, if the SSP is not be
capable of supporting video, a number of undesirable things might happen:
the call might succeed but only bring up audio due to "SDP scrubbing" by the
SSP, or call setup might fail entirely.  There are multiple ways to handle
the problem today.  One way is to dual-register the endpoints, with one of
the registrations being to a special "video network" that might assign real
(or bogus) "phone numbers" to the endpoints, allowing them to call each
other as if they were all attached to a single pseudo-PBX.  Another way is
to first try to route the call via mechanisms that try to find routes less
likely to lose functionality (e.g. an "ENUM trunk", a "DUNDI trunk" or a
"VIPR trunk").  However, those mechanisms are somewhat under-developed
today. 


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<p class=3DMsoNormal>Dean said:<o:p></o:p></p>

<p class=3DMsoPlainText>&#8220;So, other than a dynamic routing protocol =
that
meets inter/intra-ITAD routing requirements and could be derived from =
TRIP (or
built using some other model, such as a SIP event package), what are we
missing?&#8221;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>[BA] I think there is quite a bit missing, =
actually. &nbsp;In
practice, a video call may need to be routed differently from a pure =
voice call,
<i>even in the case of an INVITE with a Request-URI that represents a =
phone
number.</i>&nbsp; For example, a PBX might typically be set up to route =
calls
to a SIP trunk based on the phone number in the Request-URI. =
&nbsp;However, if
the SSP is not be capable of supporting video, a number of undesirable =
things
might happen:&nbsp; the call might succeed but only bring up audio due =
to &#8220;SDP
scrubbing&#8221; by the SSP, or call setup might fail entirely.&nbsp; =
There are
multiple ways to handle the problem today.&nbsp; One way is to =
dual-register
the endpoints, with one of the registrations being to a special =
&#8220;video
network&#8221; that might assign real (or bogus) &#8220;phone =
numbers&#8221; to
the endpoints, allowing them to call each other as if they were all =
attached to
a single pseudo-PBX.&nbsp; Another way is to first try to route the call =
via mechanisms
that try to find routes less likely to lose functionality (e.g. an =
&#8220;ENUM
trunk&#8221;, a &#8220;DUNDI trunk&#8221; or a &#8220;VIPR =
trunk&#8221;).&nbsp;
However, those mechanisms are somewhat under-developed today. =
<o:p></o:p></p>

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From pkyzivat@cisco.com  Sat Mar  6 14:53:16 2010
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Subject: Re: [dispatch] Thoughts on SIP routing, TRIP, and ENUM (Long!)
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Am I the only one that finds it absurd that one would have to route the 
signaling differently based on the media in the call?

Among other things, this means that INVITEs without offers may well not 
get the desired result, and mid-call addition of media may not work.

	Thanks,
	Paul

Bernard Aboba wrote:
> Dean said:
> 
> “So, other than a dynamic routing protocol that meets inter/intra-ITAD 
> routing requirements and could be derived from TRIP (or built using some 
> other model, such as a SIP event package), what are we missing?”
> 
>  
> 
> [BA] I think there is quite a bit missing, actually.  In practice, a 
> video call may need to be routed differently from a pure voice call, 
> /even in the case of an INVITE with a Request-URI that represents a 
> phone number./  For example, a PBX might typically be set up to route 
> calls to a SIP trunk based on the phone number in the Request-URI. 
>  However, if the SSP is not be capable of supporting video, a number of 
> undesirable things might happen:  the call might succeed but only bring 
> up audio due to “SDP scrubbing” by the SSP, or call setup might fail 
> entirely.  There are multiple ways to handle the problem today.  One way 
> is to dual-register the endpoints, with one of the registrations being 
> to a special “video network” that might assign real (or bogus) “phone 
> numbers” to the endpoints, allowing them to call each other as if they 
> were all attached to a single pseudo-PBX.  Another way is to first try 
> to route the call via mechanisms that try to find routes less likely to 
> lose functionality (e.g. an “ENUM trunk”, a “DUNDI trunk” or a “VIPR 
> trunk”).  However, those mechanisms are somewhat under-developed today.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

From bernard_aboba@hotmail.com  Sat Mar  6 15:40:23 2010
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Subject: Re: [dispatch] Thoughts on SIP routing, TRIP, and ENUM (Long!)
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I don't think you are alone here.  The current state of things is barely
workable for intra-domain operation, let alone inter-domain.  Many of the
hacks required either have lots of failure modes or have such limited
applicability or scaling characteristics they are barely worth the effort 
to configure them. 

Given the current state of things, I have seen situations where customers
have chosen public services like Skype, Live Messenger or GoogleTalk
over SIP.   

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Saturday, March 06, 2010 2:53 PM
To: Bernard Aboba
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Thoughts on SIP routing, TRIP, and ENUM (Long!)

Am I the only one that finds it absurd that one would have to route the 
signaling differently based on the media in the call?

Among other things, this means that INVITEs without offers may well not 
get the desired result, and mid-call addition of media may not work.

	Thanks,
	Paul

Bernard Aboba wrote:
> Dean said:
> 
> "So, other than a dynamic routing protocol that meets inter/intra-ITAD 
> routing requirements and could be derived from TRIP (or built using some 
> other model, such as a SIP event package), what are we missing?"
> 
>  
> 
> [BA] I think there is quite a bit missing, actually.  In practice, a 
> video call may need to be routed differently from a pure voice call, 
> /even in the case of an INVITE with a Request-URI that represents a 
> phone number./  For example, a PBX might typically be set up to route 
> calls to a SIP trunk based on the phone number in the Request-URI. 
>  However, if the SSP is not be capable of supporting video, a number of 
> undesirable things might happen:  the call might succeed but only bring 
> up audio due to "SDP scrubbing" by the SSP, or call setup might fail 
> entirely.  There are multiple ways to handle the problem today.  One way 
> is to dual-register the endpoints, with one of the registrations being 
> to a special "video network" that might assign real (or bogus) "phone 
> numbers" to the endpoints, allowing them to call each other as if they 
> were all attached to a single pseudo-PBX.  Another way is to first try 
> to route the call via mechanisms that try to find routes less likely to 
> lose functionality (e.g. an "ENUM trunk", a "DUNDI trunk" or a "VIPR 
> trunk").  However, those mechanisms are somewhat under-developed today.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From lconroy@insensate.co.uk  Sun Mar  7 03:04:38 2010
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Hi Paul, folks,
 Nope. You are not. This discussion was raised a Loooooong time ago (in =
the London/SLC meetings, IIRC), so people have had your view for at =
least that long.

The consensus then was that even if one had separate identities for =
separate services with separate media features, these would be subsumed =
by a single registrar-proxy, and you would have one common "outward =
facing" identity. That proxy would fork any requests as needed. SIP =
would negotiate any service-specific routing, so there was no problem.
In call capability negotiation would handle the rest.

The issue I had with that at the time was that "joe punter" might get =
SIP service from a set of CSPs, each of which was severely limited in =
the media features it handled. Voice might be handled by one CSP whilst =
IM might come from another CSP. There would be no practical leverage for =
mortals to convince those providers to offer full service, or to =
convince (for example) AT&T systems to register with a front-end MCI =
proxy. SIP full service might not be provided, regardless of the logic =
in the IETF's argument.

In those cases where the end user has fully functional devices, and is =
supported by fully functional CSPs, it all works. (Re-)routing based on =
media features is not an issue.

In the real world... one has to ask oneself what's absurd?

all the best,
  Lawrence

On 6 Mar 2010, at 22:53, Paul Kyzivat wrote:

> Am I the only one that finds it absurd that one would have to route =
the signaling differently based on the media in the call?
>=20
> Among other things, this means that INVITEs without offers may well =
not get the desired result, and mid-call addition of media may not work.
>=20
> 	Thanks,
> 	Paul
>=20
> Bernard Aboba wrote:
>> Dean said:
>> =93So, other than a dynamic routing protocol that meets =
inter/intra-ITAD routing requirements and could be derived from TRIP (or =
built using some other model, such as a SIP event package), what are we =
missing?=94
>> [BA] I think there is quite a bit missing, actually.  In practice, a =
video call may need to be routed differently from a pure voice call, =
/even in the case of an INVITE with a Request-URI that represents a =
phone number./  For example, a PBX might typically be set up to route =
calls to a SIP trunk based on the phone number in the Request-URI.  =
However, if the SSP is not be capable of supporting video, a number of =
undesirable things might happen:  the call might succeed but only bring =
up audio due to =93SDP scrubbing=94 by the SSP, or call setup might fail =
entirely.  There are multiple ways to handle the problem today.  One way =
is to dual-register the endpoints, with one of the registrations being =
to a special =93video network=94 that might assign real (or bogus) =
=93phone numbers=94 to the endpoints, allowing them to call each other =
as if they were all attached to a single pseudo-PBX.  Another way is to =
first try to route the call via mechanisms that try to find routes less =
likely to lose functionality (e.g. an =93ENUM trunk=94, a =93DUNDI =
trunk=94 or a =93VIPR trunk=94).  However, those mechanisms are somewhat =
under-developed today.
>> =
------------------------------------------------------------------------
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From dean.willis@softarmor.com  Sun Mar  7 10:38:01 2010
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On Mar 6, 2010, at 3:47 PM, Bernard Aboba wrote:

> Dean said:
> =93So, other than a dynamic routing protocol that meets inter/intra-=20=

> ITAD routing requirements and could be derived from TRIP (or built =20
> using some other model, such as a SIP event package), what are we =20
> missing?=94
>
> [BA] I think there is quite a bit missing, actually.  In practice, a =20=

> video call may need to be routed differently from a pure voice call, =20=

> even in the case of an INVITE with a Request-URI that represents a =20
> phone number.  For example, a PBX might typically be set up to route =20=

> calls to a SIP trunk based on the phone number in the Request-URI.  =20=

> However, if the SSP is not be capable of supporting video, a number =20=

> of undesirable things might happen:  the call might succeed but only =20=

> bring up audio due to =93SDP scrubbing=94 by the SSP, or call setup =20=

> might fail entirely.  There are multiple ways to handle the problem =20=

> today.  One way is to dual-register the endpoints, with one of the =20
> registrations being to a special =93video network=94 that might assign =
=20
> real (or bogus) =93phone numbers=94 to the endpoints, allowing them to =
=20
> call each other as if they were all attached to a single pseudo-=20
> PBX.  Another way is to first try to route the call via mechanisms =20
> that try to find routes less likely to lose functionality (e.g. an =20
> =93ENUM trunk=94, a =93DUNDI trunk=94 or a =93VIPR trunk=94).  =
However, those =20
> mechanisms are somewhat under-developed today.

Most of that routing would seem to be done between the terminating-=20
AOR's service proxy and the user agent. which is an entirely separate =20=

part of the routing problem that is currently handled by registration. =20=

I'll admit that I've also proposed a different sort of dynamic routing =20=

as an alternative to registration, but whatever that is, it doesn't =20
feel like BGP, and might well have service-specific aspects (as does =20
REGISTER, via RFC 3840/3841).

--
Dean
 =20=

--Apple-Mail-13--230416937
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 6, 2010, =
at 3:47 PM, Bernard Aboba wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple"><div class=3D"Section1"><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
11pt; font-family: Calibri, sans-serif; ">Dean =
said:<o:p></o:p></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10.5pt; =
font-family: Consolas; ">=93So, other than a dynamic routing protocol =
that meets inter/intra-ITAD routing requirements and could be derived =
from TRIP (or built using some other model, such as a SIP event =
package), what are we missing?=94<o:p></o:p></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 10.5pt; font-family: Consolas; =
"><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0in; margin-right: =
0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: 10.5pt; =
font-family: Consolas; ">[BA] I think there is quite a bit missing, =
actually. &nbsp;In practice, a video call may need to be routed =
differently from a pure voice call,<span =
class=3D"Apple-converted-space">&nbsp;</span><i>even in the case of an =
INVITE with a Request-URI that represents a phone number.</i>&nbsp; For =
example, a PBX might typically be set up to route calls to a SIP trunk =
based on the phone number in the Request-URI. &nbsp;However, if the SSP =
is not be capable of supporting video, a number of undesirable things =
might happen:&nbsp; the call might succeed but only bring up audio due =
to =93SDP scrubbing=94 by the SSP, or call setup might fail =
entirely.&nbsp; There are multiple ways to handle the problem =
today.&nbsp; One way is to dual-register the endpoints, with one of the =
registrations being to a special =93video network=94 that might assign =
real (or bogus) =93phone numbers=94 to the endpoints, allowing them to =
call each other as if they were all attached to a single =
pseudo-PBX.&nbsp; Another way is to first try to route the call via =
mechanisms that try to find routes less likely to lose functionality =
(e.g. an =93ENUM trunk=94, a =93DUNDI trunk=94 or a =93VIPR =
trunk=94).&nbsp; However, those mechanisms are somewhat under-developed =
today.<o:p></o:p></div></div></div></span></blockquote></div><br><div>Most=
 of that routing would seem to be done between the terminating-AOR's =
service proxy and the user agent. which is an entirely separate part of =
the routing problem that is currently handled by registration. I'll =
admit that I've also proposed a different sort of dynamic routing as an =
alternative to registration, but whatever that is, it doesn't feel like =
BGP, and might well have service-specific aspects (as does REGISTER, via =
RFC =
3840/3841).</div><div><br></div><div>--</div><div>Dean</div><div>&nbsp;</d=
iv></body></html>=

--Apple-Mail-13--230416937--

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Dean,

Please see attached an (unsubmitted) draft I started writing a couple of 
years ago before DRINKS really took off (in the wrong direction, IMHO).

The bilaterally exchanged information described there in could be your 
ITAD routing system.  What this does have (but your notes don't have) is 
the concept that an ITAD actually needs to be broken into smaller parts to 
achieve optimal routing.  At least in the EU/UK, carriers like "far end 
handover", because it's cheaper to keep the call in your own network and 
drop it into the recipient network as close to the called party as 
possible.  This more fine-grained split is the "destination group" term 
now used in DRINKS.

kind regards,

Ray


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<tt><font size=2>Dean,</font></tt>
<br>
<br><tt><font size=2>Please see attached an (unsubmitted) draft I started
writing a couple of years ago before DRINKS really took off (in the wrong
direction, IMHO).</font></tt>
<br>
<br><tt><font size=2>The bilaterally exchanged information described there
in could be your ITAD routing system. &nbsp;What this does have (but your
notes don't have) is the concept that an ITAD actually needs to be broken
into smaller parts to achieve optimal routing. &nbsp;At least in the EU/UK,
carriers like &quot;far end handover&quot;, because it's cheaper to keep
the call in your own network and drop it into the recipient network as
close to the called party as possible. &nbsp;This more fine-grained split
is the &quot;destination group&quot; term now used in DRINKS.</font></tt>
<br>
<br><tt><font size=2>kind regards,</font></tt>
<br>
<br><tt><font size=2>Ray</font></tt>
<br>
<br>
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=0A=0A=0ASpeermint                                                      R. =
Bellis=0AInternet-Draft                                                Nomi=
net UK=0AExpires: November 3, 2008                                   May 02=
, 2008=0A=0A=0A     IANA Registrations for the 'Destination Group' Enumserv=
ice for=0A                          Infrastructure ENUM=0A                 =
  draft-bellis-enum-destgroups-00pre=0A=0AStatus of this Memo=0A=0A   By su=
bmitting this Internet-Draft, each author represents that any=0A   applicab=
le patent or other IPR claims of which he or she is aware=0A   have been or=
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re working documents of the Internet Engineering=0A   Task Force (IETF), it=
s areas, and its working groups.  Note that=0A   other groups may also dist=
ribute working documents as Internet-=0A   Drafts.=0A=0A   Internet-Drafts =
are draft documents valid for a maximum of six months=0A   and may be updat=
ed, replaced, or obsoleted by other documents at any=0A   time.  It is inap=
propriate to use Internet-Drafts as reference=0A   material or to cite them=
 other than as "work in progress."=0A=0A   The list of current Internet-Dra=
fts can be accessed at=0A   http://www.ietf.org/ietf/1id-abstracts.txt.=0A=
=0A   The list of Internet-Draft Shadow Directories can be accessed at=0A  =
 http://www.ietf.org/shadow.html.=0A=0A   This Internet-Draft will expire o=
n November 3, 2008.=0A=0AAbstract=0A=0A   This document requests IANA regis=
tration of an Enumservice=0A   'Destination Group' and a 'dg' URI scheme.  =
A Destination Group is an=0A   opaque reference used to lookup interconnect=
ion data for inter-=0A   provider SIP sessions.=0A=0A=0A=0A=0A=0A=0A=0A=0A=
=0A=0A=0ABellis                  Expires November 3, 2008                [P=
age 1]=0A=0C=0AInternet-Draft        Destination Group Enumservice         =
    May 2008=0A=0A=0ATable of Contents=0A=0A   1.  Introduction  . . . . . =
. . . . . . . . . . . . . . . . . . . . 3=0A=0A   2.  Terminology . . . . .=
 . . . . . . . . . . . . . . . . . . . . . 3=0A=0A   3.  ENUM Service Regis=
tration for "Destination Group" . . . . . . . 3=0A=0A   4.  IANA Registrati=
on Template for URI Scheme "dg"  . . . . . . . . 4=0A=0A   5.  Description =
. . . . . . . . . . . . . . . . . . . . . . . . . . 5=0A=0A   6.  Examples =
 . . . . . . . . . . . . . . . . . . . . . . . . . . . 5=0A=0A   7.  DNS Co=
nsiderations  . . . . . . . . . . . . . . . . . . . . . . 5=0A=0A   8.  Sec=
urity Considerations . . . . . . . . . . . . . . . . . . . . 5=0A=0A   9.  =
IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 5=0A=0A   1=
0. Change Log  . . . . . . . . . . . . . . . . . . . . . . . . . . 6=0A=0A =
  11. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 6=0A=
=0A   12. References  . . . . . . . . . . . . . . . . . . . . . . . . . . 6=
=0A     12.1.  Normative References . . . . . . . . . . . . . . . . . . . 6=
=0A     12.2.  Informative References . . . . . . . . . . . . . . . . . . 6=
=0A=0A   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . . =
. 7=0A   Intellectual Property and Copyright Statements  . . . . . . . . . =
. 8=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0ABellis=
                  Expires November 3, 2008                [Page 2]=0A=0C=0A=
Internet-Draft        Destination Group Enumservice             May 2008=0A=
=0A=0A1.  Introduction=0A=0A   The requirements [RFC5067] for Infrastructur=
e ENUM=0A   [I-D.ietf-enum-infrastructure] (I-ENUM) specifies that=0A   "In=
frastructure ENUM SHALL support RRs providing a URI that can=0A   identify =
a point of interconnection for delivery to the carrier-of-=0A   record of c=
ommunications addressed to the E.164 number."=0A=0A   However storing a dir=
ect end-point URI in a publicly visible I-ENUM=0A   tree poses two problems=
:=0A=0A   1.  It does not permit a different answer to be given to differen=
t=0A       interconnect partners.=0A   2.  It exposes information about int=
erconnection infrastructure to=0A       parties who are not expected to hav=
e this information.=0A=0A   These issues are not explicitly addressed (nor =
prohibited) by the=0A   I-ENUM Requirements, but are well understood.  See =
e.g.  [ghuston]=0A=0A   Both issues could in theory be resolved by using DN=
S views to return=0A   different answers to different DNS clients.  However=
 this is not=0A   considered to scale well, particularly when the number of=
 views=0A   required might number in the hundreds or thousands.  Nor would =
this=0A   be practical if parts of the I-ENUM database are managed by a cen=
tral=0A   authority without delegation of individual number ranges to the=
=0A   terminating carriers.=0A=0A   The proposed solution is therefore to r=
egister a new Enumservice and=0A   URI type (in line with [I-D.ietf-enum-en=
umservices-guide]) which=0A   returns an opaque reference to the real inter=
connection data.  The=0A   interpretation of that opaque reference is then =
a matter of bi-=0A   lateral agreement between carriers.=0A=0A=0A2.  Termin=
ology=0A=0A   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL=
 NOT",=0A   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in=
 this=0A   document are to be interpreted as described in [RFC2119].=0A=0A=
=0A3.  ENUM Service Registration for "Destination Group"=0A=0A   The follow=
ing template contains information required for the IANA=0A   registrations =
of the 'Destination Group' Enumservice:=0A=0A   Enumservice Name: Destinati=
on Group=0A=0A=0A=0A=0ABellis                  Expires November 3, 2008    =
            [Page 3]=0A=0C=0AInternet-Draft        Destination Group Enumse=
rvice             May 2008=0A=0A=0A   Enum service Class: Ancillary Applica=
tion Enumservice=0A=0A   Enumservice Type: DG=0A=0A   Enumservice Subtype: =
n/a=0A=0A   URI Schemes: dg=0A=0A   Functional Specification: TODO=0A=0A   =
Security Considerations: see Section 8=0A=0A   Intended Usage: COMMON=0A=0A=
   Author: Ray Bellis <mailto:ray.bellis@nominet.org.uk>=0A=0A=0A4.  IANA R=
egistration Template for URI Scheme "dg"=0A=0A   URI scheme name: dg=0A=0A =
  Status: provisional=0A=0A   URI scheme syntax: (in ABNF [RFC5234])=0A=0A=
=0A                    dguri   =3D ( "dg:" dglabel )=0A                    =
dglabel =3D 1*255dgchar=0A                    dgchar  =3D ALPHA / DIGIT / "=
." / "-"=0A=0A=0A   URI scheme semantics: The URI contains opaque informati=
on that is=0A   used as a lookup key into an interconnection database.=0A=
=0A   Encoding considerations: None - all valid characters are in the US=0A=
   ASCII character set.=0A=0A   Applications: Infrastructure ENUM [I-D.ietf=
-enum-infrastructure]=0A=0A   Interoperability considerations: none=0A=0A  =
 Security considerations: see Section 8=0A=0A   Contact: Ray Bellis <mailto=
:ray.bellis@nominet.org.uk>=0A=0A=0A=0A=0A=0A=0A=0ABellis                  =
Expires November 3, 2008                [Page 4]=0A=0C=0AInternet-Draft    =
    Destination Group Enumservice             May 2008=0A=0A=0A5.  Descript=
ion=0A=0A   A Destination Group URI must be globally unique.  To avoid the =
need=0A   for a register of known Destination Groups it is RECOMMENDED that=
=0A   each carrier generate a locally unique string for each of their=0A   =
Destination Groups and then append the name of a domain name over=0A   whic=
h they have control.=0A=0A   Should an I-ENUM lookup return a Destination G=
roup URI which is=0A   unknown to the originating carrier then the originat=
ing carrier=0A   SHOULD assume that no VoIP interconnect is available and t=
he call=0A   delivered over other interconnect methods (e.g.  PSTN)=0A=0A=
=0A6.  Examples=0A=0A   TODO=0A=0A=0A7.  DNS Considerations=0A=0A   TODO=0A=
=0A   Notes - only one DG NAPTR per RRset, thus order / priority fields=0A =
  irrelevant.=0A=0A   Q: could multiple NAPTRs be used to allow alternate r=
outes?=0A=0A   Wildcards - OK=0A=0A   Non-empty leaf nodes (single numbers =
or blocks) - OK=0A=0A   Intermediate nodes - not OK=0A=0A=0A8.  Security Co=
nsiderations=0A=0A   TODO=0A=0A=0A9.  IANA Considerations=0A=0A   This docu=
ment requests the IANA registration of the Enumservice=0A   'Destination Gr=
oup' with Type 'DG' according to the definitions in=0A   this document, RFC=
xxxx [I-D.ietf-enum-enumservices-guide] and=0A   RFC3761bis [I-D.ietf-enum-=
3761bis].  The required template is=0A   contained in Section 3.=0A=0A=0A=
=0A=0ABellis                  Expires November 3, 2008                [Page=
 5]=0A=0C=0AInternet-Draft        Destination Group Enumservice            =
 May 2008=0A=0A=0A   This document requests an update to the IANA registrat=
ion of the URI=0A   scheme 'dg' according to the definitions in this docume=
nt and=0A   following the process described in [RFC4395].  The required tem=
plate=0A   is contain in Section 4.=0A=0A=0A10.  Change Log=0A=0A   draft-b=
ellis-drinks-destgroups-00=0A      Initial draft=0A=0A=0A11.  Acknowledgeme=
nts=0A=0A=0A12.  References=0A=0A12.1.  Normative References=0A=0A   [I-D.i=
etf-enum-3761bis]=0A              Bradner, S., Conroy, L., and K. Fujiwara,=
 "The E.164 to=0A              Uniform Resource Identifiers (URI) Dynamic D=
elegation=0A              Discovery  System (DDDS) Application (ENUM)",=0A =
             draft-ietf-enum-3761bis-03 (work in progress), March 2008.=0A=
=0A   [I-D.ietf-enum-infrastructure]=0A              Livingood, J., "The E.=
164 to Uniform Resource Identifiers=0A              (URI) Dynamic Delegatio=
n Discovery  System (DDDS)=0A              Application for Infrastructure E=
NUM",=0A              draft-ietf-enum-infrastructure-07 (work in progress),=
=0A              December 2007.=0A=0A   [RFC2119]  Bradner, S., "Key words =
for use in RFCs to Indicate=0A              Requirement Levels", BCP 14, RF=
C 2119, March 1997.=0A=0A   [RFC5067]  Lind, S. and P. Pfautz, "Infrastruct=
ure ENUM=0A              Requirements", RFC 5067, November 2007.=0A=0A   [R=
FC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax=0A          =
    Specifications: ABNF", STD 68, RFC 5234, January 2008.=0A=0A12.2.  Info=
rmative References=0A=0A   [I-D.ietf-enum-enumservices-guide]=0A           =
   Hoeneisen, B., Mayrhofer, A., and J. Livingood, "IANA=0A              Re=
gistration of Enumservices: Guide, Template and IANA=0A              Consid=
erations", draft-ietf-enum-enumservices-guide-09=0A              (work in p=
rogress), May 2008.=0A=0A=0A=0ABellis                  Expires November 3, =
2008                [Page 6]=0A=0C=0AInternet-Draft        Destination Grou=
p Enumservice             May 2008=0A=0A=0A   [RFC4395]  Hansen, T., Hardie=
, T., and L. Masinter, "Guidelines and=0A              Registration Procedu=
res for New URI Schemes", BCP 115,=0A              RFC 4395, February 2006.=
=0A=0A   [ghuston]  Huston, G., "Infrastructure ENUM", April 2007,=0A      =
        <http://www.circleid.com/posts/infrastructure=5Fenum/>.=0A=0A=0AAut=
hor's Address=0A=0A   Ray Bellis=0A   Nominet UK=0A   Edmund Halley Road=0A=
   Oxford  OX4 4DQ=0A   United Kingdom=0A=0A   Phone: +44 1865 332211=0A   =
Email: ray.bellis@nominet.org.uk=0A   URI:   http://www.nominet.org.uk/=0A=
=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=
=0A=0A=0A=0A=0A=0A=0ABellis                  Expires November 3, 2008      =
          [Page 7]=0A=0C=0AInternet-Draft        Destination Group Enumserv=
ice             May 2008=0A=0A=0AFull Copyright Statement=0A=0A   Copyright=
 (C) The IETF Trust (2008).=0A=0A   This document is subject to the rights,=
 licenses and restrictions=0A   contained in BCP 78, and except as set fort=
h therein, the authors=0A   retain all their rights.=0A=0A   This document =
and the information contained herein are provided on an=0A   "AS IS" basis =
and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS=0A   OR IS SPONSORE=
D BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND=0A   THE INTERNET E=
NGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS=0A   OR IMPLIED, INC=
LUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF=0A   THE INFORMATION=
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED=0A   WARRANTIES OF MERC=
HANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.=0A=0A=0AIntellectual Prope=
rty=0A=0A   The IETF takes no position regarding the validity or scope of a=
ny=0A   Intellectual Property Rights or other rights that might be claimed =
to=0A   pertain to the implementation or use of the technology described in=
=0A   this document or the extent to which any license under such rights=0A=
   might or might not be available; nor does it represent that it has=0A   =
made any independent effort to identify any such rights.  Information=0A   =
on the procedures with respect to rights in RFC documents can be=0A   found=
 in BCP 78 and BCP 79.=0A=0A   Copies of IPR disclosures made to the IETF S=
ecretariat and any=0A   assurances of licenses to be made available, or the=
 result of an=0A   attempt made to obtain a general license or permission f=
or the use of=0A   such proprietary rights by implementers or users of this=
=0A   specification can be obtained from the IETF on-line IPR repository at=
=0A   http://www.ietf.org/ipr.=0A=0A   The IETF invites any interested part=
y to bring to its attention any=0A   copyrights, patents or patent applicat=
ions, or other proprietary=0A   rights that may cover technology that may b=
e required to implement=0A   this standard.  Please address the information=
 to the IETF at=0A   ietf-ipr@ietf.org.=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=0A=
Bellis                  Expires November 3, 2008                [Page 8]=0A=
=0C=0A=
--=_mixed 003511FF802576E0_=--

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To: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 8 Mar 2010 15:32:46 +0100
Thread-Topic: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Dear Dean, all,

Please see inline.
Cheers,
Med

________________________________
De : Dean Willis [mailto:dean.willis@softarmor.com]
Envoy=E9 : vendredi 5 mars 2010 19:16
=C0 : BOUCADAIR Mohamed NCPI/NAD/TIP
Cc : Daryl Malas; dispatch@ietf.org
Objet : Re: [dispatch] Why are we TRIPped out? Is there an sip-event altern=
ative?


On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com<mailto:mo=
hamed.boucadair@orange-ftgroup.com>> <mohamed.boucadair@orange-ftgroup.com<=
mailto:mohamed.boucadair@orange-ftgroup.com>> wrote:


Dear Dean, all,

In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interc=
onnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp).
Med: http://www.eurescom.eu/message/messageMay2009/Interconnection_challeng=
es_of_IP_telephony_Eurescom_study_P1853.asp<http://www.eurescom.eu/message/=
messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_study_P1=
853.asp)>
I'm having a hard time resolving this link, as it appears to be 404. Perhap=
s there's a typo?

With respect to:


21. Extended TRIP protocol is suggested to be used as base for IP Telephony=
 routing;

We've heard here that TRIP, as currently modeled, doesn't work due to the L=
NP problem. What sorts of extensions id you have in mind? Were they intende=
d to address this problem?
[Med] An alternative would be to use TRIP in conjunction with a the current=
 Number Portability infrastructure as used in PSTN infrastructure for placi=
ng inter-ITAD calls (this means that the owner domain will always advertise=
 its aggregate in TRIP, but a NPDB (Network Portability Database) should qu=
estioned first).

Examples of extensions to TRIP (or whatever dynamic telephony routing) woul=
d be:
- Avoid AS (Autonomous System) spiral: the media traffic can cross the same=
 AS several times
- Ability to maintain multiple inter-ITAD paths
- Ability to enforce some advanced TE such as: Avoid that my VoIP traffic t=
o cross a given AS domain (TRIP only maintain an ITAD list)
- Ability to avoid congestionned links
- etc.

BTW, TRIP is only considered as starting point, this does not preclude to i=
nvestigate any new solution to meet the recommendations we identified. BTW,=
 In the recommendations list we have identified the need to encourage aggre=
gation-based models.


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Any unauthorised use or dissemination is prohibited.
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France Telecom Group shall not be liable for the message if altered, change=
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If you are not the intended addressee of this message, please cancel it imm=
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3527" name=3DGENERATOR></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; webkit-nbsp-mode: space; webkit-line-break:=
 after-white-space">
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D059155613-08032010><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>Dear Dean, all,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D059155613-08032010><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D059155613-08032010><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>Please see inline.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D059155613-08032010><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>Cheers,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D059155613-08032010><FONT face=3D"=
Courier New"=20
color=3D#0000ff size=3D2>Med</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> Dean Willis=20
[mailto:dean.willis@softarmor.com] <BR><B>Envoy=E9&nbsp;:</B> vendredi 5 ma=
rs 2010=20
19:16<BR><B>=C0&nbsp;:</B> BOUCADAIR Mohamed NCPI/NAD/TIP<BR><B>Cc&nbsp;:</=
B>=20
Daryl Malas; dispatch@ietf.org<BR><B>Objet&nbsp;:</B> Re: [dispatch] Why ar=
e we=20
TRIPped out? Is there an sip-event alternative?<BR></FONT><BR></DIV>
<DIV></DIV><BR>
<DIV>
<DIV>On Mar 5, 2010, at 1:38 AM, &lt;<A=20
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@oran=
ge-ftgroup.com</A>&gt;=20
&lt;<A=20
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@oran=
ge-ftgroup.com</A>&gt;=20
wrote:</DIV><BR class=3DApple-interchange-newline>
<BLOCKQUOTE type=3D"cite">
  <DIV><BR>Dear Dean, all,<BR><BR>In the last year, we have undertaken a st=
udy=20
  about IP telephony interconnect at large scale (URL: <A=20
  href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_cha=
llenges_of_IP_telephony_Eurescom_study_P1853.asp)">http://www.eurescom.eu/m=
essage/messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_s=
tudy_P1853.asp)</A>.</DIV></BLOCKQUOTE>
<DIV><SPAN class=3D059155613-08032010></SPAN><FONT face=3D"Courier New"><FO=
NT=20
color=3D#0000ff><FONT size=3D2>M<SPAN class=3D059155613-08032010>ed: <A=20
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_chall=
enges_of_IP_telephony_Eurescom_study_P1853.asp)"><FONT=20
face=3D"Times New Roman"=20
size=3D3>http://www.eurescom.eu/message/messageMay2009/Interconnection_chal=
lenges_of_IP_telephony_Eurescom_study_P1853.asp</FONT></A>&nbsp;</SPAN><SPA=
N=20
class=3D059155613-08032010>&nbsp;</SPAN></FONT></FONT></FONT><BR></DIV>
<DIV>I'm having a hard time resolving this link, as it appears to be 404.=
=20
Perhaps there's a typo?</DIV>
<DIV><FONT face=3D"Courier New" color=3D#0000ff size=3D2></FONT><BR></DIV>
<DIV>With respect to:</DIV>
<DIV><FONT face=3D"Courier New" color=3D#0000ff size=3D2></FONT><BR></DIV>
<BLOCKQUOTE type=3D"cite">
  <DIV><FONT class=3DApple-style-span color=3D#000000><FONT face=3D"Courier=
 New"=20
  color=3D#0000ff size=3D2></FONT><BR></FONT></DIV>
  <DIV>21.<SPAN class=3DApple-tab-span style=3D"WHITE-SPACE: pre"> </SPAN>E=
xtended=20
  TRIP protocol is suggested to be used as base for IP Telephony=20
  routing;<BR></DIV></BLOCKQUOTE></DIV><BR>
<DIV>We've heard here that TRIP, as currently modeled, doesn't work due to =
the=20
LNP problem. What sorts of extensions id you have in mind? Were they intend=
ed to=20
address this problem?<BR><SPAN class=3D059155613-08032010><FONT face=3D"Cou=
rier New"=20
color=3D#0000ff size=3D2></FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>[Med]&nbsp;An alternative would be to use TRIP&nbsp;in conjunction=
 with a=20
the current&nbsp;Number Portability infrastructure&nbsp;as used in&nbsp;PST=
N=20
infrastructure&nbsp;for&nbsp;placing&nbsp;inter-ITAD calls (this means that=
 the=20
owner domain will always advertise its aggregate in TRIP, but&nbsp;a NPDB=
=20
(Network Portability Database) should questioned=20
first).&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>Examples of extensions to TRIP (or whatever dynamic telephony rout=
ing)=20
would be:</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>- Avoid AS (Autonomous System) spiral: the media traffic can cross=
 the=20
same AS several times</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>- Ability to maintain multiple inter-ITAD paths</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>- Ability to enforce some advanced TE such as: Avoid that my VoIP =
traffic=20
to cross a given&nbsp;AS domain (TRIP only maintain an ITAD=20
list)</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>- Ability to avoid congestionned links</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>- etc.</FONT></SPAN></DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D059155613-08032010><FONT face=3D"Courier New" color=3D#0=
000ff=20
size=3D2>BTW, TRIP is only considered as starting point, this does not prec=
lude to=20
investigate any new solution to meet the recommendations we identified. BTW=
, In=20
the recommendations list we have identified the need to encourage=20
aggregation-based models.</FONT></SPAN></DIV>
<DIV><FONT face=3D"Courier New" color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV><PRE>*********************************
This message and any attachments (the "message") are confidential and inten=
ded solely for the addressees.=20
Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration.=20
France Telecom Group shall not be liable for the message if altered, change=
d or falsified.
If you are not the intended addressee of this message, please cancel it imm=
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</PRE></BODY></HTML>

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From: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>
To: <mohamed.boucadair@orange-ftgroup.com>, "Dean Willis" <dean.willis@softarmor.com>
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Just a word of caution re number portability. In some countries access =
to this data is restricted to legally qualified carriers.  Will ITADs =
always correspond to such entities? Looking at the IANA list so far =
that's not my impression. You may find that the entities that do have =
access to NP databases may not be the ones interested in a TRIP-like =
mechanism and vice versa.

=20

Penn Pfautz

AT&T Access Management

+1-732-420-4962

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On =
Behalf Of mohamed.boucadair@orange-ftgroup.com
Sent: Monday, March 08, 2010 9:33 AM
To: Dean Willis
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event =
alternative?

=20

=20

Dear Dean, all,

=20

Please see inline.

Cheers,

Med

=20

________________________________

De : Dean Willis [mailto:dean.willis@softarmor.com]=20
Envoy=E9 : vendredi 5 mars 2010 19:16
=C0 : BOUCADAIR Mohamed NCPI/NAD/TIP
Cc : Daryl Malas; dispatch@ietf.org
Objet : Re: [dispatch] Why are we TRIPped out? Is there an sip-event =
alternative?

=20

On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com> =
<mohamed.boucadair@orange-ftgroup.com> wrote:






Dear Dean, all,

In the last year, we have undertaken a study about IP telephony =
interconnect at large scale (URL: =
http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_=
of_IP_telephony_Eurescom_study_P1853.asp).

Med: =
http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_=
of_IP_telephony_Eurescom_study_P1853.asp =
<http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges=
_of_IP_telephony_Eurescom_study_P1853.asp)>  =20

I'm having a hard time resolving this link, as it appears to be 404. =
Perhaps there's a typo?

=20

With respect to:

=20

	=20

	21. Extended TRIP protocol is suggested to be used as base for IP =
Telephony routing;

=20

We've heard here that TRIP, as currently modeled, doesn't work due to =
the LNP problem. What sorts of extensions id you have in mind? Were they =
intended to address this problem?

[Med] An alternative would be to use TRIP in conjunction with a the =
current Number Portability infrastructure as used in PSTN infrastructure =
for placing inter-ITAD calls (this means that the owner domain will =
always advertise its aggregate in TRIP, but a NPDB (Network Portability =
Database) should questioned first).=20

=20

Examples of extensions to TRIP (or whatever dynamic telephony routing) =
would be:

- Avoid AS (Autonomous System) spiral: the media traffic can cross the =
same AS several times

- Ability to maintain multiple inter-ITAD paths

- Ability to enforce some advanced TE such as: Avoid that my VoIP =
traffic to cross a given AS domain (TRIP only maintain an ITAD list)

- Ability to avoid congestionned links

- etc.

=20

BTW, TRIP is only considered as starting point, this does not preclude =
to investigate any new solution to meet the recommendations we =
identified. BTW, In the recommendations list we have identified the need =
to encourage aggregation-based models.

=20

*********************************
This message and any attachments (the "message") are confidential and =
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Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration.=20
France Telecom Group shall not be liable for the message if altered, =
changed or falsified.
If you are not the intended addressee of this message, please cancel it =
immediately and inform the sender.
********************************

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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Just a word of caution re number portability. In some =
countries
access to this data is restricted to legally qualified carriers.=A0 Will =
ITADs
always correspond to such entities? Looking at the IANA list so far =
that&#8217;s
not my impression. You may find that the entities that do have access to =
NP
databases may not be the ones interested in a TRIP-like mechanism and =
vice
versa.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Penn Pfautz</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>AT&amp;T Access Management</span><span =
style=3D'color:#1F497D'><o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>+1-732-420-4962</span><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On =
Behalf Of </b>mohamed.boucadair@orange-ftgroup.com<br>
<b>Sent:</b> Monday, March 08, 2010 9:33 AM<br>
<b>To:</b> Dean Willis<br>
<b>Cc:</b> dispatch@ietf.org<br>
<b>Subject:</b> Re: [dispatch] Why are we TRIPped out? Is there an =
sip-event
alternative?<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Dear Dean, all,</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Please see inline.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Cheers,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Med</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><span =
lang=3DFR>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span lang=3DFR
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De&nbsp;:</s=
pan></b><span
lang=3DFR style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Dean Willis
[mailto:dean.willis@softarmor.com] <br>
<b>Envoy=E9&nbsp;:</b> vendredi 5 mars 2010 19:16<br>
<b>=C0&nbsp;:</b> BOUCADAIR Mohamed NCPI/NAD/TIP<br>
<b>Cc&nbsp;:</b> Daryl Malas; dispatch@ietf.org<br>
<b>Objet&nbsp;:</b> Re: [dispatch] Why are we TRIPped out? Is there an =
sip-event
alternative?</span><span lang=3DFR><o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal>On Mar 5, 2010, at 1:38 AM, &lt;<a
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@or=
ange-ftgroup.com</a>&gt;
&lt;<a =
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@or=
ange-ftgroup.com</a>&gt;
wrote:<o:p></o:p></p>

</div>

<p class=3DMsoNormal><br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal><br>
Dear Dean, all,<br>
<br>
In the last year, we have undertaken a study about IP telephony =
interconnect at
large scale (URL: <a
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_cha=
llenges_of_IP_telephony_Eurescom_study_P1853.asp)">http://www.eurescom.eu=
/message/messageMay2009/Interconnection_challenges_of_IP_telephony_Euresc=
om_study_P1853.asp)</a>.<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Med: <a
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_cha=
llenges_of_IP_telephony_Eurescom_study_P1853.asp)"><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>http://www.eurescom.eu/message/messageMay2009/Interconnec=
tion_challenges_of_IP_telephony_Eurescom_study_P1853.asp</span></a>&nbsp;=
&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>I'm having a hard time resolving this link, as it =
appears to
be 404. Perhaps there's a typo?<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>With respect to:<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<div>

<p class=3DMsoNormal>21.<span class=3Dapple-tab-span> </span>Extended =
TRIP protocol
is suggested to be used as base for IP Telephony routing;<o:p></o:p></p>

</div>

</blockquote>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>We've heard here that TRIP, as currently modeled, =
doesn't
work due to the LNP problem. What sorts of extensions id you have in =
mind? Were
they intended to address this problem?<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>[Med]&nbsp;An alternative would be to use TRIP&nbsp;in =
conjunction
with a the current&nbsp;Number Portability infrastructure&nbsp;as used
in&nbsp;PSTN infrastructure&nbsp;for&nbsp;placing&nbsp;inter-ITAD calls =
(this
means that the owner domain will always advertise its aggregate in TRIP,
but&nbsp;a NPDB (Network Portability Database) should questioned =
first).&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>Examples of extensions to TRIP (or whatever dynamic =
telephony
routing) would be:</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>- Avoid AS (Autonomous System) spiral: the media traffic can =
cross
the same AS several times</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>- Ability to maintain multiple inter-ITAD =
paths</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>- Ability to enforce some advanced TE such as: Avoid that my =
VoIP
traffic to cross a given&nbsp;AS domain (TRIP only maintain an ITAD =
list)</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>- Ability to avoid congestionned links</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>- etc.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:blue'>BTW, TRIP is only considered as starting point, this does =
not
preclude to investigate any new solution to meet the recommendations we
identified. BTW, In the recommendations list we have identified the need =
to
encourage aggregation-based models.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<pre>*********************************<o:p></o:p></pre><pre>This message =
and any attachments (the &quot;message&quot;) are confidential and =
intended solely for the addressees. <o:p></o:p></pre><pre>Any =
unauthorised use or dissemination is =
prohibited.<o:p></o:p></pre><pre>Messages are susceptible to alteration. =
<o:p></o:p></pre><pre>France Telecom Group shall not be liable for the =
message if altered, changed or falsified.<o:p></o:p></pre><pre>If you =
are not the intended addressee of this message, please cancel it =
immediately and inform the =
sender.<o:p></o:p></pre><pre>********************************<o:p></o:p><=
/pre></div>

</body>

</html>

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From: <mohamed.boucadair@orange-ftgroup.com>
To: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>, Dean Willis <dean.willis@softarmor.com>
Date: Tue, 9 Mar 2010 08:16:44 +0100
Thread-Topic: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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Dear Penn,

I fully agree with you about this "legal" frontier.

Then, do such ITADs (which have no access to NPDB) concerned about NP? Do t=
hese operators offer telephony services or only VoIP (i.e., no emergency ca=
lls, no NP, no legal intercept, and all the legal requirements)?

If these domains want to offer global telephony reachability they must have=
 contract(s) with one or several PSTN (or mobile) operators (seen as defaul=
t route). NP should be handled downstream.

Cheers,
Med

________________________________
De : PFAUTZ, PENN L (ATTCORP) [mailto:pp3129@att.com]
Envoy=E9 : mardi 9 mars 2010 00:27
=C0 : BOUCADAIR Mohamed NCPI/NAD/TIP; Dean Willis
Cc : dispatch@ietf.org
Objet : RE: [dispatch] Why are we TRIPped out? Is there an sip-event altern=
ative?

Just a word of caution re number portability. In some countries access to t=
his data is restricted to legally qualified carriers.  Will ITADs always co=
rrespond to such entities? Looking at the IANA list so far that's not my im=
pression. You may find that the entities that do have access to NP database=
s may not be the ones interested in a TRIP-like mechanism and vice versa.

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of mohamed.boucadair@orange-ftgroup.com
Sent: Monday, March 08, 2010 9:33 AM
To: Dean Willis
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alter=
native?


Dear Dean, all,

Please see inline.
Cheers,
Med

________________________________
De : Dean Willis [mailto:dean.willis@softarmor.com]
Envoy=E9 : vendredi 5 mars 2010 19:16
=C0 : BOUCADAIR Mohamed NCPI/NAD/TIP
Cc : Daryl Malas; dispatch@ietf.org
Objet : Re: [dispatch] Why are we TRIPped out? Is there an sip-event altern=
ative?

On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com<mailto:mo=
hamed.boucadair@orange-ftgroup.com>> <mohamed.boucadair@orange-ftgroup.com<=
mailto:mohamed.boucadair@orange-ftgroup.com>> wrote:



Dear Dean, all,

In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interc=
onnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp).
Med: http://www.eurescom.eu/message/messageMay2009/Interconnection_challeng=
es_of_IP_telephony_Eurescom_study_P1853.asp<http://www.eurescom.eu/message/=
messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_study_P1=
853.asp)>
I'm having a hard time resolving this link, as it appears to be 404. Perhap=
s there's a typo?

With respect to:


21. Extended TRIP protocol is suggested to be used as base for IP Telephony=
 routing;

We've heard here that TRIP, as currently modeled, doesn't work due to the L=
NP problem. What sorts of extensions id you have in mind? Were they intende=
d to address this problem?
[Med] An alternative would be to use TRIP in conjunction with a the current=
 Number Portability infrastructure as used in PSTN infrastructure for placi=
ng inter-ITAD calls (this means that the owner domain will always advertise=
 its aggregate in TRIP, but a NPDB (Network Portability Database) should qu=
estioned first).

Examples of extensions to TRIP (or whatever dynamic telephony routing) woul=
d be:
- Avoid AS (Autonomous System) spiral: the media traffic can cross the same=
 AS several times
- Ability to maintain multiple inter-ITAD paths
- Ability to enforce some advanced TE such as: Avoid that my VoIP traffic t=
o cross a given AS domain (TRIP only maintain an ITAD list)
- Ability to avoid congestionned links
- etc.

BTW, TRIP is only considered as starting point, this does not preclude to i=
nvestigate any new solution to meet the recommendations we identified. BTW,=
 In the recommendations list we have identified the need to encourage aggre=
gation-based models.


*********************************

This message and any attachments (the "message") are confidential and inten=
ded solely for the addressees.

Any unauthorised use or dissemination is prohibited.

Messages are susceptible to alteration.

France Telecom Group shall not be liable for the message if altered, change=
d or falsified.

If you are not the intended addressee of this message, please cancel it imm=
ediately and inform the sender.

********************************

*********************************
This message and any attachments (the "message") are confidential and inten=
ded solely for the addressees.=20
Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration.=20
France Telecom Group shall not be liable for the message if altered, change=
d or falsified.
If you are not the intended addressee of this message, please cancel it imm=
ediately and inform the sender.
********************************


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<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>Dear Penn,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>I fully agree with you about this "legal"=20
frontier.&nbsp;</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>Then, do such ITADs (which have no access to NPD=
B)=20
concerned about NP?&nbsp;Do these operators offer telephony services or onl=
y=20
VoIP (i.e., no emergency calls, no NP, no legal intercept, and all the lega=
l=20
requirements)? </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>If these&nbsp;domains&nbsp;want to offer global=
=20
telephony reachability they must have contract(s) with one or several PSTN =
(or=20
mobile) operators (seen as default route). NP&nbsp;should be=20
handled&nbsp;downstream.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>Cheers,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3D"Courier New" color=3D#0000ff size=
=3D2><SPAN=20
class=3D930295306-09032010>Med</SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dfr dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>De&nbsp;:</B> PFAUTZ, PENN L (ATTCORP)=20
[mailto:pp3129@att.com] <BR><B>Envoy=E9&nbsp;:</B> mardi 9 mars 2010=20
00:27<BR><B>=C0&nbsp;:</B> BOUCADAIR Mohamed NCPI/NAD/TIP; Dean=20
Willis<BR><B>Cc&nbsp;:</B> dispatch@ietf.org<BR><B>Objet&nbsp;:</B> RE:=20
[dispatch] Why are we TRIPped out? Is there an sip-event=20
alternative?<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'">Just=20
a word of caution re number portability. In some countries access to this d=
ata=20
is restricted to legally qualified carriers.&nbsp; Will ITADs always corres=
pond=20
to such entities? Looking at the IANA list so far that=92s not my impressio=
n. You=20
may find that the entities that do have access to NP databases may not be t=
he=20
ones interested in a TRIP-like mechanism and vice versa.<o:p></o:p></SPAN><=
/P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Arial','sans-serif'=
">Penn=20
Pfautz</SPAN><SPAN style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Arial','sans-serif'=
">AT&amp;T=20
Access Management</SPAN><SPAN=20
style=3D"COLOR: #1f497d"><o:p></o:p></SPAN></P></DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: #1f497d; FONT-FAMILY: 'Arial','sans-serif'=
">+1-732-420-4962</SPAN><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-seri=
f'"><o:p></o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df=
 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium non=
e; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B>On Behalf O=
f=20
</B>mohamed.boucadair@orange-ftgroup.com<BR><B>Sent:</B> Monday, March 08, =
2010=20
9:33 AM<BR><B>To:</B> Dean Willis<BR><B>Cc:</B>=20
dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] Why are we TRIPped out?=
 Is=20
there an sip-event alternative?<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Dear Dea=
n,=20
all,</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Please s=
ee=20
inline.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Cheers,<=
/SPAN><o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Med</SPA=
N><o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" align=3Dcenter><SPAN la=
ng=3DFR>
<HR align=3Dcenter width=3D"100%" SIZE=3D2>
</SPAN></DIV>
<P class=3DMsoNormal style=3D"MARGIN-BOTTOM: 12pt"><B><SPAN lang=3DFR=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">De&nbsp;:</SP=
AN></B><SPAN=20
lang=3DFR style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> De=
an Willis=20
[mailto:dean.willis@softarmor.com] <BR><B>Envoy=E9&nbsp;:</B> vendredi 5 ma=
rs 2010=20
19:16<BR><B>=C0&nbsp;:</B> BOUCADAIR Mohamed NCPI/NAD/TIP<BR><B>Cc&nbsp;:</=
B>=20
Daryl Malas; dispatch@ietf.org<BR><B>Objet&nbsp;:</B> Re: [dispatch] Why ar=
e we=20
TRIPped out? Is there an sip-event alternative?</SPAN><SPAN=20
lang=3DFR><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<DIV>
<P class=3DMsoNormal>On Mar 5, 2010, at 1:38 AM, &lt;<A=20
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@oran=
ge-ftgroup.com</A>&gt;=20
&lt;<A=20
href=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@oran=
ge-ftgroup.com</A>&gt;=20
wrote:<o:p></o:p></P></DIV>
<P class=3DMsoNormal><BR><BR><o:p></o:p></P>
<DIV>
<P class=3DMsoNormal><BR>Dear Dean, all,<BR><BR>In the last year, we have=
=20
undertaken a study about IP telephony interconnect at large scale (URL: <A=
=20
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_chall=
enges_of_IP_telephony_Eurescom_study_P1853.asp)">http://www.eurescom.eu/mes=
sage/messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_stu=
dy_P1853.asp)</A>.<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Med: <A=
=20
href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_chall=
enges_of_IP_telephony_Eurescom_study_P1853.asp)"><SPAN=20
style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman','serif'">http://ww=
w.eurescom.eu/message/messageMay2009/Interconnection_challenges_of_IP_telep=
hony_Eurescom_study_P1853.asp</SPAN></A>&nbsp;&nbsp;</SPAN><o:p></o:p></P><=
/DIV>
<DIV>
<P class=3DMsoNormal>I'm having a hard time resolving this link, as it appe=
ars to=20
be 404. Perhaps there's a typo?<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>With respect to:<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
<BLOCKQUOTE style=3D"MARGIN-TOP: 5pt; MARGIN-BOTTOM: 5pt">
  <DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal>21.<SPAN class=3Dapple-tab-span> </SPAN>Extended TRI=
P=20
  protocol is suggested to be used as base for IP Telephony=20
  routing;<o:p></o:p></P></DIV></BLOCKQUOTE></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal>We've heard here that TRIP, as currently modeled, does=
n't=20
work due to the LNP problem. What sorts of extensions id you have in mind? =
Were=20
they intended to address this problem?<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">[Med]&nb=
sp;An=20
alternative would be to use TRIP&nbsp;in conjunction with a the=20
current&nbsp;Number Portability infrastructure&nbsp;as used in&nbsp;PSTN=20
infrastructure&nbsp;for&nbsp;placing&nbsp;inter-ITAD calls (this means that=
 the=20
owner domain will always advertise its aggregate in TRIP, but&nbsp;a NPDB=
=20
(Network Portability Database) should questioned=20
first).&nbsp;</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">Examples=
 of=20
extensions to TRIP (or whatever dynamic telephony routing) would=20
be:</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">- Avoid =
AS=20
(Autonomous System) spiral: the media traffic can cross the same AS several=
=20
times</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">- Abilit=
y to=20
maintain multiple inter-ITAD paths</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">- Abilit=
y to=20
enforce some advanced TE such as: Avoid that my VoIP traffic to cross a=20
given&nbsp;AS domain (TRIP only maintain an ITAD=20
list)</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">- Abilit=
y to=20
avoid congestionned links</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">-=20
etc.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: 'Courier New'">BTW, TRI=
P is=20
only considered as starting point, this does not preclude to investigate an=
y new=20
solution to meet the recommendations we identified. BTW, In the recommendat=
ions=20
list we have identified the need to encourage aggregation-based=20
models.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV><PRE>**********************=
***********<o:p></o:p></PRE><PRE>This message and any attachments (the "mes=
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--_000_94C682931C08B048B7A8645303FDC9F30EFC01829BPUEXCB1Bnante_--

From D.Malas@cablelabs.com  Tue Mar  9 08:04:49 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, Adam Roach <adam@nostrum.com>
Date: Tue, 9 Mar 2010 09:04:52 -0700
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Dean et. al,

I am thankful my draft spurred on this discussion on potentially resolving =
a larger issue.  With this being said, the purpose of the draft is to be a =
potential component of the larger solution.  I got lost on whether or not t=
he general community understands and sees the overall value of defining an =
egress method, for ensuring a consistent approach between SSPs and vendors.

Regards,

Daryl=20

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Thursday, March 04, 2010 9:16 AM
To: Adam Roach
Cc: Daryl Malas; 'dispatch@ietf.org'
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 3, 2010, at 3:42 PM, Adam Roach wrote:

> On 3/3/10 1:26 PM, Dean Willis wrote:
>>
>> We really need a SIP-targeted analog of the BGP/OSPF route-exchange=20
>> protocols to solve the general problem.
>
> So, what, like TRIP, but for things other than phone numbers?
>

Yep. Imagine a TRIP extended to say "I have a SIP peering agreement with do=
main X". Or better yet, imagine something that actually works and is widely=
 implemented that allows sharing such route information.

A caveat: we also have simple phone-number-peering issues, and TRIP hasn't =
caught on there. Therefore, I suspect that TRIP isn't quite right. A thorou=
gh post-mortem analysis might be interesting.

--
Dean

From dean.willis@softarmor.com  Tue Mar  9 09:28:05 2010
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On Mar 9, 2010, at 10:04 AM, Daryl Malas wrote:

> Dean et. al,
>
> I am thankful my draft spurred on this discussion on potentially  
> resolving a larger issue.  With this being said, the purpose of the  
> draft is to be a potential component of the larger solution.  I got  
> lost on whether or not the general community understands and sees  
> the overall value of defining an egress method, for ensuring a  
> consistent approach between SSPs and vendors.

I definitely see the need for defining a method by which to inform  
nodes inside an ITAD of egress routes from that ITAD to other domains.  
I think that most of the people who have responded on this thread  
agree that there is such a need.

We also seem to see a need for dissemination of inter-ITAD routing  
information, although we're still discussing the relationships between  
phone numbers, ITADs, and routes.

Where we have a disagreement is on mechanism. While your proposed use  
of ITAD-specific private ENUM  servers is not completely inconsistent  
with recent trends in ENUM, I have to say we've done a lot of things  
in ENUM that I don't like, and this is pushing the envelope further in  
that direction. There are a lot of people running private DNS spaces,  
too, but we haven't standardized the discovery of default routes  
within an Intranet using private DNS for very good reasons, and those  
same reasons apply to ENUM used in the same pattern. Yes, it can be  
made to work, but it creates massive operational headaches.


We have several arguments in favor of some sort of "dynamic routing  
protocol" solution or solutions. Open questions include:

1) Are different interior and exterior routing protocols (BGP vs OSPF,  
for instance) required, or can one common protocol meet the full set f  
requirements?

2) Is  a standalone protocol such as TRIP more appropriate than  
something "inside" SIP (or DNS, XMPP, whatever)? If so, does this  
apply to both interior and exterior roles?

3) Can TRIP be made to work? This has related questions about  
aggregation models, LNP access, LNP fragmentation of aggregation, etc.

4) Can inter-ITAD routing be accomplished based on number-to-ITAD  
transformations (which ENUM gives us) accompanied by a dynamic inter- 
ITAD route exchange protocol? This may also have to factor in the LNP  
problem.

5) Can we do something combining TRIP (maybe with modifications) and  
ENUM that solves the problems? If so, will anybody deploy it, or do  
they prefer striking themselves repeatedly in the face with blunt  
objects over extended periods of time?

All-in-all, this is excellent material for the sort of problem- 
exploration process that might lead to a working group.


--
Dean


From dean.willis@softarmor.com  Tue Mar  9 10:25:10 2010
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Cc: dispatch@ietf.org, Daryl Malas <D.Malas@cablelabs.com>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On Mar 3, 2010, at 10:08 PM, Dwight, Timothy M (Tim) wrote:

> Daryl,
>
> I agree that defining a best practice often has the impact you =20
> suggest.
>
> IMHO the technique you describe has merit.  What=92s not clear is how =20=

> a different response is generated per querying domain.  The draft =20
> shows response formats that would be appropriate for SSP1, but not =20
> appropriate for any other domain.  This implies that the response =20
> would be somehow tailored for each querying domain.  I think some =20
> elaboration on how that works (e.g., what has to be provisioned, by =20=

> whom, to make it happen) would be helpful.
>


Good questions. I'm assuming (and we all know where that leads) that =20
the following describes the solution as Daryl has proposed it:

SSP1 runs its own private ENUM database, which it fully manages and =20
provisions using whatever tools it has available. All nodes at SSP1 =20
query SSP1's ENUM servers, which return responses tailored to the SSP1 =20=

domain.

SSP2 runs its own private ENUM database, which it fully manages and =20
provisions using whatever tools it has available. All nodes at SSP2 =20
query SSP2's ENUM servers, which return responses tailored to the SSP2 =20=

domain.

...

SSPn runs its own private ENUM database, which it fully manages and =20
provisions using whatever tools it has available. All nodes at SSPn =20
query SSPn's ENUM servers, which return responses tailored to the SSPn =20=

domain.


Within each SSP, each querying node must be provisioned (via some =20
unspecified process) to use the SSP's private enum servers.

Like private DNS, none of the private ENUM servers is queried by the =20
"outside world".

Unlike private DNS the private ENUM servers also do not query the =20
outside world , because SIP routing lacks the concept of a "default =20
route". Essentially, we're hacking a DNS infrastructure to provide the =20=

routing-knowlegde that in an IP network would be provided by RIP and/=20
or DHCP. And that's what makes this much messier than private DNS, =20
which is already its own world of headaches. Each external route must =20=

be manually provisioned into each private-enum root server.

With this proposal, each private ENUM domain maintained by an SSP is a =20=

completely independent name-service island, equipped with its own root =20=

server. Some benefit of DNS caching and zone transfer may occur, as it =20=

is only necessary to manually provisioning the zone files into the =20
master or "primary" private-domain root server at each SSP. Secondary =20=

roots (once provisioned to use the correct primary) acquire zone files =20=

via zone transfer, and caching servers (once provisioned to use one or =20=

more roots) simply do a recursive lookup towards the local root and =20
cache the result until its TTL expires. But one is gaining little of =20
the benefit of a global name service, and the potential for ugly =20
mistakes is relatively high with any faked-root DNS model.

--
Dean



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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 3, 2010, =
at 10:08 PM, Dwight, Timothy M (Tim) wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-size: medium; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple"><div class=3D"Section1"><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"blue" =
face=3D"Courier New"><span style=3D"font-size: 10pt; font-family: =
'Courier New'; color: blue; ">Daryl,<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"blue" face=3D"Courier New"><span =
style=3D"font-size: 10pt; font-family: 'Courier New'; color: blue; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"blue" =
face=3D"Courier New"><span style=3D"font-size: 10pt; font-family: =
'Courier New'; color: blue; ">I agree that defining a best practice =
often has the impact you suggest.<o:p></o:p></span></font></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-bottom: 0.0001pt; =
margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman'; =
"><font size=3D"2" color=3D"blue" face=3D"Courier New"><span =
style=3D"font-size: 10pt; font-family: 'Courier New'; color: blue; =
"><o:p>&nbsp;</o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"blue" =
face=3D"Courier New"><span style=3D"font-size: 10pt; font-family: =
'Courier New'; color: blue; ">IMHO the technique you describe has merit. =
&nbsp;What=92s not clear is how a different response is generated per =
querying domain. &nbsp;The draft shows response formats that would be =
appropriate for SSP1, but not appropriate for any other domain.&nbsp; =
This implies that the response would be somehow tailored for each =
querying domain. &nbsp;I think some elaboration on how that works (e.g., =
what has to be provisioned, by whom, to make it happen) would be =
helpful.<o:p></o:p></span></font></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-bottom: 0.0001pt; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman'; "><font size=3D"2" color=3D"blue" =
face=3D"Courier New"><span style=3D"font-size: 10pt; font-family: =
'Courier New'; color: blue; =
"><o:p>&nbsp;</o:p></span></font></div></div></div></span></blockquote></d=
iv><br><div><br></div><div>Good questions. I'm assuming (and we all know =
where that leads) that the following describes the solution as Daryl has =
proposed it:</div><div><br></div><div>SSP1 runs its own private ENUM =
database, which it fully manages and provisions using whatever tools it =
has available. All nodes at SSP1 query SSP1's ENUM servers, which return =
responses tailored to the SSP1 =
domain.</div><div><br></div><div><div>SSP2 runs its own private ENUM =
database, which it fully manages and provisions using whatever tools it =
has available. All nodes at SSP2 query SSP2's ENUM servers, which return =
responses tailored to the SSP2 =
domain.</div><div><br></div><div>...</div><div><br></div><div>SSPn runs =
its own private ENUM database, which it fully manages and provisions =
using whatever tools it has available. All nodes at SSPn query SSPn's =
ENUM servers, which return responses tailored to the SSPn =
domain.</div><div><br></div><div><br></div><div>Within each SSP, each =
querying node must be provisioned (via some unspecified process) to use =
the SSP's private enum =
servers.&nbsp;</div><div><br></div></div><div>Like private DNS, none of =
the private ENUM servers is queried by the "outside =
world".&nbsp;</div><div><br></div><div>Unlike private DNS the private =
ENUM servers also do not query the outside world , because SIP routing =
lacks the concept of a "default route". Essentially, we're hacking a DNS =
infrastructure to provide the routing-knowlegde that in an IP network =
would be provided by RIP and/or DHCP. And that's what makes this much =
messier than private DNS, which is already its own world of headaches. =
Each external route must be manually provisioned into each private-enum =
root server.</div><div><br></div><div>With this proposal, each private =
ENUM domain maintained by an SSP is a completely independent =
name-service island, equipped with its own root server. Some benefit of =
DNS caching and zone transfer may occur, as it is only necessary to =
manually provisioning the zone files into the master or "primary" =
private-domain root server at each SSP. Secondary roots (once =
provisioned to use the correct primary) acquire zone files via zone =
transfer, and caching servers (once provisioned to use one or more =
roots) simply do a recursive lookup towards the local root and cache the =
result until its TTL expires. But one is gaining little of the benefit =
of a global name service, and the potential for ugly mistakes is =
relatively high with any faked-root DNS =
model.</div><div><br></div><div>--</div><div>Dean&nbsp;</div><div><br></di=
v><div><br></div></body></html>=

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Date: Tue, 9 Mar 2010 14:48:47 -0500
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From: Adam Uzelac <adam.uzelac@gmail.com>
To: Dean Willis <dean.willis@softarmor.com>
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Cc: dispatch@ietf.org, "Dwight, Timothy M \(Tim\)" <timothy.dwight@verizon.com>, Daryl Malas <D.Malas@cablelabs.com>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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The only thing I would add to Dean's comments below is:

*" ....which return responses tailored to the SSPX domain."* for the _SAME_
target destination in every "hop" through the myriad of networks.  It
doesn't really help at the end of the day. Each SSP domain will return
something different based on the source of the query, and again, it's all f=
o
the same target (phone number, URI, etc).

Holy complexity, Batman!


Adam Uzelac
Global Crossing



On Tue, Mar 9, 2010 at 1:24 PM, Dean Willis <dean.willis@softarmor.com>wrot=
e:

>
>  On Mar 3, 2010, at 10:08 PM, Dwight, Timothy M (Tim) wrote:
>
>   Daryl,
>
> I agree that defining a best practice often has the impact you suggest.
>
> IMHO the technique you describe has merit.  What=92s not clear is how a
> different response is generated per querying domain.  The draft shows
> response formats that would be appropriate for SSP1, but not appropriate =
for
> any other domain.  This implies that the response would be somehow tailor=
ed
> for each querying domain.  I think some elaboration on how that works (e.=
g.,
> what has to be provisioned, by whom, to make it happen) would be helpful.
>
>
>
>
> Good questions. I'm assuming (and we all know where that leads) that the
> following describes the solution as Daryl has proposed it:
>
> SSP1 runs its own private ENUM database, which it fully manages and
> provisions using whatever tools it has available. All nodes at SSP1 query
> SSP1's ENUM servers, which return responses tailored to the SSP1 domain.
>




>
>  SSP2 runs its own private ENUM database, which it fully manages and
> provisions using whatever tools it has available. All nodes at SSP2 query
> SSP2's ENUM servers, which return responses tailored to the SSP2 domain.
>
> ...
>
> SSPn runs its own private ENUM database, which it fully manages and
> provisions using whatever tools it has available. All nodes at SSPn query
> SSPn's ENUM servers, which return responses tailored to the SSPn domain.
>
>
> Within each SSP, each querying node must be provisioned (via some
> unspecified process) to use the SSP's private enum servers.
>
> Like private DNS, none of the private ENUM servers is queried by the
> "outside world".
>
> Unlike private DNS the private ENUM servers also do not query the outside
> world , because SIP routing lacks the concept of a "default route".
> Essentially, we're hacking a DNS infrastructure to provide the
> routing-knowlegde that in an IP network would be provided by RIP and/or
> DHCP. And that's what makes this much messier than private DNS, which is
> already its own world of headaches. Each external route must be manually
> provisioned into each private-enum root server.
>
> With this proposal, each private ENUM domain maintained by an SSP is a
> completely independent name-service island, equipped with its own root
> server. Some benefit of DNS caching and zone transfer may occur, as it is
> only necessary to manually provisioning the zone files into the master or
> "primary" private-domain root server at each SSP. Secondary roots (once
> provisioned to use the correct primary) acquire zone files via zone
> transfer, and caching servers (once provisioned to use one or more roots)
> simply do a recursive lookup towards the local root and cache the result
> until its TTL expires. But one is gaining little of the benefit of a glob=
al
> name service, and the potential for ugly mistakes is relatively high with
> any faked-root DNS model.
>
> --
> Dean
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>

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<div>The only thing I would add to Dean&#39;s comments below is:</div>
<div>=A0</div>
<div><em>&quot; ....which return responses tailored to the SSPX domain.&quo=
t;</em> for the _SAME_ target destination in every &quot;hop&quot; through =
the myriad of networks.=A0 It doesn&#39;t really help at the end of the day=
. Each SSP domain will return something different based on the source of th=
e query, and again, it&#39;s all fo the same target (phone number, URI, etc=
). </div>

<div>=A0</div>
<div>Holy complexity, Batman!</div>
<div>=A0</div>
<div>=A0</div>
<div>Adam Uzelac</div>
<div>Global Crossing</div>
<div><br><br>=A0</div>
<div class=3D"gmail_quote">On Tue, Mar 9, 2010 at 1:24 PM, Dean Willis <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:dean.willis@softarmor.com">dean.willis@=
softarmor.com</a>&gt;</span> wrote:<br>
<blockquote style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex=
; PADDING-LEFT: 1ex" class=3D"gmail_quote">
<div style=3D"WORD-WRAP: break-word">
<div class=3D"im"><br>
<div>
<div>On Mar 3, 2010, at 10:08 PM, Dwight, Timothy M (Tim) wrote:</div><br>
<blockquote type=3D"cite"><span style=3D"TEXT-TRANSFORM: none; TEXT-INDENT:=
 0px; BORDER-COLLAPSE: separate; FONT: medium Helvetica; WHITE-SPACE: norma=
l; LETTER-SPACING: normal; COLOR: rgb(0,0,0); WORD-SPACING: 0px">
<div lang=3D"EN-US" vlink=3D"purple" link=3D"blue">
<div>
<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>Daryl,</span></font></div>

<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>=A0</span></font></div>

<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>I agree that defining a best practice often has the impact you suggest.</s=
pan></font></div>

<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>=A0</span></font></div>

<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>IMHO the technique you describe has merit. =A0What=92s not clear is how a =
different response is generated per querying domain. =A0The draft shows res=
ponse formats that would be appropriate for SSP1, but not appropriate for a=
ny other domain.=A0 This implies that the response would be somehow tailore=
d for each querying domain. =A0I think some elaboration on how that works (=
e.g., what has to be provisioned, by whom, to make it happen) would be help=
ful.</span></font></div>

<div style=3D"MARGIN: 0in 0in 0pt; FONT-FAMILY: &#39;Times New Roman&#39;; =
FONT-SIZE: 12pt"><font color=3D"blue" size=3D"2" face=3D"Courier New"><span=
 style=3D"FONT-FAMILY: &#39;Courier New&#39;; COLOR: blue; FONT-SIZE: 10pt"=
>=A0</span></font></div>
</div></div></span></blockquote></div><br>
<div><br></div></div>
<div>Good questions. I&#39;m assuming (and we all know where that leads) th=
at the following describes the solution as Daryl has proposed it:</div>
<div><br></div>
<div>SSP1 runs its own private ENUM database, which it fully manages and pr=
ovisions using whatever tools it has available. All nodes at SSP1 query SSP=
1&#39;s ENUM servers, which return responses tailored to the SSP1 domain.</=
div>
</div></blockquote>
<div>=A0</div>
<div>=A0</div>
<div>=A0</div>
<blockquote style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex=
; PADDING-LEFT: 1ex" class=3D"gmail_quote">
<div style=3D"WORD-WRAP: break-word">
<div><br></div>
<div>
<div>SSP2 runs its own private ENUM database, which it fully manages and pr=
ovisions using whatever tools it has available. All nodes at SSP2 query SSP=
2&#39;s ENUM servers, which return responses tailored to the SSP2 domain.</=
div>

<div><br></div>
<div>...</div>
<div><br></div>
<div>SSPn runs its own private ENUM database, which it fully manages and pr=
ovisions using whatever tools it has available. All nodes at SSPn query SSP=
n&#39;s ENUM servers, which return responses tailored to the SSPn domain.</=
div>

<div><br></div>
<div><br></div>
<div>Within each SSP, each querying node must be provisioned (via some unsp=
ecified process) to use the SSP&#39;s private enum servers.=A0</div>
<div><br></div></div>
<div>Like private DNS, none of the private ENUM servers is queried by the &=
quot;outside world&quot;.=A0</div>
<div><br></div>
<div>Unlike private DNS the private ENUM servers also do not query the outs=
ide world , because SIP routing lacks the concept of a &quot;default route&=
quot;. Essentially, we&#39;re hacking a DNS infrastructure to provide the r=
outing-knowlegde that in an IP network would be provided by RIP and/or DHCP=
. And that&#39;s what makes this much messier than private DNS, which is al=
ready its own world of headaches. Each external route must be manually prov=
isioned into each private-enum root server.</div>

<div><br></div>
<div>With this proposal, each private ENUM domain maintained by an SSP is a=
 completely independent name-service island, equipped with its own root ser=
ver. Some benefit of DNS caching and zone transfer may occur, as it is only=
 necessary to manually provisioning the zone files into the master or &quot=
;primary&quot; private-domain root server at each SSP. Secondary roots (onc=
e provisioned to use the correct primary) acquire zone files via zone trans=
fer, and caching servers (once provisioned to use one or more roots) simply=
 do a recursive lookup towards the local root and cache the result until it=
s TTL expires. But one is gaining little of the benefit of a global name se=
rvice, and the potential for ugly mistakes is relatively high with any fake=
d-root DNS model.</div>

<div><br></div>
<div>--</div>
<div>Dean=A0</div><font color=3D"#888888">
<div><br></div>
<div><br></div></font></div><br>___________________________________________=
____<br>dispatch mailing list<br><a href=3D"mailto:dispatch@ietf.org">dispa=
tch@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/dispat=
ch" target=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br=
>
<br></blockquote></div><br>

--0016363b912050ebab048163781a--

From lendl@nic.at  Tue Mar  9 13:16:41 2010
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Date: Tue, 09 Mar 2010 22:16:26 +0100
From: Otmar Lendl <lendl@nic.at>
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On 09.03.2010 18:27, Dean Willis wrote:
> 
> I definitely see the need for defining a method by which to inform nodes
> inside an ITAD of egress routes from that ITAD to other domains. I think
> that most of the people who have responded on this thread agree that
> there is such a need.

Yes, there is.

> Where we have a disagreement is on mechanism. While your proposed use of
> ITAD-specific private ENUM  servers is not completely inconsistent with
> recent trends in ENUM, I have to say we've done a lot of things in ENUM
> that I don't like, and this is pushing the envelope further in that
> direction.

As I see the problem space, we're conflating two independent problems:

a) how is routing-information exchanged between SSPs?

Here we have one approach which favors registries which digest
routing-information and spit out suitable database dumps for provisioning
the SSPs boxes. The other approach is to get some real routing-protocol up
and running between SSPs (like BGP4 for IP layer 3 routing).

b) how is this routing-information conferred to the proxies/SBCs/... which
do the actual SIP message processing / forwarding?

Now, if these proxies are actually participating in a dynamic routing
protocol (or receive full dumps from a routing registry), then b) is
internal to the proxy. Just as there is no protocol for moving routes from
the BGP table of a router to the actual packet forwarding engine, this is
purely internal to the box.

If, on the other hand, some entity distinct from the actual SIP devices
handles the routing information (either by talking to a registry, or by
running a dynamic routing protocol), then the SIP device needs a protocol
to query that routing-oracle on what to do with a given call.

It seems to me that this is what ENUM is increasingly used for within SSPs.

IMHO this is the wrong protocol choice for this task.

Why?

* ENUM is a very simple DNS application. The only key to the lookup is the
destination number, the only return value is an URL. (more or less)

* If you look at a good number of recent drafts you notice that people feel
that this is not enough: they need "ID of the SIP device" or "caller-ID" in
the query, and a sh*tload of URI-parameters (trunk-id, ...) in the answer.
See draft-mayrhofer-drinks-sed-elements for further data that might need to
get stuffed into an ENUM response.

ENUM was never intended as call-control protocol where a proxy outsources
its complete routing intelligence to an external party. It's a simple
key->URI directory lookup. The LRF (RFC 3263) and the intelligence was
supposed to be within the SIP proxy.

Now, what we need here is something closer to RADIUS than to ENUM: a
protocol where the SIP device can pack up all the information it has about
the SIP INVITE into one request to the routing-oracle which then answers
with a details set of parameters (the SED) on what the stupid proxy is
supposed to do now.

Trying to expand ENUM into doing just that is bound to fail.

cheers,

otmar
-- 
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

From lconroy@insensate.co.uk  Tue Mar  9 14:30:44 2010
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To: Otmar Lendl <lendl@nic.at>
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Hi Otmar, Dean, folks,
 OK -- this has hit my "respond" threshold. Long post, but (summary @ end).

I recall thinking at the time that TRIP was not right and couldn't readily
be fixed, and was unsurprised that it just faded away. Thus I am decidedly
sceptical of attempts to resuscitate what is no longer even a ripe corpse.
It's a dead horse - stop flogging and get off.

There is no such thing as an ENUM server -- there are only DNS servers.
Thus the idea of providing different answers to the same DNS query
depending on who asked is at best limited (as in "broken").
Dispatch is sure not the place for it (nor, I would have thought, is DRINKS).
Providing a "who's asking?"-dependent response is going to require some
fancy footwork with DNS messages, so it *has* to be a DNSEXT thing.

ENUM might, just possibly, be used alone under certain limited scenarios:
IF THERE ARE RELATIVELY FEW OPTIONS (say, <= 10),
 each could be put into its own NAPTR with its own rococo parameters.
 The ENUM client would use those parameters as its "local knowledge"
 to reject the ones it couldn't/shouldn't use.
Personally, I think that this is pushing ENUM a loooooong way, but YMMV.

Problem is, *any* selective query/response protocol (RADIUS, LDAP, HTTP, ...)
that gives the entire answer to any question from dumb proxies is going
to be scale/performance limited. LNP makes every single query potentially
have a different answer from every other query, so this is a BIG database
and/or each answer has to be synthesised by the server in real time.
There ARE a lot of parameters to fit into each exchange, and lots of such
exchanges (for every call). This seems to be an Apps area issue -- good luck.

Some of the DNS based alternatives seem to be trying to split the problem
into sets of different data. IIUC, the NICC proposals were sending part of
the answer, and the client was expected to use its local knowledge to
interpret those answers. That "local knowledge" would be delivered by
other means -- not by DNS. This is NOT the same as a "who's asking?"
scheme.

Is that ENUM? -- not as specified in the 3761 and (for SIP) in 3764.
Is that ENUM-like? Could be.
Don't assume that a single ENUM query delivering the whole answer is
the only way to use DNS.
I simply do not believe that there needs to be an entirely distinct and
discrete answer given for each (queriant/target number). There ARE a
limited number of egress points from within any ITAD, and providing a
hint on the egress point to use is enough, most of the time. If not,
then looking up the current address of that egress point should be
HEAVILY cached by clients (and might be authenticated/secured/slower).

Summary -- it doesn't have to be one grand unified solution (A or B or ...).
Discounting DNS or ENUM because it doesn't give the whole answer may
be shortsighted. DNS may be *part* of the solution.

all the best,
  Lawrence


On 9 Mar 2010, at 21:16, Otmar Lendl wrote:
> On 09.03.2010 18:27, Dean Willis wrote:
>> 
>> I definitely see the need for defining a method by which to inform nodes
>> inside an ITAD of egress routes from that ITAD to other domains. I think
>> that most of the people who have responded on this thread agree that
>> there is such a need.
> 
> Yes, there is.
> 
>> Where we have a disagreement is on mechanism. While your proposed use of
>> ITAD-specific private ENUM  servers is not completely inconsistent with
>> recent trends in ENUM, I have to say we've done a lot of things in ENUM
>> that I don't like, and this is pushing the envelope further in that
>> direction.
> 
> As I see the problem space, we're conflating two independent problems:
> 
> a) how is routing-information exchanged between SSPs?
> 
> Here we have one approach which favors registries which digest
> routing-information and spit out suitable database dumps for provisioning
> the SSPs boxes. The other approach is to get some real routing-protocol up
> and running between SSPs (like BGP4 for IP layer 3 routing).
> 
> b) how is this routing-information conferred to the proxies/SBCs/... which
> do the actual SIP message processing / forwarding?
> 
> Now, if these proxies are actually participating in a dynamic routing
> protocol (or receive full dumps from a routing registry), then b) is
> internal to the proxy. Just as there is no protocol for moving routes from
> the BGP table of a router to the actual packet forwarding engine, this is
> purely internal to the box.
> 
> If, on the other hand, some entity distinct from the actual SIP devices
> handles the routing information (either by talking to a registry, or by
> running a dynamic routing protocol), then the SIP device needs a protocol
> to query that routing-oracle on what to do with a given call.
> 
> It seems to me that this is what ENUM is increasingly used for within SSPs.
> 
> IMHO this is the wrong protocol choice for this task.
> 
> Why?
> 
> * ENUM is a very simple DNS application. The only key to the lookup is the
> destination number, the only return value is an URL. (more or less)
> 
> * If you look at a good number of recent drafts you notice that people feel
> that this is not enough: they need "ID of the SIP device" or "caller-ID" in
> the query, and a sh*tload of URI-parameters (trunk-id, ...) in the answer.
> See draft-mayrhofer-drinks-sed-elements for further data that might need to
> get stuffed into an ENUM response.
> 
> ENUM was never intended as call-control protocol where a proxy outsources
> its complete routing intelligence to an external party. It's a simple
> key->URI directory lookup. The LRF (RFC 3263) and the intelligence was
> supposed to be within the SIP proxy.
> 
> Now, what we need here is something closer to RADIUS than to ENUM: a
> protocol where the SIP device can pack up all the information it has about
> the SIP INVITE into one request to the routing-oracle which then answers
> with a details set of parameters (the SED) on what the stupid proxy is
> supposed to do now.
> 
> Trying to expand ENUM into doing just that is bound to fail.
> 
> cheers,
> 
> otmar
> -- 
> // Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


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Subject: [dispatch] SIP Forum Luncheon at IETF 77 Monday 11:30 13:00
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The SIP Forum is going to hold a open meeting for the RAI community during
IETF 77. The purpose of this meeting is remind the SIP community of what the
SIP Forum is doing and discuss relevant technical issues that relate to IETF
WG's. The agenda for this luncheon will be to review current activities in
the SIP Forum and seek additional technical input from the RAI community on
mutual concerns.

 

As many of you know the SIP Forum is the principal sponsor of the SIPit
interoperability events and currently has 4 Task groups operating. 

 

BTW registration for SIPit 26 will be open soon.

 

Our principal technical initiatives are:

 

1.      SIP Connect 1.1 which is developing an enhanced  profile of IETF
Standards for PBX to SSP interoperability. As part of that work the IETF
MARTINI WG was formed to redefine the so called Implicit registration
procedure in SIP.

2.      FAX over SIP is investigating the interoperability of SIP in T.38
networks with a goal of possible recommendations to the relevant SDO's on
how fax can work better.

3.      Client User Agent Configuration which has a pending Recommendation
on how CUA devices could auto streamline the locating, retrieving and
maintaining of configuration-related information for SIP User Agents

4.      SIP in the SmartGrid. This is a Special Interest Group that is
evaluating the appropriateness of using SIP as a protocol for a number of
Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) deployments.

 

Information on the SIP Forum is available at 

 

http//www.sipforum.org

 

If you are interested in attending this event we have set up a Doodle Poll
to indicate your interest. Seating is somewhat limited ( 150 Maximum) lunch
will be provided.

 

Location .. Monday 11:30 13:00 the Laguna room and its located on the 4th
floor of the Anaheim Hilton

The link to the poll is: 

http://www.doodle.com/hde57nbpcdykzqfz 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype: rshockey101
LinkedIn : http://www.linkedin.com/in/rshockey101



 


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<div class=3DSection1>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The SIP Forum is going to hold a open meeting for =
the RAI
community during IETF 77. The purpose of this meeting is remind the SIP
community of what the SIP Forum is doing and discuss relevant technical =
issues
that relate to IETF WG&#8217;s. The agenda for this luncheon will be to =
review
current activities in the SIP Forum and seek additional technical input =
from
the RAI community on mutual concerns.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>As many of you know the SIP Forum is the principal =
sponsor
of the SIPit interoperability events and currently has 4 Task groups =
operating.
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>BTW registration for SIPit 26 will be open =
soon.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Our principal technical initiatives =
are:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP Connect 1.1 which is developing an enhanced
&nbsp;profile of IETF Standards for PBX to SSP interoperability. As part =
of
that work the IETF MARTINI WG was formed to redefine the so called =
Implicit
registration procedure in SIP.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>FAX over SIP is investigating the =
interoperability of
SIP in T.38 networks with a goal of possible recommendations to the =
relevant
SDO&#8217;s on how fax can work better.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Client User Agent Configuration which has a =
pending
Recommendation on how CUA devices could auto streamline the locating,
retrieving and maintaining of configuration-related information for SIP =
User
Agents<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP in the SmartGrid. This is a Special Interest =
Group
that is evaluating the appropriateness of using SIP as a protocol for a =
number
of Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) =
deployments.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Information on the SIP Forum is available at =
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Times =
New Roman","serif"'>http//www.sipforum.org</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>If you are interested in attending this event we =
have set up
a Doodle Poll to indicate your interest. Seating is somewhat limited ( =
150
Maximum) lunch will be provided.<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>Location .. </span><span
style=3D'font-size:10.0pt'>Monday 11:30 13:00 </span><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>the Laguna room and its =
located on
the 4th floor of the Anaheim Hilton</span><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>The link to the poll =
is:</span><span
style=3D'font-size:10.0pt'> <br>
<br>
<a href=3D"http://www.doodle.com/hde57nbpcdykzqfz" =
target=3D"_blank">http://www.doodle.com/hde57nbpcdykzqfz</a>
</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>
Shockey Consulting<br>
Chairman of the Board of Directors SIP Forum<br>
PSTN Mobile: +1 703.593.2683<br>
&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>
skype: rshockey101<br>
LinkedIn : <a =
href=3D"http://www.linkedin.com/in/rshockey101">http://www.linkedin.com/i=
n/rshockey101</a><br>
<br>
</span><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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From dean.willis@softarmor.com  Tue Mar  9 20:09:44 2010
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From: Dean Willis <dean.willis@softarmor.com>
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Cc: dispatch@ietf.org, "Dwight, Timothy M \(Tim\)" <timothy.dwight@verizon.com>, Daryl Malas <D.Malas@cablelabs.com>
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On Mar 9, 2010, at 1:48 PM, Adam Uzelac wrote:

> The only thing I would add to Dean's comments below is:
>
> " ....which return responses tailored to the SSPX domain." for the  
> _SAME_ target destination in every "hop" through the myriad of  
> networks.  It doesn't really help at the end of the day. Each SSP  
> domain will return something different based on the source of the  
> query, and again, it's all fo the same target (phone number, URI,  
> etc).
>

Even worse, the SSPX server will return a different value based not on  
the source of the query, but on the location of the SSPX server. We  
have an implicit assumption that the queries are coming from the same  
location, which is how the ENUM server manages to return locally- 
scoped answer. The server and all its knowledge are locally-scoped,  
and getting that knowledge in there is the guano in somebody else's  
batcave.

--
Dean


From dean.willis@softarmor.com  Tue Mar  9 21:56:43 2010
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On Mar 9, 2010, at 4:30 PM, Lawrence Conroy wrote:
>
> I recall thinking at the time that TRIP was not right and couldn't  
> readily
> be fixed, and was unsurprised that it just faded away. Thus I am  
> decidedly
> sceptical of attempts to resuscitate what is no longer even a ripe  
> corpse.
> It's a dead horse - stop flogging and get off.
>

TRIP was either a wong solution to a right problem, or a right  
solution to a wrong problem. Somewhere out there is something kindof  
like TRIP (and kindof NOT like TRIP) that actually solves the right  
problem.

First we have to decide what the "right problem" is, and analysis of  
the faults of TRIP is one way to work backwards to that problem  
definition.

> ...

> Summary -- it doesn't have to be one grand unified solution (A or B  
> or ...).
> Discounting DNS or ENUM because it doesn't give the whole answer may
> be shortsighted. DNS may be *part* of the solution.

I really do think that ENUM provides an excellent means for finding  
the "destination" that is ultimately responsible for a given phone  
number. What it does NOT do is find an optimal route for getting from  
the source to the destination, given all the session-border-elements  
that have to be visited in between. We do have a lot of history on  
"finding optimal routes", and approaches for doing that look more like  
TRIP than they look like ENUM. So, we need something that adds the  
missing pieces to ENUM, and that thing (or set of things) CANOOT be  
ENUM in and of themselves, because their operational model is by  
definition different from that of DNS.

Let's think about routing at the IP level. My Mac here has only one  
route, called a "default route" that gets it to my Linksys NAT-router.  
My Mac learns this route from DHCP (but if it were statically  
configured, the default route would also be statically configured, aka  
"provisioned").

The Linksys router also learns its default route via DHCP.

The next router out, however, appears to be participating in a dynamic  
routing table (I can see it sometimes makes different routing  
decisions). It's probably running OSPF within my ISP's network, and  
exchanges its static subscriber-facing routing information dynamically  
with other routers in the ISP-cloud, while learning from those routers  
about their own adjacencies.  Eventually my traffic hits the edge of  
my ISP's network, where the routers are peering with other ISPs,  
advertising and learning routes with BGP.

Notice we haven't said anything yet about DNS. That's because DNS is  
entirely separate from routing. I think I'm actually using Google's  
DNS servers at the moment, and I suspect Google really knows very  
little about the peering topology of my ISP. So when I type "http://www.ietf.org 
" into my browser, DNS translates the name into a destination address,  
and then a combination of default, interior, and exterior routing are  
used to get my request to the IETF's web server. We're not asking  
every router along the way to look up "www.ietf.org" in the DNS again.

But for call routing to work similarly, we need 1) an analog of the  
"default route", which is probably provided by conventional  
configuration and might be well described in draft-ietf-sip-outbound,  
2) a dynamic routing protocol inside the local ISP network that helps  
my traffic get routed to an egress router, and 3) a dynamic routing  
protocol that helps get the requested from my ISP's external-facing  
SBE (an "egress route" in Daryl's draft) to an "ingress" route (and  
maybe through more egress and ingress and SBEs as it moves along  
through and between peering domains. Note that each of those peering  
domains probably have their own interior routing topology with  
steering SBCs and the like.

Both 2 and 3 are "something vaguely like TRIP" that is NOT TRIP. But  
I'd really like to concentrate on what they are, without everybody  
popping up and saying "You can't have a dynamic routing protocol  
because TRIP failed to catch on, and therefore dynamic routing  
protocols are a Bad Idea and we must instead rely on static  
configuration for everything."

--
Dean




--Apple-Mail-25--16896031
Content-Type: text/html;
	charset=US-ASCII
Content-Transfer-Encoding: quoted-printable

<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 9, 2010, =
at 4:30 PM, Lawrence Conroy wrote:</div><blockquote =
type=3D"cite"><div><font class=3D"Apple-style-span" =
color=3D"#000000"><br></font>I recall thinking at the time that TRIP was =
not right and couldn't readily<br>be fixed, and was unsurprised that it =
just faded away. Thus I am decidedly<br>sceptical of attempts to =
resuscitate what is no longer even a ripe corpse.<br>It's a dead horse - =
stop flogging and get =
off.<br><br></div></blockquote><div><br></div><div>TRIP was either a =
wong solution to a right problem, or a right solution to a wrong =
problem. Somewhere out there is something kindof like TRIP (and kindof =
NOT like TRIP) that actually solves the right =
problem.</div><div><br></div><div>First we have to decide what the =
"right problem" is, and analysis of the faults of TRIP is one way to =
work backwards to that problem =
definition.</div><div><br></div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" =
color=3D"#000000">...<br></font></div></blockquote><br><blockquote =
type=3D"cite"><div>Summary -- it doesn't have to be one grand unified =
solution (A or B or ...).<br>Discounting DNS or ENUM because it doesn't =
give the whole answer may<br>be shortsighted. DNS may be *part* of the =
solution.<br></div></blockquote><br></div><div>I really do think that =
ENUM provides an excellent means for finding the "destination" that is =
ultimately responsible for a given phone number. What it does NOT do is =
find an optimal route for getting from the source to the destination, =
given all the session-border-elements that have to be visited in =
between. We do have a lot of history on "finding optimal routes", and =
approaches for doing that look more like TRIP than they look like ENUM. =
So, we need something that adds the missing pieces to ENUM, and that =
thing (or set of things) CANOOT be ENUM in and of themselves, because =
their operational model is by definition different from that of =
DNS.</div><div><br></div><div>Let's think about routing at the IP level. =
My Mac here has only one route, called a "default route" that gets it to =
my Linksys NAT-router. My Mac learns this route from DHCP (but if it =
were statically configured, the default route would also be statically =
configured, aka "provisioned").&nbsp;</div><div><br></div><div>The =
Linksys router also learns its default route via =
DHCP.&nbsp;</div><div><br></div><div>The next router out, however, =
appears to be participating in a dynamic routing table (I can see it =
sometimes makes different routing decisions). It's probably running OSPF =
within my ISP's network, and exchanges its static subscriber-facing =
routing information dynamically with other routers in the ISP-cloud, =
while learning from those routers about their own adjacencies. =
&nbsp;Eventually my traffic hits the edge of my ISP's network, where the =
routers are peering with other ISPs, advertising and learning routes =
with BGP.</div><div><br></div><div>Notice we haven't said anything yet =
about DNS. That's because DNS is entirely separate from routing. I think =
I'm actually using Google's DNS servers at the moment, and I suspect =
Google really knows very little about the peering topology of my ISP. So =
when I type "<a href=3D"http://www.ietf.org">http://www.ietf.org</a>" =
into my browser, DNS translates the name into a destination address, and =
then a combination of default, interior, and exterior routing are used =
to get my request to the IETF's web server. We're not asking every =
router along the way to look up "<a =
href=3D"http://www.ietf.org">www.ietf.org</a>" in the DNS =
again.</div><div><br></div><div>But for call routing to work similarly, =
we need 1) an analog of the "default route", which is probably provided =
by conventional configuration and might be well described in =
draft-ietf-sip-outbound, 2) a dynamic routing protocol inside the local =
ISP network that helps my traffic get routed to an egress router, and 3) =
a dynamic routing protocol that helps get the requested from my ISP's =
external-facing SBE (an "egress route" in Daryl's draft) to an "ingress" =
route (and maybe through more egress and ingress and SBEs as it moves =
along through and between peering domains. Note that each of those =
peering domains probably have their own interior routing topology with =
steering SBCs and the like.</div><div><br></div><div>Both 2 and 3 are =
"something vaguely like TRIP" that is NOT TRIP. But I'd really like to =
concentrate on what they are, without everybody popping up and saying =
"You can't have a dynamic routing protocol because TRIP failed to catch =
on, and therefore dynamic routing protocols are a Bad Idea and we must =
instead rely on static configuration for =
everything."</div><div><br></div><div>--</div><div>Dean</div><div><br></di=
v><div><br></div><br></body></html>=

--Apple-Mail-25--16896031--

From lendl@nic.at  Wed Mar 10 02:35:39 2010
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Lawrence,

On 09.03.2010 23:30, Lawrence Conroy wrote:
>
> Summary -- it doesn't have to be one grand unified solution (A or B or ...).
> Discounting DNS or ENUM because it doesn't give the whole answer may
> be shortsighted. DNS may be *part* of the solution.

As I see it, DNS/ENUM is perfect as the LUF protocol: mapping a number to
some sort of destination identifier (ITAD, Destination-Group,
Destination-Doamin, ...). This is largely independent of who's asking and
perfectly suited for DNS' delegation hierarchy and caching architecture.

The problem starts when you add the LRF: feeding that function could be
static configuration (simple leaf SSPs), simple fallback routing (leaf SSP
with two upstreams), or a full-blown routing protocol keying off the LUF's
result (ITAD, DG, ... see above).

As long as the LRF is colocated with the SIP proxy, then we just need that
routing protocol in addition to ENUM as LUF. If it isn't (the SIP proxy is
dumb and needs to ask some routing oracle what do to), then we have
basically two choices:

a) the SIP proxy could still do LUF itself, and then ask the routing oracle
how to process a call to a given destination group.

b) the SIP proxy doesn't even do the LUF itself and asks the routing oracle
what to do with a call to a given E.164 number.

--------

If one were to do a), then this would be open to various caching
optimizations. The draft that sparked this discussion tries to strong-arm
ENUM into doing b).

That is IMHO a bad idea.

otmar
-- 
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

From Milan.Patel@InterDigital.com  Wed Mar 10 05:39:49 2010
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From: "Patel, Milan" <Milan.Patel@InterDigital.com>
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Hi,

=20

My understanding is draft-patel-dispatch-cpc-oli-parameter-02 and the
Internet Draft for the DAI tel URI parameter are to be handled in a mini
working group. I am currently working with the authors of the DAI draft
to provide a charter. The latest version of the draft for CPC and OLI
tel URI parameters includes a syntax which as been approved by 3GPP and
within 3GPP CPC and OLI are a release 7 requirement.=20

=20

I am requesting for further comments on the draft. I am aware I need to
change my contact details on the draft.=20

=20

Best regards,

Milan


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Hi,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>My understanding is =
draft-patel-dispatch-cpc-oli-parameter-02
and the Internet Draft for the DAI tel URI parameter are to be handled =
in a
mini working group. I am currently working with the authors of the DAI =
draft to
provide a charter. The latest version of the draft for CPC and OLI tel =
URI
parameters includes a syntax which as been approved by 3GPP and within =
3GPP CPC
and OLI are a release 7 requirement. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>I am requesting for further comments on the =
draft. I
am aware I need to change my contact details on the draft. =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Best regards,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Milan<o:p></o:p></span></font></p>

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From pkyzivat@cisco.com  Wed Mar 10 06:59:24 2010
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I just got around to reading this draft.
I have all of the concerns that Jon does, plus some others:

Many of these properties are in fact properties of the *call*, not 
properties of the *caller*. That is especially true of the oli values:

- some are a function of the called number rather than the caller:
   * 30 Intercept (blank) - for calls to unassigned directory number (DN)
   * 31 Intercept (trouble) - for calls to directory numbers (DN) that
        have been manually placed in trouble-busy state by Telco
        personnel
   * 32 Intercept (regular) - for calls to recently changed or
        disconnected numbers

- the value can be assigned mid-call, as a result of call handling:
   * 34 Telco Operator Handled Call - after the Telco Operator Services
        System has handled a call for an IC, it may change the standard
        ANI digits to "34", before outpulsing the sequence to the IC,
        when the Telco performs all call handling functions, e.g.,
        billing. The code tells the IC that the BOC has performed billing
        on the call and the IC only has to complete the call.
   * 52 Outward Wide Area Telecommunications Service (OUTWATS) - this
        service allows customers to make calls to a certain zone(s) or
        band(s) on a direct dialed basis for a flat monthly charge or for
        a charge based on accumulated usage. OUTWATS lines can dial
        station-to-station calls directly to points within the selected
        band(s) or zone(s). The LEC performs a screening function to
        determine the correct charging and routing for OUTWATS calls
        based on the customer's class of service and the service area of
        the call party. When these calls are routed to the interexchange
        carrier via a combined WATS-POTS trunk group, it is necessary to
        identify the WATS calls with the ANI code "52".

- (this is not an exhaustive list.)

Putting the value in From is problematic if it is a function of 
something other than the caller. Its even more problematic if it can 
change mid-call, since changing it breaks Identity.

Even when the information is a property of the caller, it seems likely 
that it will often be added by something other than the UAC, since the 
UAC is unlikely to be trusted to provide it.

If this is information that is a property of the call rather than 
caller, and it can change along the signaling path, then IMO it belongs 
in some other header, in a field that may be changed along the path.

Also a NIT:

    The "oli" parameter with value 29 indicates that the device that the
    call is initiated from is located within a prison.  An "oli" with
    value "prison" is equally valid.

The ABNF doesn't permit a value of "prison" for the oli parameter - only 
digits.


Jon Peterson wrote:
> 
> In evaluating proposals to add capabilities to SIP via a tel URI, the 
> first question we ask is, "Would this be useful in SIP requests that do 
> NOT use telephone numbers?" If the answer is "yes," then probably the 
> capability should be added to SIP in a way that works with and without 
> the tel URI. (The draft is very slightly unclear  about whether or not 
> you can include this parameter in SIP URIs that do not contain embedded 
> tel URIs but do contain non-tel-URI representations of telephone numbers 
> - but I'm speaking here to the overall intention to couple the use of 
> OLI/CPC information to telephone numbers exclusively.)
> 
> While the draft focuses on the use case where SIP is gatewaying to ISUP, 
> and thus where telephone numbers are likely to be used, I would be 
> hesitant to rule out uses that do not involve telephone numbers - 
> especially when the motivating use case given is REGISTER in an 
> emergency context. Encapsulating the opaque numeric values of OLI into a 
> SIP parameter also more or less restricts the use of this to entities 
> that understand ISUP. This begs the obvious question: should there be a 
> way in SIP (outside of gatewaying)  to express these same concepts? 
> Since you already translate the CPC values into human-readable 
> equivalents ("operator", "payphone", "test"), is there some reason why 
> OLI can't receive the same treatment? The only example of the semantics 
> of OLI that you give ("oli=29" meaning "prison") indeed seems it could 
> admit of that translation. If there is some good reason why CPC should 
> be translated and OLI should be encapsulated, the draft should 
> explicitly motivate it.
> 
> By the by, since the Acks mention that I submitted a version of this 
> proposal prior to Rohan's, just as an historical aside I in fact split 
> that out from the original draft-ietf-sip-privacy-04 by Flemming and 
> Bill Marshall, which for some reason had it tacked into an appendix. The 
> parameter proposed in that document (which can still be found via 
> Google), the Nature of Party ("np") parameter, maps to a set of 
> human-readable literals like "ordinary", "police",  "payphone", 
> "emergency", and so on. The document then proposes a mapping to the ANI 
> II digits; one could just as easily be provided for OLI or CPC in the 
> ISUP variants of interest. The example given in that appendix shows this 
> being used in a URI that doesn't contain a telephone number. In any 
> event, if you want to have a pedigree of the idea in your Acks, you 
> shouldn't leave out that document.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> Milan Patel wrote:
>> Hi,
>>
>> A new version of the Internet Draft for the Calling Party's Category and
>> Originating Line Identity URI Parameters is now available. 
>> http://www.ietf.org/id/draft-patel-dispatch-cpc-oli-parameter-00.txt
>>
>> This is based on Rohan Mahy's draft-mahy-iptel-cpc-06
>>
>> Best regards,
>> Milan
>>
>>
>> -----Original Message-----
>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] Sent: 08 
>> October 2009 18:40
>> To: Patel, Milan (MOP:EP10)
>> Cc: r.jesske@telekom.de; mdolly@att.com
>> Subject: New Version Notification for
>> draft-patel-dispatch-cpc-oli-parameter-00
>>
>> A new version of I-D, draft-patel-dispatch-cpc-oli-parameter-00.txt has
>> been successfuly submitted by Milan Patel and posted to the IETF
>> repository.
>>
>> Filename:     draft-patel-dispatch-cpc-oli-parameter
>> Revision:     00
>> Title:         Uniform Resource Identifier (URI) Parameters for
>> indicating the Calling Party's Catagory and Originating Line Identity
>> Creation_date:     2009-10-08
>> WG ID:         Independent Submission
>> Number_of_pages: 10
>>
>> Abstract:
>> This document defines two new URI parameters to describe the calling
>> party's category and toll class of service originating line
>> information which are parameters also used in SS7 ISUP and other
>> telephony signalling protocols.  The intended use of these URI
>> parameters is for the "tel" URI or equivalent SIP URI representation.
>>  
>>
>>
>>
>> The IETF Secretariat.
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>   
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From emcho@sip-communicator.org  Wed Mar 10 07:08:39 2010
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Hey all,

A couple of months ago we had a discussion here about how SIP-XMPP
clients would discover and authenticate against its SIP and XMPP parts.

Enrico and I have recently posted a draft [1] detailing the problem and
summarizing two ways of addressing it. We'd be happy to be able to
discuss this in Anaheim and comments are of course welcome at any time.

Cheers,
Emil

[1] - http://tools.ietf.org/html/draft-ivov-sipxmpp-auth-01


-- 
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31

From lendl@nic.at  Wed Mar 10 07:44:03 2010
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Date: Wed, 10 Mar 2010 16:44:04 +0100
From: Otmar Lendl <lendl@nic.at>
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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Penn,

On 09.03.2010 00:26, PFAUTZ, PENN L (ATTCORP) wrote:
> Just a word of caution re number portability. In some countries access
> to this data is restricted to legally qualified carriers.  Will ITADs
> always correspond to such entities? Looking at the IANA list so far
> that's not my impression. You may find that the entities that do have
> access to NP databases may not be the ones interested in a TRIP-like
> mechanism and vice versa.

This is indeed a relevant point.

>From my perspective, this is, while true, mostly a red herring erected by
carriers who try to shield the actual market numbers from the public.

For example, in Austria NP data used to be private and carriers (plus their
lobbyists) worked hard to keep it that way. Then one day we finally
implemented mobile NP in Austria. As calling towards mobiles is
comparatively expensive and the tariff varies by destination network, the
regulator mandated that callers have to be notified via a voice
announcement if a number has been ported. (The price for a call might have
changed from 0 cents/min (on-net for some deals) to 30 cents/min due to a
porting operation. So this was a legitimate consumer protection issue.)

In other words, we went from a "keep this info private" mindset to a "annoy
end-users with this information on each call" within a year.

So: if there is a legitimate reason why the general public needs this
information (e.g. to do correct least-cost routing on multi-homed PBXs)
then this can change quickly.

otmar
-- 
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

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Otmar:
I hear you (although in the US we didn't do calling-party-pays and
carriers generally don't reflect different termination rates in their
retail toll prices so we won't have *that* problem.)

But I'd really like to raise what I think is a larger issue - just what
are we trying to accomplish?

All the LRF and TRIP stuff seems to be tending toward a kind of
link-by-link call routing model that to me seems redolent of the kind of
trunk group by trunk group routing in the PSTN. I thought that that was
what we were trying to get away from in moving to IP!=20
The original promise of ENUM was that it would identify the target and
basic Internet connectivity would then get you there.
As you've pointed out elsewhere, that original vision didn't really work
for most carriers, but carriers have tried to replicate parts of it with
approaches like the GSMA IPX to provide a secure IP fabric for
connecting. That allows connectivity management via selective
advertising in BGP so that the SIP INVITES based on same ENUM query
response is appropriately routed from different originating networks.
The same techniques support direct bilateral interconnection which is
how much of the traffic will flow anyway.

So, when would you want something TRIP like? To me the remaining case
seems to be when you're searching for the best intermediary when you're
neither directly connected nor linked to some kind of IPX like fabric
that can get to the target. I expect intermediaries to compete and scale
to matter. When I need an intermediary I'll generally prefer that they
handle the whole job. Some pricing tables will let me choose and I'm not
sure I need/want those coming across the NNI as reachability
advertisements.=20

Add to this the downward pressure on rates that is only going accelerate
and I wonder how much need there really is. I may be biased from my
vantage point though and perhaps someone can provide the compelling use
case.


Penn Pfautz
AT&T Access Management
+1-732-420-4962
-----Original Message-----
From: Otmar Lendl [mailto:lendl@nic.at]=20
Sent: Wednesday, March 10, 2010 10:44 AM
To: PFAUTZ, PENN L (ATTCORP)
Cc: mohamed.boucadair@orange-ftgroup.com; Dean Willis; dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event
alternative?

Penn,

On 09.03.2010 00:26, PFAUTZ, PENN L (ATTCORP) wrote:
> Just a word of caution re number portability. In some countries access
> to this data is restricted to legally qualified carriers.  Will ITADs
> always correspond to such entities? Looking at the IANA list so far
> that's not my impression. You may find that the entities that do have
> access to NP databases may not be the ones interested in a TRIP-like
> mechanism and vice versa.

This is indeed a relevant point.

>From my perspective, this is, while true, mostly a red herring erected
by
carriers who try to shield the actual market numbers from the public.

For example, in Austria NP data used to be private and carriers (plus
their
lobbyists) worked hard to keep it that way. Then one day we finally
implemented mobile NP in Austria. As calling towards mobiles is
comparatively expensive and the tariff varies by destination network,
the
regulator mandated that callers have to be notified via a voice
announcement if a number has been ported. (The price for a call might
have
changed from 0 cents/min (on-net for some deals) to 30 cents/min due to
a
porting operation. So this was a legitimate consumer protection issue.)

In other words, we went from a "keep this info private" mindset to a
"annoy
end-users with this information on each call" within a year.

So: if there is a legitimate reason why the general public needs this
information (e.g. to do correct least-cost routing on multi-homed PBXs)
then this can change quickly.

otmar
--=20
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Date: Wed, 10 Mar 2010 09:24:00 -0700
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Dean,

One of the challenges is at what layer the information is routed.  Using yo=
ur example below, after the FQDN is converted into an IP address, your Mac =
now routes the payload entirely through "layer 1-3" elements to reach the d=
estination server.  With SIP, it should be considered, after the FQDN for t=
he domain is converted into an IP address it is routed through "layer 1-3" =
elements to a "layer 5" element, which is not the destination.  Our existin=
g routing protocols (e.g. RIP (just wanted to throw this one in), OSPF, BGP=
, etc.) focus on layer 3.  The address space and the ability to summarize t=
his space is well understood and defined in the industry.  I think the reas=
on we are struggling with varying solutions, whether it is TRIP or ENUM or =
whatever, is because we are working in an undefined address space, leveragi=
ng multiple layers.

Even if I am able to dynamically exchange domain names with other peers, so=
 I can do my ENUM look-up, resolve to a URI, and then do a DNS look-up on t=
he URI domain; I am still faced with layer 3 potentially routing me non-opt=
imally to my next layer 5 hop.  The reason for this is, simply, my layer 3 =
routers are not layer 5 aware.  I either statically define layer 3 or layer=
 5, but these are the only two options I see at this point.

The other thing to consider is layer 5 providers (SSPs), in my perception, =
are far more paranoid than layer 3 providers (SPs).  I believe this is the =
perception of the redundant, redundant, diversified, diversified super dupe=
r extra stable telephony infrastructure that was built over a 100 years.  I=
n another 60 years, SPs will be the same as IP is leveraged more and more f=
or "time sensitive" applications.

Regards,

Daryl

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Dean Willis
Sent: Tuesday, March 09, 2010 10:57 PM
To: Lawrence Conroy
Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 9, 2010, at 4:30 PM, Lawrence Conroy wrote:

I recall thinking at the time that TRIP was not right and couldn't readily
be fixed, and was unsurprised that it just faded away. Thus I am decidedly
sceptical of attempts to resuscitate what is no longer even a ripe corpse.
It's a dead horse - stop flogging and get off.


TRIP was either a wong solution to a right problem, or a right solution to =
a wrong problem. Somewhere out there is something kindof like TRIP (and kin=
dof NOT like TRIP) that actually solves the right problem.

First we have to decide what the "right problem" is, and analysis of the fa=
ults of TRIP is one way to work backwards to that problem definition.

...

Summary -- it doesn't have to be one grand unified solution (A or B or ...)=
.
Discounting DNS or ENUM because it doesn't give the whole answer may
be shortsighted. DNS may be *part* of the solution.

I really do think that ENUM provides an excellent means for finding the "de=
stination" that is ultimately responsible for a given phone number. What it=
 does NOT do is find an optimal route for getting from the source to the de=
stination, given all the session-border-elements that have to be visited in=
 between. We do have a lot of history on "finding optimal routes", and appr=
oaches for doing that look more like TRIP than they look like ENUM. So, we =
need something that adds the missing pieces to ENUM, and that thing (or set=
 of things) CANOOT be ENUM in and of themselves, because their operational =
model is by definition different from that of DNS.

Let's think about routing at the IP level. My Mac here has only one route, =
called a "default route" that gets it to my Linksys NAT-router. My Mac lear=
ns this route from DHCP (but if it were statically configured, the default =
route would also be statically configured, aka "provisioned").

The Linksys router also learns its default route via DHCP.

The next router out, however, appears to be participating in a dynamic rout=
ing table (I can see it sometimes makes different routing decisions). It's =
probably running OSPF within my ISP's network, and exchanges its static sub=
scriber-facing routing information dynamically with other routers in the IS=
P-cloud, while learning from those routers about their own adjacencies.  Ev=
entually my traffic hits the edge of my ISP's network, where the routers ar=
e peering with other ISPs, advertising and learning routes with BGP.

Notice we haven't said anything yet about DNS. That's because DNS is entire=
ly separate from routing. I think I'm actually using Google's DNS servers a=
t the moment, and I suspect Google really knows very little about the peeri=
ng topology of my ISP. So when I type "http://www.ietf.org" into my browser=
, DNS translates the name into a destination address, and then a combinatio=
n of default, interior, and exterior routing are used to get my request to =
the IETF's web server. We're not asking every router along the way to look =
up "www.ietf.org<http://www.ietf.org>" in the DNS again.

But for call routing to work similarly, we need 1) an analog of the "defaul=
t route", which is probably provided by conventional configuration and migh=
t be well described in draft-ietf-sip-outbound, 2) a dynamic routing protoc=
ol inside the local ISP network that helps my traffic get routed to an egre=
ss router, and 3) a dynamic routing protocol that helps get the requested f=
rom my ISP's external-facing SBE (an "egress route" in Daryl's draft) to an=
 "ingress" route (and maybe through more egress and ingress and SBEs as it =
moves along through and between peering domains. Note that each of those pe=
ering domains probably have their own interior routing topology with steeri=
ng SBCs and the like.

Both 2 and 3 are "something vaguely like TRIP" that is NOT TRIP. But I'd re=
ally like to concentrate on what they are, without everybody popping up and=
 saying "You can't have a dynamic routing protocol because TRIP failed to c=
atch on, and therefore dynamic routing protocols are a Bad Idea and we must=
 instead rely on static configuration for everything."

--
Dean




--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F671Dsrvxchg_
Content-Type: text/html; charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18876"></HEAD>
<BODY=20
style=3D"WORD-WRAP: break-word; -webkit-nbsp-mode: space; -webkit-line-brea=
k: after-white-space">
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>Dean,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>One of the challenges is at what layer the infor=
mation=20
is routed.&nbsp; Using your example below, after the FQDN is converted into=
 an=20
IP address, your Mac now routes the payload entirely through "layer 1-3"=20
elements to reach the destination server.&nbsp; With SIP, it should be=20
considered, after the FQDN for the domain is converted into an IP address i=
t is=20
routed through "layer 1-3" elements to a "layer 5" element, which is not th=
e=20
destination.&nbsp; Our existing routing protocols (e.g. RIP (just wanted to=
=20
throw this one in), OSPF, BGP, etc.) focus on layer 3.&nbsp; The&nbsp;addre=
ss=20
space&nbsp;and the ability to summarize this&nbsp;space is well understood =
and=20
defined in the industry.&nbsp; I think the reason we are struggling with va=
rying=20
solutions, whether it is TRIP or ENUM or whatever, is because we are workin=
g in=20
an undefined address space, leveraging multiple layers.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>Even if I am able to dynamically exchange domain=
 names=20
with other peers, so I can do my ENUM look-up, resolve to a URI, and then d=
o a=20
DNS look-up on the URI domain; I am still faced with layer 3 potentially ro=
uting=20
me non-optimally to my next layer 5 hop.&nbsp; The reason for this is, simp=
ly,=20
my layer 3 routers are not layer 5 aware.&nbsp; I either statically define =
layer=20
3 or layer 5, but these are the only two options I see at this=20
point.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>The other thing to consider is layer 5 providers=
=20
(SSPs), in my perception, are far more paranoid than layer 3 providers=20
(SPs).&nbsp; I believe this is the perception of the redundant, redundant,=
=20
diversified, diversified super duper extra stable telephony infrastructure =
that=20
was built over a 100 years.&nbsp; In another 60 years, SPs will be the same=
 as=20
IP is leveraged more and more for "time sensitive"=20
applications.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT color=3D#0000ff size=3D2 face=3DArial><SP=
AN=20
class=3D726180916-10032010>Daryl</SPAN></FONT></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Dean=20
Willis<BR><B>Sent:</B> Tuesday, March 09, 2010 10:57 PM<BR><B>To:</B> Lawre=
nce=20
Conroy<BR><B>Cc:</B> dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] I-=
D=20
Action: draft-malas-dispatch-sip-egress-route-00<BR></FONT><BR></DIV>
<DIV></DIV><BR>
<DIV>
<DIV>On Mar 9, 2010, at 4:30 PM, Lawrence Conroy wrote:</DIV>
<BLOCKQUOTE type=3D"cite">
  <DIV><FONT class=3DApple-style-span color=3D#000000><BR></FONT>I recall t=
hinking=20
  at the time that TRIP was not right and couldn't readily<BR>be fixed, and=
 was=20
  unsurprised that it just faded away. Thus I am decidedly<BR>sceptical of=
=20
  attempts to resuscitate what is no longer even a ripe corpse.<BR>It's a d=
ead=20
  horse - stop flogging and get off.<BR><BR></DIV></BLOCKQUOTE>
<DIV><BR></DIV>
<DIV>TRIP was either a wong solution to a right problem, or a right solutio=
n to=20
a wrong problem. Somewhere out there is something kindof like TRIP (and kin=
dof=20
NOT like TRIP) that actually solves the right problem.</DIV>
<DIV><BR></DIV>
<DIV>First we have to decide what the "right problem" is, and analysis of t=
he=20
faults of TRIP is one way to work backwards to that problem definition.</DI=
V>
<DIV><BR></DIV>
<BLOCKQUOTE type=3D"cite">
  <DIV><FONT class=3DApple-style-span=20
color=3D#000000>...<BR></FONT></DIV></BLOCKQUOTE><BR>
<BLOCKQUOTE type=3D"cite">
  <DIV>Summary -- it doesn't have to be one grand unified solution (A or B =
or=20
  ...).<BR>Discounting DNS or ENUM because it doesn't give the whole answer=
=20
  may<BR>be shortsighted. DNS may be *part* of the=20
solution.<BR></DIV></BLOCKQUOTE><BR></DIV>
<DIV>I really do think that ENUM provides an excellent means for finding th=
e=20
"destination" that is ultimately responsible for a given phone number. What=
 it=20
does NOT do is find an optimal route for getting from the source to the=20
destination, given all the session-border-elements that have to be visited =
in=20
between. We do have a lot of history on "finding optimal routes", and appro=
aches=20
for doing that look more like TRIP than they look like ENUM. So, we need=20
something that adds the missing pieces to ENUM, and that thing (or set of=20
things) CANOOT be ENUM in and of themselves, because their operational mode=
l is=20
by definition different from that of DNS.</DIV>
<DIV><BR></DIV>
<DIV>Let's think about routing at the IP level. My Mac here has only one ro=
ute,=20
called a "default route" that gets it to my Linksys NAT-router. My Mac lear=
ns=20
this route from DHCP (but if it were statically configured, the default rou=
te=20
would also be statically configured, aka "provisioned").&nbsp;</DIV>
<DIV><BR></DIV>
<DIV>The Linksys router also learns its default route via DHCP.&nbsp;</DIV>
<DIV><BR></DIV>
<DIV>The next router out, however, appears to be participating in a dynamic=
=20
routing table (I can see it sometimes makes different routing decisions). I=
t's=20
probably running OSPF within my ISP's network, and exchanges its static=20
subscriber-facing routing information dynamically with other routers in the=
=20
ISP-cloud, while learning from those routers about their own adjacencies.=20
&nbsp;Eventually my traffic hits the edge of my ISP's network, where the ro=
uters=20
are peering with other ISPs, advertising and learning routes with BGP.</DIV=
>
<DIV><BR></DIV>
<DIV>Notice we haven't said anything yet about DNS. That's because DNS is=20
entirely separate from routing. I think I'm actually using Google's DNS ser=
vers=20
at the moment, and I suspect Google really knows very little about the peer=
ing=20
topology of my ISP. So when I type "<A=20
href=3D"http://www.ietf.org">http://www.ietf.org</A>" into my browser, DNS=
=20
translates the name into a destination address, and then a combination of=20
default, interior, and exterior routing are used to get my request to the I=
ETF's=20
web server. We're not asking every router along the way to look up "<A=20
href=3D"http://www.ietf.org">www.ietf.org</A>" in the DNS again.</DIV>
<DIV><BR></DIV>
<DIV>But for call routing to work similarly, we need 1) an analog of the=20
"default route", which is probably provided by conventional configuration a=
nd=20
might be well described in draft-ietf-sip-outbound, 2) a dynamic routing=20
protocol inside the local ISP network that helps my traffic get routed to a=
n=20
egress router, and 3) a dynamic routing protocol that helps get the request=
ed=20
from my ISP's external-facing SBE (an "egress route" in Daryl's draft) to a=
n=20
"ingress" route (and maybe through more egress and ingress and SBEs as it m=
oves=20
along through and between peering domains. Note that each of those peering=
=20
domains probably have their own interior routing topology with steering SBC=
s and=20
the like.</DIV>
<DIV><BR></DIV>
<DIV>Both 2 and 3 are "something vaguely like TRIP" that is NOT TRIP. But I=
'd=20
really like to concentrate on what they are, without everybody popping up a=
nd=20
saying "You can't have a dynamic routing protocol because TRIP failed to ca=
tch=20
on, and therefore dynamic routing protocols are a Bad Idea and we must inst=
ead=20
rely on static configuration for everything."</DIV>
<DIV><BR></DIV>
<DIV>--</DIV>
<DIV>Dean</DIV>
<DIV><BR></DIV>
<DIV><BR></DIV><BR></BODY></HTML>

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From kcartwright@tnsi.com  Wed Mar 10 09:11:01 2010
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From: "Cartwright, Kenneth" <kcartwright@tnsi.com>
To: "PFAUTZ, PENN L (ATTCORP)" <pp3129@att.com>, "mohamed.boucadair@orange-ftgroup.com" <mohamed.boucadair@orange-ftgroup.com>, Dean Willis <dean.willis@softarmor.com>
Date: Wed, 10 Mar 2010 12:10:29 -0500
Thread-Topic: [dispatch] Why are we TRIPped out? Is there an sip-event alternative?
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References: <404BBC8D-04BD-4E1F-87F2-C8707EB1AE98@softarmor.com><114DAD31379DFA438C0A2E39B3B8AF5D01213F66D0@srvxchg><37B4C540-0C8C-4A3D-9493-80B9416E8815@softarmor.com><22008_1267774700_4B90B4EC_22008_38058_1_94C682931C08B048B7A8645303FDC9F30EFBC79EAA@PUEXCB1B.nanterre.francetelecom.fr><E80F9509-5B14-46CF-BB3B-89D4F2EAF3E2@softarmor.com> <22008_1268058768_4B950A90_22008_213649_1_94C682931C08B048B7A8645303FDC9F30EFC0180AB@PUEXCB1B.nanterre.francetelecom.fr> <35FE871E2B085542A35726420E29DA6B037B0E21@gaalpa1msgusr7a.ugd.att.com>
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Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event	alternative?
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...taking this discussion onto a related tangent....

These regulatory and contractual limitations on access to TN->Carrier assig=
nments are one of the reasons that existing systems that currently provide =
the envisioned LUF feature (query a TN to find out the responsible carrier =
currently assigned to that TN, more or less) are *required* to allow the LU=
F responses to vary based on the querier.  Additional requirements that nec=
essitate this are:

(1)     This TN->CarrierAssignment data set is often/usually "layered", in =
the sense that the a country code's Based Number Plan TN->CarrierAssignment=
s comprise the foundation layer, that is then overlaid by (but not co-mingl=
ed with) number portability TN->CarrierAssignmens, that is then overlaid by=
 (but not comingled with) TN->CarrierAssignments for TNs that are leased ac=
ross carriers, that is then overlaid by TN->CarrierAssignments for TNs that=
 override any of the proceeding for any number of business or operational r=
easons (although this last overlay is less structured and less relevant to =
the overall point).
(2)     Regulatory, contractual, and business reasons currently determine, =
for each country code, the layers and subsets of layers of TN->CarrierAssig=
nments that any given querier is allowed to "see".
(3)     Some queriers simply do not want to see certain layers of TN->Carri=
erAssignemnts for given country codes.
(4)     Because many country codes do not have a regulatory or any well def=
ined source for TN->CarrierAssignments, the data is sourced from multiple p=
arties.  So even within a given country code, only portions of a given laye=
r may be allowed to be seen by some queriers.

So one way of looking at the problem space of the LUF is that the above req=
uirements are not what we want to solve and standardize.  What we want to s=
olve and standardize is the much simpler and pristine end goal of ... all T=
N->CarrierAssignment information is queriable by any organization and all t=
hose organizations have legal access to all of it.  We solve and standardiz=
e this with the view of "build it and they will come".

The other way of looking at the problem is that we solve the requirements s=
tated above.

Ken

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of PFAUTZ, PENN L (ATTCORP)
Sent: Monday, March 08, 2010 6:27 PM
To: mohamed.boucadair@orange-ftgroup.com; Dean Willis
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alter=
native?

Just a word of caution re number portability. In some countries access to t=
his data is restricted to legally qualified carriers.  Will ITADs always co=
rrespond to such entities? Looking at the IANA list so far that's not my im=
pression. You may find that the entities that do have access to NP database=
s may not be the ones interested in a TRIP-like mechanism and vice versa.

Penn Pfautz
AT&T Access Management
+1-732-420-4962
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of mohamed.boucadair@orange-ftgroup.com
Sent: Monday, March 08, 2010 9:33 AM
To: Dean Willis
Cc: dispatch@ietf.org
Subject: Re: [dispatch] Why are we TRIPped out? Is there an sip-event alter=
native?


Dear Dean, all,

Please see inline.
Cheers,
Med

________________________________
De : Dean Willis [mailto:dean.willis@softarmor.com]
Envoy=E9 : vendredi 5 mars 2010 19:16
=C0 : BOUCADAIR Mohamed NCPI/NAD/TIP
Cc : Daryl Malas; dispatch@ietf.org
Objet : Re: [dispatch] Why are we TRIPped out? Is there an sip-event altern=
ative?

On Mar 5, 2010, at 1:38 AM, <mohamed.boucadair@orange-ftgroup.com<mailto:mo=
hamed.boucadair@orange-ftgroup.com>> <mohamed.boucadair@orange-ftgroup.com<=
mailto:mohamed.boucadair@orange-ftgroup.com>> wrote:


Dear Dean, all,

In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL: http://www.eurescom.eu/message/messageMay2009/Interc=
onnection_challenges_of_IP_telephony_Eurescom_study_P1853.asp).
Med: http://www.eurescom.eu/message/messageMay2009/Interconnection_challeng=
es_of_IP_telephony_Eurescom_study_P1853.asp<http://www.eurescom.eu/message/=
messageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_study_P1=
853.asp)>
I'm having a hard time resolving this link, as it appears to be 404. Perhap=
s there's a typo?

With respect to:


21. Extended TRIP protocol is suggested to be used as base for IP Telephony=
 routing;

We've heard here that TRIP, as currently modeled, doesn't work due to the L=
NP problem. What sorts of extensions id you have in mind? Were they intende=
d to address this problem?
[Med] An alternative would be to use TRIP in conjunction with a the current=
 Number Portability infrastructure as used in PSTN infrastructure for placi=
ng inter-ITAD calls (this means that the owner domain will always advertise=
 its aggregate in TRIP, but a NPDB (Network Portability Database) should qu=
estioned first).

Examples of extensions to TRIP (or whatever dynamic telephony routing) woul=
d be:
- Avoid AS (Autonomous System) spiral: the media traffic can cross the same=
 AS several times
- Ability to maintain multiple inter-ITAD paths
- Ability to enforce some advanced TE such as: Avoid that my VoIP traffic t=
o cross a given AS domain (TRIP only maintain an ITAD list)
- Ability to avoid congestionned links
- etc.

BTW, TRIP is only considered as starting point, this does not preclude to i=
nvestigate any new solution to meet the recommendations we identified. BTW,=
 In the recommendations list we have identified the need to encourage aggre=
gation-based models.


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</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"WORD-WRAP: bre=
ak-word;
webkit-nbsp-mode: space;webkit-line-break: after-white-space">
<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">&#8230;taking this discussion onto a r=
elated tangent&#8230;.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">These regulatory and contractual limit=
ations on access to TN-&gt;Carrier assignments are one of the reasons that =
existing systems that currently
 provide the envisioned LUF feature (query a TN to find out the responsible=
 carrier currently assigned to that TN, more or less) are *<b><span style=
=3D"font-weight:bold">required</span></b>* to allow the LUF responses to va=
ry based on the querier.&nbsp; Additional
 requirements that necessitate this are:<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal" style=3D"margin-left:.5in;text-indent:-.25in;mso-lis=
t:l0 level1 lfo1">
<![if !supportLists]><font size=3D"2" color=3D"navy" face=3D"Arial"><span s=
tyle=3D"font-size:10.0pt;font-family:Arial;
color:navy"><span style=3D"mso-list:Ignore">(1)<font size=3D"1" face=3D"Tim=
es New Roman"><span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;
</span></font></span></span></font><![endif]><font size=3D"2" color=3D"navy=
" face=3D"Arial"><span style=3D"font-size:10.0pt;font-family:Arial;
color:navy">This TN-&gt;CarrierAssignment data set is often/usually &#8220;=
layered&#8221;, in the sense that the a country code&#8217;s Based
 Number Plan TN-&gt;CarrierAssignments comprise the foundation layer, that =
is then overlaid by (but not co-mingled with) number portability TN-&gt;Car=
rierAssignmens, that is then overlaid by (but not comingled with) TN-&gt;Ca=
rrierAssignments for TNs that are leased
 across carriers, that is then overlaid by TN-&gt;CarrierAssignments for TN=
s that override any of the proceeding for any number of business or operati=
onal reasons (although this last overlay is less structured and less releva=
nt to the overall point).<o:p></o:p></span></font></p>
<p class=3D"MsoNormal" style=3D"margin-left:.5in;text-indent:-.25in;mso-lis=
t:l0 level1 lfo1">
<![if !supportLists]><font size=3D"2" color=3D"navy" face=3D"Arial"><span s=
tyle=3D"font-size:10.0pt;font-family:Arial;
color:navy"><span style=3D"mso-list:Ignore">(2)<font size=3D"1" face=3D"Tim=
es New Roman"><span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;
</span></font></span></span></font><![endif]><font size=3D"2" color=3D"navy=
" face=3D"Arial"><span style=3D"font-size:10.0pt;font-family:Arial;
color:navy">Regulatory, contractual, and business reasons currently determi=
ne, for each country code, the layers and subsets
 of layers of TN-&gt;CarrierAssignments that any given querier is allowed t=
o &#8220;see&#8221;.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal" style=3D"margin-left:.5in;text-indent:-.25in;mso-lis=
t:l0 level1 lfo1">
<![if !supportLists]><font size=3D"2" color=3D"navy" face=3D"Arial"><span s=
tyle=3D"font-size:10.0pt;font-family:Arial;
color:navy"><span style=3D"mso-list:Ignore">(3)<font size=3D"1" face=3D"Tim=
es New Roman"><span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;
</span></font></span></span></font><![endif]><font size=3D"2" color=3D"navy=
" face=3D"Arial"><span style=3D"font-size:10.0pt;font-family:Arial;
color:navy">Some queriers simply do not want to see certain layers of TN-&g=
t;CarrierAssignemnts for given country codes.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal" style=3D"margin-left:.5in;text-indent:-.25in;mso-lis=
t:l0 level1 lfo1">
<![if !supportLists]><font size=3D"2" color=3D"navy" face=3D"Arial"><span s=
tyle=3D"font-size:10.0pt;font-family:Arial;
color:navy"><span style=3D"mso-list:Ignore">(4)<font size=3D"1" face=3D"Tim=
es New Roman"><span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;=
&nbsp;&nbsp;&nbsp;
</span></font></span></span></font><![endif]><font size=3D"2" color=3D"navy=
" face=3D"Arial"><span style=3D"font-size:10.0pt;font-family:Arial;
color:navy">Because many country codes do not have a regulatory or any well=
 defined source for TN-&gt;CarrierAssignments, the
 data is sourced from multiple parties.&nbsp; So even within a given countr=
y code, only portions of a given layer may be allowed to be seen by some qu=
eriers.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">So one way of looking at the problem s=
pace of the LUF is that the above requirements are not what we want to solv=
e and standardize.&nbsp;
 What we want to solve and standardize is the much simpler and pristine end=
 goal of &#8230; all TN-&gt;CarrierAssignment information is queriable by a=
ny organization and all those organizations have legal access to all of it.=
&nbsp; We solve and standardize this with the
 view of &#8220;build it and they will come&#8221;.<o:p></o:p></span></font=
></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">The other way of looking at the proble=
m is that we solve the requirements stated above.<o:p></o:p></span></font><=
/p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Ken<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span style=3D"font-size:12.0pt">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> disp=
atch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
<b><span style=3D"font-weight:bold">On Behalf Of </span></b>PFAUTZ, PENN L =
(ATTCORP)<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Monday, March 08, 2010=
 6:27 PM<br>
<b><span style=3D"font-weight:bold">To:</span></b> mohamed.boucadair@orange=
-ftgroup.com; Dean Willis<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [dispatch] Why =
are we TRIPped out? Is there an sip-event alternative?</span></font><o:p></=
o:p></p>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">Just a w=
ord of caution re number portability. In some countries access to this data=
 is restricted to legally qualified carriers.&nbsp;
 Will ITADs always correspond to such entities? Looking at the IANA list so=
 far that&#8217;s not my impression. You may find that the entities that do=
 have access to NP databases may not be the ones interested in a TRIP-like =
mechanism and vice versa.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Arial"><s=
pan style=3D"font-size:10.0pt;font-family:Arial;color:#1F497D">Penn Pfautz<=
/span></font><font color=3D"#1f497d"><span style=3D"color:#1F497D"><o:p></o=
:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Arial"><s=
pan style=3D"font-size:10.0pt;font-family:Arial;color:#1F497D">AT&amp;T Acc=
ess Management</span></font><font color=3D"#1f497d"><span style=3D"color:#1=
F497D"><o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Arial"><s=
pan style=3D"font-size:10.0pt;font-family:Arial;color:#1F497D">&#43;1-732-4=
20-4962</span></font><font size=3D"2" color=3D"#1f497d" face=3D"Calibri"><s=
pan style=3D"font-size:11.0pt;font-family:
Calibri;color:#1F497D"><o:p></o:p></span></font></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> disp=
atch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]
<b><span style=3D"font-weight:bold">On Behalf Of </span></b>mohamed.boucada=
ir@orange-ftgroup.com<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Monday, March 08, 2010=
 9:33 AM<br>
<b><span style=3D"font-weight:bold">To:</span></b> Dean Willis<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> dispatch@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [dispatch] Why =
are we TRIPped out? Is there an sip-event alternative?<o:p></o:p></span></f=
ont></p>
</div>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">&nbsp;<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Dear Dean, all,</span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">&nbsp;<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Please see inline.</span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Cheers,</span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Med</span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span lang=3D"FR" style=3D"font-size:1=
2.0pt">
<hr size=3D"2" width=3D"100%" align=3D"center">
</span></font></div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><b><font size=3D"2" f=
ace=3D"Tahoma"><span lang=3D"FR" style=3D"font-size:10.0pt;font-family:Taho=
ma;font-weight:bold">De&nbsp;:</span></font></b><font size=3D"2" face=3D"Ta=
homa"><span lang=3D"FR" style=3D"font-size:10.0pt;font-family:Tahoma">
 Dean Willis [mailto:dean.willis@softarmor.com] <br>
<b><span style=3D"font-weight:bold">Envoy=E9&nbsp;:</span></b> vendredi 5 m=
ars 2010 19:16<br>
<b><span style=3D"font-weight:bold">=C0&nbsp;:</span></b> BOUCADAIR Mohamed=
 NCPI/NAD/TIP<br>
<b><span style=3D"font-weight:bold">Cc&nbsp;:</span></b> Daryl Malas; dispa=
tch@ietf.org<br>
<b><span style=3D"font-weight:bold">Objet&nbsp;:</span></b> Re: [dispatch] =
Why are we TRIPped out? Is there an sip-event alternative?</span></font><sp=
an lang=3D"FR"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">On Mar 5, 2010, at 1:38 AM, &lt;<a href=3D"mailto:mohamed.boucadair=
@orange-ftgroup.com">mohamed.boucadair@orange-ftgroup.com</a>&gt; &lt;<a hr=
ef=3D"mailto:mohamed.boucadair@orange-ftgroup.com">mohamed.boucadair@orange=
-ftgroup.com</a>&gt;
 wrote:<o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><font size=3D"3" face=
=3D"Times New Roman"><span style=3D"font-size:12.0pt"><o:p>&nbsp;</o:p></sp=
an></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><br>
Dear Dean, all,<br>
<br>
In the last year, we have undertaken a study about IP telephony interconnec=
t at large scale (URL:
<a href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_ch=
allenges_of_IP_telephony_Eurescom_study_P1853.asp)">
http://www.eurescom.eu/message/messageMay2009/Interconnection_challenges_of=
_IP_telephony_Eurescom_study_P1853.asp)</a>.<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Med:
<a href=3D"http://www.eurescom.eu/message/messageMay2009/Interconnection_ch=
allenges_of_IP_telephony_Eurescom_study_P1853.asp)">
<font size=3D"3" face=3D"Times New Roman"><span style=3D"font-size:12.0pt;f=
ont-family:&quot;Times New Roman&quot;">http://www.eurescom.eu/message/mess=
ageMay2009/Interconnection_challenges_of_IP_telephony_Eurescom_study_P1853.=
asp</span></font></a>&nbsp;&nbsp;</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">I'm having a hard time resolving this link, as it appears to be 404=
. Perhaps there's a typo?<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">With respect to:<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">21.<span class=3D"apple-tab-span">
</span>Extended TRIP protocol is suggested to be used as base for IP Teleph=
ony routing;<o:p></o:p></span></font></p>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt"><o:p>&nbsp;</o:p></span></font></p>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">We've heard here that TRIP, as currently modeled, doesn't work due =
to the LNP problem. What sorts of extensions id you have in mind? Were they=
 intended to address this
 problem?<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">[Med]&nbsp;An alternative would be to use TRIP&nbsp;in conjunction wi=
th a the current&nbsp;Number Portability infrastructure&nbsp;as used in&nbs=
p;PSTN
 infrastructure&nbsp;for&nbsp;placing&nbsp;inter-ITAD calls (this means tha=
t the owner domain will always advertise its aggregate in TRIP, but&nbsp;a =
NPDB (Network Portability Database) should questioned first).&nbsp;</span><=
/font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">Examples of extensions to TRIP (or whatever dynamic telephony routing=
) would be:</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">- Avoid AS (Autonomous System) spiral: the media traffic can cross th=
e same AS several times</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">- Ability to maintain multiple inter-ITAD paths</span></font><o:p></o=
:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">- Ability to enforce some advanced TE such as: Avoid that my VoIP tra=
ffic to cross a given&nbsp;AS domain (TRIP only maintain
 an ITAD list)</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">- Ability to avoid congestionned links</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">- etc.</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">&nbsp;<o:p></o:p></span></font></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"blue" face=3D"Courier New"=
><span style=3D"font-size:10.0pt;font-family:&quot;Courier New&quot;;color:=
blue">BTW, TRIP is only considered as starting point, this does not preclud=
e to investigate any new solution to meet the recommendations
 we identified. BTW, In the recommendations list we have identified the nee=
d to encourage aggregation-based models.</span></font><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:
12.0pt">&nbsp;<o:p></o:p></span></font></p>
</div>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>*********************************<o:p></o:p></span></font></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>This message and any attachments (the &quot;message&quot;) are confidentia=
l and intended solely for the addressees. <o:p></o:p></span></font></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>Any unauthorised use or dissemination is prohibited.<o:p></o:p></span></fo=
nt></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>Messages are susceptible to alteration. <o:p></o:p></span></font></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>France Telecom Group shall not be liable for the message if altered, chang=
ed or falsified.<o:p></o:p></span></font></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>If you are not the intended addressee of this message, please cancel it im=
mediately and inform the sender.<o:p></o:p></span></font></pre>
<pre><font size=3D"2" face=3D"Courier New"><span style=3D"font-size:10.0pt"=
>********************************<o:p></o:p></span></font></pre>
</div>
<br>
<hr>
<font face=3D"Arial" color=3D"Gray" size=3D"1">This e-mail message is for t=
he sole use of the intended recipient(s)and may<br>
contain confidential and privileged information of Transaction Network Serv=
ices.<br>
Any unauthorised review, use, disclosure or distribution is prohibited. If =
you<br>
are not the intended recipient, please contact the sender by reply e-mail a=
nd destroy all copies of the original message.<br>
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</font>
</body>
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--_000_754963199212404AB8E9CFCA6C3D0CDA0D8925D533TNSMAILNAwin2_--

From D.Malas@cablelabs.com  Wed Mar 10 15:16:07 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
Date: Wed, 10 Mar 2010 16:16:09 -0700
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Dean,

Based on your assessment below, let me try and summarize:

Two IETF efforts:
1) Egress Route Method for SIP (Intra-domain)
2) Dynamic Routing Method for SIP (Inter-domain)

I attempted to provide my response to your questions below.

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Tuesday, March 09, 2010 10:28 AM
To: Daryl Malas
Cc: Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 9, 2010, at 10:04 AM, Daryl Malas wrote:

> Dean et. al,
>
> I am thankful my draft spurred on this discussion on potentially=20
> resolving a larger issue.  With this being said, the purpose of the=20
> draft is to be a potential component of the larger solution.  I got=20
> lost on whether or not the general community understands and sees the=20
> overall value of defining an egress method, for ensuring a consistent=20
> approach between SSPs and vendors.

I definitely see the need for defining a method by which to inform nodes in=
side an ITAD of egress routes from that ITAD to other domains. =20
I think that most of the people who have responded on this thread agree tha=
t there is such a need.

We also seem to see a need for dissemination of inter-ITAD routing informat=
ion, although we're still discussing the relationships between phone number=
s, ITADs, and routes.

Where we have a disagreement is on mechanism. While your proposed use of IT=
AD-specific private ENUM  servers is not completely inconsistent with recen=
t trends in ENUM, I have to say we've done a lot of things in ENUM that I d=
on't like, and this is pushing the envelope further in that direction. Ther=
e are a lot of people running private DNS spaces, too, but we haven't stand=
ardized the discovery of default routes within an Intranet using private DN=
S for very good reasons, and those same reasons apply to ENUM used in the s=
ame pattern. Yes, it can be made to work, but it creates massive operationa=
l headaches.


We have several arguments in favor of some sort of "dynamic routing protoco=
l" solution or solutions. Open questions include:

1) Are different interior and exterior routing protocols (BGP vs OSPF, for =
instance) required, or can one common protocol meet the full set f requirem=
ents?
[DM] Good question.  I think we need to have one that works for either befo=
re we can determine whether it is internal for external only.

2) Is  a standalone protocol such as TRIP more appropriate than something "=
inside" SIP (or DNS, XMPP, whatever)? If so, does this apply to both interi=
or and exterior roles?
[DM] After some "hallway collaboration," I do not think TRIP will work, at =
all, in it's current state.

3) Can TRIP be made to work? This has related questions about aggregation m=
odels, LNP access, LNP fragmentation of aggregation, etc.
[DM] Maybe; but whether you use domains or E.164 addresses, the underlying =
layer 3-to-layer 5 issues still exists.

4) Can inter-ITAD routing be accomplished based on number-to-ITAD transform=
ations (which ENUM gives us) accompanied by a dynamic inter- ITAD route exc=
hange protocol? This may also have to factor in the LNP problem.
[DM] I think this may be a more likely potential, but still has a lot of is=
sues in forming the routing table.  A lot of dissegrated user information w=
ill cause a very large routing table.  Using domains for the routing table =
may be an option.  I am not sure what value this provides over existing ENU=
M solutions, though.  Creating another solution for the same problem does n=
ot always make for a more efficient solution.  It may just create another s=
olution, which will cause industry fragmentation in implementation preferen=
ces.

5) Can we do something combining TRIP (maybe with modifications) and ENUM t=
hat solves the problems? If so, will anybody deploy it, or do they prefer s=
triking themselves repeatedly in the face with blunt objects over extended =
periods of time?
[DM] I worked through this scenario on a whiteboard for 30 minutes or so, a=
nd I struggled to see any resolution to the problems.  It just makes it mor=
e complicated in a different way.

All-in-all, this is excellent material for the sort of problem- exploration=
 process that might lead to a working group.


--
Dean


From dean.willis@softarmor.com  Wed Mar 10 15:37:54 2010
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Subject: Re: [dispatch] is this TRIP necessary?
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On Mar 10, 2010, at 10:23 AM, PFAUTZ, PENN L (ATTCORP) wrote:

>
> All the LRF and TRIP stuff seems to be tending toward a kind of
> link-by-link call routing model that to me seems redolent of the  
> kind of
> trunk group by trunk group routing in the PSTN. I thought that that  
> was
> what we were trying to get away from in moving to IP!

Excellent point. We're moving towards application-specific routing  
infrastructure and an "intelligent network." It rather turns my  
stomach. But building tools for this model that actually work well  
turns my stomach less than hacking up DNS to commit the same atrocities.

On the other hand, "trunking" seems to be what people are actually  
willing to pay for. A democracy will get the government it deserves,  
and a market economy will get the products it buys.

--
Dean

From dean.willis@softarmor.com  Wed Mar 10 15:46:12 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On Mar 10, 2010, at 10:24 AM, Daryl Malas wrote:

> Dean,
>
> One of the challenges is at what layer the information is routed.   
> Using your example below, after the FQDN is converted into an IP  
> address, your Mac now routes the payload entirely through "layer  
> 1-3" elements to reach the destination server.  With SIP, it should  
> be considered, after the FQDN for the domain is converted into an IP  
> address it is routed through "layer 1-3" elements to a "layer 5"  
> element, which is not the destination.  Our existing routing  
> protocols (e.g. RIP (just wanted to throw this one in), OSPF, BGP,  
> etc.) focus on layer 3.  The address space and the ability to  
> summarize this space is well understood and defined in the  
> industry.  I think the reason we are struggling with varying  
> solutions, whether it is TRIP or ENUM or whatever, is because we are  
> working in an undefined address space, leveraging multiple layers.

I agree. Further, what I think that we're doing is trying to build an  
applicational-level routing later that is structurally similar to the  
IP routing layer, but operates on this other address space. It's a  
classic example of an overlay network.

"Names" are the next-level addresses. HIP and other locator-identifier  
separators are structurally very similar to what we're talking about  
doing with "domains" and "ITAD identifiers".

>
> Even if I am able to dynamically exchange domain names with other  
> peers, so I can do my ENUM look-up, resolve to a URI, and then do a  
> DNS look-up on the URI domain; I am still faced with layer 3  
> potentially routing me non-optimally to my next layer 5 hop.  The  
> reason for this is, simply, my layer 3 routers are not layer 5  
> aware.  I either statically define layer 3 or layer 5, but these are  
> the only two options I see at this point.

RIght. What I think we've been fumbling toward is a dynamic Layer 5  
routing fabric that itself is Layer 3 aware, and therefore handles

>
> The other thing to consider is layer 5 providers (SSPs), in my  
> perception, are far more paranoid than layer 3 providers (SPs).  I  
> believe this is the perception of the redundant, redundant,  
> diversified, diversified super duper extra stable telephony  
> infrastructure that was built over a 100 years.  In another 60  
> years, SPs will be the same as IP is leveraged more and more for  
> "time sensitive" applications.


We could have done what we needed to with RSVP, but the Net wasn't  
ready for resolving conflicts between real-time flows yet. In one  
possible future, everything on the Net that is real-time constrained  
has something like RSVP assuring its timeliness. In another, the "open  
Internet" is gone, replaced by lots of different application-level  
routing fabrics running over dedicated layer 3 that is over  
provisioned to handle it. I don't see a happy middle ground.

--
Dean

From dean.willis@softarmor.com  Wed Mar 10 15:48:16 2010
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On Mar 10, 2010, at 5:16 PM, Daryl Malas wrote:

> Dean,
>
> Based on your assessment below, let me try and summarize:
>
> Two IETF efforts:
> 1) Egress Route Method for SIP (Intra-domain)
> 2) Dynamic Routing Method for SIP (Inter-domain)

It's possible both could be done by one protocol, possibly with two  
modes of operation.

--
Dean

From stpeter@stpeter.im  Thu Mar 11 09:54:13 2010
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On 3/10/10 8:08 AM, Emil Ivov wrote:
> Hey all,
>=20
> A couple of months ago we had a discussion here about how SIP-XMPP
> clients would discover and authenticate against its SIP and XMPP parts.=

>=20
> Enrico and I have recently posted a draft [1] detailing the problem and=

> summarizing two ways of addressing it. We'd be happy to be able to
> discuss this in Anaheim and comments are of course welcome at any time.=


Regarding section 2, I have a few questions:

1. What kind of URI are you talking about?

2. Is this a URI that the SIP+XMPP service provides to the client? If
so, how?

3. How exactly can a URI and a password be used for authentication in
XMPP (i.e., using which SASL mechanism)?

4. Does the client derive a username from the URI so that it can use an
authentication mechanism that experts a username and password?

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi All

I am not able to upload the updated version (-04) of the I-D due to some
issue with the I-D submission tool - (Says I can submit I-D only on
2010-03-22).
Until then I am attaching a copy of the updated version (txt and html
formats).

Please review and comment.

Thanks

Note: Let me know if there is any other way to upload the I-D document
and will be happy to do it.
=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, March 02, 2010 7:19 PM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

I think this all sounds good now.

	Thanks,
	Paul

Avasarala Ranjit-A20990 wrote:
> Inline
>=20
> <Ranjit-Mar2>
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 02, 2010 2:16 AM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
> inline...
>=20
> Avasarala Ranjit-A20990 wrote:
>=20
>> I'm still a little confused here.
>> If phone is on and subscription is active, then in general=20
>> notification should be possible, regardless of registration. I=20
>> realize
>=20
>> IMS doesn't permit that, but from a general standard perspective we=20
>> shouldn't assume it. The manifestation in this case is that the=20
>> NOTIFY
>=20
>> would fail and the subscription would eventually be torn down.
>>
>> So I think the cases are:
>> - there is a subscription and notification works
>> - there is a subscription but notification fails
>> - there is no subscription.
>>
>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as=20
>> Notification fails, we can say - not sending notification due to (1)=20
>> no match of filter criteria (2) phone not registered or outside=20
>> coverage area due to which there is no connectivity and hence=20
>> notification cannot be sent Anyway if no subscription, no
> notification.
>=20
> Are you suggesting that you might have a subscription, and have the=20
> filter criteria met, and yet not attempt to send a NOTIFY because the=20
> phone is not registered or has no network connectivity???
>=20
> That seems wrong. ISTM that either you would remove the subscription,=20
> or else you would try to send the NOTIFY and then discover that it
fails.
>=20
> <Ranjit-Mar2> yes the subscription could be removed when the device is

> not reachable.
>=20
> OTOH, this probably ought not to be within scope of what is discussed=20
> in this draft.
>=20
> <Ranjit-Mar2> yes true not within the scope of this draft.
>=20
>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>> contacts: sip:line1@alice.office.atlanta.com and=20
>>> sip:line2@alice.home.atlanta.com and
>>>     both the phones ring, then it would be a case of parallel
> forking.
>>> Now say one or both the phones have set diversion rules, to divert=20
>>> the
>> In this case, what does it mean for "one or both phones have set=20
>> diversion rules"? Does that somehow imply distinct rules for each
> phone?
>> I don't see how that could work. Aren't the diversion rules for the=20
>> *AOR*?
>>
>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there=20
>> is
>=20
>> a diversion rule set for the AoR and when diversions occur, the=20
>> notifications are Sent to the AoR.
>=20
> Not sent to the AoR - sent to the remote contact of the dialog for=20
> each subscription to the AoR. Same point as below that you agreed
with.
>=20
> <Ranjit-Mar2> Yes agree with u.=20
>=20
>> I see they could be set/changed from either phone. But its a single=20
>> set of rules isn't it? So I might set the rules from the office, and=20
>> then cancel them from home.
>>
>> <Ranjit-Feb26-2> yes single set and could be set from either of the=20
>> phones. True.
>>
>>> call to say=20
>>>     sip:voicemail@alice.atlanta.com, then the notifications would=20
>>> get
>=20
>>> generated with respect to both the contacts.
>> Again I'm confused by the terminology you use, and how it relates=20
>> operationally. The diversion happens on the AOR, and the=20
>> subscriptions
>=20
>> are on the AOR, and the notifications are sent to the subscribers. I=20
>> don't see where the registered contacts enter into the picture at
all.
>>
>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>
>> If you have something else in mind, I hope you can explain it better=20
>> in the draft.
>>
>>> [4] Yes the value of N is configurable and depends on how many=20
>>> diversions the subscribe wants to keep or could be dependent on=20
>>> server policy (some maximum
>>>     number). If the server policy is to send notifications as and=20
>>> when they occur and not keep any divsersions information, then the=20
>>> value of last N
>>>     diversions will be zero. But if the server chooses to retain the

>>> last diversion always, then the value of N will be 1 and so on.
>> All is straightforward for N>=3D1.
>>
>> For N=3D0 things are messy. If we are modeling this as state, and=20
>> notifications are about state changes, then the new diversion is=20
>> added
>=20
>> to the (previously empty) state and that triggers notifications to=20
>> all
>=20
>> current subscriptions. Then if you don't want to retain even this=20
>> much
>=20
>> state (because N=3D0) you immediately remove that version from the
> state.
>> But that is again a state change, so you end up with another=20
>> notification of that state change.
>>
>> <Ranjit-Feb26-2> True. Actually the notifications are for informing=20
>> the subscriber of the call diversions that happen. The notifications=20
>> are not for informing the subscriber of state changes in the notifier

>> or the number of diversions that occurred. As I mentioned in the=20
>> abstract as well as 24.604, the main intent of CDIVN service is to=20
>> inform the subscribers of communication diversions that occur on=20
>> their
>=20
>> behalf in the network and this information will help them manage=20
>> their
>=20
>> rules better.
>=20
> I am not certain, but think we may still be talking past one another=20
> to some extent.
>=20
> The state can be considered just a formalism for defining the service=20
> clearly, regardless of whether it has intrinsic value to the CDIV=20
> service. But it *is* a formalism. So if it is introduced, then its=20
> behavior can't be ignored when inconvenient. Defining the state with=20
> N=3D0 leads to some undesired behavior. The simplest way around that =
is=20
> to insist that N be >=3D1.
>=20
> <Ranjit-Mar2> Ok in that case we can say that value of N >=3D1 and the =

> notifier has always maintains history of last diversion at a minimu.=20
> It can maintain more than 1 too depending on the policy.
>=20
> If its important to you that no state be retained once a notification=20
> has been sent about the most recent diversion, then you will have to=20
> find some other way to specify the service, that still meets the=20
> expectations of 3265. I don't know if there is anything there that=20
> Adam would consider valid. We will have to get his opinion on that.
>=20
> <Ranjit-Mar2> In that case I am OK maintaining the hostory of=20
> diversions that occurred as part of state information and retaining=20
> that state even after the notification for that diversion is sent - so

> that in case I do not get a confirmation, I would be able to re-send
the notification.
> Also maintaining state of diversions would help when the notifications

> need to sent only at a particular time (if fikter criteria mentions
it).
>=20
>=20
> 	Thanks,
> 	Paul
>=20
>> <Ranjit-Feb26-2> But if we want to include the information related to

>> how many diversions have happened till time (say from 12 AM ) till=20
>> the
>=20
>> time the notification is being sent, then as you say we can store the

>> diversions. This would be useful when the subscriber specifies in the

>> filter criteria to send notifications only during certain time like=20
>> between 5 and 6 PM and not whenever a diversion occurs. Then N is=20
>> grater than 1.
>=20
>> I presume you don't want that behavior. If you can live with N>=3D1=20
>> then
>=20
>> we don't have the problem at all. If you need to support N=3D0 and=20
>> don't
>=20
>> want the extra notify, then we have to figure out if there is some=20
>> trick to how things are modeled to get the desired effect. I've=20
>> suggested something previously, but Adam didn't like it.
>>
>> This needs sorting out.
>>
>> 	Thanks,
>> 	Paul
>=20

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From emcho@sip-communicator.org  Thu Mar 11 12:09:02 2010
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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Hey Peter,

On 11 mars 2010, at 18:33, Peter Saint-Andre <stpeter@stpeter.im> wrote:

> On 3/10/10 8:08 AM, Emil Ivov wrote:
>
> Regarding section 2, I have a few questions:
>
> 1. What kind of URI are you talking about?

The the stpeter@stpeter.im kind.

> 2. Is this a URI that the SIP+XMPP service provides to the client?

Not the service itself but rather the provider.

> If so, how?

Out of scope (but basically via a web registration or any other means  
that people are using for mail and IM registration)

> 3. How exactly can a URI and a password be used for authentication  
> in XMPP (i.e., using which SASL mechanism)?

Below.

> 4. Does the client derive a username from the URI so that it can use  
> an
> authentication mechanism that experts a username and password?

Yes, exactly. The way most clients would do for SIP and XMPP already.  
This is just about making sure that they would be able to bring the  
assumption over to a SIP+XMPP service.

Cheers,
Emil
>
> Peter
>
> -- 
> Peter Saint-Andre
> https://stpeter.im/
>
>
>

From stpeter@stpeter.im  Thu Mar 11 12:32:25 2010
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Date: Thu, 11 Mar 2010 13:20:17 -0700
From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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--------------ms000205070607040801030104
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On 3/11/10 12:59 PM, Emil Ivov wrote:
> Hey Peter,
>=20
> On 11 mars 2010, at 18:33, Peter Saint-Andre <stpeter@stpeter.im> wrote=
:
>=20
>> On 3/10/10 8:08 AM, Emil Ivov wrote:
>>
>> Regarding section 2, I have a few questions:
>>
>> 1. What kind of URI are you talking about?
>=20
> The the stpeter@stpeter.im kind.

Since it lacks a scheme, stpeter@stpeter.im is not a URI.

>> 2. Is this a URI that the SIP+XMPP service provides to the client?
>=20
> Not the service itself but rather the provider.
>=20
>> If so, how?
>=20
> Out of scope (but basically via a web registration or any other means
> that people are using for mail and IM registration)
>=20
>> 3. How exactly can a URI and a password be used for authentication in
>> XMPP (i.e., using which SASL mechanism)?
>=20
> Below.
>=20
>> 4. Does the client derive a username from the URI so that it can use a=
n
>> authentication mechanism that experts a username and password?
>=20
> Yes, exactly. The way most clients would do for SIP and XMPP already.
> This is just about making sure that they would be able to bring the
> assumption over to a SIP+XMPP service.

That is unclear to me. Do I visit a website, receive an email with a
URI, click that to authorize, and receive credentials for my combined
SIP+XMPP account?

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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--------------ms000205070607040801030104--

From pkyzivat@cisco.com  Thu Mar 11 13:05:38 2010
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Date: Thu, 11 Mar 2010 15:56:15 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Peter Saint-Andre <stpeter@stpeter.im>
References: <4B97B5F0.3030905@sip-communicator.org>	<4B99295D.9090008@stpeter.im>	<598FE5D9-781D-44A4-B465-C843FB231E9E@sip-communicator.org> <4B995081.4090201@stpeter.im>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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Peter Saint-Andre wrote:

>>> 1. What kind of URI are you talking about?
>> The the stpeter@stpeter.im kind.
> 
> Since it lacks a scheme, stpeter@stpeter.im is not a URI.

That was unclear to me too.

I was *guessing* that the intent is that

	pkyzivat@ietf.org

would imply support for a variety of url schemes that can have that as 
their body:

	sip:pkyzivat@ietf.org
	xmpp:pkyzivat@ietf.org
	mailto:pkyzivat@ietf.org

whatever protocols the domain cares to support. IOW its just a 
convention for linking things based on similar name. And then using the 
same, or related, credentials for all.

	Thanks,
	Paul

From emcho@sip-communicator.org  Thu Mar 11 14:07:44 2010
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From: Emil Ivov <emcho@sip-communicator.org>
Date: Thu, 11 Mar 2010 23:07:24 +0100
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To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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On Thu, Mar 11, 2010 at 9:20 PM, Peter Saint-Andre <stpeter@stpeter.im> wrote:
>>> 1. What kind of URI are you talking about?
>>
>> The the stpeter@stpeter.im kind.
>
> Since it lacks a scheme, stpeter@stpeter.im is not a URI.

True. I guess ID would have been a better choice. Is this a problem
for any other reason than the misuse of the term?

>>> 4. Does the client derive a username from the URI so that it can use an
>>> authentication mechanism that experts a username and password?
>>
>> Yes, exactly. The way most clients would do for SIP and XMPP already.
>> This is just about making sure that they would be able to bring the
>> assumption over to a SIP+XMPP service.
>
> That is unclear to me. Do I visit a website, receive an email with a
> URI, click that to authorize, and receive credentials for my combined
> SIP+XMPP account?

Yes, for example. Pretty much the same way you would get them at
register.jabber.org (while it worked ;) ).

Cheers,
Emil

From lendl@nic.at  Thu Mar 11 14:23:44 2010
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From: Otmar Lendl <lendl@nic.at>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] is this TRIP necessary?
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Penn,

On 10.03.2010 17:23, PFAUTZ, PENN L (ATTCORP) wrote:
>
> But I'd really like to raise what I think is a larger issue - just what 
> are we trying to accomplish?
> 

Good question: what kind of interconnection model is the right one? That is
the 1 M $ question.

> All the LRF and TRIP stuff seems to be tending toward a kind of 
> link-by-link call routing model that to me seems redolent of the kind
> of trunk group by trunk group routing in the PSTN. I thought that that
> was what we were trying to get away from in moving to IP! The original
> promise of ENUM was that it would identify the target and basic Internet
> connectivity would then get you there. As you've pointed out elsewhere,

(For those who have not followed the discussions in speermint, my arguments
are laid out in draft-lendl-speermint-background-02 )

> that original vision didn't really work for most carriers, but carriers
> have tried to replicate parts of it with approaches like the GSMA IPX to
> provide a secure IP fabric for connecting. That allows connectivity
> management via selective advertising in BGP so that the SIP INVITES
> based on same ENUM query response is appropriately routed from different
> originating networks. The same techniques support direct bilateral
> interconnection which is how much of the traffic will flow anyway.

That the carriers seem to be falling for this blatant monopoly play by the
IPX advocates is something that never fails to amaze me.

I just don't see how bilateral peerings outside the IPX can be visible in
the Layer 3 routing map of the IPX and thus be eligible to be used as
transit links. All this fancy BGP trickery only works if *all* players in
the VoIP routing space are IPX members (will the IPX admit corporate PBxs
into their net?) and all transit-capable links between them are within the
IPX framework.

> So, when would you want something TRIP like? To me the remaining case 
> seems to be when you're searching for the best intermediary when you're 
> neither directly connected nor linked to some kind of IPX like fabric 
> that can get to the target. I expect intermediaries to compete and
> scale to matter. When I need an intermediary I'll generally prefer that
> they handle the whole job. Some pricing tables will let me choose and
> I'm not sure I need/want those coming across the NNI as reachability 
> advertisements.

If the main problem is that you can't fully mesh all SSPs (which includes
PBXs), then shifting the interconnection duty to "intermediaries" does
simplify the problem a bit, but it does not solve it. That step only works
if all the intermediaries are members of the IPX.

This is not realistic.

There will be competing peering infrastructures. Starting from other
commercial undertakings in the western world (Xconnect & co) to perhaps a
Chinese peering platform and an Arabic one.

Now, will all intermediaries join all those platforms? Will any two of them
share at least one? Or will there be a hierarchy of intermediaries (just
like the tiers in the PSTN and the Internet world)? How will routing
information be passed between the tiers? Across multiple fabrics?

Maybe the IPX proponents have a plan for that. I haven't seen it.

> Add to this the downward pressure on rates that is only going
> accelerate and I wonder how much need there really is. I may be biased
> from my vantage point though and perhaps someone can provide the
> compelling use case.

>From my point of view, the pressure on rates will continue, and in the long
term they will be replaced by the cost the destination places on his attention.

A hardcore end to end interconnection model will run into the same problems
email faces these days. The price I pay to my voice operator will shift
from paying for the transmission capacity to what additional trust and
accountability he can confer the called party.

What I can imagine for the future is a scenario where PBXs/carriers will
allow calls from trusted parties (however that is defined) to reach their
ingress points without settlement. But I doubt that I will allow my phone
to ring just because some punk in Indonesia sends my PBX a few IP packets.
That will be one of the jobs of a carrier: I will want to get the call
either directly from someone who falls under my criteria of a trusted
communication partner or from some carrier that either pays me to let the
phone ring, guarantees that he polices his users, or simply charges his
users for calling me.

We're entering a phase where attention is more valuable than the 64kbit/s
transmission capacity of a traditional voice channel.

The business model of carriers might need to adjust to that, and I
definitely think there is a role for them here. If not, voice will move
from being a service to being a product that I buy at my local electronics
discounter just as I buy a wlan AP there.

(And no, Skype / Google Talk / <insert you favorite walled garden> won't
cut it. Not for replacing a word-wide, ubiquitous medium that can act as a
global default communication path.)

But we're getting slightly off tangent here ...

Anyway, I few years ago I proposed a scheme that tried to integrate both
RFC 3263-style end-2-end routing, controlled opportunistic peering, and
old-style PSTN-like transit routing into one overall scheme. This would
have allowed SSPs full control over their peering/interconnection
strategies and would have allow mixed mode arrangements and thus gradual
changes in policy to allow for an evolution of the interconnection ecosystem.

Speermint wasn't interested.

otmar
-- 
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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On 3/11/10 3:07 PM, Emil Ivov wrote:
> On Thu, Mar 11, 2010 at 9:20 PM, Peter Saint-Andre <stpeter@stpeter.im>=
 wrote:
>>>> 1. What kind of URI are you talking about?
>>>
>>> The the stpeter@stpeter.im kind.
>>
>> Since it lacks a scheme, stpeter@stpeter.im is not a URI.
>=20
> True. I guess ID would have been a better choice. Is this a problem
> for any other reason than the misuse of the term?

Life certainly is simpler if your chat ID and your voice ID are the same
-- and *really* simple if the same as your email ID. This is how service
providers such as Google do things. Naturally, not every provider wants
to offer a combined email+chat+voice service because it's a lot more work=
=2E

>>>> 4. Does the client derive a username from the URI so that it can use=
 an
>>>> authentication mechanism that expects a username and password?
>>>
>>> Yes, exactly. The way most clients would do for SIP and XMPP already.=

>>> This is just about making sure that they would be able to bring the
>>> assumption over to a SIP+XMPP service.
>>
>> That is unclear to me. Do I visit a website, receive an email with a
>> URI, click that to authorize, and receive credentials for my combined
>> SIP+XMPP account?
>=20
> Yes, for example.=20

OK. But it's still now clear what engineering work is required here. In
the flow you describe, there is some social communication with the end
user ("your chat+voice account has been created!"), but no automated way
of informing the user's software that the account supports XMPP for chat
and SIP for voice. How exactly does the provider communicate that fact
to the application, given that the user's SIP+XMPP application is not
involved in the account provisioning process (as far as I can tell)? Are
you thinking that the end user will employ their SIP+XMPP application to
complete account provisioning, or that the application will communicate
with a web-based registration system somehow? Is there discovery
involved on the part of the SIP+XMPP application?

> Pretty much the same way you would get them at
> register.jabber.org (while it worked ;) ).

Yeah yeah yeah. :P We're working on it. Running code isn't always a lot
of fun for those who run the service...

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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On 11.03.2010 00:48, Dean Willis wrote:
>
> On Mar 10, 2010, at 5:16 PM, Daryl Malas wrote:
>
>> Dean,
>>
>> Based on your assessment below, let me try and summarize:
>>
>> Two IETF efforts:
>> 1) Egress Route Method for SIP (Intra-domain)
>> 2) Dynamic Routing Method for SIP (Inter-domain)
>
> It's possible both could be done by one protocol, possibly with two 
> modes of operation.
>

I think you're onto something here. A few observations from the context 
of business-quality video conferencing/telepresence (some may be on a 
tangent, but as a different perspective on related problems):

Even within an enterprise network, there may be different administrative 
domains. E.164 numbers are starting to become really difficult to manage 
as geography is less relevant when the clients roam. Due to this and 
because of an increasing number of SIP-based non-PSTN services that 
connect into the voice-fabric of the enterprise (presence federation, 
hunt groups across voice, video, PC client), more and more a 
domain-based routing model are mixed with the traditional e.164 dial 
plan. If you do a 4Mb/s video call, choosing the right egress point from 
one administrative domain to another or towards a peering 
partner/service provider becomes critical due to the bandwidth 
requirements and need for a capable client at the other end. Choosing 
the right egress can actually influence whether the call goes through at 
all (or end up as voice only, when yuo could have had video).

We see more and more mixed environments where SIP clients either 
register to or send SIP messages over other vendors' SIP servers. If the 
call is routed (or forked) to multiple destinations, you can never be 
sure that you actually get the "best" result (as in get to the callee 
client with the most compatible functionality), and in fact, often the 
call will fail because you may get a quicker response from a 
less-capable network/device, it attempts to take the call with a 200 OK 
(or maybe Not supported here) and then something fails due to 
incompatibilities in SDP. There is not only a question of choosing the 
right egress, but when you move outside your domain, you don't have any 
control, so the problem becomes bigger.

As a service provider offering connectivity to the enterprise, there is 
no good way to plan the egress together with the enterprise. A 4 Mb/s 
video call may be attempted over a SIP trunk not capable of a video call 
(e.g. local branch office in one country), while the same call could 
have been successfully forwarded through the main office in another 
country. Well, you tell the user: you used the wrong URI or number. Or 
the hunt group was configured with wrong priorities.  Why should the 
user care?

Then once in the service provider's network, the calls have a tendency 
to use e.164 numbers (because that's what the service provider "likes") 
with a domain that sometimes seems to be accidental. If the call is a 
PSTN call, it's much cheaper to route the call over the old 
interconnects, but if it's a 4Mb/s video call, you may want to route 
over another peering fabric for video. You can solve that by separating 
voice and video (which is mostly done as video often is over the data 
network) and the address space so that the routing decision is really 
left to the user.

Some of these examples have multiple problems in them, but following the 
discussion, I have been a bit worried that the focus on fixing a problem 
for service providers doing regular voice calls and solving it with ENUM 
will make us forget that there are some other related problems that are 
looming in the not so distant future. I am therefore happy that the 
focus has shifted a bit towards solving the generic problem.

And BTW, the problems are not really only video-focused. If you want to 
peer HD voice traffic, you need a way to filter out the e.164 numbers 
you can route over an HD-capable peering network, while the remaining 
calls go the cheap route. The typical resolution I have seen is to set 
up an ENUM service. I haven't really seen how this can scale to using 
URIs, and to a complex set of capabilities (IM, presence, video of 
different flavors...)
g-)




> -- 
> Dean
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From enrico.marocco@telecomitalia.it  Fri Mar 12 01:58:56 2010
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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Peter Saint-Andre wrote:
>>>>> 1. What kind of URI are you talking about?
>>>> The the stpeter@stpeter.im kind.
>>> Since it lacks a scheme, stpeter@stpeter.im is not a URI.
>> True. I guess ID would have been a better choice. Is this a problem
>> for any other reason than the misuse of the term?
> 
> Life certainly is simpler if your chat ID and your voice ID are the same
> -- and *really* simple if the same as your email ID. This is how service
> providers such as Google do things. Naturally, not every provider wants
> to offer a combined email+chat+voice service because it's a lot more work.

The discussion is getting very close to the point. If you are a
multi-service provider with a user base of millions, you can expect
developers to offer custom integration of your services in their
platforms in order to make life easier to the users. That's basically
why iPhone and Android devices offer a shortcut for configuring
email+IM+calendar+whatever Google services altogether, simply entering
your credentials once.

Of course things work like that for Google and maybe for a few others,
but certainly not for small providers. The goal of the draft is to
stimulate discussion on the topic and to explore how existing mechanisms
(like, e.g., the domain parameter used in DIGEST authentication) can be
used to achieve seamless integration of SIP and XMPP accounts in a
standard way. We certainly do not want to mandate service providers to
offer both services, nor to standardize some exotic multi-purpose
configuration framework that will take years to define and will never
see the light.

-- 
Ciao,
Enrico

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From emcho@sip-communicator.org  Fri Mar 12 04:54:12 2010
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Date: Fri, 12 Mar 2010 13:54:11 +0100
From: Emil Ivov <emcho@sip-communicator.org>
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To: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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Adding to what Enrico said,

Peter Saint-Andre =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0:
> OK. But it's still now clear what engineering work is required here. In=

> the flow you describe, there is some social communication with the end
> user ("your chat+voice account has been created!"), but no automated wa=
y
> of informing the user's software that the account supports XMPP for cha=
t
> and SIP for voice. How exactly does the provider communicate that fact
> to the application, given that the user's SIP+XMPP application is not
> involved in the account provisioning process (as far as I can tell)?=20

The provider doesn't need to communicate that fact to the application.
The user does this by choosing to create a SIP+XMPP account. Again, it's
pretty much what what already happens with multi protocol clients for
SIP and XMPP only accounts.

Cheers,
Emil


From stpeter@stpeter.im  Fri Mar 12 07:23:47 2010
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Date: Fri, 12 Mar 2010 08:06:53 -0700
From: Peter Saint-Andre <stpeter@stpeter.im>
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To: Emil Ivov <emcho@sip-communicator.org>
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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On 3/12/10 5:54 AM, Emil Ivov wrote:
> Adding to what Enrico said,
>=20
> Peter Saint-Andre =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0:
>> OK. But it's still now clear what engineering work is required here. I=
n
>> the flow you describe, there is some social communication with the end=

>> user ("your chat+voice account has been created!"), but no automated w=
ay
>> of informing the user's software that the account supports XMPP for ch=
at
>> and SIP for voice. How exactly does the provider communicate that fact=

>> to the application, given that the user's SIP+XMPP application is not
>> involved in the account provisioning process (as far as I can tell)?=20
>=20
> The provider doesn't need to communicate that fact to the application.
> The user does this by choosing to create a SIP+XMPP account. Again, it'=
s
> pretty much what what already happens with multi protocol clients for
> SIP and XMPP only accounts.

Yes, this is all quite beautiful. But I still don't see what engineering
work is required within the IETF to make this happen.

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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From stpeter@stpeter.im  Fri Mar 12 07:27:00 2010
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Date: Fri, 12 Mar 2010 08:08:40 -0700
From: Peter Saint-Andre <stpeter@stpeter.im>
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To: Enrico Marocco <enrico.marocco@telecomitalia.it>
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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On 3/12/10 2:59 AM, Enrico Marocco wrote:
> Peter Saint-Andre wrote:
>>>>>> 1. What kind of URI are you talking about?
>>>>> The the stpeter@stpeter.im kind.
>>>> Since it lacks a scheme, stpeter@stpeter.im is not a URI.
>>> True. I guess ID would have been a better choice. Is this a problem
>>> for any other reason than the misuse of the term?
>>
>> Life certainly is simpler if your chat ID and your voice ID are the sa=
me
>> -- and *really* simple if the same as your email ID. This is how servi=
ce
>> providers such as Google do things. Naturally, not every provider want=
s
>> to offer a combined email+chat+voice service because it's a lot more w=
ork.
>=20
> The discussion is getting very close to the point. If you are a
> multi-service provider with a user base of millions, you can expect
> developers to offer custom integration of your services in their
> platforms in order to make life easier to the users. That's basically
> why iPhone and Android devices offer a shortcut for configuring
> email+IM+calendar+whatever Google services altogether, simply entering
> your credentials once.
>=20
> Of course things work like that for Google and maybe for a few others,
> but certainly not for small providers.=20

Indeed.

> The goal of the draft is to
> stimulate discussion on the topic and to explore how existing mechanism=
s
> (like, e.g., the domain parameter used in DIGEST authentication) can be=

> used to achieve seamless integration of SIP and XMPP accounts in a
> standard way.=20

OK, I do think that's worth exploring.

> We certainly do not want to mandate service providers to
> offer both services, nor to standardize some exotic multi-purpose
> configuration framework that will take years to define and will never
> see the light.

+1 to that!

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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From emcho@sip-communicator.org  Fri Mar 12 07:50:59 2010
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Date: Fri, 12 Mar 2010 16:35:33 +0100
From: Emil Ivov <emcho@sip-communicator.org>
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Subject: Re: [dispatch] Authentication for SIP+XMPP clients
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Peter Saint-Andre =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0:
> On 3/12/10 5:54 AM, Emil Ivov wrote:
>> Adding to what Enrico said,
>>
>> Peter Saint-Andre =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0:
>>> OK. But it's still now clear what engineering work is required here. =
In
>>> the flow you describe, there is some social communication with the en=
d
>>> user ("your chat+voice account has been created!"), but no automated =
way
>>> of informing the user's software that the account supports XMPP for c=
hat
>>> and SIP for voice. How exactly does the provider communicate that fac=
t
>>> to the application, given that the user's SIP+XMPP application is not=

>>> involved in the account provisioning process (as far as I can tell)? =

>> The provider doesn't need to communicate that fact to the application.=

>> The user does this by choosing to create a SIP+XMPP account. Again, it=
's
>> pretty much what what already happens with multi protocol clients for
>> SIP and XMPP only accounts.
>=20
> Yes, this is all quite beautiful. But I still don't see what engineerin=
g
> work is required within the IETF to make this happen.

Ideally - none. If we can agree that SIP+XMPP clients can assume they
can SRV query the SIP and XMPP counterparts of a SIP+XMPP using the
usual records, and that they can also assume they would be using the
same credentials to log against both of these protocols and people feel
that this fits the charter (and doesn't violate the the
no-modifications-on-the-server-side requirement), then all we need is a
paragraph in the sip+xmpp document that describes this. At worst we can
put it in a BCP doc.

Otherwise, people may think that the subject is more complicated than
that and that it needs to be explored further for some reason and in
this case there maybe something else to do, and of course I have no idea
what it may be.

Cheers,
Emil


From rjsparks@nostrum.com  Fri Mar 12 07:57:04 2010
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To: siprec@ietf.org, SIPCORE <sipcore@ietf.org>, codec@ietf.org, DISPATCH list <dispatch@ietf.org>, ecrit@ietf.org, XMPP <xmpp@ietf.org>, rai@ietf.org
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Subject: [dispatch] RAI WG chair changes
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WIth the upcoming IESG and IAB changes, we need to make=20
a bunch of chair changes. After the end of the IETF 77 the chairs=20
for the following WGs will be:

SIPREC=20
Brian Rosen
John Elwell

SIPCORE
Adam Roach
Paul Kyzivat

CODEC
Michael Knappe
Jonathan Rosenberg
Cullen Jennings

DISPATCH
Mary Barnes
Cullen Jennings

ECRIT=20
Richard Barnes
Marc Linsner

XMPP
Joe Hildebrand
Ben Campbell

To help with continuity of the IETF77 meeting, we are going to add the =
new chairs now,=20
and not remove any existing chairs until after the meeting.=20

We would like to express many thinks for all the work all these chairs =
have done and continue to do.=20

Thanks,=20
Cullen, Gonzalo, & Robert








From fluffy@cisco.com  Fri Mar 12 08:01:04 2010
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Subject: [dispatch] NEW SIPREC Working Group
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The SIPREC working group has been approved and you will see the =
announcements soon. I'd like to encourage everyone to go subscribe to =
the siprec mailing list at:

( https://www.ietf.org/mailman/listinfo/siprec )

I'd very much like to thank Brian Rosen and John Elwell for agreeing to =
chair this ship.=20

The tentative plan is to take an hour of time from the sipcore meeting =
and run a SIPREC working group meeting meeting from 10:30 to 11:30 =
Tuesday March 23 in the California B room. Please note this is a =
tentative plan, I'm trying to get this to work but it not finalized yet.=20=


Cullen <RAI AD>





From jmpolk@cisco.com  Fri Mar 12 11:31:23 2010
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At 09:49 AM 3/12/2010, Cullen Jennings wrote:

>The SIPREC working group has been approved and you will see the 
>announcements soon. I'd like to encourage everyone to go subscribe 
>to the siprec mailing list at:
>
>( https://www.ietf.org/mailman/listinfo/siprec )
>
>I'd very much like to thank Brian Rosen and John Elwell for agreeing 
>to chair this ship.

hence the mangled name of "shipwreck" for the wg... like Dean suggested... lol

(it's Friday)

>
>
>The tentative plan is to take an hour of time from the sipcore 
>meeting and run a SIPREC working group meeting meeting from 10:30 to 
>11:30 Tuesday March 23 in the California B room. Please note this is 
>a tentative plan, I'm trying to get this to work but it not finalized yet.
>
>Cullen <RAI AD>
>
>
>
>
>_______________________________________________
>RAI mailing list
>RAI@ietf.org
>https://www.ietf.org/mailman/listinfo/rai


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Hi all

Are there any registered INFO packages and can the regular SIP event
packages based on RFC 3265 be used with INFO mechanism?

Thanks

Regards
Ranjit


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<P><FONT SIZE=3D2 FACE=3D"Arial">Are there any registered INFO packages =
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From adam@nostrum.com  Mon Mar 15 08:48:04 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Greger:

For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
provide about 80% of what you're asking for.

/a

On 3/12/10 01:18, Mar 12, Greger Viken Teigre wrote:
> Even within an enterprise network, there may be different 
> administrative domains. E.164 numbers are starting to become really 
> difficult to manage as geography is less relevant when the clients 
> roam. Due to this and because of an increasing number of SIP-based 
> non-PSTN services that connect into the voice-fabric of the enterprise 
> (presence federation, hunt groups across voice, video, PC client), 
> more and more a domain-based routing model are mixed with the 
> traditional e.164 dial plan. If you do a 4Mb/s video call, choosing 
> the right egress point from one administrative domain to another or 
> towards a peering partner/service provider becomes critical due to the 
> bandwidth requirements and need for a capable client at the other end. 
> Choosing the right egress can actually influence whether the call goes 
> through at all (or end up as voice only, when yuo could have had video).
>
> We see more and more mixed environments where SIP clients either 
> register to or send SIP messages over other vendors' SIP servers. If 
> the call is routed (or forked) to multiple destinations, you can never 
> be sure that you actually get the "best" result (as in get to the 
> callee client with the most compatible functionality), and in fact, 
> often the call will fail because you may get a quicker response from a 
> less-capable network/device, it attempts to take the call with a 200 
> OK (or maybe Not supported here) and then something fails due to 
> incompatibilities in SDP. There is not only a question of choosing the 
> right egress, but when you move outside your domain, you don't have 
> any control, so the problem becomes bigger.
>
> As a service provider offering connectivity to the enterprise, there 
> is no good way to plan the egress together with the enterprise. A 4 
> Mb/s video call may be attempted over a SIP trunk not capable of a 
> video call (e.g. local branch office in one country), while the same 
> call could have been successfully forwarded through the main office in 
> another country. Well, you tell the user: you used the wrong URI or 
> number. Or the hunt group was configured with wrong priorities.  Why 
> should the user care?
>
> Then once in the service provider's network, the calls have a tendency 
> to use e.164 numbers (because that's what the service provider 
> "likes") with a domain that sometimes seems to be accidental. If the 
> call is a PSTN call, it's much cheaper to route the call over the old 
> interconnects, but if it's a 4Mb/s video call, you may want to route 
> over another peering fabric for video. You can solve that by 
> separating voice and video (which is mostly done as video often is 
> over the data network) and the address space so that the routing 
> decision is really left to the user.
>
> Some of these examples have multiple problems in them, but following 
> the discussion, I have been a bit worried that the focus on fixing a 
> problem for service providers doing regular voice calls and solving it 
> with ENUM will make us forget that there are some other related 
> problems that are looming in the not so distant future. I am therefore 
> happy that the focus has shifted a bit towards solving the generic 
> problem.
>
> And BTW, the problems are not really only video-focused. If you want 
> to peer HD voice traffic, you need a way to filter out the e.164 
> numbers you can route over an HD-capable peering network, while the 
> remaining calls go the cheap route. The typical resolution I have seen 
> is to set up an ENUM service. I haven't really seen how this can scale 
> to using URIs, and to a complex set of capabilities (IM, presence, 
> video of different flavors...)
> g-)
>


From pkyzivat@cisco.com  Mon Mar 15 09:09:00 2010
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Adam Roach wrote:
> Greger:
> 
> For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
> provide about 80% of what you're asking for.

Selecting a *UAS* that has the capabilities the caller envisions using 
(such as video) were certainly among the things that were among the 
intended use cases.

I'm dubious about their suitability for selecting a route to reach the 
UAS, assuming its the same UAS you are seeking.

Also callerprefs are no guarantee (not even a superset) of the 
capabilities that will be used on a call. It is an entirely reasonable 
use case for a UAC to invite with no offer, and then negotiate the use 
of voice and video because the UAS offers that. Its also entirely 
reasonable for a call to start out as audio only, for the caller and 
callee to learn as a result of discussion that both are video capable, 
and then add video to the call.

As non-voice media become more mainstream, and as e2e sip sessions 
become more feasible, we should be investigating how the infrastructure 
can avoid breaking these kinds of use cases. Nailing down the route in a 
way that precludes use of media not announced at call establishment time 
isn't going in a very helpful direction.

	Thanks,
	Paul

> /a
> 
> On 3/12/10 01:18, Mar 12, Greger Viken Teigre wrote:
>> Even within an enterprise network, there may be different 
>> administrative domains. E.164 numbers are starting to become really 
>> difficult to manage as geography is less relevant when the clients 
>> roam. Due to this and because of an increasing number of SIP-based 
>> non-PSTN services that connect into the voice-fabric of the enterprise 
>> (presence federation, hunt groups across voice, video, PC client), 
>> more and more a domain-based routing model are mixed with the 
>> traditional e.164 dial plan. If you do a 4Mb/s video call, choosing 
>> the right egress point from one administrative domain to another or 
>> towards a peering partner/service provider becomes critical due to the 
>> bandwidth requirements and need for a capable client at the other end. 
>> Choosing the right egress can actually influence whether the call goes 
>> through at all (or end up as voice only, when yuo could have had video).
>>
>> We see more and more mixed environments where SIP clients either 
>> register to or send SIP messages over other vendors' SIP servers. If 
>> the call is routed (or forked) to multiple destinations, you can never 
>> be sure that you actually get the "best" result (as in get to the 
>> callee client with the most compatible functionality), and in fact, 
>> often the call will fail because you may get a quicker response from a 
>> less-capable network/device, it attempts to take the call with a 200 
>> OK (or maybe Not supported here) and then something fails due to 
>> incompatibilities in SDP. There is not only a question of choosing the 
>> right egress, but when you move outside your domain, you don't have 
>> any control, so the problem becomes bigger.
>>
>> As a service provider offering connectivity to the enterprise, there 
>> is no good way to plan the egress together with the enterprise. A 4 
>> Mb/s video call may be attempted over a SIP trunk not capable of a 
>> video call (e.g. local branch office in one country), while the same 
>> call could have been successfully forwarded through the main office in 
>> another country. Well, you tell the user: you used the wrong URI or 
>> number. Or the hunt group was configured with wrong priorities.  Why 
>> should the user care?
>>
>> Then once in the service provider's network, the calls have a tendency 
>> to use e.164 numbers (because that's what the service provider 
>> "likes") with a domain that sometimes seems to be accidental. If the 
>> call is a PSTN call, it's much cheaper to route the call over the old 
>> interconnects, but if it's a 4Mb/s video call, you may want to route 
>> over another peering fabric for video. You can solve that by 
>> separating voice and video (which is mostly done as video often is 
>> over the data network) and the address space so that the routing 
>> decision is really left to the user.
>>
>> Some of these examples have multiple problems in them, but following 
>> the discussion, I have been a bit worried that the focus on fixing a 
>> problem for service providers doing regular voice calls and solving it 
>> with ENUM will make us forget that there are some other related 
>> problems that are looming in the not so distant future. I am therefore 
>> happy that the focus has shifted a bit towards solving the generic 
>> problem.
>>
>> And BTW, the problems are not really only video-focused. If you want 
>> to peer HD voice traffic, you need a way to filter out the e.164 
>> numbers you can route over an HD-capable peering network, while the 
>> remaining calls go the cheap route. The typical resolution I have seen 
>> is to set up an ENUM service. I haven't really seen how this can scale 
>> to using URIs, and to a complex set of capabilities (IM, presence, 
>> video of different flavors...)
>> g-)
>>
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From adam@nostrum.com  Mon Mar 15 09:24:36 2010
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On 3/15/10 11:09, Mar 15, Paul Kyzivat wrote:
>
>
> Adam Roach wrote:
>> Greger:
>>
>> For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
>> provide about 80% of what you're asking for.
>
> Selecting a *UAS* that has the capabilities the caller envisions using 
> (such as video) were certainly among the things that were among the 
> intended use cases.
>
> I'm dubious about their suitability for selecting a route to reach the 
> UAS, assuming its the same UAS you are seeking.

That's the other 20%. :)


> Also callerprefs are no guarantee (not even a superset) of the 
> capabilities that will be used on a call. It is an entirely reasonable 
> use case for a UAC to invite with no offer, and then negotiate the use 
> of voice and video because the UAS offers that. Its also entirely 
> reasonable for a call to start out as audio only, for the caller and 
> callee to learn as a result of discussion that both are video capable, 
> and then add video to the call.
>
> As non-voice media become more mainstream, and as e2e sip sessions 
> become more feasible, we should be investigating how the 
> infrastructure can avoid breaking these kinds of use cases. Nailing 
> down the route in a way that precludes use of media not announced at 
> call establishment time isn't going in a very helpful direction.

Nailing down a route... for signaling? I'm a bit confused about how that 
is going to preclude certain media types.

Now, if we want to talk about media steering, that's a completely 
different issue. I don't think SIP routing plays into that.

/a


From pkyzivat@cisco.com  Mon Mar 15 09:33:54 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Adam Roach wrote:
> On 3/15/10 11:09, Mar 15, Paul Kyzivat wrote:
>>
>>
>> Adam Roach wrote:
>>> Greger:
>>>
>>> For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
>>> provide about 80% of what you're asking for.
>>
>> Selecting a *UAS* that has the capabilities the caller envisions using 
>> (such as video) were certainly among the things that were among the 
>> intended use cases.
>>
>> I'm dubious about their suitability for selecting a route to reach the 
>> UAS, assuming its the same UAS you are seeking.
> 
> That's the other 20%. :)

I think we're following the 80/80 rule here. :-)

>> Also callerprefs are no guarantee (not even a superset) of the 
>> capabilities that will be used on a call. It is an entirely reasonable 
>> use case for a UAC to invite with no offer, and then negotiate the use 
>> of voice and video because the UAS offers that. Its also entirely 
>> reasonable for a call to start out as audio only, for the caller and 
>> callee to learn as a result of discussion that both are video capable, 
>> and then add video to the call.
>>
>> As non-voice media become more mainstream, and as e2e sip sessions 
>> become more feasible, we should be investigating how the 
>> infrastructure can avoid breaking these kinds of use cases. Nailing 
>> down the route in a way that precludes use of media not announced at 
>> call establishment time isn't going in a very helpful direction.
> 
> Nailing down a route... for signaling? I'm a bit confused about how that 
> is going to preclude certain media types.
> 
> Now, if we want to talk about media steering, that's a completely 
> different issue. I don't think SIP routing plays into that.

Perhaps I misinterpreted the draft, but I got the distinct impression 
that media steering, following the signaling path, was presumed, and the 
primary issue being raised.

Hence my concern.

	Paul

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On 3/15/10 11:33, Mar 15, Paul Kyzivat wrote:
>
> Adam Roach wrote:
>> Nailing down a route... for signaling? I'm a bit confused about how 
>> that is going to preclude certain media types.
>>
>> Now, if we want to talk about media steering, that's a completely 
>> different issue. I don't think SIP routing plays into that.
>
> Perhaps I misinterpreted the draft, but I got the distinct impression 
> that media steering, following the signaling path, was presumed, and 
> the primary issue being raised.

And, to be clear, that's probably what Daryl has in mind. But it doesn't 
sound like what Greger is trying to do.

Daryl is working within the strictures of reproducing the business 
relationships of the PSTN in a SIP network. And the path that has been 
taken, for better or worse, involves the heavy use of SBCs and other 
devices that impose a rather strict control on the way things are done. 
These devices largely ensure that the media path follows the signaling 
path, even when that's not the optimal thing to do. The chances of such 
networks deploying video, hi-def voice, or any of the other really neat 
things that Greger mentions are slim to none. This, of course, is NOT 
Daryl's fault. But it's where the egress-route work comes from.

Greger is speaking more towards the network you seem to have in mind -- 
one in which end-to-end SIP calls are possible, without SBCs or similar 
control boxes clobbering new services and mangling the ability to convey 
novel media from one endpoint to another. Networks where we don't need 
to define alternate connection models for protocols simply to 
accommodate network elements that do things that are explicitly 
disallowed by protocol specification.

Don't get me wrong -- the business concerns driving SBC deployment are 
legitimate. But that doesn't change the fact that they destroy innovation.

In many ways, these approaches represent an unfortunate bifurcation in 
the SIP road. One path leads to a rather stagnant network that manages 
to reproduce what we've been doing for the past century, possibly with 
some minor operational cost savings. The other leads to vibrant new 
services, end-to-end multimedia, and innovative applications.

But that fork has already been taken, and I don't think we're going to 
join the paths back again any time soon. So, for the time being, we find 
ourselves doing work in the stagnant world (SPEERMINT, MARTINI, DRINKS) 
as well as the vibrant world (P2PSIP, SIP Certs, CODEC). And, while 
there can be some commonality between solutions, the increasing 
difficulty that people in the two different worlds have acknowledging 
the other makes it nearly impossible to come up with solutions that make 
everyone happy.

The constituents of the stagnant world will find any vibrant solutions 
to be too complicated, too general, and subject to being broken by their 
intermediaries. The constituents of the vibrant world will find any 
stagnant solutions to be too brittle, too special-purpose, and too 
constrained.

Getting back to the problem at hand.

In the stagnant world, steering the signaling effectively steers the 
media. And if you can combine that steering with E.164-number lookup 
(because the stagnant world uses phone numbers), then the problem is 
solved. Anything more general is going to be too complicated, too 
general, and subject to be broken by SBCs.

In the vibrant world, signaling and media follow different paths. 
Steering the signaling is unlikely to change the media path. If you want 
to steer media, you would need to do so explicitly. We already have some 
mechanisms defined in the RAI area to do this -- TURN comes to mind -- 
and there are probably others at layer 3 that I'm not familiar with. 
This is where we may need some new work. Anything that requires E.164 
numbers as identifiers or that assumes media follows the signaling path 
is going to be too brittle, too special purpose, and too constrained.

Which is to say that I don't think there is a solution that satisfies 
both groups. The divergence from the fork between the stagnant world and 
the vibrant world is great enough that there is no common ground left 
for these kinds of issues. As odious as it is, I'm afraid we're stuck 
with solving this problem twice. The real challenge is ensuring that 
stagnant solutions don't do lasting damage to the core protocols -- we 
need to make sure they don't destroy the vibrant world with collateral 
damage.

Which is why I support a simple, non-normative, informational draft 
about the form of URLs that one might want to put in ENUM. It satisfies 
the requirements of the stagnant world, and doesn't even have any 
normative impacts on any protocol anywhere. It's harmless, and lets the 
people in the vibrant world concentrate on solving the problem in a more 
general way without having to accommodate the peculiarities of network 
elements living in the stagnant world.

/a

From gonzalo.camarillo@ericsson.com  Tue Mar 16 03:58:57 2010
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Hi Ranjit,

there is a blackout period before IETF meetings in which one cannot
submit drafts. Please, submit this version as soon as the blackout
period is over and have Paul have a look at it so that he can confirm he
is happy with it.

Thanks,

Gonzalo


On 11/03/2010 8:10 PM, Avasarala Ranjit-A20990 wrote:
> Hi All
> 
> I am not able to upload the updated version (-04) of the I-D due to some
> issue with the I-D submission tool - (Says I can submit I-D only on
> 2010-03-22).
> Until then I am attaching a copy of the updated version (txt and html
> formats).
> 
> Please review and comment.
> 
> Thanks
> 
> Note: Let me know if there is any other way to upload the I-D document
> and will be happy to do it.
> 
> 
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 02, 2010 7:19 PM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> I think this all sounds good now.
> 
>         Thanks,
>         Paul
> 
> Avasarala Ranjit-A20990 wrote:
>> Inline
>>
>> <Ranjit-Mar2>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, March 02, 2010 2:16 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
>> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
>> ofdraft-avasarala-dispatch-comm-div-notification-03]
>>
>> inline...
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>> I'm still a little confused here.
>>> If phone is on and subscription is active, then in general
>>> notification should be possible, regardless of registration. I
>>> realize
>>
>>> IMS doesn't permit that, but from a general standard perspective we
>>> shouldn't assume it. The manifestation in this case is that the
>>> NOTIFY
>>
>>> would fail and the subscription would eventually be torn down.
>>>
>>> So I think the cases are:
>>> - there is a subscription and notification works
>>> - there is a subscription but notification fails
>>> - there is no subscription.
>>>
>>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as
>>> Notification fails, we can say - not sending notification due to (1)
>>> no match of filter criteria (2) phone not registered or outside
>>> coverage area due to which there is no connectivity and hence
>>> notification cannot be sent Anyway if no subscription, no
>> notification.
>>
>> Are you suggesting that you might have a subscription, and have the
>> filter criteria met, and yet not attempt to send a NOTIFY because the
>> phone is not registered or has no network connectivity???
>>
>> That seems wrong. ISTM that either you would remove the subscription,
>> or else you would try to send the NOTIFY and then discover that it
> fails.
>>
>> <Ranjit-Mar2> yes the subscription could be removed when the device is
> 
>> not reachable.
>>
>> OTOH, this probably ought not to be within scope of what is discussed
>> in this draft.
>>
>> <Ranjit-Mar2> yes true not within the scope of this draft.
>>
>>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>>> contacts: sip:line1@alice.office.atlanta.com and
>>>> sip:line2@alice.home.atlanta.com and
>>>>     both the phones ring, then it would be a case of parallel
>> forking.
>>>> Now say one or both the phones have set diversion rules, to divert
>>>> the
>>> In this case, what does it mean for "one or both phones have set
>>> diversion rules"? Does that somehow imply distinct rules for each
>> phone?
>>> I don't see how that could work. Aren't the diversion rules for the
>>> *AOR*?
>>>
>>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there
>>> is
>>
>>> a diversion rule set for the AoR and when diversions occur, the
>>> notifications are Sent to the AoR.
>>
>> Not sent to the AoR - sent to the remote contact of the dialog for
>> each subscription to the AoR. Same point as below that you agreed
> with.
>>
>> <Ranjit-Mar2> Yes agree with u.
>>
>>> I see they could be set/changed from either phone. But its a single
>>> set of rules isn't it? So I might set the rules from the office, and
>>> then cancel them from home.
>>>
>>> <Ranjit-Feb26-2> yes single set and could be set from either of the
>>> phones. True.
>>>
>>>> call to say
>>>>     sip:voicemail@alice.atlanta.com, then the notifications would
>>>> get
>>
>>>> generated with respect to both the contacts.
>>> Again I'm confused by the terminology you use, and how it relates
>>> operationally. The diversion happens on the AOR, and the
>>> subscriptions
>>
>>> are on the AOR, and the notifications are sent to the subscribers. I
>>> don't see where the registered contacts enter into the picture at
> all.
>>>
>>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>>
>>> If you have something else in mind, I hope you can explain it better
>>> in the draft.
>>>
>>>> [4] Yes the value of N is configurable and depends on how many
>>>> diversions the subscribe wants to keep or could be dependent on
>>>> server policy (some maximum
>>>>     number). If the server policy is to send notifications as and
>>>> when they occur and not keep any divsersions information, then the
>>>> value of last N
>>>>     diversions will be zero. But if the server chooses to retain the
> 
>>>> last diversion always, then the value of N will be 1 and so on.
>>> All is straightforward for N>=1.
>>>
>>> For N=0 things are messy. If we are modeling this as state, and
>>> notifications are about state changes, then the new diversion is
>>> added
>>
>>> to the (previously empty) state and that triggers notifications to
>>> all
>>
>>> current subscriptions. Then if you don't want to retain even this
>>> much
>>
>>> state (because N=0) you immediately remove that version from the
>> state.
>>> But that is again a state change, so you end up with another
>>> notification of that state change.
>>>
>>> <Ranjit-Feb26-2> True. Actually the notifications are for informing
>>> the subscriber of the call diversions that happen. The notifications
>>> are not for informing the subscriber of state changes in the notifier
> 
>>> or the number of diversions that occurred. As I mentioned in the
>>> abstract as well as 24.604, the main intent of CDIVN service is to
>>> inform the subscribers of communication diversions that occur on
>>> their
>>
>>> behalf in the network and this information will help them manage
>>> their
>>
>>> rules better.
>>
>> I am not certain, but think we may still be talking past one another
>> to some extent.
>>
>> The state can be considered just a formalism for defining the service
>> clearly, regardless of whether it has intrinsic value to the CDIV
>> service. But it *is* a formalism. So if it is introduced, then its
>> behavior can't be ignored when inconvenient. Defining the state with
>> N=0 leads to some undesired behavior. The simplest way around that is
>> to insist that N be >=1.
>>
>> <Ranjit-Mar2> Ok in that case we can say that value of N >=1 and the
>> notifier has always maintains history of last diversion at a minimu.
>> It can maintain more than 1 too depending on the policy.
>>
>> If its important to you that no state be retained once a notification
>> has been sent about the most recent diversion, then you will have to
>> find some other way to specify the service, that still meets the
>> expectations of 3265. I don't know if there is anything there that
>> Adam would consider valid. We will have to get his opinion on that.
>>
>> <Ranjit-Mar2> In that case I am OK maintaining the hostory of
>> diversions that occurred as part of state information and retaining
>> that state even after the notification for that diversion is sent - so
> 
>> that in case I do not get a confirmation, I would be able to re-send
> the notification.
>> Also maintaining state of diversions would help when the notifications
> 
>> need to sent only at a particular time (if fikter criteria mentions
> it).
>>
>>
>>       Thanks,
>>       Paul
>>
>>> <Ranjit-Feb26-2> But if we want to include the information related to
> 
>>> how many diversions have happened till time (say from 12 AM ) till
>>> the
>>
>>> time the notification is being sent, then as you say we can store the
> 
>>> diversions. This would be useful when the subscriber specifies in the
> 
>>> filter criteria to send notifications only during certain time like
>>> between 5 and 6 PM and not whenever a diversion occurs. Then N is
>>> grater than 1.
>>
>>> I presume you don't want that behavior. If you can live with N>=1
>>> then
>>
>>> we don't have the problem at all. If you need to support N=0 and
>>> don't
>>
>>> want the extra notify, then we have to figure out if there is some
>>> trick to how things are modeled to get the desired effect. I've
>>> suggested something previously, but Adam didn't like it.
>>>
>>> This needs sorting out.
>>>
>>>      Thanks,
>>>      Paul
>>


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On Mar 15, 2010, at 11:24 AM, Adam Roach wrote:
>>
> Nailing down a route... for signaling? I'm a bit confused about how  
> that is going to preclude certain media types.
>
> Now, if we want to talk about media steering, that's a completely  
> different issue. I don't think SIP routing plays into that.

Ah, but it does.

Given the "evil peering architecture" (we really need a better  
dictionary for this) under discussion, the signaling route selected  
affects the media route, perhaps even limiting the media changes  
available once the call has been established. We're really talking  
about application-driven IP routing as an end-goal. The A-IMS model of  
using SIP to set up one's permission for using an HTTP proxy for web  
browsing is just the tip of this iceberg.

Imagine, if you will, an IP network where each router takes into  
consideration not just the destination address of a packet but all  
sorts of other information (aka "metadata") including packet source,  
application, billing relationships, current contractual quotas,  
censorship rules, and all sorts of other stuff before it chooses the  
route by which it is going to send that packet towards its next hop.

Making these routing decisions at SIP proxies is an interim step. It  
breaks the Internet up into myriad islands, each connected only by  
application gateways. IP routing as we know it is still used inside  
each island (although perhaps only on a transitional basis). But  
routing between islands is driven by more business-like (or even  
political) factors; simple route efficiency is inadequate for  
expressing the complex and shifting policy framework of the net.

The logical outgrowth of the model is that every router become both a  
policy decision and policy enforcement point, provisioned with and  
taking into account the operational and political rules that govern  
utilization. Only a tiny fraction of these rules have anything to do  
with efficient delivery of a packet from Alice to Bob. Rather, they  
include things like seeing that the application is SOX compliant,  
making sure that the appropriate watchers-and-listeners have access to  
the content, diverting widget-orders from the intended recipient  
towards an ally of the router operator's and so on. Of course,  
efficient delivery and bandwidth management is still crucial, so  
perhaps something like RSVP will be used within "islands" to separate  
the policy decisions made by the application gateways from the policy  
enforcement of packet delivery, so perhaps it won't diverge totally  
from the Internet as we know it.


Certainly this sort of system is the end-goal of very few people who  
are working on the technology today, but it is the (or at least "a")  
logical outgrowth of application-specific routing. It's what happens  
when we abandon the end-to-end principle and its related concepts of  
address-based application-independent routing, etc.

Most people aren't trying to "change the Internet".  People are just  
asking for ways to solve today's problems. Problems like "How do I  
tell my SIP nodes which SBC to use for calls to area code 212" or  
"Which NAT router should I use to reach Google most efficiently"? have  
as their basis an understanding that we've moved somewhat away from  
the Internet and towards something very different -- a route is not  
just a route, an application is not just an IP Address and a port  
number,  and no node can be trusted to handle ALL of an important  
task, because everything eventually gets hacked.

So, do we take this understanding of the difference between the common  
"real" network and the theoretical Internet and attempt to solve  
people's problems as they see them today, or do we step back and ask  
"What is the Internet way to solve this problem, why isn't it working,  
and what do we have to do to make it work right?" My guess is the  
former; it's easier, and there's more money to be made on it today. In  
the long run it destroys our world, but we just aren't geared towards  
understanding those kinds of things and would much rather just have  
today's problem fixed today.

--
Dean
--Apple-Mail-13-556657270
Content-Type: text/html;
	charset=US-ASCII
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 15, 2010, =
at 11:24 AM, Adam Roach wrote:</div><blockquote =
type=3D"cite"><div><blockquote type=3D"cite"><font =
class=3D"Apple-style-span" =
color=3D"#000000"><br></font></blockquote>Nailing down a route... for =
signaling? I'm a bit confused about how that is going to preclude =
certain media types.<br><br>Now, if we want to talk about media =
steering, that's a completely different issue. I don't think SIP routing =
plays into that.<br></div></blockquote></div><br><div>Ah, but it =
does.</div><div><br></div><div>Given the "evil peering architecture" (we =
really need a better dictionary for this) under discussion, the =
signaling route selected affects the media route, perhaps even limiting =
the media changes available once the call has been established. We're =
really talking about application-driven IP routing as an end-goal. The =
A-IMS model of using SIP to set up one's permission for using an HTTP =
proxy for web browsing is just the tip of this =
iceberg.</div><div><br></div><div>Imagine, if you will, an IP network =
where each router takes into consideration not just the destination =
address of a packet but all sorts of other information (aka "metadata") =
including packet source, application, billing relationships, current =
contractual quotas, censorship rules, and all sorts of other stuff =
before it chooses the route by which it is going to send that packet =
towards its next hop.</div><div><br></div><div>Making these routing =
decisions at SIP proxies is an interim step. It breaks the Internet up =
into myriad islands, each connected only by application gateways. IP =
routing as we know it is still used inside each island (although perhaps =
only on a transitional basis). But routing between islands is driven by =
more business-like (or even political) factors; simple route efficiency =
is inadequate for expressing the complex and shifting policy framework =
of the net.</div><div><br></div><div>The logical outgrowth of the model =
is that every router become both a policy decision and policy =
enforcement point, provisioned with and taking into account the =
operational and political rules that govern utilization. Only a tiny =
fraction of these rules have anything to do with efficient delivery of a =
packet from Alice to Bob. Rather, they include things like seeing that =
the application is SOX compliant, making sure that the appropriate =
watchers-and-listeners have access to the content, diverting =
widget-orders from the intended recipient towards an ally of the router =
operator's and so on. Of course, efficient delivery and bandwidth =
management is still crucial, so perhaps something like RSVP will be used =
within "islands" to separate the policy decisions made by the =
application gateways from the policy enforcement of packet delivery, so =
perhaps it won't diverge totally from the Internet as we know =
it.</div><div><br></div><div><br></div><div>Certainly this sort of =
system is the end-goal of very few people who are working on the =
technology today, but it is the (or at least "a") logical outgrowth of =
application-specific routing. It's what happens when we abandon the =
end-to-end principle and its related concepts of address-based =
application-independent routing, =
etc.&nbsp;</div><div><br></div><div>Most people aren't trying to "change =
the Internet". &nbsp;People are just asking for ways to solve today's =
problems. Problems like "How do I tell my SIP nodes which SBC to use for =
calls to area code 212" or "Which NAT router should I use to reach =
Google most efficiently"? have as their basis an understanding that =
we've moved somewhat away from the Internet and towards something very =
different -- a route is not just a route, an application is not just an =
IP Address and a port number, &nbsp;and no node can be trusted to handle =
ALL of an important task, because everything eventually gets =
hacked.</div><div><br></div><div>So, do we take this understanding of =
the difference between the common "real" network and the theoretical =
Internet and attempt to solve people's problems as they see them today, =
or do we step back and ask "What is the Internet way to solve this =
problem, why isn't it working, and what do we have to do to make it work =
right?" My guess is the former; it's easier, and there's more money to =
be made on it today. In the long run it destroys our world, but we just =
aren't geared towards understanding those kinds of things and would much =
rather just have today's problem fixed =
today.</div><div><br></div><div>--</div><div>Dean</div></body></html>=

--Apple-Mail-13-556657270--

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On Mar 15, 2010, at 12:33 PM, Adam Roach wrote:

> ...

> Which is to say that I don't think there is a solution that  
> satisfies both groups. The divergence from the fork between the  
> stagnant world and the vibrant world is great enough that there is  
> no common ground left for these kinds of issues. As odious as it is,  
> I'm afraid we're stuck with solving this problem twice. The real  
> challenge is ensuring that stagnant solutions don't do lasting  
> damage to the core protocols -- we need to make sure they don't  
> destroy the vibrant world with collateral damage.

Beautifully said! I read it just after my similar but less-elegant  
rant, which presumes that the stagnant world is likely to be selected  
over the vibrant through the simple processes of expediency, apathy,  
corruption, greed, and fear. But then, I'm an optimist.

--
Dean




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I believe this issue was summarized well during the 73rd IETF meeting at th=
e RAI open meeting, and captured in Jonathan's draft: http://tools.ietf.org=
/html/draft-rosenberg-rai-modest-proposal-00

I believe we MUST solve this problem twice.  Let's be realistic.  Understan=
ding the problem fully, defining the requirements, establishing the group a=
nd hammering through the varying solutions will take 5 - 10 years to comple=
te.  The telecommunications industry will develop their own proprietary mec=
hanisms for doing this, creating greater fragmentation and interoperability=
 issues; or the IETF can step up and as an interim solution say, since, you=
 are going to solve this, do it this way.  At least in this case, we will k=
now where the industry is at in their resolution rather than guessing at th=
e plethora of "potentials" down the road.

Regards,

Daryl

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Dean Willis
Sent: Tuesday, March 16, 2010 3:21 PM
To: Adam Roach
Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-0=
0


On Mar 15, 2010, at 12:33 PM, Adam Roach wrote:

> ...

> Which is to say that I don't think there is a solution that satisfies=20
> both groups. The divergence from the fork between the stagnant world=20
> and the vibrant world is great enough that there is no common ground=20
> left for these kinds of issues. As odious as it is, I'm afraid we're=20
> stuck with solving this problem twice. The real challenge is ensuring=20
> that stagnant solutions don't do lasting damage to the core protocols=20
> -- we need to make sure they don't destroy the vibrant world with=20
> collateral damage.

Beautifully said! I read it just after my similar but less-elegant rant, wh=
ich presumes that the stagnant world is likely to be selected over the vibr=
ant through the simple processes of expediency, apathy, corruption, greed, =
and fear. But then, I'm an optimist.

--
Dean



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On 3/16/10 3:59 PM, "Daryl Malas" <D.Malas@cablelabs.com> wrote:

> I believe this issue was summarized well during the 73rd IETF meeting at the
> RAI open meeting, and captured in Jonathan's draft:
> http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00

I think Jonathan Rosenberg correctly identified that there was a problem
here. And then, just like Jonathan Swift's original "Modest Proposal," he
goes on to propose that the solution to our problem is best found in eating
our own children.

I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
appropriate summary of the issues, as it gets the analysis precisely wrong.
It proposes completely abandoning the vibrant world as a lost cause, and
embracing the stagnant world as the only path forward. This isn't just
allowing collateral damage to the vibrant world for the sake of expediency
in stagnant world solutions -- it's a suggestion that we nuke the vibrant
world so it doesn't keep annoying people trying to develop stagnant
solutions.

I'll admit that it's easier to see the stagnant world, since the telcos have
quite a bit of money to throw around driving its deployment. But there are
people in the vibrant world working on new services. Go back and re-read
Greger's description of what he's seeing deployed. Check out the stuff that
AG Projects (http://ag-projects.com/) has been working on, selling, and
actually deploying. I don't recall whether you would have been at the
meeting where I wore a t-shirt showing some novel SIP devices -- I'll bring
it to Anaheim.

/a



From dean.willis@softarmor.com  Tue Mar 16 15:53:37 2010
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On Mar 16, 2010, at 6:24 PM, Adam Roach wrote:
>
> I disagree profoundly with characterizing Rosenberg's Modest  
> Proposal as an
> appropriate summary of the issues, as it gets the analysis precisely  
> wrong.
> It proposes completely abandoning the vibrant world as a lost cause,  
> and
> embracing the stagnant world as the only path forward. This isn't just
> allowing collateral damage to the vibrant world for the sake of  
> expediency
> in stagnant world solutions -- it's a suggestion that we nuke the  
> vibrant
> world so it doesn't keep annoying people trying to develop stagnant
> solutions.
>

I have long speculated that perhaps the best thing we can do is punt  
the "stagnant world" stuff to a different forum. I seem to recall many  
times suggesting that SIP should just be moved to the ITU so we can  
get on with building the Internet. I'd happily add ENUM to the same  
bucket.

The problem is that we might wake up one day and realize that the  
Internet has been turned off, and all we have left is a stagnant  
backwater of telecommunications boundaries established by fat-and- 
happy politicos eager to preserve their own domains.

The alternative, perhaps, is to embrace stagnation. Then add chlorine.

--
Dean
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Mar 16, 2010, =
at 6:24 PM, Adam Roach wrote:</div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" color=3D"#000000"><br></font>I disagree =
profoundly with characterizing Rosenberg's Modest Proposal as =
an<br>appropriate summary of the issues, as it gets the analysis =
precisely wrong.<br>It proposes completely abandoning the vibrant world =
as a lost cause, and<br>embracing the stagnant world as the only path =
forward. This isn't just<br>allowing collateral damage to the vibrant =
world for the sake of expediency<br>in stagnant world solutions -- it's =
a suggestion that we nuke the vibrant<br>world so it doesn't keep =
annoying people trying to develop =
stagnant<br>solutions.<br><br></div></blockquote><div><br></div><div>I =
have long speculated that perhaps the best thing we can do is punt the =
"stagnant world" stuff to a different forum. I seem to recall many times =
suggesting that SIP should just be moved to the ITU so we can get on =
with building the Internet. I'd happily add ENUM to the same =
bucket.</div><div><br></div><div>The problem is that we might wake up =
one day and realize that the Internet has been turned off, and all we =
have left is a stagnant backwater of telecommunications boundaries =
established by fat-and-happy politicos eager to preserve their own =
domains.</div><div><br></div><div>The alternative, perhaps, is to =
embrace stagnation. Then add =
chlorine.</div><div><br></div><div>--</div><div>Dean</div></div></body></h=
tml>=

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On 3/16/10 5:49 PM, Dean Willis wrote:
> I have long speculated that perhaps the best thing we can do is punt 
> the "stagnant world" stuff to a different forum. I seem to recall many 
> times suggesting that SIP should just be moved to the ITU so we can 
> get on with building the Internet. I'd happily add ENUM to the same 
> bucket.
>
> The problem is that we might wake up one day and realize that the 
> Internet has been turned off, and all we have left is a stagnant 
> backwater of telecommunications boundaries established by 
> fat-and-happy politicos eager to preserve their own domains.

Surely you've seen what I'm trying to do here. I've spent a 
disproportionate amount of my IETF time making sure MARTINI doesn't do 
something profoundly stupid to SIP registration (which, as you point 
out, already has some broken baked into it). I'm actually agitating for 
acceptance and publication of Daryl's draft, since it's harmless and 
will hopefully prevent some other SDO from coming up with something 
that's actually dangerous.

I think doing the stagnant world things in the IETF is still the right 
answer, as painful as it is to have to continually deal with solutions 
that willfully destroy the end-to-end principle. Because doing the work 
here at least allows us to steer things into directions that cause the 
least amount of harm.

 From that perspective, Daryl's draft is perfect[1]: it doesn't actually 
create any new protocol at all.

> The alternative, perhaps, is to embrace stagnation. Then add chlorine.

The problem, of course, is not the muck. It's the fact that it's not 
moving forward. Chlorine won't make it go.

/a


[1] Assuming we don't define new DNS record types

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I'm not sure that dispatching malas-dispatch-sip-egress-route requires us t=
o untwist this particular Gordian knot, nor to resurrect TRIP,  nor to decl=
are some kind of interplanetary Armageddon.

I think Otmar Lendl has been right on the mark in this discussion. He point=
s out that global directories like ENUM serve well as an LUF and poorly as =
an LRF, to speak in the SPEERMINT idiom - I don't think this is an especial=
ly controversial point. I read malas-dispatch-sip-egress as a document that=
 argues that knowing the results of the LUF doesn't necessarily help you to=
 find a next hop to route your call through. Again, I'd say not especially =
controversial.

malas-dispatch-sip-egress-route then goes on to specify a few NAPTR syntaxe=
s that could contain a sort of joint LUF/LRF routing information. The only =
conceptual problem I have with this approach is that the entity that typica=
lly provisions the LUF doesn't know the local topology of the originating n=
etwork well enough to say useful things about the LRF - that is indeed the =
entire motivation for considering the LUF and LRF as separate lookups. The =
draft sort of glosses this over with a lot of passive voice when describing=
 how the ENUM provisioning gets done. Is it done by the terminating side, o=
r the originating side, or somehow both?

In any event, from a DISPATCH perspective, I'd propose the problem seems to=
 lie comfortable within the parameters of SPEERMINT. Surely one of the SPEE=
RMINT chairs must think this is a worthwhile effort...

Jon Peterson
NeuStar, Inc.

On 3/16/10 3:24 PM, "Adam Roach" <adam@nostrum.com> wrote:

On 3/16/10 3:59 PM, "Daryl Malas" <D.Malas@cablelabs.com> wrote:

> I believe this issue was summarized well during the 73rd IETF meeting at =
the
> RAI open meeting, and captured in Jonathan's draft:
> http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00

I think Jonathan Rosenberg correctly identified that there was a problem
here. And then, just like Jonathan Swift's original "Modest Proposal," he
goes on to propose that the solution to our problem is best found in eating
our own children.

I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
appropriate summary of the issues, as it gets the analysis precisely wrong.
It proposes completely abandoning the vibrant world as a lost cause, and
embracing the stagnant world as the only path forward. This isn't just
allowing collateral damage to the vibrant world for the sake of expediency
in stagnant world solutions -- it's a suggestion that we nuke the vibrant
world so it doesn't keep annoying people trying to develop stagnant
solutions.

I'll admit that it's easier to see the stagnant world, since the telcos hav=
e
quite a bit of money to throw around driving its deployment. But there are
people in the vibrant world working on new services. Go back and re-read
Greger's description of what he's seeing deployed. Check out the stuff that
AG Projects (http://ag-projects.com/) has been working on, selling, and
actually deploying. I don't recall whether you would have been at the
meeting where I wore a t-shirt showing some novel SIP devices -- I'll bring
it to Anaheim.

/a


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


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</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:=
11pt'><BR>
I&#8217;m not sure that dispatching malas-dispatch-sip-egress-route require=
s us to untwist this particular Gordian knot, nor to resurrect TRIP, &nbsp;=
nor to declare some kind of interplanetary Armageddon.<BR>
<BR>
I think Otmar Lendl has been right on the mark in this discussion. He point=
s out that global directories like ENUM serve well as an LUF and poorly as =
an LRF, to speak in the SPEERMINT idiom - I don&#8217;t think this is an es=
pecially controversial point. I read malas-dispatch-sip-egress as a documen=
t that argues that knowing the results of the LUF doesn&#8217;t necessarily=
 help you to find a next hop to route your call through. Again, I&#8217;d s=
ay not especially controversial.<BR>
<BR>
malas-dispatch-sip-egress-route then goes on to specify a few NAPTR syntaxe=
s that could contain a sort of joint LUF/LRF routing information. The only =
conceptual problem I have with this approach is that the entity that typica=
lly provisions the LUF doesn&#8217;t know the local topology of the origina=
ting network well enough to say useful things about the LRF &#8211; that is=
 indeed the entire motivation for considering the LUF and LRF as separate l=
ookups. The draft sort of glosses this over with a lot of passive voice whe=
n describing how the ENUM provisioning gets done. Is it done by the termina=
ting side, or the originating side, or somehow both?<BR>
<BR>
In any event, from a DISPATCH perspective, I&#8217;d propose the problem se=
ems to lie comfortable within the parameters of SPEERMINT. Surely one of th=
e SPEERMINT chairs must think this is a worthwhile effort...<BR>
<BR>
Jon Peterson<BR>
NeuStar, Inc.<BR>
<BR>
On 3/16/10 3:24 PM, &quot;Adam Roach&quot; &lt;<a href=3D"adam@nostrum.com"=
>adam@nostrum.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:11pt'>On 3/16/10 3:59 PM, &quot;Daryl Malas&quot;=
 &lt;<a href=3D"D.Malas@cablelabs.com">D.Malas@cablelabs.com</a>&gt; wrote:=
<BR>
<BR>
&gt; I believe this issue was summarized well during the 73rd IETF meeting =
at the<BR>
&gt; RAI open meeting, and captured in Jonathan's draft:<BR>
&gt; <a href=3D"http://tools.ietf.org/html/draft-rosenberg-rai-modest-propo=
sal-00">http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00</=
a><BR>
<BR>
I think Jonathan Rosenberg correctly identified that there was a problem<BR=
>
here. And then, just like Jonathan Swift's original &quot;Modest Proposal,&=
quot; he<BR>
goes on to propose that the solution to our problem is best found in eating=
<BR>
our own children.<BR>
<BR>
I disagree profoundly with characterizing Rosenberg's Modest Proposal as an=
<BR>
appropriate summary of the issues, as it gets the analysis precisely wrong.=
<BR>
It proposes completely abandoning the vibrant world as a lost cause, and<BR=
>
embracing the stagnant world as the only path forward. This isn't just<BR>
allowing collateral damage to the vibrant world for the sake of expediency<=
BR>
in stagnant world solutions -- it's a suggestion that we nuke the vibrant<B=
R>
world so it doesn't keep annoying people trying to develop stagnant<BR>
solutions.<BR>
<BR>
I'll admit that it's easier to see the stagnant world, since the telcos hav=
e<BR>
quite a bit of money to throw around driving its deployment. But there are<=
BR>
people in the vibrant world working on new services. Go back and re-read<BR=
>
Greger's description of what he's seeing deployed. Check out the stuff that=
<BR>
AG Projects (<a href=3D"http://ag-projects.com/">http://ag-projects.com/</a=
>) has been working on, selling, and<BR>
actually deploying. I don't recall whether you would have been at the<BR>
meeting where I wore a t-shirt showing some novel SIP devices -- I'll bring=
<BR>
it to Anaheim.<BR>
<BR>
/a<BR>
<BR>
<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--_000_C7C562683B15Cjonpetersonneustarbiz_--

From D.Malas@cablelabs.com  Tue Mar 16 16:49:55 2010
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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Adam Roach'" <adam@nostrum.com>, Dean Willis <dean.willis@softarmor.com>
Date: Tue, 16 Mar 2010 17:50:02 -0600
Thread-Topic: War of the Worlds (was Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00)
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Adam and Dean,

Perhaps you have misunderstood my comments.  My comment is, simply, it is b=
etter to know and control the "ugly" truths rather than simply sweep them u=
nder the rug and blindly believe they do not exist.  In this way, you know =
what evil you will need to face, rather than being surprised by the unknown=
 proprietary solutions vendors have created based on monetary rather than t=
echnology gains.

Regards,

Daryl

-----Original Message-----
From: Adam Roach [mailto:adam@nostrum.com]=20
Sent: Tuesday, March 16, 2010 5:01 PM
To: Dean Willis
Cc: Daryl Malas; dispatch@ietf.org
Subject: Re: War of the Worlds (was Re: [dispatch] I-D Action: draft-malas-=
dispatch-sip-egress-route-00)

On 3/16/10 5:49 PM, Dean Willis wrote:
> I have long speculated that perhaps the best thing we can do is punt=20
> the "stagnant world" stuff to a different forum. I seem to recall many=20
> times suggesting that SIP should just be moved to the ITU so we can=20
> get on with building the Internet. I'd happily add ENUM to the same=20
> bucket.
>
> The problem is that we might wake up one day and realize that the=20
> Internet has been turned off, and all we have left is a stagnant=20
> backwater of telecommunications boundaries established by=20
> fat-and-happy politicos eager to preserve their own domains.

Surely you've seen what I'm trying to do here. I've spent a disproportionat=
e amount of my IETF time making sure MARTINI doesn't do something profoundl=
y stupid to SIP registration (which, as you point out, already has some bro=
ken baked into it). I'm actually agitating for acceptance and publication o=
f Daryl's draft, since it's harmless and will hopefully prevent some other =
SDO from coming up with something that's actually dangerous.

I think doing the stagnant world things in the IETF is still the right answ=
er, as painful as it is to have to continually deal with solutions that wil=
lfully destroy the end-to-end principle. Because doing the work here at lea=
st allows us to steer things into directions that cause the least amount of=
 harm.

 From that perspective, Daryl's draft is perfect[1]: it doesn't actually cr=
eate any new protocol at all.

> The alternative, perhaps, is to embrace stagnation. Then add chlorine.

The problem, of course, is not the muck. It's the fact that it's not moving=
 forward. Chlorine won't make it go.

/a


[1] Assuming we don't define new DNS record types

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From: Daryl Malas <D.Malas@cablelabs.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>, Adam Roach <adam@nostrum.com>, Dean Willis <dean.willis@softarmor.com>
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--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F6777srvxchg_
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Jon,

The provisioning of the "egress route" information is done by the originati=
ng side.  The terminating side provisions their ingress routing information=
 into an ENUM server, and the originating party evalutes the routes and ass=
ociates egress routes to them.  In this way, they append the information pr=
ovided by the terminating side (provisioning side).  If the terminating sid=
e changes the route (domain's) associated with the TNs, which an egress rou=
te was appended, then the egress route information is no longer valid and i=
s removed.  The originating side would have to append a new egress route.  =
Yes, this may seem inefficient, but there are two industry assumptions: 1) =
routes do not change very often; 2) the efficiency gained by associating an=
 egress route is greater than the inefficiency in updating egress routes.

This draft does not consider the "business" reasons for doing this, but sim=
ply says...since, we know you are already doing this, then we should make t=
he method consistent using (a) well defined mechanism(s) approved by a body=
 of experts.

I do not think this goes against what Otmar was asserting in his Speermint =
draft.  I believe Otmar was trying to address a bigger picture issue, which=
 was not ignored in Speermint.  On the contrary, we decided early to addres=
s the short-term issue of basic telephony peering using SIP, and then move =
on to the bigger more complex issue.  It just took us longer to get there t=
han we expected.  We are still listening.

Regards,

Daryl

________________________________
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: Tuesday, March 16, 2010 5:30 PM
To: Adam Roach; Daryl Malas; Dean Willis
Cc: dispatch@ietf.org
Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-=
dispatch-sip-egress-route-00)


I'm not sure that dispatching malas-dispatch-sip-egress-route requires us t=
o untwist this particular Gordian knot, nor to resurrect TRIP,  nor to decl=
are some kind of interplanetary Armageddon.

I think Otmar Lendl has been right on the mark in this discussion. He point=
s out that global directories like ENUM serve well as an LUF and poorly as =
an LRF, to speak in the SPEERMINT idiom - I don't think this is an especial=
ly controversial point. I read malas-dispatch-sip-egress as a document that=
 argues that knowing the results of the LUF doesn't necessarily help you to=
 find a next hop to route your call through. Again, I'd say not especially =
controversial.

malas-dispatch-sip-egress-route then goes on to specify a few NAPTR syntaxe=
s that could contain a sort of joint LUF/LRF routing information. The only =
conceptual problem I have with this approach is that the entity that typica=
lly provisions the LUF doesn't know the local topology of the originating n=
etwork well enough to say useful things about the LRF - that is indeed the =
entire motivation for considering the LUF and LRF as separate lookups. The =
draft sort of glosses this over with a lot of passive voice when describing=
 how the ENUM provisioning gets done. Is it done by the terminating side, o=
r the originating side, or somehow both?

In any event, from a DISPATCH perspective, I'd propose the problem seems to=
 lie comfortable within the parameters of SPEERMINT. Surely one of the SPEE=
RMINT chairs must think this is a worthwhile effort...

Jon Peterson
NeuStar, Inc.

On 3/16/10 3:24 PM, "Adam Roach" <adam@nostrum.com> wrote:

On 3/16/10 3:59 PM, "Daryl Malas" <D.Malas@cablelabs.com> wrote:

> I believe this issue was summarized well during the 73rd IETF meeting at =
the
> RAI open meeting, and captured in Jonathan's draft:
> http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00

I think Jonathan Rosenberg correctly identified that there was a problem
here. And then, just like Jonathan Swift's original "Modest Proposal," he
goes on to propose that the solution to our problem is best found in eating
our own children.

I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
appropriate summary of the issues, as it gets the analysis precisely wrong.
It proposes completely abandoning the vibrant world as a lost cause, and
embracing the stagnant world as the only path forward. This isn't just
allowing collateral damage to the vibrant world for the sake of expediency
in stagnant world solutions -- it's a suggestion that we nuke the vibrant
world so it doesn't keep annoying people trying to develop stagnant
solutions.

I'll admit that it's easier to see the stagnant world, since the telcos hav=
e
quite a bit of money to throw around driving its deployment. But there are
people in the vibrant world working on new services. Go back and re-read
Greger's description of what he's seeing deployed. Check out the stuff that
AG Projects (http://ag-projects.com/) has been working on, selling, and
actually deploying. I don't recall whether you would have been at the
meeting where I wore a t-shirt showing some novel SIP devices -- I'll bring
it to Anaheim.

/a


_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch


--_000_114DAD31379DFA438C0A2E39B3B8AF5D01213F6777srvxchg_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Re: [dispatch] War of the Worlds (was Re: I-D Action: dr=
aft-malas-dispatch-sip-egress-route-00)</TITLE>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18876"></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Jon,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The provisioning of the "egress route" information is=
 done by=20
the originating side.&nbsp; The terminating side provisions their ingress=20
routing information into an ENUM server, and the originating party evalutes=
 the=20
routes and associates egress routes to them.&nbsp; In this way, they append=
 the=20
information provided by the terminating side (provisioning side).&nbsp; If =
the=20
terminating side changes the route (domain's) associated with the TNs, whic=
h an=20
egress route was appended, then the egress route information is no longer v=
alid=20
and is removed.&nbsp; The originating side would have to append a new egres=
s=20
route.&nbsp; Yes, this may seem inefficient, but there are two industry=20
assumptions: 1) routes do not change very often; 2) the efficiency gained b=
y=20
associating an egress route is greater than the inefficiency in updating eg=
ress=20
routes.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>This draft does not consider the "business" reasons f=
or doing=20
this, but simply says...since, we know you are already doing this, then we=
=20
should make the method consistent using (a) well&nbsp;defined mechanism(s)=
=20
approved by a body of experts.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>I do not think this goes against what Otmar was asser=
ting in=20
his Speermint draft.&nbsp; I believe Otmar was trying to address a bigger=20
picture issue, which was not ignored in Speermint.&nbsp; On the contrary, w=
e=20
decided early to address the short-term issue of basic telephony peering us=
ing=20
SIP, and then move on to the bigger more complex issue.&nbsp; It just took =
us=20
longer to get there than we expected.&nbsp; We are still=20
listening.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010></SPAN><SPAN=20
class=3D425095023-16032010><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010></SPAN>&nbsp;<=
/DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D425095023-16032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Daryl</FONT>&nbsp;</SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> Peterson, Jon=20
[mailto:jon.peterson@neustar.biz] <BR><B>Sent:</B> Tuesday, March 16, 2010 =
5:30=20
PM<BR><B>To:</B> Adam Roach; Daryl Malas; Dean Willis<BR><B>Cc:</B>=20
dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] War of the Worlds (was =
Re:=20
I-D Action: draft-malas-dispatch-sip-egress-route-00)<BR></FONT><BR></DIV>
<DIV></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
style=3D"FONT-SIZE: 11pt"><BR>I&#8217;m not sure that dispatching=20
malas-dispatch-sip-egress-route requires us to untwist this particular Gord=
ian=20
knot, nor to resurrect TRIP, &nbsp;nor to declare some kind of interplaneta=
ry=20
Armageddon.<BR><BR>I think Otmar Lendl has been right on the mark in this=20
discussion. He points out that global directories like ENUM serve well as a=
n LUF=20
and poorly as an LRF, to speak in the SPEERMINT idiom - I don&#8217;t think=
 this is an=20
especially controversial point. I read malas-dispatch-sip-egress as a docum=
ent=20
that argues that knowing the results of the LUF doesn&#8217;t necessarily h=
elp you to=20
find a next hop to route your call through. Again, I&#8217;d say not especi=
ally=20
controversial.<BR><BR>malas-dispatch-sip-egress-route then goes on to speci=
fy a=20
few NAPTR syntaxes that could contain a sort of joint LUF/LRF routing=20
information. The only conceptual problem I have with this approach is that =
the=20
entity that typically provisions the LUF doesn&#8217;t know the local topol=
ogy of the=20
originating network well enough to say useful things about the LRF &#8211; =
that is=20
indeed the entire motivation for considering the LUF and LRF as separate=20
lookups. The draft sort of glosses this over with a lot of passive voice wh=
en=20
describing how the ENUM provisioning gets done. Is it done by the terminati=
ng=20
side, or the originating side, or somehow both?<BR><BR>In any event, from a=
=20
DISPATCH perspective, I&#8217;d propose the problem seems to lie comfortabl=
e within=20
the parameters of SPEERMINT. Surely one of the SPEERMINT chairs must think =
this=20
is a worthwhile effort...<BR><BR>Jon Peterson<BR>NeuStar, Inc.<BR><BR>On 3/=
16/10=20
3:24 PM, "Adam Roach" &lt;<A href=3D"adam@nostrum.com">adam@nostrum.com</A>=
&gt;=20
wrote:<BR><BR></SPAN></FONT>
<BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 11pt">On 3/16/10 3:59 PM, "Daryl Malas" &lt;<A=20
  href=3D"D.Malas@cablelabs.com">D.Malas@cablelabs.com</A>&gt; wrote:<BR><B=
R>&gt;=20
  I believe this issue was summarized well during the 73rd IETF meeting at=
=20
  the<BR>&gt; RAI open meeting, and captured in Jonathan's draft:<BR>&gt; <=
A=20
  href=3D"http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00=
">http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00</A><BR>=
<BR>I=20
  think Jonathan Rosenberg correctly identified that there was a=20
  problem<BR>here. And then, just like Jonathan Swift's original "Modest=20
  Proposal," he<BR>goes on to propose that the solution to our problem is b=
est=20
  found in eating<BR>our own children.<BR><BR>I disagree profoundly with=20
  characterizing Rosenberg's Modest Proposal as an<BR>appropriate summary o=
f the=20
  issues, as it gets the analysis precisely wrong.<BR>It proposes completel=
y=20
  abandoning the vibrant world as a lost cause, and<BR>embracing the stagna=
nt=20
  world as the only path forward. This isn't just<BR>allowing collateral da=
mage=20
  to the vibrant world for the sake of expediency<BR>in stagnant world solu=
tions=20
  -- it's a suggestion that we nuke the vibrant<BR>world so it doesn't keep=
=20
  annoying people trying to develop stagnant<BR>solutions.<BR><BR>I'll admi=
t=20
  that it's easier to see the stagnant world, since the telcos have<BR>quit=
e a=20
  bit of money to throw around driving its deployment. But there are<BR>peo=
ple=20
  in the vibrant world working on new services. Go back and re-read<BR>Greg=
er's=20
  description of what he's seeing deployed. Check out the stuff that<BR>AG=
=20
  Projects (<A href=3D"http://ag-projects.com/">http://ag-projects.com/</A>=
) has=20
  been working on, selling, and<BR>actually deploying. I don't recall wheth=
er=20
  you would have been at the<BR>meeting where I wore a t-shirt showing some=
=20
  novel SIP devices -- I'll bring<BR>it to=20
  Anaheim.<BR><BR>/a<BR><BR><BR>___________________________________________=
____<BR>dispatch=20
  mailing list<BR><A href=3D"dispatch@ietf.org">dispatch@ietf.org</A><BR><A=
=20
  href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.=
org/mailman/listinfo/dispatch</A><BR><BR></SPAN></FONT></BLOCKQUOTE></BODY>=
</HTML>

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From adam@nostrum.com  Tue Mar 16 17:47:21 2010
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Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)
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On 3/16/10 18:50, Mar 16, Daryl Malas wrote:
> Adam and Dean,
>
> Perhaps you have misunderstood my comments.  My comment is, simply, it is better to know and control the "ugly" truths rather than simply sweep them under the rug and blindly believe they do not exist.  In this way, you know what evil you will need to face, rather than being surprised by the unknown proprietary solutions vendors have created based on monetary rather than technology gains.
>    

That's exactly what I say below.

/a

> Regards,
>
> Daryl
>
> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Tuesday, March 16, 2010 5:01 PM
> To: Dean Willis
> Cc: Daryl Malas; dispatch@ietf.org
> Subject: Re: War of the Worlds (was Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00)
>
> On 3/16/10 5:49 PM, Dean Willis wrote:
>    
>> I have long speculated that perhaps the best thing we can do is punt
>> the "stagnant world" stuff to a different forum. I seem to recall many
>> times suggesting that SIP should just be moved to the ITU so we can
>> get on with building the Internet. I'd happily add ENUM to the same
>> bucket.
>>
>> The problem is that we might wake up one day and realize that the
>> Internet has been turned off, and all we have left is a stagnant
>> backwater of telecommunications boundaries established by
>> fat-and-happy politicos eager to preserve their own domains.
>>      
> Surely you've seen what I'm trying to do here. I've spent a disproportionate amount of my IETF time making sure MARTINI doesn't do something profoundly stupid to SIP registration (which, as you point out, already has some broken baked into it). I'm actually agitating for acceptance and publication of Daryl's draft, since it's harmless and will hopefully prevent some other SDO from coming up with something that's actually dangerous.
>
> I think doing the stagnant world things in the IETF is still the right answer, as painful as it is to have to continually deal with solutions that willfully destroy the end-to-end principle. Because doing the work here at least allows us to steer things into directions that cause the least amount of harm.
>
>   From that perspective, Daryl's draft is perfect[1]: it doesn't actually create any new protocol at all.
>
>    
>> The alternative, perhaps, is to embrace stagnation. Then add chlorine.
>>      
> The problem, of course, is not the muck. It's the fact that it's not moving forward. Chlorine won't make it go.
>
> /a
>
>
> [1] Assuming we don't define new DNS record types
>    


From henry.sinnreich@gmail.com  Tue Mar 16 21:19:57 2010
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Date: Tue, 16 Mar 2010 23:19:27 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Dean Willis <dean.willis@softarmor.com>, Adam Roach <adam@nostrum.com>
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Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)
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This really sums up the enigma why the IETF should do work that contradicts
every principle of the Internet architecture and its engineering.

It may be useful to remember the  _solid_   track record of the  "stagnant
world"
* OSI 
* X4.00 mail 
* IN 
* AIN 
* ISDN 
* SONET/SDH 
* ATM 
* BISDN 
* H.323 
* current re-incarnations such as PSTN/SIP and other (avoiding flames
here)...
How many companies have disappeared building this stuff? Layoffs. Capital
destruction.

The other enigma is the tolerance to associate the IETF with the
reincarnations of these failures.

>I have long speculated that perhaps the best thing we can do is punt the
"stagnant world" stuff to a different forum.
>I seem to recall many times suggesting that SIP should just be moved to the ITU
so we can get on with building the Internet.
>I'd happily add ENUM to the same bucket.

Well said.

Thanks, Henry


On 3/16/10 5:49 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> 
> On Mar 16, 2010, at 6:24 PM, Adam Roach wrote:
>> 
>> I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
>> appropriate summary of the issues, as it gets the analysis precisely wrong.
>> It proposes completely abandoning the vibrant world as a lost cause, and
>> embracing the stagnant world as the only path forward. This isn't just
>> allowing collateral damage to the vibrant world for the sake of expediency
>> in stagnant world solutions -- it's a suggestion that we nuke the vibrant
>> world so it doesn't keep annoying people trying to develop stagnant
>> solutions.
>> 
> 
> I have long speculated that perhaps the best thing we can do is punt the
> "stagnant world" stuff to a different forum. I seem to recall many times
> suggesting that SIP should just be moved to the ITU so we can get on with
> building the Internet. I'd happily add ENUM to the same bucket.
> 
> The problem is that we might wake up one day and realize that the Internet has
> been turned off, and all we have left is a stagnant backwater of
> telecommunications boundaries established by fat-and-happy politicos eager to
> preserve their own domains.
> 
> The alternative, perhaps, is to embrace stagnation. Then add chlorine.
> 
> --
> Dean
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


--B_3351626370_950792
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-di=
spatch-sip-egress-route-00)</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>This really sums up the enigma why the IETF should do work that contradict=
s every principle of the Internet architecture and its engineering.<BR>
<BR>
It may be useful to remember the &nbsp;_solid_ &nbsp;&nbsp;track record of =
the &nbsp;&quot;stagnant world&quot;<BR>
</SPAN></FONT><UL><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=
 STYLE=3D'font-size:13pt'>OSI
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>X4.00 mail
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>IN
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>AIN
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>ISDN
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>SONET/SDH
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>ATM
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>BISDN
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>H.323
</SPAN></FONT><LI><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STY=
LE=3D'font-size:13pt'>current re-incarnations such as PSTN/SIP and other (avoi=
ding flames here)...<BR>
</SPAN></FONT></UL><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN ST=
YLE=3D'font-size:13pt'>How many companies have disappeared building this stuff=
? Layoffs. Capital destruction.<BR>
<BR>
The other enigma is the tolerance to associate the IETF with the reincarnat=
ions of these failures.<BR>
<BR>
&gt;I have long speculated that perhaps the best thing we can do is punt th=
e &quot;stagnant world&quot; stuff to a different forum. <BR>
&gt;I seem to recall many times suggesting that SIP should just be moved to=
 the ITU so we can get on with building the Internet. <BR>
&gt;I'd happily add ENUM to the same bucket.<BR>
<BR>
Well said.<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 3/16/10 5:49 PM, &quot;Dean Willis&quot; &lt;<a href=3D"dean.willis@softar=
mor.com">dean.willis@softarmor.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'><BR>
On Mar 16, 2010, at 6:24 PM, Adam Roach wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'><BR>
I disagree profoundly with characterizing Rosenberg's Modest Proposal as an=
<BR>
appropriate summary of the issues, as it gets the analysis precisely wrong.=
<BR>
It proposes completely abandoning the vibrant world as a lost cause, and<BR=
>
embracing the stagnant world as the only path forward. This isn't just<BR>
allowing collateral damage to the vibrant world for the sake of expediency<=
BR>
in stagnant world solutions -- it's a suggestion that we nuke the vibrant<B=
R>
world so it doesn't keep annoying people trying to develop stagnant<BR>
solutions.<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'><BR>
I have long speculated that perhaps the best thing we can do is punt the &q=
uot;stagnant world&quot; stuff to a different forum. I seem to recall many t=
imes suggesting that SIP should just be moved to the ITU so we can get on wi=
th building the Internet. I'd happily add ENUM to the same bucket.<BR>
<BR>
The problem is that we might wake up one day and realize that the Internet =
has been turned off, and all we have left is a stagnant backwater of telecom=
munications boundaries established by fat-and-happy politicos eager to prese=
rve their own domains.<BR>
<BR>
The alternative, perhaps, is to embrace stagnation. Then add chlorine.<BR>
<BR>
--<BR>
Dean<BR>
<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"95%"></SPAN></FONT><SPAN STYLE=3D'font-size:=
13pt'><FONT FACE=3D"Consolas, Courier New, Courier">__________________________=
_____________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.o=
rg/mailman/listinfo/dispatch</a><BR>
</FONT></SPAN></BLOCKQUOTE>
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From dean.willis@softarmor.com  Tue Mar 16 22:34:03 2010
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On Mar 16, 2010, at 6:50 PM, Daryl Malas wrote:

> Adam and Dean,
>
> Perhaps you have misunderstood my comments.  My comment is, simply,  
> it is better to know and control the "ugly" truths rather than  
> simply sweep them under the rug and blindly believe they do not  
> exist.  In this way, you know what evil you will need to face,  
> rather than being surprised by the unknown proprietary solutions  
> vendors have created based on monetary rather than technology gains.


You are right. It is indeed better to know about the ugly than to  
ignore it. I have perhaps forecast its development all to clearly and  
proven unable to do anything about it, which tends to make one  
crotchety at best.

And I'd rather have a real routing protocol to handle the "LRF evil"  
within an administrative domain than hack the DNS to convey the same  
information. It is, I think, the lesser of two evils.

--
Dean


From dean.willis@softarmor.com  Tue Mar 16 22:37:21 2010
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On Mar 16, 2010, at 11:19 PM, Henry Sinnreich wrote:

> This really sums up the enigma why the IETF should do work that  
> contradicts every principle of the Internet architecture and its  
> engineering.
>
> It may be useful to remember the  _solid_   track record of the   
> "stagnant world"

Not to mention that I still can't reliably make a SIP call. Bernie and  
I had to hack on things together for thirty minutes before we could  
make a call work to talk about E2MD earlier this week. But at least we  
didn't break down and use Skype... until today, when we needed to talk  
in a hurry.

--
Dean
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<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>On Mar 16, 2010, at 11:19 PM, Henry Sinnreich wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div> <font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size:13pt">This really sums up the enigma why the IETF should do work that contradicts every principle of the Internet architecture and its engineering.<br> <br> It may be useful to remember the &nbsp;_solid_ &nbsp;&nbsp;track record of the &nbsp;"stagnant world"<br> </span></font></div></blockquote></div><br><div>Not to mention that I still can't reliably make a SIP call. Bernie and I had to hack on things together for thirty minutes before we could make a call work to talk about E2MD earlier this week. But at least we didn't break down and use Skype... until today, when we needed to talk in a hurry.</div><div><br></div><div>--</div><div>Dean</div></body></html>
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From Milan.Patel@InterDigital.com  Wed Mar 17 02:04:59 2010
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Thread-Topic: [dispatch] FW: New VersionNotification	for	draft-patel-dispatch-cpc-oli-parameter-00
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From: "Patel, Milan" <Milan.Patel@InterDigital.com>
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Hi Paul,

Thanks for pointing out the NIT. I will address that in the next
revision.=20

Your other comments seem to be specific to the semantics of certain OLI
values. That is out of scope of both the internet draft and IETF, since
they were defined elsewhere (ITU-T).=20

After receiving Jon's comments, I had extensive discussions with
operators and vendors about the use of CPC and OLI currently in the
field. The current version of the draft reflects such usage and the
syntax of the CPC and OLI as defined in version -02 of the draft has
been scrutinized by operators and vendors attending 3GPP and has
receiving unanimous support.=20

The draft defines the CPC and OLI as tel URI parameters. For usage with
SIP URIs, further syntax would need to be defined extending the SIP URI
parameter as defined in RFC 3261. Currently, we see no requirements to
include CPC or OLI for SIP URIs. If this is needed in the future, a
further Internet-Draft can be proposed.=20

Regards,
Milan

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Paul Kyzivat
Sent: Wednesday, March 10, 2010 2:59 PM
To: Jon Peterson
Cc: Milan Patel; DISPATCH
Subject: Re: [dispatch] FW: New VersionNotification for
draft-patel-dispatch-cpc-oli-parameter-00

I just got around to reading this draft.
I have all of the concerns that Jon does, plus some others:

Many of these properties are in fact properties of the *call*, not=20
properties of the *caller*. That is especially true of the oli values:

- some are a function of the called number rather than the caller:
   * 30 Intercept (blank) - for calls to unassigned directory number
(DN)
   * 31 Intercept (trouble) - for calls to directory numbers (DN) that
        have been manually placed in trouble-busy state by Telco
        personnel
   * 32 Intercept (regular) - for calls to recently changed or
        disconnected numbers

- the value can be assigned mid-call, as a result of call handling:
   * 34 Telco Operator Handled Call - after the Telco Operator Services
        System has handled a call for an IC, it may change the standard
        ANI digits to "34", before outpulsing the sequence to the IC,
        when the Telco performs all call handling functions, e.g.,
        billing. The code tells the IC that the BOC has performed
billing
        on the call and the IC only has to complete the call.
   * 52 Outward Wide Area Telecommunications Service (OUTWATS) - this
        service allows customers to make calls to a certain zone(s) or
        band(s) on a direct dialed basis for a flat monthly charge or
for
        a charge based on accumulated usage. OUTWATS lines can dial
        station-to-station calls directly to points within the selected
        band(s) or zone(s). The LEC performs a screening function to
        determine the correct charging and routing for OUTWATS calls
        based on the customer's class of service and the service area of
        the call party. When these calls are routed to the interexchange
        carrier via a combined WATS-POTS trunk group, it is necessary to
        identify the WATS calls with the ANI code "52".

- (this is not an exhaustive list.)

Putting the value in From is problematic if it is a function of=20
something other than the caller. Its even more problematic if it can=20
change mid-call, since changing it breaks Identity.

Even when the information is a property of the caller, it seems likely=20
that it will often be added by something other than the UAC, since the=20
UAC is unlikely to be trusted to provide it.

If this is information that is a property of the call rather than=20
caller, and it can change along the signaling path, then IMO it belongs=20
in some other header, in a field that may be changed along the path.

Also a NIT:

    The "oli" parameter with value 29 indicates that the device that the
    call is initiated from is located within a prison.  An "oli" with
    value "prison" is equally valid.

The ABNF doesn't permit a value of "prison" for the oli parameter - only

digits.


Jon Peterson wrote:
>=20
> In evaluating proposals to add capabilities to SIP via a tel URI, the=20
> first question we ask is, "Would this be useful in SIP requests that
do=20
> NOT use telephone numbers?" If the answer is "yes," then probably the=20
> capability should be added to SIP in a way that works with and without

> the tel URI. (The draft is very slightly unclear  about whether or not

> you can include this parameter in SIP URIs that do not contain
embedded=20
> tel URIs but do contain non-tel-URI representations of telephone
numbers=20
> - but I'm speaking here to the overall intention to couple the use of=20
> OLI/CPC information to telephone numbers exclusively.)
>=20
> While the draft focuses on the use case where SIP is gatewaying to
ISUP,=20
> and thus where telephone numbers are likely to be used, I would be=20
> hesitant to rule out uses that do not involve telephone numbers -=20
> especially when the motivating use case given is REGISTER in an=20
> emergency context. Encapsulating the opaque numeric values of OLI into
a=20
> SIP parameter also more or less restricts the use of this to entities=20
> that understand ISUP. This begs the obvious question: should there be
a=20
> way in SIP (outside of gatewaying)  to express these same concepts?=20
> Since you already translate the CPC values into human-readable=20
> equivalents ("operator", "payphone", "test"), is there some reason why

> OLI can't receive the same treatment? The only example of the
semantics=20
> of OLI that you give ("oli=3D29" meaning "prison") indeed seems it =
could

> admit of that translation. If there is some good reason why CPC should

> be translated and OLI should be encapsulated, the draft should=20
> explicitly motivate it.
>=20
> By the by, since the Acks mention that I submitted a version of this=20
> proposal prior to Rohan's, just as an historical aside I in fact split

> that out from the original draft-ietf-sip-privacy-04 by Flemming and=20
> Bill Marshall, which for some reason had it tacked into an appendix.
The=20
> parameter proposed in that document (which can still be found via=20
> Google), the Nature of Party ("np") parameter, maps to a set of=20
> human-readable literals like "ordinary", "police",  "payphone",=20
> "emergency", and so on. The document then proposes a mapping to the
ANI=20
> II digits; one could just as easily be provided for OLI or CPC in the=20
> ISUP variants of interest. The example given in that appendix shows
this=20
> being used in a URI that doesn't contain a telephone number. In any=20
> event, if you want to have a pedigree of the idea in your Acks, you=20
> shouldn't leave out that document.
>=20
> Jon Peterson
> NeuStar, Inc.
>=20
> Milan Patel wrote:
>> Hi,
>>
>> A new version of the Internet Draft for the Calling Party's Category
and
>> Originating Line Identity URI Parameters is now available.=20
>> http://www.ietf.org/id/draft-patel-dispatch-cpc-oli-parameter-00.txt
>>
>> This is based on Rohan Mahy's draft-mahy-iptel-cpc-06
>>
>> Best regards,
>> Milan
>>
>>
>> -----Original Message-----
>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] Sent:
08=20
>> October 2009 18:40
>> To: Patel, Milan (MOP:EP10)
>> Cc: r.jesske@telekom.de; mdolly@att.com
>> Subject: New Version Notification for
>> draft-patel-dispatch-cpc-oli-parameter-00
>>
>> A new version of I-D, draft-patel-dispatch-cpc-oli-parameter-00.txt
has
>> been successfuly submitted by Milan Patel and posted to the IETF
>> repository.
>>
>> Filename:     draft-patel-dispatch-cpc-oli-parameter
>> Revision:     00
>> Title:         Uniform Resource Identifier (URI) Parameters for
>> indicating the Calling Party's Catagory and Originating Line Identity
>> Creation_date:     2009-10-08
>> WG ID:         Independent Submission
>> Number_of_pages: 10
>>
>> Abstract:
>> This document defines two new URI parameters to describe the calling
>> party's category and toll class of service originating line
>> information which are parameters also used in SS7 ISUP and other
>> telephony signalling protocols.  The intended use of these URI
>> parameters is for the "tel" URI or equivalent SIP URI representation.
>> =20
>>
>>
>>
>> The IETF Secretariat.
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>  =20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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Patel, Milan wrote:
> Hi Paul,
> 
> Thanks for pointing out the NIT. I will address that in the next
> revision. 
> 
> Your other comments seem to be specific to the semantics of certain OLI
> values. That is out of scope of both the internet draft and IETF, since
> they were defined elsewhere (ITU-T). 

You miss my point. I am not suggesting there is anything wrong with the 
semantics of the values. My point is that values with these semantics do 
not belong in a URI that is being used to describe the originator of a 
call, or in the URI used to describe the destination of the call. They 
are properties of the *call*.

So, these really don't belong in a URI at all. They belong in some other 
header or headers.

> After receiving Jon's comments, I had extensive discussions with
> operators and vendors about the use of CPC and OLI currently in the
> field. The current version of the draft reflects such usage and the
> syntax of the CPC and OLI as defined in version -02 of the draft has
> been scrutinized by operators and vendors attending 3GPP and has
> receiving unanimous support. 
> 
> The draft defines the CPC and OLI as tel URI parameters. For usage with
> SIP URIs, further syntax would need to be defined extending the SIP URI
> parameter as defined in RFC 3261. Currently, we see no requirements to
> include CPC or OLI for SIP URIs. If this is needed in the future, a
> further Internet-Draft can be proposed. 

For those that truly describe properties of a URI (if there are any of 
those), you may be right.

For those that describe properties of a call, if they were carried as 
part of the call rather than as properties of the URI, then it wouldn't 
matter if the URI is sip or tel.

But in reality, even the properties that describe the caller aren't 
properly properties of the caller's URI. For instance, in principle the 
same originating "number" might be used by one phone in a prison and 
another in a police station.

ISTM that piggybacking these on the TEL uri is simply a convenient 
"ploy", because both phone numbers and these properties are managed by 
the ITU.

	Thanks,
	Paul

> Regards,
> Milan
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Wednesday, March 10, 2010 2:59 PM
> To: Jon Peterson
> Cc: Milan Patel; DISPATCH
> Subject: Re: [dispatch] FW: New VersionNotification for
> draft-patel-dispatch-cpc-oli-parameter-00
> 
> I just got around to reading this draft.
> I have all of the concerns that Jon does, plus some others:
> 
> Many of these properties are in fact properties of the *call*, not 
> properties of the *caller*. That is especially true of the oli values:
> 
> - some are a function of the called number rather than the caller:
>    * 30 Intercept (blank) - for calls to unassigned directory number
> (DN)
>    * 31 Intercept (trouble) - for calls to directory numbers (DN) that
>         have been manually placed in trouble-busy state by Telco
>         personnel
>    * 32 Intercept (regular) - for calls to recently changed or
>         disconnected numbers
> 
> - the value can be assigned mid-call, as a result of call handling:
>    * 34 Telco Operator Handled Call - after the Telco Operator Services
>         System has handled a call for an IC, it may change the standard
>         ANI digits to "34", before outpulsing the sequence to the IC,
>         when the Telco performs all call handling functions, e.g.,
>         billing. The code tells the IC that the BOC has performed
> billing
>         on the call and the IC only has to complete the call.
>    * 52 Outward Wide Area Telecommunications Service (OUTWATS) - this
>         service allows customers to make calls to a certain zone(s) or
>         band(s) on a direct dialed basis for a flat monthly charge or
> for
>         a charge based on accumulated usage. OUTWATS lines can dial
>         station-to-station calls directly to points within the selected
>         band(s) or zone(s). The LEC performs a screening function to
>         determine the correct charging and routing for OUTWATS calls
>         based on the customer's class of service and the service area of
>         the call party. When these calls are routed to the interexchange
>         carrier via a combined WATS-POTS trunk group, it is necessary to
>         identify the WATS calls with the ANI code "52".
> 
> - (this is not an exhaustive list.)
> 
> Putting the value in From is problematic if it is a function of 
> something other than the caller. Its even more problematic if it can 
> change mid-call, since changing it breaks Identity.
> 
> Even when the information is a property of the caller, it seems likely 
> that it will often be added by something other than the UAC, since the 
> UAC is unlikely to be trusted to provide it.
> 
> If this is information that is a property of the call rather than 
> caller, and it can change along the signaling path, then IMO it belongs 
> in some other header, in a field that may be changed along the path.
> 
> Also a NIT:
> 
>     The "oli" parameter with value 29 indicates that the device that the
>     call is initiated from is located within a prison.  An "oli" with
>     value "prison" is equally valid.
> 
> The ABNF doesn't permit a value of "prison" for the oli parameter - only
> 
> digits.
> 
> 
> Jon Peterson wrote:
>> In evaluating proposals to add capabilities to SIP via a tel URI, the 
>> first question we ask is, "Would this be useful in SIP requests that
> do 
>> NOT use telephone numbers?" If the answer is "yes," then probably the 
>> capability should be added to SIP in a way that works with and without
> 
>> the tel URI. (The draft is very slightly unclear  about whether or not
> 
>> you can include this parameter in SIP URIs that do not contain
> embedded 
>> tel URIs but do contain non-tel-URI representations of telephone
> numbers 
>> - but I'm speaking here to the overall intention to couple the use of 
>> OLI/CPC information to telephone numbers exclusively.)
>>
>> While the draft focuses on the use case where SIP is gatewaying to
> ISUP, 
>> and thus where telephone numbers are likely to be used, I would be 
>> hesitant to rule out uses that do not involve telephone numbers - 
>> especially when the motivating use case given is REGISTER in an 
>> emergency context. Encapsulating the opaque numeric values of OLI into
> a 
>> SIP parameter also more or less restricts the use of this to entities 
>> that understand ISUP. This begs the obvious question: should there be
> a 
>> way in SIP (outside of gatewaying)  to express these same concepts? 
>> Since you already translate the CPC values into human-readable 
>> equivalents ("operator", "payphone", "test"), is there some reason why
> 
>> OLI can't receive the same treatment? The only example of the
> semantics 
>> of OLI that you give ("oli=29" meaning "prison") indeed seems it could
> 
>> admit of that translation. If there is some good reason why CPC should
> 
>> be translated and OLI should be encapsulated, the draft should 
>> explicitly motivate it.
>>
>> By the by, since the Acks mention that I submitted a version of this 
>> proposal prior to Rohan's, just as an historical aside I in fact split
> 
>> that out from the original draft-ietf-sip-privacy-04 by Flemming and 
>> Bill Marshall, which for some reason had it tacked into an appendix.
> The 
>> parameter proposed in that document (which can still be found via 
>> Google), the Nature of Party ("np") parameter, maps to a set of 
>> human-readable literals like "ordinary", "police",  "payphone", 
>> "emergency", and so on. The document then proposes a mapping to the
> ANI 
>> II digits; one could just as easily be provided for OLI or CPC in the 
>> ISUP variants of interest. The example given in that appendix shows
> this 
>> being used in a URI that doesn't contain a telephone number. In any 
>> event, if you want to have a pedigree of the idea in your Acks, you 
>> shouldn't leave out that document.
>>
>> Jon Peterson
>> NeuStar, Inc.
>>
>> Milan Patel wrote:
>>> Hi,
>>>
>>> A new version of the Internet Draft for the Calling Party's Category
> and
>>> Originating Line Identity URI Parameters is now available. 
>>> http://www.ietf.org/id/draft-patel-dispatch-cpc-oli-parameter-00.txt
>>>
>>> This is based on Rohan Mahy's draft-mahy-iptel-cpc-06
>>>
>>> Best regards,
>>> Milan
>>>
>>>
>>> -----Original Message-----
>>> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] Sent:
> 08 
>>> October 2009 18:40
>>> To: Patel, Milan (MOP:EP10)
>>> Cc: r.jesske@telekom.de; mdolly@att.com
>>> Subject: New Version Notification for
>>> draft-patel-dispatch-cpc-oli-parameter-00
>>>
>>> A new version of I-D, draft-patel-dispatch-cpc-oli-parameter-00.txt
> has
>>> been successfuly submitted by Milan Patel and posted to the IETF
>>> repository.
>>>
>>> Filename:     draft-patel-dispatch-cpc-oli-parameter
>>> Revision:     00
>>> Title:         Uniform Resource Identifier (URI) Parameters for
>>> indicating the Calling Party's Catagory and Originating Line Identity
>>> Creation_date:     2009-10-08
>>> WG ID:         Independent Submission
>>> Number_of_pages: 10
>>>
>>> Abstract:
>>> This document defines two new URI parameters to describe the calling
>>> party's category and toll class of service originating line
>>> information which are parameters also used in SS7 ISUP and other
>>> telephony signalling protocols.  The intended use of these URI
>>> parameters is for the "tel" URI or equivalent SIP URI representation.
>>>  
>>>
>>>
>>>
>>> The IETF Secretariat.
>>>
>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>   
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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On 15.03.2010 17:24, Adam Roach wrote:
> On 3/15/10 11:09, Mar 15, Paul Kyzivat wrote:
>>
>>
>> Adam Roach wrote:
>>> Greger:
>>>
>>> For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
>>> provide about 80% of what you're asking for.
>>
>> Selecting a *UAS* that has the capabilities the caller envisions 
>> using (such as video) were certainly among the things that were among 
>> the intended use cases.
>>
>> I'm dubious about their suitability for selecting a route to reach 
>> the UAS, assuming its the same UAS you are seeking.
>
> That's the other 20%. :)

Yes, the toolset is there. Making it work in a multi-vendor, 
multi-party, multi-capability UA environment is far away.

>
>
>> Also callerprefs are no guarantee (not even a superset) of the 
>> capabilities that will be used on a call. It is an entirely 
>> reasonable use case for a UAC to invite with no offer, and then 
>> negotiate the use of voice and video because the UAS offers that. Its 
>> also entirely reasonable for a call to start out as audio only, for 
>> the caller and callee to learn as a result of discussion that both 
>> are video capable, and then add video to the call.
>>
>> As non-voice media become more mainstream, and as e2e sip sessions 
>> become more feasible, we should be investigating how the 
>> infrastructure can avoid breaking these kinds of use cases. Nailing 
>> down the route in a way that precludes use of media not announced at 
>> call establishment time isn't going in a very helpful direction.
>
> Nailing down a route... for signaling? I'm a bit confused about how 
> that is going to preclude certain media types.

I'm not sure I'm following this, but a video phone will for example be 
able to make PSTN calls over one peering relationship, high-def 
voice/AAC-LD over another, and video over a third. If the original call 
was to a URI with hunt functionality forking across all three, it's 
pretty random where the call will end up and any attempt with no offer 
or renegotiate to video afterwards is not likely to work very well from 
a user perspective.
g-)

>
> Now, if we want to talk about media steering, that's a completely 
> different issue. I don't think SIP routing plays into that.
>
> /a


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Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)
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I agree with Jon here (even though I feel partially responsible for 
starting a war of the worlds thread...)
SPEERMINT seems to be the right place. I have trying to get through all 
the current drafts there, and I see that some of the issues I raised 
here as a response to egress-route is really something that should be 
raised in SPEERMINT context.
g-)

On 17.03.2010 00:29, Peterson, Jon wrote:
>
> I'm not sure that dispatching malas-dispatch-sip-egress-route requires 
> us to untwist this particular Gordian knot, nor to resurrect TRIP, 
>  nor to declare some kind of interplanetary Armageddon.
>
> I think Otmar Lendl has been right on the mark in this discussion. He 
> points out that global directories like ENUM serve well as an LUF and 
> poorly as an LRF, to speak in the SPEERMINT idiom - I don't think this 
> is an especially controversial point. I read malas-dispatch-sip-egress 
> as a document that argues that knowing the results of the LUF doesn't 
> necessarily help you to find a next hop to route your call through. 
> Again, I'd say not especially controversial.
>
> malas-dispatch-sip-egress-route then goes on to specify a few NAPTR 
> syntaxes that could contain a sort of joint LUF/LRF routing 
> information. The only conceptual problem I have with this approach is 
> that the entity that typically provisions the LUF doesn't know the 
> local topology of the originating network well enough to say useful 
> things about the LRF -- that is indeed the entire motivation for 
> considering the LUF and LRF as separate lookups. The draft sort of 
> glosses this over with a lot of passive voice when describing how the 
> ENUM provisioning gets done. Is it done by the terminating side, or 
> the originating side, or somehow both?
>
> In any event, from a DISPATCH perspective, I'd propose the problem 
> seems to lie comfortable within the parameters of SPEERMINT. Surely 
> one of the SPEERMINT chairs must think this is a worthwhile effort...
>
> Jon Peterson
> NeuStar, Inc.
>
> On 3/16/10 3:24 PM, "Adam Roach" <adam@nostrum.com> wrote:
>
>     On 3/16/10 3:59 PM, "Daryl Malas" <D.Malas@cablelabs.com> wrote:
>
>     > I believe this issue was summarized well during the 73rd IETF
>     meeting at the
>     > RAI open meeting, and captured in Jonathan's draft:
>     > http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00
>
>     I think Jonathan Rosenberg correctly identified that there was a
>     problem
>     here. And then, just like Jonathan Swift's original "Modest
>     Proposal," he
>     goes on to propose that the solution to our problem is best found
>     in eating
>     our own children.
>
>     I disagree profoundly with characterizing Rosenberg's Modest
>     Proposal as an
>     appropriate summary of the issues, as it gets the analysis
>     precisely wrong.
>     It proposes completely abandoning the vibrant world as a lost
>     cause, and
>     embracing the stagnant world as the only path forward. This isn't just
>     allowing collateral damage to the vibrant world for the sake of
>     expediency
>     in stagnant world solutions -- it's a suggestion that we nuke the
>     vibrant
>     world so it doesn't keep annoying people trying to develop stagnant
>     solutions.
>
>     I'll admit that it's easier to see the stagnant world, since the
>     telcos have
>     quite a bit of money to throw around driving its deployment. But
>     there are
>     people in the vibrant world working on new services. Go back and
>     re-read
>     Greger's description of what he's seeing deployed. Check out the
>     stuff that
>     AG Projects (http://ag-projects.com/) has been working on,
>     selling, and
>     actually deploying. I don't recall whether you would have been at the
>     meeting where I wore a t-shirt showing some novel SIP devices --
>     I'll bring
>     it to Anaheim.
>
>     /a
>
>
>     _______________________________________________
>     dispatch mailing list
>     dispatch@ietf.org
>     https://www.ietf.org/mailman/listinfo/dispatch
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


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I agree with Jon here (even though I feel partially responsible for
starting a war of the worlds thread...)<br>
SPEERMINT seems to be the right place. I have trying to get through all
the current drafts there, and I see that some of the issues I raised
here as a response to egress-route is really something that should be
raised in SPEERMINT context.<br>
g-)<br>
<br>
On 17.03.2010 00:29, Peterson, Jon wrote:
<blockquote cite="mid:C7C56268.3B15C%25jon.peterson@neustar.biz"
 type="cite">
  <title>Re: [dispatch] War of the Worlds (was Re: I-D Action:
draft-malas-dispatch-sip-egress-route-00)</title>
  <font face="Calibri, Verdana, Helvetica, Arial"><span
 style="font-size: 11pt;"><br>
I&#8217;m not sure that dispatching malas-dispatch-sip-egress-route requires
us to untwist this particular Gordian knot, nor to resurrect TRIP, &nbsp;nor
to declare some kind of interplanetary Armageddon.<br>
  <br>
I think Otmar Lendl has been right on the mark in this discussion. He
points out that global directories like ENUM serve well as an LUF and
poorly as an LRF, to speak in the SPEERMINT idiom - I don&#8217;t think this
is an especially controversial point. I read malas-dispatch-sip-egress
as a document that argues that knowing the results of the LUF doesn&#8217;t
necessarily help you to find a next hop to route your call through.
Again, I&#8217;d say not especially controversial.<br>
  <br>
malas-dispatch-sip-egress-route then goes on to specify a few NAPTR
syntaxes that could contain a sort of joint LUF/LRF routing
information. The only conceptual problem I have with this approach is
that the entity that typically provisions the LUF doesn&#8217;t know the
local topology of the originating network well enough to say useful
things about the LRF &#8211; that is indeed the entire motivation for
considering the LUF and LRF as separate lookups. The draft sort of
glosses this over with a lot of passive voice when describing how the
ENUM provisioning gets done. Is it done by the terminating side, or the
originating side, or somehow both?<br>
  <br>
In any event, from a DISPATCH perspective, I&#8217;d propose the problem
seems to lie comfortable within the parameters of SPEERMINT. Surely one
of the SPEERMINT chairs must think this is a worthwhile effort...<br>
  <br>
Jon Peterson<br>
NeuStar, Inc.<br>
  <br>
On 3/16/10 3:24 PM, "Adam Roach" &lt;<a moz-do-not-send="true"
 href="adam@nostrum.com">adam@nostrum.com</a>&gt; wrote:<br>
  <br>
  </span></font>
  <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
 style="font-size: 11pt;">On 3/16/10 3:59 PM, "Daryl Malas" &lt;<a
 moz-do-not-send="true" href="D.Malas@cablelabs.com">D.Malas@cablelabs.com</a>&gt;
wrote:<br>
    <br>
&gt; I believe this issue was summarized well during the 73rd IETF
meeting at the<br>
&gt; RAI open meeting, and captured in Jonathan's draft:<br>
&gt; <a moz-do-not-send="true"
 href="http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00">http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00</a><br>
    <br>
I think Jonathan Rosenberg correctly identified that there was a problem<br>
here. And then, just like Jonathan Swift's original "Modest Proposal,"
he<br>
goes on to propose that the solution to our problem is best found in
eating<br>
our own children.<br>
    <br>
I disagree profoundly with characterizing Rosenberg's Modest Proposal
as an<br>
appropriate summary of the issues, as it gets the analysis precisely
wrong.<br>
It proposes completely abandoning the vibrant world as a lost cause, and<br>
embracing the stagnant world as the only path forward. This isn't just<br>
allowing collateral damage to the vibrant world for the sake of
expediency<br>
in stagnant world solutions -- it's a suggestion that we nuke the
vibrant<br>
world so it doesn't keep annoying people trying to develop stagnant<br>
solutions.<br>
    <br>
I'll admit that it's easier to see the stagnant world, since the telcos
have<br>
quite a bit of money to throw around driving its deployment. But there
are<br>
people in the vibrant world working on new services. Go back and re-read<br>
Greger's description of what he's seeing deployed. Check out the stuff
that<br>
AG Projects (<a moz-do-not-send="true" href="http://ag-projects.com/">http://ag-projects.com/</a>)
has been working on, selling, and<br>
actually deploying. I don't recall whether you would have been at the<br>
meeting where I wore a t-shirt showing some novel SIP devices -- I'll
bring<br>
it to Anaheim.<br>
    <br>
/a<br>
    <br>
    <br>
_______________________________________________<br>
dispatch mailing list<br>
    <a moz-do-not-send="true" href="dispatch@ietf.org">dispatch@ietf.org</a><br>
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 href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
    <br>
    </span></font></blockquote>
  <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
dispatch mailing list
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  </pre>
</blockquote>
<br>
</body>
</html>

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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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+1 Daryl is right here. 

Can we let people try and solve their problems? At least in the intermediate
term vs seeing chaos rule? I'm getting this same thread in the E2MD
discussions.  I though the IETF was about trying to make the internet work
better.

The meta issue here is there are ..yes classic telecommunications carriers
have substantive technical problems interoperating or transitioning from
classic analog POTS to SIP centric solutions. GEE "what am I going to do
with my DMS 250 and 500's .. hummmm what do I do when I get a EOL letter for
my hundreds of 5ESS switches."

Part of the solution MAY assume a transitional mechanism that generally
accepts that some form of the classic intercarrier interconnection model may
exist and that policy MAY exist at each intermediate hop.

This assumes that you may need to crawl before you walk and that nirvana may
be a while off in the future.

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Daryl Malas
Sent: Tuesday, March 16, 2010 6:00 PM
To: 'Dean Willis'; Adam Roach
Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00

I believe this issue was summarized well during the 73rd IETF meeting at the
RAI open meeting, and captured in Jonathan's draft:
http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00

I believe we MUST solve this problem twice.  Let's be realistic.
Understanding the problem fully, defining the requirements, establishing the
group and hammering through the varying solutions will take 5 - 10 years to
complete.  The telecommunications industry will develop their own
proprietary mechanisms for doing this, creating greater fragmentation and
interoperability issues; or the IETF can step up and as an interim solution
say, since, you are going to solve this, do it this way.  At least in this
case, we will know where the industry is at in their resolution rather than
guessing at the plethora of "potentials" down the road.

Regards,

Daryl

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
Of Dean Willis
Sent: Tuesday, March 16, 2010 3:21 PM
To: Adam Roach
Cc: dispatch@ietf.org
Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00


On Mar 15, 2010, at 12:33 PM, Adam Roach wrote:

> ...

> Which is to say that I don't think there is a solution that satisfies 
> both groups. The divergence from the fork between the stagnant world 
> and the vibrant world is great enough that there is no common ground 
> left for these kinds of issues. As odious as it is, I'm afraid we're 
> stuck with solving this problem twice. The real challenge is ensuring 
> that stagnant solutions don't do lasting damage to the core protocols 
> -- we need to make sure they don't destroy the vibrant world with 
> collateral damage.

Beautifully said! I read it just after my similar but less-elegant rant,
which presumes that the stagnant world is likely to be selected over the
vibrant through the simple processes of expediency, apathy, corruption,
greed, and fear. But then, I'm an optimist.

--
Dean



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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Let me make sure I understand this correctly. I'm pushing for adoption 
and publication of Daryl's draft, and you're trying to disagree with me 
by pushing for adoption and publication of Daryl's draft. Is that where 
we stand?

/a

On 3/17/10 4:01 PM, Richard Shockey wrote:
> +1 Daryl is right here.
>
> Can we let people try and solve their problems? At least in the intermediate
> term vs seeing chaos rule? I'm getting this same thread in the E2MD
> discussions.  I though the IETF was about trying to make the internet work
> better.
>
> The meta issue here is there are ..yes classic telecommunications carriers
> have substantive technical problems interoperating or transitioning from
> classic analog POTS to SIP centric solutions. GEE "what am I going to do
> with my DMS 250 and 500's .. hummmm what do I do when I get a EOL letter for
> my hundreds of 5ESS switches."
>
> Part of the solution MAY assume a transitional mechanism that generally
> accepts that some form of the classic intercarrier interconnection model may
> exist and that policy MAY exist at each intermediate hop.
>
> This assumes that you may need to crawl before you walk and that nirvana may
> be a while off in the future.
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Daryl Malas
> Sent: Tuesday, March 16, 2010 6:00 PM
> To: 'Dean Willis'; Adam Roach
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
>
> I believe this issue was summarized well during the 73rd IETF meeting at the
> RAI open meeting, and captured in Jonathan's draft:
> http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00
>
> I believe we MUST solve this problem twice.  Let's be realistic.
> Understanding the problem fully, defining the requirements, establishing the
> group and hammering through the varying solutions will take 5 - 10 years to
> complete.  The telecommunications industry will develop their own
> proprietary mechanisms for doing this, creating greater fragmentation and
> interoperability issues; or the IETF can step up and as an interim solution
> say, since, you are going to solve this, do it this way.  At least in this
> case, we will know where the industry is at in their resolution rather than
> guessing at the plethora of "potentials" down the road.
>
> Regards,
>
> Daryl
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf
> Of Dean Willis
> Sent: Tuesday, March 16, 2010 3:21 PM
> To: Adam Roach
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
>
>
> On Mar 15, 2010, at 12:33 PM, Adam Roach wrote:
>
>    
>> ...
>>      
>    
>> Which is to say that I don't think there is a solution that satisfies
>> both groups. The divergence from the fork between the stagnant world
>> and the vibrant world is great enough that there is no common ground
>> left for these kinds of issues. As odious as it is, I'm afraid we're
>> stuck with solving this problem twice. The real challenge is ensuring
>> that stagnant solutions don't do lasting damage to the core protocols
>> -- we need to make sure they don't destroy the vibrant world with
>> collateral damage.
>>      
> Beautifully said! I read it just after my similar but less-elegant rant,
> which presumes that the stagnant world is likely to be selected over the
> vibrant through the simple processes of expediency, apathy, corruption,
> greed, and fear. But then, I'm an optimist.
>
> --
> Dean
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>
>    


From dean.willis@softarmor.com  Wed Mar 17 17:26:17 2010
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On Mar 17, 2010, at 9:51 AM, Paul Kyzivat wrote:
>>
>
> For those that truly describe properties of a URI (if there are any  
> of those), you may be right.
>
> For those that describe properties of a call, if they were carried  
> as part of the call rather than as properties of the URI, then it  
> wouldn't matter if the URI is sip or tel.
>
> But in reality, even the properties that describe the caller aren't  
> properly properties of the caller's URI. For instance, in principle  
> the same originating "number" might be used by one phone in a prison  
> and another in a police station.
>
> ISTM that piggybacking these on the TEL uri is simply a convenient  
> "ploy", because both phone numbers and these properties are managed  
> by the ITU.


Let's run through them:

ordinary: the caller has been identified and has no special features.  
At least not on this call; they might have some on another call, or  
might re-INVITE midstream if they change their mind, so don't count in  
it. Property of "call", not "caller".

Test: This is a test call. If the same guy calls you in 5 minutes, it  
might not be a test. Property of "call", not "caller".

operator: This call was generated by an operator position. Ostensibly,  
the caller is an operator? Probably a property of the "caller". One  
could envision putting this info in RFC 4474.

payphone: The calling station is a payphone. Again, a property of the  
caller.

unknown: I don't know what this means. It's probably got nothing to do  
with the caller.

mobile-hplmn/vplmn: The call was generated by a mobile device  
(property of the calling device, anyhow) in it's home/visited plmn  
(property of this call; doesn't persist if you call it back).

Some of the ANI II digits are things that are properties of a caller.  
Some, like Code 25, are attributes of call and caller. Many seem to  
specify specifc route-handling FOR THIS CALL, like not stripping off a  
leading "63".

So I'm not sure this is a useful categorization.

I'm pretty sure I don't want a "Type 67" modifier attached to A SIP  
URI. I'm also sure that most of these parameters make no sense on a  
TEL URI when that URI is used as a Request-URI; they don't route a  
call. They kindof make sense in a From: URI (or an RFC 4474 Identity  
header field), but they lack reversibility: one couldn't store this  
From: in a phone book, then turn around and call it back 5 minutes  
later. Consequently, the called party MUST understand them, or bad  
things will happen. This calls for an option tag, which is messy.

But I'm at a loss for proposing a better mechanism. Perhaps we could  
follow the SIP-T method and define another extension header field  
carried in the request?


--
Dean



From richard@shockey.us  Wed Mar 17 18:06:22 2010
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From: "Richard Shockey" <richard@shockey.us>
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Date: Wed, 17 Mar 2010 21:06:22 -0400
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Subject: [dispatch] Reminder Last Message: SIP Forum Luncheon at IETF 77 Monday 11:30 13:00
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The SIP Forum is going to hold a open meeting for the RAI community during
IETF 77. The purpose of this meeting is remind the SIP community of what the
SIP Forum is doing and discuss relevant technical issues that relate to IETF
WG's. The agenda for this luncheon will be to review current activities in
the SIP Forum and seek additional technical input from the RAI community on
mutual concerns.

 

As many of you know the SIP Forum is the principal sponsor of the SIPit
interoperability events and currently has 4 Task groups operating. 

 

BTW registration for SIPit 26 will be open soon.

 

Our principal technical initiatives are:

 

1.      SIP Connect 1.1 which is developing an enhanced  profile of IETF
Standards for PBX to SSP interoperability. As part of that work the IETF
MARTINI WG was formed to redefine the so called Implicit registration
procedure in SIP.

2.      FAX over SIP is investigating the interoperability of SIP in T.38
networks with a goal of possible recommendations to the relevant SDO's on
how fax can work better.

3.      Client User Agent Configuration which has a pending Recommendation
on how CUA devices could auto streamline the locating, retrieving and
maintaining of configuration-related information for SIP User Agents

4.      SIP in the SmartGrid. This is a Special Interest Group that is
evaluating the appropriateness of using SIP as a protocol for a number of
Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) deployments.

 

Information on the SIP Forum is available at 

 

http//www.sipforum.org

 

If you are interested in attending this event we have set up a Doodle Poll
to indicate your interest. Seating is somewhat limited ( 150 Maximum) lunch
will be provided.

 

Location .. Monday 11:30 13:00 the Laguna room and its located on the 4th
floor of the Anaheim Hilton

The link to the poll is: 

http://www.doodle.com/hde57nbpcdykzqfz 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype: rshockey101
LinkedIn : http://www.linkedin.com/in/rshockey101

 


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<div class=3DSection1>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The SIP Forum is going to hold a open meeting for =
the RAI
community during IETF 77. The purpose of this meeting is remind the SIP
community of what the SIP Forum is doing and discuss relevant technical =
issues
that relate to IETF WG&#8217;s. The agenda for this luncheon will be to =
review
current activities in the SIP Forum and seek additional technical input =
from
the RAI community on mutual concerns.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>As many of you know the SIP Forum is the principal =
sponsor
of the SIPit interoperability events and currently has 4 Task groups =
operating.
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>BTW registration for SIPit 26 will be open =
soon.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Our principal technical initiatives =
are:<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP Connect 1.1 which is developing an enhanced
&nbsp;profile of IETF Standards for PBX to SSP interoperability. As part =
of that
work the IETF MARTINI WG was formed to redefine the so called Implicit
registration procedure in SIP.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>FAX over SIP is investigating the =
interoperability of
SIP in T.38 networks with a goal of possible recommendations to the =
relevant
SDO&#8217;s on how fax can work better.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Client User Agent Configuration which has a =
pending
Recommendation on how CUA devices could auto streamline the locating,
retrieving and maintaining of configuration-related information for SIP =
User
Agents<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>SIP in the SmartGrid. This is a Special Interest =
Group
that is evaluating the appropriateness of using SIP as a protocol for a =
number
of Smart Grid areas, including the Home Area Network (HAN), Utilities'
Operational Network, and PEV (Plug-In Vehicle) =
deployments.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Information on the SIP Forum is available at =
<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Times =
New Roman","serif"'>http//www.sipforum.org</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>If you are interested in attending this event we =
have set up
a Doodle Poll to indicate your interest. Seating is somewhat limited ( =
150
Maximum) lunch will be provided.<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>Location .. </span><span
style=3D'font-size:10.0pt'>Monday 11:30 13:00 </span><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>the Laguna room and its =
located on
the 4th floor of the Anaheim Hilton</span><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'><o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;
background:yellow;mso-highlight:yellow'>The link to the poll =
is:</span><span
style=3D'font-size:10.0pt'> <br>
<br>
<a href=3D"http://www.doodle.com/hde57nbpcdykzqfz" =
target=3D"_blank">http://www.doodle.com/hde57nbpcdykzqfz</a>
</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>
Shockey Consulting<br>
Chairman of the Board of Directors SIP Forum<br>
PSTN Mobile: +1 703.593.2683<br>
&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>
skype: rshockey101<br>
LinkedIn : <a =
href=3D"http://www.linkedin.com/in/rshockey101">http://www.linkedin.com/i=
n/rshockey101</a></span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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From Milan.Patel@InterDigital.com  Thu Mar 18 03:08:59 2010
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Thread-Topic: [dispatch] FW: New VersionNotification	for	draft-patel-dispatch-cpc-oli-parameter-00
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From: "Patel, Milan" <Milan.Patel@InterDigital.com>
To: "Dean Willis" <dean.willis@softarmor.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] FW: New VersionNotification	for	draft-patel-dispatch-cpc-oli-parameter-00
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Hi Dean, Paul,

I now understand your comments about some OLI values characterizing the
call vs caller.

The recommended use of the URI parameter is in the P-Asserted-Identity.
If this is not supported, the OLI can be added in the From header. From
a 3GPP point of view, the OLI would be added to the P-Asserted-Identity
header. The CPC and OLI values are subject to trust domain rules, so
they will not be sent to the called party. It is important they get
interworked at the Media Gateway Control Function to ISUP information
elements.

Regards,
Milan


-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]=20
Sent: Thursday, March 18, 2010 12:26 AM
To: Paul Kyzivat
Cc: Patel, Milan; DISPATCH
Subject: Re: [dispatch] FW: New VersionNotification for
draft-patel-dispatch-cpc-oli-parameter-00


On Mar 17, 2010, at 9:51 AM, Paul Kyzivat wrote:
>>
>
> For those that truly describe properties of a URI (if there are any =20
> of those), you may be right.
>
> For those that describe properties of a call, if they were carried =20
> as part of the call rather than as properties of the URI, then it =20
> wouldn't matter if the URI is sip or tel.
>
> But in reality, even the properties that describe the caller aren't =20
> properly properties of the caller's URI. For instance, in principle =20
> the same originating "number" might be used by one phone in a prison =20
> and another in a police station.
>
> ISTM that piggybacking these on the TEL uri is simply a convenient =20
> "ploy", because both phone numbers and these properties are managed =20
> by the ITU.


Let's run through them:

ordinary: the caller has been identified and has no special features. =20
At least not on this call; they might have some on another call, or =20
might re-INVITE midstream if they change their mind, so don't count in =20
it. Property of "call", not "caller".

Test: This is a test call. If the same guy calls you in 5 minutes, it =20
might not be a test. Property of "call", not "caller".

operator: This call was generated by an operator position. Ostensibly, =20
the caller is an operator? Probably a property of the "caller". One =20
could envision putting this info in RFC 4474.

payphone: The calling station is a payphone. Again, a property of the =20
caller.

unknown: I don't know what this means. It's probably got nothing to do =20
with the caller.

mobile-hplmn/vplmn: The call was generated by a mobile device =20
(property of the calling device, anyhow) in it's home/visited plmn =20
(property of this call; doesn't persist if you call it back).

Some of the ANI II digits are things that are properties of a caller. =20
Some, like Code 25, are attributes of call and caller. Many seem to =20
specify specifc route-handling FOR THIS CALL, like not stripping off a =20
leading "63".

So I'm not sure this is a useful categorization.

I'm pretty sure I don't want a "Type 67" modifier attached to A SIP =20
URI. I'm also sure that most of these parameters make no sense on a =20
TEL URI when that URI is used as a Request-URI; they don't route a =20
call. They kindof make sense in a From: URI (or an RFC 4474 Identity =20
header field), but they lack reversibility: one couldn't store this =20
From: in a phone book, then turn around and call it back 5 minutes =20
later. Consequently, the called party MUST understand them, or bad =20
things will happen. This calls for an option tag, which is messy.

But I'm at a loss for proposing a better mechanism. Perhaps we could =20
follow the SIP-T method and define another extension header field =20
carried in the request?


--
Dean



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inline

Dean Willis wrote:
> 
> On Mar 17, 2010, at 9:51 AM, Paul Kyzivat wrote:
>>>
>>
>> For those that truly describe properties of a URI (if there are any of 
>> those), you may be right.
>>
>> For those that describe properties of a call, if they were carried as 
>> part of the call rather than as properties of the URI, then it 
>> wouldn't matter if the URI is sip or tel.
>>
>> But in reality, even the properties that describe the caller aren't 
>> properly properties of the caller's URI. For instance, in principle 
>> the same originating "number" might be used by one phone in a prison 
>> and another in a police station.
>>
>> ISTM that piggybacking these on the TEL uri is simply a convenient 
>> "ploy", because both phone numbers and these properties are managed by 
>> the ITU.
> 
> 
> Let's run through them:
> 
> ordinary: the caller has been identified and has no special features. At 
> least not on this call; they might have some on another call, or might 
> re-INVITE midstream if they change their mind, so don't count in it. 
> Property of "call", not "caller".
> 
> Test: This is a test call. If the same guy calls you in 5 minutes, it 
> might not be a test. Property of "call", not "caller".
> 
> operator: This call was generated by an operator position. Ostensibly, 
> the caller is an operator? Probably a property of the "caller". One 
> could envision putting this info in RFC 4474.

> payphone: The calling station is a payphone. Again, a property of the 
> caller.
> 
> unknown: I don't know what this means. It's probably got nothing to do 
> with the caller.
> 
> mobile-hplmn/vplmn: The call was generated by a mobile device (property 
> of the calling device, anyhow) in it's home/visited plmn (property of 
> this call; doesn't persist if you call it back).
> 
> Some of the ANI II digits are things that are properties of a caller. 
> Some, like Code 25, are attributes of call and caller. Many seem to 
> specify specifc route-handling FOR THIS CALL, like not stripping off a 
> leading "63".

I included several others in my earlier post:

Properties of the callee:

   * 30 Intercept (blank) - for calls to unassigned directory number (DN)
   * 31 Intercept (trouble) - for calls to directory numbers (DN) that
        have been manually placed in trouble-busy state by Telco
        personnel
   * 32 Intercept (regular) - for calls to recently changed or
        disconnected numbers

The following can be assigned mid-call, as a result of call handling, 
and so are properties of neither the caller nor the callee. I gather 34 
is what Dean is calling "operator". But from the description, the "From" 
will probably still be as supplied by the caller, rather than 
identifying the operator. So again its a (dynamic) property of the call, 
not the caller.

   * 34 Telco Operator Handled Call - after the Telco Operator Services
        System has handled a call for an IC, it may change the standard
        ANI digits to "34", before outpulsing the sequence to the IC,
        when the Telco performs all call handling functions, e.g.,
        billing. The code tells the IC that the BOC has performed billing
        on the call and the IC only has to complete the call.
   * 52 Outward Wide Area Telecommunications Service (OUTWATS) - this
        service allows customers to make calls to a certain zone(s) or
        band(s) on a direct dialed basis for a flat monthly charge or for
        a charge based on accumulated usage. OUTWATS lines can dial
        station-to-station calls directly to points within the selected
        band(s) or zone(s). The LEC performs a screening function to
        determine the correct charging and routing for OUTWATS calls
        based on the customer's class of service and the service area of
        the call party. When these calls are routed to the interexchange
        carrier via a combined WATS-POTS trunk group, it is necessary to
        identify the WATS calls with the ANI code "52".

> So I'm not sure this is a useful categorization.
> 
> I'm pretty sure I don't want a "Type 67" modifier attached to A SIP URI. 
> I'm also sure that most of these parameters make no sense on a TEL URI 
> when that URI is used as a Request-URI; they don't route a call. They 
> kindof make sense in a From: URI (or an RFC 4474 Identity header field), 
> but they lack reversibility: one couldn't store this From: in a phone 
> book, then turn around and call it back 5 minutes later. Consequently, 
> the called party MUST understand them, or bad things will happen. This 
> calls for an option tag, which is messy.
> 
> But I'm at a loss for proposing a better mechanism. Perhaps we could 
> follow the SIP-T method and define another extension header field 
> carried in the request?

Since at least some of this stuff apparently has to change mid-call, I 
think it calls for a new header to carry it.

	Thanks,
	Paul

From pkyzivat@cisco.com  Thu Mar 18 06:19:01 2010
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Patel, Milan wrote:
> Hi Dean, Paul,
> 
> I now understand your comments about some OLI values characterizing the
> call vs caller.
> 
> The recommended use of the URI parameter is in the P-Asserted-Identity.
> If this is not supported, the OLI can be added in the From header. From
> a 3GPP point of view, the OLI would be added to the P-Asserted-Identity
> header. The CPC and OLI values are subject to trust domain rules, so
> they will not be sent to the called party. It is important they get
> interworked at the Media Gateway Control Function to ISUP information
> elements.

Using this in the PAI header does provide a vehicle for mid-call changes.

But it doesn't alter the reality that the PAI also is intended to make 
an assertion about the *caller* while some of this other information is 
about the *call* rather than the caller.

For example, how does it make any sense to include

   * 31 Intercept (trouble) - for calls to directory numbers (DN) that
        have been manually placed in trouble-busy state by Telco
        personnel

into the PAI header???

	Thanks,
	Paul

> Regards,
> Milan
> 
> 
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: Thursday, March 18, 2010 12:26 AM
> To: Paul Kyzivat
> Cc: Patel, Milan; DISPATCH
> Subject: Re: [dispatch] FW: New VersionNotification for
> draft-patel-dispatch-cpc-oli-parameter-00
> 
> 
> On Mar 17, 2010, at 9:51 AM, Paul Kyzivat wrote:
>> For those that truly describe properties of a URI (if there are any  
>> of those), you may be right.
>>
>> For those that describe properties of a call, if they were carried  
>> as part of the call rather than as properties of the URI, then it  
>> wouldn't matter if the URI is sip or tel.
>>
>> But in reality, even the properties that describe the caller aren't  
>> properly properties of the caller's URI. For instance, in principle  
>> the same originating "number" might be used by one phone in a prison  
>> and another in a police station.
>>
>> ISTM that piggybacking these on the TEL uri is simply a convenient  
>> "ploy", because both phone numbers and these properties are managed  
>> by the ITU.
> 
> 
> Let's run through them:
> 
> ordinary: the caller has been identified and has no special features.  
> At least not on this call; they might have some on another call, or  
> might re-INVITE midstream if they change their mind, so don't count in  
> it. Property of "call", not "caller".
> 
> Test: This is a test call. If the same guy calls you in 5 minutes, it  
> might not be a test. Property of "call", not "caller".
> 
> operator: This call was generated by an operator position. Ostensibly,  
> the caller is an operator? Probably a property of the "caller". One  
> could envision putting this info in RFC 4474.
> 
> payphone: The calling station is a payphone. Again, a property of the  
> caller.
> 
> unknown: I don't know what this means. It's probably got nothing to do  
> with the caller.
> 
> mobile-hplmn/vplmn: The call was generated by a mobile device  
> (property of the calling device, anyhow) in it's home/visited plmn  
> (property of this call; doesn't persist if you call it back).
> 
> Some of the ANI II digits are things that are properties of a caller.  
> Some, like Code 25, are attributes of call and caller. Many seem to  
> specify specifc route-handling FOR THIS CALL, like not stripping off a  
> leading "63".
> 
> So I'm not sure this is a useful categorization.
> 
> I'm pretty sure I don't want a "Type 67" modifier attached to A SIP  
> URI. I'm also sure that most of these parameters make no sense on a  
> TEL URI when that URI is used as a Request-URI; they don't route a  
> call. They kindof make sense in a From: URI (or an RFC 4474 Identity  
> header field), but they lack reversibility: one couldn't store this  
> From: in a phone book, then turn around and call it back 5 minutes  
> later. Consequently, the called party MUST understand them, or bad  
> things will happen. This calls for an option tag, which is messy.
> 
> But I'm at a loss for proposing a better mechanism. Perhaps we could  
> follow the SIP-T method and define another extension header field  
> carried in the request?
> 
> 
> --
> Dean
> 
> 
> 

From D.Malas@cablelabs.com  Thu Mar 18 07:58:46 2010
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To: "'Greger Viken Teigre'" <gregert@teigre.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 18 Mar 2010 08:58:48 -0600
Thread-Topic: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)
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Are there any other thoughts on "Speermint-izing" this work?  --Daryl

________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Greger Viken Teigre
Sent: Wednesday, March 17, 2010 2:33 PM
To: dispatch@ietf.org
Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-=
dispatch-sip-egress-route-00)

I agree with Jon here (even though I feel partially responsible for startin=
g a war of the worlds thread...)
SPEERMINT seems to be the right place. I have trying to get through all the=
 current drafts there, and I see that some of the issues I raised here as a=
 response to egress-route is really something that should be raised in SPEE=
RMINT context.
g-)

On 17.03.2010 00:29, Peterson, Jon wrote:

I'm not sure that dispatching malas-dispatch-sip-egress-route requires us t=
o untwist this particular Gordian knot, nor to resurrect TRIP,  nor to decl=
are some kind of interplanetary Armageddon.

I think Otmar Lendl has been right on the mark in this discussion. He point=
s out that global directories like ENUM serve well as an LUF and poorly as =
an LRF, to speak in the SPEERMINT idiom - I don't think this is an especial=
ly controversial point. I read malas-dispatch-sip-egress as a document that=
 argues that knowing the results of the LUF doesn't necessarily help you to=
 find a next hop to route your call through. Again, I'd say not especially =
controversial.

malas-dispatch-sip-egress-route then goes on to specify a few NAPTR syntaxe=
s that could contain a sort of joint LUF/LRF routing information. The only =
conceptual problem I have with this approach is that the entity that typica=
lly provisions the LUF doesn't know the local topology of the originating n=
etwork well enough to say useful things about the LRF - that is indeed the =
entire motivation for considering the LUF and LRF as separate lookups. The =
draft sort of glosses this over with a lot of passive voice when describing=
 how the ENUM provisioning gets done. Is it done by the terminating side, o=
r the originating side, or somehow both?

In any event, from a DISPATCH perspective, I'd propose the problem seems to=
 lie comfortable within the parameters of SPEERMINT. Surely one of the SPEE=
RMINT chairs must think this is a worthwhile effort...

Jon Peterson
NeuStar, Inc.

On 3/16/10 3:24 PM, "Adam Roach" <adam@nostrum.com> wrote:

On 3/16/10 3:59 PM, "Daryl Malas" <D.Malas@cablelabs.com> wrote:

> I believe this issue was summarized well during the 73rd IETF meeting at =
the
> RAI open meeting, and captured in Jonathan's draft:
> http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-00

I think Jonathan Rosenberg correctly identified that there was a problem
here. And then, just like Jonathan Swift's original "Modest Proposal," he
goes on to propose that the solution to our problem is best found in eating
our own children.

I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
appropriate summary of the issues, as it gets the analysis precisely wrong.
It proposes completely abandoning the vibrant world as a lost cause, and
embracing the stagnant world as the only path forward. This isn't just
allowing collateral damage to the vibrant world for the sake of expediency
in stagnant world solutions -- it's a suggestion that we nuke the vibrant
world so it doesn't keep annoying people trying to develop stagnant
solutions.

I'll admit that it's easier to see the stagnant world, since the telcos hav=
e
quite a bit of money to throw around driving its deployment. But there are
people in the vibrant world working on new services. Go back and re-read
Greger's description of what he's seeing deployed. Check out the stuff that
AG Projects (http://ag-projects.com/) has been working on, selling, and
actually deploying. I don't recall whether you would have been at the
meeting where I wore a t-shirt showing some novel SIP devices -- I'll bring
it to Anaheim.

/a


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Re: [dispatch] War of the Worlds (was Re: I-D Action: dr=
aft-malas-dispatch-sip-egress-route-00)</TITLE>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18876"></HEAD>
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D881235814-18032010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Are there any other thoughts on "Speermint-izing" thi=
s=20
work?&nbsp; --Daryl</FONT></SPAN></DIV><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>On Behalf Of </B>Greger Viken=20
Teigre<BR><B>Sent:</B> Wednesday, March 17, 2010 2:33 PM<BR><B>To:</B>=20
dispatch@ietf.org<BR><B>Subject:</B> Re: [dispatch] War of the Worlds (was =
Re:=20
I-D Action: draft-malas-dispatch-sip-egress-route-00)<BR></FONT><BR></DIV>
<DIV></DIV>I agree with Jon here (even though I feel partially responsible =
for=20
starting a war of the worlds thread...)<BR>SPEERMINT seems to be the right=
=20
place. I have trying to get through all the current drafts there, and I see=
 that=20
some of the issues I raised here as a response to egress-route is really=20
something that should be raised in SPEERMINT context.<BR>g-)<BR><BR>On=20
17.03.2010 00:29, Peterson, Jon wrote:=20
<BLOCKQUOTE cite=3Dmid:C7C56268.3B15C%25jon.peterson@neustar.biz=20
  type=3D"cite"><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
  style=3D"FONT-SIZE: 11pt"><BR>I&#8217;m not sure that dispatching=20
  malas-dispatch-sip-egress-route requires us to untwist this particular Go=
rdian=20
  knot, nor to resurrect TRIP, &nbsp;nor to declare some kind of interplane=
tary=20
  Armageddon.<BR><BR>I think Otmar Lendl has been right on the mark in this=
=20
  discussion. He points out that global directories like ENUM serve well as=
 an=20
  LUF and poorly as an LRF, to speak in the SPEERMINT idiom - I don&#8217;t=
 think this=20
  is an especially controversial point. I read malas-dispatch-sip-egress as=
 a=20
  document that argues that knowing the results of the LUF doesn&#8217;t ne=
cessarily=20
  help you to find a next hop to route your call through. Again, I&#8217;d =
say not=20
  especially controversial.<BR><BR>malas-dispatch-sip-egress-route then goe=
s on=20
  to specify a few NAPTR syntaxes that could contain a sort of joint LUF/LR=
F=20
  routing information. The only conceptual problem I have with this approac=
h is=20
  that the entity that typically provisions the LUF doesn&#8217;t know the =
local=20
  topology of the originating network well enough to say useful things abou=
t the=20
  LRF &#8211; that is indeed the entire motivation for considering the LUF =
and LRF as=20
  separate lookups. The draft sort of glosses this over with a lot of passi=
ve=20
  voice when describing how the ENUM provisioning gets done. Is it done by =
the=20
  terminating side, or the originating side, or somehow both?<BR><BR>In any=
=20
  event, from a DISPATCH perspective, I&#8217;d propose the problem seems t=
o lie=20
  comfortable within the parameters of SPEERMINT. Surely one of the SPEERMI=
NT=20
  chairs must think this is a worthwhile effort...<BR><BR>Jon=20
  Peterson<BR>NeuStar, Inc.<BR><BR>On 3/16/10 3:24 PM, "Adam Roach" &lt;<A=
=20
  href=3D"adam@nostrum.com" moz-do-not-send=3D"true">adam@nostrum.com</A>&g=
t;=20
  wrote:<BR><BR></SPAN></FONT>
  <BLOCKQUOTE><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
    style=3D"FONT-SIZE: 11pt">On 3/16/10 3:59 PM, "Daryl Malas" &lt;<A=20
    href=3D"D.Malas@cablelabs.com"=20
    moz-do-not-send=3D"true">D.Malas@cablelabs.com</A>&gt; wrote:<BR><BR>&g=
t; I=20
    believe this issue was summarized well during the 73rd IETF meeting at=
=20
    the<BR>&gt; RAI open meeting, and captured in Jonathan's draft:<BR>&gt;=
 <A=20
    href=3D"http://tools.ietf.org/html/draft-rosenberg-rai-modest-proposal-=
00"=20
    moz-do-not-send=3D"true">http://tools.ietf.org/html/draft-rosenberg-rai=
-modest-proposal-00</A><BR><BR>I=20
    think Jonathan Rosenberg correctly identified that there was a=20
    problem<BR>here. And then, just like Jonathan Swift's original "Modest=
=20
    Proposal," he<BR>goes on to propose that the solution to our problem is=
 best=20
    found in eating<BR>our own children.<BR><BR>I disagree profoundly with=
=20
    characterizing Rosenberg's Modest Proposal as an<BR>appropriate summary=
 of=20
    the issues, as it gets the analysis precisely wrong.<BR>It proposes=20
    completely abandoning the vibrant world as a lost cause, and<BR>embraci=
ng=20
    the stagnant world as the only path forward. This isn't just<BR>allowin=
g=20
    collateral damage to the vibrant world for the sake of expediency<BR>in=
=20
    stagnant world solutions -- it's a suggestion that we nuke the=20
    vibrant<BR>world so it doesn't keep annoying people trying to develop=20
    stagnant<BR>solutions.<BR><BR>I'll admit that it's easier to see the=20
    stagnant world, since the telcos have<BR>quite a bit of money to throw=
=20
    around driving its deployment. But there are<BR>people in the vibrant w=
orld=20
    working on new services. Go back and re-read<BR>Greger's description of=
 what=20
    he's seeing deployed. Check out the stuff that<BR>AG Projects (<A=20
    href=3D"http://ag-projects.com/"=20
    moz-do-not-send=3D"true">http://ag-projects.com/</A>) has been working =
on,=20
    selling, and<BR>actually deploying. I don't recall whether you would ha=
ve=20
    been at the<BR>meeting where I wore a t-shirt showing some novel SIP de=
vices=20
    -- I'll bring<BR>it to=20
    Anaheim.<BR><BR>/a<BR><BR><BR>_________________________________________=
______<BR>dispatch=20
    mailing list<BR><A href=3D"dispatch@ietf.org"=20
    moz-do-not-send=3D"true">dispatch@ietf.org</A><BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/dispatch"=20
    moz-do-not-send=3D"true">https://www.ietf.org/mailman/listinfo/dispatch=
</A><BR><BR></SPAN></FONT></BLOCKQUOTE><PRE wrap=3D""><FIELDSET class=3Dmim=
eAttachmentHeader></FIELDSET>
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Paul,

thank you for the comments and please excuse my late answer. I brooded
quite long about your comments.
I like very much your proposal about the "modifier"-category. I hope
Alan will agree to the change.
I talked to people involved in the DT IMS /NGN implementation
concerning some of the issues.

Please find the answers/comments in-line.

>> - Section 4.3.6 on crisis deleted
>
> Then why is it still mentioned in REQ-2?

 I will delete it in the next version.We just missed it when we did
the change.

>
>> - "tone" and "ringback" combination deleted from chapter "Combinations o=
f
>> URN".
>> - Chapter "Combination Rules for Alert URN Indications" upgraded. We
>> hope the rules are =A0much more simple now when we distinguish between
>> ring tones and ringback tones.
>
> ----------
>
> Section 1:
>
> =A0 This specification does not change the usage of the SIP Alert-Info-
> =A0 header defined in the RFC3261. =A0The Alert-Info-header can be used i=
n
> =A0 INVITE requests and 180 Ringing responses.
>
> IMO it would make sense to make an extension to 3261 to permit this heade=
r
> in any 18x response. This could help resolve issues about what ought to b=
e
> rendered to the caller in case of 182 or 183.

This would be very usefull, but can we do it in this draft? Or we need
to wait for 3261bis?


>
> Section 3:
>
> =A0 =A0 =A0... =A0The left-most
> =A0 =A0 =A0label is called "alert-category" and is separated from right-s=
ide
> =A0 =A0 =A0of the alert-identifier, the alert-indication, by a semicolon.=
..
>
> =A0 =A0 =A0 =A0 urn:alert:tone:{tone-indication}
>
> The text says "semicolon" but the example uses colon, and the ABNF also
> shows colon.

Agree. It should be "colon". I will change this.
>
> Section 4:
>
> The way the word "tone" is used in this section could cause confusion, af=
ter
> having gone to the trouble of explaining that this mechanism is for
> indicating semantic intent that might not be rendered as tones. Its furth=
er
> confused because section 4.1 PBX Tones are encoded using urn:alert:tone, =
but
> that 4.2 Service Tones are encoded using urn:alert:service.
>
> I suggest using some other word than "tone" in cases when audio rendering
> may not be used. For example "indication". Perhaps "caller indication" co=
uld
> be used instead of ringback, and "callee indication" could be used instea=
d
> of "ring". (Thats in descriptive text. Perhaps the alert-category of "ton=
e"
> should be changed too, but I'm not ready to propose that yet.)

Currently, the category "tone" is curently used only for PBX ring
tones (in case of audio rendering). It is somehow confusing, because
"service" and "ringback" are also tones. I don't like "tones", either.
I thought about changing it to ring-tone or PBX ring-tone, but I
didn't do it for the last version because such a change may affect
implementations of earlier versions of the draft. I think such changes
need a broad agreement before changing the draft. In particular, we
need Alan's agreement because the PBX tones is his part.

Personally, I would prefer "ring indication"  and "ringback
indication"  to   "caller indication" and  "callee indication". I
think the meaning is more obvious, even if the rendering is not audio.
In SIP we also habe 180 Ringing too, independent of the rendering. But
this is just a personal preference, I may be wrong .  We could ask on
the ML what other people think.


Another problem I see here is that currently "tone" (or "callee
indication" or "ring indication") is currently used only for PBX
ring-tones, not for ring tones in general. Somehow we should enable
also other kinds of ring tones. We can
a) delete the restriction to PBX ring-tones and use e.g.
urn:alert:ring-tone:normal also for public networks or
b) splitt the "tone" (or which ever new name we choose) category in
"pbx ring indications", and  "public ring indications" and maybe
others. People could then define new ring-tone categories in the
future,  e.g. "3GPP2 ring indications".
c) define the category as a chain consisting of a main-category and
sub-categories. , e.g. ring-indication:pbx,  ring-indication:public. .
The urns would be then urn:alert:ring-tone:pbx:normal,
urn:alert:ring-tone:pbx:internal,  urn:alert:ring-tone:pbx:external,
urn:alert:ring-tone:public:normal .

My preference is c), but then we have to change the ABNF Syntax and
define the alert-category in chap. 3 as being something like a chain
like

alert-categoryr=3D top-alert-category  *(":" alert-subcategory)
 top-alert-category =3D "ring-tone"/"ringback"/"service"  or
"callee-indication"/"caller-indication"/"service"
alert-subcategory =3D let-dig [*let-dig-hyp  let-dig]

I hope I don't run with this into the combination problems again....


>
> Sections 4.2.6, 4.2.7, 4.2.8:
>
> These talk about "this sub-level". But sub-level isn't defined anywhere.
> There is "sub-indication" in the ABNF, but the examples given in these
> sections don't use that.

"Sub-level" is another relic from the old version. I will replace it
with "alert indication".
"Sub-indication"  is no longer used in the current use cases. We (Alan
and I) considered the issue, but decided not to delete the
sub-indication yet, until the new format is consolidated.  For the
examples in the draft, it is still correct, just not used th the
draft. If we agree to use things like urn:alert:modifier:country:za
or  urn:alert:ring-tone:pbx:normal we have to change the format
anyway.


>
> These are shown as urn:alert:tone, which means they can only be used for
> callee indication, not caller indication. Is that intended?

Yes. This is described in chap 7.2, second paragraph.

>
> I know this might open old wounds, but perhaps it would be better to make
> these values simply be sub-indications on those values where it makes sen=
se.
> But if you don't want to go there I understand.

With the sub-indications we had the combination problem.....This is
why we changed it.

>
> Section 4.3:
>
> Aren't there also country-specific ring tones? (I know the British ones i=
n
> the movies are different from the American ones.) Should there be provisi=
on
> for specifying these? See more on this below.

Up to now, nobody stated the requirement that they are needed.
I asked peopple within DT and they don't see a need for that. The DT
ring tone is implemented in each DT end device. (I am not happy with
it, but that's it.) There is no specific German ring tone, German
carriers have their own ring tone specification, but this
specifications are more or less the ITU-T specification on ring tone,
with some minor differences.

I think it should be possible in principle to define country ring
tones, if anyone needs them. With sub-categories we could do this with
<urn:alert:ring-indication:public:normal><
urn:alert:modifier:country:za> or just
<urn:alert:ring-indication:country:za>

I would avoid defining use cases for which it is very unclear if
someone needs them. But we should try to make the syntax as fexible as
possible for later use cases.

> Why no provision for non-country-specific ringbacks? This is potentially =
a
> case for customization - perhaps the callee's presence status could be us=
ed
> by the callee to select an appropriate ringback to the caller.

It's true. We have to be change the format to allow more than country
ringback tones. I don't have any clear idea how to do it yet. Maybe
<urn:alert:ringback-indication:country:za>  or
<urn:alert:ringback-indication:public:normal><
urn:alert:modifier:country:za> if we use sub-categories?

Concerning the presence status...I think this is more complex and
dedicated presence mechanisms should be used. We may run into privacy
and all other issues connected to presence. I would define here as a
use case. But it should be possible that people add this later if
there is a need for it.
>
> Section 4.4:
>
> Are all combinations potentially valid? Or can there be at most one per
> category?

One per main category. I think we can state this if we use modifiers.
>
> How about having alert:modifier as a category? Then all the other categor=
ies
> could be mutually exclusive, and zero or more modifiers would adjust
> whatever one was supplied. In that case, priority, zip, and delayed would=
 be
> modifiers.

I like the modifier :-). But we have to check with Alan and the
Mailing list, because of possoible existing implementations.

 (Maybe normal also ought to be a modifier, and maybe country code
> should also be a modifier - urn:alert:modifier:country:za.)
I don't have any opinion yet.

>
> Section 5:
>
> I think it would be a good idea to mention something about what a UA shou=
ld
> do with these in more complex cases:
> - alert-info received but in-band media also received
> - an invite was forked and the UAC is getting responses from
> =A0multiple early dialogs, containing conflicting alert-info
> =A0and/or in-band media.
>
> It occurs to me that in some cases early media should preempt the renderi=
ng
> of alert-info, but in other cases it should not. Maybe that should be lef=
t
> to implementor discretion, but maybe there are cases where this should be
> deterministic - I haven't thought about it enough to have a firm opinion.

This is one of the basic SIP unsolved problems. I don't think we
should try to solve here this problem. Every end device, carrier...has
his own rules. DT already uses urn:alert:service: call-waiting. If a
UE receives it, the user gets the message "This is a waiting call"  on
the display if a display is there. We have no conflict with early
media. Other carrier or end devices can have other rules.

>
> Rolling all that up into some overall suggestions:
>
> You have some things that only apply to the "ring", some that only apply =
to
> the "ringback" and some that apply to both. And then there are things tha=
t
> can act as modifiers to any of the others. But the distinction between ri=
ng
> and ringback is redundant with the message it travels in. So how about:
>
> urn:alert:service:xxx has all the things you have as services plus all th=
e
> things you have as "tones". In an INVITE external/internal describe the
> caller, and in a 18x they describe the callee. Maybe some additional valu=
es
> that are useful for describing the callee status.
>
> urn:alert:modifier:xxx where xxx is priority/zip/delayed or country.cc
> where cc is a country code.
>
> Combinations allowed are one service, and one or more distinct modifiers.
> (Must be distinct based on top-level.)
>

I think this would work and makes things quite simple, at least for
the use cases we have now.    I suppose if there is no service , but a
modifier like country.cc, then the default ring or ringback tone are
assumed. We also must remove the last paragraph on page 8 because
country.cc is a valid tone but country is not.   Country is here some
kind of modifier type.
I see only one problem here, which I don't nknow how to handle.What
happens when people add new alert-indications? Just add them to the
"service" or "modifier"-category? If so, we have two unstructured
lists of alert indications, which may become long with the time. I
suppose proxies may want to remove alert-indications which are not
relevant for their users, e.g. public carrier would want to remove
pbx-specific alert-indications generated at the other end of the
network, e.g. internal/external.  If there is no simple way to
recognize the alert indication which are not relevant for them, they
will use white lists and allow exactly the alert identifiers they use
within their own domain...I don't this is what we want.  But I may be
wrong with this.

Thanks  a lot
Laura

From dworley@avaya.com  Thu Mar 18 11:35:36 2010
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From: "WORLEY, DALE R (DALE)" <dworley@avaya.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "Patel, Milan" <Milan.Patel@InterDigital.com>
Date: Thu, 18 Mar 2010 14:32:35 -0400
Thread-Topic: [dispatch] FW: New	VersionNotification	for draft-patel-dispatch-cpc-oli-parameter-00
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Subject: Re: [dispatch] FW: New	VersionNotification	for	draft-patel-dispatch-cpc-oli-parameter-00
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________________________________________
From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Pa=
ul Kyzivat [pkyzivat@cisco.com]

But it doesn't alter the reality that the PAI also is intended to make
an assertion about the *caller* while some of this other information is
about the *call* rather than the caller.

For example, how does it make any sense to include

   * 31 Intercept (trouble) - for calls to directory numbers (DN) that
        have been manually placed in trouble-busy state by Telco
        personnel
_________________________________________

It's legitimate to add a parameter to the request-URI to modify how the cal=
l is routed to its destination, and that covers a number of the CPC cases. =
 But number 31 above is neither a property of the caller or a modifier of t=
he callee, it's a dynamic (!) attribute of the call itself.  Also, there ar=
e data-domain problems, as presumably the "Intercept (trouble)" status can =
be combined with any caller-type, whereas the value 31 erases any informati=
on previously recorded in the CPC.

Dale

From pkyzivat@cisco.com  Thu Mar 18 12:09:04 2010
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Subject: Re: [dispatch] FW: New	VersionNotification	for	draft-patel-dispatch-cpc-oli-parameter-00
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WORLEY, DALE R (DALE) wrote:
> ________________________________________
> From: dispatch-bounces@ietf.org [dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat [pkyzivat@cisco.com]
> 
> But it doesn't alter the reality that the PAI also is intended to make
> an assertion about the *caller* while some of this other information is
> about the *call* rather than the caller.
> 
> For example, how does it make any sense to include
> 
>    * 31 Intercept (trouble) - for calls to directory numbers (DN) that
>         have been manually placed in trouble-busy state by Telco
>         personnel
> _________________________________________
> 
> It's legitimate to add a parameter to the request-URI to modify how the call is routed
> to its destination, and that covers a number of the CPC cases. 

It covers those cases *if* the intended use is to insert those 
properties that describe the caller into the PAI, and those that 
describe the callee into the R-URI or To-URI.

But I don't get the impression that is the intent.
Its explicitly stated the the actual content is intended to be opaque.

> But number 31 above is neither a property of the caller or a modifier of the callee,
> it's a dynamic (!) attribute of the call itself.  Also, there are data-domain problems,
> as presumably the "Intercept (trouble)" status can be combined with any caller-type,
> whereas the value 31 erases any information previously recorded in the CPC.

Yeah, even if you try to sort out those parameters that apply to the 
caller vs those that apply to the callee, then the rest don't fit anywhere.

	Thanks,
	Paul

From pkyzivat@cisco.com  Thu Mar 18 16:48:26 2010
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Cc: dispatch@ietf.org, Liess Laura <L.Liess@telekom.de>
Subject: Re: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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Laura - inline...

Laura Liess wrote:
> Paul,
> 
> thank you for the comments and please excuse my late answer. I brooded
> quite long about your comments.
> I like very much your proposal about the "modifier"-category. I hope
> Alan will agree to the change.
> I talked to people involved in the DT IMS /NGN implementation
> concerning some of the issues.
> 
> Please find the answers/comments in-line.
> 
>>> - Section 4.3.6 on crisis deleted
>> Then why is it still mentioned in REQ-2?
> 
>  I will delete it in the next version.We just missed it when we did
> the change.
> 
>>> - "tone" and "ringback" combination deleted from chapter "Combinations of
>>> URN".
>>> - Chapter "Combination Rules for Alert URN Indications" upgraded. We
>>> hope the rules are  much more simple now when we distinguish between
>>> ring tones and ringback tones.
>> ----------
>>
>> Section 1:
>>
>>   This specification does not change the usage of the SIP Alert-Info-
>>   header defined in the RFC3261.  The Alert-Info-header can be used in
>>   INVITE requests and 180 Ringing responses.
>>
>> IMO it would make sense to make an extension to 3261 to permit this header
>> in any 18x response. This could help resolve issues about what ought to be
>> rendered to the caller in case of 182 or 183.
> 
> This would be very usefull, but can we do it in this draft? Or we need
> to wait for 3261bis?

If I put on my (new) sipcore chair hat then I guess I should know the 
answer to this. But I don't. We aren't planning a bis, but changes are 
being managed. Will you be in Anaheim? Maybe we can corner a few people 
and discuss how this might be done.

>> Section 3:
>>
>>      ...  The left-most
>>      label is called "alert-category" and is separated from right-side
>>      of the alert-identifier, the alert-indication, by a semicolon...
>>
>>         urn:alert:tone:{tone-indication}
>>
>> The text says "semicolon" but the example uses colon, and the ABNF also
>> shows colon.
> 
> Agree. It should be "colon". I will change this.
>> Section 4:
>>
>> The way the word "tone" is used in this section could cause confusion, after
>> having gone to the trouble of explaining that this mechanism is for
>> indicating semantic intent that might not be rendered as tones. Its further
>> confused because section 4.1 PBX Tones are encoded using urn:alert:tone, but
>> that 4.2 Service Tones are encoded using urn:alert:service.
>>
>> I suggest using some other word than "tone" in cases when audio rendering
>> may not be used. For example "indication". Perhaps "caller indication" could
>> be used instead of ringback, and "callee indication" could be used instead
>> of "ring". (Thats in descriptive text. Perhaps the alert-category of "tone"
>> should be changed too, but I'm not ready to propose that yet.)
> 
> Currently, the category "tone" is curently used only for PBX ring
> tones (in case of audio rendering). It is somehow confusing, because
> "service" and "ringback" are also tones. I don't like "tones", either.

I found it quite disconcerting as I was reading.

> I thought about changing it to ring-tone or PBX ring-tone, but I
> didn't do it for the last version because such a change may affect
> implementations of earlier versions of the draft. I think such changes
> need a broad agreement before changing the draft. In particular, we
> need Alan's agreement because the PBX tones is his part.
> 
> Personally, I would prefer "ring indication"  and "ringback
> indication"  to   "caller indication" and  "callee indication". I

I was just looking for *something*, and that works as well as my suggestion.

> think the meaning is more obvious, even if the rendering is not audio.
> In SIP we also habe 180 Ringing too, independent of the rendering. But
> this is just a personal preference, I may be wrong .  We could ask on
> the ML what other people think.
> 
> 
> Another problem I see here is that currently "tone" (or "callee
> indication" or "ring indication") is currently used only for PBX
> ring-tones, not for ring tones in general. Somehow we should enable
> also other kinds of ring tones. We can
> a) delete the restriction to PBX ring-tones and use e.g.
> urn:alert:ring-tone:normal also for public networks or
> b) splitt the "tone" (or which ever new name we choose) category in
> "pbx ring indications", and  "public ring indications" and maybe
> others. People could then define new ring-tone categories in the
> future,  e.g. "3GPP2 ring indications".
> c) define the category as a chain consisting of a main-category and
> sub-categories. , e.g. ring-indication:pbx,  ring-indication:public. .
> The urns would be then urn:alert:ring-tone:pbx:normal,
> urn:alert:ring-tone:pbx:internal,  urn:alert:ring-tone:pbx:external,
> urn:alert:ring-tone:public:normal .
> 
> My preference is c), but then we have to change the ABNF Syntax and
> define the alert-category in chap. 3 as being something like a chain
> like

Or, urn:alert:ring-tone:normal.public, urn:alert:ring-tone:normal.pbx, ...

That way you can trim components off the end until you find something 
you understand.

The trouble with that is of course that it means that all the "pbx" 
extensions are not gathered together. If that is a problem.

Or, maybe "pbx" is a "modifier".

> alert-categoryr= top-alert-category  *(":" alert-subcategory)
>  top-alert-category = "ring-tone"/"ringback"/"service"  or
> "callee-indication"/"caller-indication"/"service"
> alert-subcategory = let-dig [*let-dig-hyp  let-dig]
> 
> I hope I don't run with this into the combination problems again....
> 
> 
>> Sections 4.2.6, 4.2.7, 4.2.8:
>>
>> These talk about "this sub-level". But sub-level isn't defined anywhere.
>> There is "sub-indication" in the ABNF, but the examples given in these
>> sections don't use that.
> 
> "Sub-level" is another relic from the old version. I will replace it
> with "alert indication".
> "Sub-indication"  is no longer used in the current use cases. We (Alan
> and I) considered the issue, but decided not to delete the
> sub-indication yet, until the new format is consolidated.  For the
> examples in the draft, it is still correct, just not used th the
> draft. If we agree to use things like urn:alert:modifier:country:za
> or  urn:alert:ring-tone:pbx:normal we have to change the format
> anyway.
> 
> 
>> These are shown as urn:alert:tone, which means they can only be used for
>> callee indication, not caller indication. Is that intended?
> 
> Yes. This is described in chap 7.2, second paragraph.
> 
>> I know this might open old wounds, but perhaps it would be better to make
>> these values simply be sub-indications on those values where it makes sense.
>> But if you don't want to go there I understand.
> 
> With the sub-indications we had the combination problem.....This is
> why we changed it.
> 
>> Section 4.3:
>>
>> Aren't there also country-specific ring tones? (I know the British ones in
>> the movies are different from the American ones.) Should there be provision
>> for specifying these? See more on this below.
> 
> Up to now, nobody stated the requirement that they are needed.
> I asked peopple within DT and they don't see a need for that. The DT
> ring tone is implemented in each DT end device. (I am not happy with
> it, but that's it.) There is no specific German ring tone, German
> carriers have their own ring tone specification, but this
> specifications are more or less the ITU-T specification on ring tone,
> with some minor differences.
> 
> I think it should be possible in principle to define country ring
> tones, if anyone needs them. With sub-categories we could do this with
> <urn:alert:ring-indication:public:normal><
> urn:alert:modifier:country:za> or just
> <urn:alert:ring-indication:country:za>
> 
> I would avoid defining use cases for which it is very unclear if
> someone needs them. But we should try to make the syntax as fexible as
> possible for later use cases.

Yes, that is what I am pushing at - leaving enough wiggle room for these 
sorts of cases to fit later in a natural way.

>> Why no provision for non-country-specific ringbacks? This is potentially a
>> case for customization - perhaps the callee's presence status could be used
>> by the callee to select an appropriate ringback to the caller.
> 
> It's true. We have to be change the format to allow more than country
> ringback tones. I don't have any clear idea how to do it yet. Maybe
> <urn:alert:ringback-indication:country:za>  or
> <urn:alert:ringback-indication:public:normal><
> urn:alert:modifier:country:za> if we use sub-categories?

It seems to me that there ought to be symmetry for ring and ringback.

> Concerning the presence status...I think this is more complex and
> dedicated presence mechanisms should be used. We may run into privacy
> and all other issues connected to presence. I would define here as a
> use case. But it should be possible that people add this later if
> there is a need for it.

I was only using presence as an example of why you might want other 
sorts of ringbacks. How they would be selected is a whole separate 
discussion.

>> Section 4.4:
>>
>> Are all combinations potentially valid? Or can there be at most one per
>> category?
> 
> One per main category. I think we can state this if we use modifiers.
>> How about having alert:modifier as a category? Then all the other categories
>> could be mutually exclusive, and zero or more modifiers would adjust
>> whatever one was supplied. In that case, priority, zip, and delayed would be
>> modifiers.
> 
> I like the modifier :-). But we have to check with Alan and the
> Mailing list, because of possoible existing implementations.
> 
>  (Maybe normal also ought to be a modifier, and maybe country code
>> should also be a modifier - urn:alert:modifier:country:za.)
> I don't have any opinion yet.

The modifier approach is a different spin on things and needs thinking 
through.

>> Section 5:
>>
>> I think it would be a good idea to mention something about what a UA should
>> do with these in more complex cases:
>> - alert-info received but in-band media also received
>> - an invite was forked and the UAC is getting responses from
>>  multiple early dialogs, containing conflicting alert-info
>>  and/or in-band media.
>>
>> It occurs to me that in some cases early media should preempt the rendering
>> of alert-info, but in other cases it should not. Maybe that should be left
>> to implementor discretion, but maybe there are cases where this should be
>> deterministic - I haven't thought about it enough to have a firm opinion.
> 
> This is one of the basic SIP unsolved problems. I don't think we
> should try to solve here this problem. Every end device, carrier...has
> his own rules. DT already uses urn:alert:service: call-waiting. If a
> UE receives it, the user gets the message "This is a waiting call"  on
> the display if a display is there. We have no conflict with early
> media. Other carrier or end devices can have other rules.
> 
>> Rolling all that up into some overall suggestions:
>>
>> You have some things that only apply to the "ring", some that only apply to
>> the "ringback" and some that apply to both. And then there are things that
>> can act as modifiers to any of the others. But the distinction between ring
>> and ringback is redundant with the message it travels in. So how about:
>>
>> urn:alert:service:xxx has all the things you have as services plus all the
>> things you have as "tones". In an INVITE external/internal describe the
>> caller, and in a 18x they describe the callee. Maybe some additional values
>> that are useful for describing the callee status.
>>
>> urn:alert:modifier:xxx where xxx is priority/zip/delayed or country.cc
>> where cc is a country code.
>>
>> Combinations allowed are one service, and one or more distinct modifiers.
>> (Must be distinct based on top-level.)
>>
> 
> I think this would work and makes things quite simple, at least for
> the use cases we have now.    I suppose if there is no service , but a
> modifier like country.cc, then the default ring or ringback tone are
> assumed.

Right.

> We also must remove the last paragraph on page 8 because
> country.cc is a valid tone but country is not.   Country is here some
> kind of modifier type.
> I see only one problem here, which I don't nknow how to handle.What
> happens when people add new alert-indications? Just add them to the
> "service" or "modifier"-category? If so, we have two unstructured
> lists of alert indications, which may become long with the time. I

It should be possible to add some hierarchy to help organize this.

> suppose proxies may want to remove alert-indications which are not
> relevant for their users, e.g. public carrier would want to remove
> pbx-specific alert-indications generated at the other end of the
> network, e.g. internal/external.  If there is no simple way to
> recognize the alert indication which are not relevant for them, they
> will use white lists and allow exactly the alert identifiers they use
> within their own domain...I don't this is what we want.  But I may be
> wrong with this.

I haven't thought about this. You are ahead of me.

	Thanks,
	Paul

From laura.liess.dt@googlemail.com  Fri Mar 19 05:43:05 2010
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Date: Fri, 19 Mar 2010 13:43:12 +0100
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From: Laura Liess <laura.liess.dt@googlemail.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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2010/3/19 Paul Kyzivat <pkyzivat@cisco.com>:
> Laura - inline...
>
> Laura Liess wrote:
>>
>> Paul,
>>
>> thank you for the comments and please excuse my late answer. I brooded
>> quite long about your comments.
>> I like very much your proposal about the "modifier"-category. I hope
>> Alan will agree to the change.
>> I talked to people involved in the DT IMS /NGN implementation
>> concerning some of the issues.
>>
>> Please find the answers/comments in-line.
>>
>>>> - Section 4.3.6 on crisis deleted
>>>
>>> Then why is it still mentioned in REQ-2?
>>
>> =A0I will delete it in the next version.We just missed it when we did
>> the change.
>>
>>>> - "tone" and "ringback" combination deleted from chapter "Combinations
>>>> of
>>>> URN".
>>>> - Chapter "Combination Rules for Alert URN Indications" upgraded. We
>>>> hope the rules are =A0much more simple now when we distinguish between
>>>> ring tones and ringback tones.
>>>
>>> ----------
>>>
>>> Section 1:
>>>
>>> =A0This specification does not change the usage of the SIP Alert-Info-
>>> =A0header defined in the RFC3261. =A0The Alert-Info-header can be used =
in
>>> =A0INVITE requests and 180 Ringing responses.
>>>
>>> IMO it would make sense to make an extension to 3261 to permit this
>>> header
>>> in any 18x response. This could help resolve issues about what ought to
>>> be
>>> rendered to the caller in case of 182 or 183.
>>
>> This would be very usefull, but can we do it in this draft? Or we need
>> to wait for 3261bis?
>
> If I put on my (new) sipcore chair hat then I guess I should know the ans=
wer
> to this. But I don't. We aren't planning a bis, but changes are being
> managed. Will you be in Anaheim? Maybe we can corner a few people and
> discuss how this might be done.

This would be great. I will be in Anaheim.
>
>>> Section 3:
>>>
>>> =A0 =A0 ... =A0The left-most
>>> =A0 =A0 label is called "alert-category" and is separated from right-si=
de
>>> =A0 =A0 of the alert-identifier, the alert-indication, by a semicolon..=
.
>>>
>>> =A0 =A0 =A0 =A0urn:alert:tone:{tone-indication}
>>>
>>> The text says "semicolon" but the example uses colon, and the ABNF also
>>> shows colon.
>>
>> Agree. It should be "colon". I will change this.
>>>
>>> Section 4:
>>>
>>> The way the word "tone" is used in this section could cause confusion,
>>> after
>>> having gone to the trouble of explaining that this mechanism is for
>>> indicating semantic intent that might not be rendered as tones. Its
>>> further
>>> confused because section 4.1 PBX Tones are encoded using urn:alert:tone=
,
>>> but
>>> that 4.2 Service Tones are encoded using urn:alert:service.
>>>
>>> I suggest using some other word than "tone" in cases when audio renderi=
ng
>>> may not be used. For example "indication". Perhaps "caller indication"
>>> could
>>> be used instead of ringback, and "callee indication" could be used
>>> instead
>>> of "ring". (Thats in descriptive text. Perhaps the alert-category of
>>> "tone"
>>> should be changed too, but I'm not ready to propose that yet.)
>>
>> Currently, the category "tone" is curently used only for PBX ring
>> tones (in case of audio rendering). It is somehow confusing, because
>> "service" and "ringback" are also tones. I don't like "tones", either.
>
> I found it quite disconcerting as I was reading.
>
>> I thought about changing it to ring-tone or PBX ring-tone, but I
>> didn't do it for the last version because such a change may affect
>> implementations of earlier versions of the draft. I think such changes
>> need a broad agreement before changing the draft. In particular, we
>> need Alan's agreement because the PBX tones is his part.
>>
>> Personally, I would prefer "ring indication" =A0and "ringback
>> indication" =A0to =A0 "caller indication" and =A0"callee indication". I
>
> I was just looking for *something*, and that works as well as my suggesti=
on.
>
>> think the meaning is more obvious, even if the rendering is not audio.
>> In SIP we also habe 180 Ringing too, independent of the rendering. But
>> this is just a personal preference, I may be wrong . =A0We could ask on
>> the ML what other people think.
>>
>>
>> Another problem I see here is that currently "tone" (or "callee
>> indication" or "ring indication") is currently used only for PBX
>> ring-tones, not for ring tones in general. Somehow we should enable
>> also other kinds of ring tones. We can
>> a) delete the restriction to PBX ring-tones and use e.g.
>> urn:alert:ring-tone:normal also for public networks or
>> b) splitt the "tone" (or which ever new name we choose) category in
>> "pbx ring indications", and =A0"public ring indications" and maybe
>> others. People could then define new ring-tone categories in the
>> future, =A0e.g. "3GPP2 ring indications".
>> c) define the category as a chain consisting of a main-category and
>> sub-categories. , e.g. ring-indication:pbx, =A0ring-indication:public. .
>> The urns would be then urn:alert:ring-tone:pbx:normal,
>> urn:alert:ring-tone:pbx:internal, =A0urn:alert:ring-tone:pbx:external,
>> urn:alert:ring-tone:public:normal .
>>
>> My preference is c), but then we have to change the ABNF Syntax and
>> define the alert-category in chap. 3 as being something like a chain
>> like
>
> Or, urn:alert:ring-tone:normal.public, urn:alert:ring-tone:normal.pbx, ..=
.
>
> That way you can trim components off the end until you find something you
> understand.
>
> The trouble with that is of course that it means that all the "pbx"
> extensions are not gathered together. If that is a problem.

We need something which helps proxies to decide quickly e.g. if it is
a public or pbx alert indication.
>
> Or, maybe "pbx" is a "modifier".
I would put  "pbx" in both "ring-indication"  fand "modifier"
alert-indications. External/internal are pbx "ring-indication" and
priority/zip/delayed are pbx modifiers.

>
>> alert-categoryr=3D top-alert-category =A0*(":" alert-subcategory)
>> =A0top-alert-category =3D "ring-tone"/"ringback"/"service" =A0or
>> "callee-indication"/"caller-indication"/"service"
>> alert-subcategory =3D let-dig [*let-dig-hyp =A0let-dig]
>>
>> I hope I don't run with this into the combination problems again....
>>
>>
>>> Sections 4.2.6, 4.2.7, 4.2.8:
>>>
>>> These talk about "this sub-level". But sub-level isn't defined anywhere=
.
>>> There is "sub-indication" in the ABNF, but the examples given in these
>>> sections don't use that.
>>
>> "Sub-level" is another relic from the old version. I will replace it
>> with "alert indication".
>> "Sub-indication" =A0is no longer used in the current use cases. We (Alan
>> and I) considered the issue, but decided not to delete the
>> sub-indication yet, until the new format is consolidated. =A0For the
>> examples in the draft, it is still correct, just not used th the
>> draft. If we agree to use things like urn:alert:modifier:country:za
>> or =A0urn:alert:ring-tone:pbx:normal we have to change the format
>> anyway.
>>
>>
>>> These are shown as urn:alert:tone, which means they can only be used fo=
r
>>> callee indication, not caller indication. Is that intended?
>>
>> Yes. This is described in chap 7.2, second paragraph.
>>
>>> I know this might open old wounds, but perhaps it would be better to ma=
ke
>>> these values simply be sub-indications on those values where it makes
>>> sense.
>>> But if you don't want to go there I understand.
>>
>> With the sub-indications we had the combination problem.....This is
>> why we changed it.
>>
>>> Section 4.3:
>>>
>>> Aren't there also country-specific ring tones? (I know the British ones
>>> in
>>> the movies are different from the American ones.) Should there be
>>> provision
>>> for specifying these? See more on this below.
>>
>> Up to now, nobody stated the requirement that they are needed.
>> I asked peopple within DT and they don't see a need for that. The DT
>> ring tone is implemented in each DT end device. (I am not happy with
>> it, but that's it.) There is no specific German ring tone, German
>> carriers have their own ring tone specification, but this
>> specifications are more or less the ITU-T specification on ring tone,
>> with some minor differences.
>>
>> I think it should be possible in principle to define country ring
>> tones, if anyone needs them. With sub-categories we could do this with
>> <urn:alert:ring-indication:public:normal><
>> urn:alert:modifier:country:za> or just
>> <urn:alert:ring-indication:country:za>
>>
>> I would avoid defining use cases for which it is very unclear if
>> someone needs them. But we should try to make the syntax as fexible as
>> possible for later use cases.
>
> Yes, that is what I am pushing at - leaving enough wiggle room for these
> sorts of cases to fit later in a natural way.
>
>>> Why no provision for non-country-specific ringbacks? This is potentiall=
y
>>> a
>>> case for customization - perhaps the callee's presence status could be
>>> used
>>> by the callee to select an appropriate ringback to the caller.
>>
>> It's true. We have to be change the format to allow more than country
>> ringback tones. I don't have any clear idea how to do it yet. Maybe
>> <urn:alert:ringback-indication:country:za> =A0or
>> <urn:alert:ringback-indication:public:normal><
>> urn:alert:modifier:country:za> if we use sub-categories?
>
> It seems to me that there ought to be symmetry for ring and ringback.
Agree.
>
>> Concerning the presence status...I think this is more complex and
>> dedicated presence mechanisms should be used. We may run into privacy
>> and all other issues connected to presence. I would define here as a
>> use case. But it should be possible that people add this later if
>> there is a need for it.
>
> I was only using presence as an example of why you might want other sorts=
 of
> ringbacks. How they would be selected is a whole separate discussion.
>
>>> Section 4.4:
>>>
>>> Are all combinations potentially valid? Or can there be at most one per
>>> category?
>>
>> One per main category. I think we can state this if we use modifiers.
>>>
>>> How about having alert:modifier as a category? Then all the other
>>> categories
>>> could be mutually exclusive, and zero or more modifiers would adjust
>>> whatever one was supplied. In that case, priority, zip, and delayed wou=
ld
>>> be
>>> modifiers.
>>
>> I like the modifier :-). But we have to check with Alan and the
>> Mailing list, because of possoible existing implementations.
>>
>> =A0(Maybe normal also ought to be a modifier, and maybe country code
>>>
>>> should also be a modifier - urn:alert:modifier:country:za.)
>>
>> I don't have any opinion yet.
>
> The modifier approach is a different spin on things and needs thinking
> through.

>
>>> Section 5:
>>>
>>> I think it would be a good idea to mention something about what a UA
>>> should
>>> do with these in more complex cases:
>>> - alert-info received but in-band media also received
>>> - an invite was forked and the UAC is getting responses from
>>> =A0multiple early dialogs, containing conflicting alert-info
>>> =A0and/or in-band media.
>>>
>>> It occurs to me that in some cases early media should preempt the
>>> rendering
>>> of alert-info, but in other cases it should not. Maybe that should be
>>> left
>>> to implementor discretion, but maybe there are cases where this should =
be
>>> deterministic - I haven't thought about it enough to have a firm opinio=
n.
>>
>> This is one of the basic SIP unsolved problems. I don't think we
>> should try to solve here this problem. Every end device, carrier...has
>> his own rules. DT already uses urn:alert:service: call-waiting. If a
>> UE receives it, the user gets the message "This is a waiting call" =A0on
>> the display if a display is there. We have no conflict with early
>> media. Other carrier or end devices can have other rules.
>>
>>> Rolling all that up into some overall suggestions:
>>>
>>> You have some things that only apply to the "ring", some that only appl=
y
>>> to
>>> the "ringback" and some that apply to both. And then there are things
>>> that
>>> can act as modifiers to any of the others. But the distinction between
>>> ring
>>> and ringback is redundant with the message it travels in. So how about:
>>>
>>> urn:alert:service:xxx has all the things you have as services plus all
>>> the
>>> things you have as "tones". In an INVITE external/internal describe the
>>> caller, and in a 18x they describe the callee. Maybe some additional
>>> values
>>> that are useful for describing the callee status.
>>>
>>> urn:alert:modifier:xxx where xxx is priority/zip/delayed or country.cc
>>> where cc is a country code.
>>>
>>> Combinations allowed are one service, and one or more distinct modifier=
s.
>>> (Must be distinct based on top-level.)
>>>
>>
>> I think this would work and makes things quite simple, at least for
>> the use cases we have now. =A0 =A0I suppose if there is no service , but=
 a
>> modifier like country.cc, then the default ring or ringback tone are
>> assumed.
>
> Right.
>
>> We also must remove the last paragraph on page 8 because
>> country.cc is a valid tone but country is not. =A0 Country is here some
>> kind of modifier type.
>> I see only one problem here, which I don't know how to handle.What
>> happens when people add new alert-indications? Just add them to the
>> "service" or "modifier"-category? If so, we have two unstructured
>> lists of alert indications, which may become long with the time. I
>
> It should be possible to add some hierarchy to help organize this.
>
>> suppose proxies may want to remove alert-indications which are not
>> relevant for their users, e.g. public carrier would want to remove
>> pbx-specific alert-indications generated at the other end of the
>> network, e.g. internal/external. =A0If there is no simple way to
>> recognize the alert indication which are not relevant for them, they
>> will use white lists and allow exactly the alert identifiers they use
>> within their own domain...I don't this is what we want. =A0But I may be
>> wrong with this.
>
> I haven't thought about this. You are ahead of me.

There are some, probably few scenarios where filters are necessary.
E.g. an ED outside a PBX network sends "internal", just to get quick
phone access to a person inside an enterprise. The carrier or the
enterprise proxy should remove it, otherwise the callee gets a wrong
information. (maybe this is a security issue?) I would like to have
the structure which allows filtering which does not force filtering
proxies to use white lists consisting of discrete values, other
methods beeing too time consuming.

Let's have a chat in Anaheim about this.

Thanks a lot
Laura






>
> =A0 =A0 =A0 =A0Thanks,
> =A0 =A0 =A0 =A0Paul
>

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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Greger Viken Teigre wrote:
> On 15.03.2010 17:24, Adam Roach wrote:
>> On 3/15/10 11:09, Mar 15, Paul Kyzivat wrote:
>>>
>>>
>>> Adam Roach wrote:
>>>> Greger:
>>>>
>>>> For what it's worth, RFCs 3840 and 3841 (caller prefs/callee caps) 
>>>> provide about 80% of what you're asking for.
>>>
>>> Selecting a *UAS* that has the capabilities the caller envisions 
>>> using (such as video) were certainly among the things that were among 
>>> the intended use cases.
>>>
>>> I'm dubious about their suitability for selecting a route to reach 
>>> the UAS, assuming its the same UAS you are seeking.
>>
>> That's the other 20%. :)
> 
> Yes, the toolset is there. Making it work in a multi-vendor, 
> multi-party, multi-capability UA environment is far away.

I can't follow what this means after a couple of replies.
The Accept-Contact for callerprefs is specific to selecting among 
registered contacts that have hopefully been tagged with callee 
capabilities.

For use by an intermediary, at best it might be able to associate callee 
capabilities to alternative paths (presumably via configuration) and 
then apply the Accept-Contact rules to those to help choose the path.

For instance, I guess a proxy that is choosing between a TDM gateway and 
a "sip trunk" might give the TDM gateway a minimal set of capabilities, 
including voice, and then perhaps give the sip trunk none, which would 
cause it to do presumed matching.

But is anybody going to implement callerprefs in routing proxies?

>>> Also callerprefs are no guarantee (not even a superset) of the 
>>> capabilities that will be used on a call. It is an entirely 
>>> reasonable use case for a UAC to invite with no offer, and then 
>>> negotiate the use of voice and video because the UAS offers that. Its 
>>> also entirely reasonable for a call to start out as audio only, for 
>>> the caller and callee to learn as a result of discussion that both 
>>> are video capable, and then add video to the call.
>>>
>>> As non-voice media become more mainstream, and as e2e sip sessions 
>>> become more feasible, we should be investigating how the 
>>> infrastructure can avoid breaking these kinds of use cases. Nailing 
>>> down the route in a way that precludes use of media not announced at 
>>> call establishment time isn't going in a very helpful direction.
>>
>> Nailing down a route... for signaling? I'm a bit confused about how 
>> that is going to preclude certain media types.
> 
> I'm not sure I'm following this, but a video phone will for example be 
> able to make PSTN calls over one peering relationship, high-def 
> voice/AAC-LD over another, and video over a third. If the original call 
> was to a URI with hunt functionality forking across all three, it's 
> pretty random where the call will end up and any attempt with no offer 
> or renegotiate to video afterwards is not likely to work very well from 
> a user perspective.

And the solution is ...?

IMO its really bad to infer only basic voice except in cases where SDP 
is present in the invite and specifies non-basic media.

*Maybe* callerprefs could help, using the approach I mention above.

My simplistic answer is that if the R-URI is sip:... then the gateway 
ought to be last resort. If the R-URI is tel:... then it means I only 
expect telephone functionality via any sort of protocol that supports 
that, so you should feel free to use the GW as your first choice, if you 
wish.

	Thanks,
	Paul

From lendl@nic.at  Fri Mar 19 13:29:28 2010
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On 18.03.2010 15:58, Daryl Malas wrote:
> Are there any other thoughts on "Speermint-izing" this work?  --Daryl
> 

It's IMHO the right place to go.

btw, you might also want to take a look at
http://tools.ietf.org/id/draft-lendl-enum-sip-lr-00.txt
and the ideas (routing vs. retargetting) referenced there.

I haven't followed up on what happened to rosenberg-sip-ua-loose-route in
the end, but looking at your problem from that PoV might be instructive.

May next week's meeting be a productive one.

/ol
-- 
// Otmar Lendl <lendl@nic.at>, T: +43 1 5056416 - 33, F: - 933 //

From gonzalo.camarillo@ericsson.com  Sun Mar 21 14:39:29 2010
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Hi,

>> This would be very usefull, but can we do it in this draft? Or we need
>> to wait for 3261bis?
> 
> If I put on my (new) sipcore chair hat then I guess I should know the
> answer to this. But I don't. We aren't planning a bis, but changes are
> being managed. Will you be in Anaheim? Maybe we can corner a few people
> and discuss how this might be done.

we have approved a few specs that update RFC 3261. So, if needed, it is
possible to update it without issues a bis draft.

Cheers,

Gonzalo

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Hi Gonzalo, Paul=20

I have uploaded the updated version of the I-D.=20
It can be accessed from:
http://www.ietf.org/id/draft-avasarala-dispatch-comm-div-notification-04
.txt

Please review and comment

Thanks

Regards
Ranjit

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]=20
Sent: Tuesday, March 16, 2010 4:29 PM
To: Avasarala Ranjit-A20990
Cc: Paul Kyzivat; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

Hi Ranjit,

there is a blackout period before IETF meetings in which one cannot
submit drafts. Please, submit this version as soon as the blackout
period is over and have Paul have a look at it so that he can confirm he
is happy with it.

Thanks,

Gonzalo


On 11/03/2010 8:10 PM, Avasarala Ranjit-A20990 wrote:
> Hi All
>=20
> I am not able to upload the updated version (-04) of the I-D due to=20
> some issue with the I-D submission tool - (Says I can submit I-D only=20
> on 2010-03-22).
> Until then I am attaching a copy of the updated version (txt and html=20
> formats).
>=20
> Please review and comment.
>=20
> Thanks
>=20
> Note: Let me know if there is any other way to upload the I-D document

> and will be happy to do it.
>=20
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 02, 2010 7:19 PM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
> I think this all sounds good now.
>=20
>         Thanks,
>         Paul
>=20
> Avasarala Ranjit-A20990 wrote:
>> Inline
>>
>> <Ranjit-Mar2>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, March 02, 2010 2:16 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
>> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review

>> ofdraft-avasarala-dispatch-comm-div-notification-03]
>>
>> inline...
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>> I'm still a little confused here.
>>> If phone is on and subscription is active, then in general=20
>>> notification should be possible, regardless of registration. I=20
>>> realize
>>
>>> IMS doesn't permit that, but from a general standard perspective we=20
>>> shouldn't assume it. The manifestation in this case is that the=20
>>> NOTIFY
>>
>>> would fail and the subscription would eventually be torn down.
>>>
>>> So I think the cases are:
>>> - there is a subscription and notification works
>>> - there is a subscription but notification fails
>>> - there is no subscription.
>>>
>>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as=20
>>> Notification fails, we can say - not sending notification due to (1)

>>> no match of filter criteria (2) phone not registered or outside=20
>>> coverage area due to which there is no connectivity and hence=20
>>> notification cannot be sent Anyway if no subscription, no
>> notification.
>>
>> Are you suggesting that you might have a subscription, and have the=20
>> filter criteria met, and yet not attempt to send a NOTIFY because the

>> phone is not registered or has no network connectivity???
>>
>> That seems wrong. ISTM that either you would remove the subscription,

>> or else you would try to send the NOTIFY and then discover that it
> fails.
>>
>> <Ranjit-Mar2> yes the subscription could be removed when the device=20
>> is
>=20
>> not reachable.
>>
>> OTOH, this probably ought not to be within scope of what is discussed

>> in this draft.
>>
>> <Ranjit-Mar2> yes true not within the scope of this draft.
>>
>>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>>> contacts: sip:line1@alice.office.atlanta.com and=20
>>>> sip:line2@alice.home.atlanta.com and
>>>>     both the phones ring, then it would be a case of parallel
>> forking.
>>>> Now say one or both the phones have set diversion rules, to divert=20
>>>> the
>>> In this case, what does it mean for "one or both phones have set=20
>>> diversion rules"? Does that somehow imply distinct rules for each
>> phone?
>>> I don't see how that could work. Aren't the diversion rules for the=20
>>> *AOR*?
>>>
>>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there=20
>>> is
>>
>>> a diversion rule set for the AoR and when diversions occur, the=20
>>> notifications are Sent to the AoR.
>>
>> Not sent to the AoR - sent to the remote contact of the dialog for=20
>> each subscription to the AoR. Same point as below that you agreed
> with.
>>
>> <Ranjit-Mar2> Yes agree with u.
>>
>>> I see they could be set/changed from either phone. But its a single=20
>>> set of rules isn't it? So I might set the rules from the office, and

>>> then cancel them from home.
>>>
>>> <Ranjit-Feb26-2> yes single set and could be set from either of the=20
>>> phones. True.
>>>
>>>> call to say
>>>>     sip:voicemail@alice.atlanta.com, then the notifications would=20
>>>> get
>>
>>>> generated with respect to both the contacts.
>>> Again I'm confused by the terminology you use, and how it relates=20
>>> operationally. The diversion happens on the AOR, and the=20
>>> subscriptions
>>
>>> are on the AOR, and the notifications are sent to the subscribers. I

>>> don't see where the registered contacts enter into the picture at
> all.
>>>
>>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>>
>>> If you have something else in mind, I hope you can explain it better

>>> in the draft.
>>>
>>>> [4] Yes the value of N is configurable and depends on how many=20
>>>> diversions the subscribe wants to keep or could be dependent on=20
>>>> server policy (some maximum
>>>>     number). If the server policy is to send notifications as and=20
>>>> when they occur and not keep any divsersions information, then the=20
>>>> value of last N
>>>>     diversions will be zero. But if the server chooses to retain=20
>>>> the
>=20
>>>> last diversion always, then the value of N will be 1 and so on.
>>> All is straightforward for N>=3D1.
>>>
>>> For N=3D0 things are messy. If we are modeling this as state, and=20
>>> notifications are about state changes, then the new diversion is=20
>>> added
>>
>>> to the (previously empty) state and that triggers notifications to=20
>>> all
>>
>>> current subscriptions. Then if you don't want to retain even this=20
>>> much
>>
>>> state (because N=3D0) you immediately remove that version from the
>> state.
>>> But that is again a state change, so you end up with another=20
>>> notification of that state change.
>>>
>>> <Ranjit-Feb26-2> True. Actually the notifications are for informing=20
>>> the subscriber of the call diversions that happen. The notifications

>>> are not for informing the subscriber of state changes in the=20
>>> notifier
>=20
>>> or the number of diversions that occurred. As I mentioned in the=20
>>> abstract as well as 24.604, the main intent of CDIVN service is to=20
>>> inform the subscribers of communication diversions that occur on=20
>>> their
>>
>>> behalf in the network and this information will help them manage=20
>>> their
>>
>>> rules better.
>>
>> I am not certain, but think we may still be talking past one another=20
>> to some extent.
>>
>> The state can be considered just a formalism for defining the service

>> clearly, regardless of whether it has intrinsic value to the CDIV=20
>> service. But it *is* a formalism. So if it is introduced, then its=20
>> behavior can't be ignored when inconvenient. Defining the state with=20
>> N=3D0 leads to some undesired behavior. The simplest way around that =
is

>> to insist that N be >=3D1.
>>
>> <Ranjit-Mar2> Ok in that case we can say that value of N >=3D1 and =
the=20
>> notifier has always maintains history of last diversion at a minimu.
>> It can maintain more than 1 too depending on the policy.
>>
>> If its important to you that no state be retained once a notification

>> has been sent about the most recent diversion, then you will have to=20
>> find some other way to specify the service, that still meets the=20
>> expectations of 3265. I don't know if there is anything there that=20
>> Adam would consider valid. We will have to get his opinion on that.
>>
>> <Ranjit-Mar2> In that case I am OK maintaining the hostory of=20
>> diversions that occurred as part of state information and retaining=20
>> that state even after the notification for that diversion is sent -=20
>> so
>=20
>> that in case I do not get a confirmation, I would be able to re-send
> the notification.
>> Also maintaining state of diversions would help when the=20
>> notifications
>=20
>> need to sent only at a particular time (if fikter criteria mentions
> it).
>>
>>
>>       Thanks,
>>       Paul
>>
>>> <Ranjit-Feb26-2> But if we want to include the information related=20
>>> to
>=20
>>> how many diversions have happened till time (say from 12 AM ) till=20
>>> the
>>
>>> time the notification is being sent, then as you say we can store=20
>>> the
>=20
>>> diversions. This would be useful when the subscriber specifies in=20
>>> the
>=20
>>> filter criteria to send notifications only during certain time like=20
>>> between 5 and 6 PM and not whenever a diversion occurs. Then N is=20
>>> grater than 1.
>>
>>> I presume you don't want that behavior. If you can live with N>=3D1=20
>>> then
>>
>>> we don't have the problem at all. If you need to support N=3D0 and=20
>>> don't
>>
>>> want the extra notify, then we have to figure out if there is some=20
>>> trick to how things are modeled to get the desired effect. I've=20
>>> suggested something previously, but Adam didn't like it.
>>>
>>> This needs sorting out.
>>>
>>>      Thanks,
>>>      Paul
>>


From fluffy@cisco.com  Tue Mar 23 10:03:59 2010
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Subject: [dispatch] Fwd: Last Call: draft-lawrence-sipforum-user-agent-config (Session Initiation Protocol (SIP) User Agent Configuration) to Informational RFC
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As discussed at the SIP Forum lunch yesterday, there has been =
considerable work on a specification for configuration of SIP User =
Agents that is reflected in this document. I believe this document is =
appropriate for progression as AD Sponsored thought it may still need a =
few changes and discussion of community consensus. To initiate this, I =
have requests a  4 week last call.=20

Please, do give this document a read, and as request in the Last Call, =
send the email to the ietf@ietf.org list.=20

Thanks,=20
Cullen <RAI AD>



Begin forwarded message:

> From: The IESG <iesg-secretary@ietf.org>
> Date: March 23, 2010 9:12:48 AM PDT
> To: IETF-Announce <ietf-announce@ietf.org>
> Subject: Last Call: draft-lawrence-sipforum-user-agent-config (Session =
Initiation Protocol (SIP) User Agent Configuration) to Informational RFC
> Reply-To: ietf@ietf.org
>=20
> The IESG has received a request from an individual submitter to =
consider=20
> the following document:
>=20
> - 'Session Initiation Protocol (SIP) User Agent Configuration '
>   <draft-lawrence-sipforum-user-agent-config-00.txt> as an =
Informational RFC
>=20
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action.  Please send substantive comments to =
the
> ietf@ietf.org mailing lists by 2010-04-20. Exceptionally,=20
> comments may be sent to iesg@ietf.org instead. In either case, please=20=

> retain the beginning of the Subject line to allow automated sorting.
>=20
> The file can be obtained via
> =
http://www.ietf.org/internet-drafts/draft-lawrence-sipforum-user-agent-con=
fig-00.txt
>=20
>=20
> IESG discussion can be tracked via
> =
https://datatracker.ietf.org/public/pidtracker.cgi?command=3Dview_id&dTag=3D=
19859&rfc_flag=3D0
>=20
> _______________________________________________
> IETF-Announce mailing list
> IETF-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf-announce


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



From krehor@cisco.com  Tue Mar 23 20:00:54 2010
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Thread-Topic: dispatch Digest, Vol 12, Issue 47
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From HKaplan@acmepacket.com  Wed Mar 24 00:43:47 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Adam Roach <adam@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 24 Mar 2010 03:43:27 -0400
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Adam Roach
> Sent: Monday, March 15, 2010 1:33 PM
>=20
> Daryl is working within the strictures of reproducing the business
> relationships of the PSTN in a SIP network. And the path that has been
> taken, for better or worse, involves the heavy use of SBCs and other
> devices that impose a rather strict control on the way things are done.
> These devices largely ensure that the media path follows the signaling
> path, even when that's not the optimal thing to do.=20

Optimal for whom, and in what way? =20

> The chances of such
> networks deploying video, hi-def voice, or any of the other really neat
> things that Greger mentions are slim to none. This, of course, is NOT
> Daryl's fault. But it's where the egress-route work comes from.

You seem to have some very odd (and typical IETF bull) preconceptions about=
 what SBC's can do or are capable of doing.  SBC's in general support video=
, HD voice, or any of the other neat things Gregor mentions.  Policies prov=
isioned on the SBC's may limit what media one can do, or in some cases if t=
ranscoding has to happen then obviously you're limited to whatever codecs t=
he transcoders can handle (as any media gateway would be), but that's reall=
y tangential.

=20
> Greger is speaking more towards the network you seem to have in mind --
> one in which end-to-end SIP calls are possible, without SBCs or similar
> control boxes clobbering new services and mangling the ability to convey
> novel media from one endpoint to another. Networks where we don't need
> to define alternate connection models for protocols simply to
> accommodate network elements that do things that are explicitly
> disallowed by protocol specification.

As a B2BUA, what SBC's do is in no way disallowed by the protocols in the I=
ETF. (although certainly I don't really care if they are ;)


> Which is why I support a simple, non-normative, informational draft
> about the form of URLs that one might want to put in ENUM. It satisfies
> the requirements of the stagnant world, and doesn't even have any
> normative impacts on any protocol anywhere. It's harmless, and lets the
> people in the vibrant world concentrate on solving the problem in a more
> general way without having to accommodate the peculiarities of network
> elements living in the stagnant world.

If you live in this "vibrant world" the rest of us don't, what problems are=
 there that need solving??  Usually one needs users and a population >1 to =
have problems.  The future always looks rosy compared to the present.

-hadriel

From HKaplan@acmepacket.com  Wed Mar 24 01:13:35 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Henry Sinnreich <henry.sinnreich@gmail.com>, Dean Willis <dean.willis@softarmor.com>, Adam Roach <adam@nostrum.com>
Date: Wed, 24 Mar 2010 04:12:30 -0400
Thread-Topic: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)
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Not to argue your general tone, but you do realize that most of those thing=
s you list were actually generally profitable and successful in their time,=
 right?  Sure their time may be past, but that doesn't mean they were failu=
res.  I mean I used to be in the IP routing world before coming to voip, an=
d SONET was the predominant long-haul transport which actually connected an=
d formed the "Internet" in the 90's and early 2000's.  I dunno what it is m=
ostly today, but I bet there's still tons of it out there.

And it's not like there weren't plenty of ultimate-failure companies buildi=
ng Ethernet switches or IP routers as well.  Or even HTTP browsers.

-hadriel
p.s. and sadly H.323 ain't dead yet either - for some reason it seems to be=
 alive and kicking in certain regions of the World, and in certain niche ma=
rket segments.


________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Henry Sinnreich
Sent: Wednesday, March 17, 2010 12:19 AM
To: Dean Willis; Adam Roach
Cc: dispatch@ietf.org; Daryl Malas
Subject: Re: [dispatch] War of the Worlds (was Re: I-D Action: draft-malas-=
dispatch-sip-egress-route-00)

This really sums up the enigma why the IETF should do work that contradicts=
 every principle of the Internet architecture and its engineering.

It may be useful to remember the  _solid_   track record of the  "stagnant =
world"

 *   OSI
 *   X4.00 mail
 *   IN
 *   AIN
 *   ISDN
 *   SONET/SDH
 *   ATM
 *   BISDN
 *   H.323
 *   current re-incarnations such as PSTN/SIP and other (avoiding flames he=
re)...
How many companies have disappeared building this stuff? Layoffs. Capital d=
estruction.

The other enigma is the tolerance to associate the IETF with the reincarnat=
ions of these failures.

>I have long speculated that perhaps the best thing we can do is punt the "=
stagnant world" stuff to a different forum.
>I seem to recall many times suggesting that SIP should just be moved to th=
e ITU so we can get on with building the Internet.
>I'd happily add ENUM to the same bucket.

Well said.

Thanks, Henry


On 3/16/10 5:49 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

On Mar 16, 2010, at 6:24 PM, Adam Roach wrote:

I disagree profoundly with characterizing Rosenberg's Modest Proposal as an
appropriate summary of the issues, as it gets the analysis precisely wrong.
It proposes completely abandoning the vibrant world as a lost cause, and
embracing the stagnant world as the only path forward. This isn't just
allowing collateral damage to the vibrant world for the sake of expediency
in stagnant world solutions -- it's a suggestion that we nuke the vibrant
world so it doesn't keep annoying people trying to develop stagnant
solutions.

I have long speculated that perhaps the best thing we can do is punt the "s=
tagnant world" stuff to a different forum. I seem to recall many times sugg=
esting that SIP should just be moved to the ITU so we can get on with build=
ing the Internet. I'd happily add ENUM to the same bucket.

The problem is that we might wake up one day and realize that the Internet =
has been turned off, and all we have left is a stagnant backwater of teleco=
mmunications boundaries established by fat-and-happy politicos eager to pre=
serve their own domains.

The alternative, perhaps, is to embrace stagnation. Then add chlorine.

--
Dean
________________________________
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Not to argue your general tone, but yo=
u do
realize that most of those things you list were actually generally profitab=
le
and successful in their time, right?&nbsp; Sure their time may be past, but
that doesn&#8217;t mean they were failures.&nbsp; I mean I used to be in th=
e IP
routing world before coming to voip, and SONET was the predominant long-hau=
l
transport which actually connected and formed the &#8220;Internet&#8221; in=
 the
90&#8217;s and early 2000&#8217;s.&nbsp; I dunno what it is mostly today, b=
ut I
bet there&#8217;s still tons of it out there.&nbsp; <o:p></o:p></span></fon=
t></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>And it&#8217;s not like there weren&#8=
217;t
plenty of ultimate-failure companies building Ethernet switches or IP route=
rs as
well. &nbsp;Or even HTTP browsers.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>-hadriel<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>p.s. and sadly H.323 ain&#8217;t dead =
yet
either &#8211; for some reason it seems to be alive and kicking in certain
regions of the World, and in certain niche market segments.<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Henry Sinnreich<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, March 17, 2=
010
12:19 AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Dean Willis; Adam Roach<=
br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> <st1:PersonName w:st=3D"=
on">dispatch@ietf.org</st1:PersonName>;
Daryl Malas<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [dispatch] War =
of the
Worlds (was Re: I-D Action: draft-malas-dispatch-sip-egress-route-00)</span=
></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D4 face=3DCalibri><span style=3D'font-size=
:13.0pt;
font-family:Calibri'>This really sums up the enigma why the IETF should do =
work
that contradicts every principle of the Internet architecture and its
engineering.<br>
<br>
It may be useful to remember the &nbsp;_solid_ &nbsp;&nbsp;track record of =
the
&nbsp;&quot;stagnant world&quot;</span></font><o:p></o:p></p>

<ul type=3Ddisc>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>OSI </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>X4.00 mail </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>IN </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>AIN </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>ISDN </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>SONET/SDH </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>ATM </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>BISDN </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>H.323 </span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D4 face=3DCalibri><span style=3D'=
font-size:
     13.0pt;font-family:Calibri'>current re-incarnations such as PSTN/SIP a=
nd
     other (avoiding flames here)...</span></font><o:p></o:p></li>
</ul>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D4 face=3DC=
alibri><span
style=3D'font-size:13.0pt;font-family:Calibri'>How many companies have
disappeared building this stuff? Layoffs. Capital destruction.<br>
<br>
The other enigma is the tolerance to associate the IETF with the reincarnat=
ions
of these failures.<br>
<br>
&gt;I have long speculated that perhaps the best thing we can do is punt th=
e
&quot;stagnant world&quot; stuff to a different forum. <br>
&gt;I seem to recall many times suggesting that SIP should just be moved to=
 the
ITU so we can get on with building the Internet. <br>
&gt;I'd happily add ENUM to the same bucket.<br>
<br>
Well said.<br>
<br>
Thanks, Henry<br>
<br>
<br>
On 3/16/10 5:49 PM, &quot;Dean Willis&quot; &lt;<a
href=3D"dean.willis@softarmor.com">dean.willis@softarmor.com</a>&gt; wrote:=
</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D4 face=3DCalibri><span style=3D'font-size=
:13.0pt;
font-family:Calibri'><br>
On Mar 16, 2010, at 6:24 PM, Adam Roach wrote:</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D4 face=3DC=
alibri><span
style=3D'font-size:13.0pt;font-family:Calibri'><br>
I disagree profoundly with characterizing <st1:City w:st=3D"on"><st1:place =
w:st=3D"on">Rosenberg</st1:place></st1:City>'s
Modest Proposal as an<br>
appropriate summary of the issues, as it gets the analysis precisely wrong.=
<br>
It proposes completely abandoning the vibrant world as a lost cause, and<br=
>
embracing the stagnant world as the only path forward. This isn't just<br>
allowing collateral damage to the vibrant world for the sake of expediency<=
br>
in stagnant world solutions -- it's a suggestion that we nuke the vibrant<b=
r>
world so it doesn't keep annoying people trying to develop stagnant<br>
solutions.</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:13.0pt'><font size=3D4 face=3DC=
alibri><span
style=3D'font-size:13.0pt;font-family:Calibri'><br>
I have long speculated that perhaps the best thing we can do is punt the
&quot;stagnant world&quot; stuff to a different forum. I seem to recall man=
y
times suggesting that SIP should just be moved to the ITU so we can get on =
with
building the Internet. I'd happily add ENUM to the same bucket.<br>
<br>
The problem is that we might wake up one day and realize that the Internet =
has
been turned off, and all we have left is a stagnant backwater of
telecommunications boundaries established by fat-and-happy politicos eager =
to
preserve their own domains.<br>
<br>
The alternative, perhaps, is to embrace stagnation. Then add chlorine.<br>
<br>
--<br>
Dean<o:p></o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D4
face=3DCalibri><span style=3D'font-size:13.0pt;font-family:Calibri'>

<hr size=3D3 width=3D"95%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal><font size=3D4 face=3DConsolas><span style=3D'font-siz=
e:13.0pt;
font-family:Consolas'>_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf=
.org/mailman/listinfo/dispatch</a></span></font><o:p></o:p></p>

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From john.elwell@siemens-enterprise.com  Wed Mar 24 06:32:51 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "dispatch@ietf.org" <dispatch@ietf.org>
Date: Wed, 24 Mar 2010 14:31:51 +0100
Thread-Topic: [dispatch] Fwd: Last Call: draft-lawrence-sipforum-user-agent-config (Session Initiation	Protocol (SIP) User Agent Configuration) to Informational RFC
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Subject: Re: [dispatch] Fwd: Last Call:	draft-lawrence-sipforum-user-agent-config (Session Initiation	Protocol (SIP) User Agent Configuration) to Informational RFC
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Also related to this is:
http://www.ietf.org/id/draft-lawrence-dispatch-sipforum-provider-alias-00.t=
xt

"This document defines procedures for how a SIP User Agent can translate an=
 identifier provided by a user into the SIP DNS domain name of a service pr=
ovider.  The form of the user supplied identifier is constrained to values =
that can be entered using only a standard 12 key telephone keypad."

This draft, from the SIP Forum, is an extension to the draft Cullen just an=
nounced IETF last call on. It allows a SIP user agent to accept a provider =
alias number (PAN) entered by the user and translate it to the domain name =
at which the configuration service for configuring the UA can be found.

The SIP Forum would like DISPATCH to consider ways of progressing this draf=
t.

John


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: 23 March 2010 10:04
> To: rai@ietf.org; dispatch@ietf.org
> Subject: [dispatch] Fwd: Last Call:=20
> draft-lawrence-sipforum-user-agent-config (Session Initiation=20
> Protocol (SIP) User Agent Configuration) to Informational RFC
>=20
>=20
> As discussed at the SIP Forum lunch yesterday, there has been=20
> considerable work on a specification for configuration of SIP=20
> User Agents that is reflected in this document. I believe=20
> this document is appropriate for progression as AD Sponsored=20
> thought it may still need a few changes and discussion of=20
> community consensus. To initiate this, I have requests a  4=20
> week last call.=20
>=20
> Please, do give this document a read, and as request in the=20
> Last Call, send the email to the ietf@ietf.org list.=20
>=20
> Thanks,=20
> Cullen <RAI AD>
>=20
>=20
>=20
> Begin forwarded message:
>=20
> > From: The IESG <iesg-secretary@ietf.org>
> > Date: March 23, 2010 9:12:48 AM PDT
> > To: IETF-Announce <ietf-announce@ietf.org>
> > Subject: Last Call:=20
> draft-lawrence-sipforum-user-agent-config (Session Initiation=20
> Protocol (SIP) User Agent Configuration) to Informational RFC
> > Reply-To: ietf@ietf.org
> >=20
> > The IESG has received a request from an individual=20
> submitter to consider=20
> > the following document:
> >=20
> > - 'Session Initiation Protocol (SIP) User Agent Configuration '
> >   <draft-lawrence-sipforum-user-agent-config-00.txt> as an=20
> Informational RFC
> >=20
> > The IESG plans to make a decision in the next few weeks,=20
> and solicits
> > final comments on this action.  Please send substantive=20
> comments to the
> > ietf@ietf.org mailing lists by 2010-04-20. Exceptionally,=20
> > comments may be sent to iesg@ietf.org instead. In either=20
> case, please=20
> > retain the beginning of the Subject line to allow automated sorting.
> >=20
> > The file can be obtained via
> >=20
> http://www.ietf.org/internet-drafts/draft-lawrence-sipforum-us
> er-agent-config-00.txt
> >=20
> >=20
> > IESG discussion can be tracked via
> >=20
> https://datatracker.ietf.org/public/pidtracker.cgi?command=3Dvie
> w_id&dTag=3D19859&rfc_flag=3D0
> >=20
> > _______________________________________________
> > IETF-Announce mailing list
> > IETF-Announce@ietf.org
> > https://www.ietf.org/mailman/listinfo/ietf-announce
>=20
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From pkyzivat@cisco.com  Wed Mar 24 07:39:46 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Hadriel,

SBCs may be *capable* of preserving everything, but AFAIK in large part 
they exist as policy enforcement points to restrict one thing or another.

If there is only one such boundary between the caller and the callee, 
then the chances of everything working that the two ends want may be 
pretty good, and if something doesn't work there is a reasonable chance 
that policy changes can be made so that it will work.

But each additional hop between the caller and callee increases the 
possibility for some loss in functionality. By the time you have the 
caller, the caller's SP, the callee's SP, the callee, and maybe one or 
more transit providers, the likelihood of much beyond basic voice 
working starts getting very questionable. Of course its still 
*possible*, but what is the likelihood?

	Thanks,
	Paul

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Adam Roach
>> Sent: Monday, March 15, 2010 1:33 PM
>>
>> Daryl is working within the strictures of reproducing the business
>> relationships of the PSTN in a SIP network. And the path that has been
>> taken, for better or worse, involves the heavy use of SBCs and other
>> devices that impose a rather strict control on the way things are done.
>> These devices largely ensure that the media path follows the signaling
>> path, even when that's not the optimal thing to do. 
> 
> Optimal for whom, and in what way?  
> 
>> The chances of such
>> networks deploying video, hi-def voice, or any of the other really neat
>> things that Greger mentions are slim to none. This, of course, is NOT
>> Daryl's fault. But it's where the egress-route work comes from.
> 
> You seem to have some very odd (and typical IETF bull) preconceptions about what SBC's can do or are capable of doing.  SBC's in general support video, HD voice, or any of the other neat things Gregor mentions.  Policies provisioned on the SBC's may limit what media one can do, or in some cases if transcoding has to happen then obviously you're limited to whatever codecs the transcoders can handle (as any media gateway would be), but that's really tangential.
> 
>  
>> Greger is speaking more towards the network you seem to have in mind --
>> one in which end-to-end SIP calls are possible, without SBCs or similar
>> control boxes clobbering new services and mangling the ability to convey
>> novel media from one endpoint to another. Networks where we don't need
>> to define alternate connection models for protocols simply to
>> accommodate network elements that do things that are explicitly
>> disallowed by protocol specification.
> 
> As a B2BUA, what SBC's do is in no way disallowed by the protocols in the IETF. (although certainly I don't really care if they are ;)
> 
> 
>> Which is why I support a simple, non-normative, informational draft
>> about the form of URLs that one might want to put in ENUM. It satisfies
>> the requirements of the stagnant world, and doesn't even have any
>> normative impacts on any protocol anywhere. It's harmless, and lets the
>> people in the vibrant world concentrate on solving the problem in a more
>> general way without having to accommodate the peculiarities of network
>> elements living in the stagnant world.
> 
> If you live in this "vibrant world" the rest of us don't, what problems are there that need solving??  Usually one needs users and a population >1 to have problems.  The future always looks rosy compared to the present.
> 
> -hadriel
> 

From HKaplan@acmepacket.com  Wed Mar 24 10:05:55 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 24 Mar 2010 13:06:09 -0400
Thread-Topic: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Sure, then what he's saying is "because of policies people want to employ a=
nd enforce, I can't do things that don't comply with those policies".  Sure=
 I believe that... it's a tautology.

With respect to media policies though, as far as I can tell, restricting th=
e type of media, or even codecs within a given media type, is not _that_ po=
pular - not for endpoint-to-endpoint sessions.  I bet maybe as much as 50% =
of the time, the codec/media-type munging by SBC's is done for fixing inter=
op, not restriction policies.  They're frequently removed because the equip=
ment _inside_ the carrier's core network (i.e., gateways and media servers)=
 can't handle anything else and don't act well when given SDP they don't ex=
pect.

But if the signaling ends up going back out of the core, like to another UA=
 or another domain, some SBC's can "undo" the changes they made (assuming o=
perator policies don't restrict that too).

Although admittedly I may be wrong on how frequently it's done for interop,=
 vs. billing/policies, vs. bandwidth issues.

-hadriel

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, March 24, 2010 10:40 AM
> To: Hadriel Kaplan
> Cc: Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route=
-
> 00
>=20
> Hadriel,
>=20
> SBCs may be *capable* of preserving everything, but AFAIK in large part
> they exist as policy enforcement points to restrict one thing or another.
>=20
> If there is only one such boundary between the caller and the callee,
> then the chances of everything working that the two ends want may be
> pretty good, and if something doesn't work there is a reasonable chance
> that policy changes can be made so that it will work.
>=20
> But each additional hop between the caller and callee increases the
> possibility for some loss in functionality. By the time you have the
> caller, the caller's SP, the callee's SP, the callee, and maybe one or
> more transit providers, the likelihood of much beyond basic voice
> working starts getting very questionable. Of course its still
> *possible*, but what is the likelihood?
>=20
> 	Thanks,
> 	Paul
>=20

From adam.uzelac@gmail.com  Wed Mar 24 10:14:15 2010
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Date: Wed, 24 Mar 2010 10:14:13 -0700
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From: Adam Uzelac <adam.uzelac@gmail.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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I can confirm the notion that Hadriel states below.  The "munging" that's
going on is more about facilitating the session than about restricting a
particular type of session.  With that said, there are indeed times that we
(as a SSP) would want to restrict, but those use cases are much fewer than
sorting out Interop issues.

Adam Uzelac
Global Crossing

On Wed, Mar 24, 2010 at 10:06 AM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>
> Sure, then what he's saying is "because of policies people want to employ
> and enforce, I can't do things that don't comply with those policies".  Sure
> I believe that... it's a tautology.
>
> With respect to media policies though, as far as I can tell, restricting
> the type of media, or even codecs within a given media type, is not _that_
> popular - not for endpoint-to-endpoint sessions.  I bet maybe as much as 50%
> of the time, the codec/media-type munging by SBC's is done for fixing
> interop, not restriction policies.  They're frequently removed because the
> equipment _inside_ the carrier's core network (i.e., gateways and media
> servers) can't handle anything else and don't act well when given SDP they
> don't expect.
>
> But if the signaling ends up going back out of the core, like to another UA
> or another domain, some SBC's can "undo" the changes they made (assuming
> operator policies don't restrict that too).
>
> Although admittedly I may be wrong on how frequently it's done for interop,
> vs. billing/policies, vs. bandwidth issues.
>
> -hadriel
>
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: Wednesday, March 24, 2010 10:40 AM
> > To: Hadriel Kaplan
> > Cc: Adam Roach; dispatch@ietf.org
> > Subject: Re: [dispatch] I-D Action:
> draft-malas-dispatch-sip-egress-route-
> > 00
> >
> > Hadriel,
> >
> > SBCs may be *capable* of preserving everything, but AFAIK in large part
> > they exist as policy enforcement points to restrict one thing or another.
> >
> > If there is only one such boundary between the caller and the callee,
> > then the chances of everything working that the two ends want may be
> > pretty good, and if something doesn't work there is a reasonable chance
> > that policy changes can be made so that it will work.
> >
> > But each additional hop between the caller and callee increases the
> > possibility for some loss in functionality. By the time you have the
> > caller, the caller's SP, the callee's SP, the callee, and maybe one or
> > more transit providers, the likelihood of much beyond basic voice
> > working starts getting very questionable. Of course its still
> > *possible*, but what is the likelihood?
> >
> >       Thanks,
> >       Paul
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--0016361e7efe2b541604828f0f5d
Content-Type: text/html; charset=ISO-8859-1
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I can confirm the notion that Hadriel states below.=A0 The &quot;munging&qu=
ot; that&#39;s going on is more about facilitating the session than about r=
estricting a particular type of session.=A0 With that said, there are indee=
d times that we (as a SSP) would want to restrict, but those use cases are =
much fewer than sorting out Interop issues.<br>
<br>Adam Uzelac<br>Global Crossing<br><br><div class=3D"gmail_quote">On Wed=
, Mar 24, 2010 at 10:06 AM, Hadriel Kaplan <span dir=3D"ltr">&lt;<a href=3D=
"mailto:HKaplan@acmepacket.com">HKaplan@acmepacket.com</a>&gt;</span> wrote=
:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt 0.8ex; borde=
r-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><br>
Sure, then what he&#39;s saying is &quot;because of policies people want to=
 employ and enforce, I can&#39;t do things that don&#39;t comply with those=
 policies&quot;. =A0Sure I believe that... it&#39;s a tautology.<br>
<br>
With respect to media policies though, as far as I can tell, restricting th=
e type of media, or even codecs within a given media type, is not _that_ po=
pular - not for endpoint-to-endpoint sessions. =A0I bet maybe as much as 50=
% of the time, the codec/media-type munging by SBC&#39;s is done for fixing=
 interop, not restriction policies. =A0They&#39;re frequently removed becau=
se the equipment _inside_ the carrier&#39;s core network (i.e., gateways an=
d media servers) can&#39;t handle anything else and don&#39;t act well when=
 given SDP they don&#39;t expect.<br>

<br>
But if the signaling ends up going back out of the core, like to another UA=
 or another domain, some SBC&#39;s can &quot;undo&quot; the changes they ma=
de (assuming operator policies don&#39;t restrict that too).<br>
<br>
Although admittedly I may be wrong on how frequently it&#39;s done for inte=
rop, vs. billing/policies, vs. bandwidth issues.<br>
<font color=3D"#888888"><br>
-hadriel<br>
</font><div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: Paul Kyzivat [mailto:<a href=3D"mailto:pkyzivat@cisco.com">pkyzi=
vat@cisco.com</a>]<br>
&gt; Sent: Wednesday, March 24, 2010 10:40 AM<br>
&gt; To: Hadriel Kaplan<br>
&gt; Cc: Adam Roach; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org=
</a><br>
&gt; Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-ro=
ute-<br>
&gt; 00<br>
&gt;<br>
</div><div class=3D"im">&gt; Hadriel,<br>
&gt;<br>
&gt; SBCs may be *capable* of preserving everything, but AFAIK in large par=
t<br>
&gt; they exist as policy enforcement points to restrict one thing or anoth=
er.<br>
&gt;<br>
&gt; If there is only one such boundary between the caller and the callee,<=
br>
&gt; then the chances of everything working that the two ends want may be<b=
r>
&gt; pretty good, and if something doesn&#39;t work there is a reasonable c=
hance<br>
&gt; that policy changes can be made so that it will work.<br>
&gt;<br>
&gt; But each additional hop between the caller and callee increases the<br=
>
&gt; possibility for some loss in functionality. By the time you have the<b=
r>
&gt; caller, the caller&#39;s SP, the callee&#39;s SP, the callee, and mayb=
e one or<br>
&gt; more transit providers, the likelihood of much beyond basic voice<br>
&gt; working starts getting very questionable. Of course its still<br>
&gt; *possible*, but what is the likelihood?<br>
&gt;<br>
&gt; =A0 =A0 =A0 Thanks,<br>
&gt; =A0 =A0 =A0 Paul<br>
&gt;<br>
</div><div><div></div><div class=3D"h5">___________________________________=
____________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</div></div></blockquote></div><br>

--0016361e7efe2b541604828f0f5d--

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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Ooh! A gauntlet! It's been a while since we had one of these. I guess we 
all need a bit of catharsis.

On 3/24/10 12:43 AM, Hadriel Kaplan wrote:
>    
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Adam Roach
>> Sent: Monday, March 15, 2010 1:33 PM
>>
>> Daryl is working within the strictures of reproducing the business
>> relationships of the PSTN in a SIP network. And the path that has been
>> taken, for better or worse, involves the heavy use of SBCs and other
>> devices that impose a rather strict control on the way things are done.
>> These devices largely ensure that the media path follows the signaling
>> path, even when that's not the optimal thing to do.
>>      
> Optimal for whom, and in what way?

I guess I should have qualified that. I meant technically optimal -- 
that is, optimizing for the lowest end-to-end latency, jitter, and 
overall network resource utilization. IP routing does a reasonable job 
of getting this right -- artificially pinning the media to various 
points around the network defeats that.

>> The chances of such
>> networks deploying video, hi-def voice, or any of the other really neat
>> things that Greger mentions are slim to none. This, of course, is NOT
>> Daryl's fault. But it's where the egress-route work comes from.
>>      
> You seem to have some very odd (and typical IETF bull) preconceptions about what SBC's can do or are capable of doing.

Defensive much? I didn't say that the SBCs were incapable of passing 
this kind of media along. I was attempting to point out that the 
operators who deploy these kinds of architectures are not typically of 
the mindset that lends itself to forward thinking. After all, they're 
sinking untold costs into a network that does quite precisely what their 
old network did. If that qualifies as "progress," then something as 
radical as hi-def audio on the same scale would probably qualify as 
"Ragnarök."

That said, if SBCs actually *are* B2BUAs as you claim below (which they 
aren't, but let's buy that conceit for a second), then they must 
implement all higher-level services that the endpoints they sit between 
expect to invoke. Because, if they really are B2BUAs, then they're the 
"other user agent" in two different (but related) calls, and they need 
to understand everything that the actual endpoints say. That means that 
handsets will always be stuck waiting on SBCs to catch up when they try 
to deploy new services.

>
>    
>> Greger is speaking more towards the network you seem to have in mind --
>> one in which end-to-end SIP calls are possible, without SBCs or similar
>> control boxes clobbering new services and mangling the ability to convey
>> novel media from one endpoint to another. Networks where we don't need
>> to define alternate connection models for protocols simply to
>> accommodate network elements that do things that are explicitly
>> disallowed by protocol specification.
>>      
> As a B2BUA, what SBC's do is in no way disallowed by the protocols in the IETF. (although certainly I don't really care if they are ;)
>    

Again, even if they were true B2BUAs, they run afoul of some rather 
well-established Internet principles. I know that things like the 
end-to-end principle seem like quaint platitudes to you, and I'm not the 
person to convince you otherwise.

But that's not the practical issue. You're smart enough to know that 
SBCs are in no way compliant B2BUAs. They don't satisfy the requirements 
of user agents, nor do they hold themselves to the restrictions of being 
a proxy. They pick-and-choose between these behaviors, sometimes within 
a single message. That's out of spec, and it causes real and 
quantifiable problems.

For example: if SBCs were actual, real, full-fledged back-to-back user 
agents, we wouldn't be stuck with defining alternative connection models 
to accommodate their peculiarities. As compliant user agents, they would 
have no issues with the MSRP protocol as defined.


>> Which is why I support a simple, non-normative, informational draft
>> about the form of URLs that one might want to put in ENUM. It satisfies
>> the requirements of the stagnant world, and doesn't even have any
>> normative impacts on any protocol anywhere. It's harmless, and lets the
>> people in the vibrant world concentrate on solving the problem in a more
>> general way without having to accommodate the peculiarities of network
>> elements living in the stagnant world.
>>      
> If you live in this "vibrant world" the rest of us don't, what problems are there that need solving??  Usually one needs users and a population>1 to have problems.  The future always looks rosy compared to the present.
>    

Although I'm sometimes guilty of reading through large backlogs of email 
and replying to messages as I come across them, I have generally taken 
it upon myself to at least read to the end of a thread before I reply in 
that thread. This lets me avoid asking questions that have already been 
answered or making assertions that have already been refuted.

http://www.ietf.org/mail-archive/web/dispatch/current/msg01549.html

(You can skip directly to the final paragraph)

/a

From dean.willis@softarmor.com  Wed Mar 24 11:08:18 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Hadriel Kaplan wrote:
> 
>> -----Original Message----- From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Adam Roach Sent:
>> Monday, March 15, 2010 1:33 PM
>> 
>> Daryl is working within the strictures of reproducing the business 
>> relationships of the PSTN in a SIP network. And the path that has
>> been taken, for better or worse, involves the heavy use of SBCs and
>> other devices that impose a rather strict control on the way things
>> are done. These devices largely ensure that the media path follows
>> the signaling path, even when that's not the optimal thing to do.
> 
> Optimal for whom, and in what way?

Optimal in the "internet" sense: the shortest/least used available
route, as would be described by the IP routing topology from address A to B.

> 
>> The chances of such networks deploying video, hi-def voice, or any
>> of the other really neat things that Greger mentions are slim to
>> none. This, of course, is NOT Daryl's fault. But it's where the
>> egress-route work comes from.
> 
> You seem to have some very odd (and typical IETF bull) preconceptions
> about what SBC's can do or are capable of doing.  SBC's in general
> support video, HD voice, or any of the other neat things Gregor
> mentions.  Policies provisioned on the SBC's may limit what media one
> can do, or in some cases if transcoding has to happen then obviously
> you're limited to whatever codecs the transcoders can handle (as any
> media gateway would be), but that's really tangential.

I believe Adam is referring to the fact that it's the old-school telecom
people who are building this sort of topology. They don't have a great
history of successful innovation. They occasionally try to offer a new
service, but only after it's been much more successfully deployed in an
"Internet" fashion by the new-school people. Does Youtube require SBCs?
How about iTunes? AOL's Instant messenger? Facebook?


> 
> 
>> Greger is speaking more towards the network you seem to have in
>> mind -- one in which end-to-end SIP calls are possible, without
>> SBCs or similar control boxes clobbering new services and mangling
>> the ability to convey novel media from one endpoint to another.
>> Networks where we don't need to define alternate connection models
>> for protocols simply to accommodate network elements that do things
>> that are explicitly disallowed by protocol specification.
> 
> As a B2BUA, what SBC's do is in no way disallowed by the protocols in
> the IETF. (although certainly I don't really care if they are ;)

Right. What they do is controlled by the policies and mindsets of those
who purchase and install them. They live in a different world, where
innovation is slow and painful. It's also a world where long-term
reliability and stability are heavily rewarded.

> 
> 
>> Which is why I support a simple, non-normative, informational draft
>>  about the form of URLs that one might want to put in ENUM. It
>> satisfies the requirements of the stagnant world, and doesn't even
>> have any normative impacts on any protocol anywhere. It's harmless,
>> and lets the people in the vibrant world concentrate on solving the
>> problem in a more general way without having to accommodate the
>> peculiarities of network elements living in the stagnant world.
> 
> If you live in this "vibrant world" the rest of us don't, what
> problems are there that need solving??  Usually one needs users and a
> population >1 to have problems.  The future always looks rosy
> compared to the present.

Why are the kids migrating towards Facebook/Youtube etc. as their
communications channels?

--
Dean

From dean.willis@softarmor.com  Wed Mar 24 11:10:45 2010
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Subject: Re: [dispatch] Fwd: Last	Call:	draft-lawrence-sipforum-user-agent-config (Session	Initiation	Protocol (SIP) User Agent Configuration) to	Informational RFC
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Elwell, John wrote:
> Also related to this is: 
> http://www.ietf.org/id/draft-lawrence-dispatch-sipforum-provider-alias-00.txt
> 
> 
> "This document defines procedures for how a SIP User Agent can
> translate an identifier provided by a user into the SIP DNS domain
> name of a service provider.  The form of the user supplied identifier
> is constrained to values that can be entered using only a standard 12
> key telephone keypad."
> 
> This draft, from the SIP Forum, is an extension to the draft Cullen
> just announced IETF last call on. It allows a SIP user agent to
> accept a provider alias number (PAN) entered by the user and
> translate it to the domain name at which the configuration service
> for configuring the UA can be found.
> 
> The SIP Forum would like DISPATCH to consider ways of progressing
> this draft.
> 

This Provider Alias Number sounds like an ITAD identifier which sounds
like a SPID.

One ought to be able to do an ENUM style query on this number and
retrieve a URI.

--
Dean

From xmlscott@gmail.com  Wed Mar 24 12:40:46 2010
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On Wed, 2010-03-24 at 13:10 -0500, Dean Willis wrote:

> This Provider Alias Number sounds like an ITAD identifier which sounds
> like a SPID.

The idea was inspired by ITADs, but the task group felt that actually
using the ITAD name and registry would not meet the needs of the
intended use (that is, high reliability consumer-facing service
providers).

> One ought to be able to do an ENUM style query on this number and
> retrieve a URI.

Which is, roughly speaking, what the draft proposes: do a NAPTR lookup
to translate the PAN into a SIP domain name, which is then used as
specified in the UA Config spec to find the configuration service for
the provider.



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On Wed, 2010-03-24 at 14:31 +0100, Elwell, John wrote:
> Also related to this is:
> http://www.ietf.org/id/draft-lawrence-dispatch-sipforum-provider-alias-00.txt
> 
> "This document defines procedures for how a SIP User Agent can
> translate an identifier provided by a user into the SIP DNS domain
> name of a service provider.  The form of the user supplied identifier
> is constrained to values that can be entered using only a standard 12
> key telephone keypad."
> 
> This draft, from the SIP Forum, is an extension to the draft Cullen
> just announced IETF last call on. It allows a SIP user agent to accept
> a provider alias number (PAN) entered by the user and translate it to
> the domain name at which the configuration service for configuring the
> UA can be found.

To be extra clear... the provider-alias draft is _not_ being last
called.   For various reasons, this part of what the SIP Forum Task
Group came up with has been separated from the rest to be progressed
independently, hence:

> The SIP Forum would like DISPATCH to consider ways of progressing this
> draft.




From hgs@cs.columbia.edu  Wed Mar 24 13:26:53 2010
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On Mar 24, 2010, at 3:41 PM, Scott Lawrence wrote:

> On Wed, 2010-03-24 at 13:10 -0500, Dean Willis wrote:
>=20
>> This Provider Alias Number sounds like an ITAD identifier which =
sounds
>> like a SPID.
>=20
> The idea was inspired by ITADs, but the task group felt that actually
> using the ITAD name and registry would not meet the needs of the
> intended use (that is, high reliability consumer-facing service
> providers).

Could you expand on the reasoning? Do you see a problem with the =
accuracy of the data set, the ability to filter registrations to keep =
out riff-raff or more operational issues (e.g., operation of a DNS =
registry)?

Henning=

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To: Adam Roach <adam@nostrum.com>
Date: Wed, 24 Mar 2010 16:56:16 -0400
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> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]
> Sent: Wednesday, March 24, 2010 1:33 PM
>
> Ooh! A gauntlet! It's been a while since we had one of these. I guess we
> all need a bit of catharsis.
>
> On 3/24/10 12:43 AM, Hadriel Kaplan wrote:
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> >> Behalf Of Adam Roach
> >> Sent: Monday, March 15, 2010 1:33 PM
> >>
> > Optimal for whom, and in what way?
>=20
> I guess I should have qualified that. I meant technically optimal --
> that is, optimizing for the lowest end-to-end latency, jitter, and
> overall network resource utilization. IP routing does a reasonable job
> of getting this right -- artificially pinning the media to various
> points around the network defeats that.

Oh, I didn't realize typical IP routing took into account latency, jitter, =
packet-loss and overall network utilization.  Oh wait, that's because it *d=
oesn't*.  MPLS-TE does more so, and its extensions for ospf/isis-te - but t=
here's this funny thing that happens with that - ALL the traffic flows that=
 way (well, all the traffic for that FEC/prefix).  So if you only want *som=
e* traffic to flow that (better) way, like oh for example money-generating =
voip traffic, you gotta make it a different FEC (which SBC's do), or apply =
router-level per-flow/per-call gates/policies.  Obviously going across the =
public Internet there's nothing much better you can do than best-effort rou=
ting, but for owners/operators of the network infrastructures, they have ot=
her options. YMMV.


> > You seem to have some very odd (and typical IETF bull) preconceptions
> about what SBC's can do or are capable of doing.
>=20
> Defensive much? I didn't say that the SBCs were incapable of passing
> this kind of media along. I was attempting to point out that the
> operators who deploy these kinds of architectures are not typically of
> the mindset that lends itself to forward thinking. After all, they're
> sinking untold costs into a network that does quite precisely what their
> old network did. If that qualifies as "progress," then something as
> radical as hi-def audio on the same scale would probably qualify as
> "Ragnar=F6k."

Actually, most operators *want* new capabilities, new services, new media t=
ypes, etc.  They're stuck in many cases because of legacy deployed gear, no=
 doubt, but they really do want to provide better and newer stuff.

=20
> That said, if SBCs actually *are* B2BUAs as you claim below (which they
> aren't, but let's buy that conceit for a second), then they must
> implement all higher-level services that the endpoints they sit between
> expect to invoke. Because, if they really are B2BUAs, then they're the
> "other user agent" in two different (but related) calls, and they need
> to understand everything that the actual endpoints say. That means that
> handsets will always be stuck waiting on SBCs to catch up when they try
> to deploy new services.

Ironically, SBC's seem to be the ones usually waiting on endpoints to imple=
ment new things. (seriously)


> That said, if SBCs actually *are* B2BUAs as you claim below (which they
> aren't, but let's buy that conceit for a second), then they must
> implement all higher-level services that the endpoints they sit between
> expect to invoke. Because, if they really are B2BUAs, then they're the
> "other user agent" in two different (but related) calls, and they need
> to understand everything that the actual endpoints say. That means that
> handsets will always be stuck waiting on SBCs to catch up when they try
> to deploy new services.

Of course you're right, in some particulars SBC's are not a "full" B2BUA, b=
ut instead walk a fine line trying to be less than a "full" B2BUA, specific=
ally to try and avoid getting in the way of new mechanisms/media/protocols.=
  But I have not seen any RFC that specifies what a B2BUA must or must not =
do - yes, it's the combination of a UAC and UAS for two distinct sessions, =
but I don't know of any doc that says what I can and cannot copy back/forth=
 among those two sessions.  And that is not specific to SBC's either - lots=
 of app-servers/softswitches/pbx's are "mostly" but not entirely B2BUA's in=
 signaling, and partial or not B2BUA's in media.


> For example: if SBCs were actual, real, full-fledged back-to-back user
> agents, we wouldn't be stuck with defining alternative connection models
> to accommodate their peculiarities. As compliant user agents, they would
> have no issues with the MSRP protocol as defined.

You're not stuck with it.  We don't *need* the alternative connection model=
 or msrp changes to function.  We *want* it, for an optimization - in the i=
pv4/v6 case to let media go "direct" when it can, or not be converted back/=
forth v6/v4/v6; in the MSRP case so it can actually scale and be cost-effec=
tive.  They're more pragmatic/practical issues than technical.


> Although I'm sometimes guilty of reading through large backlogs of email
> and replying to messages as I come across them, I have generally taken
> it upon myself to at least read to the end of a thread before I reply in
> that thread. This lets me avoid asking questions that have already been
> answered or making assertions that have already been refuted.
>=20
> http://www.ietf.org/mail-archive/web/dispatch/current/msg01549.html
>=20
> (You can skip directly to the final paragraph)

I read it, and it refutes nothing.  I didn't literally mean a population of=
 "1".  I meant a significant population, with numerous vendors, etc. (and I=
 think you knew that ;)=20

-hadriel
p.s. Is it sad that when I read "gauntlet" I first thought of a great class=
ic arcade game? :)


From HKaplan@acmepacket.com  Wed Mar 24 14:02:59 2010
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To: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 24 Mar 2010 17:03:12 -0400
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, March 24, 2010 2:08 PM
> To: Hadriel Kaplan
>=20
> Optimal in the "internet" sense: the shortest/least used available
> route, as would be described by the IP routing topology from address A to
> B.

Unless something drastic has changed in the past 6 years, the public Intern=
et routing model does not provide a shortest in the English sense of the wo=
rd.  It is not based on physical distance or latency.  Nor is it based on l=
east-used.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Wed, 24 Mar 2010 17:20:59 -0400
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, March 24, 2010 2:08 PM
> To: Hadriel Kaplan
>=20
> I believe Adam is referring to the fact that it's the old-school telecom
> people who are building this sort of topology. They don't have a great
> history of successful innovation. They occasionally try to offer a new
> service, but only after it's been much more successfully deployed in an
> "Internet" fashion by the new-school people. Does Youtube require SBCs?
> How about iTunes? AOL's Instant messenger? Facebook?

The application, protocol, delivery, usage, and expectation model for those=
 applications is very different from VoIP.  And you know that. AOL-IM is cl=
oser, but it's a closed walled-garden... how evil!

> Why are the kids migrating towards Facebook/Youtube etc. as their
> communications channels?

What does that that have to do with anything?  They're not replacements/ins=
tead-of - they're something different altogether.  Why do more kids play Xb=
ox Live and talk over the game headsets?  It must be because it's not provi=
ded by a telco!?

Why do more and more kids use SMS, or mobile phones?  Those are provided by=
 telco's.

-hadriel


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On Wed, 2010-03-24 at 13:10 -0500, Dean Willis wrote:
> >> This Provider Alias Number sounds like an ITAD identifier which sounds
> >> like a SPID.

On Mar 24, 2010, at 3:41 PM, Scott Lawrence wrote:

> > The idea was inspired by ITADs, but the task group felt that actually
> > using the ITAD name and registry would not meet the needs of the
> > intended use (that is, high reliability consumer-facing service
> > providers).

On Wed, 2010-03-24 at 16:27 -0400, Henning Schulzrinne wrote:

> Could you expand on the reasoning? Do you see a problem with the
> accuracy of the data set, the ability to filter registrations to keep
> out riff-raff or more operational issues (e.g., operation of a DNS
> registry)?

The attached is a problem statement I drafted for the SIP Forum Board
that summarized the issues raised in the task group and from some
external reviewers.




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The User Agent Configuration Task Group has included among the goals
for the Recommendation it is producing the ability to configure a User
Agent for use with a SIP domain whose administration and operation is
independent of the network to which the UA is connected.  The Task
Group believe this requirement is important in many deployment
scenarios, but a simple example serves to illustrate the challenge:

    A user wants to obtain home phone service through a particular
    commercial Internet Telephony Service Provider (ITSP).  The user
    establishes an account with the ITSP and then purchases a phone
    (UA) at an independent retail store.  The user connects that new
    UA to the home network, but our users network connection is
    provided and provisioned by a carrier that provides generic IP
    connectivity without any telephony service - the carrier is not
    affiliated with the ITSP in any way.  When the new UA is connected
    to the network, it must find the Configuration Service of the ITSP
    in order to request configuration data.  The process of locating
    that Configuration Service (as defined in the draft User Agent
    Configuration Recommendation) requires that the UA obtain the SIP
    domain name of the ITSP.  In other deployment scenarios, the UA
    can automate obtaining that SIP domain name by beginning with the
    assumption that the SIP domain is the same as the default DNS
    domain of the local network, but in this case the local network
    will provide a domain name that does not have a SIP UA
    Configuration Service, so the location procedure will fail.

A simple answer to this problem would be to require that the UA
provide a means for the user to enter the domain name of the ITSP, and
the draft Recommendation does allow that solution.  However, the
working group believes that entry of a domain name is not as simple
and user-friendly enough.  Entry of a domain name requires alphabetic
characters that a simple phone would not otherwise need, and very soon
internationalized domain names will require even more complex
characters.

The current draft calls for a mechanism that allows the user to enter
a simple string of decimal digits that the UA can use to obtain the
SIP domain name.  This solution was inspired by (and the current draft
calls for using) IP Telephony Administrative Domain (ITAD) numbers as
the numeric values.

ITAD numbers were defined as a part of RFC 3219: Telephony Routing
over IP (TRIP).  TRIP is not related to SIP and has not been widely
implemented or deployed.  In fact, the only element of the TRIP
specification that has been used at all is the ITAD registry at IANA.
The ITAD registry has been adopted as the numeric domain alias
mechanism of an experimental Internet telephony addressing scheme
based on ITAD Subscriber Numbers (ISNs), coordinated through the
freenum.org domain.  The 'ISN trial' is supported by the Internet2
Project, Packet Clearing House, and MIT.  ISN addressing uses dial
strings of the form <local-digits>*<isn>; the 'Freenum/ISN
Cookbook' [1] defines a resolution procedure that uses a DNS NAPTR
lookup of a regular expression to transform the dial string into a
SIP URL sip:<local-digits>@<domain>.

The Task Group has reached a consensus that it should explore an
alternative to ITAD as a basis for the numeric domain aliasing
mechanism in the SIP Forum Recommendation.  Reasons for this consensus
include:

   * The administrative and legal infrastructure of using the existing
     first-come first-served ITAD registry at IANA is not sufficient
     for the needs of commercial service providers.

   * Using the existing ITAD name and registry may create confusion
     regarding whether or not other parts of the protocols and
     procedures in the TRIP specification and/or the ISN/Freenum trial
     are also applicable (the intent is that no other part of these
     other specifications and procedures be used).

   * The numeric alias to domain resolution procedure in the current
     UA Configuration draft Recommendation has two problems:

     1. It uses the 'freenum.org' dns tree, which the Task Group
        believes is not scalable and robust enough to meet the needs
        of commercial SIP service providers.

     2. It is designed to take as input a full user address that
        contains an ITAD number, and produce as output a full SIP url
        (including a userpart).  The UA Configuration Recommendation
        needs as input only the domain number and as output only the
        domain name, which means that the current DNS records and ISN
        Cookbook procedures cannot be used.

   * The ITAD number is defined as a 32 bit number.  The UA
     Configuration Recommendation needs to map one arbitrary (but
     numeric) identifier to a domain name, so there is no reason to
     impose a limit of a particular number of bits as ITAD does; what
     is needed is a string of decimal digits, not a binary number of a
     particular size.

The Task Group would like the Board to consider whether or not the SIP
Forum should create a new mechanism for numeric domain aliases.  This
new mechanism would:

   * Use a DNS domain for resolution whose ownership and change
     control procedures are appropriate to the needs of SIP service
     providers and enterprises.

   * Allow the definition of a simple DNS based mapping procedure
     using only the hostname portion of the NATPR result to specify
     the domain.

[1] http://freenum.org/cookbook/

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From hgs@cs.columbia.edu  Wed Mar 24 14:37:32 2010
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Date: Wed, 24 Mar 2010 17:37:44 -0400
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>=20
> The attached is a problem statement I drafted for the SIP Forum Board
> that summarized the issues raised in the task group and from some
> external reviewers.

To quote:

The administrative and legal infrastructure of using the existing
     first-come first-served ITAD registry at IANA is not sufficient
     for the needs of commercial service providers.

* No reasons are given for this determination.

 Using the existing ITAD name and registry may create confusion
     regarding whether or not other parts of the protocols and
     procedures in the TRIP specification and/or the ISN/Freenum trial
     are also applicable (the intent is that no other part of these
     other specifications and procedures be used).

* I admit that I find this speculative. As long as the mechanism is =
clear, the users of the DNS record won't care.

The ITAD number is defined as a 32 bit number.

* This translates into (roughly) a 9-digit decimal integer. It seems =
rather unlikely that we would exceed 4 billion service providers or that =
we'd want longer identifiers than 9 digits.

In general, I agree that the ITAD registry would have to be expanded and =
the use clarified, but re-use of the same registry (say, enterprise =
numbers) for different purposes is not unusual.

Henning


--Apple-Mail-15--898311244
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><blockquote type=3D"cite"><div><font class=3D"Apple-style-span" =
color=3D"#000000"><br></font>The attached is a problem statement I =
drafted for the SIP Forum Board<br>that summarized the issues raised in =
the task group and from some<br>external =
reviewers.<br></div></blockquote></div><br><div>To =
quote:</div><div><br></div><div><div>The administrative and legal =
infrastructure of using the existing</div><div>&nbsp;&nbsp; &nbsp; =
first-come first-served ITAD registry at IANA is not =
sufficient</div><div>&nbsp;&nbsp; &nbsp; for the needs of commercial =
service providers.</div><div><br></div><div>* No reasons are given for =
this determination.</div><div><br></div><div><div>&nbsp;Using the =
existing ITAD name and registry may create =
confusion</div><div>&nbsp;&nbsp; &nbsp; regarding whether or not other =
parts of the protocols and</div><div>&nbsp;&nbsp; &nbsp; procedures in =
the TRIP specification and/or the ISN/Freenum =
trial</div><div>&nbsp;&nbsp; &nbsp; are also applicable (the intent is =
that no other part of these</div><div>&nbsp;&nbsp; &nbsp; other =
specifications and procedures be used).</div><div><br></div><div>* I =
admit that I find this speculative. As long as the mechanism is clear, =
the users of the DNS record won't care.</div><div><br></div><div>The =
ITAD number is defined as a 32 bit number.</div><div><br></div><div>* =
This translates into (roughly) a 9-digit decimal integer. It seems =
rather unlikely that we would exceed 4 billion service providers or that =
we'd want longer identifiers than 9 digits.</div><div><br></div><div>In =
general, I agree that the ITAD registry would have to be expanded and =
the use clarified, but re-use of the same registry (say, enterprise =
numbers) for different purposes is not =
unusual.</div><div><br></div><div>Henning</div><div><br></div></div></div>=
</body></html>=

--Apple-Mail-15--898311244--

From pkyzivat@cisco.com  Wed Mar 24 14:41:03 2010
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Subject: Re: [dispatch] I-D Action: draft-malas-dispatch-sip-egress-route-00
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Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Adam Roach [mailto:adam@nostrum.com]
>> Sent: Wednesday, March 24, 2010 1:33 PM
>>
>> Ooh! A gauntlet! It's been a while since we had one of these. I guess we
>> all need a bit of catharsis.
>>
>> On 3/24/10 12:43 AM, Hadriel Kaplan wrote:
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>>>> Behalf Of Adam Roach
>>>> Sent: Monday, March 15, 2010 1:33 PM
>>>>
>>> Optimal for whom, and in what way?
>> I guess I should have qualified that. I meant technically optimal --
>> that is, optimizing for the lowest end-to-end latency, jitter, and
>> overall network resource utilization. IP routing does a reasonable job
>> of getting this right -- artificially pinning the media to various
>> points around the network defeats that.
> 
> Oh, I didn't realize typical IP routing took into account latency, jitter, packet-loss and overall network utilization.  Oh wait, that's because it *doesn't*.  MPLS-TE does more so, and its extensions for ospf/isis-te - but there's this funny thing that happens with that - ALL the traffic flows that way (well, all the traffic for that FEC/prefix).  So if you only want *some* traffic to flow that (better) way, like oh for example money-generating voip traffic, you gotta make it a different FEC (which SBC's do), or apply router-level per-flow/per-call gates/policies.  Obviously going across the public Internet there's nothing much better you can do than best-effort routing, but for owners/operators of the network infrastructures, they have other options. YMMV.

RSVP?

>>> You seem to have some very odd (and typical IETF bull) preconceptions
>> about what SBC's can do or are capable of doing.
>>
>> Defensive much? I didn't say that the SBCs were incapable of passing
>> this kind of media along. I was attempting to point out that the
>> operators who deploy these kinds of architectures are not typically of
>> the mindset that lends itself to forward thinking. After all, they're
>> sinking untold costs into a network that does quite precisely what their
>> old network did. If that qualifies as "progress," then something as
>> radical as hi-def audio on the same scale would probably qualify as
>> "Ragnarök."
> 
> Actually, most operators *want* new capabilities, new services, new media types, etc.  They're stuck in many cases because of legacy deployed gear, no doubt, but they really do want to provide better and newer stuff.

They want the new capabilities that they can monetize.
But if its some new capability that can be realized by cooperation by 
the two ends, with just a "dumb pipe" connection, then there is no 
motivation to remove inhibitions.

And don't get me wrong - I think its a fine thing for SPs to monetize 
added value services that really add value that the callers want enough 
to pay for.

	Thanks,
	Paul

From pkyzivat@cisco.com  Thu Mar 25 11:50:16 2010
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Cc: dispatch@ietf.org, Gerold Pinker <Gerold.Pinker@telekom.de>, Liess Laura <L.Liess@telekom.de>
Subject: Re: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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I've been thinking about how the "namespace" of alert-info URNs might be 
reorganized. My goals in doing so are:
- allow the different "aspects" of the alert to be specified
- make it clear what the combination rules are for these
- allow for comparable expressiveness for rings and ringbacks
- provide room for extension

The following is a rough outline of a what I have in mind:

The urn syntax is as in the current draft, *except*:

alert-category  = "reason" / "priority" / "source" /
                   "duration" / "delay" / "locale" /
                   extension-category
extension-category = token

The principle for choosing these categories is that they are orthogonal. 
Hence there is a single rule for well-formedness of Alert-Info:
there can be at most one instance of each alert-category, and in 
principle any such combination is valid. (Though some may be silly or 
uninteresting.)

The above doesn't distinguish ring from ringback. That is to be 
determined from context - when present in a request it describes the 
ring. When present in a response it describes the ringback. Again, in 
principle any combination can be used for either ring or ringback, 
though some combinations may be silly or uninteresting.

The exact way in which the various categories are combined for rendering 
is left as an implementation issue. The implementation is free to ignore 
any or all at its whim.

Its my intent that things previously discussed in the draft can all be 
mapped onto these categories. Here is my swag at doing so:

urn:alert:reason:

- normal (default)
- call-waiting
- forward
- auto-callback
- recall.hold
- recall.transfer
- private.<private-name>

urn:alert:priority:

- normal (default)
- low
- high
- crisis
- private.<private-name>

urn:alert:source:

- unclassified (default)
- internal
- external
- friend
- family
- private.<private-name>

urn:alert:duration:

- normal (default)
- short (or could be "zip")
- long (for completeness, or omit if unneeded)
- private.<private-name>

urn:alert:delay:

- none (default)
- yes (not sure what to put here)
- private.<private-name>

urn:alert:locale:

- default (default)
- country.<ISO 3166-1 country code>
- private.<private-name>

In the above the "private.<private-name>" syntax is for extensions 
specific to independent organizations. The <private-name> is in the form 
of a "reverse FQDN" such as is used for Java package names. This gives a 
way of assigning unique names without the need for a new registry. The 
namespace for each alert category is independent. Those assigning new 
names must ensure they are in a position to assign names uniquely for 
the FQDN they choose.

(I have considerable reservation about introducing such an uncontrolled 
extension mechanism. I see how it can provide a lot of flexibility in 
providing tailored user experiences. But it can also lead to vendor lock 
in if a device is only capable of supporting a particular vendor's 
specific extensions. So I think this needs a lot of discussion.)

For example, I might want to define:

urn:alert:source:private.com.cisco.customer

If I were to do so, I should first ensure that I speak for cisco.com 
when coining this.

Additions to the values above other than via the "private" mechanism 
would be by standards action. The same for adding new categories.
(There could be a "private" mechanism for categories too. But I'm not at 
all confident that is a good thing.)

I'm not quite sure about "call-waiting" as a "reason" - it doesn't seem 
quite right. The reason could still be any of those other things. For 
the ring indication its far from clear to me that it is needed at all - 
the recipient already knows it is on a call, and doesn't need to be told 
that. It may make more sense for the ringback - in that case you are 
being told something about the state of the other party. Perhaps this 
should be replaced by something about the state of the party sending the 
message. E.g.

urn:alert:state:

- unknown
- waiting (in a request, means the caller is waiting
            for the call to complete. The normal case.)
- on-a-call (in a response, means the called party
              is on another call - the call waiting case.
              But could apply to a caller as well.)
- away (in a request, means the call has been generated
         automatically and the user will be summoned if/when
         the call completes. This is the converse of waiting)

(This "state" is really a form of presence. Whether its good to encode 
it here, rather than actually transmitting pidf, is an interesting 
question.)

How all of this might be mapped onto actual renderings is probably not 
something that should be standardized, but it might be worth including 
some possibilities as examples. One that comes to mind is:

(Assuming all rendering are audio):

- the device has a set of possible clips it can render
- each is attributed with the values it matches for each
   category above. (none, one, several, or all for each
   category.)

The device just finds the maximal match and renders that one.

This might be augmented if there are different sorts of renderings - 
e.g. audio, visual, tactile (vibrarion). In that case it might seek out 
a match for each sort and render them all.

	Thanks,
	Paul

From laura.liess.dt@googlemail.com  Thu Mar 25 18:26:41 2010
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From: Laura Liess <laura.liess.dt@googlemail.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Laura Liess'" <laura.liess.dt@googlemail.com>
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Date: Fri, 26 Mar 2010 02:26:51 +0100
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Cc: dispatch@ietf.org, "'Martin.Huelsemann@telekom.de'"@core3.amsl.com, "'R.Jesske@telekom.de'"@core3.amsl.com, 'Liess Laura' <L.Liess@telekom.de>, 'Gerold Pinker' <Gerold.Pinker@telekom.de>
Subject: Re: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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Paul,

Thank you for the proposal and the elaborate description. I definitively
like it.
It seems to solve the combinations problem, it is extensible and easy to
understand.=20
I have two comments on it, connected to "call-waiting".

1) It seems to me that there is some misunderstanding concerning the =
meaning
of the "call-waiting" alert urn. It is not the callee's presence state, =
but
the state of the call itself. The "call-waiting" tone tells the caller:
"Your call is temporary queued (waiting) at the callee's side because =
the
callee is currently involved in another call. The callee was informed =
that
your call is waiting." Having this information, the caller can decide to
wait for a while or to cancel the call. This is an already existing
PSTN-service which customers expect us to provide also when we migrate =
to
SIP.
Because this urn tells the user that the service "call-waiting" was
performed at the other side of the call, we name "service" was choosed =
for
this category.=20

2) There are already implementations of urn:alert:service:call-waiting. =
This
syntax is used in the 3GPP Specification for the call-waiting service =
(TS
24.615 "Communication Waiting (CW) using IP Multimedia (IM) Core Network
(CN) subsystem"). If we change this name, people either change their
software, which means money and time for them, or they leave it as it =
is,
which means lack of interoperability e2e. The "reason"-category contains =
in
principles names for services, so I think it does not harm the proposal =
if
we just replace "reason" by "service" and keep the interoperability with
existing implementations.

Thanks a lot
Laura


-----Urspr=FCngliche Nachricht-----
Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Gesendet: Donnerstag, 25. M=E4rz 2010 19:51
An: Laura Liess
Cc: alan.b.johnston@gmail.com; dispatch@ietf.org; Liess Laura; Gerold =
Pinker
Betreff: Re: [dispatch] FW: I-D
Action:draft-liess-dispatch-alert-info-urns-01.txt

I've been thinking about how the "namespace" of alert-info URNs might be =

reorganized. My goals in doing so are:
- allow the different "aspects" of the alert to be specified
- make it clear what the combination rules are for these
- allow for comparable expressiveness for rings and ringbacks
- provide room for extension

The following is a rough outline of a what I have in mind:

The urn syntax is as in the current draft, *except*:

alert-category  =3D "reason" / "priority" / "source" /
                   "duration" / "delay" / "locale" /
                   extension-category
extension-category =3D token

The principle for choosing these categories is that they are orthogonal. =

Hence there is a single rule for well-formedness of Alert-Info:
there can be at most one instance of each alert-category, and in=20
principle any such combination is valid. (Though some may be silly or=20
uninteresting.)

The above doesn't distinguish ring from ringback. That is to be=20
determined from context - when present in a request it describes the=20
ring. When present in a response it describes the ringback. Again, in=20
principle any combination can be used for either ring or ringback,=20
though some combinations may be silly or uninteresting.

The exact way in which the various categories are combined for rendering =

is left as an implementation issue. The implementation is free to ignore =

any or all at its whim.

Its my intent that things previously discussed in the draft can all be=20
mapped onto these categories. Here is my swag at doing so:

urn:alert:reason:

- normal (default)
- call-waiting
- forward
- auto-callback
- recall.hold
- recall.transfer
- private.<private-name>

urn:alert:priority:

- normal (default)
- low
- high
- crisis
- private.<private-name>

urn:alert:source:

- unclassified (default)
- internal
- external
- friend
- family
- private.<private-name>

urn:alert:duration:

- normal (default)
- short (or could be "zip")
- long (for completeness, or omit if unneeded)
- private.<private-name>

urn:alert:delay:

- none (default)
- yes (not sure what to put here)
- private.<private-name>

urn:alert:locale:

- default (default)
- country.<ISO 3166-1 country code>
- private.<private-name>

In the above the "private.<private-name>" syntax is for extensions=20
specific to independent organizations. The <private-name> is in the form =

of a "reverse FQDN" such as is used for Java package names. This gives a =

way of assigning unique names without the need for a new registry. The=20
namespace for each alert category is independent. Those assigning new=20
names must ensure they are in a position to assign names uniquely for=20
the FQDN they choose.

(I have considerable reservation about introducing such an uncontrolled=20
extension mechanism. I see how it can provide a lot of flexibility in=20
providing tailored user experiences. But it can also lead to vendor lock =

in if a device is only capable of supporting a particular vendor's=20
specific extensions. So I think this needs a lot of discussion.)

For example, I might want to define:

urn:alert:source:private.com.cisco.customer

If I were to do so, I should first ensure that I speak for cisco.com=20
when coining this.

Additions to the values above other than via the "private" mechanism=20
would be by standards action. The same for adding new categories.
(There could be a "private" mechanism for categories too. But I'm not at =

all confident that is a good thing.)

I'm not quite sure about "call-waiting" as a "reason" - it doesn't seem=20
quite right. The reason could still be any of those other things. For=20
the ring indication its far from clear to me that it is needed at all -=20
the recipient already knows it is on a call, and doesn't need to be told =

that. It may make more sense for the ringback - in that case you are=20
being told something about the state of the other party. Perhaps this=20
should be replaced by something about the state of the party sending the =

message. E.g.

urn:alert:state:

- unknown
- waiting (in a request, means the caller is waiting
            for the call to complete. The normal case.)
- on-a-call (in a response, means the called party
              is on another call - the call waiting case.
              But could apply to a caller as well.)
- away (in a request, means the call has been generated
         automatically and the user will be summoned if/when
         the call completes. This is the converse of waiting)

(This "state" is really a form of presence. Whether its good to encode=20
it here, rather than actually transmitting pidf, is an interesting=20
question.)

How all of this might be mapped onto actual renderings is probably not=20
something that should be standardized, but it might be worth including=20
some possibilities as examples. One that comes to mind is:

(Assuming all rendering are audio):

- the device has a set of possible clips it can render
- each is attributed with the values it matches for each
   category above. (none, one, several, or all for each
   category.)

The device just finds the maximal match and renders that one.

This might be augmented if there are different sorts of renderings -=20
e.g. audio, visual, tactile (vibrarion). In that case it might seek out=20
a match for each sort and render them all.

	Thanks,
	Paul


From pkyzivat@cisco.com  Thu Mar 25 19:02:02 2010
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Cc: dispatch@ietf.org, "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "Huelsemann, Martin" <Martin.Huelsemann@telekom.de>, 'Gerold Pinker' <Gerold.Pinker@telekom.de>
Subject: Re: [dispatch] FW: I-D Action:draft-liess-dispatch-alert-info-urns-01.txt
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Laura Liess wrote:
> Paul,
> 
> Thank you for the proposal and the elaborate description. I definitively
> like it.
> It seems to solve the combinations problem, it is extensible and easy to
> understand. 
> I have two comments on it, connected to "call-waiting".
> 
> 1) It seems to me that there is some misunderstanding concerning the meaning
> of the "call-waiting" alert urn. It is not the callee's presence state, but
> the state of the call itself. The "call-waiting" tone tells the caller:
> "Your call is temporary queued (waiting) at the callee's side because the
> callee is currently involved in another call. The callee was informed that
> your call is waiting." Having this information, the caller can decide to
> wait for a while or to cancel the call. This is an already existing
> PSTN-service which customers expect us to provide also when we migrate to
> SIP.
> Because this urn tells the user that the service "call-waiting" was
> performed at the other side of the call, we name "service" was choosed for
> this category. 

I wasn't sure from the draft whether this was about the ringback to the 
caller (which is what you are describing), or whether it was about the 
"ring" to the callee, which is of typically different in this case.

As I noted, I didn't think it made that much sense as an indication to 
the callee, since the callee knows its status. So I think we are in 
agreement about the important use - for ringback when the callee is 
currently handling another call.

As I was writing things down I realized that this is really a statement 
about the state of the callee that happens to be known to the UAS and 
hence reportable. I realize in the commonly used case this is a 
"service" in the network, and hence is different from a feature of the 
UAS - such as a multi-line phone. But the two are functionally 
equivalent to both the caller and the callee - the callee is occupied on 
some other call, and may (or may not) be delayed in answering. (Whether 
this delay is likely to be longer, or shorter, than would be the case if 
not on another call, is debatable. I guess just reporting it to the 
caller as a datum does permit the caller to make his own judgement.

If this is reported as a distinct category, then it can be reported in 
an orthogonal way. For instance maybe the ringback will be "normal" but 
there will be a display on the phone - perhaps akin to a "on hold" light.

> 2) There are already implementations of urn:alert:service:call-waiting. This
> syntax is used in the 3GPP Specification for the call-waiting service (TS
> 24.615 "Communication Waiting (CW) using IP Multimedia (IM) Core Network
> (CN) subsystem"). If we change this name, people either change their
> software, which means money and time for them, or they leave it as it is,
> which means lack of interoperability e2e. The "reason"-category contains in
> principles names for services, so I think it does not harm the proposal if
> we just replace "reason" by "service" and keep the interoperability with
> existing implementations.

How to name things is a second order issue. If its just a matter of 
"service" vs. "reason" then I don't care very much. (I'm not especially 
attached to "reason", but I also find "service" somewhat confusing for 
the purpose.)

The orthogonality is more important. The way I partitioned the data is 
just a first cut, and I'm entirely open to alternative ways of doing 
that. But I would hate to see the categorization warped just to fit one 
preexisting use. IMO it would make more sense to special-case that one - 
perhaps treat it as a separate category of its own.

	Thanks,
	Paul

> Thanks a lot
> Laura
> 
> 
> -----Ursprüngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Gesendet: Donnerstag, 25. März 2010 19:51
> An: Laura Liess
> Cc: alan.b.johnston@gmail.com; dispatch@ietf.org; Liess Laura; Gerold Pinker
> Betreff: Re: [dispatch] FW: I-D
> Action:draft-liess-dispatch-alert-info-urns-01.txt
> 
> I've been thinking about how the "namespace" of alert-info URNs might be 
> reorganized. My goals in doing so are:
> - allow the different "aspects" of the alert to be specified
> - make it clear what the combination rules are for these
> - allow for comparable expressiveness for rings and ringbacks
> - provide room for extension
> 
> The following is a rough outline of a what I have in mind:
> 
> The urn syntax is as in the current draft, *except*:
> 
> alert-category  = "reason" / "priority" / "source" /
>                    "duration" / "delay" / "locale" /
>                    extension-category
> extension-category = token
> 
> The principle for choosing these categories is that they are orthogonal. 
> Hence there is a single rule for well-formedness of Alert-Info:
> there can be at most one instance of each alert-category, and in 
> principle any such combination is valid. (Though some may be silly or 
> uninteresting.)
> 
> The above doesn't distinguish ring from ringback. That is to be 
> determined from context - when present in a request it describes the 
> ring. When present in a response it describes the ringback. Again, in 
> principle any combination can be used for either ring or ringback, 
> though some combinations may be silly or uninteresting.
> 
> The exact way in which the various categories are combined for rendering 
> is left as an implementation issue. The implementation is free to ignore 
> any or all at its whim.
> 
> Its my intent that things previously discussed in the draft can all be 
> mapped onto these categories. Here is my swag at doing so:
> 
> urn:alert:reason:
> 
> - normal (default)
> - call-waiting
> - forward
> - auto-callback
> - recall.hold
> - recall.transfer
> - private.<private-name>
> 
> urn:alert:priority:
> 
> - normal (default)
> - low
> - high
> - crisis
> - private.<private-name>
> 
> urn:alert:source:
> 
> - unclassified (default)
> - internal
> - external
> - friend
> - family
> - private.<private-name>
> 
> urn:alert:duration:
> 
> - normal (default)
> - short (or could be "zip")
> - long (for completeness, or omit if unneeded)
> - private.<private-name>
> 
> urn:alert:delay:
> 
> - none (default)
> - yes (not sure what to put here)
> - private.<private-name>
> 
> urn:alert:locale:
> 
> - default (default)
> - country.<ISO 3166-1 country code>
> - private.<private-name>
> 
> In the above the "private.<private-name>" syntax is for extensions 
> specific to independent organizations. The <private-name> is in the form 
> of a "reverse FQDN" such as is used for Java package names. This gives a 
> way of assigning unique names without the need for a new registry. The 
> namespace for each alert category is independent. Those assigning new 
> names must ensure they are in a position to assign names uniquely for 
> the FQDN they choose.
> 
> (I have considerable reservation about introducing such an uncontrolled 
> extension mechanism. I see how it can provide a lot of flexibility in 
> providing tailored user experiences. But it can also lead to vendor lock 
> in if a device is only capable of supporting a particular vendor's 
> specific extensions. So I think this needs a lot of discussion.)
> 
> For example, I might want to define:
> 
> urn:alert:source:private.com.cisco.customer
> 
> If I were to do so, I should first ensure that I speak for cisco.com 
> when coining this.
> 
> Additions to the values above other than via the "private" mechanism 
> would be by standards action. The same for adding new categories.
> (There could be a "private" mechanism for categories too. But I'm not at 
> all confident that is a good thing.)
> 
> I'm not quite sure about "call-waiting" as a "reason" - it doesn't seem 
> quite right. The reason could still be any of those other things. For 
> the ring indication its far from clear to me that it is needed at all - 
> the recipient already knows it is on a call, and doesn't need to be told 
> that. It may make more sense for the ringback - in that case you are 
> being told something about the state of the other party. Perhaps this 
> should be replaced by something about the state of the party sending the 
> message. E.g.
> 
> urn:alert:state:
> 
> - unknown
> - waiting (in a request, means the caller is waiting
>             for the call to complete. The normal case.)
> - on-a-call (in a response, means the called party
>               is on another call - the call waiting case.
>               But could apply to a caller as well.)
> - away (in a request, means the call has been generated
>          automatically and the user will be summoned if/when
>          the call completes. This is the converse of waiting)
> 
> (This "state" is really a form of presence. Whether its good to encode 
> it here, rather than actually transmitting pidf, is an interesting 
> question.)
> 
> How all of this might be mapped onto actual renderings is probably not 
> something that should be standardized, but it might be worth including 
> some possibilities as examples. One that comes to mind is:
> 
> (Assuming all rendering are audio):
> 
> - the device has a set of possible clips it can render
> - each is attributed with the values it matches for each
>    category above. (none, one, several, or all for each
>    category.)
> 
> The device just finds the maximal match and renders that one.
> 
> This might be augmented if there are different sorts of renderings - 
> e.g. audio, visual, tactile (vibrarion). In that case it might seek out 
> a match for each sort and render them all.
> 
> 	Thanks,
> 	Paul
> 
> 

From mary.ietf.barnes@gmail.com  Fri Mar 26 10:49:29 2010
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Subject: [dispatch] DISPATCH WG deadlines for IETF-78
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Hi folks,

Per the discussion in the SIPCLF WG this morning, the following are
the deadlines for topics for DISPATCH for IETF-78:

* May 17th, 2010. Cutoff date to notify the chairs/DISPATCH WG of
plans to submit a proposal. [Two weeks prior to BoF proposal deadline]

* May 31st, 2010. Cutoff for charter proposals for topics. [One day
prior to BoF proposal deadline]

* June 7th, 2010. Topics that are to be the focus of IETF-78 are
announced. [One week before AD BoF approval and deadline to request WG
slot]

* July 5th, 2010. -00 draft deadline.

* July 12th, 2010. Draft deadline.

These deadlines have been published on the DISPATCH WG wiki page:
http://trac.tools.ietf.org/wg/dispatch/trac/wiki

Regards,
Mary
DISPATCH WG co-chair

From adam@nostrum.com  Mon Mar 29 08:35:24 2010
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Subject: Re: [dispatch] Fwd: Last	Call:	draft-lawrence-sipforum-user-agent-config (Session	Initiation	Protocol (SIP) User Agent Configuration)	to	Informational RFC
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I tend to agree with Henning on all the points he raises below, and want 
to expand just a little bit.

The use of ITADs has in no way confused people participating in the 
Freenum trial into thinking that TRIP was required or even an option. In 
fact, the separation of ITAD from TRIP in that experiment is so 
inherently clear that freenum.org doesn't even feel the need to discuss 
the hypothesized confusion in their FAQ (http://freenum.org/faq.html). 
Unless we expect a materially dimmer group of people to be consumers of 
the SIP Forum's output, there is historical information that TRIP 
conflation concerns are unfounded.

In addition to the points that Henning raised, I'm a bit confused by the 
assertion that the use of "freenum.org" does not scale, given that it is 
modeled on the general DNS delegation model, and is basically the same 
mechanism as ENUM uses. The current sipforum draft appears to 
counter-propose the use of a single, flat, non-delegated database at 
"sipforum.org." Could you clarify how trading a hierarchical, 
distributed database technology that provides domain naming services for 
approximately 400 million Internet hosts will have inferior scaling 
properties when compared against what is presumably a flat relational 
database hosted by SIPForum?

/a

On 3/24/10 4:37 PM, Henning Schulzrinne wrote:
>>
>> The attached is a problem statement I drafted for the SIP Forum Board
>> that summarized the issues raised in the task group and from some
>> external reviewers.
>
> To quote:
>
> The administrative and legal infrastructure of using the existing
>      first-come first-served ITAD registry at IANA is not sufficient
>      for the needs of commercial service providers.
>
> * No reasons are given for this determination.
>
>  Using the existing ITAD name and registry may create confusion
>      regarding whether or not other parts of the protocols and
>      procedures in the TRIP specification and/or the ISN/Freenum trial
>      are also applicable (the intent is that no other part of these
>      other specifications and procedures be used).
>
> * I admit that I find this speculative. As long as the mechanism is 
> clear, the users of the DNS record won't care.
>
> The ITAD number is defined as a 32 bit number.
>
> * This translates into (roughly) a 9-digit decimal integer. It seems 
> rather unlikely that we would exceed 4 billion service providers or 
> that we'd want longer identifiers than 9 digits.
>
> In general, I agree that the ITAD registry would have to be expanded 
> and the use clarified, but re-use of the same registry (say, 
> enterprise numbers) for different purposes is not unusual.
>
> Henning
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


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 http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I tend to agree with Henning on all the points he raises below, and
want to expand just a little bit. <br>
<br>
The use of ITADs has in no way confused people participating in the
Freenum trial into thinking that TRIP was required or even an option.
In fact, the separation of ITAD from TRIP in that experiment is so
inherently clear that freenum.org doesn't even feel the need to discuss
the hypothesized confusion in their FAQ (<a class="moz-txt-link-freetext" href="http://freenum.org/faq.html">http://freenum.org/faq.html</a>).
Unless we expect a materially dimmer group of people to be consumers of
the SIP Forum's output, there is historical information that TRIP
conflation concerns are unfounded.<br>
<br>
In addition to the points that Henning raised, I'm a bit confused by
the assertion that the use of "freenum.org" does not scale, given that
it is modeled on the general DNS delegation model, and is basically the
same mechanism as ENUM uses. The current sipforum draft appears to
counter-propose the use of a single, flat, non-delegated database at
"sipforum.org." Could you clarify how trading a hierarchical,
distributed database technology that provides domain naming services
for approximately 400 million Internet hosts will have inferior scaling
properties when compared against what is presumably a flat relational
database hosted by SIPForum?<br>
<br>
/a<br>
<br>
On 3/24/10 4:37 PM, Henning Schulzrinne wrote:
<blockquote
 cite="mid:A1C3BB28-8582-436E-A2CE-320E1C97423C@cs.columbia.edu"
 type="cite">
  <div>
  <blockquote type="cite">
    <div><font class="Apple-style-span" color="#000000"><br>
    </font>The attached is a problem statement I drafted for the SIP
Forum Board<br>
that summarized the issues raised in the task group and from some<br>
external reviewers.<br>
    </div>
  </blockquote>
  </div>
  <br>
  <div>To quote:</div>
  <div><br>
  </div>
  <div>
  <div>The administrative and legal infrastructure of using the existing</div>
  <div>&nbsp;&nbsp; &nbsp; first-come first-served ITAD registry at IANA is not
sufficient</div>
  <div>&nbsp;&nbsp; &nbsp; for the needs of commercial service providers.</div>
  <div><br>
  </div>
  <div>* No reasons are given for this determination.</div>
  <div><br>
  </div>
  <div>
  <div>&nbsp;Using the existing ITAD name and registry may create confusion</div>
  <div>&nbsp;&nbsp; &nbsp; regarding whether or not other parts of the protocols and</div>
  <div>&nbsp;&nbsp; &nbsp; procedures in the TRIP specification and/or the ISN/Freenum
trial</div>
  <div>&nbsp;&nbsp; &nbsp; are also applicable (the intent is that no other part of
these</div>
  <div>&nbsp;&nbsp; &nbsp; other specifications and procedures be used).</div>
  <div><br>
  </div>
  <div>* I admit that I find this speculative. As long as the mechanism
is clear, the users of the DNS record won't care.</div>
  <div><br>
  </div>
  <div>The ITAD number is defined as a 32 bit number.</div>
  <div><br>
  </div>
  <div>* This translates into (roughly) a 9-digit decimal integer. It
seems rather unlikely that we would exceed 4 billion service providers
or that we'd want longer identifiers than 9 digits.</div>
  <div><br>
  </div>
  <div>In general, I agree that the ITAD registry would have to be
expanded and the use clarified, but re-use of the same registry (say,
enterprise numbers) for different purposes is not unusual.</div>
  <div><br>
  </div>
  <div>Henning</div>
  <div><br>
  </div>
  </div>
  </div>
  <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
dispatch mailing list
<a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a>
  </pre>
</blockquote>
<br>
</body>
</html>

--------------000803090600030201030502--

From xmlscott@gmail.com  Mon Mar 29 13:52:48 2010
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Subject: Re: [dispatch] draft-lawrence-sipforum-provider-alias-00
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(changed the subject line to accurately reflect the draft being
discussed)

On Mon, 2010-03-29 at 10:35 -0500, Adam Roach wrote:

> The use of ITADs has in no way confused people participating in the
> Freenum trial into thinking that TRIP was required or even an option.
> In fact, the separation of ITAD from TRIP in that experiment is so
> inherently clear that freenum.org doesn't even feel the need to
> discuss the hypothesized confusion in their FAQ
> (http://freenum.org/faq.html). Unless we expect a materially dimmer
> group of people to be consumers of the SIP Forum's output, there is
> historical information that TRIP conflation concerns are unfounded.

I don't personally think that this is the most important of the
problems, but we did get people asking questions about this - more were
about how the new usage related to what Freenum.org is doing than about
TRIP.  

Which serves to highlight that while the ITAD registry identifies
domains, that ISN does is _not_ the same as what this draft proposes.
The ISN dialing transform takes as input a dial string (usually carried
as the user part of a sip uri) into a full sip uri - the input has both
a domain identifier and a local part to be resolved in that domain.
This draft proposes a transform of one domain name into another, to be
used not for routing any sip request, but for subsequent resolution
(also through the DNS) into an http url.

When I was editing the SIP Forum task group draft, the difference was
brought home to me by the fact that there was no ISN 'standard' I could
claim to be profiling for this - we were doing an entirely different
operation for a significantly different purpose.  Of course, as it
turned out the Board decided we needed to take the whole thing elsewhere
to be made a standard anyway... 

> In addition to the points that Henning raised, I'm a bit confused by
> the assertion that the use of "freenum.org" does not scale, given that
> it is modeled on the general DNS delegation model, and is basically
> the same mechanism as ENUM uses.

I wasn't saying that DNS doesn't scale - I said that the _current_
freenum.org infrastructure was not appropriate to the needs of
commercial service providers.  Not a protocol issue at all.

Suppose that you are selling a small business or consumer phone service,
and you are relying on the (appropriately simple) instruction that the
customer set up a phone by entering '555' (your alias) when connecting a
new phone.   If it doesn't work because the translation can't be done,
do you think that the average small business person or consumer will
blame the operator of the DNS service or you?

The current freenum.org infrastructure is some non-profit research
institutions and universities; a fine thing and a great organization.
That doesn't make it appropriate as a key component in commercial
services (if there are significant commercial deployments of ISN I am
not aware of them, and if they are making money then I want to meet
their sales people).

 The current sipforum draft appears to counter-propose the use of a
single, flat, non-delegated database at "sipforum.org." 

If you read it as proposing any such thing, then I accept that editorial
work is needed (if you can divine how you got that idea, please point
out text).  It proposes the use of DNS NAPTR lookups with all the
marvelous scaling and redundancy properties you so justifiably admire.

As for 'sipforum.org', the draft says:

   The lookup key for the S-NAPTR request is the Provider Alias Number
   concatenated with a fixed root suffix (for the purposes of this
   initial draft, the value ".pan.sipforum.org." is used) to form a DNS
   domain name. 

... the key words here being 'a fixed suffix (for the purposes of this
initial draft'...  The actual value chosen can be anything, but it
seemed clearer to use a real example.  No one (most emphatically
including the SIP Forum itself) believes that the Forum is the
appropriate organization to actually operate the resolution
infrastructure regardless of the domain name chosen as the fixed suffix.

Personally, I think that the most important reason not to use ITADs or
freenum.org was the first one I listed:

   * The administrative and legal infrastructure of using the existing
     first-come first-served ITAD registry at IANA is not sufficient
     for the needs of commercial service providers.

If I were a service provider and was going to count on this to provide
me with that vital first connection service, I'd want to know that the
rules on how registrations were added to and modified in the registry
were subject to carefully spelled out contractual terms with enforceable
agreements as to service levels.  I'd want to know that that asset could
be transferred under well specified terms.  RFC 3219 has no such
infrastructure.  



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Subject: [dispatch] draft-lawrence-sipforum-provider-alias-00
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Scott accurately described what I believe is the business case for the SIP
Forum to sponsor such a registry, though that decision has not been made
yet. The SIP Forum would NEVER actually operate the registry. There are lots
of companies that can do that.

 

The PAN registry IMHO needs the kind of strong contractual agreements common
to operators of ICANN TLD's for instance or public / private ENUM
registries.

 

 

 

... the key words here being 'a fixed suffix (for the purposes of this

initial draft'...  The actual value chosen can be anything, but it

seemed clearer to use a real example.  No one (most emphatically

including the SIP Forum itself) believes that the Forum is the

appropriate organization to actually operate the resolution

infrastructure regardless of the domain name chosen as the fixed suffix.

 

Personally, I think that the most important reason not to use ITADs or

freenum.org was the first one I listed:

 

   * The administrative and legal infrastructure of using the existing

     first-come first-served ITAD registry at IANA is not sufficient

     for the needs of commercial service providers.

 

If I were a service provider and was going to count on this to provide

me with that vital first connection service, I'd want to know that the

rules on how registrations were added to and modified in the registry

were subject to carefully spelled out contractual terms with enforceable

agreements as to service levels.  I'd want to know that that asset could

be transferred under well specified terms.  RFC 3219 has no such

infrastructure.  

 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
< <mailto:richard(at)shockey.us> mailto:richard(at)shockey.us>
skype: rshockey101
LinkedIn :  <http://www.linkedin.com/in/rshockey101>
http://www.linkedin.com/in/rshockey101
http//www.sipforum.org

 


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<p class=3DMsoNormal>Scott accurately described what I believe is the =
business
case for the SIP Forum to sponsor such a registry, though that decision =
has not
been made yet. The SIP Forum would NEVER actually operate the registry. =
There
are lots of companies that can do that.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>The PAN registry IMHO needs the kind of strong =
contractual
agreements common to operators of ICANN TLD&#8217;s for instance or =
public /
private ENUM registries.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div =
style=3D'mso-element:para-border-div;border:none;border-bottom:dotted =
windowtext 3.0pt;
padding:0in 0in 1.0pt 0in'>

<p class=3DMsoPlainText =
style=3D'border:none;padding:0in'><o:p>&nbsp;</o:p></p>

</div>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>... the key words here being 'a fixed suffix =
(for the
purposes of this<o:p></o:p></p>

<p class=3DMsoPlainText>initial draft'...&nbsp; The actual value chosen =
can be
anything, but it<o:p></o:p></p>

<p class=3DMsoPlainText>seemed clearer to use a real example.&nbsp; No =
one (most
emphatically<o:p></o:p></p>

<p class=3DMsoPlainText>including the SIP Forum itself) believes that =
the Forum
is the<o:p></o:p></p>

<p class=3DMsoPlainText>appropriate organization to actually operate the
resolution<o:p></o:p></p>

<p class=3DMsoPlainText>infrastructure regardless of the domain name =
chosen as
the fixed suffix.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Personally, I think that the most important =
reason not to
use ITADs or<o:p></o:p></p>

<p class=3DMsoPlainText>freenum.org was the first one I =
listed:<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp; * The administrative and legal =
infrastructure of using
the existing<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp; first-come first-served =
ITAD registry at IANA is not
sufficient<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp; for the needs of =
commercial service providers.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>If I were a service provider and was going to =
count on
this to provide<o:p></o:p></p>

<p class=3DMsoPlainText>me with that vital first connection service, I'd =
want to
know that the<o:p></o:p></p>

<p class=3DMsoPlainText>rules on how registrations were added to and =
modified in
the registry<o:p></o:p></p>

<p class=3DMsoPlainText>were subject to carefully spelled out =
contractual terms
with enforceable<o:p></o:p></p>

<p class=3DMsoPlainText>agreements as to service levels.&nbsp; I'd want =
to know that
that asset could<o:p></o:p></p>

<p class=3DMsoPlainText>be transferred under well specified terms.&nbsp; =
RFC 3219 has
no such<o:p></o:p></p>

<p class=3DMsoPlainText>infrastructure.&nbsp; <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>
Shockey Consulting<br>
Chairman of the Board of Directors SIP Forum<br>
PSTN Mobile: +1 703.593.2683<br>
&lt;<a href=3D"mailto:richard(at)shockey.us"><span =
style=3D'color:blue'>mailto:richard(at)shockey.us</span></a>&gt;<br>
skype: rshockey101<br>
LinkedIn : <a href=3D"http://www.linkedin.com/in/rshockey101"><span
style=3D'color:blue'>http://www.linkedin.com/in/rshockey101</span></a><br=
>
http//www.sipforum.org</span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

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Thread-Topic: [dispatch] [Fwd: Re: Preliminaryversion	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] [Fwd: Re: Preliminaryversion	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Hi All

Any update on this I-D?

Thanks=20


Regards
Ranjit

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Avasarala Ranjit-A20990
Sent: Monday, March 22, 2010 10:53 PM
To: Gonzalo Camarillo
Cc: dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminaryversion ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

Hi Gonzalo, Paul=20

I have uploaded the updated version of the I-D.=20
It can be accessed from:
http://www.ietf.org/id/draft-avasarala-dispatch-comm-div-notification-04
.txt

Please review and comment

Thanks

Regards
Ranjit

-----Original Message-----
From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
Sent: Tuesday, March 16, 2010 4:29 PM
To: Avasarala Ranjit-A20990
Cc: Paul Kyzivat; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

Hi Ranjit,

there is a blackout period before IETF meetings in which one cannot
submit drafts. Please, submit this version as soon as the blackout
period is over and have Paul have a look at it so that he can confirm he
is happy with it.

Thanks,

Gonzalo


On 11/03/2010 8:10 PM, Avasarala Ranjit-A20990 wrote:
> Hi All
>=20
> I am not able to upload the updated version (-04) of the I-D due to=20
> some issue with the I-D submission tool - (Says I can submit I-D only=20
> on 2010-03-22).
> Until then I am attaching a copy of the updated version (txt and html=20
> formats).
>=20
> Please review and comment.
>=20
> Thanks
>=20
> Note: Let me know if there is any other way to upload the I-D document

> and will be happy to do it.
>=20
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, March 02, 2010 7:19 PM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
> I think this all sounds good now.
>=20
>         Thanks,
>         Paul
>=20
> Avasarala Ranjit-A20990 wrote:
>> Inline
>>
>> <Ranjit-Mar2>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, March 02, 2010 2:16 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
>> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review

>> ofdraft-avasarala-dispatch-comm-div-notification-03]
>>
>> inline...
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>> I'm still a little confused here.
>>> If phone is on and subscription is active, then in general=20
>>> notification should be possible, regardless of registration. I=20
>>> realize
>>
>>> IMS doesn't permit that, but from a general standard perspective we=20
>>> shouldn't assume it. The manifestation in this case is that the=20
>>> NOTIFY
>>
>>> would fail and the subscription would eventually be torn down.
>>>
>>> So I think the cases are:
>>> - there is a subscription and notification works
>>> - there is a subscription but notification fails
>>> - there is no subscription.
>>>
>>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as=20
>>> Notification fails, we can say - not sending notification due to (1)

>>> no match of filter criteria (2) phone not registered or outside=20
>>> coverage area due to which there is no connectivity and hence=20
>>> notification cannot be sent Anyway if no subscription, no
>> notification.
>>
>> Are you suggesting that you might have a subscription, and have the=20
>> filter criteria met, and yet not attempt to send a NOTIFY because the

>> phone is not registered or has no network connectivity???
>>
>> That seems wrong. ISTM that either you would remove the subscription,

>> or else you would try to send the NOTIFY and then discover that it
> fails.
>>
>> <Ranjit-Mar2> yes the subscription could be removed when the device=20
>> is
>=20
>> not reachable.
>>
>> OTOH, this probably ought not to be within scope of what is discussed

>> in this draft.
>>
>> <Ranjit-Mar2> yes true not within the scope of this draft.
>>
>>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>>> contacts: sip:line1@alice.office.atlanta.com and=20
>>>> sip:line2@alice.home.atlanta.com and
>>>>     both the phones ring, then it would be a case of parallel
>> forking.
>>>> Now say one or both the phones have set diversion rules, to divert=20
>>>> the
>>> In this case, what does it mean for "one or both phones have set=20
>>> diversion rules"? Does that somehow imply distinct rules for each
>> phone?
>>> I don't see how that could work. Aren't the diversion rules for the=20
>>> *AOR*?
>>>
>>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there=20
>>> is
>>
>>> a diversion rule set for the AoR and when diversions occur, the=20
>>> notifications are Sent to the AoR.
>>
>> Not sent to the AoR - sent to the remote contact of the dialog for=20
>> each subscription to the AoR. Same point as below that you agreed
> with.
>>
>> <Ranjit-Mar2> Yes agree with u.
>>
>>> I see they could be set/changed from either phone. But its a single=20
>>> set of rules isn't it? So I might set the rules from the office, and

>>> then cancel them from home.
>>>
>>> <Ranjit-Feb26-2> yes single set and could be set from either of the=20
>>> phones. True.
>>>
>>>> call to say
>>>>     sip:voicemail@alice.atlanta.com, then the notifications would=20
>>>> get
>>
>>>> generated with respect to both the contacts.
>>> Again I'm confused by the terminology you use, and how it relates=20
>>> operationally. The diversion happens on the AOR, and the=20
>>> subscriptions
>>
>>> are on the AOR, and the notifications are sent to the subscribers. I

>>> don't see where the registered contacts enter into the picture at
> all.
>>>
>>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>>
>>> If you have something else in mind, I hope you can explain it better

>>> in the draft.
>>>
>>>> [4] Yes the value of N is configurable and depends on how many=20
>>>> diversions the subscribe wants to keep or could be dependent on=20
>>>> server policy (some maximum
>>>>     number). If the server policy is to send notifications as and=20
>>>> when they occur and not keep any divsersions information, then the=20
>>>> value of last N
>>>>     diversions will be zero. But if the server chooses to retain=20
>>>> the
>=20
>>>> last diversion always, then the value of N will be 1 and so on.
>>> All is straightforward for N>=3D1.
>>>
>>> For N=3D0 things are messy. If we are modeling this as state, and=20
>>> notifications are about state changes, then the new diversion is=20
>>> added
>>
>>> to the (previously empty) state and that triggers notifications to=20
>>> all
>>
>>> current subscriptions. Then if you don't want to retain even this=20
>>> much
>>
>>> state (because N=3D0) you immediately remove that version from the
>> state.
>>> But that is again a state change, so you end up with another=20
>>> notification of that state change.
>>>
>>> <Ranjit-Feb26-2> True. Actually the notifications are for informing=20
>>> the subscriber of the call diversions that happen. The notifications

>>> are not for informing the subscriber of state changes in the=20
>>> notifier
>=20
>>> or the number of diversions that occurred. As I mentioned in the=20
>>> abstract as well as 24.604, the main intent of CDIVN service is to=20
>>> inform the subscribers of communication diversions that occur on=20
>>> their
>>
>>> behalf in the network and this information will help them manage=20
>>> their
>>
>>> rules better.
>>
>> I am not certain, but think we may still be talking past one another=20
>> to some extent.
>>
>> The state can be considered just a formalism for defining the service

>> clearly, regardless of whether it has intrinsic value to the CDIV=20
>> service. But it *is* a formalism. So if it is introduced, then its=20
>> behavior can't be ignored when inconvenient. Defining the state with=20
>> N=3D0 leads to some undesired behavior. The simplest way around that =
is

>> to insist that N be >=3D1.
>>
>> <Ranjit-Mar2> Ok in that case we can say that value of N >=3D1 and =
the=20
>> notifier has always maintains history of last diversion at a minimu.
>> It can maintain more than 1 too depending on the policy.
>>
>> If its important to you that no state be retained once a notification

>> has been sent about the most recent diversion, then you will have to=20
>> find some other way to specify the service, that still meets the=20
>> expectations of 3265. I don't know if there is anything there that=20
>> Adam would consider valid. We will have to get his opinion on that.
>>
>> <Ranjit-Mar2> In that case I am OK maintaining the hostory of=20
>> diversions that occurred as part of state information and retaining=20
>> that state even after the notification for that diversion is sent -=20
>> so
>=20
>> that in case I do not get a confirmation, I would be able to re-send
> the notification.
>> Also maintaining state of diversions would help when the=20
>> notifications
>=20
>> need to sent only at a particular time (if fikter criteria mentions
> it).
>>
>>
>>       Thanks,
>>       Paul
>>
>>> <Ranjit-Feb26-2> But if we want to include the information related=20
>>> to
>=20
>>> how many diversions have happened till time (say from 12 AM ) till=20
>>> the
>>
>>> time the notification is being sent, then as you say we can store=20
>>> the
>=20
>>> diversions. This would be useful when the subscriber specifies in=20
>>> the
>=20
>>> filter criteria to send notifications only during certain time like=20
>>> between 5 and 6 PM and not whenever a diversion occurs. Then N is=20
>>> grater than 1.
>>
>>> I presume you don't want that behavior. If you can live with N>=3D1=20
>>> then
>>
>>> we don't have the problem at all. If you need to support N=3D0 and=20
>>> don't
>>
>>> want the extra notify, then we have to figure out if there is some=20
>>> trick to how things are modeled to get the desired effect. I've=20
>>> suggested something previously, but Adam didn't like it.
>>>
>>> This needs sorting out.
>>>
>>>      Thanks,
>>>      Paul
>>

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Cc: dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminaryversion	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Ranjit,

Sorry! I was busy last week at ietf.
I will try to get to it this week.

	Paul

Avasarala Ranjit-A20990 wrote:
> Hi All
> 
> Any update on this I-D?
> 
> Thanks 
> 
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Avasarala Ranjit-A20990
> Sent: Monday, March 22, 2010 10:53 PM
> To: Gonzalo Camarillo
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminaryversion ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> Hi Gonzalo, Paul 
> 
> I have uploaded the updated version of the I-D. 
> It can be accessed from:
> http://www.ietf.org/id/draft-avasarala-dispatch-comm-div-notification-04
> .txt
> 
> Please review and comment
> 
> Thanks
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: Gonzalo Camarillo [mailto:Gonzalo.Camarillo@ericsson.com]
> Sent: Tuesday, March 16, 2010 4:29 PM
> To: Avasarala Ranjit-A20990
> Cc: Paul Kyzivat; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> Hi Ranjit,
> 
> there is a blackout period before IETF meetings in which one cannot
> submit drafts. Please, submit this version as soon as the blackout
> period is over and have Paul have a look at it so that he can confirm he
> is happy with it.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> On 11/03/2010 8:10 PM, Avasarala Ranjit-A20990 wrote:
>> Hi All
>>
>> I am not able to upload the updated version (-04) of the I-D due to 
>> some issue with the I-D submission tool - (Says I can submit I-D only 
>> on 2010-03-22).
>> Until then I am attaching a copy of the updated version (txt and html 
>> formats).
>>
>> Please review and comment.
>>
>> Thanks
>>
>> Note: Let me know if there is any other way to upload the I-D document
> 
>> and will be happy to do it.
>>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, March 02, 2010 7:19 PM
>> To: Avasarala Ranjit-A20990
>> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
>> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review 
>> ofdraft-avasarala-dispatch-comm-div-notification-03]
>>
>> I think this all sounds good now.
>>
>>         Thanks,
>>         Paul
>>
>> Avasarala Ranjit-A20990 wrote:
>>> Inline
>>>
>>> <Ranjit-Mar2>
>>>
>>>
>>> Regards
>>> Ranjit
>>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: Tuesday, March 02, 2010 2:16 AM
>>> To: Avasarala Ranjit-A20990
>>> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
>>> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> 
>>> ofdraft-avasarala-dispatch-comm-div-notification-03]
>>>
>>> inline...
>>>
>>> Avasarala Ranjit-A20990 wrote:
>>>
>>>> I'm still a little confused here.
>>>> If phone is on and subscription is active, then in general 
>>>> notification should be possible, regardless of registration. I 
>>>> realize
>>>> IMS doesn't permit that, but from a general standard perspective we 
>>>> shouldn't assume it. The manifestation in this case is that the 
>>>> NOTIFY
>>>> would fail and the subscription would eventually be torn down.
>>>>
>>>> So I think the cases are:
>>>> - there is a subscription and notification works
>>>> - there is a subscription but notification fails
>>>> - there is no subscription.
>>>>
>>>> <Ranjit-Feb26-2> Agreed. Actually instead of calling it as 
>>>> Notification fails, we can say - not sending notification due to (1)
> 
>>>> no match of filter criteria (2) phone not registered or outside 
>>>> coverage area due to which there is no connectivity and hence 
>>>> notification cannot be sent Anyway if no subscription, no
>>> notification.
>>>
>>> Are you suggesting that you might have a subscription, and have the 
>>> filter criteria met, and yet not attempt to send a NOTIFY because the
> 
>>> phone is not registered or has no network connectivity???
>>>
>>> That seems wrong. ISTM that either you would remove the subscription,
> 
>>> or else you would try to send the NOTIFY and then discover that it
>> fails.
>>> <Ranjit-Mar2> yes the subscription could be removed when the device 
>>> is
>>> not reachable.
>>>
>>> OTOH, this probably ought not to be within scope of what is discussed
> 
>>> in this draft.
>>>
>>> <Ranjit-Mar2> yes true not within the scope of this draft.
>>>
>>>>> [3] in the example u suggested where sip:alice@atlanta.org with two
>>>>> contacts: sip:line1@alice.office.atlanta.com and 
>>>>> sip:line2@alice.home.atlanta.com and
>>>>>     both the phones ring, then it would be a case of parallel
>>> forking.
>>>>> Now say one or both the phones have set diversion rules, to divert 
>>>>> the
>>>> In this case, what does it mean for "one or both phones have set 
>>>> diversion rules"? Does that somehow imply distinct rules for each
>>> phone?
>>>> I don't see how that could work. Aren't the diversion rules for the 
>>>> *AOR*?
>>>>
>>>> <Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there 
>>>> is
>>>> a diversion rule set for the AoR and when diversions occur, the 
>>>> notifications are Sent to the AoR.
>>> Not sent to the AoR - sent to the remote contact of the dialog for 
>>> each subscription to the AoR. Same point as below that you agreed
>> with.
>>> <Ranjit-Mar2> Yes agree with u.
>>>
>>>> I see they could be set/changed from either phone. But its a single 
>>>> set of rules isn't it? So I might set the rules from the office, and
> 
>>>> then cancel them from home.
>>>>
>>>> <Ranjit-Feb26-2> yes single set and could be set from either of the 
>>>> phones. True.
>>>>
>>>>> call to say
>>>>>     sip:voicemail@alice.atlanta.com, then the notifications would 
>>>>> get
>>>>> generated with respect to both the contacts.
>>>> Again I'm confused by the terminology you use, and how it relates 
>>>> operationally. The diversion happens on the AOR, and the 
>>>> subscriptions
>>>> are on the AOR, and the notifications are sent to the subscribers. I
> 
>>>> don't see where the registered contacts enter into the picture at
>> all.
>>>> <Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.
>>>>
>>>> If you have something else in mind, I hope you can explain it better
> 
>>>> in the draft.
>>>>
>>>>> [4] Yes the value of N is configurable and depends on how many 
>>>>> diversions the subscribe wants to keep or could be dependent on 
>>>>> server policy (some maximum
>>>>>     number). If the server policy is to send notifications as and 
>>>>> when they occur and not keep any divsersions information, then the 
>>>>> value of last N
>>>>>     diversions will be zero. But if the server chooses to retain 
>>>>> the
>>>>> last diversion always, then the value of N will be 1 and so on.
>>>> All is straightforward for N>=1.
>>>>
>>>> For N=0 things are messy. If we are modeling this as state, and 
>>>> notifications are about state changes, then the new diversion is 
>>>> added
>>>> to the (previously empty) state and that triggers notifications to 
>>>> all
>>>> current subscriptions. Then if you don't want to retain even this 
>>>> much
>>>> state (because N=0) you immediately remove that version from the
>>> state.
>>>> But that is again a state change, so you end up with another 
>>>> notification of that state change.
>>>>
>>>> <Ranjit-Feb26-2> True. Actually the notifications are for informing 
>>>> the subscriber of the call diversions that happen. The notifications
> 
>>>> are not for informing the subscriber of state changes in the 
>>>> notifier
>>>> or the number of diversions that occurred. As I mentioned in the 
>>>> abstract as well as 24.604, the main intent of CDIVN service is to 
>>>> inform the subscribers of communication diversions that occur on 
>>>> their
>>>> behalf in the network and this information will help them manage 
>>>> their
>>>> rules better.
>>> I am not certain, but think we may still be talking past one another 
>>> to some extent.
>>>
>>> The state can be considered just a formalism for defining the service
> 
>>> clearly, regardless of whether it has intrinsic value to the CDIV 
>>> service. But it *is* a formalism. So if it is introduced, then its 
>>> behavior can't be ignored when inconvenient. Defining the state with 
>>> N=0 leads to some undesired behavior. The simplest way around that is
> 
>>> to insist that N be >=1.
>>>
>>> <Ranjit-Mar2> Ok in that case we can say that value of N >=1 and the 
>>> notifier has always maintains history of last diversion at a minimu.
>>> It can maintain more than 1 too depending on the policy.
>>>
>>> If its important to you that no state be retained once a notification
> 
>>> has been sent about the most recent diversion, then you will have to 
>>> find some other way to specify the service, that still meets the 
>>> expectations of 3265. I don't know if there is anything there that 
>>> Adam would consider valid. We will have to get his opinion on that.
>>>
>>> <Ranjit-Mar2> In that case I am OK maintaining the hostory of 
>>> diversions that occurred as part of state information and retaining 
>>> that state even after the notification for that diversion is sent - 
>>> so
>>> that in case I do not get a confirmation, I would be able to re-send
>> the notification.
>>> Also maintaining state of diversions would help when the 
>>> notifications
>>> need to sent only at a particular time (if fikter criteria mentions
>> it).
>>>
>>>       Thanks,
>>>       Paul
>>>
>>>> <Ranjit-Feb26-2> But if we want to include the information related 
>>>> to
>>>> how many diversions have happened till time (say from 12 AM ) till 
>>>> the
>>>> time the notification is being sent, then as you say we can store 
>>>> the
>>>> diversions. This would be useful when the subscriber specifies in 
>>>> the
>>>> filter criteria to send notifications only during certain time like 
>>>> between 5 and 6 PM and not whenever a diversion occurs. Then N is 
>>>> grater than 1.
>>>> I presume you don't want that behavior. If you can live with N>=1 
>>>> then
>>>> we don't have the problem at all. If you need to support N=0 and 
>>>> don't
>>>> want the extra notify, then we have to figure out if there is some 
>>>> trick to how things are modeled to get the desired effect. I've 
>>>> suggested something previously, but Adam didn't like it.
>>>>
>>>> This needs sorting out.
>>>>
>>>>      Thanks,
>>>>      Paul
> 
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