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Date: Thu, 01 Apr 2004 10:32:56 +0300
From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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To: SIMPLE mailing list <simple@ietf.org>,
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CC: Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
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Subject: [Sipping] MESSAGE Exploders draft
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Hi:

I have just submitted a draft that discusses a exploder of MESSAGEs.

While it appears in the repository, you can download them from:

http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.txt
http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.html

Since this is an area covered by SIMPLE (Instant Messaging) and SIPPING 
(exploders), I am copying both mailing lists.

Regards,

      Miguel
-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland


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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: [Sipping] Redirection by a UAS
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This message is in MIME format. Since your mail reader does not understand
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SIP allows a UAS to reject an INVITE request with a 3xx response. A
particular application of this is to achieve call forwarding no reply or
deflection. With forwarding no reply the 3xx response is issued in
accordance with a script in the UA after alerting the user for a certain
time. With deflection the 3xx response is issued as a direct result of user
action in response to alerting.

A problem arises if the request cannot successfully be redirected in
accordance with the contact URI in the 3xx response. Reasons could include,
but are not limited to, the following: URI not found, not authorised, busy.
The question is, how to notify the redirecting UAS that the request has
failed. In the case of forwarding no reply, it might be desirable to
continue alerting the user at that UAS until it can be determined that
forwarding is successful (e.g., when another UAS begins alerting). In the
case of deflection it might be useful to inform the user that deflection was
unsuccessful and give the user another opportunity to answer the call.

The only way of achieving this at present is for the proxy that retargets
the request as a result of the 3xx response to repeat the request to the
original contact on detecting failure of the retargeted request. This is not
entirely satisfactory because it does not explicitly indicate the reason for
repeating the request and it may have undesirable impact on call detail
recording / missed call logging at the UAS.

Does anyone else see this as a problem and is anyone aware of alternative
solutions?

John Elwell (john.elwell@siemens.com)


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<TITLE>Redirection by a UAS</TITLE>
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<P><FONT SIZE=2 FACE="Arial">SIP allows a UAS to reject an INVITE request with a 3xx response. A particular application of this is to achieve call forwarding no reply or deflection. With forwarding no reply the 3xx response is issued in accordance with a script in the UA after alerting the user for a certain time. With deflection the 3xx response is issued as a direct result of user action in response to alerting.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">A problem arises if the request cannot successfully be redirected in accordance with the contact URI in the 3xx response. Reasons could include, but are not limited to, the following: URI not found, not authorised, busy. The question is, how to notify the redirecting UAS that the request has failed. In the case of forwarding no reply, it might be desirable to continue alerting the user at that UAS until it can be determined that forwarding is successful (e.g., when another UAS begins alerting). In the case of deflection it might be useful to inform the user that deflection was unsuccessful and give the user another opportunity to answer the call.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">The only way of achieving this at present is for the proxy that retargets the request as a result of the 3xx response to repeat the request to the original contact on detecting failure of the retargeted request. This is not entirely satisfactory because it does not explicitly indicate the reason for repeating the request and it may have undesirable impact on call detail recording / missed call logging at the UAS.</FONT></P>

<P><FONT SIZE=2 FACE="Arial">Does anyone else see this as a problem and is anyone aware of alternative solutions?</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">John Elwell (john.elwell@siemens.com)</FONT>
</P>

</BODY>
</HTML>

--Boundary_(ID_ed4bpVGpKt60kEnc8/yhzw)--

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Apr  2 10:23:22 2004
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sipping@ietf.org>
Subject: RE: [Sipping] Redirection by a UAS
Date: Thu, 1 Apr 2004 19:36:52 -0500
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> SIP allows a UAS to reject an INVITE request 
> with a 3xx response. A particular application 
> of this is to achieve call forwarding no reply 
> or deflection. With forwarding no reply the 
> 3xx response is issued in accordance with a 
> script in the UA after alerting the user for 
> a certain time. With deflection the 3xx 
> response is issued as a direct result of user 
> action in response to alerting.
>
> A problem arises if the request cannot 
> successfully be redirected in accordance with 
> the contact URI in the 3xx response. Reasons 
> could include, but are not limited to, the 
> following: URI not found, not authorized, busy. 
> The question is, how to notify the redirecting 
> UAS that the request has failed. In the case of 
> forwarding no reply, it might be desirable to 
> continue alerting the user at that UAS until 
> it can be determined that forwarding is 
> successful (e.g., when another UAS begins 
> alerting). In the case of deflection it might 
> be useful to inform the user that deflection 
> was unsuccessful and give the user another 
> opportunity to answer the call.
>
> The only way of achieving this at present is 
> for the proxy that retargets the request as 
> a result of the 3xx response to repeat the 
> request to the original contact on detecting 
> failure of the retargeted request. This is not 
> entirely satisfactory because it does not 
> explicitly indicate the reason for repeating 
> the request and it may have undesirable impact 
> on call detail recording / missed call logging 
> at the UAS.
>
> Does anyone else see this as a problem and is 
> anyone aware of alternative solutions? 

solution 1: 
  Instead of sending a 3xx, it could act as a
  transaction-stateful-proxy and proxy the
  INVITE to the forwarded-to party.

solution 2:
  Go through the standards process to define
  a mechanism that allows service policies and 
  relationships to be clearly communicated
  within a 3xx that contains multiple Contact 
  entries.  Using the Contact's q values helps
  with fault tolerance; however more information
  is needed to clearly communicate other common 
  services like simultaneous-ringing,
  call-forward-busy, and call-forward-no-answer.


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Date: Fri, 02 Apr 2004 14:26:21 -0500
To: <brett@broadsoft.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: RE: [Sipping] Redirection by a UAS
Cc: <sipping@ietf.org>
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At 07:36 PM 4/1/2004 -0500, Brett Tate wrote:
> > SIP allows a UAS to reject an INVITE request
> > with a 3xx response. A particular application
> > of this is to achieve call forwarding no reply
> > or deflection. With forwarding no reply the
> > 3xx response is issued in accordance with a
> > script in the UA after alerting the user for
> > a certain time. With deflection the 3xx
> > response is issued as a direct result of user
> > action in response to alerting.
> >
> > A problem arises if the request cannot
> > successfully be redirected in accordance with
> > the contact URI in the 3xx response. Reasons
> > could include, but are not limited to, the
> > following: URI not found, not authorized, busy.
> > The question is, how to notify the redirecting
> > UAS that the request has failed. In the case of
> > forwarding no reply, it might be desirable to
> > continue alerting the user at that UAS until
> > it can be determined that forwarding is
> > successful (e.g., when another UAS begins
> > alerting). In the case of deflection it might
> > be useful to inform the user that deflection
> > was unsuccessful and give the user another
> > opportunity to answer the call.
> >
> > The only way of achieving this at present is
> > for the proxy that retargets the request as
> > a result of the 3xx response to repeat the
> > request to the original contact on detecting
> > failure of the retargeted request. This is not
> > entirely satisfactory because it does not
> > explicitly indicate the reason for repeating
> > the request and it may have undesirable impact
> > on call detail recording / missed call logging
> > at the UAS.
> >
> > Does anyone else see this as a problem and is
> > anyone aware of alternative solutions?
>
>solution 1:
>   Instead of sending a 3xx, it could act as a
>   transaction-stateful-proxy and proxy the
>   INVITE to the forwarded-to party.
>
>solution 2:
>   Go through the standards process to define
>   a mechanism that allows service policies and
>   relationships to be clearly communicated
>   within a 3xx that contains multiple Contact
>   entries.  Using the Contact's q values helps
>   with fault tolerance; however more information
>   is needed to clearly communicate other common
>   services like simultaneous-ringing,
>   call-forward-busy, and call-forward-no-answer.

Would it be possible to include a last resort Contact that refers back to 
itself, but in such a way that the local part has a convention that 
indicates that the forwarding failed?

Mike



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


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



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From: "Brett Tate" <brett@broadsoft.com>
To: "'Michael Hammer'" <mhammer@cisco.com>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] Redirection by a UAS
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> > > >solution 2:
> > > >   Go through the standards process to define
> > > >   a mechanism that allows service policies and
> > > >   relationships to be clearly communicated
> > > >   within a 3xx that contains multiple Contact
> > > >   entries.  Using the Contact's q values helps
> > > >   with fault tolerance; however more information
> > > >   is needed to clearly communicate other common
> > > >   services like simultaneous-ringing,
> > > >   call-forward-busy, and call-forward-no-answer.
> > >
> > > Would it be possible to include a last resort
> > > Contact that refers back to itself, but in such
> > > a way that the local part has a convention that
> > > indicates that the forwarding failed?
> >
> >Yes, if both parties agree upon a solution.
> 
> Why both?  The contact is provided by the 
> called party.  The calling party simply uses 
> the contact.  Only the called party needs to know 
> what the convention means.  Did I miss something?

Trying the contact entry with the highest q value
is relatively obvious; however if I recall correctly,
rfc3261 does not even make choosing the highest 
q value a MUST.  The relationship between this 
contact entry and other contact entries are
currently not obvious or well standardized.

Some vendors simultaneous-ring if the q values 
are the same; other vendors do not.

Some vendors advance contact entries upon
any 4xx-6xx response; others do not.  Some 
vendors only advance if no response is received.

There is currently no defined parameter within
a contact entry which can communicate a no-answer
timer indicating the desire to advance to the 
next contact.  However I assume that some vendors
have toyed with trying to use the expires parameter
for this purpose.


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



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From: "Brett Tate" <brett@broadsoft.com>
To: "'Michael Hammer'" <mhammer@cisco.com>
Cc: <sipping@ietf.org>
Subject: RE: [Sipping] Redirection by a UAS
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> > > Does anyone else see this as a problem and is
> > > anyone aware of alternative solutions?
> >
> >solution 1:
> >   Instead of sending a 3xx, it could act as a
> >   transaction-stateful-proxy and proxy the
> >   INVITE to the forwarded-to party.
> >
> >solution 2:
> >   Go through the standards process to define
> >   a mechanism that allows service policies and
> >   relationships to be clearly communicated
> >   within a 3xx that contains multiple Contact
> >   entries.  Using the Contact's q values helps
> >   with fault tolerance; however more information
> >   is needed to clearly communicate other common
> >   services like simultaneous-ringing,
> >   call-forward-busy, and call-forward-no-answer.
> 
> Would it be possible to include a last resort 
> Contact that refers back to itself, but in such 
> a way that the local part has a convention that 
> indicates that the forwarding failed?

Yes, if both parties agree upon a solution.

Many vendors have either chosen to implement
a b2bua or proxy to resolve the issue.  Others
have implemented 3xx proprietary solutions.  

I am not aware on anyone having tried to
standardize a 3xx solution.  The main reason
is because it typically isn't the best solution
for complex services.  However it definitely 
would be beneficial for basic services.


_______________________________________________
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This list is for NEW development of the application of SIP
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At 02:55 PM 4/2/2004 -0500, Brett Tate wrote:
> > >solution 2:
> > >   Go through the standards process to define
> > >   a mechanism that allows service policies and
> > >   relationships to be clearly communicated
> > >   within a 3xx that contains multiple Contact
> > >   entries.  Using the Contact's q values helps
> > >   with fault tolerance; however more information
> > >   is needed to clearly communicate other common
> > >   services like simultaneous-ringing,
> > >   call-forward-busy, and call-forward-no-answer.
> >
> > Would it be possible to include a last resort
> > Contact that refers back to itself, but in such
> > a way that the local part has a convention that
> > indicates that the forwarding failed?
>
>Yes, if both parties agree upon a solution.

Why both?  The contact is provided by the called party.  The calling party 
simply uses the contact.  Only the called party needs to know what the 
convention means.  Did I miss something?

Mike


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(Comments to self inline)

Ben Campbell wrote:
[...]
> Saying the exploder MUST NOT include the URI list in the outbound 
> messages seems a little severe. I can imagine applications where I want 
> the recipients to know the list. This could be useful for "reply to all" 
> sort of functions. But if we do allow this, we would need to give the 
> sender a way to choose, possibly on a per-URI basis. (Sort of a "cc" vs. 
> "bcc" thing.)

This sort of functionality seems to be required by REQ-GROUP-2 and 
REQ-GROUP-4 from the advanced IM reqs draft.

[...]

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I like this approach overall.

A couple of comments and questions. Apologies if these have already been 
discussed.

Is it legal for the list parameter to point to something other than a 
body part? That is, can I have external lists?

Saying the exploder MUST NOT include the URI list in the outbound 
messages seems a little severe. I can imagine applications where I want 
the recipients to know the list. This could be useful for "reply to all" 
sort of functions. But if we do allow this, we would need to give the 
sender a way to choose, possibly on a per-URI basis. (Sort of a "cc" vs. 
"bcc" thing.)

Instead of saying that message exploders must always send message 
requests, we might consider saying the outbound request method must 
match the outbound method. This would allow for exploders that know how 
to explode more than one method.


When condidering DoS issues, it might be worth suggesting behavior if 
the exploder finds duplicate URIs on the list. For example, I could put 
the same URI in a bunch of times, and have a nice DoS amplifier.


Miguel Garcia wrote:
> Hi:
> 
> I have just submitted a draft that discusses a exploder of MESSAGEs.
> 
> While it appears in the repository, you can download them from:
> 
> http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.txt 
> 
> http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.html 
> 
> 
> Since this is an area covered by SIMPLE (Instant Messaging) and SIPPING 
> (exploders), I am copying both mailing lists.
> 
> Regards,
> 
>      Miguel


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To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
CC: SIMPLE mailing list <simple@ietf.org>,
        SIPPING mailing list <sipping@ietf.org>,
        Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
Subject: Re: [Sipping] MESSAGE Exploders draft
References: <406BC5A8.4080003@nokia.com>
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Miguel,

In general the approach in the draft seems reasonable to me.

I see one area of ambiguity, having to do with how multipart bodies are 
handled. The draft says to remove the list and include everything else. 
But even your example doesn't do that - it also removes the 
multipart/mixed wrapper. Of course in this particular example that makes 
perfect sense, but it goes beyond what the draft says to do.

There are more complex cases:

- for security there could be signed body parts.

   - The signature could be on the multipart containing the message
     and list. It would have to be removed.

   - The signature could be on just the message. Presumably
     it should remain.

- There could be other body parts referenced by cid: urls in
   other headers in the message.

I think there needs to be a more prescriptive way of defining what gets 
passed on in the exploded messages. This is really just a manifestation 
of a more general problem of how to decide how different body parts 
should be processed. It potentially exists in a normal, unexploded, MESSAGE.

I think at least part of the answer lies with Content-Disposition. I 
think there should be clarity about the content-disposition should be 
for the body part(s) that MESSAGE interprets as message content. 
Probably "render" (which is default for sip) would be an appropriate 
content-disposition for message content.

Also, I think there should be some clarity about what the 
content-disposition should be for various kinds of multipart containers 
when used in the various ways they may be used in sip. I'm not quite 
sure what the answer is here.

The list itself is of course referenced from a cid: url. It is the url 
and its placement that determines how the corresponding body part should 
be processed. For these, the content-disposition should be some value 
that doesn't imply some other sort of processing. I think it can't be 
any of "session", "render", "alert", or "icon". Maybe it could be 
"attachment".

	Paul



Miguel Garcia wrote:
> Hi:
> 
> I have just submitted a draft that discusses a exploder of MESSAGEs.
> 
> While it appears in the repository, you can download them from:
> 
> http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.txt 
> 
> http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.html 
> 
> 
> Since this is an area covered by SIMPLE (Instant Messaging) and SIPPING 
> (exploders), I am copying both mailing lists.
> 
> Regards,
> 
>      Miguel


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



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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] Redirection by a UAS
To: "'Michael Hammer'" <mhammer@cisco.com>, brett@broadsoft.com
Cc: sipping@ietf.org
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Michael,

You wrote:
> Would it be possible to include a last resort Contact that 
> refers back to 
> itself, but in such a way that the local part has a convention that 
> indicates that the forwarding failed?
> 
This sounds interesting. Would the fact that the Call-ID has not changed be
a sufficient indication that it is a repeat attempt, presumably because
forwarding failed?

John (john.elwell@siemens.com)

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



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Date: Mon, 05 Apr 2004 13:20:02 -0400
To: "Elwell, John" <john.elwell@siemens.com>
From: Michael Hammer <mhammer@cisco.com>
Subject: RE: [Sipping] Redirection by a UAS
Cc: brett@broadsoft.com, sipping@ietf.org
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At 05:28 PM 4/5/2004 +0100, Elwell, John wrote:
>Michael,
>
>You wrote:
> > Would it be possible to include a last resort Contact that
> > refers back to
> > itself, but in such a way that the local part has a convention that
> > indicates that the forwarding failed?
> >
>This sounds interesting. Would the fact that the Call-ID has not changed be
>a sufficient indication that it is a repeat attempt, presumably because
>forwarding failed?
>
>John (john.elwell@siemens.com)

Because of cases where re-transmissions are used (which could be 
temporarily mis-routed), I would not want to rely on an implicit 
indication.  An explicit index into a stored call state (which I assume 
exists) would seem better to me.

Mike


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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sipping] Redirection by a UAS
To: "'Michael Hammer'" <mhammer@cisco.com>
Cc: brett@broadsoft.com, sipping@ietf.org
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> -----Original Message-----
> From: Michael Hammer [mailto:mhammer@cisco.com] 
> Sent: 05 April 2004 18:20
> To: Elwell, John
> Cc: brett@broadsoft.com; sipping@ietf.org
> Subject: RE: [Sipping] Redirection by a UAS
> 
> 
> At 05:28 PM 4/5/2004 +0100, Elwell, John wrote:
> >Michael,
> >
> >You wrote:
> > > Would it be possible to include a last resort Contact that
> > > refers back to
> > > itself, but in such a way that the local part has a 
> convention that
> > > indicates that the forwarding failed?
> > >
> >This sounds interesting. Would the fact that the Call-ID has 
> not changed be
> >a sufficient indication that it is a repeat attempt, 
> presumably because
> >forwarding failed?
> >
> >John (john.elwell@siemens.com)
> 
> Because of cases where re-transmissions are used (which could be 
> temporarily mis-routed), I would not want to rely on an implicit 
> indication.  An explicit index into a stored call state 
> (which I assume 
> exists) would seem better to me.
> 
> Mike
>

In that case, perhaps History-Info would suffice. The UAS could see from
History-Info that the request has been redirected from itself elsewhere and
then back again.

John 

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Cc: Dean Willis <dean.willis@softarmor.com>,
        Hisham Khartabil <hisham.khartabil@nokia.com>,
        Allison Mankin <mankin@psg.com>, Robert Sparks <rjsparks@nostrum.com>,
        John Peterson <jon.peterson@neustar.biz>,
        Ted Hardie <hardie@qualcomm.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        Rohan Mahy <rohan@cisco.com>, Alan Johnston <alan.johnston@wcom.com>,
        Adam Roach <adam@dynamicsoft.com>
From: Rohan Mahy <rohan@cisco.com>
Date: Tue, 6 Apr 2004 10:53:31 -0700
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Subject: [Sipping] Joint Interim Meeting Mon May 24 - Wed May 26
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Hello All,

Please mark your calendars.

I'd like to announce a joint interim of the SIP, SIPPING, SIMPLE, and  
XCON WGs.
Nokia has generously agreed to host the event either at their offices  
near Boston or in a neighboring hotel during the week of May 24.

Please reserve the following dates for the WGs that you are interested  
in:
Monday May 24 SIP and SIPPING  tentatively 9 am start
Tuesday May 25 SIMPLE   All day
Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch  
flights.
(the actual agenda is subject to change, including the dates of various  
WG meetings).

There will be no fee, however please let Hisham and myself know if you  
are planning to attend by clicking on the Yes or Maybe links below:

mailto:rohan@cisco.com? 
Cc=hisham.khartabil@nokia.com&Subject=[interim]yes
mailto:rohan@cisco.com? 
Cc=hisham.khartabil@nokia.com&Subject=[interim]maybe

Wireless will be provided. As usual, I will be making T-shirts.  If you  
are interested in buying one, please send your shirt size to me.

More details on the agenda and hotel accommodations will be forthcoming  
shortly.

thanks,
-rohan


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Subject: [Sipping] Last Call: 'The Early Session Disposition Type for the Session
 Initiation Protocol (SIP)' to Proposed Standard
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The IESG has received a request from the Session Initiation Proposal 
Investigation WG to consider the following documents:

- 'The Early Session Disposition Type for the Session Initiation Protocol 
   (SIP) '
   <draft-ietf-sipping-early-disposition-01.txt> as a Proposed Standard
- 'Early Media and Ringing Tone Generation in the Session Initiation 
   Protocol '
   <draft-ietf-sipping-early-media-01.txt> as an Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2004-04-19.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sipping-early-disposition-
01.txt
http://www.ietf.org/internet-drafts/draft-ietf-sipping-early-media-01.txt


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Thread-Topic: Initiating TLS from behind a firewall&nat
Thread-Index: AcQclw9wAjzYEbNLTqe5OdDB3/n8MQ==
From: =?iso-8859-2?Q?Petr=F3czi_Attila?= <Attila.Petroczi@t-systems.co.hu>
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Subject: [Sipping] Initiating TLS from behind a firewall&nat
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Hi All,

does anybody know about some standard for initiating a TLS connection =
from behind a firewall with NAT?

In details: Our client is sitting behind a firewall with NAT and wants =
to connect to a SIP UA somewhere on the Internet. We have an http/https =
proxy installed on the firewall&nat device.=20

Our client first has to build up a TCP connection through the proxy to =
the other SIP device before starting the TLS handshake to secure the =
channel. For example in case of https the browser sends a plain connect =
message to the 443 port of the distant https server before starting the =
TLS handshake on a new socket layered over the TCP socket. What about =
SIP?

I think we may send an INFO message to build up the TCP connection, and =
after the TLS handshake can we send our first INVITE message through the =
secured TLS channel. Am I right?=20

I have found =
http://www.jdrosen.net/papers/draft-rosenberg-sip-entfw-01.txt and=20
http://ncl.cgu.edu/publications/paulsamy_chatterjee.PDF, but they do not =
deal with such details.=20

Best regards,

Attila
=20
Attila Petr=F3czi
T-Systems RIC Kft.
H-1097 Budapest, K=F6nyves K=E1lm=E1n krt. 12-14.
Telefon: +36 1 456 5361
Fax: +36 1 456 5499
mailto:HYPERLINK =
"mailto:attila.petroczi@t-systems.co.hu"attila.petroczi@t-systems.co.hu =
<mailto:HYPERLINK >=20
www.t-systems.co.hu =
<file:///C:/Documents%20and%20Settings/apetroczi/Application%20Data/Micro=
soft/Signatures/www.t-systems.co.hu>=20



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Date: Wed, 07 Apr 2004 17:30:27 -0700
From: Cullen Jennings <fluffy@cisco.com>
To: "'sipping@ietf.org'" <sipping@ietf.org>
CC: Orit Levin <oritl@microsoft.com>,
        Henning Schulzrinne <hgs@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: [Sipping] User with multiple sessions in Conference Package
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In a conference, here are many cases where a single user may have set up
multiple SIP dialogs with the focus. For example, using one UA instance for
audio and another UA instance for IM. Or connected on both cell phone and on
desk phone in office. Right now the conference package seems a little
underspecified in how to deal with this and we need to clear this up.

I'm tossing out two proposal below. Comments? Ideas? Can we choose one?
Could be  close something like this on this list?

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

Proposal 1: 

The User Element gets a new attribute called contact. This attribute
contains the contact that the focus used for this dialog. Each dialog
results in a separate User Element instance and the uri is used to correlate
the fact that they are the same user. For anonymous calls the system needs
to generate separate anonymous user uri's so they can be separated from one
another. 

For Example,

<conference-info>
    <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
        <media-stream> <ssrc> 1234 </ssrc> </media-stream>
    </user>    
    <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
        <media-stream> <ssrc> 5678 </ssrc> </media-stream>
    </user>
</conference-info>

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

Proposal 2: 

The data in the User Element gets split between a User Element and a new
Session Element that is placed inside the User Element. The media stream
elements are put inside the Session.

<conference-info>
    <user uri="fluffy@cisco.com">
        <session contact="1111@focus.cisco.com">
            <media-stream> <ssrc> 1234 </ssrc> </media-stream>
        </session> 
        <session contact="2222@focus.cisco.com">
            <media-stream> <ssrc> 5678 </ssrc> </media-stream>
        </session>
    </user>
</conference-info>


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



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From: hisham.khartabil@nokia.com
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Subject: RE: [Sipping] User with multiple sessions in Conference Package
Date: Thu, 8 Apr 2004 15:11:22 +0300
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To: <fluffy@cisco.com>, <sipping@ietf.org>
Cc: <oritl@microsoft.com>, <hgs@cs.columbia.edu>, <jdrosen@dynamicsoft.com>
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Why do you need the focus to know that it is the same user (human)? I =
don't understand the use case. Can you elaborate?

Thanks,
Hisham

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Cullen Jennings
> Sent: 08.April.2004 03:30
> To: 'sipping@ietf.org'
> Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
> Subject: [Sipping] User with multiple sessions in Conference Package
>=20
>=20
>=20
> In a conference, here are many cases where a single user may=20
> have set up
> multiple SIP dialogs with the focus. For example, using one=20
> UA instance for
> audio and another UA instance for IM. Or connected on both=20
> cell phone and on
> desk phone in office. Right now the conference package seems a little
> underspecified in how to deal with this and we need to clear this up.
>=20
> I'm tossing out two proposal below. Comments? Ideas? Can we=20
> choose one?
> Could be  close something like this on this list?
>=20
> --------------------------------------------------------------
>=20
> Proposal 1:=20
>=20
> The User Element gets a new attribute called contact. This attribute
> contains the contact that the focus used for this dialog. Each dialog
> results in a separate User Element instance and the uri is=20
> used to correlate
> the fact that they are the same user. For anonymous calls the=20
> system needs
> to generate separate anonymous user uri's so they can be=20
> separated from one
> another.=20
>=20
> For Example,
>=20
> <conference-info>
>     <user uri=3D"fluffy@cisco.com" contact=3D"1111@focus.cisco.com">
>         <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>     </user>   =20
>     <user uri=3D"fluffy@cisco.com" contact=3D"2222@focus.cisco.com">
>         <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>     </user>
> </conference-info>
>=20
> ---------------------------------------------------------------
>=20
> Proposal 2:=20
>=20
> The data in the User Element gets split between a User=20
> Element and a new
> Session Element that is placed inside the User Element. The=20
> media stream
> elements are put inside the Session.
>=20
> <conference-info>
>     <user uri=3D"fluffy@cisco.com">
>         <session contact=3D"1111@focus.cisco.com">
>             <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>         </session>=20
>         <session contact=3D"2222@focus.cisco.com">
>             <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>         </session>
>     </user>
> </conference-info>
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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I just read thru this again. I found one loose end. In section 4:

    Sending early media using a different offer/answer exchange than the
    one used for sending regular media helps avoid media clipping in case
    of forking. The UAC can reject or mute new offers for early media
    without muting the sessions that will carry media when the original
    INVITE is accepted. The UAC can give priority to media received over
    the latter sessions. This way, the application server model
    transitions from early to regular media at the right moment.

There is one case where the UAC still may have difficulty knowing which 
media to render - the case when multiple 200 responses make it back to 
the UAC, because multiple UAS's answered at almost the same time. This 
is similar to the case of receiving multiple early media streams in the 
gateway model - you can't necessarily decide which media stream matches 
which sip message, so if the intent is to hang up on one call and retain 
the other, it may take time to figure out which was the right media 
stream to play.

I don't know that this changes the arguments in a material way. An 
implementation that wants to cover all the bases will in this case want 
to take some of the same remediation steps as in the gateway case, which 
is a lot of pain. Most likely the typical approach will be to ignore the 
problem on the assumption that it will rarely happen, and that clipping 
is acceptable in that case.

I don't have strong feelings if this should be mentioned in the draft or 
not.

	Paul


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From: Paul Kyzivat <pkyzivat@cisco.com>
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Section 4 says:

    As for any other SIP request, the size of requests carrying URI lists
    MUST NOT exceed 1300 bytes, unless the user agent client has positive
    knowledge that the message will not traverse a congestion-unsafe link
    at any hop, or that the message size is at least 200 bytes less than
    the lowest MTU value found en route to the server.

I was curious about the implications of this, so I did some counting. I 
took the example in section 9 and stripped out all the leading blanks to 
get a plausible size estimate. The number I got was 1242, with four urls 
in the list. Each additional URL (of the size in the example) adds 37 
bytes. So a list with more than 5 urls exceeds the 1300 byte limit.

That says to me that this technique is pretty much useless except in 
carefully controlled networks, or when compression is used. Seems pretty 
limiting. So it seems likely that in practice content indirection will 
have to be used.

	Paul


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From: Cullen Jennings <fluffy@cisco.com>
To: Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, <woof@snowshore.com>,
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Their is a bit of weird inconsistency in the having the parameters that are
passed to the __sip-conf service invocation embedded in the user portion.

In the annc and dialog service, there are no parameters in the user portion
of the URI. Yet in the conf service, the conf id is embedded in the user
portion. 

I think this is wrong. It should be

__sip-conf@mediaserver.example.net;conf=uniqueIdentifier

This makes it feasible to specify routing rules for proxies that correctly
get all the traffic for the service to the media server. Particularly if the
media server is behind a NAT.

Could you change the draft to make the conf service invocation consistent
with the annc and dialog?

Cullen










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From exim@www1.ietf.org  Mon Apr 12 13:34:34 2004
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From: Cullen Jennings <fluffy@cisco.com>
To: Andy Spitzer <woof@snowshore.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, <jvandyke@snowshore.com>
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Thanks for the fast response. I'm lost on how you imagine this working - I
think you will need to walk me through it.  Let's say the domain is
cisco.com and I have two media servers ms1.cisco.com and ms2.cisco.com. And
my conference ID's are 6 digit numbers.

Are you suggesting that ms1.cisco.com and ms2.cisco.com know that
__sip-conf=1 belongs to ms1 and __sip-conf=2 belongs to ms2 and they both
send a REGISTER message to the proxy and ms2 registers
__sip-conf=2@cisco.com with a contact pointing to ms2.cisco.com? I don't see
how they could register all the possible conference Ids in any useful way.
Mostly because the message needs to routed to the ms before the ms will
create the conference and conference needs to be created before it's id can
be registered and the registration needs to happen before the routing and
thus we have a circular loop of dependencies in time.
 
I'm proposing that the proxy route to the correct media server by having the
host portion of the URI be ms1.cisco.com or ms2.cisco.com


On 4/12/04 9:28 AM, "Andy Spitzer" <woof@snowshore.com> wrote:

> Cullen,
> 
> The reason for having the conf id as part of the user part, was because
> unlike the other services (annc, ivr, and dialog), conferencing is not
> stand alone.  With annc, ivr, and dialog, each session is unique in
> independent.  With conferencing, however, multiple sessions make up a
> particular conference.
> 
> As such, a particular conference must live on a particular media server.
> All participants into that conference must reach the same media server.
> By placing the conf id in the user part, it makes the conference look like
> a user on a particular media server, and as such any proxy can use the
> correct routing rules for users in order to reach the correct media server
> assigned to that conference.  Just the same way they would route users to
> a particular endpoint.
> 
> If the conf id was done as you describe, it would mean proxies would have
> to look into the parameters to determine which conference was being
> requested in order to determine which media server to INVITE.  We didn't
> want to force proxies to do that, so it was decided the user part was more
> proxy friendly.
> 
> Your approach is certainly feasible, and was indeed originally
> considered.  I believe it may be possible to support both simultainously.
> 
> --Andy Spitzer
> woof@snowshore.com
> 
> On Mon, 12 Apr 2004 08:59:14 -0700, Cullen Jennings <fluffy@cisco.com>
> wrote:
> 
>> 
>> Their is a bit of weird inconsistency in the having the parameters that
>> are
>> passed to the __sip-conf service invocation embedded in the user portion.
>> 
>> In the annc and dialog service, there are no parameters in the user
>> portion
>> of the URI. Yet in the conf service, the conf id is embedded in the user
>> portion.
>> 
>> I think this is wrong. It should be
>> 
>> __sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>> 
>> This makes it feasible to specify routing rules for proxies that
>> correctly
>> get all the traffic for the service to the media server. Particularly if
>> the
>> media server is behind a NAT.
>> 
>> Could you change the draft to make the conf service invocation consistent
>> with the annc and dialog?
>> 
>> Cullen
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
> 
> 
> 
> 


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Date: Mon, 12 Apr 2004 10:43:47 -0700
Subject: Re: [Sipping] User with multiple sessions in Conference Package
From: Cullen Jennings <fluffy@cisco.com>
To: <hisham.khartabil@nokia.com>, <sipping@ietf.org>
CC: Orit Levin <oritl@microsoft.com>,
        Henning Schulzrinne <hgs@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Message-ID: <BCA02363.3956C%fluffy@cisco.com>
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I'll take a stab at explaining :-)

If someone has a roster displayed on some device - they want to see that one
person, Cullen is in the conference they don't want to sees that their are 4
participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More importantly, they want to
be able to use a user interface to boot Cullen out of the conference not
have to go and somehow magically figure out which devices to boot out since
they are being used by Cullen.

If their are 10 participants that are all Anonymous, you still want to be
able to know their are 10 participants not 1 participant called Anonymous
that seems to have dialed in from 10 devices.

Given that SIP AOR correspond to a user and is not a single device. There
are basically three levels off aggregation happening here.

There are users. Each user can have multiple SIP dialogs that got them into
the conference. Each one of these SIP dialogs can be started and terminated
independently. For example one may be dialed out while another dialed in and
one may be active while others may be ongoing. Inside of each SIP dialog,
there can be multiple media stream (audio, video, etc). The three levels are
Users, SIP Sessions or Dialogs, Media Sessions. Unfortunately, the draft as
written supports two things and the User level information and the SIP
session level information is mixed together. I think this needs to get
fixed. 

One more use case, many audio conference systems announce when a user joins
of leaves. This is really annoying when someone transfers from the cell
phone to desk phone. Let me give you a scenario that I think XCON should
support. 

Cullen dials in to conf from cell phone and is announced. Cullen then dials
in from desk phone and is *not* announced because the user Cullen is already
in the conference. Cullen hangs up cell phone and this is *not* announced
because Cullen is still in the conference. Cullen hangs up his desk phone
and the conference announces that "Fluffy has left the building" because he
is no longer in the conference.


On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com" <hisham.khartabil@nokia.com>
wrote:

> Why do you need the focus to know that it is the same user (human)? I don't
> understand the use case. Can you elaborate?
> 
> Thanks,
> Hisham
> 
>> -----Original Message-----
>> From: sipping-admin@ietf.org
>> [mailto:sipping-admin@ietf.org]On Behalf Of
>> ext Cullen Jennings
>> Sent: 08.April.2004 03:30
>> To: 'sipping@ietf.org'
>> Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
>> Subject: [Sipping] User with multiple sessions in Conference Package
>> 
>> 
>> 
>> In a conference, here are many cases where a single user may
>> have set up
>> multiple SIP dialogs with the focus. For example, using one
>> UA instance for
>> audio and another UA instance for IM. Or connected on both
>> cell phone and on
>> desk phone in office. Right now the conference package seems a little
>> underspecified in how to deal with this and we need to clear this up.
>> 
>> I'm tossing out two proposal below. Comments? Ideas? Can we
>> choose one?
>> Could be  close something like this on this list?
>> 
>> --------------------------------------------------------------
>> 
>> Proposal 1: 
>> 
>> The User Element gets a new attribute called contact. This attribute
>> contains the contact that the focus used for this dialog. Each dialog
>> results in a separate User Element instance and the uri is
>> used to correlate
>> the fact that they are the same user. For anonymous calls the
>> system needs
>> to generate separate anonymous user uri's so they can be
>> separated from one
>> another. 
>> 
>> For Example,
>> 
>> <conference-info>
>>     <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
>>         <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>     </user>    
>>     <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
>>         <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>     </user>
>> </conference-info>
>> 
>> ---------------------------------------------------------------
>> 
>> Proposal 2: 
>> 
>> The data in the User Element gets split between a User
>> Element and a new
>> Session Element that is placed inside the User Element. The
>> media stream
>> elements are put inside the Session.
>> 
>> <conference-info>
>>     <user uri="fluffy@cisco.com">
>>         <session contact="1111@focus.cisco.com">
>>             <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>         </session>
>>         <session contact="2222@focus.cisco.com">
>>             <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>         </session>
>>     </user>
>> </conference-info>
>> 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>> 
> 


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



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Date: Mon, 12 Apr 2004 14:57:55 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Conf stuff in natann draft
In-reply-to: <BCA00AE2.39526%fluffy@cisco.com>
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Cullen,

You are right - there is an inconsistency.  However, if we make netann 
consistent as you propose (and previous versions of netann use exactly this 
format), then netann becomes inconsistent with the conferencing framework 
and the rest of the conferencing work.

The conferencing framework assumes that the conference ID is embedded in 
the user portion of the URI, not in URI parameters.

Can you explain your NAT issue with this approach?

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

At 08:59 AM 4/12/2004 -0700, Cullen Jennings wrote:

>Their is a bit of weird inconsistency in the having the parameters that are
>passed to the __sip-conf service invocation embedded in the user portion.
>
>In the annc and dialog service, there are no parameters in the user portion
>of the URI. Yet in the conf service, the conf id is embedded in the user
>portion.
>
>I think this is wrong. It should be
>
>__sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>
>This makes it feasible to specify routing rules for proxies that correctly
>get all the traffic for the service to the media server. Particularly if the
>media server is behind a NAT.
>
>Could you change the draft to make the conf service invocation consistent
>with the annc and dialog?
>
>Cullen
>
>
>
>
>
>
>
>
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


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



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Date: Mon, 12 Apr 2004 15:02:38 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] User with multiple sessions in Conference Package
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Cullen,

I agree with your explanation - this is definitely a goal.

However, I don't understand your initial proposals.  The conference package 
should, as you argue, just indicate that Cullen is in the conference, and 
perhaps note the media types used.  Whether these media types are supported 
by a single device, or multiple devices doesn't matter as far as the 
conference package is concerned.  What exactly is the problem you are 
trying to solve with your two proposals?  I don't think that Contact URIs 
belong in the conference package - if a participant wants detailed 
information about the way in which a participant is connected to the 
conference, the dialog package can be used.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

At 10:43 AM 4/12/2004 -0700, Cullen Jennings wrote:

>I'll take a stab at explaining :-)
>
>If someone has a roster displayed on some device - they want to see that one
>person, Cullen is in the conference they don't want to sees that their are 4
>participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
>Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More importantly, they want to
>be able to use a user interface to boot Cullen out of the conference not
>have to go and somehow magically figure out which devices to boot out since
>they are being used by Cullen.
>
>If their are 10 participants that are all Anonymous, you still want to be
>able to know their are 10 participants not 1 participant called Anonymous
>that seems to have dialed in from 10 devices.
>
>Given that SIP AOR correspond to a user and is not a single device. There
>are basically three levels off aggregation happening here.
>
>There are users. Each user can have multiple SIP dialogs that got them into
>the conference. Each one of these SIP dialogs can be started and terminated
>independently. For example one may be dialed out while another dialed in and
>one may be active while others may be ongoing. Inside of each SIP dialog,
>there can be multiple media stream (audio, video, etc). The three levels are
>Users, SIP Sessions or Dialogs, Media Sessions. Unfortunately, the draft as
>written supports two things and the User level information and the SIP
>session level information is mixed together. I think this needs to get
>fixed.
>
>One more use case, many audio conference systems announce when a user joins
>of leaves. This is really annoying when someone transfers from the cell
>phone to desk phone. Let me give you a scenario that I think XCON should
>support.
>
>Cullen dials in to conf from cell phone and is announced. Cullen then dials
>in from desk phone and is *not* announced because the user Cullen is already
>in the conference. Cullen hangs up cell phone and this is *not* announced
>because Cullen is still in the conference. Cullen hangs up his desk phone
>and the conference announces that "Fluffy has left the building" because he
>is no longer in the conference.
>
>
>On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com" <hisham.khartabil@nokia.com>
>wrote:
>
> > Why do you need the focus to know that it is the same user (human)? I don't
> > understand the use case. Can you elaborate?
> >
> > Thanks,
> > Hisham
> >
> >> -----Original Message-----
> >> From: sipping-admin@ietf.org
> >> [mailto:sipping-admin@ietf.org]On Behalf Of
> >> ext Cullen Jennings
> >> Sent: 08.April.2004 03:30
> >> To: 'sipping@ietf.org'
> >> Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
> >> Subject: [Sipping] User with multiple sessions in Conference Package
> >>
> >>
> >>
> >> In a conference, here are many cases where a single user may
> >> have set up
> >> multiple SIP dialogs with the focus. For example, using one
> >> UA instance for
> >> audio and another UA instance for IM. Or connected on both
> >> cell phone and on
> >> desk phone in office. Right now the conference package seems a little
> >> underspecified in how to deal with this and we need to clear this up.
> >>
> >> I'm tossing out two proposal below. Comments? Ideas? Can we
> >> choose one?
> >> Could be  close something like this on this list?
> >>
> >> --------------------------------------------------------------
> >>
> >> Proposal 1:
> >>
> >> The User Element gets a new attribute called contact. This attribute
> >> contains the contact that the focus used for this dialog. Each dialog
> >> results in a separate User Element instance and the uri is
> >> used to correlate
> >> the fact that they are the same user. For anonymous calls the
> >> system needs
> >> to generate separate anonymous user uri's so they can be
> >> separated from one
> >> another.
> >>
> >> For Example,
> >>
> >> <conference-info>
> >>     <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
> >>         <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>     </user>
> >>     <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
> >>         <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>     </user>
> >> </conference-info>
> >>
> >> ---------------------------------------------------------------
> >>
> >> Proposal 2:
> >>
> >> The data in the User Element gets split between a User
> >> Element and a new
> >> Session Element that is placed inside the User Element. The
> >> media stream
> >> elements are put inside the Session.
> >>
> >> <conference-info>
> >>     <user uri="fluffy@cisco.com">
> >>         <session contact="1111@focus.cisco.com">
> >>             <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>         </session>
> >>         <session contact="2222@focus.cisco.com">
> >>             <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>         </session>
> >>     </user>
> >> </conference-info>
> >>
> >>
> >> _______________________________________________
> >> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >> This list is for NEW development of the application of SIP
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sip@ietf.org for new developments of core SIP
> >>
> >
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


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



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Date: Mon, 12 Apr 2004 19:41:31 -0700
Subject: Re: [Sipping] User with multiple sessions in Conference Package
From: Cullen Jennings <fluffy@cisco.com>
To: Alan Johnston <alan.johnston@mci.com>, <hisham.khartabil@nokia.com>,
        <sipping@ietf.org>
CC: Orit Levin <oritl@microsoft.com>,
        Henning Schulzrinne <hgs@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Message-ID: <BCA0A16B.39680%fluffy@cisco.com>
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> The conference package
> should, as you argue, just indicate that Cullen is in the conference, and
> perhaps note the media types used.

Right now the conference package seems to have a lot more than that in it.
It has if I dialed in or out, disconnect reasons, on-hold, etc. that are
specific to the session not the user.

BTW. I really like the definition of the uri element in the user. I think
that was the hard problem and it is right.

On 4/12/04 1:02 PM, "Alan Johnston" <alan.johnston@mci.com> wrote:

> Cullen,
> 
> I agree with your explanation - this is definitely a goal.
> 
> However, I don't understand your initial proposals.  The conference package
> should, as you argue, just indicate that Cullen is in the conference, and
> perhaps note the media types used.  Whether these media types are supported
> by a single device, or multiple devices doesn't matter as far as the
> conference package is concerned.  What exactly is the problem you are
> trying to solve with your two proposals?  I don't think that Contact URIs
> belong in the conference package - if a participant wants detailed
> information about the way in which a participant is connected to the
> conference, the dialog package can be used.
> 
> Thanks,
> Alan Johnston
> MCI
> sip:alan@sipstation.com
> 
> At 10:43 AM 4/12/2004 -0700, Cullen Jennings wrote:
> 
>> I'll take a stab at explaining :-)
>> 
>> If someone has a roster displayed on some device - they want to see that one
>> person, Cullen is in the conference they don't want to sees that their are 4
>> participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
>> Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More importantly, they want to
>> be able to use a user interface to boot Cullen out of the conference not
>> have to go and somehow magically figure out which devices to boot out since
>> they are being used by Cullen.
>> 
>> If their are 10 participants that are all Anonymous, you still want to be
>> able to know their are 10 participants not 1 participant called Anonymous
>> that seems to have dialed in from 10 devices.
>> 
>> Given that SIP AOR correspond to a user and is not a single device. There
>> are basically three levels off aggregation happening here.
>> 
>> There are users. Each user can have multiple SIP dialogs that got them into
>> the conference. Each one of these SIP dialogs can be started and terminated
>> independently. For example one may be dialed out while another dialed in and
>> one may be active while others may be ongoing. Inside of each SIP dialog,
>> there can be multiple media stream (audio, video, etc). The three levels are
>> Users, SIP Sessions or Dialogs, Media Sessions. Unfortunately, the draft as
>> written supports two things and the User level information and the SIP
>> session level information is mixed together. I think this needs to get
>> fixed.
>> 
>> One more use case, many audio conference systems announce when a user joins
>> of leaves. This is really annoying when someone transfers from the cell
>> phone to desk phone. Let me give you a scenario that I think XCON should
>> support.
>> 
>> Cullen dials in to conf from cell phone and is announced. Cullen then dials
>> in from desk phone and is *not* announced because the user Cullen is already
>> in the conference. Cullen hangs up cell phone and this is *not* announced
>> because Cullen is still in the conference. Cullen hangs up his desk phone
>> and the conference announces that "Fluffy has left the building" because he
>> is no longer in the conference.
>> 
>> 
>> On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com" <hisham.khartabil@nokia.com>
>> wrote:
>> 
>>> Why do you need the focus to know that it is the same user (human)? I don't
>>> understand the use case. Can you elaborate?
>>> 
>>> Thanks,
>>> Hisham
>>> 
>>>> -----Original Message-----
>>>> From: sipping-admin@ietf.org
>>>> [mailto:sipping-admin@ietf.org]On Behalf Of
>>>> ext Cullen Jennings
>>>> Sent: 08.April.2004 03:30
>>>> To: 'sipping@ietf.org'
>>>> Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
>>>> Subject: [Sipping] User with multiple sessions in Conference Package
>>>> 
>>>> 
>>>> 
>>>> In a conference, here are many cases where a single user may
>>>> have set up
>>>> multiple SIP dialogs with the focus. For example, using one
>>>> UA instance for
>>>> audio and another UA instance for IM. Or connected on both
>>>> cell phone and on
>>>> desk phone in office. Right now the conference package seems a little
>>>> underspecified in how to deal with this and we need to clear this up.
>>>> 
>>>> I'm tossing out two proposal below. Comments? Ideas? Can we
>>>> choose one?
>>>> Could be  close something like this on this list?
>>>> 
>>>> --------------------------------------------------------------
>>>> 
>>>> Proposal 1:
>>>> 
>>>> The User Element gets a new attribute called contact. This attribute
>>>> contains the contact that the focus used for this dialog. Each dialog
>>>> results in a separate User Element instance and the uri is
>>>> used to correlate
>>>> the fact that they are the same user. For anonymous calls the
>>>> system needs
>>>> to generate separate anonymous user uri's so they can be
>>>> separated from one
>>>> another.
>>>> 
>>>> For Example,
>>>> 
>>>> <conference-info>
>>>>     <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
>>>>         <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>     </user>
>>>>     <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
>>>>         <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>     </user>
>>>> </conference-info>
>>>> 
>>>> ---------------------------------------------------------------
>>>> 
>>>> Proposal 2:
>>>> 
>>>> The data in the User Element gets split between a User
>>>> Element and a new
>>>> Session Element that is placed inside the User Element. The
>>>> media stream
>>>> elements are put inside the Session.
>>>> 
>>>> <conference-info>
>>>>     <user uri="fluffy@cisco.com">
>>>>         <session contact="1111@focus.cisco.com">
>>>>             <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>         </session>
>>>>         <session contact="2222@focus.cisco.com">
>>>>             <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>         </session>
>>>>     </user>
>>>> </conference-info>
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>> 
>>> 
>> 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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



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Date: Mon, 12 Apr 2004 20:04:08 -0700
Subject: Re: [Sipping] Conf stuff in natann draft
From: Cullen Jennings <fluffy@cisco.com>
To: Alan Johnston <alan.johnston@mci.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, <woof@snowshore.com>,
        <jvandyke@snowshore.com>
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I think I understand the confusion. The name of the focus should have the
identify the conference by the user part. However, this is very different
than the service invocation that the focus uses to control the media mixer.
The term __sip-conf is a bit of a red hearing __sip-mixer would be more
appropriate. 

I think that sip:fluffy-staff@mcisco.com is a perfectly reasonable AOR for a
conference. Note my choice of the word AOR and yes this should be routable
by a proxy. Now this focus is controlling several media servers. One might
be doing video, another audio, and so on. There is certainly an argument to
be made that Megaco might be a better protocol for this but the netann draft
is about using SIP to perform service invocation on these devices. Let's say
that video is done ms1 and audio on ms2 for the same device. The focus needs
to be able to send a SIP service invocations to a proxy and get it routed to
the appropriate ms.

If I was generating a URL to give out externally to join a certain
conference. I might give lsik83dk8d@cisco.com to folks that were not the
moderator but give 89kkdiek8dke8@cisco.com to folks that did have moderator
privileges. Perhaps it is the type of conference that does not start mixing
until the moderator arrives. Now these are both valid names to reach the
same focus and they will result in service invocation to add them to a
mixer. I would expect the conference ID in the service invocation to be some
completely other string that perhaps has a conference id of 1234 in it.
Presumable folks that were moderators and ones that were not would all be
going to 1234 on the audio mixer and perhaps some other id 4567 on a
separate video mixer. I want to be able to route to the audio and video
mixer via a proxy for firewall traversal and to make TLS work without
devices other than the proxy having a TLS certificate.

I'll give it all more thought but I don't see how to make it work with a
name like sip:conf=1234@ms1.cisco.com being sent to the proxy but I see how
to make it work if I send sip:__sip-conf@ms1.cisco.com;conf=1234 even though
it is ugly. I'll think about more.

In summary, as the the netann draft nicely points out, I don't think I see
much of a relation between the external AOR that route to the focus and the
internal service invocation names that the focus sends to the mixer.

Cullen


On 4/12/04 12:57 PM, "Alan Johnston" <alan.johnston@mci.com> wrote:

> Cullen,
> 
> You are right - there is an inconsistency.  However, if we make netann
> consistent as you propose (and previous versions of netann use exactly this
> format), then netann becomes inconsistent with the conferencing framework
> and the rest of the conferencing work.
> 
> The conferencing framework assumes that the conference ID is embedded in
> the user portion of the URI, not in URI parameters.
> 
> Can you explain your NAT issue with this approach?
> 
> Thanks,
> Alan Johnston
> MCI
> sip:alan@sipstation.com
> 
> At 08:59 AM 4/12/2004 -0700, Cullen Jennings wrote:
> 
>> Their is a bit of weird inconsistency in the having the parameters that are
>> passed to the __sip-conf service invocation embedded in the user portion.
>> 
>> In the annc and dialog service, there are no parameters in the user portion
>> of the URI. Yet in the conf service, the conf id is embedded in the user
>> portion.
>> 
>> I think this is wrong. It should be
>> 
>> __sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>> 
>> This makes it feasible to specify routing rules for proxies that correctly
>> get all the traffic for the service to the media server. Particularly if the
>> media server is behind a NAT.
>> 
>> Could you change the draft to make the conf service invocation consistent
>> with the annc and dialog?
>> 
>> Cullen
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
> 
> 


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Date: Mon, 12 Apr 2004 20:04:01 -0700
From: Cullen Jennings <fluffy@cisco.com>
To: Andy Spitzer <woof@snowshore.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, <jvandyke@snowshore.com>
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First of all let me say thank you for the very complete response. Few
questions inline. Also I put some of my reply to this in the my reply to
Alan's email. 

On 4/12/04 11:32 AM, "Andy Spitzer" <woof@snowshore.com> wrote:

> Cullen,
> 
> It is expected that all sessions into the conference are controlled by an
> application server via third party call control.  The media servers and
> conf id's need not be externally visible.  The calls first go to the AS,
> then are sent to an MS for IVR (conference selection, record name, collect
> PIN, etc), then are sent to (possibly another) MS for the conference.  MS
> conf id's need not be seen externally, as the AS can re-write the messages
> any way it sees fit, acting as a B2BUA

Great, this is exactly the type of model I am interested in.

> 
> The application server responsible for controlling the conferences (and
> thus creating the conference ids) is the one which decides which
> conference lives on which MS.  It is up to the AS to REGISTER the
> conferences with the proxies if the conf ids are externally visible, as
> well as creating the conferences on the MS.

Agree the AS is picking the ids. In my case, I don't want to make the id
that are valid on the ms externally visible but I do need to route traffic
from the AS to the MS through a proxy because the MS is behind a firewall
and connecting over TLS.

> 
> You are correct in that the MS doesn't know the conf id's a-prioi.  The AS
> DOES know the conf id's, however.  It knows when they are supposed to
> start, end, etc., and as such it sends the appropriate REGSISTER methods

My problem is I can't do 3rd Party registration with TLS. The MS connects to
the proxy over TLS and needs to bind the TLS connection in the proxy to this
particular registration so 3rd party registrations don't work.

> to any proxies and any <create conference> MSCML commands to the
> appropriate MS.  It also removes the conferences from the proxies and MS's
> when the alloted time has expired.
> 
> The approach of externally visible ms1.cisco.com and ms2.cisco.com media
> servers does indeed mean you don't have to REGISTER each conference as a
> user, as the host name is used for routing instead.  That's a fine
> solution as well.  Even with that approach, something has to figure out
> the resources available on any given MS at a particular time, and divide
> up the usage across the MS's.  That is still in the realm of an AS.

Agreed it is still the AS making the decision about which resources to use
when. 

> 
> So, to walk through an example where the conf id is externally visible,
> the MS host name is not, and there is no initial IVR for users:
> 
> 1.  AS determines a 1 hour conference is to start at 10:00.  It assignes a
> conf id to the conference.  The conf id it picks is (to stick with your 6
> digit number example) 012345.
> 
> 2.  At 9:55 (just an example) the AS figures out which is the best MS to
> host this conference on.  It creates a control session by sending a
> <create conference> command to the appropriate MS (it picked ms1) to
> reserve resources:  sip:conf=012345@ms1.cisco.com
> 
> 3.  The AS then sends a REGISTER to the proxy for user conf=012345 on
> ms1.cisco.com

I assume here (based on next point) that you mean conf-012345@cisco.com is
bound to the contact ms1.cisco.com.  Note the domain of the registrar was
cisco.com not ms1.cisco.com. If I was allocating ms1 and ms2 to this
conference, I assume I would give them different conference Ids and focus
would keep track of how they all mapped together.

> 
> 4.  As external calls arrive, they can go via the proxy to the correct
> MS.  External calls are directed to conf=012345@cisco.com.  The proxy
> routes it to conf=012345@ms1.cisco.com
> 

Ok, I think it is this step where the 3rd party registration fails to work
for me. Rest of the steps make sense.

> 5.  At 10:05 the AS determines the conference is over.  It sends a BYE to
> the control session on the MS (the one it created in step 2).  This
> releases the conference resources and ends up sending BYEs to any
> remaining participant legs.
> 
> 6.  The AS removes the registration for user conf=012345 from the proxy.
> 
> 
> For more control over conference participants, usually all calls will go
> to the AS first, which will control an IVR session to decide which
> conference the caller is to be placed into, record the name, collect other
> info, etc.  Then it will place that user into the conference.
> 
> So, to walk through this example:
> 
> 1.  AS determines a 1 hour conference is to start at 10:00.  It assignes a
> conf id to the conference.  The conf id it picks is (to stick with your 6
> digit number example) 012345.
> 
> 2.  At 9:55 (just an example) the AS figures out which is the best MS to
> host this conference on.  It creates a control session by sending a
> <create conference> command to the appropriate MS (it picked ms1) to
> reserve resources:  sip:conf=012345@ms1.cisco.com
> 
> 3. External caller INVITES sip:conferences@cicso.com.  That hits the AS
> which selects an available IVR MS and collects the conference info from
> the caller via DTMF.  The AS sends an INVITE to the selected IVR MS:
> sip:ivr@ms2.cisco.com and uses MSCML in INFO methods to play prompts,
> collect digits, and record names.  When the IVR session is over it sends a
> BYE to the IVR MS.
> 
> 4. The AS verifies the caller via PIN or whatever, and places the caller
> into the conference by sending an INVITE to the selected conference MS:
> conf=012345@ms1.cisco.com
> 
> 5. After the user joins the conference, the AS uses MSCML in INFO methods
> sent to the control leg created in step 2 to play the name of the new
> participant to the conference, as in "{recorded name} has joined the
> conference".
> 
> 6. As participants leave, they send BYE's to the AS, which sends BYE's to
> the conference MS.  It uses MSCML in INFO methods sent to the control leg
> to inform the other participants "{recorded name} has left the conference".
> 
> 7. At 10:05, the AS decides to destroy the conference so it clears all the
> sessions it has with the external users and the MS.
> 
> No external proxies need to know which MS's own which conferences, as that
> is the AS's responsibility.
> 
> --Andy Spitzer
> woof@snowshore.com

Once again, thanks for the completely write up, it really did spell it out
clearly. 

> 
> 
> On Mon, 12 Apr 2004 10:29:19 -0700, Cullen Jennings <fluffy@cisco.com>
> wrote:
> 
>> 
>> Thanks for the fast response. I'm lost on how you imagine this working -
>> I
>> think you will need to walk me through it.  Let's say the domain is
>> cisco.com and I have two media servers ms1.cisco.com and ms2.cisco.com.
>> And
>> my conference ID's are 6 digit numbers.
>> 
>> Are you suggesting that ms1.cisco.com and ms2.cisco.com know that
>> __sip-conf=1 belongs to ms1 and __sip-conf=2 belongs to ms2 and they both
>> send a REGISTER message to the proxy and ms2 registers
>> __sip-conf=2@cisco.com with a contact pointing to ms2.cisco.com? I don't
>> see
>> how they could register all the possible conference Ids in any useful
>> way.
>> Mostly because the message needs to routed to the ms before the ms will
>> create the conference and conference needs to be created before it's id
>> can
>> be registered and the registration needs to happen before the routing and
>> thus we have a circular loop of dependencies in time.
>> 
>> I'm proposing that the proxy route to the correct media server by having
>> the
>> host portion of the URI be ms1.cisco.com or ms2.cisco.com
>> 
>> 
>> On 4/12/04 9:28 AM, "Andy Spitzer" <woof@snowshore.com> wrote:
>> 
>>> Cullen,
>>> 
>>> The reason for having the conf id as part of the user part, was because
>>> unlike the other services (annc, ivr, and dialog), conferencing is not
>>> stand alone.  With annc, ivr, and dialog, each session is unique in
>>> independent.  With conferencing, however, multiple sessions make up a
>>> particular conference.
>>> 
>>> As such, a particular conference must live on a particular media server.
>>> All participants into that conference must reach the same media server.
>>> By placing the conf id in the user part, it makes the conference look
>>> like
>>> a user on a particular media server, and as such any proxy can use the
>>> correct routing rules for users in order to reach the correct media
>>> server
>>> assigned to that conference.  Just the same way they would route users
>>> to
>>> a particular endpoint.
>>> 
>>> If the conf id was done as you describe, it would mean proxies would
>>> have
>>> to look into the parameters to determine which conference was being
>>> requested in order to determine which media server to INVITE.  We didn't
>>> want to force proxies to do that, so it was decided the user part was
>>> more
>>> proxy friendly.
>>> 
>>> Your approach is certainly feasible, and was indeed originally
>>> considered.  I believe it may be possible to support both
>>> simultainously.
>>> 
>>> --Andy Spitzer
>>> woof@snowshore.com
>>> 
>>> On Mon, 12 Apr 2004 08:59:14 -0700, Cullen Jennings <fluffy@cisco.com>
>>> wrote:
>>> 
>>>> 
>>>> Their is a bit of weird inconsistency in the having the parameters that
>>>> are
>>>> passed to the __sip-conf service invocation embedded in the user
>>>> portion.
>>>> 
>>>> In the annc and dialog service, there are no parameters in the user
>>>> portion
>>>> of the URI. Yet in the conf service, the conf id is embedded in the
>>>> user
>>>> portion.
>>>> 
>>>> I think this is wrong. It should be
>>>> 
>>>> __sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>>>> 
>>>> This makes it feasible to specify routing rules for proxies that
>>>> correctly
>>>> get all the traffic for the service to the media server. Particularly
>>>> if
>>>> the
>>>> media server is behind a NAT.
>>>> 
>>>> Could you change the draft to make the conf service invocation
>>>> consistent
>>>> with the annc and dialog?
>>>> 
>>>> Cullen
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>> 
>>> 
>>> 
>>> 
>> 
>> 
> 
> 
> 
> 


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Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7017978D4@esebe019.ntc.nokia.com>
Thread-Topic: Joint Interim Meeting Mon May 24 - Wed May 26
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To: <rohan@cisco.com>, <sipping@ietf.org>
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Subject: [Sipping] RE: Joint Interim Meeting Mon May 24 - Wed May 26
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Here is the hotel information:

Renaissance Boston Hotel Bedford
44 Middlesex Turnpike Bedford, MA 01730 USA
Phone: 1 781-275-5500 Fax: 1 781-275-3042

http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB

There are 50 rooms booked for this event at the rate of $99 per night =
(from Sunday 23rd until Wednesday 26th). These rooms will be held until =
3th May, so please reserve your room on or before that date.

You can reserve online or by telephone using the Group Code: NONOKA

Also, please use the links that Rohan provided to confirm your =
attendance. This will aid us in determining the catering requirements. =
Here is the link again:

mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim=
]yes

Thanks,
Hisham

> -----Original Message-----
> From: ext Rohan Mahy [mailto:rohan@cisco.com]
> Sent: 06.April.2004 20:54
> To: sipping (E-mail)
> Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo;
> Rohan Mahy; Alan Johnston; Adam Roach
> Subject: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> Hello All,
>=20
> Please mark your calendars.
>=20
> I'd like to announce a joint interim of the SIP, SIPPING,=20
> SIMPLE, and =20
> XCON WGs.
> Nokia has generously agreed to host the event either at their=20
> offices =20
> near Boston or in a neighboring hotel during the week of May 24.
>=20
> Please reserve the following dates for the WGs that you are=20
> interested =20
> in:
> Monday May 24 SIP and SIPPING  tentatively 9 am start
> Tuesday May 25 SIMPLE   All day
> Wednesday May 26 XCON  tentatively finish at 4pm so folks can catch =20
> flights.
> (the actual agenda is subject to change, including the dates=20
> of various =20
> WG meetings).
>=20
> There will be no fee, however please let Hisham and myself=20
> know if you =20
> are planning to attend by clicking on the Yes or Maybe links below:
>=20
> mailto:rohan@cisco.com?=20
> Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> mailto:rohan@cisco.com?=20
> Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
>=20
> Wireless will be provided. As usual, I will be making=20
> T-shirts.  If you =20
> are interested in buying one, please send your shirt size to me.
>=20
> More details on the agenda and hotel accommodations will be=20
> forthcoming =20
> shortly.
>=20
> thanks,
> -rohan
>=20
>=20

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



From exim@www1.ietf.org  Tue Apr 13 03:39:49 2004
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Thread-Topic: Joint Interim Meeting Mon May 24 - Wed May 26
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To: <rohan@cisco.com>, <sipping@ietf.org>
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        <jon.peterson@neustar.biz>, <hardie@qualcomm.com>,
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One small correction:

Rooms will be help until 3rd May.

Thanks,
Hisham

> -----Original Message-----
> From: Khartabil Hisham (Nokia-TP-MSW/Helsinki)=20
> Sent: 13.April.2004 10:00
> To: 'ext Rohan Mahy'; sipping (E-mail)
> Cc: Dean Willis; Allison Mankin; Robert Sparks; John Peterson; Ted
> Hardie; Gonzalo Camarillo; Alan Johnston; Adam Roach
> Subject: RE: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> Here is the hotel information:
>=20
> Renaissance Boston Hotel Bedford
> 44 Middlesex Turnpike Bedford, MA 01730 USA
> Phone: 1 781-275-5500 Fax: 1 781-275-3042
>=20
> http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB
>=20
> There are 50 rooms booked for this event at the rate of $99=20
> per night (from Sunday 23rd until Wednesday 26th). These=20
> rooms will be held until 3th May, so please reserve your room=20
> on or before that date.
>=20
> You can reserve online or by telephone using the Group Code: NONOKA
>=20
> Also, please use the links that Rohan provided to confirm=20
> your attendance. This will aid us in determining the catering=20
> requirements. Here is the link again:
>=20
> mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[
> interim]yes
>=20
> Thanks,
> Hisham
>=20
> > -----Original Message-----
> > From: ext Rohan Mahy [mailto:rohan@cisco.com]
> > Sent: 06.April.2004 20:54
> > To: sipping (E-mail)
> > Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> > Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo Camarillo;
> > Rohan Mahy; Alan Johnston; Adam Roach
> > Subject: Joint Interim Meeting Mon May 24 - Wed May 26
> >=20
> >=20
> > Hello All,
> >=20
> > Please mark your calendars.
> >=20
> > I'd like to announce a joint interim of the SIP, SIPPING,=20
> > SIMPLE, and =20
> > XCON WGs.
> > Nokia has generously agreed to host the event either at their=20
> > offices =20
> > near Boston or in a neighboring hotel during the week of May 24.
> >=20
> > Please reserve the following dates for the WGs that you are=20
> > interested =20
> > in:
> > Monday May 24 SIP and SIPPING  tentatively 9 am start
> > Tuesday May 25 SIMPLE   All day
> > Wednesday May 26 XCON  tentatively finish at 4pm so folks=20
> can catch =20
> > flights.
> > (the actual agenda is subject to change, including the dates=20
> > of various =20
> > WG meetings).
> >=20
> > There will be no fee, however please let Hisham and myself=20
> > know if you =20
> > are planning to attend by clicking on the Yes or Maybe links below:
> >=20
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> > mailto:rohan@cisco.com?=20
> > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
> >=20
> > Wireless will be provided. As usual, I will be making=20
> > T-shirts.  If you =20
> > are interested in buying one, please send your shirt size to me.
> >=20
> > More details on the agenda and hotel accommodations will be=20
> > forthcoming =20
> > shortly.
> >=20
> > thanks,
> > -rohan
> >=20
> >=20
>=20

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



From exim@www1.ietf.org  Tue Apr 13 09:22:12 2004
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Subject: RE: [Sipping] RE: Joint Interim Meeting Mon May 24 - Wed May 26
Date: Tue, 13 Apr 2004 15:58:29 +0300
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It was pointed out to me that the Group Code for booking was incorrect. =
Here is the correct Group Code: NOKNOKA.

Apologies for the inconvenience.
Hisham

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext hisham.khartabil@nokia.com
> Sent: 13.April.2004 10:03
> To: rohan@cisco.com; sipping@ietf.org
> Cc: dean.willis@softarmor.com; mankin@psg.com; rjsparks@nostrum.com;
> jon.peterson@neustar.biz; hardie@qualcomm.com;
> Gonzalo.Camarillo@ericsson.com; alan.johnston@wcom.com;
> adam@dynamicsoft.com
> Subject: [Sipping] RE: Joint Interim Meeting Mon May 24 - Wed May 26
>=20
>=20
> One small correction:
>=20
> Rooms will be help until 3rd May.
>=20
> Thanks,
> Hisham
>=20
> > -----Original Message-----
> > From: Khartabil Hisham (Nokia-TP-MSW/Helsinki)=20
> > Sent: 13.April.2004 10:00
> > To: 'ext Rohan Mahy'; sipping (E-mail)
> > Cc: Dean Willis; Allison Mankin; Robert Sparks; John Peterson; Ted
> > Hardie; Gonzalo Camarillo; Alan Johnston; Adam Roach
> > Subject: RE: Joint Interim Meeting Mon May 24 - Wed May 26
> >=20
> >=20
> > Here is the hotel information:
> >=20
> > Renaissance Boston Hotel Bedford
> > 44 Middlesex Turnpike Bedford, MA 01730 USA
> > Phone: 1 781-275-5500 Fax: 1 781-275-3042
> >=20
> > http://marriott.com/property/propertyPage.mi?marshaCode=3DBOSSB
> >=20
> > There are 50 rooms booked for this event at the rate of $99=20
> > per night (from Sunday 23rd until Wednesday 26th). These=20
> > rooms will be held until 3th May, so please reserve your room=20
> > on or before that date.
> >=20
> > You can reserve online or by telephone using the Group Code: NONOKA
> >=20
> > Also, please use the links that Rohan provided to confirm=20
> > your attendance. This will aid us in determining the catering=20
> > requirements. Here is the link again:
> >=20
> > mailto:rohan@cisco.com?Cc=3Dhisham.khartabil@nokia.com&Subject=3D[
> > interim]yes
> >=20
> > Thanks,
> > Hisham
> >=20
> > > -----Original Message-----
> > > From: ext Rohan Mahy [mailto:rohan@cisco.com]
> > > Sent: 06.April.2004 20:54
> > > To: sipping (E-mail)
> > > Cc: Dean Willis; Khartabil Hisham (Nokia-TP-MSW/Helsinki); Allison
> > > Mankin; Robert Sparks; John Peterson; Ted Hardie; Gonzalo=20
> Camarillo;
> > > Rohan Mahy; Alan Johnston; Adam Roach
> > > Subject: Joint Interim Meeting Mon May 24 - Wed May 26
> > >=20
> > >=20
> > > Hello All,
> > >=20
> > > Please mark your calendars.
> > >=20
> > > I'd like to announce a joint interim of the SIP, SIPPING,=20
> > > SIMPLE, and =20
> > > XCON WGs.
> > > Nokia has generously agreed to host the event either at their=20
> > > offices =20
> > > near Boston or in a neighboring hotel during the week of May 24.
> > >=20
> > > Please reserve the following dates for the WGs that you are=20
> > > interested =20
> > > in:
> > > Monday May 24 SIP and SIPPING  tentatively 9 am start
> > > Tuesday May 25 SIMPLE   All day
> > > Wednesday May 26 XCON  tentatively finish at 4pm so folks=20
> > can catch =20
> > > flights.
> > > (the actual agenda is subject to change, including the dates=20
> > > of various =20
> > > WG meetings).
> > >=20
> > > There will be no fee, however please let Hisham and myself=20
> > > know if you =20
> > > are planning to attend by clicking on the Yes or Maybe=20
> links below:
> > >=20
> > > mailto:rohan@cisco.com?=20
> > > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]yes
> > > mailto:rohan@cisco.com?=20
> > > Cc=3Dhisham.khartabil@nokia.com&Subject=3D[interim]maybe
> > >=20
> > > Wireless will be provided. As usual, I will be making=20
> > > T-shirts.  If you =20
> > > are interested in buying one, please send your shirt size to me.
> > >=20
> > > More details on the agenda and hotel accommodations will be=20
> > > forthcoming =20
> > > shortly.
> > >=20
> > > thanks,
> > > -rohan
> > >=20
> > >=20
> >=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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Message-ID: <407BF1AD.2000201@cisco.com>
Date: Tue, 13 Apr 2004 09:57:01 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Cullen Jennings <fluffy@cisco.com>
CC: hisham.khartabil@nokia.com, sipping@ietf.org,
        Orit Levin <oritl@microsoft.com>,
        Henning Schulzrinne <hgs@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sipping] User with multiple sessions in Conference Package
References: <BCA02363.3956C%fluffy@cisco.com>
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Cullen,

I think what you are trying to achieve is a good thing. But I think the 
focus may need to be psychic to achieve it. How is it supposed to know 
which dialogs to group together? I guess it could group all the things 
that share the same From address, but that won't get all the cases - 
probably not the interesting ones.

	Paul

Cullen Jennings wrote:
> I'll take a stab at explaining :-)
> 
> If someone has a roster displayed on some device - they want to see that one
> person, Cullen is in the conference they don't want to sees that their are 4
> participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
> Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More importantly, they want to
> be able to use a user interface to boot Cullen out of the conference not
> have to go and somehow magically figure out which devices to boot out since
> they are being used by Cullen.
> 
> If their are 10 participants that are all Anonymous, you still want to be
> able to know their are 10 participants not 1 participant called Anonymous
> that seems to have dialed in from 10 devices.
> 
> Given that SIP AOR correspond to a user and is not a single device. There
> are basically three levels off aggregation happening here.
> 
> There are users. Each user can have multiple SIP dialogs that got them into
> the conference. Each one of these SIP dialogs can be started and terminated
> independently. For example one may be dialed out while another dialed in and
> one may be active while others may be ongoing. Inside of each SIP dialog,
> there can be multiple media stream (audio, video, etc). The three levels are
> Users, SIP Sessions or Dialogs, Media Sessions. Unfortunately, the draft as
> written supports two things and the User level information and the SIP
> session level information is mixed together. I think this needs to get
> fixed. 
> 
> One more use case, many audio conference systems announce when a user joins
> of leaves. This is really annoying when someone transfers from the cell
> phone to desk phone. Let me give you a scenario that I think XCON should
> support. 
> 
> Cullen dials in to conf from cell phone and is announced. Cullen then dials
> in from desk phone and is *not* announced because the user Cullen is already
> in the conference. Cullen hangs up cell phone and this is *not* announced
> because Cullen is still in the conference. Cullen hangs up his desk phone
> and the conference announces that "Fluffy has left the building" because he
> is no longer in the conference.
> 
> 
> On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com" <hisham.khartabil@nokia.com>
> wrote:
> 
> 
>>Why do you need the focus to know that it is the same user (human)? I don't
>>understand the use case. Can you elaborate?
>>
>>Thanks,
>>Hisham
>>
>>
>>>-----Original Message-----
>>>From: sipping-admin@ietf.org
>>>[mailto:sipping-admin@ietf.org]On Behalf Of
>>>ext Cullen Jennings
>>>Sent: 08.April.2004 03:30
>>>To: 'sipping@ietf.org'
>>>Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
>>>Subject: [Sipping] User with multiple sessions in Conference Package
>>>
>>>
>>>
>>>In a conference, here are many cases where a single user may
>>>have set up
>>>multiple SIP dialogs with the focus. For example, using one
>>>UA instance for
>>>audio and another UA instance for IM. Or connected on both
>>>cell phone and on
>>>desk phone in office. Right now the conference package seems a little
>>>underspecified in how to deal with this and we need to clear this up.
>>>
>>>I'm tossing out two proposal below. Comments? Ideas? Can we
>>>choose one?
>>>Could be  close something like this on this list?
>>>
>>>--------------------------------------------------------------
>>>
>>>Proposal 1: 
>>>
>>>The User Element gets a new attribute called contact. This attribute
>>>contains the contact that the focus used for this dialog. Each dialog
>>>results in a separate User Element instance and the uri is
>>>used to correlate
>>>the fact that they are the same user. For anonymous calls the
>>>system needs
>>>to generate separate anonymous user uri's so they can be
>>>separated from one
>>>another. 
>>>
>>>For Example,
>>>
>>><conference-info>
>>>    <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
>>>        <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>    </user>    
>>>    <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
>>>        <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>    </user>
>>></conference-info>
>>>
>>>---------------------------------------------------------------
>>>
>>>Proposal 2: 
>>>
>>>The data in the User Element gets split between a User
>>>Element and a new
>>>Session Element that is placed inside the User Element. The
>>>media stream
>>>elements are put inside the Session.
>>>
>>><conference-info>
>>>    <user uri="fluffy@cisco.com">
>>>        <session contact="1111@focus.cisco.com">
>>>            <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>        </session>
>>>        <session contact="2222@focus.cisco.com">
>>>            <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>        </session>
>>>    </user>
>>></conference-info>
>>>
>>>
>>>_______________________________________________
>>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>This list is for NEW development of the application of SIP
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sip@ietf.org for new developments of core SIP
>>>
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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



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Subject: [Sipping] RFC 3680 on A Session Initiation Protocol (SIP) Event Package for Registrations
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3680

        Title:      A Session Initiation Protocol (SIP) Event Package
                    for Registrations
        Author(s):  J. Rosenberg
        Status:     Standards Track
        Date:       March 2004
        Mailbox:    jdrosen@dynamicsoft.com
        Pages:      26
        Characters: 35403
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sipping-reg-event-00.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3680.txt


This document defines a Session Initiation Protocol (SIP) event
package for registrations.  Through its REGISTER method, SIP allows a
user agent to create, modify, and delete registrations.  Registrations
can also be altered by administrators in order to enforce policy.  As
a result, these registrations represent a piece of state in the
network that can change dynamically.  There are many cases where a
user agent would like to be notified of changes in this state.  This
event package defines a mechanism by which those user agents can
request and obtain such notifications.

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
for improvements.  Please refer to the current edition of the
"Internet Official Protocol Standards" (STD 1) for the
standardization state and status of this protocol.  Distribution
of this memo is unlimited.

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Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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A new Request for Comments is now available in online RFC libraries.


        BCP 85
        RFC 3725

        Title:      Best Current Practices for Third Party Call
                    Control (3pcc) in the Session Initiation Protocol
                    (SIP)
        Author(s):  J. Rosenberg, J. Peterson, H. Schulzrinne,
                    G. Camarillo
        Status:     Best Current Practice
        Date:       April 2004
        Mailbox:    jdrosen@dynamicsoft.com, jon.peterson@neustar.biz,
                    schulzrinne@cs.columbia.edu
        Pages:      31
        Characters: 77308
        SeeAlso:    BCP 85

        I-D Tag:    draft-ietf-sipping-3pcc-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3725.txt


Third party call control refers to the ability of one entity to
create a call in which communication is actually between other
parties.  Third party call control is possible using the mechanisms
specified within the Session Initiation Protocol (SIP).  However,
there are several possible approaches, each with different benefits
and drawbacks.  This document discusses best current practices for the
usage of SIP for third party call control.

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements.  Distribution of this memo is unlimited.

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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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To: Cullen Jennings <fluffy@cisco.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, jvandyke@snowshore.com
References: <BCA00AE2.39526%fluffy@cisco.com>
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Cullen,

The reason for having the conf id as part of the user part, was because 
unlike the other services (annc, ivr, and dialog), conferencing is not 
stand alone.  With annc, ivr, and dialog, each session is unique in 
independent.  With conferencing, however, multiple sessions make up a 
particular conference.

As such, a particular conference must live on a particular media server.  
All participants into that conference must reach the same media server.  
By placing the conf id in the user part, it makes the conference look like 
a user on a particular media server, and as such any proxy can use the 
correct routing rules for users in order to reach the correct media server 
assigned to that conference.  Just the same way they would route users to 
a particular endpoint.

If the conf id was done as you describe, it would mean proxies would have 
to look into the parameters to determine which conference was being 
requested in order to determine which media server to INVITE.  We didn't 
want to force proxies to do that, so it was decided the user part was more 
proxy friendly.

Your approach is certainly feasible, and was indeed originally 
considered.  I believe it may be possible to support both simultainously.

--Andy Spitzer
woof@snowshore.com

On Mon, 12 Apr 2004 08:59:14 -0700, Cullen Jennings <fluffy@cisco.com> 
wrote:

>
> Their is a bit of weird inconsistency in the having the parameters that 
> are
> passed to the __sip-conf service invocation embedded in the user portion.
>
> In the annc and dialog service, there are no parameters in the user 
> portion
> of the URI. Yet in the conf service, the conf id is embedded in the user
> portion.
>
> I think this is wrong. It should be
>
> __sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>
> This makes it feasible to specify routing rules for proxies that 
> correctly
> get all the traffic for the service to the media server. Particularly if 
> the
> media server is behind a NAT.
>
> Could you change the draft to make the conf service invocation consistent
> with the annc and dialog?
>
> Cullen
>
>
>
>
>
>
>
>
>
>




_______________________________________________
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This list is for NEW development of the application of SIP
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To: Cullen Jennings <fluffy@cisco.com>, Eric Burger <eburger@snowshore.com>,
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Cullen,

It is expected that all sessions into the conference are controlled by an 
application server via third party call control.  The media servers and 
conf id's need not be externally visible.  The calls first go to the AS, 
then are sent to an MS for IVR (conference selection, record name, collect 
PIN, etc), then are sent to (possibly another) MS for the conference.  MS 
conf id's need not be seen externally, as the AS can re-write the messages 
any way it sees fit, acting as a B2BUA

The application server responsible for controlling the conferences (and 
thus creating the conference ids) is the one which decides which 
conference lives on which MS.  It is up to the AS to REGISTER the 
conferences with the proxies if the conf ids are externally visible, as 
well as creating the conferences on the MS.

You are correct in that the MS doesn't know the conf id's a-prioi.  The AS 
DOES know the conf id's, however.  It knows when they are supposed to 
start, end, etc., and as such it sends the appropriate REGSISTER methods 
to any proxies and any <create conference> MSCML commands to the 
appropriate MS.  It also removes the conferences from the proxies and MS's 
when the alloted time has expired.

The approach of externally visible ms1.cisco.com and ms2.cisco.com media 
servers does indeed mean you don't have to REGISTER each conference as a 
user, as the host name is used for routing instead.  That's a fine 
solution as well.  Even with that approach, something has to figure out 
the resources available on any given MS at a particular time, and divide 
up the usage across the MS's.  That is still in the realm of an AS.

So, to walk through an example where the conf id is externally visible, 
the MS host name is not, and there is no initial IVR for users:

1.  AS determines a 1 hour conference is to start at 10:00.  It assignes a 
conf id to the conference.  The conf id it picks is (to stick with your 6 
digit number example) 012345.

2.  At 9:55 (just an example) the AS figures out which is the best MS to 
host this conference on.  It creates a control session by sending a 
<create conference> command to the appropriate MS (it picked ms1) to 
reserve resources:  sip:conf=012345@ms1.cisco.com

3.  The AS then sends a REGISTER to the proxy for user conf=012345 on 
ms1.cisco.com

4.  As external calls arrive, they can go via the proxy to the correct 
MS.  External calls are directed to conf=012345@cisco.com.  The proxy 
routes it to conf=012345@ms1.cisco.com

5.  At 10:05 the AS determines the conference is over.  It sends a BYE to 
the control session on the MS (the one it created in step 2).  This 
releases the conference resources and ends up sending BYEs to any 
remaining participant legs.

6.  The AS removes the registration for user conf=012345 from the proxy.


For more control over conference participants, usually all calls will go 
to the AS first, which will control an IVR session to decide which 
conference the caller is to be placed into, record the name, collect other 
info, etc.  Then it will place that user into the conference.

So, to walk through this example:

1.  AS determines a 1 hour conference is to start at 10:00.  It assignes a 
conf id to the conference.  The conf id it picks is (to stick with your 6 
digit number example) 012345.

2.  At 9:55 (just an example) the AS figures out which is the best MS to 
host this conference on.  It creates a control session by sending a 
<create conference> command to the appropriate MS (it picked ms1) to 
reserve resources:  sip:conf=012345@ms1.cisco.com

3. External caller INVITES sip:conferences@cicso.com.  That hits the AS 
which selects an available IVR MS and collects the conference info from 
the caller via DTMF.  The AS sends an INVITE to the selected IVR MS: 
sip:ivr@ms2.cisco.com and uses MSCML in INFO methods to play prompts, 
collect digits, and record names.  When the IVR session is over it sends a 
BYE to the IVR MS.

4. The AS verifies the caller via PIN or whatever, and places the caller 
into the conference by sending an INVITE to the selected conference MS: 
conf=012345@ms1.cisco.com

5. After the user joins the conference, the AS uses MSCML in INFO methods 
sent to the control leg created in step 2 to play the name of the new 
participant to the conference, as in "{recorded name} has joined the 
conference".

6. As participants leave, they send BYE's to the AS, which sends BYE's to 
the conference MS.  It uses MSCML in INFO methods sent to the control leg 
to inform the other participants "{recorded name} has left the conference".

7. At 10:05, the AS decides to destroy the conference so it clears all the 
sessions it has with the external users and the MS.

No external proxies need to know which MS's own which conferences, as that 
is the AS's responsibility.

--Andy Spitzer
woof@snowshore.com


On Mon, 12 Apr 2004 10:29:19 -0700, Cullen Jennings <fluffy@cisco.com> 
wrote:

>
> Thanks for the fast response. I'm lost on how you imagine this working - 
> I
> think you will need to walk me through it.  Let's say the domain is
> cisco.com and I have two media servers ms1.cisco.com and ms2.cisco.com. 
> And
> my conference ID's are 6 digit numbers.
>
> Are you suggesting that ms1.cisco.com and ms2.cisco.com know that
> __sip-conf=1 belongs to ms1 and __sip-conf=2 belongs to ms2 and they both
> send a REGISTER message to the proxy and ms2 registers
> __sip-conf=2@cisco.com with a contact pointing to ms2.cisco.com? I don't 
> see
> how they could register all the possible conference Ids in any useful 
> way.
> Mostly because the message needs to routed to the ms before the ms will
> create the conference and conference needs to be created before it's id 
> can
> be registered and the registration needs to happen before the routing and
> thus we have a circular loop of dependencies in time.
>
> I'm proposing that the proxy route to the correct media server by having 
> the
> host portion of the URI be ms1.cisco.com or ms2.cisco.com
>
>
> On 4/12/04 9:28 AM, "Andy Spitzer" <woof@snowshore.com> wrote:
>
>> Cullen,
>>
>> The reason for having the conf id as part of the user part, was because
>> unlike the other services (annc, ivr, and dialog), conferencing is not
>> stand alone.  With annc, ivr, and dialog, each session is unique in
>> independent.  With conferencing, however, multiple sessions make up a
>> particular conference.
>>
>> As such, a particular conference must live on a particular media server.
>> All participants into that conference must reach the same media server.
>> By placing the conf id in the user part, it makes the conference look 
>> like
>> a user on a particular media server, and as such any proxy can use the
>> correct routing rules for users in order to reach the correct media 
>> server
>> assigned to that conference.  Just the same way they would route users 
>> to
>> a particular endpoint.
>>
>> If the conf id was done as you describe, it would mean proxies would 
>> have
>> to look into the parameters to determine which conference was being
>> requested in order to determine which media server to INVITE.  We didn't
>> want to force proxies to do that, so it was decided the user part was 
>> more
>> proxy friendly.
>>
>> Your approach is certainly feasible, and was indeed originally
>> considered.  I believe it may be possible to support both 
>> simultainously.
>>
>> --Andy Spitzer
>> woof@snowshore.com
>>
>> On Mon, 12 Apr 2004 08:59:14 -0700, Cullen Jennings <fluffy@cisco.com>
>> wrote:
>>
>>>
>>> Their is a bit of weird inconsistency in the having the parameters that
>>> are
>>> passed to the __sip-conf service invocation embedded in the user 
>>> portion.
>>>
>>> In the annc and dialog service, there are no parameters in the user
>>> portion
>>> of the URI. Yet in the conf service, the conf id is embedded in the 
>>> user
>>> portion.
>>>
>>> I think this is wrong. It should be
>>>
>>> __sip-conf@mediaserver.example.net;conf=uniqueIdentifier
>>>
>>> This makes it feasible to specify routing rules for proxies that
>>> correctly
>>> get all the traffic for the service to the media server. Particularly 
>>> if
>>> the
>>> media server is behind a NAT.
>>>
>>> Could you change the draft to make the conf service invocation 
>>> consistent
>>> with the annc and dialog?
>>>
>>> Cullen
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>
>>
>>
>>
>
>




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



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From: hisham.khartabil@nokia.com
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Subject: RE: [Sipping] User with multiple sessions in Conference Package
Date: Wed, 14 Apr 2004 12:40:01 +0300
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To: <pkyzivat@cisco.com>, <fluffy@cisco.com>
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It could based on the asserted identity (unless it is a public =
conference and no authentication is needed). By asserted identity I mean =
any authentication method available to the conference server/focus.

/Hisham

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Paul Kyzivat
> Sent: 13.April.2004 16:57
> To: Cullen Jennings
> Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org; Orit
> Levin; Henning Schulzrinne; Jonathan Rosenberg
> Subject: Re: [Sipping] User with multiple sessions in=20
> Conference Package
>=20
>=20
> Cullen,
>=20
> I think what you are trying to achieve is a good thing. But I=20
> think the=20
> focus may need to be psychic to achieve it. How is it=20
> supposed to know=20
> which dialogs to group together? I guess it could group all=20
> the things=20
> that share the same From address, but that won't get all the cases -=20
> probably not the interesting ones.
>=20
> 	Paul
>=20
> Cullen Jennings wrote:
> > I'll take a stab at explaining :-)
> >=20
> > If someone has a roster displayed on some device - they=20
> want to see that one
> > person, Cullen is in the conference they don't want to sees=20
> that their are 4
> > participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
> > Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More=20
> importantly, they want to
> > be able to use a user interface to boot Cullen out of the=20
> conference not
> > have to go and somehow magically figure out which devices=20
> to boot out since
> > they are being used by Cullen.
> >=20
> > If their are 10 participants that are all Anonymous, you=20
> still want to be
> > able to know their are 10 participants not 1 participant=20
> called Anonymous
> > that seems to have dialed in from 10 devices.
> >=20
> > Given that SIP AOR correspond to a user and is not a single=20
> device. There
> > are basically three levels off aggregation happening here.
> >=20
> > There are users. Each user can have multiple SIP dialogs=20
> that got them into
> > the conference. Each one of these SIP dialogs can be=20
> started and terminated
> > independently. For example one may be dialed out while=20
> another dialed in and
> > one may be active while others may be ongoing. Inside of=20
> each SIP dialog,
> > there can be multiple media stream (audio, video, etc). The=20
> three levels are
> > Users, SIP Sessions or Dialogs, Media Sessions.=20
> Unfortunately, the draft as
> > written supports two things and the User level information=20
> and the SIP
> > session level information is mixed together. I think this=20
> needs to get
> > fixed.=20
> >=20
> > One more use case, many audio conference systems announce=20
> when a user joins
> > of leaves. This is really annoying when someone transfers=20
> from the cell
> > phone to desk phone. Let me give you a scenario that I=20
> think XCON should
> > support.=20
> >=20
> > Cullen dials in to conf from cell phone and is announced.=20
> Cullen then dials
> > in from desk phone and is *not* announced because the user=20
> Cullen is already
> > in the conference. Cullen hangs up cell phone and this is=20
> *not* announced
> > because Cullen is still in the conference. Cullen hangs up=20
> his desk phone
> > and the conference announces that "Fluffy has left the=20
> building" because he
> > is no longer in the conference.
> >=20
> >=20
> > On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com"=20
> <hisham.khartabil@nokia.com>
> > wrote:
> >=20
> >=20
> >>Why do you need the focus to know that it is the same user=20
> (human)? I don't
> >>understand the use case. Can you elaborate?
> >>
> >>Thanks,
> >>Hisham
> >>
> >>
> >>>-----Original Message-----
> >>>From: sipping-admin@ietf.org
> >>>[mailto:sipping-admin@ietf.org]On Behalf Of
> >>>ext Cullen Jennings
> >>>Sent: 08.April.2004 03:30
> >>>To: 'sipping@ietf.org'
> >>>Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
> >>>Subject: [Sipping] User with multiple sessions in=20
> Conference Package
> >>>
> >>>
> >>>
> >>>In a conference, here are many cases where a single user may
> >>>have set up
> >>>multiple SIP dialogs with the focus. For example, using one
> >>>UA instance for
> >>>audio and another UA instance for IM. Or connected on both
> >>>cell phone and on
> >>>desk phone in office. Right now the conference package=20
> seems a little
> >>>underspecified in how to deal with this and we need to=20
> clear this up.
> >>>
> >>>I'm tossing out two proposal below. Comments? Ideas? Can we
> >>>choose one?
> >>>Could be  close something like this on this list?
> >>>
> >>>--------------------------------------------------------------
> >>>
> >>>Proposal 1:=20
> >>>
> >>>The User Element gets a new attribute called contact. This=20
> attribute
> >>>contains the contact that the focus used for this dialog.=20
> Each dialog
> >>>results in a separate User Element instance and the uri is
> >>>used to correlate
> >>>the fact that they are the same user. For anonymous calls the
> >>>system needs
> >>>to generate separate anonymous user uri's so they can be
> >>>separated from one
> >>>another.=20
> >>>
> >>>For Example,
> >>>
> >>><conference-info>
> >>>    <user uri=3D"fluffy@cisco.com" =
contact=3D"1111@focus.cisco.com">
> >>>        <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>>    </user>   =20
> >>>    <user uri=3D"fluffy@cisco.com" =
contact=3D"2222@focus.cisco.com">
> >>>        <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>>    </user>
> >>></conference-info>
> >>>
> >>>---------------------------------------------------------------
> >>>
> >>>Proposal 2:=20
> >>>
> >>>The data in the User Element gets split between a User
> >>>Element and a new
> >>>Session Element that is placed inside the User Element. The
> >>>media stream
> >>>elements are put inside the Session.
> >>>
> >>><conference-info>
> >>>    <user uri=3D"fluffy@cisco.com">
> >>>        <session contact=3D"1111@focus.cisco.com">
> >>>            <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>>        </session>
> >>>        <session contact=3D"2222@focus.cisco.com">
> >>>            <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>>        </session>
> >>>    </user>
> >>></conference-info>
> >>>
> >>>
> >>>_______________________________________________
> >>>Sipping mailing list =20
https://www1.ietf.org/mailman/listinfo/sipping
>>>This list is for NEW development of the application of SIP
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sip@ietf.org for new developments of core SIP
>>>
>>
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20


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

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



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Date: Wed, 14 Apr 2004 09:22:53 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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        hgs@cs.columbia.edu, jdrosen@dynamicsoft.com
Subject: Re: [Sipping] User with multiple sessions in Conference Package
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hisham.khartabil@nokia.com wrote:
> It could based on the asserted identity (unless it is a public conference and no authentication is needed). By asserted identity I mean any authentication method available to the conference server/focus.

I wonder how often the endpoints will be authenticated. In most of the 
conference servers I am familiar with today, authentication is simply 
based on knowing the conference id.

And it probably won't be likely that all the devices a user employs will 
be bound to the the user in a way known to the conference server. (E.g. 
I am calling in from the phone tel:+1-978-555-1234, which may just be 
some phone in a conference room, as well as the IM client 
sip:pkyzivat@cisco.com. How does the phone get associated with me?)

	Paul

>>-----Original Message-----
>>From: sipping-admin@ietf.org 
>>[mailto:sipping-admin@ietf.org]On Behalf Of
>>ext Paul Kyzivat
>>Sent: 13.April.2004 16:57
>>To: Cullen Jennings
>>Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org; Orit
>>Levin; Henning Schulzrinne; Jonathan Rosenberg
>>Subject: Re: [Sipping] User with multiple sessions in 
>>Conference Package
>>
>>
>>Cullen,
>>
>>I think what you are trying to achieve is a good thing. But I 
>>think the 
>>focus may need to be psychic to achieve it. How is it 
>>supposed to know 
>>which dialogs to group together? I guess it could group all 
>>the things 
>>that share the same From address, but that won't get all the cases - 
>>probably not the interesting ones.
>>
>>	Paul
>>
>>Cullen Jennings wrote:
>>
>>>I'll take a stab at explaining :-)
>>>
>>>If someone has a roster displayed on some device - they 
>>
>>want to see that one
>>
>>>person, Cullen is in the conference they don't want to sees 
>>
>>that their are 4
>>
>>>participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
>>>Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More 
>>
>>importantly, they want to
>>
>>>be able to use a user interface to boot Cullen out of the 
>>
>>conference not
>>
>>>have to go and somehow magically figure out which devices 
>>
>>to boot out since
>>
>>>they are being used by Cullen.
>>>
>>>If their are 10 participants that are all Anonymous, you 
>>
>>still want to be
>>
>>>able to know their are 10 participants not 1 participant 
>>
>>called Anonymous
>>
>>>that seems to have dialed in from 10 devices.
>>>
>>>Given that SIP AOR correspond to a user and is not a single 
>>
>>device. There
>>
>>>are basically three levels off aggregation happening here.
>>>
>>>There are users. Each user can have multiple SIP dialogs 
>>
>>that got them into
>>
>>>the conference. Each one of these SIP dialogs can be 
>>
>>started and terminated
>>
>>>independently. For example one may be dialed out while 
>>
>>another dialed in and
>>
>>>one may be active while others may be ongoing. Inside of 
>>
>>each SIP dialog,
>>
>>>there can be multiple media stream (audio, video, etc). The 
>>
>>three levels are
>>
>>>Users, SIP Sessions or Dialogs, Media Sessions. 
>>
>>Unfortunately, the draft as
>>
>>>written supports two things and the User level information 
>>
>>and the SIP
>>
>>>session level information is mixed together. I think this 
>>
>>needs to get
>>
>>>fixed. 
>>>
>>>One more use case, many audio conference systems announce 
>>
>>when a user joins
>>
>>>of leaves. This is really annoying when someone transfers 
>>
>>from the cell
>>
>>>phone to desk phone. Let me give you a scenario that I 
>>
>>think XCON should
>>
>>>support. 
>>>
>>>Cullen dials in to conf from cell phone and is announced. 
>>
>>Cullen then dials
>>
>>>in from desk phone and is *not* announced because the user 
>>
>>Cullen is already
>>
>>>in the conference. Cullen hangs up cell phone and this is 
>>
>>*not* announced
>>
>>>because Cullen is still in the conference. Cullen hangs up 
>>
>>his desk phone
>>
>>>and the conference announces that "Fluffy has left the 
>>
>>building" because he
>>
>>>is no longer in the conference.
>>>
>>>
>>>On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com" 
>>
>><hisham.khartabil@nokia.com>
>>
>>>wrote:
>>>
>>>
>>>
>>>>Why do you need the focus to know that it is the same user 
>>>
>>(human)? I don't
>>
>>>>understand the use case. Can you elaborate?
>>>>
>>>>Thanks,
>>>>Hisham
>>>>
>>>>
>>>>
>>>>>-----Original Message-----
>>>>>From: sipping-admin@ietf.org
>>>>>[mailto:sipping-admin@ietf.org]On Behalf Of
>>>>>ext Cullen Jennings
>>>>>Sent: 08.April.2004 03:30
>>>>>To: 'sipping@ietf.org'
>>>>>Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
>>>>>Subject: [Sipping] User with multiple sessions in 
>>>>
>>Conference Package
>>
>>>>>
>>>>>
>>>>>In a conference, here are many cases where a single user may
>>>>>have set up
>>>>>multiple SIP dialogs with the focus. For example, using one
>>>>>UA instance for
>>>>>audio and another UA instance for IM. Or connected on both
>>>>>cell phone and on
>>>>>desk phone in office. Right now the conference package 
>>>>
>>seems a little
>>
>>>>>underspecified in how to deal with this and we need to 
>>>>
>>clear this up.
>>
>>>>>I'm tossing out two proposal below. Comments? Ideas? Can we
>>>>>choose one?
>>>>>Could be  close something like this on this list?
>>>>>
>>>>>--------------------------------------------------------------
>>>>>
>>>>>Proposal 1: 
>>>>>
>>>>>The User Element gets a new attribute called contact. This 
>>>>
>>attribute
>>
>>>>>contains the contact that the focus used for this dialog. 
>>>>
>>Each dialog
>>
>>>>>results in a separate User Element instance and the uri is
>>>>>used to correlate
>>>>>the fact that they are the same user. For anonymous calls the
>>>>>system needs
>>>>>to generate separate anonymous user uri's so they can be
>>>>>separated from one
>>>>>another. 
>>>>>
>>>>>For Example,
>>>>>
>>>>><conference-info>
>>>>>   <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
>>>>>       <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>>   </user>    
>>>>>   <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
>>>>>       <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>>   </user>
>>>>></conference-info>
>>>>>
>>>>>---------------------------------------------------------------
>>>>>
>>>>>Proposal 2: 
>>>>>
>>>>>The data in the User Element gets split between a User
>>>>>Element and a new
>>>>>Session Element that is placed inside the User Element. The
>>>>>media stream
>>>>>elements are put inside the Session.
>>>>>
>>>>><conference-info>
>>>>>   <user uri="fluffy@cisco.com">
>>>>>       <session contact="1111@focus.cisco.com">
>>>>>           <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>>       </session>
>>>>>       <session contact="2222@focus.cisco.com">
>>>>>           <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>>       </session>
>>>>>   </user>
>>>>></conference-info>
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Sipping mailing list  
>>>>
> https://www1.ietf.org/mailman/listinfo/sipping
> 
>>>>This list is for NEW development of the application of SIP
>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>Use sip@ietf.org for new developments of core SIP
>>>>
>>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 


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



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> -----Original Message-----
> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 14.April.2004 16:23
> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
> Cc: fluffy@cisco.com; sipping@ietf.org; oritl@microsoft.com;
> hgs@cs.columbia.edu; jdrosen@dynamicsoft.com
> Subject: Re: [Sipping] User with multiple sessions in=20
> Conference Package
>=20
>=20
>=20
>=20
> hisham.khartabil@nokia.com wrote:
> > It could based on the asserted identity (unless it is a=20
> public conference and no authentication is needed). By=20
> asserted identity I mean any authentication method available=20
> to the conference server/focus.
>=20
> I wonder how often the endpoints will be authenticated. In=20
> most of the=20
> conference servers I am familiar with today, authentication is simply=20
> based on knowing the conference id.

In this case, I don't think it is possible to achieve what Cullen wants. =
What I had in mind was that either the network hosting the conference =
server authenticates the user and asserts him/her to the conference =
server or the conference server itself authenticates the USER.

>=20
> And it probably won't be likely that all the devices a user=20
> employs will=20
> be bound to the the user in a way known to the conference=20
> server. (E.g.=20
> I am calling in from the phone tel:+1-978-555-1234, which may just be=20
> some phone in a conference room, as well as the IM client=20
> sip:pkyzivat@cisco.com. How does the phone get associated with me?)

This does not affect the authentication of the user/caller in the manner =
I described above.

Hisham

>=20
> 	Paul
>=20
> >>-----Original Message-----
> >>From: sipping-admin@ietf.org=20
> >>[mailto:sipping-admin@ietf.org]On Behalf Of
> >>ext Paul Kyzivat
> >>Sent: 13.April.2004 16:57
> >>To: Cullen Jennings
> >>Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org; Orit
> >>Levin; Henning Schulzrinne; Jonathan Rosenberg
> >>Subject: Re: [Sipping] User with multiple sessions in=20
> >>Conference Package
> >>
> >>
> >>Cullen,
> >>
> >>I think what you are trying to achieve is a good thing. But I=20
> >>think the=20
> >>focus may need to be psychic to achieve it. How is it=20
> >>supposed to know=20
> >>which dialogs to group together? I guess it could group all=20
> >>the things=20
> >>that share the same From address, but that won't get all=20
> the cases -=20
> >>probably not the interesting ones.
> >>
> >>	Paul
> >>
> >>Cullen Jennings wrote:
> >>
> >>>I'll take a stab at explaining :-)
> >>>
> >>>If someone has a roster displayed on some device - they=20
> >>
> >>want to see that one
> >>
> >>>person, Cullen is in the conference they don't want to sees=20
> >>
> >>that their are 4
> >>
> >>>participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
> >>>Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More=20
> >>
> >>importantly, they want to
> >>
> >>>be able to use a user interface to boot Cullen out of the=20
> >>
> >>conference not
> >>
> >>>have to go and somehow magically figure out which devices=20
> >>
> >>to boot out since
> >>
> >>>they are being used by Cullen.
> >>>
> >>>If their are 10 participants that are all Anonymous, you=20
> >>
> >>still want to be
> >>
> >>>able to know their are 10 participants not 1 participant=20
> >>
> >>called Anonymous
> >>
> >>>that seems to have dialed in from 10 devices.
> >>>
> >>>Given that SIP AOR correspond to a user and is not a single=20
> >>
> >>device. There
> >>
> >>>are basically three levels off aggregation happening here.
> >>>
> >>>There are users. Each user can have multiple SIP dialogs=20
> >>
> >>that got them into
> >>
> >>>the conference. Each one of these SIP dialogs can be=20
> >>
> >>started and terminated
> >>
> >>>independently. For example one may be dialed out while=20
> >>
> >>another dialed in and
> >>
> >>>one may be active while others may be ongoing. Inside of=20
> >>
> >>each SIP dialog,
> >>
> >>>there can be multiple media stream (audio, video, etc). The=20
> >>
> >>three levels are
> >>
> >>>Users, SIP Sessions or Dialogs, Media Sessions.=20
> >>
> >>Unfortunately, the draft as
> >>
> >>>written supports two things and the User level information=20
> >>
> >>and the SIP
> >>
> >>>session level information is mixed together. I think this=20
> >>
> >>needs to get
> >>
> >>>fixed.=20
> >>>
> >>>One more use case, many audio conference systems announce=20
> >>
> >>when a user joins
> >>
> >>>of leaves. This is really annoying when someone transfers=20
> >>
> >>from the cell
> >>
> >>>phone to desk phone. Let me give you a scenario that I=20
> >>
> >>think XCON should
> >>
> >>>support.=20
> >>>
> >>>Cullen dials in to conf from cell phone and is announced.=20
> >>
> >>Cullen then dials
> >>
> >>>in from desk phone and is *not* announced because the user=20
> >>
> >>Cullen is already
> >>
> >>>in the conference. Cullen hangs up cell phone and this is=20
> >>
> >>*not* announced
> >>
> >>>because Cullen is still in the conference. Cullen hangs up=20
> >>
> >>his desk phone
> >>
> >>>and the conference announces that "Fluffy has left the=20
> >>
> >>building" because he
> >>
> >>>is no longer in the conference.
> >>>
> >>>
> >>>On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com"=20
> >>
> >><hisham.khartabil@nokia.com>
> >>
> >>>wrote:
> >>>
> >>>
> >>>
> >>>>Why do you need the focus to know that it is the same user=20
> >>>
> >>(human)? I don't
> >>
> >>>>understand the use case. Can you elaborate?
> >>>>
> >>>>Thanks,
> >>>>Hisham
> >>>>
> >>>>
> >>>>
> >>>>>-----Original Message-----
> >>>>>From: sipping-admin@ietf.org
> >>>>>[mailto:sipping-admin@ietf.org]On Behalf Of
> >>>>>ext Cullen Jennings
> >>>>>Sent: 08.April.2004 03:30
> >>>>>To: 'sipping@ietf.org'
> >>>>>Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
> >>>>>Subject: [Sipping] User with multiple sessions in=20
> >>>>
> >>Conference Package
> >>
> >>>>>
> >>>>>
> >>>>>In a conference, here are many cases where a single user may
> >>>>>have set up
> >>>>>multiple SIP dialogs with the focus. For example, using one
> >>>>>UA instance for
> >>>>>audio and another UA instance for IM. Or connected on both
> >>>>>cell phone and on
> >>>>>desk phone in office. Right now the conference package=20
> >>>>
> >>seems a little
> >>
> >>>>>underspecified in how to deal with this and we need to=20
> >>>>
> >>clear this up.
> >>
> >>>>>I'm tossing out two proposal below. Comments? Ideas? Can we
> >>>>>choose one?
> >>>>>Could be  close something like this on this list?
> >>>>>
> >>>>>--------------------------------------------------------------
> >>>>>
> >>>>>Proposal 1:=20
> >>>>>
> >>>>>The User Element gets a new attribute called contact. This=20
> >>>>
> >>attribute
> >>
> >>>>>contains the contact that the focus used for this dialog.=20
> >>>>
> >>Each dialog
> >>
> >>>>>results in a separate User Element instance and the uri is
> >>>>>used to correlate
> >>>>>the fact that they are the same user. For anonymous calls the
> >>>>>system needs
> >>>>>to generate separate anonymous user uri's so they can be
> >>>>>separated from one
> >>>>>another.=20
> >>>>>
> >>>>>For Example,
> >>>>>
> >>>>><conference-info>
> >>>>>   <user uri=3D"fluffy@cisco.com" =
contact=3D"1111@focus.cisco.com">
> >>>>>       <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>>>>   </user>   =20
> >>>>>   <user uri=3D"fluffy@cisco.com" =
contact=3D"2222@focus.cisco.com">
> >>>>>       <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>>>>   </user>
> >>>>></conference-info>
> >>>>>
> >>>>>---------------------------------------------------------------
> >>>>>
> >>>>>Proposal 2:=20
> >>>>>
> >>>>>The data in the User Element gets split between a User
> >>>>>Element and a new
> >>>>>Session Element that is placed inside the User Element. The
> >>>>>media stream
> >>>>>elements are put inside the Session.
> >>>>>
> >>>>><conference-info>
> >>>>>   <user uri=3D"fluffy@cisco.com">
> >>>>>       <session contact=3D"1111@focus.cisco.com">
> >>>>>           <media-stream> <ssrc> 1234 </ssrc> </media-stream>
> >>>>>       </session>
> >>>>>       <session contact=3D"2222@focus.cisco.com">
> >>>>>           <media-stream> <ssrc> 5678 </ssrc> </media-stream>
> >>>>>       </session>
> >>>>>   </user>
> >>>>></conference-info>
> >>>>>
> >>>>>
> >>>>>_______________________________________________
> >>>>>Sipping mailing list =20
> >>>>
> > https://www1.ietf.org/mailman/listinfo/sipping
> >=20
> >>>>This list is for NEW development of the application of SIP
> >>>>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>>>Use sip@ietf.org for new developments of core SIP
> >>>>
> >>>
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sip@ietf.org for new developments of core SIP
> >>
> >=20
> >=20
> >=20
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >=20
>=20
>=20

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



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Subject: RE: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
Date: Thu, 15 Apr 2004 13:53:38 +0300
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Thread-Topic: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
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I think this paragraph can be entirely removed.

/Hisham

> -----Original Message-----
> From: sipping-admin@ietf.org=20
> [mailto:sipping-admin@ietf.org]On Behalf Of
> ext Paul Kyzivat
> Sent: 10.April.2004 01:06
> To: sipping@ietf.org
> Subject: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
>=20
>=20
> Section 4 says:
>=20
>     As for any other SIP request, the size of requests=20
> carrying URI lists
>     MUST NOT exceed 1300 bytes, unless the user agent client=20
> has positive
>     knowledge that the message will not traverse a=20
> congestion-unsafe link
>     at any hop, or that the message size is at least 200=20
> bytes less than
>     the lowest MTU value found en route to the server.
>=20
> I was curious about the implications of this, so I did some=20
> counting. I=20
> took the example in section 9 and stripped out all the=20
> leading blanks to=20
> get a plausible size estimate. The number I got was 1242,=20
> with four urls=20
> in the list. Each additional URL (of the size in the example) adds 37=20
> bytes. So a list with more than 5 urls exceeds the 1300 byte limit.
>=20
> That says to me that this technique is pretty much useless except in=20
> carefully controlled networks, or when compression is used.=20
> Seems pretty=20
> limiting. So it seems likely that in practice content=20
> indirection will=20
> have to be used.
>=20
> 	Paul
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>=20

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



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Date: Thu, 15 Apr 2004 09:23:39 -0700
Subject: Re: [Sipping] User with multiple sessions in Conference Package
From: Cullen Jennings <fluffy@cisco.com>
To: <hisham.khartabil@nokia.com>, Paul H Kyzivat <pkyzivat@cisco.com>
CC: <sipping@ietf.org>
Message-ID: <BCA4051B.39C97%fluffy@cisco.com>
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I'm very happy with the only correlation to know that two SIP dialogs are
the same user is the definition of the "user" uri in the conference package
today. If I authenticated as some particular user to the conference system
this will work, if not it is not useful. There are several ways this might
of happened:

1) my strong sip identity information (ala IAB, S/MIME, etc)
2) my weak sip identity information (whatever I put in my From header)
3) I entered a PIN code to an IVR system
4) the uri of the conference was specific for me and identifies me

If people are not willing to provide any information correlate or
uncorrelated two different devices then sure this won't work.

It's true that many service provider hosted conference systems do not
separately identify each participant. However it is the opposite for
enterprise based conferencing systems. I have an account on the conference
system I use at work. When I dial in from my office phone it recognizes it
is me and log me in. When I dial in from some other phone, I enter a user id
and pin. It saves all my preferences, what language I want, what time zone I
am in, and does not make me record my name for ever conference. If I plan to
use a particular conference room, I could add that conference room as a
participant to the meeting and it's phone gets placed in the right
conference when it dials in.

Cullen



On 4/15/04 1:43 AM, "hisham.khartabil@nokia.com"
<hisham.khartabil@nokia.com> wrote:

> 
> 
>> -----Original Message-----
>> From: ext Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: 14.April.2004 16:23
>> To: Khartabil Hisham (Nokia-TP-MSW/Helsinki)
>> Cc: fluffy@cisco.com; sipping@ietf.org; oritl@microsoft.com;
>> hgs@cs.columbia.edu; jdrosen@dynamicsoft.com
>> Subject: Re: [Sipping] User with multiple sessions in
>> Conference Package
>> 
>> 
>> 
>> 
>> hisham.khartabil@nokia.com wrote:
>>> It could based on the asserted identity (unless it is a
>> public conference and no authentication is needed). By
>> asserted identity I mean any authentication method available
>> to the conference server/focus.
>> 
>> I wonder how often the endpoints will be authenticated. In
>> most of the 
>> conference servers I am familiar with today, authentication is simply
>> based on knowing the conference id.
> 
> In this case, I don't think it is possible to achieve what Cullen wants. What
> I had in mind was that either the network hosting the conference server
> authenticates the user and asserts him/her to the conference server or the
> conference server itself authenticates the USER.
> 
>> 
>> And it probably won't be likely that all the devices a user
>> employs will 
>> be bound to the the user in a way known to the conference
>> server. (E.g. 
>> I am calling in from the phone tel:+1-978-555-1234, which may just be
>> some phone in a conference room, as well as the IM client
>> sip:pkyzivat@cisco.com. How does the phone get associated with me?)
> 
> This does not affect the authentication of the user/caller in the manner I
> described above.
> 
> Hisham
> 
>> 
>> Paul
>> 
>>>> -----Original Message-----
>>>> From: sipping-admin@ietf.org
>>>> [mailto:sipping-admin@ietf.org]On Behalf Of
>>>> ext Paul Kyzivat
>>>> Sent: 13.April.2004 16:57
>>>> To: Cullen Jennings
>>>> Cc: Khartabil Hisham (Nokia-TP-MSW/Helsinki); sipping@ietf.org; Orit
>>>> Levin; Henning Schulzrinne; Jonathan Rosenberg
>>>> Subject: Re: [Sipping] User with multiple sessions in
>>>> Conference Package
>>>> 
>>>> 
>>>> Cullen,
>>>> 
>>>> I think what you are trying to achieve is a good thing. But I
>>>> think the 
>>>> focus may need to be psychic to achieve it. How is it
>>>> supposed to know
>>>> which dialogs to group together? I guess it could group all
>>>> the things 
>>>> that share the same From address, but that won't get all
>> the cases - 
>>>> probably not the interesting ones.
>>>> 
>>>> Paul
>>>> 
>>>> Cullen Jennings wrote:
>>>> 
>>>>> I'll take a stab at explaining :-)
>>>>> 
>>>>> If someone has a roster displayed on some device - they
>>>> 
>>>> want to see that one
>>>> 
>>>>> person, Cullen is in the conference they don't want to sees
>>>> 
>>>> that their are 4
>>>> 
>>>>> participants, Cullen-Cell-Phone, Cullen-Speaker-Phone,
>>>>> Cullen-JABBER_GATEWAY, Cullen-VIDEO-ICHAT_GW. More
>>>> 
>>>> importantly, they want to
>>>> 
>>>>> be able to use a user interface to boot Cullen out of the
>>>> 
>>>> conference not
>>>> 
>>>>> have to go and somehow magically figure out which devices
>>>> 
>>>> to boot out since
>>>> 
>>>>> they are being used by Cullen.
>>>>> 
>>>>> If their are 10 participants that are all Anonymous, you
>>>> 
>>>> still want to be
>>>> 
>>>>> able to know their are 10 participants not 1 participant
>>>> 
>>>> called Anonymous
>>>> 
>>>>> that seems to have dialed in from 10 devices.
>>>>> 
>>>>> Given that SIP AOR correspond to a user and is not a single
>>>> 
>>>> device. There
>>>> 
>>>>> are basically three levels off aggregation happening here.
>>>>> 
>>>>> There are users. Each user can have multiple SIP dialogs
>>>> 
>>>> that got them into
>>>> 
>>>>> the conference. Each one of these SIP dialogs can be
>>>> 
>>>> started and terminated
>>>> 
>>>>> independently. For example one may be dialed out while
>>>> 
>>>> another dialed in and
>>>> 
>>>>> one may be active while others may be ongoing. Inside of
>>>> 
>>>> each SIP dialog,
>>>> 
>>>>> there can be multiple media stream (audio, video, etc). The
>>>> 
>>>> three levels are
>>>> 
>>>>> Users, SIP Sessions or Dialogs, Media Sessions.
>>>> 
>>>> Unfortunately, the draft as
>>>> 
>>>>> written supports two things and the User level information
>>>> 
>>>> and the SIP
>>>> 
>>>>> session level information is mixed together. I think this
>>>> 
>>>> needs to get
>>>> 
>>>>> fixed. 
>>>>> 
>>>>> One more use case, many audio conference systems announce
>>>> 
>>>> when a user joins
>>>> 
>>>>> of leaves. This is really annoying when someone transfers
>>>> 
>>>> from the cell
>>>> 
>>>>> phone to desk phone. Let me give you a scenario that I
>>>> 
>>>> think XCON should
>>>> 
>>>>> support. 
>>>>> 
>>>>> Cullen dials in to conf from cell phone and is announced.
>>>> 
>>>> Cullen then dials
>>>> 
>>>>> in from desk phone and is *not* announced because the user
>>>> 
>>>> Cullen is already
>>>> 
>>>>> in the conference. Cullen hangs up cell phone and this is
>>>> 
>>>> *not* announced
>>>> 
>>>>> because Cullen is still in the conference. Cullen hangs up
>>>> 
>>>> his desk phone
>>>> 
>>>>> and the conference announces that "Fluffy has left the
>>>> 
>>>> building" because he
>>>> 
>>>>> is no longer in the conference.
>>>>> 
>>>>> 
>>>>> On 4/8/04 5:11 AM, "hisham.khartabil@nokia.com"
>>>> 
>>>> <hisham.khartabil@nokia.com>
>>>> 
>>>>> wrote:
>>>>> 
>>>>> 
>>>>> 
>>>>>> Why do you need the focus to know that it is the same user
>>>>> 
>>>> (human)? I don't
>>>> 
>>>>>> understand the use case. Can you elaborate?
>>>>>> 
>>>>>> Thanks,
>>>>>> Hisham
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>>> -----Original Message-----
>>>>>>> From: sipping-admin@ietf.org
>>>>>>> [mailto:sipping-admin@ietf.org]On Behalf Of
>>>>>>> ext Cullen Jennings
>>>>>>> Sent: 08.April.2004 03:30
>>>>>>> To: 'sipping@ietf.org'
>>>>>>> Cc: Orit Levin; Henning Schulzrinne; Jonathan Rosenberg
>>>>>>> Subject: [Sipping] User with multiple sessions in
>>>>>> 
>>>> Conference Package
>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> In a conference, here are many cases where a single user may
>>>>>>> have set up
>>>>>>> multiple SIP dialogs with the focus. For example, using one
>>>>>>> UA instance for
>>>>>>> audio and another UA instance for IM. Or connected on both
>>>>>>> cell phone and on
>>>>>>> desk phone in office. Right now the conference package
>>>>>> 
>>>> seems a little
>>>> 
>>>>>>> underspecified in how to deal with this and we need to
>>>>>> 
>>>> clear this up.
>>>> 
>>>>>>> I'm tossing out two proposal below. Comments? Ideas? Can we
>>>>>>> choose one?
>>>>>>> Could be  close something like this on this list?
>>>>>>> 
>>>>>>> --------------------------------------------------------------
>>>>>>> 
>>>>>>> Proposal 1:
>>>>>>> 
>>>>>>> The User Element gets a new attribute called contact. This
>>>>>> 
>>>> attribute
>>>> 
>>>>>>> contains the contact that the focus used for this dialog.
>>>>>> 
>>>> Each dialog
>>>> 
>>>>>>> results in a separate User Element instance and the uri is
>>>>>>> used to correlate
>>>>>>> the fact that they are the same user. For anonymous calls the
>>>>>>> system needs
>>>>>>> to generate separate anonymous user uri's so they can be
>>>>>>> separated from one
>>>>>>> another. 
>>>>>>> 
>>>>>>> For Example,
>>>>>>> 
>>>>>>> <conference-info>
>>>>>>>   <user uri="fluffy@cisco.com" contact="1111@focus.cisco.com">
>>>>>>>       <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>>>>   </user>  
>>>>>>>   <user uri="fluffy@cisco.com" contact="2222@focus.cisco.com">
>>>>>>>       <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>>>>   </user>
>>>>>>> </conference-info>
>>>>>>> 
>>>>>>> ---------------------------------------------------------------
>>>>>>> 
>>>>>>> Proposal 2:
>>>>>>> 
>>>>>>> The data in the User Element gets split between a User
>>>>>>> Element and a new
>>>>>>> Session Element that is placed inside the User Element. The
>>>>>>> media stream
>>>>>>> elements are put inside the Session.
>>>>>>> 
>>>>>>> <conference-info>
>>>>>>>   <user uri="fluffy@cisco.com">
>>>>>>>       <session contact="1111@focus.cisco.com">
>>>>>>>           <media-stream> <ssrc> 1234 </ssrc> </media-stream>
>>>>>>>       </session>
>>>>>>>       <session contact="2222@focus.cisco.com">
>>>>>>>           <media-stream> <ssrc> 5678 </ssrc> </media-stream>
>>>>>>>       </session>
>>>>>>>   </user>
>>>>>>> </conference-info>
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> Sipping mailing list
>>>>>> 
>>> https://www1.ietf.org/mailman/listinfo/sipping
>>> 
>>>>>> This list is for NEW development of the application of SIP
>>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>> Use sip@ietf.org for new developments of core SIP
>>>>>> 
>>>>> 
>>>> 
>>>> _______________________________________________
>>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>> 
>> 
>> 
> 


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From: Cullen Jennings <fluffy@cisco.com>
To: Andy Spitzer <woof@snowshore.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, <jvandyke@snowshore.com>,
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I'm going to try and take a second take at this this stuff.

Things I think we agree on ...

I assume we agree there will be multiple MS and multiple AS and we will use
a proxy to broker connections so that we only require the proxy to act as a
TLS server instead of every device being a TLS server.

There are conference names that link to a particular focus. They route to
the AS. We are not talking about conference names. This has nothing to do
with conference names.

The AS is trying to invoke a service (such as a mixer) on the DSP farm. We
are talking about the name of that service. There may be many mixers
involved with a single conference. I need a name that is routable to a
particular mixer via the proxy. The AS will pick which mixer it wants to
use. 

Where we disagree ....

That you can use 3rd party registration to tell a proxy to send invites down
some particular TLS connection. I don't think you can.

That my proxy needs to keep a binding for every instance of every service.
I'm fine with one biding per media service but not per instance of the
service. I think it is lots of state that is just not needed in a proxy.

That a mixer service is somehow different than other services. I don't see
how it is. An IVR system also has to run on a particular box. It may the
only box that has the voice prompts needed.

My proposal .... 

Lots of people will build systems differently. We should try and support
that. I want to be able to have a GRUU for the contact of a particular
service so I can reach it via a proxy over TLS. Others may want to hand
configure routes in the proxy. I don't care. However, the parameters for
service invocations should not be in the user part because that takes away
the ability to configure the name. An example of this is not being able to
use a GRU. There are other examples.

What I think people should do ....

Is there some reason that this mixer service is somehow different and needs
to have it parameters in the user part.  I just see why - help explain this.
Also help explain how we are going to make this 3rd party registration work.








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On 4/15/04 3:38 PM, "Andy Spitzer" <woof@brooktrout.com> wrote:

> Ahh, but for IVR, each and every CALL to the service is independant of
> every other call.  If call A went to MS 1, and call B went to MS 2, both
> would work just fine, they do not need to know about each other; they are
> independentof each other.

I don't think so. The AS needs to be able to say if it wants MS1 or MS2 even
in the IVR case. Perhaps it knows that one of them is busy. Perhaps one of
them does ASR and the other does not. I agree there are case were you media
servers are interchangeable but there are also cases where they are not and
the AS needs to be able to make the choice about that.

I agree with you that you for the mixer, you need to the the right one. I
just think that is true of all services.

I will make an incredible fluffy analogy here ... You are claiming that
services invocations are like AOR URI's. I don't think they are. I think
they are more like contact URIs. Yes the name of the meeting that gets
routed to the focus may be more like an AOR but we are talking about the
mixer name.
 



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Subject: Re: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
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Hi,

yes, I had already got this comment before, and I had logged a bug 
against it. The next revision of the draft will not have this paragraph.

Thanks,

Gonzalo

hisham.khartabil@nokia.com wrote:

> I think this paragraph can be entirely removed.
> 
> /Hisham
> 
> 
>>-----Original Message-----
>>From: sipping-admin@ietf.org 
>>[mailto:sipping-admin@ietf.org]On Behalf Of
>>ext Paul Kyzivat
>>Sent: 10.April.2004 01:06
>>To: sipping@ietf.org
>>Subject: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
>>
>>
>>Section 4 says:
>>
>>    As for any other SIP request, the size of requests 
>>carrying URI lists
>>    MUST NOT exceed 1300 bytes, unless the user agent client 
>>has positive
>>    knowledge that the message will not traverse a 
>>congestion-unsafe link
>>    at any hop, or that the message size is at least 200 
>>bytes less than
>>    the lowest MTU value found en route to the server.
>>
>>I was curious about the implications of this, so I did some 
>>counting. I 
>>took the example in section 9 and stripped out all the 
>>leading blanks to 
>>get a plausible size estimate. The number I got was 1242, 
>>with four urls 
>>in the list. Each additional URL (of the size in the example) adds 37 
>>bytes. So a list with more than 5 urls exceeds the 1300 byte limit.
>>
>>That says to me that this technique is pretty much useless except in 
>>carefully controlled networks, or when compression is used. 
>>Seems pretty 
>>limiting. So it seems likely that in practice content 
>>indirection will 
>>have to be used.
>>
>>	Paul
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>>
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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



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Date: Fri, 16 Apr 2004 10:11:34 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
CC: hisham.khartabil@nokia.com, sipping@ietf.org
Subject: Re: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
References: <2038BCC78B1AD641891A0D1AE133DBB701797908@esebe019.ntc.nokia.com> <407F8265.6070605@ericsson.com>
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Why can this be removed? Isn't it still a real issue?

Or is this assumed to be fixed by the 3261 requirement that large 
messages be delivered over TCP?

	Paul

Gonzalo Camarillo wrote:
> Hi,
> 
> yes, I had already got this comment before, and I had logged a bug 
> against it. The next revision of the draft will not have this paragraph.
> 
> Thanks,
> 
> Gonzalo
> 
> hisham.khartabil@nokia.com wrote:
> 
>> I think this paragraph can be entirely removed.
>>
>> /Hisham
>>
>>
>>> -----Original Message-----
>>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
>>> ext Paul Kyzivat
>>> Sent: 10.April.2004 01:06
>>> To: sipping@ietf.org
>>> Subject: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
>>>
>>>
>>> Section 4 says:
>>>
>>>    As for any other SIP request, the size of requests carrying URI lists
>>>    MUST NOT exceed 1300 bytes, unless the user agent client has positive
>>>    knowledge that the message will not traverse a congestion-unsafe link
>>>    at any hop, or that the message size is at least 200 bytes less than
>>>    the lowest MTU value found en route to the server.
>>>
>>> I was curious about the implications of this, so I did some counting. 
>>> I took the example in section 9 and stripped out all the leading 
>>> blanks to get a plausible size estimate. The number I got was 1242, 
>>> with four urls in the list. Each additional URL (of the size in the 
>>> example) adds 37 bytes. So a list with more than 5 urls exceeds the 
>>> 1300 byte limit.
>>>
>>> That says to me that this technique is pretty much useless except in 
>>> carefully controlled networks, or when compression is used. Seems 
>>> pretty limiting. So it seems likely that in practice content 
>>> indirection will have to be used.
>>>
>>>     Paul
>>>
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>>
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>>
> 


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



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Message-ID: <C90B77EFE2FAF6459B42E1C686F6E2CB03ABA3B3@MSTELEFONICA8>
From: De Leo Walter <deleow@TELEFONICA.COM.AR>
To: "'Cullen Jennings'" <fluffy@cisco.com>
Cc: "'sipping@ietf.org'" <sipping@ietf.org>
Subject: [Sipping] Use of P-Asserted
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Cullen:

I have one concern about a use of a P-Asserted Identity header that some
folks are using. It is, from my point of view out of the scope of RFC 3325,
but I think it is a reasonable use.  The use I am referring to is the
following:

Calls initiated by UACs inside the trust domain include the P-Asserted
Identity (they are part of the network who asserts the identity) header in
their own INVITEs.

I refer specifically to this scenario:

Suppose a SIP an ingress Media Gateway Controller, a B2BUA Service Platform
and an egress MGC.
Calls initiated (from the SIP viewpoint) by ingress MGC or B2BUA carry the
P-Asserted Identity header inserted by themselves. The egress Media Gateway
Controller maps allways the content of the P-Assertd Identity header to the
ANI field in corresponding ISUP message and the presentation restriction
information in the Privacy Header to the corresponding ISUP field.

So the egress MGC has always the same policy, INVITES which don't have a
P-Asserted Identity header are not trusted, and if they have one either it
comes from a UAC authenticated by a proxy in the Trust domain or it comes
from inside the Trust domain itself and are, hence, trusted. (I am asuming
the communication inside the trust domain is safe).

Do you see any problem in this use of P-Asserted Identity header.

I write to you because you are co-author of RFC 3325. I hope not having
disturbed you with this question. I copy the sipping list just in case
someone else has seen this use of P-Asserted Identity header and has
something to say.

Best Regards.

Walter D. De Leo



------_=_NextPart_001_01C423C0.88EF8CB0
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>[Sipping] Use of P-Asserted</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Cullen:</FONT>
</P>

<P><FONT SIZE=3D2>I have one concern about a use of a P-Asserted =
Identity header that some folks are using. It is, from my point of view =
out of the scope of RFC 3325, but I think it is a reasonable use.&nbsp; =
The use I am referring to is the following:</FONT></P>

<P><FONT SIZE=3D2>Calls initiated by UACs inside the trust domain =
include the P-Asserted Identity (they are part of the network who =
asserts the identity) header in their own INVITEs.</FONT></P>

<P><FONT SIZE=3D2>I refer specifically to this scenario:</FONT>
</P>

<P><FONT SIZE=3D2>Suppose a SIP an ingress Media Gateway Controller, a =
B2BUA Service Platform and an egress MGC.</FONT>
<BR><FONT SIZE=3D2>Calls initiated (from the SIP viewpoint) by ingress =
MGC or B2BUA carry the P-Asserted Identity header inserted by =
themselves. The egress Media Gateway Controller maps allways the =
content of the P-Assertd Identity header to the ANI field in =
corresponding ISUP message and the presentation restriction information =
in the Privacy Header to the corresponding ISUP field.</FONT></P>

<P><FONT SIZE=3D2>So the egress MGC has always the same policy, INVITES =
which don't have a P-Asserted Identity header are not trusted, and if =
they have one either it comes from a UAC authenticated by a proxy in =
the Trust domain or it comes from inside the Trust domain itself and =
are, hence, trusted. (I am asuming the communication inside the trust =
domain is safe).</FONT></P>

<P><FONT SIZE=3D2>Do you see any problem in this use of P-Asserted =
Identity header.</FONT>
</P>

<P><FONT SIZE=3D2>I write to you because you are co-author of RFC 3325. =
I hope not having disturbed you with this question. I copy the sipping =
list just in case someone else has seen this use of P-Asserted Identity =
header and has something to say.</FONT></P>

<P><FONT SIZE=3D2>Best Regards.</FONT>
</P>

<P><FONT SIZE=3D2>Walter D. De Leo</FONT>
</P>
<BR>

</BODY>
</HTML>
------_=_NextPart_001_01C423C0.88EF8CB0--

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



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Date: Fri, 16 Apr 2004 10:16:33 -0500
From: Alex Audu <alex.audu@alcatel.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
CC: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>,
        hisham.khartabil@nokia.com, sipping@ietf.org
Subject: Re: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
References: <2038BCC78B1AD641891A0D1AE133DBB701797908@esebe019.ntc.nokia.com> <407F8265.6070605@ericsson.com> <407FE996.6000206@cisco.com>
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UDP induced message size limitation should be transparent to the user.  
3261 has
already specified proper handling for large messages. I think that 
should be enough
to allow the client side deal with the problem transparently.

Regards,
Alex.

Paul Kyzivat wrote:

> Why can this be removed? Isn't it still a real issue?
>
> Or is this assumed to be fixed by the 3261 requirement that large 
> messages be delivered over TCP?
>
>     Paul
>
> Gonzalo Camarillo wrote:
>
>> Hi,
>>
>> yes, I had already got this comment before, and I had logged a bug 
>> against it. The next revision of the draft will not have this paragraph.
>>
>> Thanks,
>>
>> Gonzalo
>>
>> hisham.khartabil@nokia.com wrote:
>>
>>> I think this paragraph can be entirely removed.
>>>
>>> /Hisham
>>>
>>>
>>>> -----Original Message-----
>>>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On 
>>>> Behalf Of
>>>> ext Paul Kyzivat
>>>> Sent: 10.April.2004 01:06
>>>> To: sipping@ietf.org
>>>> Subject: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
>>>>
>>>>
>>>> Section 4 says:
>>>>
>>>>    As for any other SIP request, the size of requests carrying URI 
>>>> lists
>>>>    MUST NOT exceed 1300 bytes, unless the user agent client has 
>>>> positive
>>>>    knowledge that the message will not traverse a congestion-unsafe 
>>>> link
>>>>    at any hop, or that the message size is at least 200 bytes less 
>>>> than
>>>>    the lowest MTU value found en route to the server.
>>>>
>>>> I was curious about the implications of this, so I did some 
>>>> counting. I took the example in section 9 and stripped out all the 
>>>> leading blanks to get a plausible size estimate. The number I got 
>>>> was 1242, with four urls in the list. Each additional URL (of the 
>>>> size in the example) adds 37 bytes. So a list with more than 5 urls 
>>>> exceeds the 1300 byte limit.
>>>>
>>>> That says to me that this technique is pretty much useless except 
>>>> in carefully controlled networks, or when compression is used. 
>>>> Seems pretty limiting. So it seems likely that in practice content 
>>>> indirection will have to be used.
>>>>
>>>>     Paul
>>>>
>>>>
>>>> _______________________________________________
>>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>>
>>>
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>>
>>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP



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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: alex.audu@alcatel.com
CC: Paul Kyzivat <pkyzivat@cisco.com>, hisham.khartabil@nokia.com,
        sipping@ietf.org
Subject: Re: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
References: <2038BCC78B1AD641891A0D1AE133DBB701797908@esebe019.ntc.nokia.com> <407F8265.6070605@ericsson.com> <407FE996.6000206@cisco.com> <407FF8D1.4020701@alcatel.com>
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Hi,

yes, the idea is to reference RFC 3261. That is, we will not define any 
special handling for large messages in the uri list draft.

Thanks,

Gonzalo

Alex Audu wrote:
> UDP induced message size limitation should be transparent to the user.  
> 3261 has
> already specified proper handling for large messages. I think that 
> should be enough
> to allow the client side deal with the problem transparently.
> 
> Regards,
> Alex.
> 
> Paul Kyzivat wrote:
> 
>> Why can this be removed? Isn't it still a real issue?
>>
>> Or is this assumed to be fixed by the 3261 requirement that large 
>> messages be delivered over TCP?
>>
>>     Paul
>>
>> Gonzalo Camarillo wrote:
>>
>>> Hi,
>>>
>>> yes, I had already got this comment before, and I had logged a bug 
>>> against it. The next revision of the draft will not have this paragraph.
>>>
>>> Thanks,
>>>
>>> Gonzalo
>>>
>>> hisham.khartabil@nokia.com wrote:
>>>
>>>> I think this paragraph can be entirely removed.
>>>>
>>>> /Hisham
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On 
>>>>> Behalf Of
>>>>> ext Paul Kyzivat
>>>>> Sent: 10.April.2004 01:06
>>>>> To: sipping@ietf.org
>>>>> Subject: [Sipping] Re:draft-camarillo-sipping-uri-list-02.txt
>>>>>
>>>>>
>>>>> Section 4 says:
>>>>>
>>>>>    As for any other SIP request, the size of requests carrying URI 
>>>>> lists
>>>>>    MUST NOT exceed 1300 bytes, unless the user agent client has 
>>>>> positive
>>>>>    knowledge that the message will not traverse a congestion-unsafe 
>>>>> link
>>>>>    at any hop, or that the message size is at least 200 bytes less 
>>>>> than
>>>>>    the lowest MTU value found en route to the server.
>>>>>
>>>>> I was curious about the implications of this, so I did some 
>>>>> counting. I took the example in section 9 and stripped out all the 
>>>>> leading blanks to get a plausible size estimate. The number I got 
>>>>> was 1242, with four urls in the list. Each additional URL (of the 
>>>>> size in the example) adds 37 bytes. So a list with more than 5 urls 
>>>>> exceeds the 1300 byte limit.
>>>>>
>>>>> That says to me that this technique is pretty much useless except 
>>>>> in carefully controlled networks, or when compression is used. 
>>>>> Seems pretty limiting. So it seems likely that in practice content 
>>>>> indirection will have to be used.
>>>>>
>>>>>     Paul
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>>> This list is for NEW development of the application of SIP
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sip@ietf.org for new developments of core SIP
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>>> This list is for NEW development of the application of SIP
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sip@ietf.org for new developments of core SIP
>>>>
>>>
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
> 
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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



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To: Cullen Jennings <fluffy@cisco.com>, Eric Burger <eburger@snowshore.com>,
        "'sipping@ietf.org'" <sipping@ietf.org>, jvandyke@snowshore.com,
        Alan Johnston <alan.johnston@mci.com>
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Subject: [Sipping] Re: Service name  stuff in natann draft
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Cullen,

On Thu, 15 Apr 2004 15:23:23 -0700, Cullen Jennings <fluffy@cisco.com> 
wrote:
> That a mixer service is somehow different than other services. I don't 
> see
> how it is. An IVR system also has to run on a particular box. It may the
> only box that has the voice prompts needed.

Ahh, but for IVR, each and every CALL to the service is independant of 
every other call.  If call A went to MS 1, and call B went to MS 2, both 
would work just fine, they do not need to know about each other; they are 
independentof each other.

But for mixing, this is not true.  Calls for each mix must find the media 
server hosting THAT mix.  If call A went to MS 1, and call B when to MS 2, 
they would not be talking to each other!  Each individual call is NOT 
indepentant of the other calls in that mix, thus they must "find" the 
correct MS that hosts that mix.

This is why we put the conference id in the user part...so that each 
conference mix can be assigned a unique name (user part), and calls for 
that "user" find the correct MS that hosts that particular mix.  Using the 
"normal" user@domain approach seemed like the right way to pin users (aka 
conference ids) to endpoints.

--Andy Spitzer
woof@brooktrout.com (new address)


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This list is for NEW development of the application of SIP
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Date: Sun, 18 Apr 2004 20:08:47 -0700
Subject: Re: [Sipping] Use of P-Asserted
From: Cullen Jennings <fluffy@cisco.com>
To: De Leo Walter <deleow@telefonica.com.ar>
CC: "'sipping@ietf.org'" <sipping@ietf.org>
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Not sure I fully understand but I think what you are saying sounds like a
reasonable way to use it.

By UAC=B9s in the trust domain I assume you mean GWs, Call Agents, and B2BUA,
that are trusted not IP Phones sitting on someone's desk.


On 4/16/04 7:38 AM, "De Leo Walter" <deleow@TELEFONICA.COM.AR> wrote:

> Cullen:=20
>=20
> I have one concern about a use of a P-Asserted Identity header that some =
folks
> are using. It is, from my point of view out of the scope of RFC 3325, but=
 I
> think it is a reasonable use.  The use I am referring to is the following=
:
>=20
> Calls initiated by UACs inside the trust domain include the P-Asserted
> Identity (they are part of the network who asserts the identity) header i=
n
> their own INVITEs.
>=20
> I refer specifically to this scenario:
>=20
> Suppose a SIP an ingress Media Gateway Controller, a B2BUA Service Platfo=
rm
> and an egress MGC.
> Calls initiated (from the SIP viewpoint) by ingress MGC or B2BUA carry th=
e
> P-Asserted Identity header inserted by themselves. The egress Media Gatew=
ay
> Controller maps allways the content of the P-Assertd Identity header to t=
he
> ANI field in corresponding ISUP message and the presentation restriction
> information in the Privacy Header to the corresponding ISUP field.
>=20
> So the egress MGC has always the same policy, INVITES which don't have a
> P-Asserted Identity header are not trusted, and if they have one either i=
t
> comes from a UAC authenticated by a proxy in the Trust domain or it comes=
 from
> inside the Trust domain itself and are, hence, trusted. (I am asuming the
> communication inside the trust domain is safe).
>=20
> Do you see any problem in this use of P-Asserted Identity header.
>=20
> I write to you because you are co-author of RFC 3325. I hope not having
> disturbed you with this question. I copy the sipping list just in case so=
meone
> else has seen this use of P-Asserted Identity header and has something to=
 say.
>=20
> Best Regards.=20
>=20
> Walter D. De Leo=20
>=20
>=20



--B_3165167932_2076171
Content-type: text/html; charset="ISO-8859-1"
Content-Transfer-Encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [Sipping] Use of P-Asserted</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0px'><BR>
Not sure I fully understand but I think what you are saying sounds like a r=
easonable way to use it.<BR>
<BR>
By UAC&#8217;s in the trust domain I assume you mean GWs, Call Agents, and =
B2BUA, that are trusted not IP Phones sitting on someone's desk. <BR>
<BR>
<BR>
On 4/16/04 7:38 AM, &quot;De Leo Walter&quot; &lt;deleow@TELEFONICA.COM.AR&=
gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0p=
x'>Cullen: <BR>
<BR>
I have one concern about a use of a P-Asserted Identity header that some fo=
lks are using. It is, from my point of view out of the scope of RFC 3325, bu=
t I think it is a reasonable use. &nbsp;The use I am referring to is the fol=
lowing:<BR>
<BR>
Calls initiated by UACs inside the trust domain include the P-Asserted Iden=
tity (they are part of the network who asserts the identity) header in their=
 own INVITEs.<BR>
<BR>
I refer specifically to this scenario: <BR>
<BR>
Suppose a SIP an ingress Media Gateway Controller, a B2BUA Service Platform=
 and an egress MGC. <BR>
Calls initiated (from the SIP viewpoint) by ingress MGC or B2BUA carry the =
P-Asserted Identity header inserted by themselves. The egress Media Gateway =
Controller maps allways the content of the P-Assertd Identity header to the =
ANI field in corresponding ISUP message and the presentation restriction inf=
ormation in the Privacy Header to the corresponding ISUP field.<BR>
<BR>
So the egress MGC has always the same policy, INVITES which don't have a P-=
Asserted Identity header are not trusted, and if they have one either it com=
es from a UAC authenticated by a proxy in the Trust domain or it comes from =
inside the Trust domain itself and are, hence, trusted. (I am asuming the co=
mmunication inside the trust domain is safe).<BR>
<BR>
Do you see any problem in this use of P-Asserted Identity header. <BR>
<BR>
I write to you because you are co-author of RFC 3325. I hope not having dis=
turbed you with this question. I copy the sipping list just in case someone =
else has seen this use of P-Asserted Identity header and has something to sa=
y.<BR>
<BR>
Best Regards. <BR>
<BR>
Walter D. De Leo <BR>
<BR>
<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Verdana"><SPAN STYLE=3D'font-size:14.0=
px'><BR>
</SPAN></FONT>
</BODY>
</HTML>


--B_3165167932_2076171--


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



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Thread-Topic: [Sip-implementors] B2BUA for testing..
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Look at: www.vovida.org you can find B2B implementation.


SReeram


-----Original Message-----
From: sip-implementors-bounces@cs.columbia.edu
[mailto:sip-implementors-bounces@cs.columbia.edu] On Behalf Of
badrisree@npd.hcltech.com
Sent: Monday, April 19, 2004 5:16 PM
To: sip-implementors@cs.columbia.edu
Cc: sipping@ietf.org
Subject: [Sip-implementors] B2BUA for testing..


Hi:

I am developing a test suite for testing a B2BUA implementation..Are
there any freely available B2BUA implementation available for this
purpose. Any pointers would be of great help.

Thanks,
Badri

Badri Srinivasan S
Networking Technology Labs,
HCL Technologies,
Plot No:66 South Phase
Ambattur Industrial Estate
Chennai-58
Phone:91 44 26521077, 26521270  Xtn: 2243
Fax 52060485

------------------------------------------------------------------------
----
-----------
"After much research, Polycom selected HCL's SIP protocol tester to
ensure our products deliver the carrier-grade reliability our
conferencing service providers demand. In addition, HCL's tools provide
the in-depth testing we need to speed the certification of our Voice
over IP solutions and guarantee interoperability," said Matt Taylor,
Staff Engineer for Polycom
------------------------------------------------------------------------
----
------------

visit us @ http://www.hcltechnologies.com/voip
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors

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



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Date: Mon, 19 Apr 2004 09:38:41 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: [Sipping] Re: [Sip-implementors] Call Transfer through  B2BUA..
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badrisree@npd.hcltech.com wrote:
> Hi All,
> 
> The 3pcc RFC 3725 depicts call flows that are best current practices for
> B2BUAs...
> In all the Call Flows the signalling initiates from the B2BUA i.e the INVITE
> is sent from the B2BUA that then puts the 2 UAs into a Call.
> But in the real life scenario like Call Transfer, the Call originates from
> an End Point A that then reaches a IP-PBX (B2BUA) that then tranfers the
> Call to End Point B. Pl. find below the flows equivalent to those given in
> the RFC 3725 where the Call originates from an End Point.

[snip]

> Flow 4 in RFC 3725 sends out an INVITE from the B2BUA without any media line
> in the Sdp.
> This flow is not possible for an INVITE that comes from an external enppoint
> that will always have an SDP with some media line.

It may be improbable, but it isn't impossible.

For instance, suppose A was yet another B2BUA that is itself carrying 
out flow 4.

> I am in the process of coming up with a test Suite for B2BUA. I am assumuing
> that a B2BUA that implements Call Transfer functionality will use any one of
> the above flows to achieve Call transfer. Am I correct in assuming this. 

A B2BUA is a very general concept. There are many uses for them. You 
cannot infer very much about the behavior of a B2BUA *in general*, so it 
is nearly impossible to test that one is acting correctly, except that 
both sides of it adhere to the rules for a UA.

I suggest you abandon any attempt to generically test a B2BUA. Instead, 
decide on the behavior you require of the kind of B2BUA you are 
interested in and test that. If you are interested in a B2BUA that acts 
as a control point in the middle of calls handled by an IP-PBX, then 
test for that. You will have to figure out the requirements that need 
testing, because there are no standards for such a beast.

> Also in general in any B2BUA call flow, in a real life situation the Call
> originates from an Endpoint and reaches the B2BUA that then ends & initiates
> the Call and controls it. Is that correct..?

Not in general. For instance it isn't true for a "click to dial" 
application. Maybe it is true for the kind of B2BUA you are interested in.

	Paul


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: [Sipping] Call Transfer through  B2BUA..
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Hi All,

The 3pcc RFC 3725 depicts call flows that are best current practices for
B2BUAs...
In all the Call Flows the signalling initiates from the B2BUA i.e the INVITE
is sent from the B2BUA that then puts the 2 UAs into a Call.
But in the real life scenario like Call Transfer, the Call originates from
an End Point A that then reaches a IP-PBX (B2BUA) that then tranfers the
Call to End Point B. Pl. find below the flows equivalent to those given in
the RFC 3725 where the Call originates from an End Point.

Equivalent to Flow 1

	A			B2BUA		B
	 INV (Sdp A)	
	------------>
	100 Trying			INV (Sdp A)
	<------------		------------>
	 				200 (Sdp B)
					<--------------
					 ACK
	200 (Sdp B)			----------->
	<-------------
	ACK
	------------>
			RTP
	<----------------------------------->

Equivalent to Flow 2

	A			B2BUA		B
	 INV (Sdp A)
	-------------->
	200 ( blackhole Sdp)
	<--------------
				INV (Sdp A)
	   ACK		------------->
	<---------------
				200 (Sdp B)
				<-------------
				ACK
				-------------->
	ReINV (Sdp B)
	<-------------- 
	200
	-------------->
	ACK
	<--------------
			RTP
	<-------------------------------->

Equivalent to Flow 3

	A			B2BUA		B
	INV (Sdp A)
	-------------->
	200 (black hole Sdp)
	<--------------
	ACK
	--------------->		INV (No Sdp)
				  --------------->
					200 (Sdp B)
				   <--------------
	ReINV (Sdp B)
	<---------------
	200 (Sdp C)
	--------------->		ACK (Sdp C)
				  ---------------->
	ACK
	<---------------
			RTP
	<--------------------------------->

Flow 4 in RFC 3725 sends out an INVITE from the B2BUA without any media line
in the Sdp.
This flow is not possible for an INVITE that comes from an external enppoint
that will always have an SDP with some media line.

I am in the process of coming up with a test Suite for B2BUA. I am assumuing
that a B2BUA that implements Call Transfer functionality will use any one of
the above flows to achieve Call transfer. Am I correct in assuming this. 

Also in general in any B2BUA call flow, in a real life situation the Call
originates from an Endpoint and reaches the B2BUA that then ends & initiates
the Call and controls it. Is that correct..?

Thanks,
Badri

						


Badri Srinivasan S
Networking Technology Labs,
HCL Technologies,
Plot No:66 South Phase
Ambattur Industrial Estate
Chennai-58
Phone:91 44 26521077, 26521270  Xtn: 2243
Fax 52060485

----------------------------------------------------------------------------
-----------
"After much research, Polycom selected HCL's SIP protocol tester to ensure
our products deliver the carrier-grade reliability our conferencing service
providers demand. In addition, HCL's tools provide the in-depth testing we
need to speed the certification of our Voice over IP solutions and guarantee
interoperability," said Matt Taylor, Staff Engineer for Polycom
----------------------------------------------------------------------------
------------

visit us @ http://www.hcltechnologies.com/voip

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



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Subject: [Sipping] B2BUA for testing..
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Hi:

I am developing a test suite for testing a B2BUA implementation..Are there
any freely available B2BUA implementation available for this purpose.
Any pointers would be of great help.

Thanks,
Badri

Badri Srinivasan S
Networking Technology Labs,
HCL Technologies,
Plot No:66 South Phase
Ambattur Industrial Estate
Chennai-58
Phone:91 44 26521077, 26521270  Xtn: 2243
Fax 52060485

----------------------------------------------------------------------------
-----------
"After much research, Polycom selected HCL's SIP protocol tester to ensure
our products deliver the carrier-grade reliability our conferencing service
providers demand. In addition, HCL's tools provide the in-depth testing we
need to speed the certification of our Voice over IP solutions and guarantee
interoperability," said Matt Taylor, Staff Engineer for Polycom
----------------------------------------------------------------------------
------------

visit us @ http://www.hcltechnologies.com/voip

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



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look @ sipcenter.com =20
  u can find quite some info on test gears..etc,,,
=20
Rama


badrisree@npd.hcltech.com wrote:
Hi:

I am developing a test suite for testing a B2BUA implementation..Are ther=
e
any freely available B2BUA implementation available for this purpose.
Any pointers would be of great help.

Thanks,
Badri

Badri Srinivasan S
Networking Technology Labs,
HCL Technologies,
Plot No:66 South Phase
Ambattur Industrial Estate
Chennai-58
Phone:91 44 26521077, 26521270 Xtn: 2243
Fax 52060485

-------------------------------------------------------------------------=
---
-----------
"After much research, Polycom selected HCL's SIP protocol tester to ensur=
e
our products deliver the carrier-grade reliability our conferencing servi=
ce
providers demand. In addition, HCL's tools provide the in-depth testing w=
e
need to speed the certification of our Voice over IP solutions and guaran=
tee
interoperability," said Matt Taylor, Staff Engineer for Polycom
-------------------------------------------------------------------------=
---
------------

visit us @ http://www.hcltechnologies.com/voip
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
	=09
---------------------------------
Do you Yahoo!?
Yahoo! Photos: High-quality 4x6 digital prints for 25=A2
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<DIV>look @ sipcenter.com&nbsp; </DIV>
<DIV>&nbsp; u can find quite some info on test gears..etc,,,</DIV>
<DIV>&nbsp;</DIV>
<DIV>Rama</DIV>
<DIV><BR><BR><B><I>badrisree@npd.hcltech.com</I></B> wrote:</DIV>
<BLOCKQUOTE class=3Dreplbq style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #1010ff 2px solid">Hi:<BR><BR>I am developing a test suite f=
or testing a B2BUA implementation..Are there<BR>any freely available B2BU=
A implementation available for this purpose.<BR>Any pointers would be of =
great help.<BR><BR>Thanks,<BR>Badri<BR><BR>Badri Srinivasan S<BR>Networki=
ng Technology Labs,<BR>HCL Technologies,<BR>Plot No:66 South Phase<BR>Amb=
attur Industrial Estate<BR>Chennai-58<BR>Phone:91 44 26521077, 26521270 X=
tn: 2243<BR>Fax 52060485<BR><BR>-----------------------------------------=
-----------------------------------<BR>-----------<BR>"After much researc=
h, Polycom sselected HCL's SIP protocol tester to ensure<BR>our products =
deliver the carrier-grade reliability our conferencing service<BR>provide=
rs demand. In addition, HCL's tools provide the in-depth testing we<BR>ne=
ed to speed the certification of our Voice over IP solutions and guarante=
e<BR>interoperability," said Matt Taylor, Staff
 Engineer for Polycom<BR>------------------------------------------------=
----------------------------<BR>------------<BR><BR>visit us @ http://www=
.hclttechnologies.com/voip<BR>___________________________________________=
____<BR>Sip-implementors mailing list<BR>Sip-implementors@cs.columbia.edu=
<BR>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors</BLOCK=
QUOTE><p>
		<hr size=3D1><font face=3Darial size=3D-1>Do you Yahoo!?<br>
Yahoo! Photos: <a href=3D"http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=
=3D23765/*http://photos.yahoo.c
om/ph/print_splash">High-quality 4x6 digital prints for 25=A2</a>
--0-303367069-1082377356=:73637--

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From exim@www1.ietf.org  Tue Apr 20 08:05:39 2004
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Reply-To: <skeagy@emergicom.com>
From: "Scott Keagy" <skeagy@emergicom.com>
To: <nena-ltd@cs.columbia.edu>, <sipping@ietf.org>
Date: Tue, 20 Apr 2004 04:43:02 -0700
Organization: Emergicom, Inc.
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Subject: [Sipping] emergency call-back scenarios and requirements
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Need to be specific in managing emergency call-backs (don't know if this
has already been discussed).

If I'm sips:skeagy@emergicom.com as my call-back, and I have 3 devices
(one of which is in my wife's possession in a different state/country
while I'm traveling for work), I want to make sure that my wife doesn't
answer a call-back first and cut me out of the loop accidentally. I'm
sure she'd be happy to be aware of the incident (and stressed it's
happening), but it would be disconcerting to me to not have updates or
be able to provide more useful details to speed a response.

This is aggravated in an enterprise where you have multiple line
appearances with same phone number or identifier, and they can be in
separate states and answered by people with no real knowledge of or
vested interest in each other.

So here's a set of requirements (or best practice suggestions) to the
problem of handling emergency call-backs:
1) need to recognize when an inbound call to arbitrary subscriber is
related to an emergency incident. Would be useful classification for
applying call-based precedence/preemption policies (can argue all day
about whether or not this is a good thing, but having the nerd knob lets
the affected user make the decision), as well as recognizing when to
apply the rule of steering the call to a given device.

2) Default configuration of proxies to do a serial search for an end
device associated with a URI (as opposed to a forked/parallel search)
when the inbound call is recognized as an emergency callback.

3) The first device on the list of serial searches for emergency
call-backs is the last one cached that made an emergency call. Apply
whatever policies for remainder (for example, whatever is preferred
policy for normal calls, or perhaps a forced parallel/forked search to
get a quick response on any other related line if no response on primary
device).

4) This implies that all emergency calls with a reply-to or From: header
are from devices that are in the search-list for the URI in the reply-to
or From: header. Client devices should be able to verify whether this is
the case, and warn the user or administrator during configuration if
it's not the case)


Regards,
Scott



_______________________________________________
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Message-ID: <C90B77EFE2FAF6459B42E1C686F6E2CB03ABA3BA@MSTELEFONICA8>
From: De Leo Walter <deleow@TELEFONICA.COM.AR>
Cc: sipping@ietf.org
Subject: RE: [Sipping] Use of P-Asserted
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Thank you for your answer. It is very useful to me to know your opinion.
More inline
Regards.

-----Mensaje original-----
De: Cullen Jennings [mailto:fluffy@cisco.com] 
Enviado el: Lunes, 19 de Abril de 2004 12:09 a.m.
Para: De Leo Walter
CC: 'sipping@ietf.org'
Asunto: Re: [Sipping] Use of P-Asserted

Not sure I fully understand but I think what you are saying sounds like a
reasonable way to use it.

By UAC's in the trust domain I assume you mean GWs, Call Agents, and B2BUA,
that are trusted not IP Phones sitting on someone's desk. [WDL]  That's
correct. We are talking of Call Agents, GW and B2BUA service platforms.


On 4/16/04 7:38 AM, "De Leo Walter" <deleow@TELEFONICA.COM.AR> wrote:



Cullen: 

I have one concern about a use of a P-Asserted Identity header that some
folks are using. It is, from my point of view out of the scope of RFC 3325,
but I think it is a reasonable use.  The use I am referring to is the
following:

Calls initiated by UACs inside the trust domain include the P-Asserted
Identity (they are part of the network who asserts the identity) header in
their own INVITEs.

I refer specifically to this scenario: 

Suppose a SIP an ingress Media Gateway Controller, a B2BUA Service Platform
and an egress MGC. 
Calls initiated (from the SIP viewpoint) by ingress MGC or B2BUA carry the
P-Asserted Identity header inserted by themselves. The egress Media Gateway
Controller maps allways the content of the P-Assertd Identity header to the
ANI field in corresponding ISUP message and the presentation restriction
information in the Privacy Header to the corresponding ISUP field.

So the egress MGC has always the same policy, INVITES which don't have a
P-Asserted Identity header are not trusted, and if they have one either it
comes from a UAC authenticated by a proxy in the Trust domain or it comes
from inside the Trust domain itself and are, hence, trusted. (I am asuming
the communication inside the trust domain is safe).

Do you see any problem in this use of P-Asserted Identity header. 

I write to you because you are co-author of RFC 3325. I hope not having
disturbed you with this question. I copy the sipping list just in case
someone else has seen this use of P-Asserted Identity header and has
something to say.

Best Regards. 

Walter D. De Leo 








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<DIV><SPAN class=260521313-20042004><FONT face="Courier New" color=#0000ff 
size=2>Thank you for your answer. It is very useful to me to know your 
opinion.<BR></FONT></SPAN><FONT size=2><SPAN class=260521313-20042004><FONT 
face="Courier New"><FONT color=#0000ff>More inline<BR></FONT></FONT></SPAN><SPAN 
class=260521313-20042004><FONT face="Courier New" 
color=#0000ff>Regards.</FONT></SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV><FONT face=Tahoma size=2>-----Mensaje original-----<BR><B>De:</B> Cullen 
  Jennings [mailto:fluffy@cisco.com] <BR><B>Enviado el:</B> Lunes, 19 de Abril 
  de 2004 12:09 a.m.<BR><B>Para:</B> De Leo Walter<BR><B>CC:</B> 
  'sipping@ietf.org'<BR><B>Asunto:</B> Re: [Sipping] Use of 
  P-Asserted<BR><BR></FONT><FONT face=Verdana><SPAN style="FONT-SIZE: 14px">Not 
  sure I fully understand but I think what you are saying sounds like a 
  reasonable way to use it.<BR><BR>By UAC's in the trust domain I assume you 
  mean GWs, Call Agents, and B2BUA, that are trusted not IP Phones sitting on 
  someone's desk. <SPAN class=260521313-20042004><FONT face="Courier New" 
  color=#0000ff size=2>[WDL]&nbsp; That's correct.&nbsp;We are talking of Call 
  Agents, GW and B2BUA service platforms.</FONT></SPAN><BR><BR><BR>On 4/16/04 
  7:38 AM, "De Leo Walter" &lt;deleow@TELEFONICA.COM.AR&gt; 
  wrote:<BR><BR></SPAN></FONT></DIV>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <BLOCKQUOTE><FONT face=Verdana><SPAN style="FONT-SIZE: 14px">Cullen: 
      <BR><BR>I have one concern about a use of a P-Asserted Identity header 
      that some folks are using. It is, from my point of view out of the scope 
      of RFC 3325, but I think it is a reasonable use. &nbsp;The use I am 
      referring to is the following:<BR><BR>Calls initiated by UACs inside the 
      trust domain include the P-Asserted Identity (they are part of the network 
      who asserts the identity) header in their own INVITEs.<BR><BR>I refer 
      specifically to this scenario: <BR><BR>Suppose a SIP an ingress Media 
      Gateway Controller, a B2BUA Service Platform and an egress MGC. <BR>Calls 
      initiated (from the SIP viewpoint) by ingress MGC or B2BUA carry the 
      P-Asserted Identity header inserted by themselves. The egress Media 
      Gateway Controller maps allways the content of the P-Assertd Identity 
      header to the ANI field in corresponding ISUP message and the presentation 
      restriction information in the Privacy Header to the corresponding ISUP 
      field.<BR><BR>So the egress MGC has always the same policy, INVITES which 
      don't have a P-Asserted Identity header are not trusted, and if they have 
      one either it comes from a UAC authenticated by a proxy in the Trust 
      domain or it comes from inside the Trust domain itself and are, hence, 
      trusted. (I am asuming the communication inside the trust domain is 
      safe).<BR><BR>Do you see any problem in this use of P-Asserted Identity 
      header. <BR><BR>I write to you because you are co-author of RFC 3325. I 
      hope not having disturbed you with this question. I copy the sipping list 
      just in case someone else has seen this use of P-Asserted Identity header 
      and has something to say.<BR><BR>Best Regards. <BR><BR>Walter D. De Leo 
      <BR><BR><BR></SPAN></FONT></BLOCKQUOTE><FONT face=Verdana><SPAN 
    style="FONT-SIZE: 14px"><BR></BLOCKQUOTE></BLOCKQUOTE></SPAN></FONT></BODY></HTML>

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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Tue, 20 Apr 2004 17:30:32 +0300
From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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        Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
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Subject: [Sipping] Re: [Simple] MESSAGE Exploders draft
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Ben:

Thanks for your comments. Sorry for my late answer. See my inline comments

ext Ben Campbell wrote:

> I like this approach overall.
> 
> A couple of comments and questions. Apologies if these have already been 
> discussed.
> 
> Is it legal for the list parameter to point to something other than a 
> body part? That is, can I have external lists?

I believe this question lies in the field of the URI list draft 
(draft-camarillo-sipping-uri-list-02.txt). Section 8 of that draft has a 
discussion.


> Saying the exploder MUST NOT include the URI list in the outbound 
> messages seems a little severe. I can imagine applications where I want 
> the recipients to know the list. This could be useful for "reply to all" 
> sort of functions. But if we do allow this, we would need to give the 
> sender a way to choose, possibly on a per-URI basis. (Sort of a "cc" vs. 
> "bcc" thing.)

It seems reasonable to me. I think what we wanted to say is that the 
Request-URI of the outboud message SHOULD NOT contain a list parameter. 
The body need not carry the list, but may carry it for other purposes 
(such as the one you mentioned).


> 
> Instead of saying that message exploders must always send message 
> requests, we might consider saying the outbound request method must 
> match the outbound method. This would allow for exploders that know how 
> to explode more than one method.

No, I don't agree. In the past we got an agreement that we will create 
a message-specific exploder draft. There is a draft that says how to 
deal with INVITE, other for SUBSCRIBE, other for REFER, and this draft 
is about MESSAGE.  We don't want to go back and make the draft generic, 
to be applicable to any method. So I believe the wording is correct as 
it is now.

> 
> 
> When condidering DoS issues, it might be worth suggesting behavior if 
> the exploder finds duplicate URIs on the list. For example, I could put 
> the same URI in a bunch of times, and have a nice DoS amplifier.'

Good point. We will add a discussion on the subject.

Miguel.
> 
> 
> Miguel Garcia wrote:
> 
>>Hi:
>>
>>I have just submitted a draft that discusses a exploder of MESSAGEs.
>>
>>While it appears in the repository, you can download them from:
>>
>>http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.txt 
>>
>>http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.html 
>>
>>
>>Since this is an area covered by SIMPLE (Instant Messaging) and SIPPING 
>>(exploders), I am copying both mailing lists.
>>
>>Regards,
>>
>>     Miguel
> 
> 
> 
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland


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Miguel Garcia wrote:

[...]
>>
>> Instead of saying that message exploders must always send message 
>> requests, we might consider saying the outbound request method must 
>> match the outbound method. This would allow for exploders that know 
>> how to explode more than one method.
> 
> 
> No, I don't agree. In the past we got an agreement that we will create a 
> message-specific exploder draft. There is a draft that says how to deal 
> with INVITE, other for SUBSCRIBE, other for REFER, and this draft is 
> about MESSAGE.  We don't want to go back and make the draft generic, to 
> be applicable to any method. So I believe the wording is correct as it 
> is now.

My concern is, the text as written can be interpreted to mean that 
device cannot be both a MESSAGE exploder and an INVITE exploder, because 
exploding INVITE explicitly violates a normative requirement for MESSAGE 
exploders. I know that is not the intent, but I think it can be misread. 
Perhaps we simply need to say that exploding non-MESSAGE methods is out 
of scope for this document.

Thanks!

Ben.

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Hi Paul:

Thanks for your comments. Answers inline.

ext Paul Kyzivat wrote:

> Miguel,
> 
> In general the approach in the draft seems reasonable to me.
> 
> I see one area of ambiguity, having to do with how multipart bodies are 
> handled. The draft says to remove the list and include everything else. 
> But even your example doesn't do that - it also removes the 
> multipart/mixed wrapper. Of course in this particular example that makes 
> perfect sense, but it goes beyond what the draft says to do.

We start from the the fact that there will be as a minimum, two bodies, 
the list, and the actual payload. There might be more bodies, such as 
multiple payloads, S/MIME signatures, etc.

I guess the algorithm should be:

- If there is any kind of security body for the exploder (e.g., signed 
with the public key of the exploder), then remove that body.

- Remove the list itself (in a typical use case, unless an extension 
says the oposite; I am referring to Ben's suggestion to allow it).

- If there is only one body left, then remove the multipart/mixed wrapper.

- Send the remaining stuff.


> 
> There are more complex cases:
> 
> - for security there could be signed body parts.
> 
>    - The signature could be on the multipart containing the message
>      and list. It would have to be removed.
> 
>    - The signature could be on just the message. Presumably
>      it should remain.
 >

I agree.

> 
> - There could be other body parts referenced by cid: urls in
>    other headers in the message.

Yes, they should remain.
> 
> I think there needs to be a more prescriptive way of defining what gets 
> passed on in the exploded messages. This is really just a manifestation 
> of a more general problem of how to decide how different body parts 
> should be processed. It potentially exists in a normal, unexploded, MESSAGE.
> 
> I think at least part of the answer lies with Content-Disposition. I 
> think there should be clarity about the content-disposition should be 
> for the body part(s) that MESSAGE interprets as message content. 
> Probably "render" (which is default for sip) would be an appropriate 
> content-disposition for message content.

Agree so far. It may happen that some of this stuff is generic enough 
(not MESSAGE specific).

> 
> Also, I think there should be some clarity about what the 
> content-disposition should be for various kinds of multipart containers 
> when used in the various ways they may be used in sip. I'm not quite 
> sure what the answer is here.
> 
> The list itself is of course referenced from a cid: url. It is the url 
> and its placement that determines how the corresponding body part should 
> be processed. For these, the content-disposition should be some value 
> that doesn't imply some other sort of processing. I think it can't be 
> any of "session", "render", "alert", or "icon". Maybe it could be 
> "attachment".

I will be happy to have a very clear deterministic way to find out what 
to pass and what to keep to the other side of the exploder, base on 
presumabley the content-disposition. Let's see if I can come up with 
something.

Thanks,

     Miguel

> 
> 	Paul
> 
> 
> 
> Miguel Garcia wrote:
> 
>>Hi:
>>
>>I have just submitted a draft that discusses a exploder of MESSAGEs.
>>
>>While it appears in the repository, you can download them from:
>>
>>http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.txt 
>>
>>http://people.nokia.net/~miguel/drafts/draft-garcia-simple-message-exploder-00.html 
>>
>>
>>Since this is an area covered by SIMPLE (Instant Messaging) and SIPPING 
>>(exploders), I am copying both mailing lists.
>>
>>Regards,
>>
>>     Miguel
> 
> 
> 
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland


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CC: SIMPLE mailing list <simple@ietf.org>,
        SIPPING mailing list <sipping@ietf.org>,
        Gonzalo Camarillo <gonzalo.camarillo@ericsson.com>
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ext Ben Campbell wrote:
> 
> Miguel Garcia wrote:
> 
> [...]
> 
>>>
>>> Instead of saying that message exploders must always send message 
>>> requests, we might consider saying the outbound request method must 
>>> match the outbound method. This would allow for exploders that know 
>>> how to explode more than one method.
>>
>>
>>
>> No, I don't agree. In the past we got an agreement that we will create 
>> a message-specific exploder draft. There is a draft that says how to 
>> deal with INVITE, other for SUBSCRIBE, other for REFER, and this draft 
>> is about MESSAGE.  We don't want to go back and make the draft 
>> generic, to be applicable to any method. So I believe the wording is 
>> correct as it is now.
> 
> 
> My concern is, the text as written can be interpreted to mean that 
> device cannot be both a MESSAGE exploder and an INVITE exploder, because 
> exploding INVITE explicitly violates a normative requirement for MESSAGE 
> exploders. I know that is not the intent, but I think it can be misread. 
> Perhaps we simply need to say that exploding non-MESSAGE methods is out 
> of scope for this document.
> 
> Thanks!
> 
> Ben.


Ok, I understand your concern, and it sounds fine to me. We will specify 
that other messages are outside the scope of the document. Thanks.

    Miguel

-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland


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Andy Spitzer wrote:

> This is why we put the conference id in the user part...so that each 
> conference mix can be assigned a unique name (user part), and calls for 
> that "user" find the correct MS that hosts that particular mix.  Using 
> the "normal" user@domain approach seemed like the right way to pin users 
> (aka conference ids) to endpoints.

Zork! It's a GRUU!

--
Dean

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Scott Keagy wrote:

> Need to be specific in managing emergency call-backs (don't know if this
> has already been discussed).

Isn't it enough to say that the Contact used for emergency calls SHOULD 
be a GRUU, and that call-backs made from emergency call centers MUST use 
the Contact of the original call, not the AOR indicated in the From: field?

--
Dean

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Date: Wed, 21 Apr 2004 11:15:50 +0300
From: "Niemi Aki (Nokia-M/Espoo)" <aki.niemi@nokia.com>
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To: ext Miguel Garcia <Miguel.An.Garcia@nokia.com>
CC: SIPPING WG <sipping@ietf.org>
Subject: Re: [Sipping] Re: [Simple] MESSAGE Exploders draft
References: <406DE1C8.3040309@dynamicsoft.com> <40853408.1020804@nokia.com>
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ext Miguel Garcia wrote:

[snip]

>> Saying the exploder MUST NOT include the URI list in the outbound 
>> messages seems a little severe. I can imagine applications where I
>>  want the recipients to know the list. This could be useful for
>> "reply to all" sort of functions. But if we do allow this, we would
>> need to give the sender a way to choose, possibly on a per-URI
>> basis. (Sort of a "cc" vs. "bcc" thing.)
> 
> 
> It seems reasonable to me. I think what we wanted to say is that the
>  Request-URI of the outboud message SHOULD NOT contain a list
> parameter. The body need not carry the list, but may carry it for
> other purposes (such as the one you mentioned).

I agree.

One other thing, to convey such Cc information to the final recipients
of the explosion, message/cpim actually seems like a very good fit. It
already has the To, Cc headers defined, and the sender can populate them
as it sees fit and the exploder need not care. Including a URI in the
MESSAGE resource list, but omitting it from the To, CC in message/cpim
would result in exactly the Bcc-type behavior.

Cheers,
Aki


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



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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Barry Dingle" <barry.dingle@bigfoot.com.au>,
        "Rosen, Brian" <Brian.Rosen@marconi.com>,
        "Arnoud van Wijk" <a.vwijk@viataal.nl>,
        <Gonzalo.Camarillo@ericsson.com>, <sipping@ietf.org>
Cc: <rrroy@att.com>, <henry.sinnreich@mci.com>, <paulej@packetizer.com>,
        <gv@trace.wisc.edu>
Subject: RE: Betr.: [Sipping] ToIP Requirements
Date: Wed, 21 Apr 2004 18:56:52 +0200
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Just found this mail that I want to add some clarification to.

A calling V.18 signals the V.8 CI tone signal with the "textphone" call
type.
It is a tone and a specific V.21 bit pattern.

An old answering V.18 answers with ANS followed by a V.21 modulated signa=
l
called "txp".

A newer answering V.18 answers with the full V.8 answering procedure
starting with ANSAM.

That is valid for proper V.18s

Under v.18 interoperability and described in its Annex A-F are older
textphone types that do not have these signals. The detection logic of V.=
18
tells you how they signal.


All of these calls come as audio or telephony calls in ISDN.

Gunnar

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


> -----Original Message-----
> From: Barry Dingle [mailto:barry.dingle@bigfoot.com.au]
> Sent: Monday, March 01, 2004 3:10 AM
> To: Rosen, Brian; Gunnar Hellstrom; Arnoud van Wijk;
> Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> Cc: rrroy@att.com; henry.sinnreich@mci.com; paulej@packetizer.com;
> gv@trace.wisc.edu; Barry Dingle
> Subject: Re: Betr.: [Sipping] ToIP Requirements
>
>
> Hi Brian,
>
> There are two issues. The old Textphones that must continue to
> be supported
> and the more recent V.18 Textphones that have in-built
> interworking with the
> old Textphone protocols (and will revert to the detected protocol).
>
> A V.18 Textphone detects the type of Textphone with which it is
> interworking
> by sending out a sequence of modem signals based on the V.8 modem
> identification and control sequence plus it listens for modem
> signals from
> up to 7 Textphone modem protocols.
>
> Some Textphone require manual intervention to create a modem
> signal. In that
> case, V.18 has to assume that someone will send some text so it
> can detect
> the incoming protocol.
>
> In summary, V.18 Textphones are easily detectable but some older
> Textphone
> protocols are not. This becomes a more significant issue when
> talking about
> emergency service.
>
> BTW - V.18 only works on analog PSTN lines - not on ISDN lines directly.
>
> Cheers,
> Barry DINGLE
> Telecommunications Consultant
> Project Manager - ACIF Any-to-Any Text Connectivity Options WG
> Fellow of the University of Melbourne
>    barry.dingle@bigfoot.com.au
> Mob:   +61 (0)41 911 7578
> Fixed: +61 (0)3 9725 3937
> ----- Original Message -----
> From: "Rosen, Brian" <Brian.Rosen@marconi.com>
> To: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>; "Arnoud van Wijk"
> <a.vwijk@viataal.nl>; <Gonzalo.Camarillo@ericsson.com>;
> <sipping@ietf.org>
> Cc: <rrroy@att.com>; <barry.dingle@bigfoot.com.au>;
> <henry.sinnreich@mci.com>; <paulej@packetizer.com>; <gv@trace.wisc.edu>
> Sent: Monday, March 01, 2004 11:16 AM
> Subject: RE: Betr.: [Sipping] ToIP Requirements
>
>
> > How does V.18 work?  Is the "calling identifier" something in (ISDN
> presumably) signalling?  Or is it a distinctive tone
> > pattern?
> >
> > Brian
> >
> > > -----Original Message-----
> > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > Sent: Friday, February 27, 2004 1:45 AM
> > > To: Arnoud van Wijk; 'Rosen, Brian'; Gonzalo.Camarillo@ericsson.com=
;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > The INCOM group of the European commission made
> recommendations that the
> > > same emergency number shall be used for all kinds of terminals.
> > > It is dangerous to have different prefixes or other tricks
> in different
> countries. How shall you keep track of that if you are a visitor?
> > >
> > > From PSTN, it is only the old textphones that cause problems.
> > > A calling V.18 sends a very distinct calling identifier that can be
> catched by a gateway or
> > > an emergency centre and be properly handled. On the SIP side, the
> options are even better.
> > >
> > > I do not mean that all problems are solved by this
> statement, but I do
> not
> > > think that the prefix is the only solution, and I do not
> think we shall
> > > recommend it in connection to emergency.
> > >
> > >
> > > Gunnar
> > >
> > > -------------------------------------------
> > > Gunnar Hellstrom
> > > Omnitor AB
> > > Renathvagen 2
> > > SE 121 37 Johanneshov
> > > SWEDEN
> > > +46 8 556 002 03
> > > Mob: +46 708 204 288
> > > www.omnitor.se
> > > Gunnar.Hellstrom@Omnitor.se
> > > --------------------------------------------
> > >
> > >
> > > -----Original Message-----
> > > From: sipping-admin@ietf.org
> > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > Arnoud van Wijk
> > > Sent: Thursday, February 26, 2004 4:57 PM
> > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > >
> > > Hello Brian,
> > >
> > > Yes you are right about that the PSTN cannot detect if it is
> > > a voice call or
> > > TDD/Textphone call. They are both audio.
> > > The PSTN would if it can detect it is a textphone call, route
> > > it to the text
> > > gateway. And transcode the call into T.140 over RTP as
> > > decribed in RFC 2793.
> > > This is for PSTNs which use IP to interconnect with other
> > > PSTNs or with any
> > > SIP telephones. So it is for analoge telephony into SIP
> > > telephony here.
> > >
> > > In the ToIP draft:
> > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-to
> > > ip-01.txt
> > > A possible solution is a prefix for textphone users before
> > > dialling the
> > > selected number.
> > > It indeed would mean that you dial e.g. 600-112 instead 112 directl=
y.
> > > Is that bad?
> > > I believe it is not.
> > >
> > > Most textphones have the emergency number pre programmed. And
> > > when textphone
> > > users have to use a pre-fix for any textphone call, then the
> > > same prefix
> > > before 112 or 911 will be fine.
> > > In the Netherlands, we have for emergency calls 800-112
> > > instead of 112. It
> > > works fine, only almost no deaf users call 112, since they
> > > are then mostly
> > > in a situation where they do not have a textphone nearby!
> > >
> > > Maybe there are other solutions that allow the PSTN to
> detect that the
> > > incoming call is a textphone user and acts on it.
> > >
> > > For example it is possible to create a database where the
> > > PSTN can check if
> > > the incoming number is registered as a textphone.
> > >
> > > But using a text gateway to connect PSTN textphones/TDDs to
> > > ToIP is still
> > > the best solution, even the only solution in my opinion.
> > > There should be no attempts to transport the text telephone
> > > tones (e.g.
> > > Baudot) over the IP voice channel! I believe that will open
> > > Pandora's box.
> > >
> > > I am not a PSTN expert, and we can surely look into this
> > > matter to find a
> > > solution. Let it be prefix or any other solution that lets the PSTN
> > > immediately route the call to the text gateway.
> > >
> > > What do you PSTN people think? How would you tackle this
> > > problem? Your input
> > > will be very welcome.
> > >
> > > Greetz
> > >
> > > Arnoud
> > >
> > > Drs. Arnoud A. T. van Wijk
> > > Viataal
> > > Research & Development
> > > Afdeling RDS
> > > Theerestraat 42
> > > 5271 GD Sint-Michielsgestel
> > > The Netherlands.
> > > Mobile: +31651921948
> > > International text telephone: +31735588408
> > >
> > >
> > > -----Original Message-----
> > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > Sent: donderdag 26 februari 2004 15:36
> > > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > > Gonzalo.Camarillo@ericsson.com;
> > > sipping@ietf.org
> > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > henry.sinnreich@mci.com;
> > > paulej@packetizer.com; gv@trace.wisc.edu
> > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > >
> > > Arnoud
> > >
> > > I do not believe this can work.  A TDD user dials 112 or 911.
> > >  Today, this
> > > will be routed directly to a "selective router" (a TDM
> > > switch) which sends
> > > the
> > > call to the correct Emergency Response Center.  You can't
> > > insert a gateway,
> > > because you don't know that it's a TDD call until it is
> > > answered.  To make
> > > it work, you would have to establish a special PSTN number
> > > for emergency
> > > TDD,
> > > which would route to the gateway, which would then call the
> > > ERC.  I don't
> > > believe that is practical.
> > >
> > > Brian
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > > Sent: Thursday, February 26, 2004 9:17 AM
> > > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > > Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > >
> > > > Hello Brian,
> > > > That is also why we need to look at the emergency call system.
> > > > This text gateway is only at the PSTN side, thus another
> > > > requirement for a
> > > > T140/RTP gateway for the PSTN is that when a local
> > > emergency number is
> > > > dialled (112, or 911 etc). That the gateway must be able to
> > > > receive that
> > > > call (even dropping existing calls if necessary). Such a call
> > > > is easy to
> > > > recognize as an emergency call. (PSTN side).
> > > > From that gateway on, it is all IP with RFC2793 interactive
> > > > text. And then
> > > > it is just all existing SIP as if it was from an IP end-point.
> > > >
> > > > This gives emergency centers an advantage, all calls will be
> > > > T.140/RTP, so
> > > > integration in their computer systems will be easy instead
> > > > being forced to
> > > > have analog textphones available for a textphone call. (or a
> > > > separate text
> > > > telephone emergency center).
> > > >
> > > > /Arnoud
> > > >
> > > > Drs. Arnoud A. T. van Wijk
> > > > Viataal
> > > > Research & Development
> > > > Afdeling RDS
> > > > Theerestraat 42
> > > > 5271 GD Sint-Michielsgestel
> > > > The Netherlands.
> > > > Mobile: +31651921948
> > > > International text telephone: +31735588408
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
> > > > Sent: donderdag 26 februari 2004 15:04
> > > > To: 'Arnoud van Wijk'; 'Gunnar Hellstrom';
> > > > Gonzalo.Camarillo@ericsson.com;
> > > > sipping@ietf.org
> > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > henry.sinnreich@mci.com;
> > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > >
> > > > Arnoud
> > > >
> > > > Generally, you don't want to rely on very many SIP elements
> > > > to complete
> > > > an emergency call.  Present systems accept TDD at all answering
> > > > points because they don't know before the calls is answered that =
it
> > > > is a TDD call.  I certainly don't want the user to have to
> > > do anything
> > > > other than dial his local emergency number, or complete a device
> > > > specific key sequence to complete an emergency call, and I don't
> > > > want a gateway busy to prevent an emergency call.
> > > >
> > > > Brian
> > > >
> > > > > -----Original Message-----
> > > > > From: Arnoud van Wijk [mailto:a.vwijk@viataal.nl]
> > > > > Sent: Thursday, February 26, 2004 6:07 AM
> > > > > To: 'Rosen, Brian'; 'Gunnar Hellstrom';
> > > > > Gonzalo.Camarillo@ericsson.com;
> > > > > sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > >
> > > > > Actually,
> > > > > When using the text gateway, there is no need to detect the
> > > > > TDD/Analogue
> > > > > text telephone tones/signals. That is the job of the
> text gateway.
> > > > >
> > > > > And for the emergency calling. We have to look closely at this =
in
> > > > > combination with a text gateway. Possible behaviour/action.
> > > > > And possible
> > > > > action in SIP.
> > > > >
> > > > > Right now I cannot say what has to happen, I must study the
> > > > > emergency calls
> > > > > in depth. But as I see it, using the text gateway, all SIP
> > > > > originating and
> > > > > receiving emergency calls will be possible for any
> textphone user.
> > > > >
> > > > > Cheers
> > > > >
> > > > > Arnoud
> > > > >
> > > > >
> > > > > Drs. Arnoud A. T. van Wijk
> > > > > Viataal
> > > > > Research & Development
> > > > > Afdeling RDS
> > > > > Theerestraat 42
> > > > > 5271 GD Sint-Michielsgestel
> > > > > The Netherlands.
> > > > > Mobile: +31651921948
> > > > > International text telephone: +31735588408
> > > > >
> > > > > -----Original Message-----
> > > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]
> > > > > On Behalf Of
> > > > > Rosen, Brian
> > > > > Sent: woensdag 25 februari 2004 23:08
> > > > > To: 'Gunnar Hellstrom'; A vWijk; Gonzalo.Camarillo@ericsson.com=
;
> > > > > sipping@ietf.org
> > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > henry.sinnreich@mci.com;
> > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > >
> > > > > I'm particularly interested in the last one, emergency calling.
> > > > >
> > > > > Given that you propose an appropriate SDP indicating
> > > > support for text,
> > > > > do you think there is anything else we need to do to in SIP
> > > > > to make this
> > > > > happen?  I don't see anything.
> > > > >
> > > > > I know that the problem of using a legacy TDD device on
> > > any kind of
> > > > > adapter is as problematic as current TDD; the call will answere=
d
> > > > > as if it was a voice call, the baudot/whatever tones will be
> > > > > detected,
> > > > > and the operator will look for text.
> > > > > Can't think of any way to make that better.
> > > > >
> > > > > Do note that language preference headers should be set!!!
> > > > >
> > > > > Brian
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Gunnar Hellstrom [mailto:gunnar.hellstrom@omnitor.se]
> > > > > > Sent: Wednesday, February 25, 2004 4:54 PM
> > > > > > To: A vWijk; Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > > henry.sinnreich@mci.com;
> > > > > > paulej@packetizer.com; gv@trace.wisc.edu
> > > > > > Subject: RE: Betr.: [Sipping] ToIP Requirements
> > > > > >
> > > > > >
> > > > > > I think PSTN textphone gateway invocation is one area where
> > > > > > we either need
> > > > > > protocol extensions, or need agreements on what current
> > > > > > mechanisms to use.
> > > > > >
> > > > > > Another is invocation of various kinds of relay services or
> > > > > > translation
> > > > > > services.
> > > > > >
> > > > > > A third is calling emergency services and reach positions
> > > > capable of
> > > > > > handling the type of communication that the user prefers.
> > > > > >
> > > > > >
> > > > > >
> > > > > > Gunnar
> > > > > > -------------------------------------------
> > > > > > Gunnar Hellstrom
> > > > > > Omnitor AB
> > > > > > Renathvagen 2
> > > > > > SE 121 37 Johanneshov
> > > > > > SWEDEN
> > > > > > +46 8 556 002 03
> > > > > > Mob: +46 708 204 288
> > > > > > www.omnitor.se
> > > > > > Gunnar.Hellstrom@Omnitor.se
> > > > > > --------------------------------------------
> > > > > >
> > > > > >
> > > > > > -----Original Message-----
> > > > > > From: sipping-admin@ietf.org
> > > > > > [mailto:sipping-admin@ietf.org]On Behalf Of
> > > > > > A vWijk
> > > > > > Sent: Tuesday, February 24, 2004 1:22 PM
> > > > > > To: Gonzalo.Camarillo@ericsson.com; sipping@ietf.org
> > > > > > Cc: rrroy@att.com; barry.dingle@bigfoot.com.au;
> > > > > > henry.sinnreich@mci.com;
> > > > > > gunnar.hellstrom@omnitor.se; paulej@packetizer.com;
> > > > > gv@trace.wisc.edu
> > > > > > Subject: Betr.: [Sipping] ToIP Requirements
> > > > > >
> > > > > >
> > > > > > Hello Gonzalo,
> > > > > > While there are device requirements in the text. Those
> > > > > > requirements and the
> > > > > > explanation on the different analogue text telephone systems
> > > > > > is included to
> > > > > > give the reader an understanding what kind of devices will
> > > > > > connect with
> > > > > > ToIP. And the need for the text gateways. And what the
> > > > > > different behaviour
> > > > > > can/may be compared to end to end IP devices.
> > > > > > The most important part will be the ToIP framwork and the
> > > > > > text gateways and
> > > > > > how such gateways will be used and work with the ToIP
> framework.
> > > > > >
> > > > > > There is a ToIP framework where a terminal uses streaming
> > > > > > text as defined in
> > > > > > RFC2793 to connect with any other Interactive
> > > > > > texting/real-time streaming
> > > > > > text device.
> > > > > > This can be end-to-end IP, and a direct T140/RTP stream is
> > > > > > established with
> > > > > > SIP. But in other cases, a text gateway must be used to allow
> > > > > > T.140/RTP to
> > > > > > connect with a non IP text telephony device.
> > > > > > It can be an analogue text telephone (TDD/TTY) that uses a
> > > > > > text gateway to
> > > > > > be able to understand RFC2793 text. (PSTN-IP)
> > > > > > It can be a mobile phone running GPRS that uses the text
> > > > gateway to
> > > > > > understand RFC2793 text. (this exist right NOW using SIP! And
> > > > > > those gateways
> > > > > > will be used in mobile networks this year!)
> > > > > > Or a mobile text telephone using CTM via the text gateway to
> > > > > > communicate
> > > > > > with RFC2793 text.
> > > > > > Or even one analogue text telephone connecting to another
> > > > > > analogue text
> > > > > > telephone in a different country, and the PSTN's use IP to
> > > > > > connect the call.
> > > > > >
> > > > > > The question is for example, how do we use that gateway from
> > > > > > the IP side? Is
> > > > > > it invisible for the IP side? Or do we need special
> > > > > > extensions in SIP to
> > > > > > that the IP terminal knows it is communicating with the
> > > > > > end-device through
> > > > > > the text gateway?
> > > > > > There may be a slightly different call setup flow when a text
> > > > > > gateway is
> > > > > > included.
> > > > > > Or additional SIP messages indicating gateway activity. When
> > > > > > you connect for
> > > > > > example to an analogue text telephone with an RFC2793 capable
> > > > > > IP terminal
> > > > > > the capabilities will be less then IP to IP terminal directly
> > > > > > (different
> > > > > > transmission speeds, half/full duplex etc).
> > > > > > Or what to do with emergency calls that are in RFC 2793
> > > text only?
> > > > > >
> > > > > > That are requirements that will be defined and worked out in
> > > > > > this draft.
> > > > > > Also, we want to make everybody in SIP and SIPPING aware
> > > > > > about those text
> > > > > > gateways. So that complicated matters like how to send
> > > > analogue text
> > > > > > telephone text through a PSTN and connect with a SIP
> > > > > > telephone using RFC2793
> > > > > > interactive texting. When using the text gateway, there is
> > > > > > then no need for
> > > > > > RFC 2833 events at the PSTN for text telephones.
> > > > > >
> > > > > > So, yes. I believe that this should be in the SIPPING WG. And
> > > > > > it will fit in
> > > > > > the charter - support for hearing-/speech-impaired calling.
> > > > > >
> > > > > > greetz
> > > > > >
> > > > > > Arnoud
> > > > > >
> > > > > >
> > > > > >
> > > > > > Drs. Arnoud A. T. van Wijk
> > > > > > Viataal
> > > > > > Research & Development
> > > > > > Afdeling RDS
> > > > > > Theerestraat 42
> > > > > > 5271 GD Sint-Michielsgestel
> > > > > > The Netherlands.
> > > > > > Mobile: +31651921948
> > > > > > International text telephone: +31735588408
> > > > > >
> > > > > > >>> Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
> > > > > > 24-02-04 10:21 >>>
> > > > > > Hello,
> > > > > >
> > > > > > the draft below provides a set of requirements for real-time
> > > > > > conversations using SIP.
> > > > > > http://www.ietf.org/internet-drafts/draft-manyfolks-sipping-t=
o
> > > > > ip-01.txt
> > > > >
> > > > > Nevertheless, I have found a lot of device requirements in
> > > > > the text, but
> > > > > I fail to understand which particular protocol requirements
> > > > we need to
> > > > > fulfill with new protocol extensions.
> > > > >
> > > > > While SIPPING is the right place to handle requirements
> > > > that will lead
> > > > > to new SIP protocol extensions, SIPPING may not be the
> > > > right place to
> > > > > deal with device requirements (e.g., text phones MUST have a
> > > > > display of
> > > > > at least 100 characters).
> > > > >
> > > > > Thanks,
> > > > >
> > > > > Gonzalo
> > > > >
> > > > >
> > > > > This communication is confidential and intended solely for
> > > > > the addressee(s).
> > > > > Any unauthorized review, use, disclosure or distribution is
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> > > > > please notify the
> > > > > sender by replying to this transmission and delete the
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> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on current s=
ip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
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> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on current s=
ip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on current s=
ip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > > _______________________________________________
> > > > > Sipping mailing list
> > > https://www1.ietf.org/mailman/listinfo/sipping
> > > > > This list is for NEW development of the application of SIP
> > > > > Use sip-implementors@cs.columbia.edu for questions on current s=
ip
> > > > > Use sip@ietf.org for new developments of core SIP
> > > > >
> > > > >
> > > >
> > > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sippin=
g
> > > This list is for NEW development of the application of SIP
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sip@ietf.org for new developments of core SIP
> > >
> > >
> > >
>
>
>



_______________________________________________
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Date: Wed, 21 Apr 2004 17:02:02 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: [Sipping] Managing multiple body parts
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I changed the subject of this because it is generalized.

See below for details. The general question is:

Is there a need for some sort of draft (beyond what is already in 3261) 
explaining how body parts are to be arranged and tagged so that they 
will be found and processed in an interoperable way?

I think there is. I have spent time working on sip stack support for 
managing body parts, and I came up with all kinds of questions about 
what is valid and what is not.

	Paul

Miguel Garcia wrote:
> Hi Paul:
> 
> Thanks for your comments. Answers inline.
> 
> ext Paul Kyzivat wrote:
> 
>> Miguel,
>>
>> In general the approach in the draft seems reasonable to me.
>>
>> I see one area of ambiguity, having to do with how multipart bodies 
>> are handled. The draft says to remove the list and include everything 
>> else. But even your example doesn't do that - it also removes the 
>> multipart/mixed wrapper. Of course in this particular example that 
>> makes perfect sense, but it goes beyond what the draft says to do.
> 
> 
> We start from the the fact that there will be as a minimum, two bodies, 
> the list, and the actual payload. There might be more bodies, such as 
> multiple payloads, S/MIME signatures, etc.
> 
> I guess the algorithm should be:
> 
> - If there is any kind of security body for the exploder (e.g., signed 
> with the public key of the exploder), then remove that body.
> 
> - Remove the list itself (in a typical use case, unless an extension 
> says the oposite; I am referring to Ben's suggestion to allow it).
> 
> - If there is only one body left, then remove the multipart/mixed wrapper.
> 
> - Send the remaining stuff.

This seems like the ad hoc approach. It will probably work in a lot of 
cases. In the absence of more general rules this is probably a pragmatic 
approach to take.

But I think there is a more general problem - not specific to message 
exploders, or exploders, or message - that applies to management of body 
parts in general.

Part of it is just:
- in a hierarchy of mime entities, how do I identify the body parts that 
interest me?

And the flip side of that is:
- How do I arrange the body parts that I need to convey into a hierarchy 
of mime entities so that the recipient will find and process them correctly?

This is then further complicated by servers in the signaling path that 
are permitted to mess with body parts. (That currently is only B2BUAs, 
but with the m2m and m2e discussions it may eventually include proxies.) 
For them there are the following additional questions:

- Should I (and if so how) modify the mime entity hierarchy to remove 
body parts that I have processed?

- How should I modify the mime entity hierarchy to add additional body 
parts that are to be processed further downstream, so that the recipient 
will process them correctly?

>> There are more complex cases:
>>
>> - for security there could be signed body parts.
>>
>>    - The signature could be on the multipart containing the message
>>      and list. It would have to be removed.
>>
>>    - The signature could be on just the message. Presumably
>>      it should remain.
> 
>  >
> 
> I agree.
> 
>>
>> - There could be other body parts referenced by cid: urls in
>>    other headers in the message.
> 
> 
> Yes, they should remain.
> 
>>
>> I think there needs to be a more prescriptive way of defining what 
>> gets passed on in the exploded messages. This is really just a 
>> manifestation of a more general problem of how to decide how different 
>> body parts should be processed. It potentially exists in a normal, 
>> unexploded, MESSAGE.
>>
>> I think at least part of the answer lies with Content-Disposition. I 
>> think there should be clarity about the content-disposition should be 
>> for the body part(s) that MESSAGE interprets as message content. 
>> Probably "render" (which is default for sip) would be an appropriate 
>> content-disposition for message content.
> 
> 
> Agree so far. It may happen that some of this stuff is generic enough 
> (not MESSAGE specific).

Yes, very much so.

>> Also, I think there should be some clarity about what the 
>> content-disposition should be for various kinds of multipart 
>> containers when used in the various ways they may be used in sip. I'm 
>> not quite sure what the answer is here.
>>
>> The list itself is of course referenced from a cid: url. It is the url 
>> and its placement that determines how the corresponding body part 
>> should be processed. For these, the content-disposition should be some 
>> value that doesn't imply some other sort of processing. I think it 
>> can't be any of "session", "render", "alert", or "icon". Maybe it 
>> could be "attachment".
> 
> 
> I will be happy to have a very clear deterministic way to find out what 
> to pass and what to keep to the other side of the exploder, base on 
> presumabley the content-disposition. Let's see if I can come up with 
> something.

I'm interested in this. Do you think this is suitable as a new work 
item? (Probably for sipping at first.)

	Paul


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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
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        SIPPING mailing list <sipping@ietf.org>
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Hi Paul,

typically you either tag the body with a content disposition tag, or 
have a pointer somewhere in the message pointing to the body part... or 
both.

What sort of things do you have in mind, besides the ones I mentioned?

Thanks,

Gonzalo

Paul Kyzivat wrote:

> I changed the subject of this because it is generalized.
> 
> See below for details. The general question is:
> 
> Is there a need for some sort of draft (beyond what is already in 3261) 
> explaining how body parts are to be arranged and tagged so that they 
> will be found and processed in an interoperable way?
> 
> I think there is. I have spent time working on sip stack support for 
> managing body parts, and I came up with all kinds of questions about 
> what is valid and what is not.
> 
>     Paul
> 
> Miguel Garcia wrote:
> 
>> Hi Paul:
>>
>> Thanks for your comments. Answers inline.
>>
>> ext Paul Kyzivat wrote:
>>
>>> Miguel,
>>>
>>> In general the approach in the draft seems reasonable to me.
>>>
>>> I see one area of ambiguity, having to do with how multipart bodies 
>>> are handled. The draft says to remove the list and include everything 
>>> else. But even your example doesn't do that - it also removes the 
>>> multipart/mixed wrapper. Of course in this particular example that 
>>> makes perfect sense, but it goes beyond what the draft says to do.
>>
>>
>>
>> We start from the the fact that there will be as a minimum, two 
>> bodies, the list, and the actual payload. There might be more bodies, 
>> such as multiple payloads, S/MIME signatures, etc.
>>
>> I guess the algorithm should be:
>>
>> - If there is any kind of security body for the exploder (e.g., signed 
>> with the public key of the exploder), then remove that body.
>>
>> - Remove the list itself (in a typical use case, unless an extension 
>> says the oposite; I am referring to Ben's suggestion to allow it).
>>
>> - If there is only one body left, then remove the multipart/mixed 
>> wrapper.
>>
>> - Send the remaining stuff.
> 
> 
> This seems like the ad hoc approach. It will probably work in a lot of 
> cases. In the absence of more general rules this is probably a pragmatic 
> approach to take.
> 
> But I think there is a more general problem - not specific to message 
> exploders, or exploders, or message - that applies to management of body 
> parts in general.
> 
> Part of it is just:
> - in a hierarchy of mime entities, how do I identify the body parts that 
> interest me?
> 
> And the flip side of that is:
> - How do I arrange the body parts that I need to convey into a hierarchy 
> of mime entities so that the recipient will find and process them 
> correctly?
> 
> This is then further complicated by servers in the signaling path that 
> are permitted to mess with body parts. (That currently is only B2BUAs, 
> but with the m2m and m2e discussions it may eventually include proxies.) 
> For them there are the following additional questions:
> 
> - Should I (and if so how) modify the mime entity hierarchy to remove 
> body parts that I have processed?
> 
> - How should I modify the mime entity hierarchy to add additional body 
> parts that are to be processed further downstream, so that the recipient 
> will process them correctly?
> 
>>> There are more complex cases:
>>>
>>> - for security there could be signed body parts.
>>>
>>>    - The signature could be on the multipart containing the message
>>>      and list. It would have to be removed.
>>>
>>>    - The signature could be on just the message. Presumably
>>>      it should remain.
>>
>>
>>  >
>>
>> I agree.
>>
>>>
>>> - There could be other body parts referenced by cid: urls in
>>>    other headers in the message.
>>
>>
>>
>> Yes, they should remain.
>>
>>>
>>> I think there needs to be a more prescriptive way of defining what 
>>> gets passed on in the exploded messages. This is really just a 
>>> manifestation of a more general problem of how to decide how 
>>> different body parts should be processed. It potentially exists in a 
>>> normal, unexploded, MESSAGE.
>>>
>>> I think at least part of the answer lies with Content-Disposition. I 
>>> think there should be clarity about the content-disposition should be 
>>> for the body part(s) that MESSAGE interprets as message content. 
>>> Probably "render" (which is default for sip) would be an appropriate 
>>> content-disposition for message content.
>>
>>
>>
>> Agree so far. It may happen that some of this stuff is generic enough 
>> (not MESSAGE specific).
> 
> 
> Yes, very much so.
> 
>>> Also, I think there should be some clarity about what the 
>>> content-disposition should be for various kinds of multipart 
>>> containers when used in the various ways they may be used in sip. I'm 
>>> not quite sure what the answer is here.
>>>
>>> The list itself is of course referenced from a cid: url. It is the 
>>> url and its placement that determines how the corresponding body part 
>>> should be processed. For these, the content-disposition should be 
>>> some value that doesn't imply some other sort of processing. I think 
>>> it can't be any of "session", "render", "alert", or "icon". Maybe it 
>>> could be "attachment".
>>
>>
>>
>> I will be happy to have a very clear deterministic way to find out 
>> what to pass and what to keep to the other side of the exploder, base 
>> on presumabley the content-disposition. Let's see if I can come up 
>> with something.
> 
> 
> I'm interested in this. Do you think this is suitable as a new work 
> item? (Probably for sipping at first.)
> 
>     Paul

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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



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Cc: sipping@ietf.org
To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sipping] Re: Service name  stuff in natann draft
References: <BCA4596B.39D9C%fluffy@cisco.com> <opr6iru6h0gjeb4n@pobox2.brooktrout.com> <4085ACCD.6010204@softarmor.com>
Message-ID: <opr6s7v5jvgjeb4n@pobox2.brooktrout.com>
From: Andy Spitzer <woof@brooktrout.com>
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Dean,

Yep.  But netann pre-dates GRUU.

--Andy Spitzer

On Tue, 20 Apr 2004 18:05:49 -0500, Dean Willis 
<dean.willis@softarmor.com> wrote:

> Andy Spitzer wrote:
>
>> This is why we put the conference id in the user part...so that each 
>> conference mix can be assigned a unique name (user part), and calls for 
>> that "user" find the correct MS that hosts that particular mix.  Using 
>> the "normal" user@domain approach seemed like the right way to pin 
>> users (aka conference ids) to endpoints.
>
> Zork! It's a GRUU!
>
> --
> Dean



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Subject: RE: [Sipping] Re: Service name  stuff in natann draft
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Gonzalo Camarillo wrote:
> Hi Paul,
> 
> typically you either tag the body with a content disposition tag, or 
> have a pointer somewhere in the message pointing to the body part... or 
> both.

Note that even if you don't explicitly tag a body with a content 
disposition it implicitly has one, because there is a defaulting rule.

> What sort of things do you have in mind, besides the ones I mentioned?

Here are a few fuzzy issues:

- what content-disposition should a multipart/mixed body have?
   Should the recipient always look inside one for interesting
   content, regardless of the disposition?

- It is ok to nest body parts arbitrarily deep in nested
   multipart/mixed bodies? Or is there a cannonical way that
   body parts should be composed?

- What should happen, in an invite, to a body part with
   content-type of application/sdp but with a content-disposition
   of something other than "session"? (E.g. "render".)

- What should happen if an invite contains two body parts that
   both have content-type of application/sdp and content-disposition
   of session. (In a multipart/mixed, not multipart/alternative.)

- What content-disposition should be used for the content of
   a MESSAGE message? A NOTIFY message? A SUBSCRIBE message?

- What content-disposition should be used for multipart/signed
   and multipart/encrypted bodies? Should it match whatever it
   is that is signed or encrypted?

- What content-disposition should be used for body parts that
   are referenced via a cid: uri somewhere in the message?

- Does the fact that a body part is referenced by a cid: uri
   mean that it shouldn't also be processed as it would have been
   had it not been referenced by a cid: uri? What if the reference
   is from an extension header that the recipient doesn't understand?

- The content-disposition header has a handling parameter.
   An error should be generated if any parts with handling=required
   aren't able to be processed. In the context of sip, how is a
   determination made whether a part has been processed? Is it
   sufficient that it be understood even if it isn't acted upon?

I can probably come up with more, but that is a start.

	Paul


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Date: Thu, 22 Apr 2004 14:51:23 -0500
To: Hisham Khartabil <hisham.khartabil@nokia.com>,
        Rohan Mahy <rohan@cisco.com>, Dean Willis <dean.willis@softarmor.com>
From: "James M. Polk" <jmpolk@cisco.com>
Cc: sipping@ietf.org, sip@ietf.org
In-Reply-To: <200404221531.LAA05708@ietf.org>
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Subject: [Sipping] Re: SIMPLE/SIP/SIPPING/XCON Working Groups Joint Interim
 Meeting Announcement
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What happened to time for Emergency and E2M Security? In Seoul, these were 
mentioned as two of the prevailing topic areas for the interim (along with 
Session Policy and Exploders).

At 11:31 AM 4/22/2004 -0400, Hisham Khartabil wrote:
>An joint interim meeting for the SIMPLE, SIP, SIPPING and XCON working groups
>will be help from Monday 24th May until Wednesday 26th May in Boston, 
>Massachusetts.
>
>Location:
>Renaissance Boston Hotel Bedford
>44 Middlesex Turnpike Bedford, MA 01730 USA
>Phone: 1 781-275-5500 Fax: 1 781-275-3042
>
>http://marriott.com/property/propertyPage.mi?marshaCode=BOSSB
>
>Agenda:
>
>MONDAY - SIMPLE
>
>9:00-12:00 - MSRP/MSRP Relays
>12:00-13:00 - Lunch
>13:00-16:00 - XCAP
>16:00-16:30 - Filtering
>16:30-17:00 - Partial Publish
>17:00-17:30 - Advanced Messaging
>
>TUESDAY - SIP/SIPPING
>
>9:00-10:30 - KPML
>10:30-11:00 - GRUU
>11:00-12:00 - Session Policies
>12:00-13:00 - Lunch
>13:00-14:30 - Session Policies Cont.
>14:30-15:30 - Exploders
>time permitting:
>15:30-16:15 - Connected Party
>16:15-17:00 - Application Interaction
>17:00-17:30 - Dialog Package
>17:30-18:00 - Connection Re-use
>
>WEDNESDAY - XCON
>
>9:00-11:00 - Floor Control
>11:00-12:00 - CPCP
>12:00-13:00 - Lunch
>13:00-14:00 - CPCP Cont.
>14:00-16:00 - MPCP
>
>The chairs would like to thank Nokia for sponsoring the event.
>
>
>
>_______________________________________________
>IETF-Announce mailing list
>IETF-Announce@ietf.org
>https://www1.ietf.org/mailman/listinfo/ietf-announce


cheers,
James

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


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



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From: "Paul D.Smith" <Paul.D.Smith@dataconnection.com>
To: SIPPING WG <sipping@ietf.org>
Date: Fri, 23 Apr 2004 10:20:05 +0100
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Subject: [Sipping] FW: I-D ACTION:draft-smith-sip-auth-examples-00.txt
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Please note that I am aware that the name of this draft, and the indicated
WG, names the "SIP WG".  After discussion with the WG chairs, this draft is
offered to the "Sipping WG".  Later version of this draft will reflect the
Sipping WG.

Paul DS.

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: 22 April 2004 20:39
To: i-d-announce@ietf.org
Subject: I-D ACTION:draft-smith-sip-auth-examples-00.txt


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


	Title		: Digest Authentication Examples for Session
Initiation 
			  Protocol (SIP)
	Author(s)	: P. Smith, I. Clarkson
	Filename	: draft-smith-sip-auth-examples-00.txt
	Pages		: 29
	Date		: 2004-4-22
	
RFC3261 [2] recommends that SIP requests be authenticated using HTTP
   Digest authentication as defined in RFC2617 [3], and many vendors
   have successfully added this function to their products.  However
   implementation bugs are still being observed both at interoperability
   events and in the field.

   This document provides worked examples of Digest authentication usage
   in SIP.  It is intended to help new SIP implementers, and to provide
   a common set of test authentication examples against which an
   implementation may be checked.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-smith-sip-auth-examples-00.txt

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



From exim@www1.ietf.org  Mon Apr 26 18:07:47 2004
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From: "Yu, James" <james.yu@neustar.biz>
To: "'Deepak.Pant@utstar.com'" <Deepak.Pant@utstar.com>
Cc: sipping@ietf.org, iptel@ietf.org
Date: Mon, 26 Apr 2004 17:21:21 -0400
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Subject: [Sipping] RE: Question about draft-yu-tel-url-08.txt
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Deepak,

Sorry that I saw your questions just now.  Please see responses below.

James

-----Original Message-----
From: Deepak.Pant@utstar.com [mailto:Deepak.Pant@utstar.com]
Sent: Friday, March 19, 2004 8:53 AM
To: sipping@ietf.org; iptel@ietf.org; james.yu@neustar.biz
Subject: Question about draft-yu-tel-url-08.txt



I have the following questions about conveying carrier
identitification code in SIP.

On the ISUP side we get three information
1. Type of network identification
2. Network identification plan(3 digits or 4 digits)
3. Digits 1-3 or 1-4

Q1. The draft draft-yu-tel-url-08.txt provides a mechanism to convey the
digit
information on the sip network. However how do we convey the
type of network identification information on the SIP network.

[YU]  I guess that you were referring to the carrier selection info.
parameter.   I'm working with other co-authors to propose tel URI parameters
such as the dail around indicator (dai) that would indicate the type of cic.


Q2. On a UAS receiving an INVITE message with cic=+11234 or cic=1234,
what is the difference between the two cic information. Does this mean
something different needs to be conveyed on the ISUP side.

[YU]  If the cic does not begin with "+" it should have a "cic-content=+1"
if cic=1234.


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



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Date: Wed, 28 Apr 2004 08:05:44 -0500
From: Dean Willis <dean.willis@softarmor.com>
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 SIP)
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Moving to SIPPING, because that's where this belongs.

Following up on a previous thread:

Paul Kyzivat wrote:

 > Dean Willis wrote
>> Of course, I'm still deeply confused between:
>> 1) preference/capability negotiation enabled by offer/answer
>> 2) preference/capability negotiation enabled by callee-caps/caller-pref.
> 
> Forking and offer/answer provide a partial solution. But offer/answer 
> doesn't give you a way to say you don't want the call unless your 
> offered media are accepted. At least not an easy way. Conceivably some 
> new extension to preconditions could be made to handle this kind of 
> thing. Lacking that you run the risk of having an unsuitable device 
> answer, even if there is a suitable device present.
> 
> Callerprefs gives you an increased likelihood that you will end up with 
> a suitable device.

The above is a "good start".

One of the challenges I'm faced with regularly is helping SIP-using 
designers decide what to negotiate with O/A and what to negotiate with 
prefs/caps.

Someone even asked me yesterday why we don't just do everything with 
prefs/caps and deprecate O/A.

So, if somebody out there is looking for a meaningful task, it might be 
reasonable to undertake a clear explanation of the relationship and 
usage of these related mechanisms. Perhaps we should see what kind of 
consensus we have here on the list . . .

Here's a debate to consider: Let's say I want to make a voice 
push-to-talk call to somebody. This means I'm wanting to use 
push-to-talk behavior, and negotiate a floor control protocol and voice 
codec to be used within the call.

We seem to have consensus that the voice codec, port numbers, and IP 
addresses involved are to be described in SDP and negotiated in O/A 
mode. That's "traditional SIP".

How about the floor control protocol to use? Prefs/Caps or O/A? My 
personal guess is that this is O/A.

And here's the hard one -- Is the desire for push-to-talk behavior 
something that should be expressed as caller preferences and matched by 
the routing function to a termination whose callee-caps indicate 
compatibility, or is it something that should be described in an SDP 
offer/answer? Or something else entirely?

--
Dean





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



From exim@www1.ietf.org  Wed Apr 28 12:22:11 2004
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From: Gordon Ledgard <gledgard@iperia.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        Paul Kyzivat
	 <pkyzivat@cisco.com>, sipping@ietf.org
Subject: RE: [Sipping] Application identification with callee caps/caller 
	preferences (Was SIP)
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Isn't the difference between O/A and Pref/Caps simply
the difference between media negotiation (transport layer)
and application capability?

I would not have thought this to be a complicated distinction.

O/A says "I have all the hardware and infrastructure to support
this".

Pref/Caps says "I'm not set up to deal with this (I do not have 
certain features loaded) or I don't want to right now."

Life is a grey area. :)

Gordon Ledgard
IPeria, Inc.





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, April 28, 2004 9:06 AM
> To: Paul Kyzivat; sipping@ietf.org
> Subject: [Sipping] Application identification with callee caps/caller
> preferences (Was SIP)
> 
> 
> 
> Moving to SIPPING, because that's where this belongs.
> 
> Following up on a previous thread:
> 
> Paul Kyzivat wrote:
> 
>  > Dean Willis wrote
> >> Of course, I'm still deeply confused between:
> >> 1) preference/capability negotiation enabled by offer/answer
> >> 2) preference/capability negotiation enabled by 
> callee-caps/caller-pref.
> > 
> > Forking and offer/answer provide a partial solution. But 
> offer/answer 
> > doesn't give you a way to say you don't want the call unless your 
> > offered media are accepted. At least not an easy way. 
> Conceivably some 
> > new extension to preconditions could be made to handle this kind of 
> > thing. Lacking that you run the risk of having an unsuitable device 
> > answer, even if there is a suitable device present.
> > 
> > Callerprefs gives you an increased likelihood that you will 
> end up with 
> > a suitable device.
> 
> The above is a "good start".
> 
> One of the challenges I'm faced with regularly is helping SIP-using 
> designers decide what to negotiate with O/A and what to 
> negotiate with 
> prefs/caps.
> 
> Someone even asked me yesterday why we don't just do everything with 
> prefs/caps and deprecate O/A.
> 
> So, if somebody out there is looking for a meaningful task, 
> it might be 
> reasonable to undertake a clear explanation of the relationship and 
> usage of these related mechanisms. Perhaps we should see what kind of 
> consensus we have here on the list . . .
> 
> Here's a debate to consider: Let's say I want to make a voice 
> push-to-talk call to somebody. This means I'm wanting to use 
> push-to-talk behavior, and negotiate a floor control protocol 
> and voice 
> codec to be used within the call.
> 
> We seem to have consensus that the voice codec, port numbers, and IP 
> addresses involved are to be described in SDP and negotiated in O/A 
> mode. That's "traditional SIP".
> 
> How about the floor control protocol to use? Prefs/Caps or O/A? My 
> personal guess is that this is O/A.
> 
> And here's the hard one -- Is the desire for push-to-talk behavior 
> something that should be expressed as caller preferences and 
> matched by 
> the routing function to a termination whose callee-caps indicate 
> compatibility, or is it something that should be described in an SDP 
> offer/answer? Or something else entirely?
> 
> --
> Dean
> 
> 
> 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

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



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Date: Wed, 28 Apr 2004 11:33:13 -0400
From: David Oran <oran@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>,
        sipping@ietf.org
Subject: Re: [Sipping] Application identification with callee caps/caller
 preferences (Was SIP)
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References: <0C1353ABB1DEB74DB067ADFF749C4EEF029E151D@esebe004.ntc.nokia.com
 > <F586CA84FE30BD374874DF77@[10.32.245.156]> <408E841B.70900@softarmor.com>
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--On Wednesday, April 28, 2004 8:05 AM -0500 Dean Willis 
<dean.willis@softarmor.com> wrote:

>
[snip]
> Here's a debate to consider: Let's say I want to make a voice
> push-to-talk call to somebody. This means I'm wanting to use push-to-talk
> behavior, and negotiate a floor control protocol and voice codec to be
> used within the call.
>
> We seem to have consensus that the voice codec, port numbers, and IP
> addresses involved are to be described in SDP and negotiated in O/A mode.
> That's "traditional SIP".
>
and traditional everything else multimedia as well...

> How about the floor control protocol to use? Prefs/Caps or O/A? My
> personal guess is that this is O/A.
>
I'd say O/A as well, since for me the dividing line is:
- am I singaling WHAT media I want to do or can do: Callerprefs/calleecaps
- am I signaling HOW I can do it (codec, protocol, etc.): O/A

So, I guess I'd express the need to a cnference with floor control through 
callerprefs and negotiate the floor control protocol and parameters thereof 
with O/A.

> And here's the hard one -- Is the desire for push-to-talk behavior
> something that should be expressed as caller preferences and matched by
> the routing function to a termination whose callee-caps indicate
> compatibility, or is it something that should be described in an SDP
> offer/answer? Or something else entirely?
>
What exactly is "push-to-talk" as an application? Is it a half-duplex 
session? If so, I suspect I'd signal that I want a half-duplex session in 
the prefs/caps. Whether push-to-talk is just a form of floor control is, I 
think, the interesting design question.

Dave.

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





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



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Date: Wed, 28 Apr 2004 11:34:57 -0500
From: Alan Johnston <alan.johnston@mci.com>
Subject: Re: [Sipping] Application identification with callee  caps/caller
 preferences (Was SIP)
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        Paul Kyzivat <pkyzivat@cisco.com>, sipping@ietf.org
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At 11:33 AM 4/28/2004 -0400, David Oran wrote:


>--On Wednesday, April 28, 2004 8:05 AM -0500 Dean Willis 
><dean.willis@softarmor.com> wrote:
>
>[snip]
>>Here's a debate to consider: Let's say I want to make a voice
>>push-to-talk call to somebody. This means I'm wanting to use push-to-talk
>>behavior, and negotiate a floor control protocol and voice codec to be
>>used within the call.
>>
>>We seem to have consensus that the voice codec, port numbers, and IP
>>addresses involved are to be described in SDP and negotiated in O/A mode.
>>That's "traditional SIP".
>and traditional everything else multimedia as well...
>
>>How about the floor control protocol to use? Prefs/Caps or O/A? My
>>personal guess is that this is O/A.
>I'd say O/A as well, since for me the dividing line is:
>- am I singaling WHAT media I want to do or can do: Callerprefs/calleecaps
>- am I signaling HOW I can do it (codec, protocol, etc.): O/A
>
>So, I guess I'd express the need to a cnference with floor control through 
>callerprefs and negotiate the floor control protocol and parameters 
>thereof with O/A.

It depends.

SIP is often used to establish multimedia sessions, which result in RTP 
flows.  If the floor control "session" is established using SIP, then O/A 
applies.

However, if floor control is not established through SIP but is a parallel 
control stream, then what is needed is a conference focus that supports the 
required floor control protocol, and an ability for the focus to provide a 
URI to the user to establish this separate channel.

In this case, then a Require/420 exchange is appropriate.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

>>And here's the hard one -- Is the desire for push-to-talk behavior
>>something that should be expressed as caller preferences and matched by
>>the routing function to a termination whose callee-caps indicate
>>compatibility, or is it something that should be described in an SDP
>>offer/answer? Or something else entirely?
>What exactly is "push-to-talk" as an application? Is it a half-duplex 
>session? If so, I suspect I'd signal that I want a half-duplex session in 
>the prefs/caps. Whether push-to-talk is just a form of floor control is, I 
>think, the interesting design question.
>
>Dave.
>
>>--
>>Dean
>>
>>
>>
>>
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sip@ietf.org for new developments of core SIP
>
>
>
>
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


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



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Date: Wed, 28 Apr 2004 12:49:53 -0400
From: David Oran <oran@cisco.com>
To: Alan Johnston <alan.johnston@mci.com>,
        Dean Willis <dean.willis@softarmor.com>,
        Paul Kyzivat <pkyzivat@cisco.com>, sipping@ietf.org
Subject: Re: [Sipping] Application identification with callee  caps/caller
 preferences (Was SIP)
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--On Wednesday, April 28, 2004 11:34 AM -0500 Alan Johnston 
<alan.johnston@mci.com> wrote:

> At 11:33 AM 4/28/2004 -0400, David Oran wrote:
>
[snip]
>> So, I guess I'd express the need to a cnference with floor control
>> through  callerprefs and negotiate the floor control protocol and
>> parameters  thereof with O/A.
>
> It depends.
>
> SIP is often used to establish multimedia sessions, which result in RTP
> flows.  If the floor control "session" is established using SIP, then O/A
> applies.
>
Sure. I agree.

> However, if floor control is not established through SIP but is a
> parallel control stream, then what is needed is a conference focus that
> supports the required floor control protocol, and an ability for the
> focus to provide a URI to the user to establish this separate channel.
>
Yes, I agree.

> In this case, then a Require/420 exchange is appropriate.
>
It seems that if the two protocols are "parallel" then I'm not sure SIP 
should have any say on how the state synchronization is done. Using 
requires/420 is one possibility, of course, an alternative might be using 
SDP preconditions.

Dave.

> Thanks,
> Alan Johnston
> MCI
> sip:alan@sipstation.com
>
>>> And here's the hard one -- Is the desire for push-to-talk behavior
>>> something that should be expressed as caller preferences and matched by
>>> the routing function to a termination whose callee-caps indicate
>>> compatibility, or is it something that should be described in an SDP
>>> offer/answer? Or something else entirely?
>> What exactly is "push-to-talk" as an application? Is it a half-duplex
>> session? If so, I suspect I'd signal that I want a half-duplex session
>> in  the prefs/caps. Whether push-to-talk is just a form of floor control
>> is, I  think, the interesting design question.
>>
>> Dave.
>>
>>> --
>>> Dean
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sip@ietf.org for new developments of core SIP
>>
>>
>>
>>
>>
>> _______________________________________________
>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>> This list is for NEW development of the application of SIP
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sip@ietf.org for new developments of core SIP
>





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



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Subject: [Sipping] Re: Application identification with callee caps/caller preferences
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Dean Willis wrote:
> 
> Moving to SIPPING, because that's where this belongs.
> 
> Following up on a previous thread:
> 
> Paul Kyzivat wrote:
> 
>  > Dean Willis wrote
> 
>>> Of course, I'm still deeply confused between:
>>> 1) preference/capability negotiation enabled by offer/answer
>>> 2) preference/capability negotiation enabled by callee-caps/caller-pref.
>>
>>
>> Forking and offer/answer provide a partial solution. But offer/answer 
>> doesn't give you a way to say you don't want the call unless your 
>> offered media are accepted. At least not an easy way. Conceivably some 
>> new extension to preconditions could be made to handle this kind of 
>> thing. Lacking that you run the risk of having an unsuitable device 
>> answer, even if there is a suitable device present.
>>
>> Callerprefs gives you an increased likelihood that you will end up 
>> with a suitable device.
> 
> 
> The above is a "good start".
> 
> One of the challenges I'm faced with regularly is helping SIP-using 
> designers decide what to negotiate with O/A and what to negotiate with 
> prefs/caps.
> 
> Someone even asked me yesterday why we don't just do everything with 
> prefs/caps and deprecate O/A.

Well, that would be silly. Prefs/caps don't provide sufficient detail to 
establish a media session.

Clearly prefs/caps is the way to go for things that aren't expressed at 
all in O/A - things that don't have to do with the media streams themselves.

There is an area of overlap around media-related things. IMO it would be 
better to enhance O/A to cover those cases. For instance, saying "I only 
want a session if I can have both audio and video", or "I want an audio 
session but I'm not yet prepared to tell you where to send it - please 
respond, if you can, with an audio offer".

This sort of thing *could* be done with preconditions. But there seems 
to be a (justifiable) reluctance to invent new uses for preconditions. 
(Or even to use the existing ones.)

Prefs/caps provides *some" of this functionality using what is perceived 
to be a lighter weight mechanism. In the absence of something better, it 
is reasonable to use it.

> So, if somebody out there is looking for a meaningful task, it might be 
> reasonable to undertake a clear explanation of the relationship and 
> usage of these related mechanisms. Perhaps we should see what kind of 
> consensus we have here on the list . . .

Does the above help?

> Here's a debate to consider: Let's say I want to make a voice 
> push-to-talk call to somebody. This means I'm wanting to use 
> push-to-talk behavior, and negotiate a floor control protocol and voice 
> codec to be used within the call.
> 
> We seem to have consensus that the voice codec, port numbers, and IP 
> addresses involved are to be described in SDP and negotiated in O/A 
> mode. That's "traditional SIP".
> 
> How about the floor control protocol to use? Prefs/Caps or O/A? My 
> personal guess is that this is O/A.

I need to know more about how the floor control is intended to work 
before I would begin to answer this.

> And here's the hard one -- Is the desire for push-to-talk behavior 
> something that should be expressed as caller preferences and matched by 
> the routing function to a termination whose callee-caps indicate 
> compatibility, or is it something that should be described in an SDP 
> offer/answer? Or something else entirely?

I don't think this can be done entirely with O/A if you really require 
the push-to-talk behavior, at least for any plausible definition of 
push-to-talk that I can imagine. I think anything you might offer would 
be answered by a regular sip phone, ignoring/refusing the push-to-talk 
features.

	Paul


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



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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sipping] Re: WGLC for draft-ietf-sipping-transc-3pcc-00
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I'd like to announce working group last call for the 3PCC transcoding
draft:

     draft-ietf-sipping-transc-3pcc-00.txt

This last call period starts immediately and will run through May 15, 2004.

There are two known minor textual issues with this draft which will be
corrected following the WGLC comments:

1) Split the references into normative and non-normative.

2) Correct the example domains -- s/domain.com/example.com.

Please review this draft and get your comments, if any, back to the
author (and to the list, if reasonable) as soon as possible.

As I understand it, we'll be requesting that this draft be published as 
an informational RFC. The procedure for informationals seems to be 
evolving, and my understanding is that they will now receive 
significantly less scrutiny from the IESG. So, we need to make sure we 
do a VERY thorough job of reviewing this in the working group.

Thanks,

-- 
Dean Willis
SIPPING WG co-chair



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



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Subject: Re: [Sipping] Re: WGLC for draft-ietf-sipping-transc-3pcc-00
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Somehow I missed this draft altogether. (Asleep at the wheel I guess.)

I see a couple of limitations in the approaches recommended, and have 
some ideas for how to fix them:

3.2 (Figure 2):

This may be appropriate if B wants to hide its actual capabilities and 
expose T's instead. But suppose A is also deaf and also prefers text? In 
that case you might end up with two transcoders in the call. (Although 
there is no flow here that would cause that.) Also, the transcoding 
service may cost something so it would be good to delay requesting it 
until it is known to be needed.

I understand that there is a desire to offer voice, because without that 
a call may never be established a dumb voice peer.

As an alternative, if T is flexible in what codecs it can use, and B 
knows what those are, it can respons with an initial offer that includes 
a text stream as well as a voice stream that is black-holed. (C=0). That 
doesn't actually permit voice communication but offers voice and gives a 
a chance to pick what it wants. And it doesn't yet bother T.

Then, if A's answer accepts the voice, B will invite T and then reinvite 
A to splice the streams. If A accepts text and refuses voice, all is cool.

3.3 (Figure 3):

This again will be suboptimal if B prefers text. A perhaps more serious 
problem is that the handshake between A and T may unnecessarily restrict 
the set of codecs that A offers B. A may offer a bunch to T, but T may 
only be able to use one at a time, and so may restrict its answer to a 
single codec even though it could also do one of the others.

Both of these issues could be solved using the same tactic I just 
suggested above - offering test and black-holed voice. This however 
might result in clipping B, who doesn't know the situation and is likely 
to start talking immediately. This could be dealt with using PRACK and 
UPDATE to refine the media stream selections before the call is answered.

An alternative that works reasonably well with my suggetion above is for 
A to initiate the call with an offerless invite. If the offer that comes 
back contains only voice, then A can invite T at that time.

	Paul


Dean Willis wrote:
> 
> I'd like to announce working group last call for the 3PCC transcoding
> draft:
> 
>     draft-ietf-sipping-transc-3pcc-00.txt
> 
> This last call period starts immediately and will run through May 15, 2004.
> 
> There are two known minor textual issues with this draft which will be
> corrected following the WGLC comments:
> 
> 1) Split the references into normative and non-normative.
> 
> 2) Correct the example domains -- s/domain.com/example.com.
> 
> Please review this draft and get your comments, if any, back to the
> author (and to the list, if reasonable) as soon as possible.
> 
> As I understand it, we'll be requesting that this draft be published as 
> an informational RFC. The procedure for informationals seems to be 
> evolving, and my understanding is that they will now receive 
> significantly less scrutiny from the IESG. So, we need to make sure we 
> do a VERY thorough job of reviewing this in the working group.
> 
> Thanks,
> 


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



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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
CC: Dean Willis <dean.willis@softarmor.com>, sipping@ietf.org,
        Rohan Mahy <rohan@cisco.com>, Allison Mankin <mankin@psg.com>
Subject: Re: [Sipping] Re: WGLC for draft-ietf-sipping-transc-3pcc-00
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Paul,

thanks for your comments.

The flows you are referring to describe how to invoke a transcoder by 
the caller and by the callee. Nevertheless, this document does not 
recommend *when* you have to invoke it. This is dealt with by the 
framework document: draft-ietf-sipping-transc-framework-00.txt

In particular, the framework says:

" It is recommended that an offerer does not invoke transcoding
    services before making sure that the answerer does not support the
    capabilities needed for the session. Making wrong assumptions about
    the answerer's capabilities can lead to situations where two
    transcoders are introduced (one by the offerer and one by the
    answerer) in a session that would not need any transcoding services
    at all."

I guess the paragraph above addresses your concern. In any event, if you 
want to see more explanatory text included in any of these drafts, let 
me know.

Thanks,

Gonzalo

Paul Kyzivat wrote:

> Somehow I missed this draft altogether. (Asleep at the wheel I guess.)
> 
> I see a couple of limitations in the approaches recommended, and have 
> some ideas for how to fix them:
> 
> 3.2 (Figure 2):
> 
> This may be appropriate if B wants to hide its actual capabilities and 
> expose T's instead. But suppose A is also deaf and also prefers text? In 
> that case you might end up with two transcoders in the call. (Although 
> there is no flow here that would cause that.) Also, the transcoding 
> service may cost something so it would be good to delay requesting it 
> until it is known to be needed.
> 
> I understand that there is a desire to offer voice, because without that 
> a call may never be established a dumb voice peer.
> 
> As an alternative, if T is flexible in what codecs it can use, and B 
> knows what those are, it can respons with an initial offer that includes 
> a text stream as well as a voice stream that is black-holed. (C=0). That 
> doesn't actually permit voice communication but offers voice and gives a 
> a chance to pick what it wants. And it doesn't yet bother T.
> 
> Then, if A's answer accepts the voice, B will invite T and then reinvite 
> A to splice the streams. If A accepts text and refuses voice, all is cool.
> 
> 3.3 (Figure 3):
> 
> This again will be suboptimal if B prefers text. A perhaps more serious 
> problem is that the handshake between A and T may unnecessarily restrict 
> the set of codecs that A offers B. A may offer a bunch to T, but T may 
> only be able to use one at a time, and so may restrict its answer to a 
> single codec even though it could also do one of the others.
> 
> Both of these issues could be solved using the same tactic I just 
> suggested above - offering test and black-holed voice. This however 
> might result in clipping B, who doesn't know the situation and is likely 
> to start talking immediately. This could be dealt with using PRACK and 
> UPDATE to refine the media stream selections before the call is answered.
> 
> An alternative that works reasonably well with my suggetion above is for 
> A to initiate the call with an offerless invite. If the offer that comes 
> back contains only voice, then A can invite T at that time.
> 
>     Paul
> 
> 
> Dean Willis wrote:
> 
>>
>> I'd like to announce working group last call for the 3PCC transcoding
>> draft:
>>
>>     draft-ietf-sipping-transc-3pcc-00.txt
>>
>> This last call period starts immediately and will run through May 15, 
>> 2004.
>>
>> There are two known minor textual issues with this draft which will be
>> corrected following the WGLC comments:
>>
>> 1) Split the references into normative and non-normative.
>>
>> 2) Correct the example domains -- s/domain.com/example.com.
>>
>> Please review this draft and get your comments, if any, back to the
>> author (and to the list, if reasonable) as soon as possible.
>>
>> As I understand it, we'll be requesting that this draft be published 
>> as an informational RFC. The procedure for informationals seems to be 
>> evolving, and my understanding is that they will now receive 
>> significantly less scrutiny from the IESG. So, we need to make sure we 
>> do a VERY thorough job of reviewing this in the working group.
>>
>> Thanks,
>>

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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Cc: Dean Willis <dean.willis@softarmor.com>,
        sipping (E-mail) <sipping@ietf.org>, Rohan Mahy <rohan@cisco.com>,
        Hisham Khartabil <hisham.khartabil@nokia.com>,
        Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
From: Rohan Mahy <rohan@cisco.com>
Date: Thu, 29 Apr 2004 12:59:05 -0700
To: "James M. Polk" <jmpolk@cisco.com>
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Subject: [Sipping] Re: SIMPLE/SIP/SIPPING/XCON Working Groups Joint Interim Meeting Announcement
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Hi,

Narrowing this to just SIPPING...

For E2M:  I believe that we will discuss the most fundamental issue of 
E2M (piggyback vs. redirect) at the interim as part of Session Policy.  
If we decide this issue, there will be much more work that needs to be 
done before the WG as a whole can comment on the specifics of any 
proposed mechanisms.

For Emergency: I don't think the working group will be familiar enough 
with the output of the emergency design team to have a productive 
discussion at the interim. There were more than a dozen open issues in 
Korea that the design team had not yet addressed; we should get to the 
point where the design team has something like 2 or 4 open issues 
before we use significant meeting time with the WG.

thanks,
-rohan


On Apr 22, 2004, at 12:51 PM, James M. Polk wrote:

> What happened to time for Emergency and E2M Security? In Seoul, these 
> were mentioned as two of the prevailing topic areas for the interim 
> (along with Session Policy and Exploders).
>
> At 11:31 AM 4/22/2004 -0400, Hisham Khartabil wrote:
>> An joint interim meeting for the SIMPLE, SIP, SIPPING and XCON 
>> working groups
>> will be help from Monday 24th May until Wednesday 26th May in Boston, 
>> Massachusetts.
>>
>> Location:
>> Renaissance Boston Hotel Bedford
>> 44 Middlesex Turnpike Bedford, MA 01730 USA
>> Phone: 1 781-275-5500 Fax: 1 781-275-3042
>>
>> http://marriott.com/property/propertyPage.mi?marshaCode=BOSSB
>>
>> Agenda:
>>
>> MONDAY - SIMPLE
>>
>> 9:00-12:00 - MSRP/MSRP Relays
>> 12:00-13:00 - Lunch
>> 13:00-16:00 - XCAP
>> 16:00-16:30 - Filtering
>> 16:30-17:00 - Partial Publish
>> 17:00-17:30 - Advanced Messaging
>>
>> TUESDAY - SIP/SIPPING
>>
>> 9:00-10:30 - KPML
>> 10:30-11:00 - GRUU
>> 11:00-12:00 - Session Policies
>> 12:00-13:00 - Lunch
>> 13:00-14:30 - Session Policies Cont.
>> 14:30-15:30 - Exploders
>> time permitting:
>> 15:30-16:15 - Connected Party
>> 16:15-17:00 - Application Interaction
>> 17:00-17:30 - Dialog Package
>> 17:30-18:00 - Connection Re-use
>>
>> WEDNESDAY - XCON
>>
>> 9:00-11:00 - Floor Control
>> 11:00-12:00 - CPCP
>> 12:00-13:00 - Lunch
>> 13:00-14:00 - CPCP Cont.
>> 14:00-16:00 - MPCP
>>
>> The chairs would like to thank Nokia for sponsoring the event.
>>
>>
>>
>> _______________________________________________
>> IETF-Announce mailing list
>> IETF-Announce@ietf.org
>> https://www1.ietf.org/mailman/listinfo/ietf-announce
>
>
> cheers,
> James
>
>                                *******************
>                 Truth is not to be argued... it is to be presented
>


_______________________________________________
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This list is for NEW development of the application of SIP
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Subject: Re: [Sipping] Re: WGLC for draft-ietf-sipping-transc-3pcc-00
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Gonzalo Camarillo wrote:
> Paul,
> 
> thanks for your comments.
> 
> The flows you are referring to describe how to invoke a transcoder by 
> the caller and by the callee. Nevertheless, this document does not 
> recommend *when* you have to invoke it. This is dealt with by the 
> framework document: draft-ietf-sipping-transc-framework-00.txt
> 
> In particular, the framework says:
> 
> " It is recommended that an offerer does not invoke transcoding
>    services before making sure that the answerer does not support the
>    capabilities needed for the session. Making wrong assumptions about
>    the answerer's capabilities can lead to situations where two
>    transcoders are introduced (one by the offerer and one by the
>    answerer) in a session that would not need any transcoding services
>    at all."
> 
> I guess the paragraph above addresses your concern.

I don't know. Putting it in the framework doesn't make it happen. It has 
to be in something else other than just the framework before it becomes 
functional.

I suppose you could view this as a problem for the human operator of the 
device - to select an appropriate phone feature based on known 
capabilities of the callee. But that is pretty much a cop out. And it 
doesn't work when the decision has to be made  by the callee, as when 
the initial invite has no offer, unless you expect the callee to select 
different answer modes depending on who is calling.

I would expect that the stuff you quote from the framework would appear 
as steps in the call flow that determine those capabilities.

Also, the framework has nothing to do with the codec restriction problem.

 > In any event, if you
> want to see more explanatory text included in any of these drafts, let 
> me know.

This isn't a hot button of mine. If nobody else is bothered then I will 
be happy to leave it be. This isn't normative stuff anyway.

	Paul

> Thanks,
> 
> Gonzalo
> 
> Paul Kyzivat wrote:
> 
>> Somehow I missed this draft altogether. (Asleep at the wheel I guess.)
>>
>> I see a couple of limitations in the approaches recommended, and have 
>> some ideas for how to fix them:
>>
>> 3.2 (Figure 2):
>>
>> This may be appropriate if B wants to hide its actual capabilities and 
>> expose T's instead. But suppose A is also deaf and also prefers text? 
>> In that case you might end up with two transcoders in the call. 
>> (Although there is no flow here that would cause that.) Also, the 
>> transcoding service may cost something so it would be good to delay 
>> requesting it until it is known to be needed.
>>
>> I understand that there is a desire to offer voice, because without 
>> that a call may never be established a dumb voice peer.
>>
>> As an alternative, if T is flexible in what codecs it can use, and B 
>> knows what those are, it can respons with an initial offer that 
>> includes a text stream as well as a voice stream that is black-holed. 
>> (C=0). That doesn't actually permit voice communication but offers 
>> voice and gives a a chance to pick what it wants. And it doesn't yet 
>> bother T.
>>
>> Then, if A's answer accepts the voice, B will invite T and then 
>> reinvite A to splice the streams. If A accepts text and refuses voice, 
>> all is cool.
>>
>> 3.3 (Figure 3):
>>
>> This again will be suboptimal if B prefers text. A perhaps more 
>> serious problem is that the handshake between A and T may 
>> unnecessarily restrict the set of codecs that A offers B. A may offer 
>> a bunch to T, but T may only be able to use one at a time, and so may 
>> restrict its answer to a single codec even though it could also do one 
>> of the others.
>>
>> Both of these issues could be solved using the same tactic I just 
>> suggested above - offering test and black-holed voice. This however 
>> might result in clipping B, who doesn't know the situation and is 
>> likely to start talking immediately. This could be dealt with using 
>> PRACK and UPDATE to refine the media stream selections before the call 
>> is answered.
>>
>> An alternative that works reasonably well with my suggetion above is 
>> for A to initiate the call with an offerless invite. If the offer that 
>> comes back contains only voice, then A can invite T at that time.
>>
>>     Paul
>>
>>
>> Dean Willis wrote:
>>
>>>
>>> I'd like to announce working group last call for the 3PCC transcoding
>>> draft:
>>>
>>>     draft-ietf-sipping-transc-3pcc-00.txt
>>>
>>> This last call period starts immediately and will run through May 15, 
>>> 2004.
>>>
>>> There are two known minor textual issues with this draft which will be
>>> corrected following the WGLC comments:
>>>
>>> 1) Split the references into normative and non-normative.
>>>
>>> 2) Correct the example domains -- s/domain.com/example.com.
>>>
>>> Please review this draft and get your comments, if any, back to the
>>> author (and to the list, if reasonable) as soon as possible.
>>>
>>> As I understand it, we'll be requesting that this draft be published 
>>> as an informational RFC. The procedure for informationals seems to be 
>>> evolving, and my understanding is that they will now receive 
>>> significantly less scrutiny from the IESG. So, we need to make sure 
>>> we do a VERY thorough job of reviewing this in the working group.
>>>
>>> Thanks,
>>>
> 


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From: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sipping] Re: WGLC for draft-ietf-sipping-transc-3pcc-00
References: <408F9D5F.6020407@softarmor.com> <4090B7A5.4020607@softarmor.com> <40913C65.3050305@cisco.com> <409152F0.70209@ericsson.com> <40917F16.90604@cisco.com>
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Paul Kyzivat wrote:

Gonzalo quoted:
>> " It is recommended that an offerer does not invoke transcoding
>>    services before making sure that the answerer does not support the
>>    capabilities needed for the session. Making wrong assumptions about
>>    the answerer's capabilities can lead to situations where two
>>    transcoders are introduced (one by the offerer and one by the
>>    answerer) in a session that would not need any transcoding services
>>    at all."
...

Paul pondered:
> I suppose you could view this as a problem for the human operator of the 
> device - to select an appropriate phone feature based on known 
> capabilities of the callee. But that is pretty much a cop out. And it 
> doesn't work when the decision has to be made  by the callee, as when 
> the initial invite has no offer, unless you expect the callee to select 
> different answer modes depending on who is calling.

Would something like a priority-ordered preference/capability set in 
caller-preferences/callee caps be a big aid here?

--
Dean

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Dean Willis wrote:
> 
> Paul pondered:

`:-)

>> I suppose you could view this as a problem for the human operator of 
>> the device - to select an appropriate phone feature based on known 
>> capabilities of the callee. But that is pretty much a cop out. And it 
>> doesn't work when the decision has to be made  by the callee, as when 
>> the initial invite has no offer, unless you expect the callee to 
>> select different answer modes depending on who is calling.
> 
> Would something like a priority-ordered preference/capability set in 
> caller-preferences/callee caps be a big aid here?

Well, the normal use of callerprefs is to select among alternative 
advertised contacts. That doesn't seem to be the case here. Of course an 
endpoint *could* choose to alter its behavior based on callerprefs, but 
I don't know if we want to suggest that.

However I suppose a variant is possible. An endpoint with these 
capabilities could register multiple contacts, tagged with capabilities 
for voice and text. Depending upon which one was called, it could do the 
right thing. And it could prioritize its native capabilities higher, so 
that a forking proxy would try that first even if the caller expresses 
no preferences.

If that were done, when a call arrives at the contact associated with 
text, only text would be offered. When a call arrives at the contact 
associated with voice then all the games described in this draft would 
be carried out.

That sounds like an interesting approach.

	Paul


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