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From iptel-bounces@ietf.org  Thu Jan  6 16:15:16 2005
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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To: Shan Lu <shanlu@sentito.com>
Subject: Re: [Iptel] Finalizing trunk-group I-D
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Shan Lu wrote:
> Vijay,
> 
> I like your proposal in spirit, but am weary of the (over)use of
> "anonymous". 

Shan Lu:

Sorry for the delayed response.  Please see inline.

> For everybody's benefit, you propose the following format when
> calling party number is absent:
> 
> Contact: <sip:anonymous;phone-context=example.com;tgrp=foo@example.com>,
> 
> where the origination TG is "foo".
[...]
> OTOH, what we want is to convey to the recipient of Contact header the
> following: the SIP URI userinfo in Contact is a "quasi-" tel URI in that any
> subsequent fields following the first semi-colon must be treated as tel URI
> parameters, including ";tgrp=". 
> 
> I believe a keyword different than "anonymous" serves this purpose better.
> For example, we don't have to consider the subtle difference between
> "presentation restricted" and "address unavailable". My suggestion is
> "telurifmt", i.e.:
> 
> Contact: <sip:telurifmt;tgrp=foo@example.com>.

Personally, I have no problem with this; however, I am a
little confused...I believe that we were avoiding registering
a new token with IANA, hence "anonymous" was suggested since its use
and semantics are well established in the field.

Using "telurifmt" puts us back to the same place: what if
"telurifmt" actually means something to someone?

If you don't think this will be a problem, then I am more
inclined to proceed on it and close this issue down.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From iptel-bounces@ietf.org  Fri Jan  7 10:06:02 2005
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From: "Shan Lu" <shanlu@sentito.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Subject: RE: [Iptel] Finalizing trunk-group I-D
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>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of Vijay K. Gurbani
>Sent: Thursday, January 06, 2005 3:44 PM
>To: Shan Lu
>Cc: 'IETF IPTEL WG'; tu-sawada@kddi.com
>Subject: Re: [Iptel] Finalizing trunk-group I-D
>
>
<snip>
>
>Personally, I have no problem with this; however, I am a
>little confused...I believe that we were avoiding registering
>a new token with IANA, hence "anonymous" was suggested since its use
>and semantics are well established in the field.
>

True. But none of existing usages carries the same semantic we want for
tgrp. Worse, they could interfere with each other.
   
>Using "telurifmt" puts us back to the same place: what if
>"telurifmt" actually means something to someone?
>
>If you don't think this will be a problem, then I am more
>inclined to proceed on it and close this issue down.
>

Pick your favorite poison :-). My vote is for a new keyword different than
"anonymous".

Thanks, Shan Lu


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From iptel-bounces@ietf.org  Thu Jan 13 18:25:21 2005
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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Subject: [Iptel] Is trunk group a new namespace?
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Folks,

In order to close the last open issue with trunk-group, I had
a series of email exchanges with the WG chairs, the
results of which now need to go before the WG to decide how to
move forward.

The last open issue was how to resolve the case where an
incoming call request does not have a calling party's number
available.  How do we indicate the originating trunk group?

We had almost reached a solution in the form of the following
URI: Contact: <sip:telurifmt;tgrp=foo@example.com> (the
I-D exhorts originating trunk groups to go in the Contact
header).

In the email exchange with the WG chairs, Jonathan has
suggested a couple of issues: first, the fact that a trunk
group can exist without a phone number appears to imply that
it is cannot be associated with a tel URI and may, in fact,
indicate that this is a new resource complete with its
own scheme.  Something like:

    trgrp:trunk-group-id[:phone-number]@domain

Second, saving the originating trunk group in the Contact
header is not semantically correct; if the originating trunk
group is to be used for routing, then Contact is not the
semantically correct header to save this information in.

Here is my take on these issues.  The second one is easier
to tackle, so let me try that first.  Jonathan is accurate
in pointing out the semantic use of Contact header.  How-
ever, we reached a decision long time ago in the life of
the trunk-group I-D that we would like NOT to have parameters
like ;o-tgrp=xxx;t-tgrp=yyy in the R-URI.  Hence we
decided to put the originating trunk group in the Contact
header with the understanding that when it is used for
routing in the reverse side, it will appear in the R-URI
position.  But what if it is used in routing on the initiating
side?

The first issue -- trunk-group being a new resource --
is an interesting one.  It has ramificiations, if we decide
to go that route.  Where will it appear?  In a new header(s)?
as parameters to the R-URI or the Contact URI?  We will need
a short paragraph on how to convert trgrp URIs to sip URIs
or tell URIs, ...

So, we have some choices to make:

Regarding putting originating trunk group information: do we
want to leave it in the Contact header or find a new place
for it?

Regarding representing trunk group information itself: do we
want hash out more abstractly the relationship between the
trunk group and the phone number from a namespace prespective?

To make matters worse, I will point out that we already have
a few implementations that are following
http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt.
So depending on what we decide, the implementations will be impacted.

Opinions and comments welcome.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Subject: [Iptel] Implementing "language-switch" tag in CPL
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I find the documentation of the language-switch tag (RFC3880 chapter 
4.3) unclear. Lets take this example script (modified version of 
example in RFC 3880 chapter 12.5):
   	.....
          <language-switch>
              <not-present>
		.....
              </not-present>
              <language matches="es">
                <location url="sip:spanish@operator.example.com">
                  <proxy/>
                </location>
              </language>
              <language matches="en">
              <otherwise>
                <location url="sip:english@operator.example.com">
                  <proxy/>
                </location>
              </language>
              <otherwise>
		.....
              </otherwise>
            </language-switch>
	.....

There are several possible cases here:

* the SIP request supplied no accept-language
- execute the content of not-present.

* the SIP request supplied accept-language that didn't match any of the 
languages, in this case spanish (es) and english (en)
-  execute the content of otherwise.

* the SIP request supplied accept-language that matches e.g.:
"Accept-Language: en, es;q=0.8 da;q=0.0"

+ if matching is done in regular switch order (as language-switch is 
documented to be a switch in the RFC - it's part of chapter 4 - 
"Switches") the first <language matches="es"> condition, would be 
matched so the caller would end up at a spanish operator, even though 
he/she has a higher q-value for "en" (= "en;q=1").

+ This seams odd, as this would mean that the callers q-values are 
completely ignored, this interpretation also appears to be in 
contradiction with the statement "Languages with a 'q' value of 0 are 
also ignored." - RFC 3880 chapter 4.3. (why are q-value = 0 treated 
specially ?).

+ The HTTP 1.1 specification (in regards to accept-language) appears to 
imply that each <language matches="..."> condition is assigned a 
q-value - same as the matching language-range in the request or 0 if 
not supplied in the request. The condition with the largest q-value is 
then selected and executed - is this how it is supposed to work ?


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From: Jonathan Lennox <lennox@cnr.cs.columbia.edu>
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To: =?ISO-8859-1?Q?Administrat=F6r?= <hakan.stenholm@it.su.se>
Subject: Re: [Iptel] Implementing "language-switch" tag in CPL
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I agree that this is somewhat unfortunate.

The problem is, fundamentally, that the "first-match" semantics of CPL
switches don't mesh very well with the "best-match" semantics of HTTP/S=
IP
Accept-Language.

To resolve this, I decided to just impose CPL's semantics, to keep CPL
engines from having to scan an entire language-switch before deciding w=
hich
output to take.

Conceivably this was the wrong choice; a CPL extension called something=
 like
best-match-language-switch (in an appropriate namespace) might be in or=
der.

On Friday, January 14 2005, "Administrat=F6r" wrote to "iptel@ietf.org"=
 saying:

> I find the documentation of the language-switch tag (RFC3880 chapter=20=

> 4.3) unclear. Lets take this example script (modified version of=20
> example in RFC 3880 chapter 12.5):
>    =09.....
>           <language-switch>
>               <not-present>
> =09=09.....
>               </not-present>
>               <language matches=3D"es">
>                 <location url=3D"sip:spanish@operator.example.com">
>                   <proxy/>
>                 </location>
>               </language>
>               <language matches=3D"en">
>               <otherwise>
>                 <location url=3D"sip:english@operator.example.com">
>                   <proxy/>
>                 </location>
>               </language>
>               <otherwise>
> =09=09.....
>               </otherwise>
>             </language-switch>
> =09.....
>=20
> There are several possible cases here:
>=20
> * the SIP request supplied no accept-language
> - execute the content of not-present.
>=20
> * the SIP request supplied accept-language that didn't match any of t=
he=20
> languages, in this case spanish (es) and english (en)
> -  execute the content of otherwise.
>=20
> * the SIP request supplied accept-language that matches e.g.:
> "Accept-Language: en, es;q=3D0.8 da;q=3D0.0"
>=20
> + if matching is done in regular switch order (as language-switch is=20=

> documented to be a switch in the RFC - it's part of chapter 4 -=20
> "Switches") the first <language matches=3D"es"> condition, would be=20=

> matched so the caller would end up at a spanish operator, even though=
=20
> he/she has a higher q-value for "en" (=3D "en;q=3D1").
>=20
> + This seams odd, as this would mean that the callers q-values are=20=

> completely ignored, this interpretation also appears to be in=20
> contradiction with the statement "Languages with a 'q' value of 0 are=
=20
> also ignored." - RFC 3880 chapter 4.3. (why are q-value =3D 0 treated=
=20
> specially =3F).
>=20
> + The HTTP 1.1 specification (in regards to accept-language) appears =
to=20
> imply that each <language matches=3D"..."> condition is assigned a=20=

> q-value - same as the matching language-range in the request or 0 if=20=

> not supplied in the request. The condition with the largest q-value i=
s=20
> then selected and executed - is this how it is supposed to work =3F

--=20
Jonathan Lennox
lennox@cs.columbia.edu

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From: =?ISO-8859-1?Q?Administrat=F6r?= <hakan.stenholm@it.su.se>
Subject: Re: [Iptel] Implementing "language-switch" tag in CPL
Date: Mon, 17 Jan 2005 14:14:23 +0100
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I don't have any real experience working in the RFC process, in regards
to how and to what extent a RFC can be modified - but here are my
thoughts and opinions about language-switches:

* Replace it with "language-select", described below (or similar CPL
   Node).
* Add "language-select" as some kind of extension.
* is language-switch needed and truly useful or could it be made
   optional / removed from the RFC (3880) ?:

- We, in this case Stockholm University how work on
   http://www.stacken.kth.se/projekt/yxa/ don't see any immediate need
   for this functionality, this may of course apply to other parties.
- Are regular users actually going supply reasonable.
   "accepted-language" values, or will 90% of all phones send some kind
   of useless default or supply no values at all ? - For language-switch
   to be truly useful, almost all phone must supply correct values for
   "accepted-language", for CPL scripts to be able to rely on this
   information.

-----------------------------------------------------------------------
Notes on proposed language-select below:

* I tried to model it after "proxy" - with matching conditions
   "language" and "default" a no matching condition.

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

Language Selects

    Language selects allow a CPL script to make decisions based on the
    languages in which the originator of the call wishes to communicate.
    They are summarized in Figure X. This a modified version of
    "language-switch" acting in a manner more compatible with other
    related RFCs like RFC 2616 (see chapter 14.4).

             Node:  "language-select"
          Outputs:  "language"         Specific string to match
       Parameters:  None

           Output:  "language"
       Parameters:  "matches"          Match if the given language
                                       matches a language-range of the
                                       call.
           Output:  "default"
       Parameters:  none

       Figure X: Syntax of the "language-select" node

    Language selects take no parameters.

    The "language" output takes one parameter, "matches".  The value of
    the parameter is a language-tag, as defined in RFC 3066.  The
    caller may have specified a set of language-ranges, also as defined
    in RFC 3066. The match is done as follows:

    * Each language-tag (i.e. each "matches") is matched against the
      language-ranges supplied by the request, each language-tag value =
is
      paired with the best "q" value found, "q" =3D 0 if no value was
      supplied.
      The "language" output with the best fit is then executed, best =
fit:
      - select all "language"-"q" pairs, that have the maximum "q" =
value.
      - select all the "language" outputs that have the highest number =
of
        segements =3D number of "-" separators + 1.
      - if there are still identical matches the "language" output will
        be selected in the order they are declared in "language-select".

    * If no "language" output matches, the "default" output is selected.
      This happens if the request supplied no language information or if
      all the "language"-"q" pairs got "q" =3D 0.

    See RFC 3066 for the details of how language-ranges match language-
    tags.  Briefly, a language-range matches a language-tag if it =
exactly
    equals the tag, or if it exactly equals a prefix of the tag such =
that
    the first character following the prefix is "-".

    If the caller specified the special language-range "*", it is =
ignored
    for the purpose of matching.  Languages with a "q" value of 0 are
    also ignored ().

Usage of "language-select" with SIP

    The language-ranges for the "language-select" select are obtained
    from the SIP "Accept-Language" header field.  The select is not-
    present if the initial SIP request did not contain this header =
field.
-----------------------------------------------------------------------

On 2005-01-14, at 23.05, Jonathan Lennox wrote:

> I agree that this is somewhat unfortunate.
>
> The problem is, fundamentally, that the "first-match" semantics of CPL
> switches don't mesh very well with the "best-match" semantics of=20
> HTTP/SIP
> Accept-Language.
>
> To resolve this, I decided to just impose CPL's semantics, to keep CPL
> engines from having to scan an entire language-switch before deciding=20=

> which
> output to take.
>
> Conceivably this was the wrong choice; a CPL extension called=20
> something like
> best-match-language-switch (in an appropriate namespace) might be in=20=

> order.
>
> On Friday, January 14 2005, "Administrat=F6r" wrote to =
"iptel@ietf.org"=20
> saying:
>
>> I find the documentation of the language-switch tag (RFC3880 chapter
>> 4.3) unclear. Lets take this example script (modified version of
>> example in RFC 3880 chapter 12.5):
>>    	.....
>>           <language-switch>
>>               <not-present>
>> 		.....
>>               </not-present>
>>               <language matches=3D"es">
>>                 <location url=3D"sip:spanish@operator.example.com">
>>                   <proxy/>
>>                 </location>
>>               </language>
>>               <language matches=3D"en">
>>               <otherwise>
>>                 <location url=3D"sip:english@operator.example.com">
>>                   <proxy/>
>>                 </location>
>>               </language>
>>               <otherwise>
>> 		.....
>>               </otherwise>
>>             </language-switch>
>> 	.....
>>
>> There are several possible cases here:
>>
>> * the SIP request supplied no accept-language
>> - execute the content of not-present.
>>
>> * the SIP request supplied accept-language that didn't match any of=20=

>> the
>> languages, in this case spanish (es) and english (en)
>> -  execute the content of otherwise.
>>
>> * the SIP request supplied accept-language that matches e.g.:
>> "Accept-Language: en, es;q=3D0.8 da;q=3D0.0"
>>
>> + if matching is done in regular switch order (as language-switch is
>> documented to be a switch in the RFC - it's part of chapter 4 -
>> "Switches") the first <language matches=3D"es"> condition, would be
>> matched so the caller would end up at a spanish operator, even though
>> he/she has a higher q-value for "en" (=3D "en;q=3D1").
>>
>> + This seams odd, as this would mean that the callers q-values are
>> completely ignored, this interpretation also appears to be in
>> contradiction with the statement "Languages with a 'q' value of 0 are
>> also ignored." - RFC 3880 chapter 4.3. (why are q-value =3D 0 treated
>> specially ?).
>>
>> + The HTTP 1.1 specification (in regards to accept-language) appears=20=

>> to
>> imply that each <language matches=3D"..."> condition is assigned a
>> q-value - same as the matching language-range in the request or 0 if
>> not supplied in the request. The condition with the largest q-value =
is
>> then selected and executed - is this how it is supposed to work ?
>
> --=20
> Jonathan Lennox
> lennox@cs.columbia.edu
>


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Flemming Andreasen wrote:

> Greetings
>
> Below, please find review comments for draft-ietf-iptel-tel-np-03.txt.=20
> In general, I think the document is close to being done, but there are=20
> a few minor issues to address as well as some nits.
>
>
> Non-Editorial
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> - Section 4, ABNF
> The production for "global-hex-digits" ends with "*phonedigit-hex". I=20
> don't understand what that is for, not least considering the ensuing=20
> text for "local-rn"
> <quote>
>   For a "local-rn", the routing number in the "rn" parameter MUST be
>   meaningful in terms of "rn-context".  For example, if a national
>   routing number is in the "rn" parameter, the "rn-context" MUST
>   contain a valid E.164 country code after "+" if it is in the
>   "global-hex-digits" format.
> </quote>
>
>
> Section 5.1, 6th paragraph
> "The network node SHALL ignore the "cic" parameter if it identifies a=20
> carrier or service provider associated with that node, or if that=20
> parameter contains a code for indicating that a geographic number is=20
> supplied (e.g., +1-0110 means "local, translated geographical=20
> telephone number provided"). ". Given the normative requirement, we=20
> either need some more text or references to explain how we determine=20
> if the parameter contains a code for indicating that a geographic=20
> number is supplied.
>
> Section 5.1, 6th paragraph
> "The network node SHALL remove the "cic" parameter and look for the=20
> "rn" parameter for making the routing decision." I don't see why the=20
> network node SHALL remove the "cic" parameter and would in fact argue=20
> for at most a MAY. I agree it shall look for the "rn" parameter for=20
> making the routing decision though.
>
> Section 5.1, 9th and 10th paragraph.
> Again, I don't see why "the network SHALL remove the "rn" parameter".
>
> Section 5.2.4, 1st paragraph
> "A Public Switched Telephone Network (PSTN) gateway needs to convert
>   between SS7 ISUP and the VoIP protocol such as SIP or H.323.  This
>   type of network node SHALL add the corresponding information from
>   the ISUP to the defined parameters to the "tel" URI for routing and
>   the "tel" URI associated with the caller and vice versa."
> I disagree with the latter part ("and the "tel" URI associated with=20
> the caller and vice versa"). We are getting into privacy and security=20
> related issues and hence cannot make it mandatory for all.
>
> Section 7
> Need to add security considerations for using the extensions for the=20
> caller (JIP considerations)
>
>
> Editorial and/or nits:
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> - Global
> Three spaces after period in several places.
> Not sure if formatting adheres to RFC Editor guidelines (at least I=20
> can't get it to print with proper page breaks)
>
> - Abstract
> Should not describe the actual extension parameters in the abstract=20
> but rather describe the problem and outline the solution.
>
> - Abstract  "that NP database dip" =3D> "that an NP database dip"
>                                ^^
> - Section 2, 1st paragraph
> "It has been identified that NP has impacts on several=20
> works-in-progress at the IETF." =3D>
>
> "NP impacts call signaling." or something like that.
>
> - Section 2, 3rd paragraph
> "call set up" =3D> "call setup"
>
> - Section 2, 5th paragraph
> The sentence
>    "Section 5 describes the rules for a network node that deals with=20
> some or all of the defined parameters in a "tel" URI"
> is not accurate since we are only talking about some specific=20
> extensions here. The phrase "defined parameters" is used throughout=20
> the document to refer to the extensions defined in this document. I'd=20
> suggest using another more accurate phrase throughout.
>
> - Section 4, 1st paragraph
> Add some words explaining how this ABNF simply defines new parameters=20
> in accordance with the ABNF for "tel" URI provided in RFC 3966.
>
> - Section 4, 5th paragraph
> "...the routing number in the "rn" parameter MUST be meaningful in=20
> terms of "rn-context". Although the example that follows helps clarify=20
> it for E.164 numbers, it is in general not well defined what=20
> "meamingful" means. Please rephrase to make it clearer.
>
> - Section 4, last paragraph
> "...the CIC value in the "cic" parameter MUST be meaningful in terms=20
> of "cic-context". Semantics unclear - please rephrase to make it cleare=
r.
>
> Section 5.1, 4th paragraph
> "or routing information" =3D> "or routing number information" (?)
>
> Section 5.1, last paragraph
> "making routing decision" =3D> "making routing decisions"
>                                                     ^
> "set to do so or may route the call to a designated network node that=20
> will access the NP database or may route the call" =3D>
>              ^^
> "set to do so, it may route the call to a designated network node that=20
> will access the NP database, or it may route the call"
>            =20
> ^^^                                                                    =
           =20
>  ^    ^^
>
> Section 5.2.2, 6th paragraph
> "telephone number that is the typical case for the second freephone=20
> database access, the" =3D>
>
> "telephone number (which is the typical case for the second freephone=20
> database access), the"
>                 =20
> ^^^^^^                                                             ^
>
> Section 5.2.3, 2nd paragraph
> "For example, if the network node is a PSTN gateway that receives an=20
> ISUP message that contains the JIP, the "rn" parameter in the "tel"=20
> URI would normally contain the correct location information." Please=20
> rephrase (it is the ISUP message that contains the correct location=20
> information which is therefore placed in the "rn" parameter in the=20
> "tel" URI).
>
>
> Section 5.2.4, 2nd paragraph
> "put it after the "tel:" in the "tel" URI that is used for routing" =3D=
>
>
> "put it in the global-number-digits or local-number-digits (see RFC=20
> 3966) part of the "tel" URI that is used for routing"
>       =20
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^=
^^
>
> Section 5.2.4, 3rd paragraph
> "put it after "tel:" in the "tel" URI that is used for routing" =3D>
>
> "put it in the global-number-digits or local-number-digits (see RFC=20
> 3966) part of the "tel" URI that is used for routing"
>       =20
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^=
^^
>
> Section 5.2.4, 3rd paragraph
> "No "rn" SHALL appear in the "tel" URI." =3D>
>
> "An "rn" SHALL NOT appear in the "tel" URI."
> ^^            ^^^
>
> Section 5.2.4, 3rd paragraph
> "It is outside the scope of this document how to include the "rn"=20
> parameter if the local policies require the network node to do so." I=20
> don't follow - the previous sentence just said that you are not=20
> allowed to do so.
>
> Section 6, example B
> "provider=C6s" =3D> "provider's"
>
> Section 6, example E
> "   E. A "tel" URI, tel:+1-202-533-1234;rn=3D+1-202-000-0000;npdi, cont=
ains
>     an invalid routing number (e.g., no routing information on "+1-
>     202-000"), "
> Example seems to assume prefix-based routing - might be good to=20
> clarify that.
>
> Section 8 and 9
> There are gaps in the reference numbers. In general, it would probably=20
> be better to use symbolic references instead of plain numbers.
> 2806bis-09 is now RFC 3966.
>
>
> -- Flemming
>
>
>
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>

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Date: Sun, 16 Jan 2005 14:42:19 -1000
Subject: Re: [Iptel] Is trunk group a new namespace?
From: Cullen Jennings <fluffy@cisco.com>
To: "Vijay K. Gurbani" <vkg@lucent.com>, "iptel@ietf.org" <iptel@ietf.org>
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inline comments - not as chair..

Summary - I am very against the new URI type idea. It is not deployable -
more inline

On 1/13/05 3:06 PM, "Vijay K. Gurbani" <vkg@lucent.com> wrote:

> Folks,
> 
> In order to close the last open issue with trunk-group, I had
> a series of email exchanges with the WG chairs, the
> results of which now need to go before the WG to decide how to
> move forward.
> 
> The last open issue was how to resolve the case where an
> incoming call request does not have a calling party's number
> available.  How do we indicate the originating trunk group?
> 
> We had almost reached a solution in the form of the following
> URI: Contact: <sip:telurifmt;tgrp=foo@example.com> (the
> I-D exhorts originating trunk groups to go in the Contact
> header).
>

I don't like the making up telurifmt as a special SIP user name. However,
what the user port of a contact is is a GW local issue and not defined by
SIP. The GW could put line1 or ds0-22 or a random number. This problem only
comes up in the context of what originating information that is currently in
the contact header.
 
> In the email exchange with the WG chairs, Jonathan has
> suggested a couple of issues: first, the fact that a trunk
> group can exist without a phone number appears to imply that
> it is cannot be associated with a tel URI and may, in fact,
> indicate that this is a new resource complete with its
> own scheme.  Something like:
> 
>     trgrp:trunk-group-id[:phone-number]@domain

The trunk group without a phone number would clearly be useless as a
destination. As a GW source information, I'm not keen on it. Were would you
put it? In the Contact header? You would have to map it to a sip URI to do
this which would result in a SIP uri in the contact that looks almost
identical to what everyone has already implemented. This would be a huge
amount of work that in the end resulted in protocol on the wire that was
semantically identical and differed by at most a few bits in syntax.

I am really against the new URI type idea - many SIP elements support tel
and the current proposal works in a backwards compatible way with all those
devices. If we make a new trgrp URI, no proxy or UA will know about it or be
able to do anything with it. There is no migration path from tel to trgrp
until they are all replaced. No one will every do it because they are
already using tel. 

> 
> Second, saving the originating trunk group in the Contact
> header is not semantically correct; if the originating trunk
> group is to be used for routing, then Contact is not the
> semantically correct header to save this information in.
>

I'm not sure I agree with this. The Contact identifies the resource on the
UA. The trunk group is part of the resource name on the GW. Saying you can
only route on things in the request URI it totally bogus. I'll point at CPL
for example. It is fine to route on many things in the SIP message. The
contact header, as used in a registration if pretty fundamental to how
requests get routed in SIP. GRU is going to work because some elements are
going to look at tag in the contact and route based on them.

 > Here is my take on these issues.  The second one is easier
> to tackle, so let me try that first.  Jonathan is accurate
> in pointing out the semantic use of Contact header.  How-
> ever, we reached a decision long time ago in the life of
> the trunk-group I-D that we would like NOT to have parameters
> like ;o-tgrp=xxx;t-tgrp=yyy in the R-URI.  Hence we
> decided to put the originating trunk group in the Contact
> header with the understanding that when it is used for
> routing in the reverse side, it will appear in the R-URI
> position.  But what if it is used in routing on the initiating
> side?
> 
> The first issue -- trunk-group being a new resource --
> is an interesting one.  It has ramificiations, if we decide
> to go that route.  Where will it appear?  In a new header(s)?
> as parameters to the R-URI or the Contact URI?  We will need
> a short paragraph on how to convert trgrp URIs to sip URIs
> or tell URIs, ...
> 
> So, we have some choices to make:
> 
> Regarding putting originating trunk group information: do we
> want to leave it in the Contact header or find a new place
> for it?

I view the originating trunk group as part of the name of the resource on
the GW and thus reasonable to put in the contact.

> 
> Regarding representing trunk group information itself: do we
> want hash out more abstractly the relationship between the
> trunk group and the phone number from a namespace prespective?
> 
> To make matters worse, I will point out that we already have
> a few implementations that are following
> http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt.
> So depending on what we decide, the implementations will be impacted.
> 
> Opinions and comments welcome.

If someone can point out the harm that results in trunk group being a
parameter of a tel URI that would be fixed by having a separate namespace, I
would probably change my opinion.

> 
> Thanks,
> 
> - vijay


Part of this discussion was brought up by the example of an IAM with no
originating phone number but where the GW still sends an INVITE. In this
case SIP does nto say what to put in the From but this is a SIP/tel problem
not a trunk group problem.

My proposal .... More or less keep what we have and

Allow a trgp and phone-context parameter in a tel URI.

Also allows these is a SIP uri even if that SIP URI is not mapable to a
valid tel URI

When this is seen in a contact, it identifies the PSTN associated resource
on the GW identified by the contact header

When this is seen in a request URI, it specifies the desired trunk group for
reaching the PSTN resource identified in the request URI

We have been discussing this for a long long time as a WG item. Unless
someone has some specific harm or problem caused by the proposed solutions,
I'm not keen on totally throwing it out.

PS - I'm sick and can't think straight - I lost my crypto token and can't
run my VPN so can't do email so have no idea when I will send this or
receive responses. I reserve the option to claim my evil twin skip wrote
this and I disagree with all of it :-) Someone convince me I'm nuts.







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Subject: Re: [Iptel] Implementing "language-switch" tag in CPL
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On Monday, January 17 2005, "Administrat=F6r" wrote to "iptel@ietf.org,=
 Jonathan Lennox" saying:

> I don't have any real experience working in the RFC process, in regar=
ds
> to how and to what extent a RFC can be modified - but here are my
> thoughts and opinions about language-switches:

The best approach would be to define language-select (or whatever you w=
ant
to call it) as an extension to CPL.  See section 11 of RFC 3880 for det=
ails
on how CPL extensions work.  Since CPL extensions are defined using XML=

namespaces, they don't necessarily have to go through the IETF
standardization process (though if they're generally useful, I'd encour=
age
them to be).

> - Are regular users actually going supply reasonable.
>    "accepted-language" values, or will 90% of all phones send some ki=
nd
>    of useless default or supply no values at all =3F - For language-s=
witch
>    to be truly useful, almost all phone must supply correct values fo=
r
>    "accepted-language", for CPL scripts to be able to rely on this
>    information.

This I don't know.  If your implementation targets SIP, I'd suggest ask=
ing
on the SIP-Implementors mailing list.  See
<http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors>.

--=20
Jonathan Lennox
lennox at cs dot columbia dot edu

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Date: Thu, 20 Jan 2005 20:18:49 -0500
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To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Iptel] Is trunk group a new namespace?
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Also not as chair.

I've considered this further, and concluded that here is a case where 
the pragmatic approach in the current draft is probably a better choice 
than the more theoretical considerations that led me to consider the 
idea of a new namespace.

As such, I agree with everything cullen suggests, but disagree on one 
point. Cullen suggests:

 > Allow a trgp and phone-context parameter in a tel URI.
 >
 > Also allows these is a SIP uri even if that SIP URI is not mapable to a
 > valid tel URI

I'd rather not do that. Instead, I'd rather declare this problem 
(indicating a trunk group without a phone number) as out of scope. This 
is not a tel URI issue at all - its an ISUP to SIP mapping issue. Our 
job is to define a way to extend the tel URI to be able to indicate 
trunk groups, and nothing more.

Thanks,
Jonathan R.

Cullen Jennings wrote:

> inline comments - not as chair..
> 
> Summary - I am very against the new URI type idea. It is not deployable -
> more inline
> 
> On 1/13/05 3:06 PM, "Vijay K. Gurbani" <vkg@lucent.com> wrote:
> 
> 
>>Folks,
>>
>>In order to close the last open issue with trunk-group, I had
>>a series of email exchanges with the WG chairs, the
>>results of which now need to go before the WG to decide how to
>>move forward.
>>
>>The last open issue was how to resolve the case where an
>>incoming call request does not have a calling party's number
>>available.  How do we indicate the originating trunk group?
>>
>>We had almost reached a solution in the form of the following
>>URI: Contact: <sip:telurifmt;tgrp=foo@example.com> (the
>>I-D exhorts originating trunk groups to go in the Contact
>>header).
>>
> 
> 
> I don't like the making up telurifmt as a special SIP user name. However,
> what the user port of a contact is is a GW local issue and not defined by
> SIP. The GW could put line1 or ds0-22 or a random number. This problem only
> comes up in the context of what originating information that is currently in
> the contact header.
>  
> 
>>In the email exchange with the WG chairs, Jonathan has
>>suggested a couple of issues: first, the fact that a trunk
>>group can exist without a phone number appears to imply that
>>it is cannot be associated with a tel URI and may, in fact,
>>indicate that this is a new resource complete with its
>>own scheme.  Something like:
>>
>>    trgrp:trunk-group-id[:phone-number]@domain
> 
> 
> The trunk group without a phone number would clearly be useless as a
> destination. As a GW source information, I'm not keen on it. Were would you
> put it? In the Contact header? You would have to map it to a sip URI to do
> this which would result in a SIP uri in the contact that looks almost
> identical to what everyone has already implemented. This would be a huge
> amount of work that in the end resulted in protocol on the wire that was
> semantically identical and differed by at most a few bits in syntax.
> 
> I am really against the new URI type idea - many SIP elements support tel
> and the current proposal works in a backwards compatible way with all those
> devices. If we make a new trgrp URI, no proxy or UA will know about it or be
> able to do anything with it. There is no migration path from tel to trgrp
> until they are all replaced. No one will every do it because they are
> already using tel. 
> 
> 
>>Second, saving the originating trunk group in the Contact
>>header is not semantically correct; if the originating trunk
>>group is to be used for routing, then Contact is not the
>>semantically correct header to save this information in.
>>
> 
> 
> I'm not sure I agree with this. The Contact identifies the resource on the
> UA. The trunk group is part of the resource name on the GW. Saying you can
> only route on things in the request URI it totally bogus. I'll point at CPL
> for example. It is fine to route on many things in the SIP message. The
> contact header, as used in a registration if pretty fundamental to how
> requests get routed in SIP. GRU is going to work because some elements are
> going to look at tag in the contact and route based on them.
> 
>  > Here is my take on these issues.  The second one is easier
> 
>>to tackle, so let me try that first.  Jonathan is accurate
>>in pointing out the semantic use of Contact header.  How-
>>ever, we reached a decision long time ago in the life of
>>the trunk-group I-D that we would like NOT to have parameters
>>like ;o-tgrp=xxx;t-tgrp=yyy in the R-URI.  Hence we
>>decided to put the originating trunk group in the Contact
>>header with the understanding that when it is used for
>>routing in the reverse side, it will appear in the R-URI
>>position.  But what if it is used in routing on the initiating
>>side?
>>
>>The first issue -- trunk-group being a new resource --
>>is an interesting one.  It has ramificiations, if we decide
>>to go that route.  Where will it appear?  In a new header(s)?
>>as parameters to the R-URI or the Contact URI?  We will need
>>a short paragraph on how to convert trgrp URIs to sip URIs
>>or tell URIs, ...
>>
>>So, we have some choices to make:
>>
>>Regarding putting originating trunk group information: do we
>>want to leave it in the Contact header or find a new place
>>for it?
> 
> 
> I view the originating trunk group as part of the name of the resource on
> the GW and thus reasonable to put in the contact.
> 
> 
>>Regarding representing trunk group information itself: do we
>>want hash out more abstractly the relationship between the
>>trunk group and the phone number from a namespace prespective?
>>
>>To make matters worse, I will point out that we already have
>>a few implementations that are following
>>http://www.ietf.org/internet-drafts/draft-ietf-iptel-trunk-group-02.txt.
>>So depending on what we decide, the implementations will be impacted.
>>
>>Opinions and comments welcome.
> 
> 
> If someone can point out the harm that results in trunk group being a
> parameter of a tel URI that would be fixed by having a separate namespace, I
> would probably change my opinion.
> 
> 
>>Thanks,
>>
>>- vijay
> 
> 
> 
> Part of this discussion was brought up by the example of an IAM with no
> originating phone number but where the GW still sends an INVITE. In this
> case SIP does nto say what to put in the From but this is a SIP/tel problem
> not a trunk group problem.
> 
> My proposal .... More or less keep what we have and
> 
> Allow a trgp and phone-context parameter in a tel URI.
> 
> Also allows these is a SIP uri even if that SIP URI is not mapable to a
> valid tel URI
> 
> When this is seen in a contact, it identifies the PSTN associated resource
> on the GW identified by the contact header
> 
> When this is seen in a request URI, it specifies the desired trunk group for
> reaching the PSTN resource identified in the request URI
> 
> We have been discussing this for a long long time as a WG item. Unless
> someone has some specific harm or problem caused by the proposed solutions,
> I'm not keen on totally throwing it out.
> 
> PS - I'm sick and can't think straight - I lost my crypto token and can't
> run my VPN so can't do email so have no idea when I will send this or
> receive responses. I reserve the option to claim my evil twin skip wrote
> this and I disagree with all of it :-) Someone convince me I'm nuts.
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [Iptel] Is trunk group a new namespace?
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Jonathan Rosenberg wrote:
> Also not as chair.
> 
> I've considered this further, and concluded that here is a case where 
> the pragmatic approach in the current draft is probably a better choice 
> than the more theoretical considerations that led me to consider the 
> idea of a new namespace.

Jonathan:

OK; pragmatism wins this one.

> As such, I agree with everything cullen suggests, but disagree on one 
> point. Cullen suggests:
> 
>  > Allow a trgp and phone-context parameter in a tel URI.
>  >
>  > Also allows these is a SIP uri even if that SIP URI is not mapable to a
>  > valid tel URI
> 
> I'd rather not do that. Instead, I'd rather declare this problem 
> (indicating a trunk group without a phone number) as out of scope. This 
> is not a tel URI issue at all - its an ISUP to SIP mapping issue. Our 
> job is to define a way to extend the tel URI to be able to indicate 
> trunk groups, and nothing more.

As I have so learnt in the past couple of weeks, the calling number
is optional in ISUP.  So, I would also agree that mapping an
absent ISUP parameter to a SIP element is an ISUP to SIP mapping
issue and outside the scopr of the trunk-group I-D.

I will consider this issue resolved, and will rev up the draft
with the WGLC and other discussions shortly.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>,
        "'Cullen Jennings'" <fluffy@cisco.com>
Subject: RE: [Iptel] Is trunk group a new namespace?
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All,

First, I agree with the consensus that trunk group should not be a new
namespace.

More inline.

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Jonathan Rosenberg
>Sent: Thursday, January 20, 2005 8:19 PM
>To: Cullen Jennings
>Cc: iptel@ietf.org; Vijay K. Gurbani
>Subject: Re: [Iptel] Is trunk group a new namespace?

<snip>
>
>
>I'd rather not do that. Instead, I'd rather declare this problem=20
>(indicating a trunk group without a phone number) as out of=20
>scope.=20

Declaring out of scope does not make the problem go away, does it? I =
would
further argue that not solving the problem would severely limit the
proposal's usability. Think of calls that requested privacy. A solution =
that
can fail on any given call is not a real solution.

But you have a point that the I-D went out of its way to address the
problem. The real issue, in my opinion, lies in RFC3966. Had 3966 =
allowed
tel uri to have empty phone digits (but meaningful parameters), we won't
have this discussion at all. As a side note, sip uri allows completely =
bogus
userinfo/hostport but valid uri parameters.

The proposal in the I-D may be ugly and the I-D may even be the wrong =
avenue
to address the issue. But that's what you get for not doing right at =
first
try in 3966. Not to mention it does solve a real issue.

Regards,

Shan Lu


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To: Shan Lu <shanlu@sentito.com>
Subject: Re: [Iptel] Is trunk group a new namespace?
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Shan Lu wrote:
> Declaring out of scope does not make the problem go away, does it? 
> I would further argue that not solving the problem would severely 
> limit the proposal's usability. Think of calls that requested 
> privacy. A solution that can fail on any given call is not 
> a real solution.

Cullen, in an earlier email, wrote the following which I think is
worth bringing up:

    Now the problem came up when trying to overload where a call
    came from into the contact header field value. In this case
    people are trying to specify the trunk group but leave the
    telephone number unspecified. However, the contact identifies
    the GW (and the trunk group on GW) - it does not necessarily
    identify the PSTN endpoint that called the originating GW.
    The From would identify the originating endpoint on the PSTN.
    Keep in mind the calling party in the IAM gets mapped to the
    From not the contact header - it is a far end issue. The
    trunk group, which is a local to the GW issues, is what makes
    sense to map into the contact for the GW. A GW could form the
    user portion of the contact by just saying which DS0 was in
    use and adding the logical trunk group.

The last sentence above provides a decent way out to generate a
user portion and add the trunk-group parameters to it without
us having to mandate a certain user token.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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From: "Shan Lu" <shanlu@sentito.com>
To: "'Vijay K. Gurbani'" <vkg@lucent.com>
Subject: RE: [Iptel] Is trunk group a new namespace?
Date: Fri, 21 Jan 2005 13:55:39 -0500
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Vijay,

Is it the proposal to use the following:

Contact:<sip:1;tgrp=3D2@3.4.5.6>, where DS0 number is 1 and TG is 2?

The why don't you use the same format even when calling number is =
available?
I have no issue with it, but don't call it tel-uri because it isn't.

Regards,=20

Shan Lu
>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Vijay K. Gurbani
>Sent: Friday, January 21, 2005 11:36 AM
>To: Shan Lu
>Cc: iptel@ietf.org; Cullen Jennings
>Subject: Re: [Iptel] Is trunk group a new namespace?
>
>
>Shan Lu wrote:
>> Declaring out of scope does not make the problem go away, does it?=20
>> I would further argue that not solving the problem would severely=20
>> limit the proposal's usability. Think of calls that requested=20
>> privacy. A solution that can fail on any given call is not=20
>> a real solution.
>
>Cullen, in an earlier email, wrote the following which I think is
>worth bringing up:
>
>    Now the problem came up when trying to overload where a call
>    came from into the contact header field value. In this case
>    people are trying to specify the trunk group but leave the
>    telephone number unspecified. However, the contact identifies
>    the GW (and the trunk group on GW) - it does not necessarily
>    identify the PSTN endpoint that called the originating GW.
>    The From would identify the originating endpoint on the PSTN.
>    Keep in mind the calling party in the IAM gets mapped to the
>    From not the contact header - it is a far end issue. The
>    trunk group, which is a local to the GW issues, is what makes
>    sense to map into the contact for the GW. A GW could form the
>    user portion of the contact by just saying which DS0 was in
>    use and adding the logical trunk group.
>
>The last sentence above provides a decent way out to generate a
>user portion and add the trunk-group parameters to it without
>us having to mandate a certain user token.
>
>- vijay
>--=20
>Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
>Wireless Networks Group/Internet Software and Services
>Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
>Naperville, Illinois 60566     Voice: +1 630 224 0216
>
>_______________________________________________
>Iptel mailing list
>Iptel@ietf.org
>https://www1.ietf.org/mailman/listinfo/iptel
>


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Subject: Re: [Iptel] Is trunk group a new namespace?
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Shan Lu wrote:
> Vijay,
> 
> Is it the proposal to use the following:
> 
> Contact:<sip:1;tgrp=2@3.4.5.6>, where DS0 number is 1 and TG is 2?

Yes, something to that effect.  I will flesh it out further
in the rev'd I-D.

> The why don't you use the same format even when calling number 
> is available?

The I-D, as it currently stands, is focused on where to place
the originating trunk group, not what the user part of the
Contact SIP URI should be.  If an implementation wants to
substitute something other than the calling party number (if
present), they can do so.  On the other hand, if the calling
party number is not present, the next rev of the I-D will
provide some guidelines on what to populate in the user part.

> I have no issue with it, but don't call it tel-uri because it isn't.

I like your "no issue with it" part ;-)

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216

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Subject: RE: [Iptel] Is trunk group a new namespace?
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>From: Vijay K. Gurbani <vkg@lucent.com>
>To: Shan Lu <shanlu@sentito.com>
>CC: iptel@ietf.org, Cullen Jennings <fluffy@cisco.com>
>Date: Jan 21, 2005 - 3:32pm

>Shan Lu wrote:
>> Vijay,
>>
>> Is it the proposal to use the following:
>>
>> Contact:<sip:1;tgrp=2@3.4.5.6>, where DS0 number is 1 and TG is 2?

>Yes, something to that effect. I will flesh it out further
>in the rev'd I-D.

Wait. Doesn't this take a full circle back to our earlier discussion? I
thought we agreed upon the need for a keyword, such as "anonymous" or
"teluri", no?

Without such a keyword, how do you inform recipient of the above sip uri
that "tgrp" stands for "Trunk Group" and not "The Gourmet Roasted Pizza"?
Unless you think the DS0 number is sufficient.

Regards,

Shan Lu

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Actually, I don't see why its needed to specify any kind of keyword at all.

Let us remember that we are talking about the contact header field. As 
Cullen has pointed out, it doesn't identify the originator but rather a 
port on a gateway. As such, just because there was no calling number 
doesn't mean one could not use a tel URI to identify the gateway line. 
Seems like something of the form:

tel:<lrn>;phone-context="<provider-domain>"

would suffice, in which case one could add a trunk group to this and 
then turn it into a sip uri:

sip:732766;phone-context=provider.com;tgrp=TG-1@provider.com;user=phone


I don't think that we need to say anything about how the tel URI is 
constructed, merely that any URI that is a reasonable descriptor for the 
gateway "port" resource itself will suffice.

-Jonathan R.



shanlu@sentito.com wrote:

>>From: Vijay K. Gurbani <vkg@lucent.com>
>>To: Shan Lu <shanlu@sentito.com>
>>CC: iptel@ietf.org, Cullen Jennings <fluffy@cisco.com>
>>Date: Jan 21, 2005 - 3:32pm
> 
> 
>>Shan Lu wrote:
>>
>>>Vijay,
>>>
>>>Is it the proposal to use the following:
>>>
>>>Contact:<sip:1;tgrp=2@3.4.5.6>, where DS0 number is 1 and TG is 2?
> 
> 
>>Yes, something to that effect. I will flesh it out further
>>in the rev'd I-D.
> 
> 
> Wait. Doesn't this take a full circle back to our earlier discussion? I
> thought we agreed upon the need for a keyword, such as "anonymous" or
> "teluri", no?
> 
> Without such a keyword, how do you inform recipient of the above sip uri
> that "tgrp" stands for "Trunk Group" and not "The Gourmet Roasted Pizza"?
> Unless you think the DS0 number is sufficient.
> 
> Regards,
> 
> Shan Lu
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Date: Sat, 22 Jan 2005 20:15:55 -0800
Subject: Re: [Iptel] Is trunk group a new namespace?
From: Cullen Jennings <fluffy@cisco.com>
To: "Vijay K. Gurbani" <vkg@lucent.com>,
        Jonathan Rosenberg <jdrosen@cisco.com>
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On 1/21/05 6:03 AM, "Vijay K. Gurbani" <vkg@lucent.com> wrote:

>> I'd rather not do that. Instead, I'd rather declare this problem
>> (indicating a trunk group without a phone number) as out of scope. This
>> is not a tel URI issue at all - its an ISUP to SIP mapping issue. Our
>> job is to define a way to extend the tel URI to be able to indicate
>> trunk groups, and nothing more.
> 
> As I have so learnt in the past couple of weeks, the calling number
> is optional in ISUP.  So, I would also agree that mapping an
> absent ISUP parameter to a SIP element is an ISUP to SIP mapping
> issue and outside the scopr of the trunk-group I-D.
> 
> I will consider this issue resolved, and will rev up the draft
> with the WGLC and other discussions shortly.

This sounds like the best path forward to me. I'm happy with this.

Cullen - not as chair.

Actually as chair, after listening to all the options, I really think this
is the only path that has a good chance of closing this issues anytime in
next year to two. Given the WG desire to finish this work, I think this is
the best path forward with my chair hat on too :-)



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Date: Sun, 23 Jan 2005 06:49:21 -0800
Subject: Re: [Iptel] Is trunk group a new namespace?
From: Cullen Jennings <fluffy@cisco.com>
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On 1/21/05 6:27 AM, "Shan Lu" <shanlu@sentito.com> wrote:

> 
> But you have a point that the I-D went out of its way to address the
> problem. The real issue, in my opinion, lies in RFC3966. Had 3966 allowed
> tel uri to have empty phone digits (but meaningful parameters), we won't
> have this discussion at all. As a side note, sip uri allows completely bogus
> userinfo/hostport but valid uri parameters.

Not as chair...

The issue with a tel URI with no telephone number is that the semantics of
what you do with are unclear. Imagine a HTTP URI with no host. What does it
mean? My personally view that is unless we explain to the IESG what it means
they are likely to tell we are doing the wrong thing and go invent a new URI
type - this is a long road.

It's true that for some problems, all you need is trunk group information. I
think this draft on trunk group attributes on a tel is not attempting to
solve that class of problems. Something else can be done to solve them.
However, having trunk group information attributes with a tel is useful for
some other problems and that is what this is well defined for. 

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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [Iptel] Is trunk group a new namespace?
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Jonathan,

While I like the practical side of the proposal, two observations came to
mind:

1. It is not at all clear that contact header is limited to gateway "port"
resource. RFC3398 does not say so. The rest of the TG draft actually uses
contact to mean calling party tel-uri. I would venture to guess that most
existing ISUP gateways use contact header to indicate IAM calling party
attribute in some shape or form.

2. We now have created one precedence where a tel uri is only a tel uri in
syntax that is not intended to represent any real telephone resource. I
believe it is important that the TG draft say something on how such tel uri
can be formed in contact header. Further it must point out the possibility
that tel uri contained in contact header may not be a real tel uri.

BTW, I think it is even less likely to have calling exchange's LRN in IAM
(if I interpret your example correctly).

Regards,

Shan Lu       

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org] 
>On Behalf Of Jonathan Rosenberg
>Sent: Sunday, January 23, 2005 12:11 AM
>To: shanlu@sentito.com
>Cc: iptel@ietf.org; 'Vijay K. Gurbani'
>Subject: Re: [Iptel] Is trunk group a new namespace?
>
>
>Actually, I don't see why its needed to specify any kind of 
>keyword at all.
>
>Let us remember that we are talking about the contact header field. As 
>Cullen has pointed out, it doesn't identify the originator but 
>rather a 
>port on a gateway. As such, just because there was no calling number 
>doesn't mean one could not use a tel URI to identify the gateway line. 
>Seems like something of the form:
>
>tel:<lrn>;phone-context="<provider-domain>"
>
>would suffice, in which case one could add a trunk group to this and 
>then turn it into a sip uri:
>
>sip:732766;phone-context=provider.com;tgrp=TG-1@provider.com;user=phone
>
>
>I don't think that we need to say anything about how the tel URI is 
>constructed, merely that any URI that is a reasonable 
>descriptor for the 
>gateway "port" resource itself will suffice.
>
>-Jonathan R.
>
>
>
>shanlu@sentito.com wrote:
>
>>>From: Vijay K. Gurbani <vkg@lucent.com>
>>>To: Shan Lu <shanlu@sentito.com>
>>>CC: iptel@ietf.org, Cullen Jennings <fluffy@cisco.com>
>>>Date: Jan 21, 2005 - 3:32pm
>> 
>> 
>>>Shan Lu wrote:
>>>
>>>>Vijay,
>>>>
>>>>Is it the proposal to use the following:
>>>>
>>>>Contact:<sip:1;tgrp=2@3.4.5.6>, where DS0 number is 1 and TG is 2?
>> 
>> 
>>>Yes, something to that effect. I will flesh it out further
>>>in the rev'd I-D.
>> 
>> 
>> Wait. Doesn't this take a full circle back to our earlier 
>discussion? I
>> thought we agreed upon the need for a keyword, such as "anonymous" or
>> "teluri", no?
>> 
>> Without such a keyword, how do you inform recipient of the 
>above sip uri
>> that "tgrp" stands for "Trunk Group" and not "The Gourmet 
>Roasted Pizza"?
>> Unless you think the DS0 number is sufficient.
>> 
>> Regards,
>> 
>> Shan Lu
>> 
>> _______________________________________________
>> Iptel mailing list
>> Iptel@ietf.org
>> https://www1.ietf.org/mailman/listinfo/iptel
>> 
>
>-- 
>Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
>Director, Service Provider VoIP Architecture   Parsippany, NJ 
>07054-2711
>Cisco Systems
>jdrosen@cisco.com                              FAX:   (973) 952-5050
>http://www.jdrosen.net                         PHONE: (973) 952-5000
>http://www.cisco.com
>
>_______________________________________________
>Iptel mailing list
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>https://www1.ietf.org/mailman/listinfo/iptel
>


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Cullen,

Agreed - I wasn't suggesting the long road. Just wanted to have a tel =
uri
similar to sip:invalid@invalid.net. For instance,
"tel:-;phone-context=3Dinternal" could be useful to indicate attributes =
of the
resource w/o having to reveal the number. But wasn't suggesting to =
modify
3966.

Regards,

Shan Lu

>-----Original Message-----
>From: iptel-bounces@ietf.org [mailto:iptel-bounces@ietf.org]=20
>On Behalf Of Cullen Jennings
>Sent: Sunday, January 23, 2005 9:49 AM
>To: Shan Lu; Jonathan Rosenberg
>Cc: iptel@ietf.org; Vijay K. Gurbani
>Subject: Re: [Iptel] Is trunk group a new namespace?
>
>
>On 1/21/05 6:27 AM, "Shan Lu" <shanlu@sentito.com> wrote:
>
>>=20
>> But you have a point that the I-D went out of its way to address the
>> problem. The real issue, in my opinion, lies in RFC3966. Had=20
>3966 allowed
>> tel uri to have empty phone digits (but meaningful=20
>parameters), we won't
>> have this discussion at all. As a side note, sip uri allows=20
>completely bogus
>> userinfo/hostport but valid uri parameters.
>
>Not as chair...
>
>The issue with a tel URI with no telephone number is that the=20
>semantics of
>what you do with are unclear. Imagine a HTTP URI with no host.=20
>What does it
>mean? My personally view that is unless we explain to the IESG=20
>what it means
>they are likely to tell we are doing the wrong thing and go=20
>invent a new URI
>type - this is a long road.
>
>It's true that for some problems, all you need is trunk group=20
>information. I
>think this draft on trunk group attributes on a tel is not=20
>attempting to
>solve that class of problems. Something else can be done to solve them.
>However, having trunk group information attributes with a tel=20
>is useful for
>some other problems and that is what this is well defined for.=20
>
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