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From: Itamar Gilad <ItamarG@tlv.radvision.com>
To: "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Call services
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Am I right in assuming that draf-ietf-mmusic-sip-cc-01.txt ('SIP Call
Control Services") and draft-sparks-sip-service-examples-00.txt ("SIP
Telephony Service Examples With Call Flows") are the only authoritative
documents describing call services in SIP?  Any others?
 
Thanks
 
   Itamar Gilad
  

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<DIV><FONT face=Arial size=2><SPAN class=058334209-02012000>Am I right in 
assuming that draf-ietf-mmusic-sip-cc-01.txt ('SIP Call Control Services") and 
draft-sparks-sip-service-examples-00.txt ("SIP Telephony Service Examples With 
Call Flows") are the only authoritative documents describing call services in 
SIP?&nbsp; Any others?</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN 
class=058334209-02012000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN 
class=058334209-02012000>Thanks</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN 
class=058334209-02012000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=Arial size=2><SPAN class=058334209-02012000>&nbsp;&nbsp; Itamar 
Gilad</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=058334209-02012000>&nbsp; 
</SPAN></FONT></DIV></BODY></HTML>

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From owner-sip-outgoing@lists.research.bell-labs.com  Sun Jan  2 09:31:57 2000
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Date: Sun, 02 Jan 2000 08:28:53 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: Call services
In-reply-to: <00Jan2.105532ist.115202@firewall.tlv.radvision.com>
To: Itamar Gilad <ItamarG@tlv.radvision.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
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See also johnston-sip-call flows. A pdf. version is available at
http://www.greycouncil.com/sip/drafts/draft-johnston-sip-call-flows-00.pdf

There is also a draft on multi-proxy authorization at

 http://www.greycouncil.com/sip/drafts/

Henry
  -----Original Message-----
  From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Itamar Gilad
  Sent: Sunday, January 02, 2000 2:56 AM
  To: 'SIP List'
  Subject: Call services


  Am I right in assuming that draf-ietf-mmusic-sip-cc-01.txt ('SIP Call
Control Services") and draft-sparks-sip-service-examples-00.txt ("SIP
Telephony Service Examples With Call Flows") are the only authoritative
documents describing call services in SIP?  Any others?

  Thanks

     Itamar Gilad


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<META content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D910311614-02012000>See=20
also johnston-sip-call flows. A pdf. version is available =
at</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><A=20
href=3D"http://www.greycouncil.com/sip/drafts/draft-johnston-sip-call-flo=
ws-00.pdf">http://www.greycouncil.com/sip/drafts/draft-johnston-sip-call-=
flows-00.pdf</A></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D910311614-02012000>There=20
is also a draft on multi-proxy authorization at</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D910311614-02012000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D910311614-02012000>&nbsp;<A=20
href=3D"http://www.greycouncil.com/sip/drafts/">http://www.greycouncil.co=
m/sip/drafts/</A></SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D910311614-02012000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D910311614-02012000>Henry</SPAN></FONT></DIV>
<BLOCKQUOTE style=3D"MARGIN-RIGHT: 0px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B>=20
  owner-sip@lists.research.bell-labs.com=20
  [mailto:owner-sip@lists.research.bell-labs.com]<B>On Behalf Of =
</B>Itamar=20
  Gilad<BR><B>Sent:</B> Sunday, January 02, 2000 2:56 AM<BR><B>To:</B> =
'SIP=20
  List'<BR><B>Subject:</B> Call services<BR><BR></DIV></FONT>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D058334209-02012000>Am I =
right in=20
  assuming that draf-ietf-mmusic-sip-cc-01.txt ('SIP Call Control =
Services") and=20
  draft-sparks-sip-service-examples-00.txt ("SIP Telephony Service =
Examples With=20
  Call Flows") are the only authoritative documents describing call =
services in=20
  SIP?&nbsp; Any others?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D058334209-02012000></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D058334209-02012000>Thanks</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D058334209-02012000></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D058334209-02012000>&nbsp;&nbsp;=20
  Itamar Gilad</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D058334209-02012000>&nbsp;=20
  </SPAN></FONT></DIV></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_zmDJoW2w5ltG6YJMB/ezVw)--



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Cc: Scott Petrack <scott.petrack@metatel.com>,
        Henning Schulzrinne <Schulz-Rinne@t-online.de>,
        "Daniel G. Petrie" <dpetrie@pingtel.com>,
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Subject: Re: JPEG in INVITE
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Jonathan Rosenberg writes:
 > The issue is, if the content is referenced by indirection, is it better
 > to be in the body or in a header. Henning is arguing in favor of a
 > header. I agree with him that this is simpler. Both he and Dan are also
 > arguing that we need a way to identify the purpose of the content.
 > Providing such data also argues for it being a header. My concern,
 > though, is that there are so many uses for displaying of content, we
 > have no hope of enumerating them all. I suppose we could enumerate a few
 > and leave others to future standardization. Also, using a URI list in
 > the body is something thats doable now (since we support MIME already),
 > whereas a new header is yet-another-extension that I'd rather avoid. The
 > notion of a generic "body contains content to display", with the body
 > either in-line or referenced via a URI list, is appealing to me because
 > of its generality. Yes, we could have both, but then this seems to make
 > things even more complicated for a designer that now has to choose
 > whether to use the header or the indirect content in the body.

   With UDP, fragmentation, etc, vs other MIME
   headers would they not always have to choose?
 
 > Am I the only one who prefers the indirect references in the body
 > approach?

   I tend to think that URI is a much better
   method overall since the receiving side can
   make a determination of whether they want the 
   content or not, and the timing of fetching 
   the content. However, one thing to be said is 
   that absent an existing IPSEC session between 
   the two hosts (or possibly a third party),
   there will be delay to pay for the indirect
   method. Maybe this extra delay is managable for
   things like X-faces though.

	       Mike



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan  3 00:24:58 2000
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Many services are possible without having to explicitly define how they
are done. A classic example is call screening. There are many varieties
of this serivice (caller based, time of day based, priority based), all
of which are possible in SIP, but none of which I think are described in
the documents you mention. It only requires the server to reject the
call based on some logic. 

-Jonathan R.

> Itamar Gilad wrote:
> 
> Am I right in assuming that draf-ietf-mmusic-sip-cc-01.txt ('SIP Call
> Control Services") and draft-sparks-sip-service-examples-00.txt ("SIP
> Telephony Service Examples With Call Flows") are the only
> authoritative documents describing call services in SIP?  Any others?
> 
> Thanks
> 
>    Itamar Gilad
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hi:

It is given in the SIP spec, that retrans for INVITE stops when either a
definitive or provisioan response is received.  My question is on
provisional response. If the caller gets a 100 Trying response,, which
is essentially that the user has not yet been contacted, should the
retrans. stop?

Thanks.

--
Sunitha Kumar



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi:
<p>It is given in the SIP spec, that retrans for INVITE stops when either
a definitive or provisioan response is received.&nbsp; My question is on
provisional response. If the caller gets a 100 Trying response,, which
is essentially that the user has not yet been contacted, should the retrans.
stop?
<p>Thanks.
<pre>--&nbsp;
Sunitha Kumar
</pre>
&nbsp;</html>

--------------C9A1664D9A0F8A488B4B0348--




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Yes. Since request reliability is hop by hop, the next hop will
retransmit the request to achieve end to end reliability. A UA should
stop retransmitting an INVITE when it receives any response to it.

-Jonathan R.

Sunitha Kumar wrote:
> 
> Hi:
> 
> It is given in the SIP spec, that retrans for INVITE stops when either
> a definitive or provisioan response is received.  My question is on
> provisional response. If the caller gets a 100 Trying response,, which
> is essentially that the user has not yet been contacted, should the
> retrans. stop?
> 
> Thanks.
> 
> --
> Sunitha Kumar
> 
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: JPEG in INVITE
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Tue, 4 Jan 2000 14:07:39 -0600 (CST)
Cc: eussean@exu.ericsson.se (Sean Olson),
        scott.petrack@metatel.com (Scott Petrack), schulzrinne@cs.columbia.edu,
        Rachid.Taji@srit.siemens.fr, Alvir.Alisic@ecs.ericsson.se,
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In-Reply-To: <3867BEC6.52B1A1AD@dynamicsoft.com> from "Jonathan Rosenberg" at Dec 27, 1999 02:32:22 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>There are some definite problems with this, but interestingly its more
>or less conceptually how the PSTN guarantees caller ID service (sans the
>crypto, of course). The service provider verifies the identity of the
>caller's terminal based on the physical wires the call comes in on, and
>inserts the name of the caller. This name is assumed known from offline
>administrative procedures.

...which actually aren't even as secure as you suggest. I could call
up Southwestern Bell and tell them that the name I want to go out
when I call people is "Albert Einstein," "Superman," or "Jonathan
Rosenberg." There is no verification of ID with caller ID.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Subject: Re: URL Generalization for Portability (was TRIP)
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Tue, 4 Jan 2000 13:49:40 -0600 (CST)
Cc: Adam.Roach@ericsson.com (Adam B. Roach), dean.willis@wcom.com,
        sip@lists.research.bell-labs.com
In-Reply-To: <385AB575.60B1CFCD@dynamicsoft.com> from "Jonathan Rosenberg" at Dec 17, 1999 05:13:09 PM
From: "Adam B. Roach" <Adam.Roach@ericsson.com>
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>I'm honestly queasy about this. One of things IP did right was to make
>its addresses and domain names always well defined independent of where
>you are located. With IP addresses, we never send just the last byte,
>assuming that the router fills in the rest. In SIP, we did the same. We
>pass around URLs, which are interpretable independent of the context.

I think this is an argument, however unintentional, to eliminate
the tel: URI.  Without a tag to indicate context (which you are
arguing against), they are not "Uniform" Resource Indicators.

>I fully agree that users should not need to know about dial and number
>plans. I also agree that users should be able to dial extensions and
>local numbers as they do today in a PBX. I'd like to understand the
>costs involved in having internationalization of numbers done within the
>UA. Obtaining a dial plan automatically through a REGISTER response
>would still allow an administrator to control it, and to modify it
>(registrations get refreshed, so you can send updated versions in the
>responses of refreshes). So, my questions are: how complex can a dial
>plan be? How many entries in a dial plan table are there usually? How
>much processing power does it take to compare a number against a dial
>plan and internationalize it?

It's not as much a matter of processing power as the code to perform
internationalization and the size of the data necessary to do so.
As both Dean and Rick have pointed out, the dialing plans which
a phone may be required to understand can be very large and complicated
(especially within an enterprise or VPN environment). As Rick has
pointed out, he is currently running into very real memory limitations 
that make this processing impossible on the user agent.

You can argue about theoretical processing power and memeory necessary 
to perform these sorts of things, but I'm afraid that real-world
experience is going to have to trump theory. Imbedded clients
are *already* running into limitations that prevent them from doing
what you suggest they do.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Date: Tue, 04 Jan 2000 14:46:49 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: URL Generalization for Portability (was TRIP)
In-reply-to: <200001041949.NAA15095@b04a24.exu.ericsson.se>
To: "Adam B. Roach" <Adam.Roach@ericsson.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: dean.willis@wcom.com, sip@lists.research.bell-labs.com
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Adam,

>As both Dean and Rick have pointed out, the dialing plans which
>a phone may be required to understand can be very large and complicated
>(especially within an enterprise or VPN environment).

IP telephony gateways are now the priority model, but will not stay so
forever. At that point in time, a non-legacy oriented solution may be far
preferable, IMHO. A PDA or display phone screen may just be right for
name@host.net...

BTW: The practically the only place in the WSJ where you can find phone
numbers are in ads. URLs dominate in all articles (quip credited to Chris
Cunningham).

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Adam B.
Roach
Sent: Tuesday, January 04, 2000 1:50 PM
To: Jonathan Rosenberg
Cc: Adam B. Roach; dean.willis@wcom.com;
sip@lists.research.bell-labs.com
Subject: Re: URL Generalization for Portability (was TRIP)


>I'm honestly queasy about this. One of things IP did right was to make
>its addresses and domain names always well defined independent of where
>you are located. With IP addresses, we never send just the last byte,
>assuming that the router fills in the rest. In SIP, we did the same. We
>pass around URLs, which are interpretable independent of the context.

I think this is an argument, however unintentional, to eliminate
the tel: URI.  Without a tag to indicate context (which you are
arguing against), they are not "Uniform" Resource Indicators.

>I fully agree that users should not need to know about dial and number
>plans. I also agree that users should be able to dial extensions and
>local numbers as they do today in a PBX. I'd like to understand the
>costs involved in having internationalization of numbers done within the
>UA. Obtaining a dial plan automatically through a REGISTER response
>would still allow an administrator to control it, and to modify it
>(registrations get refreshed, so you can send updated versions in the
>responses of refreshes). So, my questions are: how complex can a dial
>plan be? How many entries in a dial plan table are there usually? How
>much processing power does it take to compare a number against a dial
>plan and internationalize it?

It's not as much a matter of processing power as the code to perform
internationalization and the size of the data necessary to do so.
As both Dean and Rick have pointed out, the dialing plans which
a phone may be required to understand can be very large and complicated
(especially within an enterprise or VPN environment). As Rick has
pointed out, he is currently running into very real memory limitations
that make this processing impossible on the user agent.

You can argue about theoretical processing power and memeory necessary
to perform these sorts of things, but I'm afraid that real-world
experience is going to have to trump theory. Imbedded clients
are *already* running into limitations that prevent them from doing
what you suggest they do.

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA




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"Adam B. Roach" wrote:
> 
> >I'm honestly queasy about this. One of things IP did right was to make
> >its addresses and domain names always well defined independent of where
> >you are located. With IP addresses, we never send just the last byte,
> >assuming that the router fills in the rest. In SIP, we did the same. We
> >pass around URLs, which are interpretable independent of the context.
> 
> I think this is an argument, however unintentional, to eliminate
> the tel: URI.  Without a tag to indicate context (which you are
> arguing against), they are not "Uniform" Resource Indicators.

No, its more an argument for using the tel URL when in its
internationalized form:

tel:+17327417244

The tel URL does allow for phone contexts that help identify where the
URL is valid. For
example, you can include a prefix as a context (like +1 516) to indicate
that the number
can only be called from that area:

tel:5551212;phone-context=+1516

This is useful when you imbed a tel URL in a web page in an intranet for
a company on 
Long Island. This way, the modem in the PC just dials 5551212; no area
code is needed
from there. Our application of the tel URL is slightly different. We are
not using them
to actual dial numbers from web pages. Rather, they point to resources
in the PSTN. In
that sense, I agree with Adam that what I think we want is for them to
be uniform, so
that they are resolvable from everywhere. This means they should either
be international
numbers, or be sip URLs which indicate where to go to get them resolved.

So, in the case of a gateway sending just dialed digits to a proxy, I
argued that we should use a SIP URL and put the contexts and stuff into
the user portion. Several folks complained about that; is it more
palatable to use the phone-context from the tel URL as part of the SIP
URL?

sip:334@wcom.com;phone-context=0000034

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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> Our application of the tel URL is slightly different. We are
> not using them
> to actual dial numbers from web pages. Rather, they point to resources
> in the PSTN. In
> that sense, I agree with Adam that what I think we want is for them to
> be uniform, so
> that they are resolvable from everywhere. This means they should either
> be international
> numbers, or be sip URLs which indicate where to go to get them resolved.
> 

But you can do this with the tel: URL as well by providing the tsp parameter.
The main distinction that I see between the sip: and tel: URLs is that
with the tel: URL, you know DEFINITIVELY that the resource that you are
referring to is a PSTN telephone. With the sip: URL this becomes trickier.
Are you referring to an IP/PC SIP client with a numeric ID that is 
convenient to dial from a plain black phone, or are you actually 
referring to a genuine PSTN terminal? The distinction is (usefully) blurred.

> So, in the case of a gateway sending just dialed digits to a proxy, I
> argued that we should use a SIP URL and put the contexts and stuff into
> the user portion. Several folks complained about that; is it more
> palatable to use the phone-context from the tel URL as part of the SIP
> URL?
> 
> sip:334@wcom.com;phone-context=0000034
>

It seems more logical, IMHO, to stuff everything into the user portion.
This allows you to easily distinguish between parameters directly
related to the user and parameters related to, for example, the 
preferred SIP transport.


> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 

-----------------------------------------------------------------
Sean Olson            mailto:sean.olson@ericsson.com
Ericsson Inc.         tel:(972)583-5472 
                      fax:(972)669-0154



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Date: Tue, 04 Jan 2000 18:06:31 -0600
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From: "James M. Polk" <jmpolk@cisco.com>
Subject: Prioritization limitation question
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Anyone/everyone

Reading the ANSI document T1.619-1992 a colleague pointed out to me, I came
across its MLPP Prioritization requirements as being the following:

1)      0       Flash Override (Highest)
2)      1       Flash
3)      2       Immediate
4)      3       Priority
5)      4       Routine  (Lowest)

But looking at SIP grammar/syntax, it only stipulates 4 levels of priority as
the following:

priority-value = "emergency" | "urgent" | "normal" | "non-urgent" 

Question: Has there been any discussion (which I haven't found yet) on matching
these two specifications? And for clarification, I strongly prefer SIP moving
up to having 5 levels instead of ANSI moving down to 4 levels in order to meet
current US Government/Military Regulations I really don't want to ask them to
change.

Comments/clarifications are welcome, thank you.

_______________________________________________________ 
A sobering reflection upon this century's greatest accomplishments and
discoveries:
"There is no keystroke that rewrites racism... nor is there any software that
takes care of Mother Earth"

James M. Polk 
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5199
www.cisco.com
--=====================_26036514==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>Anyone/everyone</div>
<br>
<div>Reading the ANSI document T1.619-1992 a colleague pointed out to me,
I came across its MLPP Prioritization requirements as being the
following:</div>
<br>
<div>1)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>0<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Flash
Override (Highest)</div>
<div>2)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>1<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Flash</div>
<div>3)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>2<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Immediate</div>
<div>4)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>3<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Priority</div>
<div>5)<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>4<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Routine&nbsp;
(Lowest)</div>
<br>
<div>But looking at SIP grammar/syntax, it only stipulates 4 levels of
priority as the following:</div>
<br>
<div>priority-value = &quot;emergency&quot; | &quot;urgent&quot; |
&quot;normal&quot; | &quot;non-urgent&quot; </div>
<br>
<div>Question: Has there been any discussion (which I haven't found yet)
on matching these two specifications? And for clarification, I strongly
prefer SIP moving up to having 5 levels instead of ANSI moving down to 4
levels in order to meet current US Government/Military Regulations I
really don't want to ask them to change.</div>
<br>
<div>Comments/clarifications are welcome, thank you.</div>
<br>

<div align="center">
_______________________________________________________ <br>
A sobering reflection upon this century's greatest accomplishments and
discoveries:<br>
&quot;There is no keystroke that rewrites racism... nor is there any
software that takes care of Mother Earth&quot;<br>
<br>
</div>
James M. Polk <br>
Sr. Product Manager, Multiservice Architecture and Standards<br>
Enterprise Voice Business Unit<br>
Cisco Systems<br>
Dallas, Texas<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5199<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_26036514==_.ALT--




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Subject: Re: Prioritization limitation question
To: jmpolk@cisco.com (James M. Polk)
Date: Tue, 4 Jan 2000 19:35:00 -0600 (CST)
Cc: sip@lists.research.bell-labs.com, rtbell@cisco.com
In-Reply-To: <4.1.20000104175635.00bcc6e0@diablo.cisco.com> from "James M. Polk" at Jan 04, 2000 06:06:31 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>Anyone/everyone
>
>Reading the ANSI document T1.619-1992 a colleague pointed out to me, I came
>across its MLPP Prioritization requirements as being the following:
>
>1)      0       Flash Override (Highest)
>2)      1       Flash
>3)      2       Immediate
>4)      3       Priority
>5)      4       Routine  (Lowest)

>But looking at SIP grammar/syntax, it only stipulates 4 levels of priority as
>the following:
>
>priority-value = "emergency" | "urgent" | "normal" | "non-urgent" 
>
>Question: Has there been any discussion (which I haven't found yet) on matching
>these two specifications? And for clarification, I strongly prefer SIP moving
>up to having 5 levels instead of ANSI moving down to 4 levels in order to meet
>current US Government/Military Regulations I really don't want to ask them to
>change.

No, this has not been discussed; however, consider that SIP clients are
not to be treated as trusted nodes (since users have direct control over
them). If some PSTN gateway were taking my client's word for it, I'd
hack mine to send out whatever would generate "flash override" as the 
priority whenever I couldn't get through. Not the sort of thing you 
really want me to be able to do, I imagine.

So, given that you will only trust this information when it comes from
another PSTN gateway, I would propose that, instead of trying to mold
the SIP architecture to fit ITU and ANSI documents, it would seem to
make sense to tunnel your ISUP and TCAP messages end-to-end (which
you're obviously *already* doing, since you can't activate CCBS 
without receiving an appropriate indicator in the diagnostic field of
the cause parameter of a REL message, right?).

It wouldn't hurt to map MLPP priorities down into the SIP priorities, 
just in case you *do* terminate on a native SIP client (flash override 
and flash translate to emergency; immediate and priority translate to 
urgent; routine translates to normal) -- but I wouldn't actually trust 
these going back *into* the PSTN. The reason for this is that, in SIP, 
the priorities are a heuristic, meant more for conveying information to 
the user. In the PSTN, the MLPP priorities are absolutes, meant for
conveying information to the network.

Finally, you'll note that there are actually sixteen priorities available
for MLPP (even though only five are defined by the ITU and ANSI). It's 
possible that current proprietary or future ISUP protocols will define 
further priorities. It would, of course, be absurd to come up with sixteen 
SIP priority names to accomodate this possible eventuality, and the prospect 
of having numeric priorities for SIP has already been discussed and soundly 
rejected (if my occasonally fuzzy memory serves me well today).

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Sean Olson wrote:
> 
> > Our application of the tel URL is slightly different. We are
> > not using them
> > to actual dial numbers from web pages. Rather, they point to resources
> > in the PSTN. In
> > that sense, I agree with Adam that what I think we want is for them to
> > be uniform, so
> > that they are resolvable from everywhere. This means they should either
> > be international
> > numbers, or be sip URLs which indicate where to go to get them resolved.
> >
> 
> But you can do this with the tel: URL as well by providing the tsp parameter.

I think it means something different in the context of the tel URL. In
the tel URL, it would indicate that the call should be routed over the
circuit switched network of the telco indicated in the tsp parameter
(thats my interpretation, at least). With the SIP URL, the domain name
is used to obtain the address of a server to send the request to, that
server being the only server capable of resolving the name on the LHS of
the @ sign in the URL. These are really different, and when using
private dialing plans, I think the meaning is more in line with the SIP
URL form. Syntactically, its really no different and doesn't matter.

> The main distinction that I see between the sip: and tel: URLs is that
> with the tel: URL, you know DEFINITIVELY that the resource that you are
> referring to is a PSTN telephone. With the sip: URL this becomes trickier.
> Are you referring to an IP/PC SIP client with a numeric ID that is
> convenient to dial from a plain black phone, or are you actually
> referring to a genuine PSTN terminal? The distinction is (usefully) blurred.

But in the case we are discussing, we really don't know. All we know is
that the gateway has dialed some string of digits.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan  5 08:20:06 2000
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From: "Rosen, Brian" <brosen@fore.com>
To: "'Adam B. Roach'" <Adam.Roach@Ericsson.com>
Cc: sip@lists.research.bell-labs.com
Subject: RE: Prioritization limitation question
Date: Wed, 5 Jan 2000 08:16:46 -0500 
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What if you were trying to implement MLPP where the individual
stations were trusted, in particular, for a customer like the
government?  You are saying SIP cannot be used for a government application,
which typically require MLPP.

There isn't any reason NOT to use the existing convention; they
do everything the current SIP priority does.  If you make them
backwards compatible, you lose nothing, you gain compatibility.

Brian
------------
Brian Rosen, Principal Engineer
Marconi (Formerly FORE Systems)
1000 FORE Drive, Warrendale, PA 15086
(724) 742-6826  mailto:brosen@eng.fore.com 



> -----Original Message-----
> From: Adam B. Roach [mailto:Adam.Roach@Ericsson.com]
> Sent: Tuesday, January 04, 2000 8:35 PM
> To: jmpolk@cisco.com
> Cc: sip@lists.research.bell-labs.com; rtbell@cisco.com
> Subject: Re: Prioritization limitation question
> 
> 
> >Anyone/everyone
> >
> >Reading the ANSI document T1.619-1992 a colleague pointed 
> out to me, I came
> >across its MLPP Prioritization requirements as being the following:
> >
> >1)      0       Flash Override (Highest)
> >2)      1       Flash
> >3)      2       Immediate
> >4)      3       Priority
> >5)      4       Routine  (Lowest)
> 
> >But looking at SIP grammar/syntax, it only stipulates 4 
> levels of priority as
> >the following:
> >
> >priority-value = "emergency" | "urgent" | "normal" | "non-urgent" 
> >
> >Question: Has there been any discussion (which I haven't 
> found yet) on matching
> >these two specifications? And for clarification, I strongly 
> prefer SIP moving
> >up to having 5 levels instead of ANSI moving down to 4 
> levels in order to meet
> >current US Government/Military Regulations I really don't 
> want to ask them to
> >change.
> 
> No, this has not been discussed; however, consider that SIP 
> clients are
> not to be treated as trusted nodes (since users have direct 
> control over
> them). If some PSTN gateway were taking my client's word for it, I'd
> hack mine to send out whatever would generate "flash override" as the 
> priority whenever I couldn't get through. Not the sort of thing you 
> really want me to be able to do, I imagine.
> 
> So, given that you will only trust this information when it comes from
> another PSTN gateway, I would propose that, instead of trying to mold
> the SIP architecture to fit ITU and ANSI documents, it would seem to
> make sense to tunnel your ISUP and TCAP messages end-to-end (which
> you're obviously *already* doing, since you can't activate CCBS 
> without receiving an appropriate indicator in the diagnostic field of
> the cause parameter of a REL message, right?).
> 
> It wouldn't hurt to map MLPP priorities down into the SIP priorities, 
> just in case you *do* terminate on a native SIP client (flash 
> override 
> and flash translate to emergency; immediate and priority translate to 
> urgent; routine translates to normal) -- but I wouldn't 
> actually trust 
> these going back *into* the PSTN. The reason for this is 
> that, in SIP, 
> the priorities are a heuristic, meant more for conveying 
> information to 
> the user. In the PSTN, the MLPP priorities are absolutes, meant for
> conveying information to the network.
> 
> Finally, you'll note that there are actually sixteen 
> priorities available
> for MLPP (even though only five are defined by the ITU and 
> ANSI). It's 
> possible that current proprietary or future ISUP protocols 
> will define 
> further priorities. It would, of course, be absurd to come up 
> with sixteen 
> SIP priority names to accomodate this possible eventuality, 
> and the prospect 
> of having numeric priorities for SIP has already been 
> discussed and soundly 
> rejected (if my occasonally fuzzy memory serves me well today).
> 
> --
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. 
> Arapaho, MS L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | 
> Richardson, TX 75081 USA
> 



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To: "Rosen, Brian" <brosen@fore.com>,
        "'Adam B. Roach'" <Adam.Roach@Ericsson.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: Prioritization limitation question
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Adam

Brian states the conclusions I wish to accomplish. There are quite a few IP
only Voice environments being planned and implemented -- with PSTN access in
and out very controlled or non-existent. In this case, having the 5 priorities 
explicitly allows SIP into environments requiring MLPP. Not having all 5
completely eliminates SIP -- as "Flash Override" is a very required
feature/application. I do understand why there needs to be a great amount of
individuality within the development of a protocol with specific
interoperability interfacing requirements defined and limited to not overwhelm
the new protocol's intent. But with inclusion of this 5th priority, virtually
all MLPP implementations will at that point include SIP as viable; likely not
before this time though.

I'm just suggesting the addition of a single value within a defined
variable-set. I'm not suggesting a new variable, or taking the protocol in a
new direction.

At 08:16 AM 1/5/2000 -0500, Rosen, Brian wrote:
>What if you were trying to implement MLPP where the individual
>stations were trusted, in particular, for a customer like the
>government?  You are saying SIP cannot be used for a government application,
>which typically require MLPP.
>
>There isn't any reason NOT to use the existing convention; they
>do everything the current SIP priority does.  If you make them
>backwards compatible, you lose nothing, you gain compatibility.
>
>Brian
>------------
>Brian Rosen, Principal Engineer
>Marconi (Formerly FORE Systems)
>1000 FORE Drive, Warrendale, PA 15086
>(724) 742-6826  mailto:brosen@eng.fore.com 
>
>
>
>> -----Original Message-----
>> From: Adam B. Roach [mailto:Adam.Roach@Ericsson.com]
>> Sent: Tuesday, January 04, 2000 8:35 PM
>> To: jmpolk@cisco.com
>> Cc: sip@lists.research.bell-labs.com; rtbell@cisco.com
>> Subject: Re: Prioritization limitation question
>> 
>> 
>> >Anyone/everyone
>> >
>> >Reading the ANSI document T1.619-1992 a colleague pointed 
>> out to me, I came
>> >across its MLPP Prioritization requirements as being the following:
>> >
>> >1)      0       Flash Override (Highest)
>> >2)      1       Flash
>> >3)      2       Immediate
>> >4)      3       Priority
>> >5)      4       Routine  (Lowest)
>> 
>> >But looking at SIP grammar/syntax, it only stipulates 4 
>> levels of priority as
>> >the following:
>> >
>> >priority-value = "emergency" | "urgent" | "normal" | "non-urgent" 
>> >
>> >Question: Has there been any discussion (which I haven't 
>> found yet) on matching
>> >these two specifications? And for clarification, I strongly 
>> prefer SIP moving
>> >up to having 5 levels instead of ANSI moving down to 4 
>> levels in order to meet
>> >current US Government/Military Regulations I really don't 
>> want to ask them to
>> >change.
>> 
>> No, this has not been discussed; however, consider that SIP 
>> clients are
>> not to be treated as trusted nodes (since users have direct 
>> control over
>> them). If some PSTN gateway were taking my client's word for it, I'd
>> hack mine to send out whatever would generate "flash override" as the 
>> priority whenever I couldn't get through. Not the sort of thing you 
>> really want me to be able to do, I imagine.
>> 
>> So, given that you will only trust this information when it comes from
>> another PSTN gateway, I would propose that, instead of trying to mold
>> the SIP architecture to fit ITU and ANSI documents, it would seem to
>> make sense to tunnel your ISUP and TCAP messages end-to-end (which
>> you're obviously *already* doing, since you can't activate CCBS 
>> without receiving an appropriate indicator in the diagnostic field of
>> the cause parameter of a REL message, right?).
>> 
>> It wouldn't hurt to map MLPP priorities down into the SIP priorities, 
>> just in case you *do* terminate on a native SIP client (flash 
>> override 
>> and flash translate to emergency; immediate and priority translate to 
>> urgent; routine translates to normal) -- but I wouldn't 
>> actually trust 
>> these going back *into* the PSTN. The reason for this is 
>> that, in SIP, 
>> the priorities are a heuristic, meant more for conveying 
>> information to 
>> the user. In the PSTN, the MLPP priorities are absolutes, meant for
>> conveying information to the network.
>> 
>> Finally, you'll note that there are actually sixteen 
>> priorities available
>> for MLPP (even though only five are defined by the ITU and 
>> ANSI). It's 
>> possible that current proprietary or future ISUP protocols 
>> will define 
>> further priorities. It would, of course, be absurd to come up 
>> with sixteen 
>> SIP priority names to accomodate this possible eventuality, 
>> and the prospect 
>> of having numeric priorities for SIP has already been 
>> discussed and soundly 
>> rejected (if my occasonally fuzzy memory serves me well today).
>> 
>> --
>> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. 
>> Arapaho, MS L-04
>> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | 
>> Richardson, TX 75081 USA
>> 
>
>

_______________________________________________________ 
A sobering reflection upon this century's greatest accomplishments and
discoveries:
"There is no keystroke that rewrites racism... nor is there any software that
takes care of Mother Earth"

James M. Polk 
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5199
www.cisco.com
--=====================_1236468==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>Adam</div>
<br>
<div>Brian states the conclusions I wish to accomplish. There are quite a
few IP only Voice environments being planned and implemented -- with PSTN
access in and out very controlled or non-existent. In this case, having
the 5 priorities&nbsp; explicitly allows SIP into environments requiring
MLPP. Not having all 5 completely eliminates SIP -- as &quot;Flash
Override&quot; is a very required feature/application. I do understand
why there needs to be a great amount of individuality within the
development of a protocol with specific interoperability interfacing
requirements defined and limited to not overwhelm the new protocol's
intent. But with inclusion of this 5th priority, virtually all MLPP
implementations will at that point include SIP as viable; likely not
before this time though.</div>
<br>
<div>I'm just suggesting the addition of a single value within a defined
variable-set. I'm not suggesting a new variable, or taking the protocol
in a new direction.</div>
<br>
<div>At 08:16 AM 1/5/2000 -0500, Rosen, Brian wrote:</div>
<div>&gt;What if you were trying to implement MLPP where the
individual</div>
<div>&gt;stations were trusted, in particular, for a customer like
the</div>
<div>&gt;government?&nbsp; You are saying SIP cannot be used for a
government application,</div>
<div>&gt;which typically require MLPP.</div>
<div>&gt;</div>
<div>&gt;There isn't any reason NOT to use the existing convention;
they</div>
<div>&gt;do everything the current SIP priority does.&nbsp; If you make
them</div>
<div>&gt;backwards compatible, you lose nothing, you gain
compatibility.</div>
<div>&gt;</div>
<div>&gt;Brian</div>
<div>&gt;------------</div>
<div>&gt;Brian Rosen, Principal Engineer</div>
<div>&gt;Marconi (Formerly FORE Systems)</div>
<div>&gt;1000 FORE Drive, Warrendale, PA 15086</div>
<div>&gt;(724) 742-6826&nbsp;
<a href="mailto:brosen@eng.fore.com" EUDORA=AUTOURL>mailto:brosen@eng.fore.com</a>
</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;&gt; -----Original Message-----</div>
<div>&gt;&gt; From: Adam B. Roach [<a href="mailto:Adam.Roach@Ericsson.com" EUDORA=AUTOURL>mailto:Adam.Roach@Ericsson.com</a>]</div>
<div>&gt;&gt; Sent: Tuesday, January 04, 2000 8:35 PM</div>
<div>&gt;&gt; To: jmpolk@cisco.com</div>
<div>&gt;&gt; Cc: sip@lists.research.bell-labs.com; rtbell@cisco.com</div>
<div>&gt;&gt; Subject: Re: Prioritization limitation question</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; </div>
<div>&gt;&gt; &gt;Anyone/everyone</div>
<div>&gt;&gt; &gt;</div>
<div>&gt;&gt; &gt;Reading the ANSI document T1.619-1992 a colleague pointed </div>
<div>&gt;&gt; out to me, I came</div>
<div>&gt;&gt; &gt;across its MLPP Prioritization requirements as being the following:</div>
<div>&gt;&gt; &gt;</div>
<div>&gt;&gt; &gt;1)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Flash Override (Highest)</div>
<div>&gt;&gt; &gt;2)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Flash</div>
<div>&gt;&gt; &gt;3)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Immediate</div>
<div>&gt;&gt; &gt;4)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Priority</div>
<div>&gt;&gt; &gt;5)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Routine&nbsp; (Lowest)</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; &gt;But looking at SIP grammar/syntax, it only stipulates 4 </div>
<div>&gt;&gt; levels of priority as</div>
<div>&gt;&gt; &gt;the following:</div>
<div>&gt;&gt; &gt;</div>
<div>&gt;&gt; &gt;priority-value = &quot;emergency&quot; | &quot;urgent&quot; | &quot;normal&quot; | &quot;non-urgent&quot; </div>
<div>&gt;&gt; &gt;</div>
<div>&gt;&gt; &gt;Question: Has there been any discussion (which I haven't </div>
<div>&gt;&gt; found yet) on matching</div>
<div>&gt;&gt; &gt;these two specifications? And for clarification, I strongly </div>
<div>&gt;&gt; prefer SIP moving</div>
<div>&gt;&gt; &gt;up to having 5 levels instead of ANSI moving down to 4 </div>
<div>&gt;&gt; levels in order to meet</div>
<div>&gt;&gt; &gt;current US Government/Military Regulations I really don't </div>
<div>&gt;&gt; want to ask them to</div>
<div>&gt;&gt; &gt;change.</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; No, this has not been discussed; however, consider that SIP </div>
<div>&gt;&gt; clients are</div>
<div>&gt;&gt; not to be treated as trusted nodes (since users have direct </div>
<div>&gt;&gt; control over</div>
<div>&gt;&gt; them). If some PSTN gateway were taking my client's word for it, I'd</div>
<div>&gt;&gt; hack mine to send out whatever would generate &quot;flash override&quot; as the </div>
<div>&gt;&gt; priority whenever I couldn't get through. Not the sort of thing you </div>
<div>&gt;&gt; really want me to be able to do, I imagine.</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; So, given that you will only trust this information when it comes from</div>
<div>&gt;&gt; another PSTN gateway, I would propose that, instead of trying to mold</div>
<div>&gt;&gt; the SIP architecture to fit ITU and ANSI documents, it would seem to</div>
<div>&gt;&gt; make sense to tunnel your ISUP and TCAP messages end-to-end (which</div>
<div>&gt;&gt; you're obviously *already* doing, since you can't activate CCBS </div>
<div>&gt;&gt; without receiving an appropriate indicator in the diagnostic field of</div>
<div>&gt;&gt; the cause parameter of a REL message, right?).</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; It wouldn't hurt to map MLPP priorities down into the SIP priorities, </div>
<div>&gt;&gt; just in case you *do* terminate on a native SIP client (flash </div>
<div>&gt;&gt; override </div>
<div>&gt;&gt; and flash translate to emergency; immediate and priority translate to </div>
<div>&gt;&gt; urgent; routine translates to normal) -- but I wouldn't </div>
<div>&gt;&gt; actually trust </div>
<div>&gt;&gt; these going back *into* the PSTN. The reason for this is </div>
<div>&gt;&gt; that, in SIP, </div>
<div>&gt;&gt; the priorities are a heuristic, meant more for conveying </div>
<div>&gt;&gt; information to </div>
<div>&gt;&gt; the user. In the PSTN, the MLPP priorities are absolutes, meant for</div>
<div>&gt;&gt; conveying information to the network.</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; Finally, you'll note that there are actually sixteen </div>
<div>&gt;&gt; priorities available</div>
<div>&gt;&gt; for MLPP (even though only five are defined by the ITU and </div>
<div>&gt;&gt; ANSI). It's </div>
<div>&gt;&gt; possible that current proprietary or future ISUP protocols </div>
<div>&gt;&gt; will define </div>
<div>&gt;&gt; further priorities. It would, of course, be absurd to come up </div>
<div>&gt;&gt; with sixteen </div>
<div>&gt;&gt; SIP priority names to accomodate this possible eventuality, </div>
<div>&gt;&gt; and the prospect </div>
<div>&gt;&gt; of having numeric priorities for SIP has already been </div>
<div>&gt;&gt; discussed and soundly </div>
<div>&gt;&gt; rejected (if my occasonally fuzzy memory serves me well today).</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; --</div>
<div>&gt;&gt; Adam Roach, Ericsson Inc. |&nbsp; Ph: +1 972 583 7594 | 1010 E. </div>
<div>&gt;&gt; Arapaho, MS L-04</div>
<div>&gt;&gt; adam.roach@ericsson.com&nbsp;&nbsp; | Fax: +1 972 669 0154 | </div>
<div>&gt;&gt; Richardson, TX 75081 USA</div>
<div>&gt;&gt; </div>
<div>&gt;</div>
<div>&gt;</div>
<br>

<div align="center">
_______________________________________________________ <br>
A sobering reflection upon this century's greatest accomplishments and discoveries:<br>
&quot;There is no keystroke that rewrites racism... nor is there any software that takes care of Mother Earth&quot;<br>
<br>
</div>
James M. Polk <br>
Sr. Product Manager, Multiservice Architecture and Standards<br>
Enterprise Voice Business Unit<br>
Cisco Systems<br>
Dallas, Texas<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5199<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_1236468==_.ALT--




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Subject: Re: Prioritization limitation question
To: jmpolk@cisco.com (James M. Polk)
Date: Wed, 5 Jan 2000 11:45:17 -0600 (CST)
Cc: brosen@fore.com (Rosen Brian), Adam.Roach@ericsson.com ('Adam B. Roach'),
        sip@lists.research.bell-labs.com, rtbell@cisco.com
In-Reply-To: <4.1.20000105105551.00a50340@diablo.cisco.com> from "James M. Polk" at Jan 05, 2000 11:05:55 AM
From: "Adam B. Roach" <Adam.Roach@ericsson.com>
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>Brian states the conclusions I wish to accomplish. There are quite a few IP
>only Voice environments being planned and implemented -- with PSTN access in
>and out very controlled or non-existent. In this case, having the 5 priorities 
>explicitly allows SIP into environments requiring MLPP. 

I mistook your problem to be a PSTN interworking one; sorry.

>Not having all 5
>completely eliminates SIP -- as "Flash Override" is a very required
>feature/application. I do understand why there needs to be a great amount of
>individuality within the development of a protocol with specific
>interoperability interfacing requirements defined and limited to not overwhelm
>the new protocol's intent. But with inclusion of this 5th priority, virtually
>all MLPP implementations will at that point include SIP as viable; likely not
>before this time though.

The SIP Priority: header does not currently provide anything like MLPP. 
It is not specified that it is to be used for call pre-emption. I beleive 
you'll find that most clients use it to render the incoming call (e.g. 
making the screen flash, or making the phone ring particularly urgently), 
if at all.

Extending the number of priorites will merely change the syntax without
adding the semantics you desire.

I propose that, if you need this functionality, you write up an
internet draft specifying how to negotiate between clients that
MLPP is available, the exact behavior expected, and the method
of conveying MLPP priority levels. For simplicity of mapping,
I'd take a look at ITU Q.733.3 and Q.955.3. As an overview,
I'd include a header (similar to the CCBS diagnostic) in the
486, 600, and 603 (and possibly 182) responses to INVITE which 
indicates that the client is CCBS-capable. The originating client 
would then re-launch the INVITE with some indication that it is 
invoking the CCBS service, along with the priority of the incoming 
call.

But merely extending the enumeration of values allowed in the
Priority: header is *wildly* insufficient for what you require.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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From: "Donovan, Steven R." <Steven.R.Donovan@wcom.com>
Subject: tel:# versus sip:#@host;user=phone
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Discussions in the "URL Generalization for Portability (was TRIP)" thread
have me a little confused.

It was my understanding that the reason for the user=phone construct in the
sip url was because the tel url definition was still in draft stage.  As
such, I had always thought that a sip url with user=phone represented the
same address as a tel url with the same number.

As such, in my mind (warped as it might be) the following two INVITE
messages are used to call the same PSTN phone number:

INVITE sip:+19725551212@wcom.com;user=phone SIP/2.0
...

INVITE tel:+19725551212 SIP/2.0
...

In both of these cases, the caller wants to reach someone at the number
19725551212.

Note that this does not necessarily mean in either case that the call will
routed to a PSTN phone that has 19725551212 printed on the front of it or
that is connected to a switch via a line interface that is identified by
that number.  There are many points between the caller and callee of the
call where the caller address can be changed.  It is equally valid for both
of the above cases to change +19725551212 to dallas@directory-assistance.com
and have the call fowarded to a SIP based directory assistance platform as
illustrated in the following:

Callee to Proxy Server
INVITE tel:+19725551212 SIP/2.0
From: joe@home
To: tel:+19725551212
...

Proxy Server to Directory Assistance Platform

INVITE dallas@directory-assistance.com SIP/2.0
From: joe@home
To: tel:+19725551212
...

The resent discussions imply that the tel url and sip user=phone addresses
would be treated differently.

This does not seem wise to me.

Steve



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I agree 100% with Adam here. 

The semantics of Priority are not what you want. If this feature is
critical to your deployment, please do write up a draft with a proposal
on how it works. 

-Jonathan R.

"Adam B. Roach" wrote:
> 
> The SIP Priority: header does not currently provide anything like MLPP.
> It is not specified that it is to be used for call pre-emption. I beleive
> you'll find that most clients use it to render the incoming call (e.g.
> making the screen flash, or making the phone ring particularly urgently),
> if at all.
> 
> Extending the number of priorites will merely change the syntax without
> adding the semantics you desire.
> 
> I propose that, if you need this functionality, you write up an
> internet draft specifying how to negotiate between clients that
> MLPP is available, the exact behavior expected, and the method
> of conveying MLPP priority levels. For simplicity of mapping,
> I'd take a look at ITU Q.733.3 and Q.955.3. As an overview,
> I'd include a header (similar to the CCBS diagnostic) in the
> 486, 600, and 603 (and possibly 182) responses to INVITE which
> indicates that the client is CCBS-capable. The originating client
> would then re-launch the INVITE with some indication that it is
> invoking the CCBS service, along with the priority of the incoming
> call.
> 
> But merely extending the enumeration of values allowed in the
> Priority: header is *wildly* insufficient for what you require.
> 
> --
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Jonathan Rosenberg wrote:
> 
> The issue is, if the content is referenced by indirection, is it better
> to be in the body or in a header. Henning is arguing in favor of a
> header. I agree with him that this is simpler. Both he and Dan are also
> arguing that we need a way to identify the purpose of the content.
> Providing such data also argues for it being a header.

I would say that the reason body parts are preferrable over headers is
that we're likely to wish to associate various content descriptors with
the "extra" content, regardless of whether it is included inline or by
reference. There may be a variety of such descriptors, some of them
which we know about now (most of which are probably already
well-defined), and some which we can't imagine now and which may have a
purpose completely unlike other descriptors.

To me it seems to make sense to have a single mechanism for associating
descriptors with directly and indirectly included content (I believe
this is how MHTML works also). This indicates that using MIME body parts
is the more prudent choice here, even though it's tempting to go for the
admittedly simpler but also more shortsighted approach of sticking URLs
into headers.

Incidentally, the X-Face header recognized by some XEmacs mailer holds
an image inline in some obscure bitmap format. The idea was great and
it's a mystery why this isn't a widely supported feature today - maybe
because of the pain involved in generating a decent quality image that
fits in just a couple of hundred bytes.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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"Donovan, Steven R." wrote:
> 
> Discussions in the "URL Generalization for Portability (was TRIP)" thread
> have me a little confused.
> 
> It was my understanding that the reason for the user=phone construct in the
> sip url was because the tel url definition was still in draft stage.  As
> such, I had always thought that a sip url with user=phone represented the
> same address as a tel url with the same number.

No; the user=phone simply stated that the user portion is built using
the
BNF of telephone-subscriber. Beyond that, its still a SIP URL.

> The resent discussions imply that the tel url and sip user=phone addresses
> would be treated differently.

Not differently in the sense you describe. In either case, it would
still be true that the call might be routed to some other URL which is
not associated with the telephone number at all
(sip:directory-service@wcom.com, for example).

The debate is really around numbers that are not uniform, in the sense
that they aren't usable everywhere. For an international e164 number,
using the tel URL for:

tel:+17327417244

or

sip:+17327417244@wcom.com

is likely to have the same effect. But, with a number which is private,
the tel URL doesn't (I've been arguing) provide sufficient information
to figure out how to resolve the number, while the SIP URL form does.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: Prioritization limitation question
To: brosen@fore.com (Rosen Brian)
Date: Wed, 5 Jan 2000 13:44:15 -0600 (CST)
Cc: sip@lists.research.bell-labs.com
In-Reply-To: <4FBEA8857476D311A03300204840E1CF0685E9@whq-msgusr-02.fore.com> from "Rosen, Brian" at Jan 05, 2000 02:05:20 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>I agree we will have to have a fully worked out proposal
>not just an additional level or two.
>
>Do you believe that the existing SIP priority field should be 
>reused (essentially having a way to state that the priority is 
>mandatory, not advisory), or add a whole new field?
>I'd rather the former.

I'd propose using different headers, simply because I don't
like the philosophy of overloading existing headers in extensions
to mean semanticaly different things. I propose that your callflow
might look like:

* I launch a normal invite to Joe:

  INVITE sip:joe@bob.com SIP/2.0
  To: <sip:joe@bob.com>
  From: <sip:adam@ericsson.com>;tag=13948753987
  Call-Id: 13078245@ws0985.ericsson.com
  CSeq: 1 INVITE
  Via: SIP/2.0/UDP ws0985.ericsson.com

* His MLPP-capable device responds with a busy, and an
  indication that he supports MLPP (along with a version,
  in case the semantics need updating in the future):

  SIP/2.0 486 Busy Here
  To: <sip:joe@bob.com>;tag=a872-27bc-287e-7d7f
  From: <sip:adam@ericsson.com>;tag=13948753987
  Call-Id: 13078245@ws0985.ericsson.com
  CSeq: 1 INVITE
  MLPP-Version: 1

* Seeing that his phone supports a version of MLPP that my phone is
  capable of, my phone gives me the option of invoking the CCBS service.
  I activate it with a level of immediate (2). The phone re-launches
  my INVITE request to Joe's phone:

  INVITE sip:joe@bob.com SIP/2.0
  To: <sip:joe@bob.com>
  From: <sip:adam@ericsson.com>;tag=13948753987
  Call-Id: 13078245@ws0985.ericsson.com
  CSeq: 2 INVITE
  Via: SIP/2.0/UDP ws0985.ericsson.com
  MLPP-Version: 1
  MLPP-Priority: 2

* Joe's phone "does the right thing" and takes my call:

  SIP/2.0 200 Okay
  To: <sip:joe@bob.com>;tag=7f27-237e-ab71-0277
  From: <sip:adam@ericsson.com>;tag=13948753987
  Call-Id: 13078245@ws0985.ericsson.com
  CSeq: 2 INVITE

Before soneone starts a discussion on the fact that
the INVITE must be launched twice: note that this is the
same way this service is handled in the PSTN currently. A
call attempt gets a busy indication with a CCBS flag set;
the call attempt is then re-launched with MLPP. Presumably,
this is done for a reason. The above solution should
satisfy the same requirements as the PSTN version of this
service.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Rather than using this MLPP-Version thing to indicate support for CCBS,
I'd rather use the more generic Supported/Require mechanism (a draft on
this is pending within the next few days). So,

C->S INVITE

S->C 421 Feature Available
Require: com.cisco.mlpp

C->S INVITE
Supported: com.cisco.mlpp
MLPP: 2;version=1

S->C 200 OK

-Jonathan R.

"Adam B. Roach" wrote:
> 
> >I agree we will have to have a fully worked out proposal
> >not just an additional level or two.
> >
> >Do you believe that the existing SIP priority field should be
> >reused (essentially having a way to state that the priority is
> >mandatory, not advisory), or add a whole new field?
> >I'd rather the former.
> 
> I'd propose using different headers, simply because I don't
> like the philosophy of overloading existing headers in extensions
> to mean semanticaly different things. I propose that your callflow
> might look like:
> 
> * I launch a normal invite to Joe:
> 
>   INVITE sip:joe@bob.com SIP/2.0
>   To: <sip:joe@bob.com>
>   From: <sip:adam@ericsson.com>;tag=13948753987
>   Call-Id: 13078245@ws0985.ericsson.com
>   CSeq: 1 INVITE
>   Via: SIP/2.0/UDP ws0985.ericsson.com
> 
> * His MLPP-capable device responds with a busy, and an
>   indication that he supports MLPP (along with a version,
>   in case the semantics need updating in the future):
> 
>   SIP/2.0 486 Busy Here
>   To: <sip:joe@bob.com>;tag=a872-27bc-287e-7d7f
>   From: <sip:adam@ericsson.com>;tag=13948753987
>   Call-Id: 13078245@ws0985.ericsson.com
>   CSeq: 1 INVITE
>   MLPP-Version: 1
> 
> * Seeing that his phone supports a version of MLPP that my phone is
>   capable of, my phone gives me the option of invoking the CCBS service.
>   I activate it with a level of immediate (2). The phone re-launches
>   my INVITE request to Joe's phone:
> 
>   INVITE sip:joe@bob.com SIP/2.0
>   To: <sip:joe@bob.com>
>   From: <sip:adam@ericsson.com>;tag=13948753987
>   Call-Id: 13078245@ws0985.ericsson.com
>   CSeq: 2 INVITE
>   Via: SIP/2.0/UDP ws0985.ericsson.com
>   MLPP-Version: 1
>   MLPP-Priority: 2
> 
> * Joe's phone "does the right thing" and takes my call:
> 
>   SIP/2.0 200 Okay
>   To: <sip:joe@bob.com>;tag=7f27-237e-ab71-0277
>   From: <sip:adam@ericsson.com>;tag=13948753987
>   Call-Id: 13078245@ws0985.ericsson.com
>   CSeq: 2 INVITE
> 
> Before soneone starts a discussion on the fact that
> the INVITE must be launched twice: note that this is the
> same way this service is handled in the PSTN currently. A
> call attempt gets a busy indication with a CCBS flag set;
> the call attempt is then re-launched with MLPP. Presumably,
> this is done for a reason. The above solution should
> satisfy the same requirements as the PSTN version of this
> service.
> 
> --
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: Prioritization limitation question
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Wed, 5 Jan 2000 17:50:02 -0600 (CST)
Cc: Adam.Roach@ericsson.com (Adam B. Roach), brosen@fore.com (Rosen Brian),
        sip@lists.research.bell-labs.com
In-Reply-To: <3873CF03.AEE128B9@dynamicsoft.com> from "Jonathan Rosenberg" at Jan 05, 2000 06:08:51 PM
From: "Adam B. Roach" <Adam.Roach@ericsson.com>
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Jonathan Rosenberg writes:

>Rather than using this MLPP-Version thing to indicate support for CCBS,
>I'd rather use the more generic Supported/Require mechanism (a draft on
>this is pending within the next few days). So,
>
>C->S INVITE
>
>S->C 421 Feature Available
>Require: com.cisco.mlpp
>
>C->S INVITE
>Supported: com.cisco.mlpp
>MLPP: 2;version=1
>
>S->C 200 OK

It sounds like you're adding yet another round trip:
the first to agree on the feature, the second to try
an invite (which gets a "busy" response), and the third
to invoke the feature (am I misreading something here?)

Keep in mind that MLPP is relevant only when the far side
returns an indication like "busy." Only *after* the calling 
party is notified that the far side is busy does he make a 
decision that he wants to break in on the call.

Also, in PSTN interoperation scenarios, the gateway won't
know that you can break in on the called party until the busy
indication arrives (since we're just pumping the call back
into the network and don't necessarily know the capability
of the end office to which the subscriber is physically 
connected) -- so negotiation of the feature cannot occur before 
an attempt is made to reach the called party.

If we're going to go out of our way to borrow a service from
ISUP, we'd be exceedingly foolish not to make them able to
interoperate.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 09:32:21 2000
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To: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        jmpolk@cisco.com (James M. Polk)
From: Bob Bell <rtbell@cisco.com>
Subject: Re: Prioritization limitation question
Cc: sip@lists.research.bell-labs.com
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Adam -

Comments imbedded.

At 06:35 PM  1/4/00, Adam B. Roach wrote:
> >Anyone/everyone
> >
> >Reading the ANSI document T1.619-1992 a colleague pointed out to me, I came
> >across its MLPP Prioritization requirements as being the following:
> >
> >1)      0       Flash Override (Highest)
> >2)      1       Flash
> >3)      2       Immediate
> >4)      3       Priority
> >5)      4       Routine  (Lowest)
>
> >But looking at SIP grammar/syntax, it only stipulates 4 levels of 
> priority as
> >the following:
> >
> >priority-value = "emergency" | "urgent" | "normal" | "non-urgent"
> >
> >Question: Has there been any discussion (which I haven't found yet) on 
> matching
> >these two specifications? And for clarification, I strongly prefer SIP 
> moving
> >up to having 5 levels instead of ANSI moving down to 4 levels in order 
> to meet
> >current US Government/Military Regulations I really don't want to ask 
> them to
> >change.
>
>No, this has not been discussed; however, consider that SIP clients are
>not to be treated as trusted nodes (since users have direct control over
>them). If some PSTN gateway were taking my client's word for it, I'd
>hack mine to send out whatever would generate "flash override" as the
>priority whenever I couldn't get through. Not the sort of thing you
>really want me to be able to do, I imagine.

 >>>Bob
In a totally IP environment, there would be no PSTN gateways. The 
precedence values would be used to determine the order in which packets 
would be dropped at congestion points so as to allow the higher priority 
"calls" to be undegraded under congested conditions. For the enforcement, 
perhaps this service requires the use of a "trusted" proxy to authenticate 
a user request for a high priority call.
<<<Bob

>So, given that you will only trust this information when it comes from
>another PSTN gateway, I would propose that, instead of trying to mold
>the SIP architecture to fit ITU and ANSI documents, it would seem to
>make sense to tunnel your ISUP and TCAP messages end-to-end (which
>you're obviously *already* doing, since you can't activate CCBS
>without receiving an appropriate indicator in the diagnostic field of
>the cause parameter of a REL message, right?).
>
>It wouldn't hurt to map MLPP priorities down into the SIP priorities,
>just in case you *do* terminate on a native SIP client (flash override
>and flash translate to emergency; immediate and priority translate to
>urgent; routine translates to normal) -- but I wouldn't actually trust

 >>>Bob
Mapping the Flash and Flash Override into the same value basically 
eliminates the differences between them. There is a real, tactical 
difference for the distinction.
<<<Bob

>these going back *into* the PSTN. The reason for this is that, in SIP,
>the priorities are a heuristic, meant more for conveying information to
>the user. In the PSTN, the MLPP priorities are absolutes, meant for
>conveying information to the network.
>
>Finally, you'll note that there are actually sixteen priorities available
>for MLPP (even though only five are defined by the ITU and ANSI). It's
>possible that current proprietary or future ISUP protocols will define
>further priorities. It would, of course, be absurd to come up with sixteen
>SIP priority names to accomodate this possible eventuality, and the prospect
>of having numeric priorities for SIP has already been discussed and soundly
>rejected (if my occasonally fuzzy memory serves me well today).

 >>>Bob
Perhaps using only a keyword followed by a numeric value could be used.
<<<Bob


>--
>Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
>adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 09:34:58 2000
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Date: Thu, 06 Jan 2000 09:25:43 +0530
From: "ANIRBAN ROY" <anirban.roy@wipro.com>
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hi,
I am trying to figure out how to get a home phone functionality in SIP.
The home phone has the following functionality
1    When a call comes all phones in the home ring
2    When one picks up, all other line must stop ringing
3    A user can picks up from any other telephone in the home and join
an existing call.


                                this a scenario where A calls a Home
Phone which has Three connection B, C, D in his home



+-------------|
                                                    A---------------|
Proxy        |---------------D

|                    |

+------------+

/     \

/       \

/         \

/           \

B           C


When a calls to a home phone which has 3 connection (B, C, D) lands on
proxy, then proxy forks the INVITE message to B, C, D. so every phone
starts ringing. This is the first character of Home Phone. Now Let's say
B picks up the phone then Proxy sends CANCEL message to C and D so that
the ringing can stop. This fulfills the Second point of Home Phone.

I am not able to get the call flow for the Third point in home Phone. If
lets say C picks up the phone then he should also be in session with A
and B. How can this be done because When a CANCEL message is received by
C and D then they both reached the state where they can only expect
INVITE.


Can any one please tell me how the call flow for the third point in the
home phone.

The above points of Home phone were taken from the -----Bell Labs
Journal (October-December 1998)



regards
Anirban




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 09:38:16 2000
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hi,
I am trying to figure out how to get a home phone functionality in SIP.
The home phone has the following functionality
1    When a call comes all phones in the home ring
2    When one picks up, all other line must stop ringing
3    A user can picks up from any other telephone in the home and join
an existing call.


                                this a scenario where A calls a Home
Phone which has Three connection B, C, D in his home



+-------------|
                                                    A---------------|
Proxy        |---------------D

|                    |

+------------+

/     \

/       \

/         \

/           \

B           C


When a calls to a home phone which has 3 connection (B, C, D) lands on
proxy, then proxy forks the INVITE message to B, C, D. so every phone
starts ringing. This is the first character of Home Phone. Now Let's say
B picks up the phone then Proxy sends CANCEL message to C and D so that
the ringing can stop. This fulfills the Second point of Home Phone.

I am not able to get the call flow for the Third point in home Phone. If
lets say C picks up the phone then he should also be in session with A
and B. How can this be done because When a CANCEL message is received by
C and D then they both reached the state where they can only expect
INVITE.


Can any one please tell me how the call flow for the third point in the
home phone.

The above points of Home phone were taken from the -----Bell Labs
Journal (October-December 1998)



regards
Anirban




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 09:59:48 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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You are right in that a SIP UA, as currently specified, won't be able to
get home line type of service. It can be done in the way described in
this paper, but it requires changes to the behavior of the UA, although
no changes to the protocol itself, depending on how its done. It also
requires the caller to be "cooperative" - they will receive multiple 200
OK as each extension picks up, and they need to initiate a multi-party
conference to do it. Basically, if you use multicast instead of forking,
you don't need protocol extensions, just some new behaviors in the UA. 
If you want to rely on forking rather than multicast, you need somewhat
more complex extensions.

When using multicast, the UA basically snoops on the multicast group for
BYEs and other 200 OKs. This allows it to build a complete picture of
what other extensions have picked up, hung up, or been hung up on. Based
on this, it can, in a distributed fashion, figure out the state of the
call. If there are calls active, a user picking up the phone would cause
the phone to send a 200 OK for the call. 

For forking, snooping of BYEs and other messages is not feasible, so
some extensions are needed. However, doing this means you home line
extensions could actually be anywhere in the world. Pretty cool. 

There are other ways to do it which hide from the caller the fact that
multiple extensions have picked up. This is more like how the current
home phone service works. 

Are many people interested in this? If there are many folks who would
like to see some work on SIP extensions required for home line service,
it might be something worthwhile to consider.

-Jonathan R.

ANIRBAN ROY wrote:
> 
> hi,
> I am trying to figure out how to get a home phone functionality in SIP.
> The home phone has the following functionality
> 1    When a call comes all phones in the home ring
> 2    When one picks up, all other line must stop ringing
> 3    A user can picks up from any other telephone in the home and join
> an existing call.
> 
>                                A scenario is there where A calls a Home
> Phone which has Three connection B, C, D in his home
> 
> When a calls to a home phone which has 3 connection (B, C, D) lands on
> proxy, then proxy forks the INVITE message to B, C, D. so every phone
> starts ringing. This is the first character of Home Phone. Now Let's say
> 
> B picks up the phone then Proxy sends CANCEL message to C and D so that
> the ringing can stop. This fulfills the Second point of Home Phone.
> 
> I am not able to get the call flow for the Third point in home Phone. If
> 
> lets say C picks up the phone (when both A and C in conversation)then he
> should also be in session with A
> and B. How can this be done because When a CANCEL message is received by
> 
> C and D then they both reached the state where they can only expect
> INVITE.
> 
> Can any one please tell me how the call flow for the third point in the
> home phone.
> 
> The above points of Home phone were taken from the -----Bell Labs
> Journal (October-December 1998)
> 
> regards
> Anirban

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Rosen, Brian" <brosen@fore.com>
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Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: RE: not able to understand the call flow for Home Phone implement
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> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home 
> line service,
> it might be something worthwhile to consider.
> 

Count me in.  It has applications beyond home.

Brian



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 10:48:11 2000
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        jdrosen@dynamicsoft.com (Jonathan Rosenberg)
From: Bob Bell <rtbell@cisco.com>
Subject: Re: Prioritization limitation question
Cc: Adam.Roach@ericsson.com (Adam B. Roach), brosen@fore.com (Rosen Brian),
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Gentlemen -

There is another aspect of MLPP which is being ignored here. That is that I 
don't want to talk to you, I just need the bandwidth your communications is 
consuming over a WAN link of limited bandwidth. So, I (or a proxy which has 
a better network view that I do) needs to tell you that you must terminate 
the call. Thus, there is no opportunity for me to present an INVITE to you 
and for you to return a Busy indication.

Bob

At 04:50 PM  1/5/00, Adam B. Roach wrote:

>Jonathan Rosenberg writes:
>
> >Rather than using this MLPP-Version thing to indicate support for CCBS,
> >I'd rather use the more generic Supported/Require mechanism (a draft on
> >this is pending within the next few days). So,
> >
> >C->S INVITE
> >
> >S->C 421 Feature Available
> >Require: com.cisco.mlpp
> >
> >C->S INVITE
> >Supported: com.cisco.mlpp
> >MLPP: 2;version=1
> >
> >S->C 200 OK
>
>It sounds like you're adding yet another round trip:
>the first to agree on the feature, the second to try
>an invite (which gets a "busy" response), and the third
>to invoke the feature (am I misreading something here?)
>
>Keep in mind that MLPP is relevant only when the far side
>returns an indication like "busy." Only *after* the calling
>party is notified that the far side is busy does he make a
>decision that he wants to break in on the call.
>
>Also, in PSTN interoperation scenarios, the gateway won't
>know that you can break in on the called party until the busy
>indication arrives (since we're just pumping the call back
>into the network and don't necessarily know the capability
>of the end office to which the subscriber is physically
>connected) -- so negotiation of the feature cannot occur before
>an attempt is made to reach the called party.
>
>If we're going to go out of our way to borrow a service from
>ISUP, we'd be exceedingly foolish not to make them able to
>interoperate.
>
>--
>Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
>adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA
>

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)



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From: Bob Bell <rtbell@cisco.com>
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  implement ation
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I also would have a personal interest in this activity. So count me in also.

Bob Bell

At 08:21 AM  1/6/00, Rosen, Brian wrote:
>
> > Are many people interested in this? If there are many folks who would
> > like to see some work on SIP extensions required for home
> > line service,
> > it might be something worthwhile to consider.
> >
>
>Count me in.  It has applications beyond home.
>
>Brian
>

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)



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	Hi all!

	Can anyone name a good book concerning SIP?

	Petri Saarela




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 12:02:08 2000
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Date: Thu, 06 Jan 2000 12:05:08 -0500
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There are no books (yet ;) ) that I know of. However, there is a wealth
of papers, tutorials, and slides at the SIP homepage:

http://www.cs.columbia.edu/~hgs/sip/papers.html

-Jonathan R.

petri.saarela@nokia.com wrote:
> 
>         Hi all!
> 
>         Can anyone name a good book concerning SIP?
> 
>         Petri Saarela

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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        sip@lists.research.bell-labs.com
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This is a much different service. As the SIP path and media path are
different (and, in fact, may flow through completely different service
providers), there is really no way for this to be handled at the SIP
layer. A proxy can't know to tear down someone elses call if bandwidth
were available. Even the UA probably can't know that his call is
consuming bandwidth that is preventing some other call from being made.

Rather, if the *service* we need is that some generals call gets enough
bandwidth, overriding some private, this is best handled by some kind of
diffserv or rsvp operation.


-Jonathan R.

Bob Bell wrote:
> 
> Gentlemen -
> 
> There is another aspect of MLPP which is being ignored here. That is that I
> don't want to talk to you, I just need the bandwidth your communications is
> consuming over a WAN link of limited bandwidth. So, I (or a proxy which has
> a better network view that I do) needs to tell you that you must terminate
> the call. Thus, there is no opportunity for me to present an INVITE to you
> and for you to return a Busy indication.
> 
> Bob
> 
> At 04:50 PM  1/5/00, Adam B. Roach wrote:
> 
> >Jonathan Rosenberg writes:
> >
> > >Rather than using this MLPP-Version thing to indicate support for CCBS,
> > >I'd rather use the more generic Supported/Require mechanism (a draft on
> > >this is pending within the next few days). So,
> > >
> > >C->S INVITE
> > >
> > >S->C 421 Feature Available
> > >Require: com.cisco.mlpp
> > >
> > >C->S INVITE
> > >Supported: com.cisco.mlpp
> > >MLPP: 2;version=1
> > >
> > >S->C 200 OK
> >
> >It sounds like you're adding yet another round trip:
> >the first to agree on the feature, the second to try
> >an invite (which gets a "busy" response), and the third
> >to invoke the feature (am I misreading something here?)
> >
> >Keep in mind that MLPP is relevant only when the far side
> >returns an indication like "busy." Only *after* the calling
> >party is notified that the far side is busy does he make a
> >decision that he wants to break in on the call.
> >
> >Also, in PSTN interoperation scenarios, the gateway won't
> >know that you can break in on the called party until the busy
> >indication arrives (since we're just pumping the call back
> >into the network and don't necessarily know the capability
> >of the end office to which the subscriber is physically
> >connected) -- so negotiation of the feature cannot occur before
> >an attempt is made to reach the called party.
> >
> >If we're going to go out of our way to borrow a service from
> >ISUP, we'd be exceedingly foolish not to make them able to
> >interoperate.
> >
> >--
> >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
> >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA
> >
> 
> Bob Bell
> Cisco Systems Inc.
> 801-294-3034(v)
> 801-294-3023(f)

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Thu, 06 Jan 2000 12:18:52 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: "Adam B. Roach" <Adam.Roach@ericsson.com>
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"Adam B. Roach" wrote:
> 
> >C->S INVITE
> >
> >S->C 421 Feature Available
> >Require: com.cisco.mlpp
> >
> >C->S INVITE
> >Supported: com.cisco.mlpp
> >MLPP: 2;version=1
> >
> >S->C 200 OK
> 
> It sounds like you're adding yet another round trip:
> the first to agree on the feature, the second to try
> an invite (which gets a "busy" response), and the third
> to invoke the feature (am I misreading something here?)

I wasn't suggesting that. I was suggesting that if the far side is busy,
and has this CCBS thing, it returns a 421 feature available with this 
Require header.

Anyway, based on Bob's mail, it may not matter.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hello SIP-er's;

IP Telephony, by Hersent, Gurle & Petit, has a Chapter (# 2) on SIP, which

seems fairly good (although I haven't read that far yet)!
It is out, as I have one in front of me, purchased last week from Borders
for $49.95.

Published by Addison-Wesley, 2000.


Regards

Marshall Eubanks

T.M. Eubanks
CTO
Multicast Technologies, LLC
P.O. Box 99
Clifton, Virginia 20124
Phone : 703-803-8141
Fax     : 703-222-3250
e-mail : tme@21rst-century.com



Jonathan Rosenberg wrote:

> There are no books (yet ;) ) that I know of. However, there is a wealth
> of papers, tutorials, and slides at the SIP homepage:
>
> http://www.cs.columbia.edu/~hgs/sip/papers.html
>
> -Jonathan R.
>
> petri.saarela@nokia.com wrote:
> >
> >         Hi all!
> >
> >         Can anyone name a good book concerning SIP?
> >
> >         Petri Saarela
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com







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Date: Thu, 06 Jan 2000 12:13:41 -0700
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Bob Bell <rtbell@cisco.com>
Subject: Re: Prioritization limitation question
Cc: "Adam B. Roach" <Adam.Roach@ericsson.com>, Rosen Brian <brosen@fore.com>,
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Jonathan -

I understand that this is a different aspect of the problem. It does bear 
on SIP in that if a lower priority call is "preempted", there is supposed 
to be a signal given to the end stations indicating that they have been or 
are being preempted. That signal would need to be SIP. How and where it is 
originated is a totally different issue and one I don't currently know how 
to solve. There would need to be some form of feedback from RSVP or 
DIFFSERV or ???? to allow for that interaction.

Bob Bell

At 10:11 AM  1/6/00, Jonathan Rosenberg wrote:
>This is a much different service. As the SIP path and media path are
>different (and, in fact, may flow through completely different service
>providers), there is really no way for this to be handled at the SIP
>layer. A proxy can't know to tear down someone elses call if bandwidth
>were available. Even the UA probably can't know that his call is
>consuming bandwidth that is preventing some other call from being made.
>
>Rather, if the *service* we need is that some generals call gets enough
>bandwidth, overriding some private, this is best handled by some kind of
>diffserv or rsvp operation.
>
>
>-Jonathan R.
>
>Bob Bell wrote:
> >
> > Gentlemen -
> >
> > There is another aspect of MLPP which is being ignored here. That is that I
> > don't want to talk to you, I just need the bandwidth your communications is
> > consuming over a WAN link of limited bandwidth. So, I (or a proxy which has
> > a better network view that I do) needs to tell you that you must terminate
> > the call. Thus, there is no opportunity for me to present an INVITE to you
> > and for you to return a Busy indication.
> >
> > Bob
> >
> > At 04:50 PM  1/5/00, Adam B. Roach wrote:
> >
> > >Jonathan Rosenberg writes:
> > >
> > > >Rather than using this MLPP-Version thing to indicate support for CCBS,
> > > >I'd rather use the more generic Supported/Require mechanism (a draft on
> > > >this is pending within the next few days). So,
> > > >
> > > >C->S INVITE
> > > >
> > > >S->C 421 Feature Available
> > > >Require: com.cisco.mlpp
> > > >
> > > >C->S INVITE
> > > >Supported: com.cisco.mlpp
> > > >MLPP: 2;version=1
> > > >
> > > >S->C 200 OK
> > >
> > >It sounds like you're adding yet another round trip:
> > >the first to agree on the feature, the second to try
> > >an invite (which gets a "busy" response), and the third
> > >to invoke the feature (am I misreading something here?)
> > >
> > >Keep in mind that MLPP is relevant only when the far side
> > >returns an indication like "busy." Only *after* the calling
> > >party is notified that the far side is busy does he make a
> > >decision that he wants to break in on the call.
> > >
> > >Also, in PSTN interoperation scenarios, the gateway won't
> > >know that you can break in on the called party until the busy
> > >indication arrives (since we're just pumping the call back
> > >into the network and don't necessarily know the capability
> > >of the end office to which the subscriber is physically
> > >connected) -- so negotiation of the feature cannot occur before
> > >an attempt is made to reach the called party.
> > >
> > >If we're going to go out of our way to borrow a service from
> > >ISUP, we'd be exceedingly foolish not to make them able to
> > >interoperate.
> > >
> > >--
> > >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS 
> L-04
> > >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 
> 75081 USA
> > >
> >
> > Bob Bell
> > Cisco Systems Inc.
> > 801-294-3034(v)
> > 801-294-3023(f)
>
>--
>Jonathan D. Rosenberg                       200 Executive Drive
>Chief Scientist                             Suite 120
>dynamicsoft                                 West Orange, NJ 07052
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)



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Date: Thu, 06 Jan 2000 13:11:49 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: not able to understand the call flow for Home Phone implementation
In-reply-to: <4FBEA8857476D311A03300204840E1CF0685FC@whq-msgusr-02.fore.com>
To: "Rosen, Brian" <brosen@fore.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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>It has applications beyond home.
Agree. 

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Rosen, Brian
Sent: Thursday, January 06, 2000 9:22 AM
To: 'Jonathan Rosenberg'
Cc: 'sip@lists.research.bell-labs.com'
Subject: RE: not able to understand the call flow for Home Phone
implement ation


 
> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home 
> line service,
> it might be something worthwhile to consider.
> 

Count me in.  It has applications beyond home.

Brian




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Date: Thu, 06 Jan 2000 12:15:35 -0700
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "Adam B. Roach" <Adam.Roach@ericsson.com>
From: Bob Bell <rtbell@cisco.com>
Subject: Re: Prioritization limitation question
Cc: Rosen Brian <brosen@fore.com>, sip@lists.research.bell-labs.com
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Jonathan -

This form, where I need to preempt a call that you have currently active so 
that I can talk to you is also a valid case. Don't abandon this part just 
because the other part also exists.

Bob

At 10:18 AM  1/6/00, Jonathan Rosenberg wrote:


>"Adam B. Roach" wrote:
> >
> > >C->S INVITE
> > >
> > >S->C 421 Feature Available
> > >Require: com.cisco.mlpp
> > >
> > >C->S INVITE
> > >Supported: com.cisco.mlpp
> > >MLPP: 2;version=1
> > >
> > >S->C 200 OK
> >
> > It sounds like you're adding yet another round trip:
> > the first to agree on the feature, the second to try
> > an invite (which gets a "busy" response), and the third
> > to invoke the feature (am I misreading something here?)
>
>I wasn't suggesting that. I was suggesting that if the far side is busy,
>and has this CCBS thing, it returns a 421 feature available with this
>Require header.
>
>Anyway, based on Bob's mail, it may not matter.
>
>-Jonathan R.
>
>--
>Jonathan D. Rosenberg                       200 Executive Drive
>Chief Scientist                             Suite 120
>dynamicsoft                                 West Orange, NJ 07052
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)



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Subject: RE: not able to understand the call flow for Home Phone implementation
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Yes  I'm definitely interested.
Again beyond the home applications

Mike

-----Original Message-----
From: brosen [mailto:brosen@fore.com]
Sent: Thursday, January 06, 2000 7:22 AM
To: jdrosen
Cc: brosen; sip
Subject: RE: not able to understand the call flow for Home Phone
implementation


 
> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home 
> line service,
> it might be something worthwhile to consider.
> 

Count me in.  It has applications beyond home.

Brian



--openmail-part-0fe51ed0-00000001--




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From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: Prioritization limitation question
In-reply-to: <4.2.0.58.20000106121019.00ae37d0@orinoco.cisco.com>
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Bob, I will give it a try.
A QoS enabled SIP client, like a SIP phone would actually have an RSVP
client as well. If bandwidth is preempted during a QoS assured call, it is
up to the client to react and advise the user with a notice "Lost QoS for
this call" and give the user the choice to continue the call with best
effort service or to hang up. Since QoS assured calls will probably command
some premium rates, the knowledgeable user may not be completely unhappy not
to be charged for QoS minutes any more. It will also allow for a more
graceful termination of the call by the user himself/herself instead of
being just cut off in the middle of a conversation. Or do I understand the
problem right?

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Bob Bell
Sent: Thursday, January 06, 2000 1:14 PM
To: Jonathan Rosenberg
Cc: Adam B. Roach; Rosen Brian; sip@lists.research.bell-labs.com
Subject: Re: Prioritization limitation question


Jonathan -

I understand that this is a different aspect of the problem. It does bear
on SIP in that if a lower priority call is "preempted", there is supposed
to be a signal given to the end stations indicating that they have been or
are being preempted. That signal would need to be SIP. How and where it is
originated is a totally different issue and one I don't currently know how
to solve. There would need to be some form of feedback from RSVP or
DIFFSERV or ???? to allow for that interaction.

Bob Bell

At 10:11 AM  1/6/00, Jonathan Rosenberg wrote:
>This is a much different service. As the SIP path and media path are
>different (and, in fact, may flow through completely different service
>providers), there is really no way for this to be handled at the SIP
>layer. A proxy can't know to tear down someone elses call if bandwidth
>were available. Even the UA probably can't know that his call is
>consuming bandwidth that is preventing some other call from being made.
>
>Rather, if the *service* we need is that some generals call gets enough
>bandwidth, overriding some private, this is best handled by some kind of
>diffserv or rsvp operation.
>
>
>-Jonathan R.
>
>Bob Bell wrote:
> >
> > Gentlemen -
> >
> > There is another aspect of MLPP which is being ignored here. That is
that I
> > don't want to talk to you, I just need the bandwidth your communications
is
> > consuming over a WAN link of limited bandwidth. So, I (or a proxy which
has
> > a better network view that I do) needs to tell you that you must
terminate
> > the call. Thus, there is no opportunity for me to present an INVITE to
you
> > and for you to return a Busy indication.
> >
> > Bob
> >
> > At 04:50 PM  1/5/00, Adam B. Roach wrote:
> >
> > >Jonathan Rosenberg writes:
> > >
> > > >Rather than using this MLPP-Version thing to indicate support for
CCBS,
> > > >I'd rather use the more generic Supported/Require mechanism (a draft
on
> > > >this is pending within the next few days). So,
> > > >
> > > >C->S INVITE
> > > >
> > > >S->C 421 Feature Available
> > > >Require: com.cisco.mlpp
> > > >
> > > >C->S INVITE
> > > >Supported: com.cisco.mlpp
> > > >MLPP: 2;version=1
> > > >
> > > >S->C 200 OK
> > >
> > >It sounds like you're adding yet another round trip:
> > >the first to agree on the feature, the second to try
> > >an invite (which gets a "busy" response), and the third
> > >to invoke the feature (am I misreading something here?)
> > >
> > >Keep in mind that MLPP is relevant only when the far side
> > >returns an indication like "busy." Only *after* the calling
> > >party is notified that the far side is busy does he make a
> > >decision that he wants to break in on the call.
> > >
> > >Also, in PSTN interoperation scenarios, the gateway won't
> > >know that you can break in on the called party until the busy
> > >indication arrives (since we're just pumping the call back
> > >into the network and don't necessarily know the capability
> > >of the end office to which the subscriber is physically
> > >connected) -- so negotiation of the feature cannot occur before
> > >an attempt is made to reach the called party.
> > >
> > >If we're going to go out of our way to borrow a service from
> > >ISUP, we'd be exceedingly foolish not to make them able to
> > >interoperate.
> > >
> > >--
> > >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS
> L-04
> > >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
> 75081 USA
> > >
> >
> > Bob Bell
> > Cisco Systems Inc.
> > 801-294-3034(v)
> > 801-294-3023(f)
>
>--
>Jonathan D. Rosenberg                       200 Executive Drive
>Chief Scientist                             Suite 120
>dynamicsoft                                 West Orange, NJ 07052
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com

Bob Bell
Cisco Systems Inc.
801-294-3034(v)
801-294-3023(f)




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The first problem is that there is no way I know of in RSVP for one
reservation to preempt another, and for notification of preemption to be
delivered (there has been some work on MPLS extensions to RSVP which
allow replacement, but I don't know if thats useful for this
application). 

Even if it did, I'm not really sure it matters. This isn't a circuit
network; pre-emption does not imply a one to one replacement of one
circuit with another. One could add priorities to an rsvp reservation so
it always succeeds; all this means is that the loss rate of existing
connections is increased if there really wasn't enough bandwidth for the
new one. You can rely on human response to that to hang up if it gets
really bad. In most cases, I suspect a WAN connection that supports any
kind of data traffic at all will have sufficient resources so that these
high priority reservations won't impact one drop existing calls.
Furthermore, adaptive mechanisms that use FEC or redundanct codecs will
allow performance to be probably not bad for loss rates that are quite
high.

-Jonathan R.



Bob Bell wrote:
> 
> Jonathan -
> 
> I understand that this is a different aspect of the problem. It does bear
> on SIP in that if a lower priority call is "preempted", there is supposed
> to be a signal given to the end stations indicating that they have been or
> are being preempted. That signal would need to be SIP. How and where it is
> originated is a totally different issue and one I don't currently know how
> to solve. There would need to be some form of feedback from RSVP or
> DIFFSERV or ???? to allow for that interaction.
> 
> Bob Bell
> 
> At 10:11 AM  1/6/00, Jonathan Rosenberg wrote:
> >This is a much different service. As the SIP path and media path are
> >different (and, in fact, may flow through completely different service
> >providers), there is really no way for this to be handled at the SIP
> >layer. A proxy can't know to tear down someone elses call if bandwidth
> >were available. Even the UA probably can't know that his call is
> >consuming bandwidth that is preventing some other call from being made.
> >
> >Rather, if the *service* we need is that some generals call gets enough
> >bandwidth, overriding some private, this is best handled by some kind of
> >diffserv or rsvp operation.
> >
> >
> >-Jonathan R.
> >
> >Bob Bell wrote:
> > >
> > > Gentlemen -
> > >
> > > There is another aspect of MLPP which is being ignored here. That is that I
> > > don't want to talk to you, I just need the bandwidth your communications is
> > > consuming over a WAN link of limited bandwidth. So, I (or a proxy which has
> > > a better network view that I do) needs to tell you that you must terminate
> > > the call. Thus, there is no opportunity for me to present an INVITE to you
> > > and for you to return a Busy indication.
> > >
> > > Bob
> > >
> > > At 04:50 PM  1/5/00, Adam B. Roach wrote:
> > >
> > > >Jonathan Rosenberg writes:
> > > >
> > > > >Rather than using this MLPP-Version thing to indicate support for CCBS,
> > > > >I'd rather use the more generic Supported/Require mechanism (a draft on
> > > > >this is pending within the next few days). So,
> > > > >
> > > > >C->S INVITE
> > > > >
> > > > >S->C 421 Feature Available
> > > > >Require: com.cisco.mlpp
> > > > >
> > > > >C->S INVITE
> > > > >Supported: com.cisco.mlpp
> > > > >MLPP: 2;version=1
> > > > >
> > > > >S->C 200 OK
> > > >
> > > >It sounds like you're adding yet another round trip:
> > > >the first to agree on the feature, the second to try
> > > >an invite (which gets a "busy" response), and the third
> > > >to invoke the feature (am I misreading something here?)
> > > >
> > > >Keep in mind that MLPP is relevant only when the far side
> > > >returns an indication like "busy." Only *after* the calling
> > > >party is notified that the far side is busy does he make a
> > > >decision that he wants to break in on the call.
> > > >
> > > >Also, in PSTN interoperation scenarios, the gateway won't
> > > >know that you can break in on the called party until the busy
> > > >indication arrives (since we're just pumping the call back
> > > >into the network and don't necessarily know the capability
> > > >of the end office to which the subscriber is physically
> > > >connected) -- so negotiation of the feature cannot occur before
> > > >an attempt is made to reach the called party.
> > > >
> > > >If we're going to go out of our way to borrow a service from
> > > >ISUP, we'd be exceedingly foolish not to make them able to
> > > >interoperate.
> > > >
> > > >--
> > > >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS
> > L-04
> > > >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
> > 75081 USA
> > > >
> > >
> > > Bob Bell
> > > Cisco Systems Inc.
> > > 801-294-3034(v)
> > > 801-294-3023(f)
> >
> >--
> >Jonathan D. Rosenberg                       200 Executive Drive
> >Chief Scientist                             Suite 120
> >dynamicsoft                                 West Orange, NJ 07052
> >jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> >http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> >http://www.dynamicsoft.com
> 
> Bob Bell
> Cisco Systems Inc.
> 801-294-3034(v)
> 801-294-3023(f)

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: Scott Bradner <sob@harvard.edu>
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To: jdrosen@dynamicsoft.com, rtbell@cisco.com
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> The first problem is that there is no way I know of in RSVP for one
> reservation to preempt another, and for notification of preemption to be
> delivered 

at least the 1st part is in COPS

Scott



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Date: Thu, 06 Jan 2000 16:05:04 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: Prioritization limitation question
In-reply-to: <3875096A.163E5DE6@dynamicsoft.com>
To: Bob Bell <rtbell@cisco.com>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@lists.research.bell-labs.com, Rosen Brian <brosen@fore.com>,
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An access network, such as an ISP or intranet may want to manage their
premium rate QoS pool from a policy server talking COPS to the edge router
(that enforces the policy) connecting to the IP backbone. If the policy
server is smart enough to understand priorities, it can de-install the QoS
policy for some unfortunate lower level callers and install precious QoS for
the higher priority application, SIP telephony or anything else. Maybe a
corporate video announcement from the president, or an emergency. The
initial draft for such a model is
http://search.ietf.org/internet-drafts/draft-sinnreich-interdomain-sip-qos-o
sp-00.txt
though we are planning on an improved version soon.

Interested to know if this model for QoS management from a policy server is
broken or OK, so your comments are appreciated. If the model holds, than we
have an answer to Bob's problem.

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
Rosenberg
Sent: Thursday, January 06, 2000 3:30 PM
To: Bob Bell
Cc: Adam B. Roach; Rosen Brian; sip@lists.research.bell-labs.com
Subject: Re: Prioritization limitation question


The first problem is that there is no way I know of in RSVP for one
reservation to preempt another, and for notification of preemption to be
delivered (there has been some work on MPLS extensions to RSVP which
allow replacement, but I don't know if thats useful for this
application).

Even if it did, I'm not really sure it matters. This isn't a circuit
network; pre-emption does not imply a one to one replacement of one
circuit with another. One could add priorities to an rsvp reservation so
it always succeeds; all this means is that the loss rate of existing
connections is increased if there really wasn't enough bandwidth for the
new one. You can rely on human response to that to hang up if it gets
really bad. In most cases, I suspect a WAN connection that supports any
kind of data traffic at all will have sufficient resources so that these
high priority reservations won't impact one drop existing calls.
Furthermore, adaptive mechanisms that use FEC or redundanct codecs will
allow performance to be probably not bad for loss rates that are quite
high.

-Jonathan R.



Bob Bell wrote:
>
> Jonathan -
>
> I understand that this is a different aspect of the problem. It does bear
> on SIP in that if a lower priority call is "preempted", there is supposed
> to be a signal given to the end stations indicating that they have been or
> are being preempted. That signal would need to be SIP. How and where it is
> originated is a totally different issue and one I don't currently know how
> to solve. There would need to be some form of feedback from RSVP or
> DIFFSERV or ???? to allow for that interaction.
>
> Bob Bell
>
> At 10:11 AM  1/6/00, Jonathan Rosenberg wrote:
> >This is a much different service. As the SIP path and media path are
> >different (and, in fact, may flow through completely different service
> >providers), there is really no way for this to be handled at the SIP
> >layer. A proxy can't know to tear down someone elses call if bandwidth
> >were available. Even the UA probably can't know that his call is
> >consuming bandwidth that is preventing some other call from being made.
> >
> >Rather, if the *service* we need is that some generals call gets enough
> >bandwidth, overriding some private, this is best handled by some kind of
> >diffserv or rsvp operation.
> >
> >
> >-Jonathan R.
> >
> >Bob Bell wrote:
> > >
> > > Gentlemen -
> > >
> > > There is another aspect of MLPP which is being ignored here. That is
that I
> > > don't want to talk to you, I just need the bandwidth your
communications is
> > > consuming over a WAN link of limited bandwidth. So, I (or a proxy
which has
> > > a better network view that I do) needs to tell you that you must
terminate
> > > the call. Thus, there is no opportunity for me to present an INVITE to
you
> > > and for you to return a Busy indication.
> > >
> > > Bob
> > >
> > > At 04:50 PM  1/5/00, Adam B. Roach wrote:
> > >
> > > >Jonathan Rosenberg writes:
> > > >
> > > > >Rather than using this MLPP-Version thing to indicate support for
CCBS,
> > > > >I'd rather use the more generic Supported/Require mechanism (a
draft on
> > > > >this is pending within the next few days). So,
> > > > >
> > > > >C->S INVITE
> > > > >
> > > > >S->C 421 Feature Available
> > > > >Require: com.cisco.mlpp
> > > > >
> > > > >C->S INVITE
> > > > >Supported: com.cisco.mlpp
> > > > >MLPP: 2;version=1
> > > > >
> > > > >S->C 200 OK
> > > >
> > > >It sounds like you're adding yet another round trip:
> > > >the first to agree on the feature, the second to try
> > > >an invite (which gets a "busy" response), and the third
> > > >to invoke the feature (am I misreading something here?)
> > > >
> > > >Keep in mind that MLPP is relevant only when the far side
> > > >returns an indication like "busy." Only *after* the calling
> > > >party is notified that the far side is busy does he make a
> > > >decision that he wants to break in on the call.
> > > >
> > > >Also, in PSTN interoperation scenarios, the gateway won't
> > > >know that you can break in on the called party until the busy
> > > >indication arrives (since we're just pumping the call back
> > > >into the network and don't necessarily know the capability
> > > >of the end office to which the subscriber is physically
> > > >connected) -- so negotiation of the feature cannot occur before
> > > >an attempt is made to reach the called party.
> > > >
> > > >If we're going to go out of our way to borrow a service from
> > > >ISUP, we'd be exceedingly foolish not to make them able to
> > > >interoperate.
> > > >
> > > >--
> > > >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho,
MS
> > L-04
> > > >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
> > 75081 USA
> > > >
> > >
> > > Bob Bell
> > > Cisco Systems Inc.
> > > 801-294-3034(v)
> > > 801-294-3023(f)
> >
> >--
> >Jonathan D. Rosenberg                       200 Executive Drive
> >Chief Scientist                             Suite 120
> >dynamicsoft                                 West Orange, NJ 07052
> >jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> >http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> >http://www.dynamicsoft.com
>
> Bob Bell
> Cisco Systems Inc.
> 801-294-3034(v)
> 801-294-3023(f)

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 17:21:52 2000
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Date: Thu, 06 Jan 2000 16:16:29 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: not able to understand the call flow for Home Phone implementation
In-reply-to: <3874AEBC.6F547BA2@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        ANIRBAN ROY <anirban.roy@wipro.com>
Cc: sip@lists.research.bell-labs.com
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Jonathan,

>Basically, if you use multicast instead of forking,
>you don't need protocol extensions, just some new behaviors in the UA.

I believe the zeroconf WG
http://www.ietf.org/html.charters/zeroconf-charter.html
is counting on multicast for home networks for a number of reasons.
Multicast may be a valid approach.

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
Rosenberg
Sent: Thursday, January 06, 2000 9:03 AM
To: ANIRBAN ROY
Cc: sip@lists.research.bell-labs.com
Subject: Re: not able to understand the call flow for Home Phone
implementation


You are right in that a SIP UA, as currently specified, won't be able to
get home line type of service. It can be done in the way described in
this paper, but it requires changes to the behavior of the UA, although
no changes to the protocol itself, depending on how its done. It also
requires the caller to be "cooperative" - they will receive multiple 200
OK as each extension picks up, and they need to initiate a multi-party
conference to do it. Basically, if you use multicast instead of forking,
you don't need protocol extensions, just some new behaviors in the UA.
If you want to rely on forking rather than multicast, you need somewhat
more complex extensions.

When using multicast, the UA basically snoops on the multicast group for
BYEs and other 200 OKs. This allows it to build a complete picture of
what other extensions have picked up, hung up, or been hung up on. Based
on this, it can, in a distributed fashion, figure out the state of the
call. If there are calls active, a user picking up the phone would cause
the phone to send a 200 OK for the call.

For forking, snooping of BYEs and other messages is not feasible, so
some extensions are needed. However, doing this means you home line
extensions could actually be anywhere in the world. Pretty cool.

There are other ways to do it which hide from the caller the fact that
multiple extensions have picked up. This is more like how the current
home phone service works.

Are many people interested in this? If there are many folks who would
like to see some work on SIP extensions required for home line service,
it might be something worthwhile to consider.

-Jonathan R.

ANIRBAN ROY wrote:
>
> hi,
> I am trying to figure out how to get a home phone functionality in SIP.
> The home phone has the following functionality
> 1    When a call comes all phones in the home ring
> 2    When one picks up, all other line must stop ringing
> 3    A user can picks up from any other telephone in the home and join
> an existing call.
>
>                                A scenario is there where A calls a Home
> Phone which has Three connection B, C, D in his home
>
> When a calls to a home phone which has 3 connection (B, C, D) lands on
> proxy, then proxy forks the INVITE message to B, C, D. so every phone
> starts ringing. This is the first character of Home Phone. Now Let's say
>
> B picks up the phone then Proxy sends CANCEL message to C and D so that
> the ringing can stop. This fulfills the Second point of Home Phone.
>
> I am not able to get the call flow for the Third point in home Phone. If
>
> lets say C picks up the phone (when both A and C in conversation)then he
> should also be in session with A
> and B. How can this be done because When a CANCEL message is received by
>
> C and D then they both reached the state where they can only expect
> INVITE.
>
> Can any one please tell me how the call flow for the third point in the
> home phone.
>
> The above points of Home phone were taken from the -----Bell Labs
> Journal (October-December 1998)
>
> regards
> Anirban

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 17:49:45 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Gateways and registration
Date: Thu, 6 Jan 2000 17:45:15 -0500 
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The SIP specification does not require that gateways register with a SIP
server.  Thus, the server is never notified if the gateway goes offline.  By
contrast, protocols such as H.323 have a different concept for Registration
and a server can detect when a gateway goes down or offline if registration
expires.

Is there some way to provide equivalent functionality in SIP?  If not, is
the concept of gateway registration under consideration for a future draft
of the spec?

Thanks.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 17:55:51 2000
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Date: Thu, 06 Jan 2000 16:21:22 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: A case related to "home Phone" type of functionality in SIP
In-reply-to: <3874123E.5E4F14DD@wipsys.soft.net>
To: ANIRBAN ROY <anirban.roy@wipro.com>, sip@lists.research.bell-labs.com
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In a business environment, we call this a "single line extension".

SIP approaches tend to have the property that each phone is uniquely
addressed, whereas in the PSTN each circuit is uniquely addressed.

The obvious solution is to combine a parallel search proxy and some UAS
behavior.

The sequence is:
1) Incoming call reaches parallel proxy.
2) Parallel proxy invites all UASes in extension group.
3) User answers one extension.
4) Parallel proxy cancels unanswered legs. Those phones stop ringing.
5) Answering phone send INVITE to other phones in group, with "NO RING"
specified somehow, and long expire timer.


Case 1: 2nd party answers at extension phone:
6) First answering phone bridges all call legs

Case 2: User hangs up answering phone
6) Answering phone CANCELS open INVITES to other phones in group
7) Answering phone BYES all active call legs

Case 3: 3rd party places call to extension phone
6) Extension phone rings, ignoring open INVITE from first answering phone or
leaving it "attached" to a line button

etc.

Another approach is to use a chained INVITE, which has been propsed using
INVITE with Also headers, along with a conferencing bridge:

The sequence here is:
1) Incoming call reaches extension group proxy
2) Extension group proxy INVITES or extension bridge, with an Also: or
whatever header for each phone in extension group
3) Extension bridge INVITES each phone in group
3) User answeres one extension
4) Extension bridge OKs call leg to user
5) Extension bridge OKs call from caller
6) Exenstion bridge CANCELS legs to other phones in group
7) Extension bridge reINVITES other phones in group with a "NO RING" option
and long expire timer.
. . .
with similar tear-down and 2nd call scenarios

And there are dozens of other variations on these themes possible.

--
Dean


> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of ANIRBAN ROY
> Sent: Wednesday, January 05, 2000 9:56 PM
> To: sip@lists.research.bell-labs.com
> Subject: A case related to "home Phone" type of functionality in SIP
>
>
> hi,
> I am trying to figure out how to get a home phone functionality in SIP.
> The home phone has the following functionality
> 1    When a call comes all phones in the home ring
> 2    When one picks up, all other line must stop ringing
> 3    A user can picks up from any other telephone in the home and join
> an existing call.
>
>
>                                 this a scenario where A calls a Home
> Phone which has Three connection B, C, D in his home
>
>
>
> +-------------|
>                                                     A---------------|
> Proxy        |---------------D
>
> |                    |
>
> +------------+
>
> /     \
>
> /       \
>
> /         \
>
> /           \
>
> B           C
>
>
> When a calls to a home phone which has 3 connection (B, C, D) lands on
> proxy, then proxy forks the INVITE message to B, C, D. so every phone
> starts ringing. This is the first character of Home Phone. Now Let's say
> B picks up the phone then Proxy sends CANCEL message to C and D so that
> the ringing can stop. This fulfills the Second point of Home Phone.
>
> I am not able to get the call flow for the Third point in home Phone. If
> lets say C picks up the phone then he should also be in session with A
> and B. How can this be done because When a CANCEL message is received by
> C and D then they both reached the state where they can only expect
> INVITE.
>
>
> Can any one please tell me how the call flow for the third point in the
> home phone.
>
> The above points of Home phone were taken from the -----Bell Labs
> Journal (October-December 1998)
>
>
>
> regards
> Anirban
>
>




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This function doesn't belong in the realm of SIP, IMHO. Rather, it is
within the realm of TRIP, which supports learning of when a gateway goes
offline.


-Jonathan R.

Linden deCarmo wrote:
> 
> The SIP specification does not require that gateways register with a SIP
> server.  Thus, the server is never notified if the gateway goes offline.  By
> contrast, protocols such as H.323 have a different concept for Registration
> and a server can detect when a gateway goes down or offline if registration
> expires.
> 
> Is there some way to provide equivalent functionality in SIP?  If not, is
> the concept of gateway registration under consideration for a future draft
> of the spec?
> 
> Thanks.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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To: sip@lists.research.bell-labs.com, ldeCarmo@netspeak.com
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> 
> The SIP specification does not require that gateways register with a SIP
> server.  Thus, the server is never notified if the gateway goes offline.  By
> contrast, protocols such as H.323 have a different concept for Registration
> and a server can detect when a gateway goes down or offline if registration
> expires.
> 
> Is there some way to provide equivalent functionality in SIP?  If not, is
> the concept of gateway registration under consideration for a future draft
> of the spec?
> 

Although not required by the SIP standard, there is absolutely nothing to 
keep you from doing this other than the fact that there are better 
mechanisms. TRIP might be one avenue to consider. 

> Thanks.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487

-----------------------------------------------------------------
Sean Olson            mailto:sean.olson@ericsson.com
Ericsson Inc.         tel:(972)583-5472 
                      fax:(972)669-0154




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To: dean.willis@wcom.com
Subject: RE: A case related to "home Phone" type of functionality in SIP
Cc: sip@lists.research.bell-labs.com
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> 
> In a business environment, we call this a "single line extension".
> 
> SIP approaches tend to have the property that each phone is uniquely
> addressed, whereas in the PSTN each circuit is uniquely addressed.
> 
> The obvious solution is to combine a parallel search proxy and some UAS
> behavior.
> 
> The sequence is:
> 1) Incoming call reaches parallel proxy.
> 2) Parallel proxy invites all UASes in extension group.
> 3) User answers one extension.
> 4) Parallel proxy cancels unanswered legs. Those phones stop ringing.
> 5) Answering phone send INVITE to other phones in group, with "NO RING"
> specified somehow, and long expire timer.
> 
> 

In a business environment, this is a perfectly acceptable and reasonable
solution. For a home environment, multicast/DHCP seems more reasonable
because Joe consumer can go to Radio Shack, buy a 3com SIP phone, and
plug it into his home IP network and it works with zero (or 
almost-zero) configuration. Plus, it doesn't require the home user 
to buy/maintain a parallel proxy.

(By multicast, I mean both signalling and media).

-----------------------------------------------------------------
Sean Olson            mailto:sean.olson@ericsson.com
Ericsson Inc.         tel:(972)583-5472 
                      fax:(972)669-0154




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 20:48:47 2000
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Date: Thu, 06 Jan 2000 19:43:04 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: A case related to "home Phone" type of functionality in SIP
In-reply-to: <200001062349.RAA02400@b04a45.exu.ericsson.se>
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Are you saying that the first phone would multicast the media streams to the
others on the local LAN, or that the INVITES and media would be sent
multicast across the ISP network?

If the former, it might work. COuld be pretty cool.

If the latter, no, it's not going to happen that way.

1) No ISP that I know of will provide a multicast feed, especially for a
lowly dialup user.

2) Every SIP phone in the country is going to listen to a universal
multicast group, then make application discard decisions on the INVITES that
don't apply to it  . . . Or, we could statically allocate a multicast
address for each home network . . .not. The INVITE will come in UNICAST.

3) Personally, I've never managed to get multicast routing working right
over the bizarre equipment that has accumulated at my house.

4) In our own internal network, we often saw join times that took seven or
eight minutes to propaget back to the source of a multicast. That's some
serious post-dial delay. I'm not sure we even try to route multicast
anymore.

5) The firewall issues that occur with cross-domain multicast are mind
boggling. Anything coming into my house is by-definition cross-domain.

Hey, I love multicast. I spent more than two years building really cool
businesses (Real Broadcast Netork, skyMCI, etc.) around it, and I'm here to
tell you it just isn't going to happen across an Internet backbone for the
average user any time soon.

--
Dean

> -----Original Message-----
> From: Sean Olson [mailto:eussean@exu.ericsson.se]
> Sent: Thursday, January 06, 2000 5:49 PM
> To: dean.willis@wcom.com
> Cc: sip@lists.research.bell-labs.com
> Subject: RE: A case related to "home Phone" type of functionality in SIP
>
>
> >
> > In a business environment, we call this a "single line extension".
> >
> > SIP approaches tend to have the property that each phone is uniquely
> > addressed, whereas in the PSTN each circuit is uniquely addressed.
> >
> > The obvious solution is to combine a parallel search proxy and some UAS
> > behavior.
> >
> > The sequence is:
> > 1) Incoming call reaches parallel proxy.
> > 2) Parallel proxy invites all UASes in extension group.
> > 3) User answers one extension.
> > 4) Parallel proxy cancels unanswered legs. Those phones stop ringing.
> > 5) Answering phone send INVITE to other phones in group, with "NO RING"
> > specified somehow, and long expire timer.
> >
> >
>
> In a business environment, this is a perfectly acceptable and reasonable
> solution. For a home environment, multicast/DHCP seems more reasonable
> because Joe consumer can go to Radio Shack, buy a 3com SIP phone, and
> plug it into his home IP network and it works with zero (or
> almost-zero) configuration. Plus, it doesn't require the home user
> to buy/maintain a parallel proxy.
>
> (By multicast, I mean both signalling and media).
>
> -----------------------------------------------------------------
> Sean Olson            mailto:sean.olson@ericsson.com
> Ericsson Inc.         tel:(972)583-5472
>                       fax:(972)669-0154
>




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan  6 22:37:54 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: not able to understand the call flow for Home Phone implementation
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This is not far from, if at all different than, what some call "multiple
station appearances".  This is a feature for which I would like to see a
a standard solution.  We proposed this to the TIA as a recommended
feature for SIP IP phones.

Jonathan Rosenberg wrote:

> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home line service,
> it might be something worthwhile to consider.




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 01:00:01 2000
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Cc: sip@lists.research.bell-labs.com
Subject: Re: A case related to "home Phone" type of functionality in SIP
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From: Magnus Ahltorp <map@stacken.kth.se>
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> The home phone has the following functionality
> 1    When a call comes all phones in the home ring
> 2    When one picks up, all other line must stop ringing
> 3    A user can picks up from any other telephone in the home and join
> an existing call.

The third point is not always a feature of a home phone system. For
example, in Sweden, the default is a priority system, where the phones
are connected in serial, and only one phone at a time can be in use.
This is supposed to make it harder for people to listen in to other
people's calls.

I would rather want a dial tone in my ear when I lift the phone in
point three, for convenience and privacy.

/Magnus



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To: Dean Willis <dean.willis@wcom.com>
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Subject: Re: A case related to "home Phone" type of functionality in SIP
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hi ,

lets say in a house there are 3 phones B, C, D. now User "A" calls to this house

Please tell me one question

When a call is going on between two user A and B and the third person "C" picks
up the phone from the same house what happens then????.

#       does a 200 Response goes back to the caller. Or the 200 Response just
append the Multicast Address with address "C".

#   RTP data goes directly from caller to callee. When there are more than one
user is in conversation how Proxy makes it sure that the RTP data will be send
to particular multicast address. As because Proxy only work in the signalling
layer.


Please reply

Anirban

Dean Willis wrote:

> Are you saying that the first phone would multicast the media streams to the
> others on the local LAN, or that the INVITES and media would be sent
> multicast across the ISP network?
>
> If the former, it might work. COuld be pretty cool.
>
> If the latter, no, it's not going to happen that way.
>
> 1) No ISP that I know of will provide a multicast feed, especially for a
> lowly dialup user.
>
> 2) Every SIP phone in the country is going to listen to a universal
> multicast group, then make application discard decisions on the INVITES that
> don't apply to it  . . . Or, we could statically allocate a multicast
> address for each home network . . .not. The INVITE will come in UNICAST.
>
> 3) Personally, I've never managed to get multicast routing working right
> over the bizarre equipment that has accumulated at my house.
>
> 4) In our own internal network, we often saw join times that took seven or
> eight minutes to propaget back to the source of a multicast. That's some
> serious post-dial delay. I'm not sure we even try to route multicast
> anymore.
>
> 5) The firewall issues that occur with cross-domain multicast are mind
> boggling. Anything coming into my house is by-definition cross-domain.
>
> Hey, I love multicast. I spent more than two years building really cool
> businesses (Real Broadcast Netork, skyMCI, etc.) around it, and I'm here to
> tell you it just isn't going to happen across an Internet backbone for the
> average user any time soon.
>
> --
> Dean
>
> > -----Original Message-----
> > From: Sean Olson [mailto:eussean@exu.ericsson.se]
> > Sent: Thursday, January 06, 2000 5:49 PM
> > To: dean.willis@wcom.com
> > Cc: sip@lists.research.bell-labs.com
> > Subject: RE: A case related to "home Phone" type of functionality in SIP
> >
> >
> > >
> > > In a business environment, we call this a "single line extension".
> > >
> > > SIP approaches tend to have the property that each phone is uniquely
> > > addressed, whereas in the PSTN each circuit is uniquely addressed.
> > >
> > > The obvious solution is to combine a parallel search proxy and some UAS
> > > behavior.
> > >
> > > The sequence is:
> > > 1) Incoming call reaches parallel proxy.
> > > 2) Parallel proxy invites all UASes in extension group.
> > > 3) User answers one extension.
> > > 4) Parallel proxy cancels unanswered legs. Those phones stop ringing.
> > > 5) Answering phone send INVITE to other phones in group, with "NO RING"
> > > specified somehow, and long expire timer.
> > >
> > >
> >
> > In a business environment, this is a perfectly acceptable and reasonable
> > solution. For a home environment, multicast/DHCP seems more reasonable
> > because Joe consumer can go to Radio Shack, buy a 3com SIP phone, and
> > plug it into his home IP network and it works with zero (or
> > almost-zero) configuration. Plus, it doesn't require the home user
> > to buy/maintain a parallel proxy.
> >
> > (By multicast, I mean both signalling and media).
> >
> > -----------------------------------------------------------------
> > Sean Olson           mailto:sean.olson@ericsson.com
> > Ericsson Inc.         tel:(972)583-5472
> >                       fax:(972)669-0154
> >




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Subject: General question on SIP used b/w MGCs
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Hi,
I recently went thru all the drafts related to PSTN interworking in
hgs' site.

Now the impression I might have incorrectly formed seems to be that
all SIP is doing is providing a wrapper for any payload (ISUP, etc) that
needs to be transmitted b/w MGCs.

I had a look at both the INFO and the MIME Types - my current
understanding is that if an MGC wants to transmit a signal to another MGC,
it just creates a SIP Message (INVITE, INFO etc) and just puts the
complete signal
to be transmitted in the boddy. 
Now the receiving MGC extracts the body and interprets the signal in what
ever way it was meant to be interpreted.

Now if SIP is only acting as a wrapper, why is it that people seem to
mention that SIP is the preferred interaction scheme b/w MGC - MGC ?

Could some one point me to papers (besides the pstn drafts I mentioned)
helping me out with this ?

Actually the statements abovr must be quite naive, as I dont know much
about the MGC side  :-)

Anyway, I would like to know if besides being just a wrapper which is only
tunelling signals to and fro, is there any other advantage of SIP b/w two
MGCs ?

Thx
Regds
Arjun
Arjun Roychowdhury @ Hughes Software Systems



On Thu, 6 Jan 2000, Jonathan Rosenberg wrote:

> 
> 
> 
> 
> The first problem is that there is no way I know of in RSVP for one
> reservation to preempt another, and for notification of preemption to be
> delivered (there has been some work on MPLS extensions to RSVP which
> allow replacement, but I don't know if thats useful for this
> application).
> 
> Even if it did, I'm not really sure it matters. This isn't a circuit
> network; pre-emption does not imply a one to one replacement of one
> circuit with another. One could add priorities to an rsvp reservation so
> it always succeeds; all this means is that the loss rate of existing
> connections is increased if there really wasn't enough bandwidth for the
> new one. You can rely on human response to that to hang up if it gets
> really bad. In most cases, I suspect a WAN connection that supports any
> kind of data traffic at all will have sufficient resources so that these
> high priority reservations won't impact one drop existing calls.
> Furthermore, adaptive mechanisms that use FEC or redundanct codecs will
> allow performance to be probably not bad for loss rates that are quite
> high.
> 
> -Jonathan R.
> 
> 
> 
> Bob Bell wrote:
> >
> > Jonathan -
> >
> > I understand that this is a different aspect of the problem. It does bear
> > on SIP in that if a lower priority call is "preempted", there is supposed
> > to be a signal given to the end stations indicating that they have been
> or
> > are being preempted. That signal would need to be SIP. How and where it
> is
> > originated is a totally different issue and one I don't currently know
> how
> > to solve. There would need to be some form of feedback from RSVP or
> > DIFFSERV or ???? to allow for that interaction.
> >
> > Bob Bell
> >
> > At 10:11 AM  1/6/00, Jonathan Rosenberg wrote:
> > >This is a much different service. As the SIP path and media path are
> > >different (and, in fact, may flow through completely different service
> > >providers), there is really no way for this to be handled at the SIP
> > >layer. A proxy can't know to tear down someone elses call if bandwidth
> > >were available. Even the UA probably can't know that his call is
> > >consuming bandwidth that is preventing some other call from being made.
> > >
> > >Rather, if the *service* we need is that some generals call gets enough
> > >bandwidth, overriding some private, this is best handled by some kind of
> > >diffserv or rsvp operation.
> > >
> > >
> > >-Jonathan R.
> > >
> > >Bob Bell wrote:
> > > >
> > > > Gentlemen -
> > > >
> > > > There is another aspect of MLPP which is being ignored here. That is
> that I
> > > > don't want to talk to you, I just need the bandwidth your
> communications is
> > > > consuming over a WAN link of limited bandwidth. So, I (or a proxy
> which has
> > > > a better network view that I do) needs to tell you that you must
> terminate
> > > > the call. Thus, there is no opportunity for me to present an INVITE
> to you
> > > > and for you to return a Busy indication.
> > > >
> > > > Bob
> > > >
> > > > At 04:50 PM  1/5/00, Adam B. Roach wrote:
> > > >
> > > > >Jonathan Rosenberg writes:
> > > > >
> > > > > >Rather than using this MLPP-Version thing to indicate support for
> CCBS,
> > > > > >I'd rather use the more generic Supported/Require mechanism (a
> draft on
> > > > > >this is pending within the next few days). So,
> > > > > >
> > > > > >C->S INVITE
> > > > > >
> > > > > >S->C 421 Feature Available
> > > > > >Require: com.cisco.mlpp
> > > > > >
> > > > > >C->S INVITE
> > > > > >Supported: com.cisco.mlpp
> > > > > >MLPP: 2;version=1
> > > > > >
> > > > > >S->C 200 OK
> > > > >
> > > > >It sounds like you're adding yet another round trip:
> > > > >the first to agree on the feature, the second to try
> > > > >an invite (which gets a "busy" response), and the third
> > > > >to invoke the feature (am I misreading something here?)
> > > > >
> > > > >Keep in mind that MLPP is relevant only when the far side
> > > > >returns an indication like "busy." Only *after* the calling
> > > > >party is notified that the far side is busy does he make a
> > > > >decision that he wants to break in on the call.
> > > > >
> > > > >Also, in PSTN interoperation scenarios, the gateway won't
> > > > >know that you can break in on the called party until the busy
> > > > >indication arrives (since we're just pumping the call back
> > > > >into the network and don't necessarily know the capability
> > > > >of the end office to which the subscriber is physically
> > > > >connected) -- so negotiation of the feature cannot occur before
> > > > >an attempt is made to reach the called party.
> > > > >
> > > > >If we're going to go out of our way to borrow a service from
> > > > >ISUP, we'd be exceedingly foolish not to make them able to
> > > > >interoperate.
> > > > >
> > > > >--
> > > > >Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho,
> MS
> > > L-04
> > > > >adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
> > > 75081 USA
> > > > >
> > > >
> > > > Bob Bell
> > > > Cisco Systems Inc.
> > > > 801-294-3034(v)
> > > > 801-294-3023(f)
> > >
> > >--
> > >Jonathan D. Rosenberg                       200 Executive Drive
> > >Chief Scientist                             Suite 120
> > >dynamicsoft                                 West Orange, NJ 07052
> > >jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > >http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > >http://www.dynamicsoft.com
> >
> > Bob Bell
> > Cisco Systems Inc.
> > 801-294-3034(v)
> > 801-294-3023(f)
> 
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 




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archow@hss.hns.com wrote:
> 
> Hi,
> Now if SIP is only acting as a wrapper, why is it that people seem to
> mention that SIP is the preferred interaction scheme b/w MGC - MGC ?

Its not just a transfer function in this application. The call control
is driven from SIP; the encapsulated data is extra info that can be used
at an ISUP gateway if needed.

The result of this is that you get SIPs powerful proxy, forking, search,
routing, multi-provider, multi-hop  and scaling features, and at the
same time, you get ISUP transparency if you want it. Should an INVITE
with ISUP payloads hit a non-ISUP gateway (or even a PC phone) the data
can be ignored and the call is set up. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 09:09:42 2000
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Cc: sip@lists.research.bell-labs.com
Subject: Re: General question on SIP used b/w MGCs
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Hi Jonathan,
so you mean that the real  use of using SIP is if I want to merge proxy
server features into an MGC is not so useful if I have a config. like
another full fledged SIP Proxy server already in the network which already
provides these services you mentioned, and my gateway does not therefore
need these features ?

Regds
Arjun



On Fri, 7 Jan 2000, Jonathan Rosenberg wrote:

> 
> 
> 
> 
> 
> 
> archow@hss.hns.com wrote:
> >
> > Hi,
> > Now if SIP is only acting as a wrapper, why is it that people seem to
> > mention that SIP is the preferred interaction scheme b/w MGC - MGC ?
> 
> Its not just a transfer function in this application. The call control
> is driven from SIP; the encapsulated data is extra info that can be used
> at an ISUP gateway if needed.
> 
> The result of this is that you get SIPs powerful proxy, forking, search,
> routing, multi-provider, multi-hop  and scaling features, and at the
> same time, you get ISUP transparency if you want it. Should an INVITE
> with ISUP payloads hit a non-ISUP gateway (or even a PC phone) the data
> can be ignored and the call is set up.
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 09:21:53 2000
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Sorry, I can't completley parse this sentence. I think you are asking if
the benefit of using SIP is that you can merge proxy servers into an
MGC? If this is the question, the answer is no, thats not at all (in
fact, the OPPOSITE) of what I am saying. I'm saying, if the MGC uses
SIP, you get to reuse your existing proxy servers, since they already
provide routing, scaling, inter-provider access, multi-hop, etc., but
you get end to end ISUP transparency as well.

-Jonathan R.

archow@hss.hns.com wrote:
> 
> Hi Jonathan,
> so you mean that the real  use of using SIP is if I want to merge proxy
> server features into an MGC is not so useful if I have a config. like
> another full fledged SIP Proxy server already in the network which already
> provides these services you mentioned, and my gateway does not therefore
> need these features ?
> 
> Regds
> Arjun
> 
> On Fri, 7 Jan 2000, Jonathan Rosenberg wrote:
> 
> >
> >
> >
> >
> >
> >
> > archow@hss.hns.com wrote:
> > >
> > > Hi,
> > > Now if SIP is only acting as a wrapper, why is it that people seem to
> > > mention that SIP is the preferred interaction scheme b/w MGC - MGC ?
> >
> > Its not just a transfer function in this application. The call control
> > is driven from SIP; the encapsulated data is extra info that can be used
> > at an ISUP gateway if needed.
> >
> > The result of this is that you get SIPs powerful proxy, forking, search,
> > routing, multi-provider, multi-hop  and scaling features, and at the
> > same time, you get ISUP transparency if you want it. Should an INVITE
> > with ISUP payloads hit a non-ISUP gateway (or even a PC phone) the data
> > can be ignored and the call is set up.
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >
> >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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> 
> Although not required by the SIP standard, there is absolutely nothing to
> keep you from doing this other than the fact that there are better
> mechanisms. TRIP might be one avenue to consider.
> 

The slight difficulty is that SIP registers users, so a gateway
"representing" phone numbers would have to register a lot of individual
users. This is another reason TRIP is preferable, since it has
mechanisms to aggregate reachability information. It is not clear to me
that creating a wildcard user, with all the possible permutations, would
be wise. Obviously, if you know your proxy server, nothing prevents you
from registering

415*@gateway

or whatevever format you choose.

Henning



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 09:51:42 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        ldeCarmo@netspeak.com
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The additional difficult with SIP REGISTER isn't just the wildcarding
problem, but that the semantics are wrong. If multiple gateways register
a single number (aka, multiple Contacts with the same To field), the
result is that the proxy will *fork* a call to these, in all likelihood,
which is not at all what you want. Choosing of a gateway for terminating
calls depends a lot on policy information, and requires additional
information also not provided in a REGISTER message. TRIP does all this
for you, and also works for getting routing information from other
domains (thats its main job, in fact). A TRIP implementation on a
gateway is actually very simple, since you don't need any of the policy,
import, aggregation processing from TRIP, just the exporting operation.
In fact, I would argue that a gateway implementation of TRIP is simpler
than implementing some kind of REGISTER extension.

-Jonathan R.

Henning Schulzrinne wrote:
> 
> >
> > Although not required by the SIP standard, there is absolutely nothing to
> > keep you from doing this other than the fact that there are better
> > mechanisms. TRIP might be one avenue to consider.
> >
> 
> The slight difficulty is that SIP registers users, so a gateway
> "representing" phone numbers would have to register a lot of individual
> users. This is another reason TRIP is preferable, since it has
> mechanisms to aggregate reachability information. It is not clear to me
> that creating a wildcard user, with all the possible permutations, would
> be wise. Obviously, if you know your proxy server, nothing prevents you
> from registering
> 
> 415*@gateway
> 
> or whatevever format you choose.
> 
> Henning

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
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I'm also interested.

> -----Original Message-----
> From:	Rosen, Brian [SMTP:brosen@fore.com]
> Sent:	Thursday, January 06, 2000 10:22 AM
> To:	'Jonathan Rosenberg'
> Cc:	'sip@lists.research.bell-labs.com'
> Subject:	RE: not able to understand the call flow for Home Phone
> implement ation
> 
>  
> > Are many people interested in this? If there are many folks who would
> > like to see some work on SIP extensions required for home 
> > line service,
> > it might be something worthwhile to consider.
> > 
> 
> Count me in.  It has applications beyond home.
> 
> Brian
> 

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<TITLE>RE: not able to understand the call flow for Home Phone =
implement ation</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I'm also =
interested.</FONT>
</P>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Rosen, Brian [SMTP:brosen@fore.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Thursday, January 06, 2000 10:22 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">'Jonathan Rosenberg'</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">'sip@lists.research.bell-labs.com'</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">RE: not able to understand the call =
flow for Home Phone implement ation</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&nbsp;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Are many people interested in =
this? If there are many folks who would</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; like to see some work on SIP =
extensions required for home </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; line service,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; it might be something worthwhile =
to consider.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Count me in.&nbsp; It has applications =
beyond home.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Brian</FONT>
</P>
</UL>
</BODY>
</HTML>
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From: "Mark Gibson" <mrg@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: RE: General question on SIP used b/w MGCs
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	I'm saying, if the MGC uses
	SIP, you get to reuse your existing proxy servers, since they
already
	provide routing, 

	I had understood you to say in the past that a SIP server, and
therefore a SIP proxy, used a signalling path that is independent of the
media path. I am therefore eager to learn how a proxy provides routing as,
as you're doubtless already aware, I have a real interest in this problem.
Or is it SIP message routing that you're referring to here?

	scaling, inter-provider access, multi-hop, etc., but
	you get end to end ISUP transparency as well.

	-Jonathan R.

	I'd appreciate a clarification. 

	Mark

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<UL>
<P><FONT SIZE=3D2 FACE=3D"Arial">I</FONT><A NAME=3D"_MailData"><FONT =
SIZE=3D2 FACE=3D"Arial">'m saying, if the MGC uses</FONT></A>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP, you get to reuse your existing =
proxy servers, since they already</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">provide routing,</FONT><U> </U>
</P>

<P><U><FONT SIZE=3D2 FACE=3D"Arial">I had understood you to say in the =
past that a SIP server, and therefore a SIP proxy, used a signalling =
path that is</FONT></U><U> <FONT SIZE=3D2 =
FACE=3D"Arial">independent</FONT></U><U><FONT SIZE=3D2 =
FACE=3D"Arial"></FONT></U><U> <FONT SIZE=3D2 FACE=3D"Arial">of the =
media path. I am therefore eager to learn how a proxy provides routing =
as, as you're doubtless already aware, I have a real interest in this =
problem.</FONT></U><U><FONT SIZE=3D2 FACE=3D"Arial"> Or is it SIP =
message routing that you're referring to here?</FONT></U><U></U></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">scaling, inter-provider access, =
multi-hop, etc., but</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">you get end to end ISUP transparency =
as well.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-Jonathan R.</FONT>
</P>

<P><U><FONT SIZE=3D2 FACE=3D"Arial">I'd appreciate a =
clarification.</FONT></U><U> </U>
</P>

<P><U><FONT SIZE=3D2 FACE=3D"Arial">Mark</FONT></U>
</P>
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From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 10:41:46 2000
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> Mark Gibson wrote:
> 
>      I'm saying, if the MGC uses
>      SIP, you get to reuse your existing proxy servers, since they
>      already
>      provide routing,
> 
>      I had understood you to say in the past that a SIP server, and
>      therefore a SIP proxy, used a signalling path that is independent
>      of the media path. I am therefore eager to learn how a proxy
>      provides routing as, as you're doubtless already aware, I have a
>      real interest in this problem. Or is it SIP message routing that
>      you're referring to here?

Yes, it is SIP message routing - issues like telephone number routing
are the things the proxy worries about. IP packet routing is already
taken care of for us ;)

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 10:58:33 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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To: "James M. Polk" <jmpolk@cisco.com>
Cc: sip@lists.research.bell-labs.com, rtbell@cisco.com
Subject: Re: Prioritization limitation question
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While the conclusion of this discussion appears to be that a more
elaborate mechanism is required to serve this purpose, I wonder if it
would be helpful to allow extensions to priority-value in the SIP spec,
with the obvious caveat that an application may simply ignore new
values, although a reasonable application may decide to render the token
as-is, without interpretation. If there are local conventions that
encompass the existing priorities, this would seem preferable to forcing
people to use 

X-Priority: local-label

and the normal Priority otherwise. I18N, among other reasons, argues
against allowing this.

James M. Polk wrote:
> 
> Anyone/everyone
> 
> Reading the ANSI document T1.619-1992 a colleague pointed out to me, I
> came across its MLPP Prioritization requirements as being the
> following:
> 
> 1)      0       Flash Override (Highest)
> 2)      1       Flash
> 3)      2       Immediate
> 4)      3       Priority
> 5)      4       Routine  (Lowest)
> 
> But looking at SIP grammar/syntax, it only stipulates 4 levels of
> priority as the following:
> 
> priority-value = "emergency" | "urgent" | "normal" | "non-urgent"
> 
> Question: Has there been any discussion (which I haven't found yet) on
> matching these two specifications? And for clarification, I strongly
> prefer SIP moving up to having 5 levels instead of ANSI moving down to
> 4 levels in order to meet current US Government/Military Regulations I
> really don't want to ask them to change.
> 
> Comments/clarifications are welcome, thank you.
> 
>        _______________________________________________________
> A sobering reflection upon this century's greatest accomplishments and
>                              discoveries:
>     "There is no keystroke that rewrites racism... nor is there any
>                software that takes care of Mother Earth"
> 
> James M. Polk
> Sr. Product Manager, Multiservice Architecture and Standards
> Enterprise Voice Business Unit
> Cisco Systems
> Dallas, Texas
> w) 972.813.5208
> f)  972.813.5199
> www.cisco.com



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It should also be noted that the issue of session-level preemption for a
pure IP environment may be less of an issue than in the PSTN, since a
multiple call presences, with full information about who's calling,
priority, etc., are easily handled.

Whether, from a user-interface point of view, it would be nice to have
localized versions of call priority, so that a military officer, say,
would see call priorities in familiar renditions, is a separate issue
(see my earlier email). The question is whether this should be done
explicitly, as in

ANSI-Priority: Flash-Override

or as extensions to the Priority header. The former is probably cleaner,
as it is unlikely that the two ranking systems are to be mixed within
the same environment.



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From: "David Oran" <oran@cisco.com>
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Subject: RE: not able to understand the call flow for Home Phone implementation
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Of course, another way to think about this is that there is only one UA, and
the different instruments in the house are not in fact separate SIP UAs,
since there's only one phone number and one "call" at a time on all of them.
In this case, you could use MGCP or Megaco to have the user agent control
the various instruments and use multicast or an RTP mixer to distribute the
audio properly.

Whether this approach is at all desirable hinges on a number of factors.
Here's some test cases:
a)If two of the home phones are currently talking to someone and one of the
two hangs up, can it now make an outgoing call? If the answer is yes, then
this isn't really the same as multiple instruments sharing a "line".
b)If I pick up a phone while somebody else is talking on another, do I get a
choice as to whether I join the current call or not?
c)If the far end hangs up, do the home phones stay in communication
indefinitely, or do they eventually get howler tone indicating you forgot to
hang up the line?
d)If one of the phones is currently off hook dialing out, are incoming calls
blocked to all the other phones?

So, do you REALLY want to exactly emulate the shared-line semantics of
current POTS phones?

Dave.

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
> Rosenberg
> Sent: Thursday, January 06, 2000 10:03 AM
> To: ANIRBAN ROY
> Cc: sip@lists.research.bell-labs.com
> Subject: Re: not able to understand the call flow for Home Phone
> implementation
>
>
> You are right in that a SIP UA, as currently specified, won't be able to
> get home line type of service. It can be done in the way described in
> this paper, but it requires changes to the behavior of the UA, although
> no changes to the protocol itself, depending on how its done. It also
> requires the caller to be "cooperative" - they will receive multiple 200
> OK as each extension picks up, and they need to initiate a multi-party
> conference to do it. Basically, if you use multicast instead of forking,
> you don't need protocol extensions, just some new behaviors in the UA.
> If you want to rely on forking rather than multicast, you need somewhat
> more complex extensions.
>
> When using multicast, the UA basically snoops on the multicast group for
> BYEs and other 200 OKs. This allows it to build a complete picture of
> what other extensions have picked up, hung up, or been hung up on. Based
> on this, it can, in a distributed fashion, figure out the state of the
> call. If there are calls active, a user picking up the phone would cause
> the phone to send a 200 OK for the call.
>
> For forking, snooping of BYEs and other messages is not feasible, so
> some extensions are needed. However, doing this means you home line
> extensions could actually be anywhere in the world. Pretty cool.
>
> There are other ways to do it which hide from the caller the fact that
> multiple extensions have picked up. This is more like how the current
> home phone service works.
>
> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home line service,
> it might be something worthwhile to consider.
>
> -Jonathan R.
>
> ANIRBAN ROY wrote:
> >
> > hi,
> > I am trying to figure out how to get a home phone functionality in SIP.
> > The home phone has the following functionality
> > 1    When a call comes all phones in the home ring
> > 2    When one picks up, all other line must stop ringing
> > 3    A user can picks up from any other telephone in the home and join
> > an existing call.
> >
> >                                A scenario is there where A calls a Home
> > Phone which has Three connection B, C, D in his home
> >
> > When a calls to a home phone which has 3 connection (B, C, D) lands on
> > proxy, then proxy forks the INVITE message to B, C, D. so every phone
> > starts ringing. This is the first character of Home Phone. Now Let's say
> >
> > B picks up the phone then Proxy sends CANCEL message to C and D so that
> > the ringing can stop. This fulfills the Second point of Home Phone.
> >
> > I am not able to get the call flow for the Third point in home Phone. If
> >
> > lets say C picks up the phone (when both A and C in conversation)then he
> > should also be in session with A
> > and B. How can this be done because When a CANCEL message is received by
> >
> > C and D then they both reached the state where they can only expect
> > INVITE.
> >
> > Can any one please tell me how the call flow for the third point in the
> > home phone.
> >
> > The above points of Home phone were taken from the -----Bell Labs
> > Journal (October-December 1998)
> >
> > regards
> > Anirban
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
>
>




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David Oran wrote:
> 
> Whether this approach is at all desirable hinges on a number of factors.
> Here's some test cases:
> a)If two of the home phones are currently talking to someone and one of the
> two hangs up, can it now make an outgoing call? If the answer is yes, then
> this isn't really the same as multiple instruments sharing a "line".
> b)If I pick up a phone while somebody else is talking on another, do I get a
> choice as to whether I join the current call or not?
> c)If the far end hangs up, do the home phones stay in communication
> indefinitely, or do they eventually get howler tone indicating you forgot to
> hang up the line?
> d)If one of the phones is currently off hook dialing out, are incoming calls
> blocked to all the other phones?
> 
> So, do you REALLY want to exactly emulate the shared-line semantics of
> current POTS phones?

I'm not at all sure we want to emulate the current semantics. For one, I
kind of like the capability of knowing when (and who) picks up on
several extensions; this would argue for these devices being SIP UAs and
sending a 200 OK as each picks up. If we take on this work (which there
does seem sufficient interest to do), the first task is to figure out
which scenarios are supported. It may turn out that the spec is some
general extensions and then a description of how some number of
different cases might be supported.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 13:11:51 2000
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Subject: Re: A case related to "home Phone" type of functionality in SIP
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I'd rather not delve too much farther into the technical details of how
until we have understood what we want to accomplish. There have been
lots of ideas already in the last few days on how to do this, and they
all are solutions to slightly different problems.

So, given there appears to be interest in working on this, let me ask
the next question - who is willing to volunteer to work on a design team
to develop some documents for this? First step is requirements. If there
are sufficient people who are willing to actually do the work (phone
conferences, mail discussions and possibly a face to face), I will
discuss the possibility of adding this as a work item with my co-chairs
and the ADs.

-Jonathan R.

ANIRBAN ROY wrote:
> 
> hi ,
> 
> lets say in a house there are 3 phones B, C, D. now User "A" calls to this house
> 
> Please tell me one question
> 
> When a call is going on between two user A and B and the third person "C" picks
> up the phone from the same house what happens then????.
> 
> #       does a 200 Response goes back to the caller. Or the 200 Response just
> append the Multicast Address with address "C".
> 
> #   RTP data goes directly from caller to callee. When there are more than one
> user is in conversation how Proxy makes it sure that the RTP data will be send
> to particular multicast address. As because Proxy only work in the signalling
> layer.
> 
> Please reply
> 
> Anirban
> 
> Dean Willis wrote:
> 
> > Are you saying that the first phone would multicast the media streams to the
> > others on the local LAN, or that the INVITES and media would be sent
> > multicast across the ISP network?
> >
> > If the former, it might work. COuld be pretty cool.
> >
> > If the latter, no, it's not going to happen that way.
> >
> > 1) No ISP that I know of will provide a multicast feed, especially for a
> > lowly dialup user.
> >
> > 2) Every SIP phone in the country is going to listen to a universal
> > multicast group, then make application discard decisions on the INVITES that
> > don't apply to it  . . . Or, we could statically allocate a multicast
> > address for each home network . . .not. The INVITE will come in UNICAST.
> >
> > 3) Personally, I've never managed to get multicast routing working right
> > over the bizarre equipment that has accumulated at my house.
> >
> > 4) In our own internal network, we often saw join times that took seven or
> > eight minutes to propaget back to the source of a multicast. That's some
> > serious post-dial delay. I'm not sure we even try to route multicast
> > anymore.
> >
> > 5) The firewall issues that occur with cross-domain multicast are mind
> > boggling. Anything coming into my house is by-definition cross-domain.
> >
> > Hey, I love multicast. I spent more than two years building really cool
> > businesses (Real Broadcast Netork, skyMCI, etc.) around it, and I'm here to
> > tell you it just isn't going to happen across an Internet backbone for the
> > average user any time soon.
> >
> > --
> > Dean
> >
> > > -----Original Message-----
> > > From: Sean Olson [mailto:eussean@exu.ericsson.se]
> > > Sent: Thursday, January 06, 2000 5:49 PM
> > > To: dean.willis@wcom.com
> > > Cc: sip@lists.research.bell-labs.com
> > > Subject: RE: A case related to "home Phone" type of functionality in SIP
> > >
> > >
> > > >
> > > > In a business environment, we call this a "single line extension".
> > > >
> > > > SIP approaches tend to have the property that each phone is uniquely
> > > > addressed, whereas in the PSTN each circuit is uniquely addressed.
> > > >
> > > > The obvious solution is to combine a parallel search proxy and some UAS
> > > > behavior.
> > > >
> > > > The sequence is:
> > > > 1) Incoming call reaches parallel proxy.
> > > > 2) Parallel proxy invites all UASes in extension group.
> > > > 3) User answers one extension.
> > > > 4) Parallel proxy cancels unanswered legs. Those phones stop ringing.
> > > > 5) Answering phone send INVITE to other phones in group, with "NO RING"
> > > > specified somehow, and long expire timer.
> > > >
> > > >
> > >
> > > In a business environment, this is a perfectly acceptable and reasonable
> > > solution. For a home environment, multicast/DHCP seems more reasonable
> > > because Joe consumer can go to Radio Shack, buy a 3com SIP phone, and
> > > plug it into his home IP network and it works with zero (or
> > > almost-zero) configuration. Plus, it doesn't require the home user
> > > to buy/maintain a parallel proxy.
> > >
> > > (By multicast, I mean both signalling and media).
> > >
> > > -----------------------------------------------------------------
> > > Sean Olson           mailto:sean.olson@ericsson.com
> > > Ericsson Inc.         tel:(972)583-5472
> > >                       fax:(972)669-0154
> > >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Rosen, Brian" <brosen@fore.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        ANIRBAN ROY <anirban.roy@wipro.com>
Cc: Dean Willis <dean.willis@wcom.com>, Sean Olson <eussean@exu.ericsson.se>,
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Subject: RE: A case related to "home Phone" type of functionality in SIP
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I'll work on it.

Brian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, January 07, 2000 1:15 PM
> To: ANIRBAN ROY
> Cc: Dean Willis; Sean Olson; sip@lists.research.bell-labs.com
> Subject: Re: A case related to "home Phone" type of 
> functionality in SIP
> 
> 
> I'd rather not delve too much farther into the technical 
> details of how
> until we have understood what we want to accomplish. There have been
> lots of ideas already in the last few days on how to do this, and they
> all are solutions to slightly different problems.
> 
> So, given there appears to be interest in working on this, let me ask
> the next question - who is willing to volunteer to work on a 
> design team
> to develop some documents for this? First step is 
> requirements. If there
> are sufficient people who are willing to actually do the work (phone
> conferences, mail discussions and possibly a face to face), I will
> discuss the possibility of adding this as a work item with my 
> co-chairs
> and the ADs.
> 
> -Jonathan R.
> 
> ANIRBAN ROY wrote:
> > 
> > hi ,
> > 
> > lets say in a house there are 3 phones B, C, D. now User 
> "A" calls to this house
> > 
> > Please tell me one question
> > 
> > When a call is going on between two user A and B and the 
> third person "C" picks
> > up the phone from the same house what happens then????.
> > 
> > #       does a 200 Response goes back to the caller. Or the 
> 200 Response just
> > append the Multicast Address with address "C".
> > 
> > #   RTP data goes directly from caller to callee. When 
> there are more than one
> > user is in conversation how Proxy makes it sure that the 
> RTP data will be send
> > to particular multicast address. As because Proxy only work 
> in the signalling
> > layer.
> > 
> > Please reply
> > 
> > Anirban
> > 
> > Dean Willis wrote:
> > 
> > > Are you saying that the first phone would multicast the 
> media streams to the
> > > others on the local LAN, or that the INVITES and media 
> would be sent
> > > multicast across the ISP network?
> > >
> > > If the former, it might work. COuld be pretty cool.
> > >
> > > If the latter, no, it's not going to happen that way.
> > >
> > > 1) No ISP that I know of will provide a multicast feed, 
> especially for a
> > > lowly dialup user.
> > >
> > > 2) Every SIP phone in the country is going to listen to a 
> universal
> > > multicast group, then make application discard decisions 
> on the INVITES that
> > > don't apply to it  . . . Or, we could statically allocate 
> a multicast
> > > address for each home network . . .not. The INVITE will 
> come in UNICAST.
> > >
> > > 3) Personally, I've never managed to get multicast 
> routing working right
> > > over the bizarre equipment that has accumulated at my house.
> > >
> > > 4) In our own internal network, we often saw join times 
> that took seven or
> > > eight minutes to propaget back to the source of a 
> multicast. That's some
> > > serious post-dial delay. I'm not sure we even try to 
> route multicast
> > > anymore.
> > >
> > > 5) The firewall issues that occur with cross-domain 
> multicast are mind
> > > boggling. Anything coming into my house is by-definition 
> cross-domain.
> > >
> > > Hey, I love multicast. I spent more than two years 
> building really cool
> > > businesses (Real Broadcast Netork, skyMCI, etc.) around 
> it, and I'm here to
> > > tell you it just isn't going to happen across an Internet 
> backbone for the
> > > average user any time soon.
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: Sean Olson [mailto:eussean@exu.ericsson.se]
> > > > Sent: Thursday, January 06, 2000 5:49 PM
> > > > To: dean.willis@wcom.com
> > > > Cc: sip@lists.research.bell-labs.com
> > > > Subject: RE: A case related to "home Phone" type of 
> functionality in SIP
> > > >
> > > >
> > > > >
> > > > > In a business environment, we call this a "single 
> line extension".
> > > > >
> > > > > SIP approaches tend to have the property that each 
> phone is uniquely
> > > > > addressed, whereas in the PSTN each circuit is 
> uniquely addressed.
> > > > >
> > > > > The obvious solution is to combine a parallel search 
> proxy and some UAS
> > > > > behavior.
> > > > >
> > > > > The sequence is:
> > > > > 1) Incoming call reaches parallel proxy.
> > > > > 2) Parallel proxy invites all UASes in extension group.
> > > > > 3) User answers one extension.
> > > > > 4) Parallel proxy cancels unanswered legs. Those 
> phones stop ringing.
> > > > > 5) Answering phone send INVITE to other phones in 
> group, with "NO RING"
> > > > > specified somehow, and long expire timer.
> > > > >
> > > > >
> > > >
> > > > In a business environment, this is a perfectly 
> acceptable and reasonable
> > > > solution. For a home environment, multicast/DHCP seems 
> more reasonable
> > > > because Joe consumer can go to Radio Shack, buy a 3com 
> SIP phone, and
> > > > plug it into his home IP network and it works with zero (or
> > > > almost-zero) configuration. Plus, it doesn't require 
> the home user
> > > > to buy/maintain a parallel proxy.
> > > >
> > > > (By multicast, I mean both signalling and media).
> > > >
> > > > 
> -----------------------------------------------------------------
> > > > Sean Olson           mailto:sean.olson@ericsson.com
> > > > Ericsson Inc.         tel:(972)583-5472
> > > >                       fax:(972)669-0154
> > > >
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 13:37:45 2000
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From: "David Oran" <oran@cisco.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "ANIRBAN ROY" <anirban.roy@wipro.com>, <sip@lists.research.bell-labs.com>
Subject: RE: not able to understand the call flow for Home Phone implementation
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> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, January 07, 2000 1:08 PM
> To: David Oran
> Cc: ANIRBAN ROY; sip@lists.research.bell-labs.com
> Subject: Re: not able to understand the call flow for Home Phone
> implementation
> 
[snip] 
> I'm not at all sure we want to emulate the current semantics. 
Bingo. My view exactly. That's why I started listing the mind-benders.




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 13:51:45 2000
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From: Wing Man Kwok <wkwok@telogy.com>
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Current status of DCS
Date: Fri, 7 Jan 2000 13:48:38 -0500 
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Hi all,

I would like to ask what is the current status of DCS?  

I heard from one of my colleagues that the 2 stage call setup as proposed in
the earlier DCS draft is being modified to a one stage call setup.  Is this
true?  

Any pointer to an updated version of DCS draft or related documents is
greatly appreciated.  

Thanks.

Wing Man Kwok



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> > 
> > Although not required by the SIP standard, there is absolutely nothing to
> > keep you from doing this other than the fact that there are better
> > mechanisms. TRIP might be one avenue to consider.
> > 
> 
> The slight difficulty is that SIP registers users, so a gateway
> "representing" phone numbers would have to register a lot of individual
> users. This is another reason TRIP is preferable, since it has
> mechanisms to aggregate reachability information. It is not clear to me
> that creating a wildcard user, with all the possible permutations, would
> be wise. Obviously, if you know your proxy server, nothing prevents you
> from registering
> 
> 415*@gateway
> 
> or whatevever format you choose.
> 
> Henning
> 

True, it would require some interesting syntax but I believe the semantics
he had in mind are different. The problem he is trying to solve is not
registering a gateway that is capable of answering for a set of users, but
more simply, registering the fact that a given gateway is up and in service.
The set of users that a given gateway can support would then have to be
either statically configured or negotiated through some different protocol
such as TRIP. 

sean




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan  7 14:43:04 2000
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> 
> True, it would require some interesting syntax but I believe the semantics
> he had in mind are different. The problem he is trying to solve is not
> registering a gateway that is capable of answering for a set of users, but
> more simply, registering the fact that a given gateway is up and in service.
> The set of users that a given gateway can support would then have to be
> either statically configured or negotiated through some different protocol
> such as TRIP.

However, registration of a gateway only shows that the gateway was
available, say, an hour ago, unless the gateway is kind enough to "sign
out" before it crashes. If you already know that you want to send a call
to the gateway and you only have one, you might as well try and find out
quickly whether it's up or not. If there's a choice of gateways, you
presumably need something like TRIP to determine the correct one, in the
absence of information which "users" (phone #) each gateway handles.

> 
> sean



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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'Sean Olson'" <eussean@exu.ericsson.se>, schulzrinne@cs.columbia.edu
Cc: sip@lists.research.bell-labs.com
Subject: RE: Gateways and registration
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	>True, it would require some interesting syntax but I believe the
semantics
	>he had in mind are different. The problem he is trying to solve is
not
	>registering a gateway that is capable of answering for a set of
users, but
	>more simply, registering the fact that a given gateway is up and in
service.
	>The set of users that a given gateway can support would then have
to be
	>either statically configured or negotiated through some different
protocol
	>such as TRIP. 

This is EXACTLY what I'm trying to solve Sean. 

I'd prefer to do it in SIP rather than a companion protocol, because we're
trying to interface with 3rd party gateways that may/may not talk TRIP.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487





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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        Sean Olson <eussean@exu.ericsson.se>
Cc: sip@lists.research.bell-labs.com
Subject: RE: Gateways and registration
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	>However, registration of a gateway only shows that the gateway was
	>available, say, an hour ago, unless the gateway is kind enough to
"sign
	>out" before it crashes. 

	Proper use of the Expires header would cause the gateway to
re-register on a periodic basis and allow the Server to detect if it went
down.  Granted, it could crash between registrations, but I could live with
that.

	>to the gateway and you only have one, you might as well try and
find out
	>quickly whether it's up or not. If there's a choice of gateways,
you
	>presumably need something like TRIP to determine the correct one,
in the
	>absence of information which "users" (phone #) each gateway
handles.

      If user information was statically provisioned at the Server, a simple
Register-like function would suffice to determine availability of the
gateway.  More advanced information could be obtained from a TRIP-like
protocol.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487






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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Linden deCarmo <ldeCarmo@netspeak.com>
Cc: "'Sean Olson'" <eussean@exu.ericsson.se>, schulzrinne@cs.columbia.edu,
        sip@lists.research.bell-labs.com
Subject: Re: Gateways and registration
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Linden deCarmo wrote:
> 
> This is EXACTLY what I'm trying to solve Sean.
> 
> I'd prefer to do it in SIP rather than a companion protocol, because we're
> trying to interface with 3rd party gateways that may/may not talk TRIP.

I don't see why they would be any more likely to behave properly in a
non-standard usage of REGISTER. TRIP handles keepalives; its keepalive
mechanism is quite flexible (borrows from BGP, actually), and you also
get the routing information you need. IMHO, I'd rather solve the general
problem with a solid solution than make many small extensions to SIP
here and there to solve only a subset of the real problem.

A complete VoIP solution requires many components. The general approach
we have taken into IETF is to break the various components into
independent, manageable pieces, and then define protocols for solving
each of those pieces. By keeping these pieces standalone and
independent, we can swap in different versions or alternatives down the
road without affecting other components. This helps tremendously in
extensibility and growth; we have seen this in routing protocols, for
example (separation of intra and inter domain protocols, even though
both run on a router). It also means we can optimize the solution for
the problem at hand. 

For many different reasons, some of which have already been outlined
here, the SIP REGISTER method is not a good tool for gateways to use. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Fri, 07 Jan 2000 14:41:25 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: not able to understand the call flow for Home Phone implementation
In-reply-to: <NDBBKHCGKKIOOIJEGCOEKEHOCOAA.oran@cisco.com>
To: David Oran <oran@cisco.com>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        ANIRBAN ROY <anirban.roy@wipro.com>
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Dave,
>In this case, you could use MGCP or Megaco to have the user agent control
>the various instruments

and have the call agent for consumers in the LEC CO? Perish the thought.

Henry

-----Original Message-----
From: owner-sip@lists.research.bell-labs.com
[mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of David Oran
Sent: Friday, January 07, 2000 11:45 AM
To: Jonathan Rosenberg; ANIRBAN ROY
Cc: sip@lists.research.bell-labs.com
Subject: RE: not able to understand the call flow for Home Phone
implementation


Of course, another way to think about this is that there is only one UA, and
the different instruments in the house are not in fact separate SIP UAs,
since there's only one phone number and one "call" at a time on all of them.
In this case, you could use MGCP or Megaco to have the user agent control
the various instruments and use multicast or an RTP mixer to distribute the
audio properly.

Whether this approach is at all desirable hinges on a number of factors.
Here's some test cases:
a)If two of the home phones are currently talking to someone and one of the
two hangs up, can it now make an outgoing call? If the answer is yes, then
this isn't really the same as multiple instruments sharing a "line".
b)If I pick up a phone while somebody else is talking on another, do I get a
choice as to whether I join the current call or not?
c)If the far end hangs up, do the home phones stay in communication
indefinitely, or do they eventually get howler tone indicating you forgot to
hang up the line?
d)If one of the phones is currently off hook dialing out, are incoming calls
blocked to all the other phones?

So, do you REALLY want to exactly emulate the shared-line semantics of
current POTS phones?

Dave.

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
> Rosenberg
> Sent: Thursday, January 06, 2000 10:03 AM
> To: ANIRBAN ROY
> Cc: sip@lists.research.bell-labs.com
> Subject: Re: not able to understand the call flow for Home Phone
> implementation
>
>
> You are right in that a SIP UA, as currently specified, won't be able to
> get home line type of service. It can be done in the way described in
> this paper, but it requires changes to the behavior of the UA, although
> no changes to the protocol itself, depending on how its done. It also
> requires the caller to be "cooperative" - they will receive multiple 200
> OK as each extension picks up, and they need to initiate a multi-party
> conference to do it. Basically, if you use multicast instead of forking,
> you don't need protocol extensions, just some new behaviors in the UA.
> If you want to rely on forking rather than multicast, you need somewhat
> more complex extensions.
>
> When using multicast, the UA basically snoops on the multicast group for
> BYEs and other 200 OKs. This allows it to build a complete picture of
> what other extensions have picked up, hung up, or been hung up on. Based
> on this, it can, in a distributed fashion, figure out the state of the
> call. If there are calls active, a user picking up the phone would cause
> the phone to send a 200 OK for the call.
>
> For forking, snooping of BYEs and other messages is not feasible, so
> some extensions are needed. However, doing this means you home line
> extensions could actually be anywhere in the world. Pretty cool.
>
> There are other ways to do it which hide from the caller the fact that
> multiple extensions have picked up. This is more like how the current
> home phone service works.
>
> Are many people interested in this? If there are many folks who would
> like to see some work on SIP extensions required for home line service,
> it might be something worthwhile to consider.
>
> -Jonathan R.
>
> ANIRBAN ROY wrote:
> >
> > hi,
> > I am trying to figure out how to get a home phone functionality in SIP.
> > The home phone has the following functionality
> > 1    When a call comes all phones in the home ring
> > 2    When one picks up, all other line must stop ringing
> > 3    A user can picks up from any other telephone in the home and join
> > an existing call.
> >
> >                                A scenario is there where A calls a Home
> > Phone which has Three connection B, C, D in his home
> >
> > When a calls to a home phone which has 3 connection (B, C, D) lands on
> > proxy, then proxy forks the INVITE message to B, C, D. so every phone
> > starts ringing. This is the first character of Home Phone. Now Let's say
> >
> > B picks up the phone then Proxy sends CANCEL message to C and D so that
> > the ringing can stop. This fulfills the Second point of Home Phone.
> >
> > I am not able to get the call flow for the Third point in home Phone. If
> >
> > lets say C picks up the phone (when both A and C in conversation)then he
> > should also be in session with A
> > and B. How can this be done because When a CANCEL message is received by
> >
> > C and D then they both reached the state where they can only expect
> > INVITE.
> >
> > Can any one please tell me how the call flow for the third point in the
> > home phone.
> >
> > The above points of Home phone were taken from the -----Bell Labs
> > Journal (October-December 1998)
> >
> > regards
> > Anirban
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>
>
>





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Date: Sat, 08 Jan 2000 12:03:15 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: A case related to "home Phone" type of functionality in SIP
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Jonathan asked:

> So, given there appears to be interest in working on this, let me ask
> the next question - who is willing to volunteer to work on a design team
> to develop some documents for this? First step is requirements. If there
> are sufficient people who are willing to actually do the work (phone
> conferences, mail discussions and possibly a face to face), I will
> discuss the possibility of adding this as a work item with my co-chairs
> and the ADs.


My team is already working on a variation of this to meet the business
expectation of "single line extension" as supported off of common PBX and
Centrex systems.

The terse and badly written spec we have says:
-----
"Feature: Single Line Extension
Description: Provides the ability to give the customer single line set
extensions, either on the same or different premises - even in different
buildings.   This "line" will ring at each location and work just like an
extension.  There are several variations of Single Line Extension.
Sequential-First Wins, Parallel, etc.
-----

So, the basic idea is to capture the effect of this function as would be
experienced by a PBX user. This is slightly richer than the traditional home
experience, as most of these PBX phones are also "multiline" phones. This
means that if another user picks up a phone in the extension group, that
user can select whether to participate in the ongoing group call (an
"extension") or select another line for new dial tone and a new call. SIP
phones are basically multiline phones in this context.

If we added the capability for phones to display status on extensions that
join or leave the call, we would satisfy most of the requirements expressed
so far on this list. It would also be a terribly useful feature for
conference bridges . . .

Essentially, I view an extension group as a dynamically extended conference
brdige group.

I would be willing to support a common effort to standardize some kind of
solution.

I would like to see a solution that works outside the scope of a single
broadcast/multicast domain.

--
Dean


> So, given there appears to be interest in working on this, let me ask
> the next question - who is willing to volunteer to work on a design team
> to develop some documents for this? First step is requirements. If there
> are sufficient people who are willing to actually do the work (phone
> conferences, mail discussions and possibly a face to face), I will
> discuss the possibility of adding this as a work item with my co-chairs
> and the ADs.
>
> -Jonathan R.
>
> ANIRBAN ROY wrote:
> >
> > hi ,
> >
> > lets say in a house there are 3 phones B, C, D. now User "A"
> calls to this house
> >
> > Please tell me one question
> >
> > When a call is going on between two user A and B and the third
> person "C" picks
> > up the phone from the same house what happens then????.
> >
> > #       does a 200 Response goes back to the caller. Or the 200
> Response just
> > append the Multicast Address with address "C".
> >
> > #   RTP data goes directly from caller to callee. When there
> are more than one
> > user is in conversation how Proxy makes it sure that the RTP
> data will be send
> > to particular multicast address. As because Proxy only work in
> the signalling
> > layer.
> >
> > Please reply
> >
> > Anirban
> >
> > Dean Willis wrote:
> >
> > > Are you saying that the first phone would multicast the media
> streams to the
> > > others on the local LAN, or that the INVITES and media would be sent
> > > multicast across the ISP network?
> > >
> > > If the former, it might work. COuld be pretty cool.
> > >
> > > If the latter, no, it's not going to happen that way.
> > >
> > > 1) No ISP that I know of will provide a multicast feed,
> especially for a
> > > lowly dialup user.
> > >
> > > 2) Every SIP phone in the country is going to listen to a universal
> > > multicast group, then make application discard decisions on
> the INVITES that
> > > don't apply to it  . . . Or, we could statically allocate a multicast
> > > address for each home network . . .not. The INVITE will come
> in UNICAST.
> > >
> > > 3) Personally, I've never managed to get multicast routing
> working right
> > > over the bizarre equipment that has accumulated at my house.
> > >
> > > 4) In our own internal network, we often saw join times that
> took seven or
> > > eight minutes to propaget back to the source of a multicast.
> That's some
> > > serious post-dial delay. I'm not sure we even try to route multicast
> > > anymore.
> > >
> > > 5) The firewall issues that occur with cross-domain multicast are mind
> > > boggling. Anything coming into my house is by-definition cross-domain.
> > >
> > > Hey, I love multicast. I spent more than two years building
> really cool
> > > businesses (Real Broadcast Netork, skyMCI, etc.) around it,
> and I'm here to
> > > tell you it just isn't going to happen across an Internet
> backbone for the
> > > average user any time soon.
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: Sean Olson [mailto:eussean@exu.ericsson.se]
> > > > Sent: Thursday, January 06, 2000 5:49 PM
> > > > To: dean.willis@wcom.com
> > > > Cc: sip@lists.research.bell-labs.com
> > > > Subject: RE: A case related to "home Phone" type of
> functionality in SIP
> > > >
> > > >
> > > > >
> > > > > In a business environment, we call this a "single line extension".
> > > > >
> > > > > SIP approaches tend to have the property that each phone
> is uniquely
> > > > > addressed, whereas in the PSTN each circuit is uniquely addressed.
> > > > >
> > > > > The obvious solution is to combine a parallel search
> proxy and some UAS
> > > > > behavior.
> > > > >
> > > > > The sequence is:
> > > > > 1) Incoming call reaches parallel proxy.
> > > > > 2) Parallel proxy invites all UASes in extension group.
> > > > > 3) User answers one extension.
> > > > > 4) Parallel proxy cancels unanswered legs. Those phones
> stop ringing.
> > > > > 5) Answering phone send INVITE to other phones in group,
> with "NO RING"
> > > > > specified somehow, and long expire timer.
> > > > >
> > > > >
> > > >
> > > > In a business environment, this is a perfectly acceptable
> and reasonable
> > > > solution. For a home environment, multicast/DHCP seems more
> reasonable
> > > > because Joe consumer can go to Radio Shack, buy a 3com SIP
> phone, and
> > > > plug it into his home IP network and it works with zero (or
> > > > almost-zero) configuration. Plus, it doesn't require the home user
> > > > to buy/maintain a parallel proxy.
> > > >
> > > > (By multicast, I mean both signalling and media).
> > > >
> > > > -----------------------------------------------------------------
> > > > Sean Olson           mailto:sean.olson@ericsson.com
> > > > Ericsson Inc.         tel:(972)583-5472
> > > >                       fax:(972)669-0154
> > > >
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>




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Wing Man Kwok wrote:
> 
> Hi all,
> 
> I would like to ask what is the current status of DCS?
> 
> I heard from one of my colleagues that the 2 stage call setup as proposed in
> the earlier DCS draft is being modified to a one stage call setup.  Is this
> true?

There was some discussion amongst the authors of the two alternative
drafts, and a very tentative compromise was reached that uses a single
INVITE, but makes use of provisional response reliability to add the
additional round trip needed for DCS.  Some additional work remained to
ensure this was compatible with normal SIP and DQoS.

> 
> Any pointer to an updated version of DCS draft or related documents is
> greatly appreciated.

Try searching the I-D archive for "DCS" which will turn up the ones
presented at the interim meeting in November. Those are the most recent.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Peter Lewis" <peter.lewis@upperside.fr>
To: <sip@lists.research.bell-labs.com>
Subject: SIP 2000
Date: Tue, 11 Jan 2000 13:38:47 +0100
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=20
SIP 2000: beyond H.323?=20
Discussing and debating in Paris May 10-12.
A CFP is online at:
http://www.upperside.fr/basip.htm

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<DIV><FONT color=3D#000000 size=3D2>Discussing and debating in Paris May =

10-12.</FONT></DIV>
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Message-ID: <387ABDC5.5746B32F@wipsys.soft.net>
Date: Tue, 11 Jan 2000 10:51:09 +0530
From: "ANIRBAN ROY" <anirban.roy@wipro.com>
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To: sip@lists.research.bell-labs.com
Subject: Call flow for the Home Phone Application.
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hi ,
I have tried to dipict the Call Flow for the Home Phone with some
diagram to make it easy to understand. I have some doubts in those Call
Flow. Please look into it and give some suggestion for this flow.


 I am attaching a windows word doc file with it which has the call flow
with diagramatic representation of each step.

I am sorry that i am attaching Windows word doc file as I don't have any
other tool for making the Diagram.


regards
Anirban

--------------2DB1CFB8439B57A7E1633E85
Content-Type: application/msword;
 name="Call flow.doc"
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 filename="Call flow.doc"
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From owner-sip-outgoing@lists.research.bell-labs.com  Tue Jan 11 09:06:57 2000
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Date: Tue, 11 Jan 2000 16:58:32 +0530
From: "ANIRBAN ROY" <anirban.roy@wipro.com>
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To: sip@lists.research.bell-labs.com
Cc: jdrosen@dynamicsoft.com
Subject: Call flow for home phone Application
Content-Type: multipart/mixed;
 boundary="------------DCC0C60C268CBEBBFAA7F2A9"
Sender: owner-sip@lists.research.bell-labs.com
Precedence: bulk

This is a multi-part message in MIME format.
--------------DCC0C60C268CBEBBFAA7F2A9
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hi ,
I have tried to dipict the Call Flow for the Home Phone with some
diagram to make it easy to understand. I have some doubts in those Call
Flow. Please look into it and give some suggestion for this flow.


 I am attaching a windows word doc file with it which has the call flow
with diagramatic representation of each step.

I am sorry that i am attaching Windows word doc file as I don't have any

other tool for making the Diagram.


regards
Anirban

--------------DCC0C60C268CBEBBFAA7F2A9
Content-Type: application/msword;
 name="Call flow.doc"
Content-Disposition: inline;
 filename="Call flow.doc"
Content-Transfer-Encoding: base64

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--------------DCC0C60C268CBEBBFAA7F2A9--




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Jan 11 13:33:56 2000
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To: ANIRBAN ROY <anirban.roy@wipro.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Call flow for the Home Phone Application.
References: <387ABDC5.5746B32F@wipsys.soft.net>
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I would like to contribute some requirements:

I agree with Dean's points on multicast and would like to see
a solution which does not require multicast.

If a media "bridge" is used in the solution, it should be possible
for the answering party to be the bridge (as opposed to requiring
an additional entity).

The called parties which do not initially join the call, need some
minimal state information (i.e. to light a lamp to indicate that the
"line" is being used and/or the call is joinable).  This state information
should be provided in an event driven means (i.e. not polled).

Cheers,
Dan


ANIRBAN ROY wrote:

> hi ,
> I have tried to dipict the Call Flow for the Home Phone with some
> diagram to make it easy to understand. I have some doubts in those Call
> Flow. Please look into it and give some suggestion for this flow.
>
>  I am attaching a windows word doc file with it which has the call flow
> with diagramatic representation of each step.
>
> I am sorry that i am attaching Windows word doc file as I don't have any
> other tool for making the Diagram.
>
> regards
> Anirban
>
>   ------------------------------------------------------------
>                     Name: Call flow.doc
>    Call flow.doc    Type: WINWORD File (application/msword)
>                 Encoding: base64




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Jan 11 15:21:53 2000
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From: rick@fdd.com
To: jon.peterson@level3.com
Cc: sip@lists.research.bell-labs.com
Subject: Re: SIP 3rd-party call creation?
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Jon,

Yes, and the worst case is a Palm which is only present
for a small part of a call.  Please check out
http://www.normos.org/ietf/draft/draft-dean-phonectl-00.txt
and let me know what you think.

--
Rick
rick_dean@mw.3com.com


Jon.Peterson@Level3.com wrote:
> 
> Has any consideration been given to uninvolved third parties originating
> sessions between two SIP endpoints?
> 
> If I might provide a clarifying example without endorsing the described
> implementation - given two users A and B, each of whom has a SIP phone, and
> an automated entity C (the third party): C somehow decides that a session
> should be initiated between A and B, and thus sends an INVITE to A which,
> due to special properties and/or headers, does not result in a session
> between A and C, but rather prompts A to send a new INVITE to B after some
> sort of approval on A's part. Subsequently C might or might not maintain a
> relationship with A throughout the session establishment, and so forth.
> 
> I haven't been able to find any existing work that addresses this sort of
> situation; has something been written on this that I've missed? The SIP Call
> Control Services document covers some nearby territory (the Requested-By and
> Also headers could be relevant), but I don't think, from my reading of the
> document, that it endorses third-party call setup unless the third party is
> a participant in the session.
> 
> If this hasn't been explored to date, what is the prevailing opinion on
> third-party session initiation? Is it something to which SIP can and should
> be applied? Are the accounting and security issues too damaging?
> 
> Jon Peterson



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 12 11:11:23 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        sip@lists.research.bell-labs.com
Subject: Re: Call flow for the Home Phone Application.
References: <387ABDC5.5746B32F@wipsys.soft.net> <387C1498.63E47220@dynamicsoft.com>
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hi,
Yes I will like to work on the feature of Home phone feature for the SIP Proxy.
Please include me in your list.

regards
Anirban Roy

Jonathan Rosenberg wrote:

> As I have mentioned, it is too premature to begin discussions on the
> implementation. We first must decide whether there:
>
> 1. is sufficient people to work on this,
> 2. the ADs approve adding it to the charter,
> 3. the requirements for the service are defined
>
> then we can figure out how to extend SIP. Regarding (1), I have received
> few offers of people willing to actually work on this (two, to be
> exact). Feel free to respond privately if you are willing to work.
>
> -Jonathan R.
>
> ANIRBAN ROY wrote:
> >
> > hi ,
> > I have tried to dipict the Call Flow for the Home Phone with some
> > diagram to make it easy to understand. I have some doubts in those Call
> > Flow. Please look into it and give some suggestion for this flow.
> >
> >  I am attaching a windows word doc file with it which has the call flow
> > with diagramatic representation of each step.
> >
> > I am sorry that i am attaching Windows word doc file as I don't have any
> > other tool for making the Diagram.
> >
> > regards
> > Anirban
> >
> >   ------------------------------------------------------------------------
> >                     Name: Call flow.doc
> >    Call flow.doc    Type: WINWORD File (application/msword)
> >                 Encoding: base64
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com




From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 12 11:23:07 2000
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Message-ID: <387C291E.9BB056E8@obsoft.com>
Date: Wed, 12 Jan 2000 01:11:26 -0600
From: Bobby Sardana <sardana@obsoft.com>
Organization: ObjectSoftware, Inc.
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To: ANIRBAN ROY <anirban.roy@wipro.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Call flow for home phone Application
References: <387B13E0.B8883290@wipsys.soft.net>
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Hello Anirban:

I tried to print your document and the mentioned diagrams did not print at
all. I was wondering if you can check your document and post it again.

I was wondering if others had this problem too!

Thanks.

Bobby Sardana.
sardana@obsoft.com

ANIRBAN ROY wrote:

> hi ,
> I have tried to dipict the Call Flow for the Home Phone with some
> diagram to make it easy to understand. I have some doubts in those Call
> Flow. Please look into it and give some suggestion for this flow.
>
>  I am attaching a windows word doc file with it which has the call flow
> with diagramatic representation of each step.
>
> I am sorry that i am attaching Windows word doc file as I don't have any
>
> other tool for making the Diagram.
>
> regards
> Anirban
>
>   ------------------------------------------------------------------------
>                     Name: Call flow.doc
>    Call flow.doc    Type: Microsoft Word Document (application/msword)
>                 Encoding: base64




From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 12 11:28:03 2000
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To: ANIRBAN ROY <anirban.roy@wipro.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Call flow for the Home Phone Application.
References: <387ABDC5.5746B32F@wipsys.soft.net>
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As I have mentioned, it is too premature to begin discussions on the
implementation. We first must decide whether there:

1. is sufficient people to work on this,
2. the ADs approve adding it to the charter,
3. the requirements for the service are defined

then we can figure out how to extend SIP. Regarding (1), I have received
few offers of people willing to actually work on this (two, to be
exact). Feel free to respond privately if you are willing to work.

-Jonathan R.

ANIRBAN ROY wrote:
> 
> hi ,
> I have tried to dipict the Call Flow for the Home Phone with some
> diagram to make it easy to understand. I have some doubts in those Call
> Flow. Please look into it and give some suggestion for this flow.
> 
>  I am attaching a windows word doc file with it which has the call flow
> with diagramatic representation of each step.
> 
> I am sorry that i am attaching Windows word doc file as I don't have any
> other tool for making the Diagram.
> 
> regards
> Anirban
> 
>   ------------------------------------------------------------------------
>                     Name: Call flow.doc
>    Call flow.doc    Type: WINWORD File (application/msword)
>                 Encoding: base64

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 12 16:04:04 2000
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Message-ID: <387CEB72.EADAC4E0@cs.columbia.edu>
Date: Wed, 12 Jan 2000 16:00:34 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Cc: Kundan Singh <kns10@cs.columbia.edu>
Subject: New draft 
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Note: this is **NOT** a work item for the SIP working group. This note
is for information only, since the relevant experts are probably at
least partially gathered around the SIP list camp fire...

Kundan Singh and I wrote a new Internet-Draft:

        Title           : Interworking Between SIP/SDP and H.323
        Author(s)       : K. Singh, H. Schulzrinne 
        Filename        : draft-singh-sip-h323-00.txt,.ps
        Pages           : 46
        Date            : 11-Jan-00

Comments are appreciated. If there are lots of comments or interest in
this topic, we may set up a mailing list. Please let us know.

Unless the chairs advise differently, please do NOT send comments to
this (the SIP-WG) mailing list.

Thank you.

Henning
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
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From: Chip Sharp <chsharp@cisco.com>
Subject: Re: New draft 
Cc: Kundan Singh <kns10@cs.columbia.edu>
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FYI,
It looks like work is being initiated in IUT-T SG16 on this as well.
Chip

At 04:00 PM 1/12/00 -0500, Henning Schulzrinne wrote:
>Note: this is **NOT** a work item for the SIP working group. This note
>is for information only, since the relevant experts are probably at
>least partially gathered around the SIP list camp fire...
>
>Kundan Singh and I wrote a new Internet-Draft:
>
>         Title           : Interworking Between SIP/SDP and H.323
>         Author(s)       : K. Singh, H. Schulzrinne
>         Filename        : draft-singh-sip-h323-00.txt,.ps
>         Pages           : 46
>         Date            : 11-Jan-00
>
>Comments are appreciated. If there are lots of comments or interest in
>this topic, we may set up a mailing list. Please let us know.
>
>Unless the chairs advise differently, please do NOT send comments to
>this (the SIP-WG) mailing list.
>
>Thank you.
>
>Henning
>--
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

Support NetAid!  http://www.netaid.org
--------------------------------------------------
Chip Sharp                 Consulting Engineering
Cisco Systems              Telco Bio-region
Reality - Love it or Leave it.			
--------------------------------------------------




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hi all .I'm new to this concept. Could you please
explain to me in what contexts the "call leg" is used
and in what contexts is the "call id" used ?

thanx

__________________________________________________
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From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 13 04:37:46 2000
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  Hi,
    I'm new to this service and to SIP as a whole. Please tell me, is the ACK
message also sent for a fixed number of retransmissions in UDP? Or is it not?
If it isn't how does the callee know that the response it sent has reached?
(for eg. in the case that it sent a response 200 OK to an INVITE request)
    Also could you please tell me the difference between a stateful and stateless
proxy? Where is the former necessary?
   Thanks.



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 13 04:42:02 2000
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could u please clarify my doubt regarding stateful
proxies that have been mentioned in the RFCs ..it says
that stateful proxies make a comparison of the tuple
with the existing records and if it exists, it is
recoginsed as a retransmission. However, it forwards
the same tuple, just as a stateless proxy. How is a
stateless proxy different from a stateful proxy.
thanx
__________________________________________________
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Talk to your friends online with Yahoo! Messenger.
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Date: Thu, 13 Jan 2000 09:42:33 +0000
From: Marc Petit-Huguenin <petithug@8x8.com>
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-- 
Marc Petit-Huguenin
Home: mailto:petithug@cyberservices.com
Work: mailto:petithug@8x8.com
"Everything I love is either immoral, illegal or makes you fat"



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Hi all,

Sorry for the previous empty message. 

I have some questions/propositions about SIP voice mail. 

I find some information about Voice mail implementation in: 
 http://www.cs.columbia.edu/~hgs/papers/Schu9802_Signaling.ps.gz
 http://www.cs.columbia.edu/~hgs/sip/drafts/requirements.pdf

but I have two problems:

The first problem is for the Voice-mail to choose the correct mailbox. 
If a SIP entity makes a direct connection to a voicemail, the voice mail
can use the From header to select the mailbox (In this case, the SIP
entity is the owner of the mailbox and is connected in administrative
mode, for retrieving messages or configure his mail box). The voice mail
asks for PIN code or rejects with 407.
But if the connection is an indirect connection, the voice mail do not
known how to select the mailbox:

Imagine A and B are UA SIP and V is the SIP voicemail for B.
A calls B (perhaps through a proxy) then a forwarding rule on B
redirects A to V. V do not know that the call was forwarded by B so it
cannot directly connect A to the mailbox of B.

One solution for B is to act as a SIP proxy for this call. V can check
the first Via header (or best, the first Record-Route), and deduce the
correct mailbox from this information.

Another solution is to add a token in the SIP address returned in the
Contact field of the 302:
 Contact: <sip:V@8x8.com;com.8x8.redirect="A@8x8.com">

(I am not sure that a SIP entity have the obligation to keep all the
tokens)

My second problem is to notify a SIP entity that new voicemails are
arrived.

A solution is to use the register method to convey this information. My
email are checked every 10 minutes, so we can imagine that if a SIP UA
registers each 10 minutes, the mailbox informations can be send in the
200 response:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.36:5060
From: sip:A@8x8.com
To: sip:A@8x8.com
Call-Id: 1234@192.168.0.36
CSeq: 523 INVITE
Voice-Mail: true;new=5;urgent=0;archived=0

The problem is that we must have a link between the register server and
the voice-mail.

Any comments?

-- 
Marc Petit-Huguenin
Home: mailto:petithug@cyberservices.com
Work: mailto:petithug@8x8.com
"Everything I love is either immoral, illegal or makes you fat"



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As a general rule, folks who are new to SIP are encouraged to browse
through the many tutorials and papers at
http://www.cs.columbia.edu/~hgs/sip


Siddharth Toshniwal wrote:
> 
>   Hi,
>     I'm new to this service and to SIP as a whole. Please tell me, is the ACK
> message also sent for a fixed number of retransmissions in UDP? Or is it not?

No. ACK is sent when a response retransmission is received.

> If it isn't how does the callee know that the response it sent has reached?

Reliability is achieved because the response is retransmitted until an
ACK comes,
and the ACK is retransmitted on response retransmissions. Note this is
only for INVITE.

> (for eg. in the case that it sent a response 200 OK to an INVITE request)
>     Also could you please tell me the difference between a stateful and stateless
> proxy? Where is the former necessary?

Stateless proxies forget about the request once its forwarded. Stateful
proxies remember the request once its forwarded, so they can associate
the response with some internal state. Stateless proxies scale very
well, and can be very fast. They are good for network cores. Stateful
proxies can do more (they can fork, for example) and can provide
services stateless ones can't (call forward busy, for example). They
don't scale as big as stateless ones. An admininstrator gets to decide
which to use. These are also logical entities; a physical proxy is
likely to act as a stateless proxy for some calls, stateful for others,
and as a redirect server for even others.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Newbies are encouraged to read the tutorials at the SIP homepage,
http://www.cs.columbia.edu/~hgs/sip.

Rajan Pillai wrote:
> 
> hi all .I'm new to this concept. Could you please
> explain to me in what contexts the "call leg" is used
> and in what contexts is the "call id" used ?

Call leg refers to the one to one signaling relationship between two
UAs. The Call-ID is an identifier, carried in the SIP messages, that
refers to the call. A call is a collection of call legs. A UAC starts by
sending an INVITE; because of forking, it may receive multiple 200 OKs
from different UAS. Each corresponds to a different call leg within the
same call. Call is thus a grouping of call legs. In the call control
spec, additional call legs are created through the Also header.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 13 10:13:29 2000
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Rajan Pillai wrote:
> 
> could u please clarify my doubt regarding stateful
> proxies that have been mentioned in the RFCs ..it says
> that stateful proxies make a comparison of the tuple
> with the existing records and if it exists, it is
> recoginsed as a retransmission. However, it forwards
> the same tuple, just as a stateless proxy. How is a
> stateless proxy different from a stateful proxy.
> thanx

Tuples aren't forwarded. Messages are forwarded. A stateful
proxy recognizes the request as a retransmission, and DOESN'T
forward it. The stateful proxy will retransmit requests on its
own as if it were a UA, but a stateless proxy just forwards
requests that it receives.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 13 10:39:56 2000
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Marc Petit-Huguenin wrote:
> 
> but I have two problems:
> 
> The first problem is for the Voice-mail to choose the correct mailbox.
> If a SIP entity makes a direct connection to a voicemail, the voice mail
> can use the From header to select the mailbox (In this case, the SIP
> entity is the owner of the mailbox and is connected in administrative
> mode, for retrieving messages or configure his mail box). The voice mail
> asks for PIN code or rejects with 407.
> But if the connection is an indirect connection, the voice mail do not
> known how to select the mailbox:
> 
> Imagine A and B are UA SIP and V is the SIP voicemail for B.
> A calls B (perhaps through a proxy) then a forwarding rule on B
> redirects A to V. V do not know that the call was forwarded by B so it
> cannot directly connect A to the mailbox of B.

The problem is that you are trying to address the mailbox indirectly.
Rather, a voice mailbox is identified by an address, for example:

sip:joe_voicemail@company.com

or even:

sip:voicemail@company.com

In the latter case, the user might be prompted for a pin or password to
identify whose mailbox is being accessed.

So, in your example, B would redirect A to sip:B_voicemail@company.com.
It works fine in this case.


> My second problem is to notify a SIP entity that new voicemails are
> arrived.
> 
> A solution is to use the register method to convey this information. My
> email are checked every 10 minutes, so we can imagine that if a SIP UA
> registers each 10 minutes, the mailbox informations can be send in the
> 200 response:
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.0.36:5060
> From: sip:A@8x8.com
> To: sip:A@8x8.com
> Call-Id: 1234@192.168.0.36
> CSeq: 523 INVITE
> Voice-Mail: true;new=5;urgent=0;archived=0
> 
> The problem is that we must have a link between the register server and
> the voice-mail.
> 
> Any comments?

I happen to dislike using SIP for voicemail notifications, particularly
in this way. The reason is that voicemail is a separate service, very
possibly provided by a provider that is not by SIP proxy/registrar
service. Therefore, linking these together won't work for separate
providers. I don't want to rule that case out. I personally like IMAP;
this allows for unified notifications of both email and voice. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 13 11:07:47 2000
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I've set a mailing list for those interested in the discussion of this
topic. (Given the apparent flurry of interest in the topic, this may
move elsewhere in the future.)  Discussion is *not* limited to the draft
we just put out.

You can subscribe at http://www.egroups.com/group/sip-h323/ by clicking
on "group info" or by sending email to sip-h323-subscribe@egroups.com

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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From: Hemon Philippe <Philippe.Hemon@SRIT.siemens.fr>
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Subject: Contact header grammar
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Hi all,

I've a problem with the Contact header in the BNF grammar. What does <"
means in contact params ?

contact-params  = "q" "=" qvalue| "action" "=" "proxy" | "redirect" |
"expires" "=" delta-seconds | <" SIP-date <" |  extension-attribute 

Does it correspond to "<"?
Thanks.

Philippe





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What document are you looking at? Copying from rfc2543:

> Contact = ( "Contact" | "m" ) ":" 
>              ("*" | (1# (( name-addr | addr-spec )
>              [ *( ";" contact-params ) ] [ comment ] )))
> 
>    name-addr      = [ display-name ] "<" addr-spec ">"
>    addr-spec      = SIP-URL | URI
>    display-name   = *token | quoted-string
> 
>    contact-params = "q"       "=" qvalue
>                   | "action"  "=" "proxy" | "redirect"
>                   | "expires" "=" delta-seconds | <"> SIP-date <">
>                   | extension-attribute
> 
>    extension-attribute = extension-name [ "=" extension-value ]
> 

<"> means the quote character itself:

Contact: sip:jdrosen@dynamicsoft.com; expires="Mon, 01 Jan 9999 00:00:00
GMT"

-Jonathan R.

Hemon Philippe wrote:
> 
> Hi all,
> 
> I've a problem with the Contact header in the BNF grammar. What does <"
> means in contact params ?
> 
> contact-params  = "q" "=" qvalue| "action" "=" "proxy" | "redirect" |
> "expires" "=" delta-seconds | <" SIP-date <" |  extension-attribute
> 
> Does it correspond to "<"?
> Thanks.
> 
> Philippe

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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--------------C4A53FD6F8879021E7B50194
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

The ACK msg, doesnt need to be retransmitted. The reason being that the callee
retransmits 200 OK. So, in the case the ACK gets lost, the 200 OK is retransmitted,
in which case the ACK is sent back again.

Hope that helps
sunitha



Siddharth Toshniwal wrote:

>   Hi,
>     I'm new to this service and to SIP as a whole. Please tell me, is the ACK
> message also sent for a fixed number of retransmissions in UDP? Or is it not?
> If it isn't how does the callee know that the response it sent has reached?
> (for eg. in the case that it sent a response 200 OK to an INVITE request)
>     Also could you please tell me the difference between a stateful and stateless
> proxy? Where is the former necessary?
>    Thanks.

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



--------------C4A53FD6F8879021E7B50194
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
The ACK&nbsp;msg, doesnt need to be retransmitted. The reason being that
the callee retransmits 200 OK. So, in the case the ACK&nbsp;gets lost,
the 200 OK&nbsp;is retransmitted, in which case the ACK is sent back again.
<p>Hope that helps
<br>sunitha
<br>&nbsp;
<br>&nbsp;
<p>Siddharth Toshniwal wrote:
<blockquote TYPE=CITE>&nbsp; Hi,
<br>&nbsp;&nbsp;&nbsp; I'm new to this service and to SIP as a whole. Please
tell me, is the ACK
<br>message also sent for a fixed number of retransmissions in UDP? Or
is it not?
<br>If it isn't how does the callee know that the response it sent has
reached?
<br>(for eg. in the case that it sent a response 200 OK to an INVITE request)
<br>&nbsp;&nbsp;&nbsp; Also could you please tell me the difference between
a stateful and stateless
<br>proxy? Where is the former necessary?
<br>&nbsp;&nbsp; Thanks.</blockquote>

<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------C4A53FD6F8879021E7B50194--




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When SIP is used for SS7 to SS7 interworking, which SIP provisional response can
be used to carry an Early ACM which just means that the address is complete and
the called party is being located (an ACM with "No indication" for 'called
party's status indicator' and without any 'optional backward call indicator')? I
can not use '100 trying' as it is not passed end-to-end while going through
proxies. Can I use 183? In that case, what should be the "Session" header field
value?

Any suggestions?

Thanks,
 - hariram





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Hi Hemon,
I think you are referring to an old copy of the hyperlinked SIP grammar in
Henning's site ?
There was a problem in the old  hyperlinked version, where mysteriously,
the ">" character
was swallowed  (along with many other errors)

The grammar mistakes have been corrected and SDP hyperlink has also been
included in the new one.
Please re-cache the latest grammar from henning's site and discard the old
one.

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems








Hemon Philippe <Philippe.Hemon@SRIT.siemens.fr> on 01/13/2000 09:24:48 PM

To:   sip@lists.research.bell-labs.com
cc:

Subject:  Contact header grammar




Hi all,

I've a problem with the Contact header in the BNF grammar. What does <"
means in contact params ?

contact-params  = "q" "=" qvalue| "action" "=" "proxy" | "redirect" |
"expires" "=" delta-seconds | <" SIP-date <" |  extension-attribute

Does it correspond to "<"?
Thanks.

Philippe










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What happens when User 1 sends an invite request to
User 2 and simultaneously User2 sends an invite
request to User 1 ? Do both accept each others'
connections - in which case, each receives 2 copies of
each message ?
__________________________________________________
Do You Yahoo!?
Talk to your friends online with Yahoo! Messenger.
http://im.yahoo.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan 14 09:14:20 2000
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From: LeFloch Yannick <Yannick.LeFloch@SRIT.siemens.fr>
To: sip@lists.research.bell-labs.com
Subject: Question about WWW-Authenticate and Authorization Headers
Date: Fri, 14 Jan 2000 15:05:47 +0100
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Hi All,

  I have some difficulties understanding some parts of the SIP grammar
definitions.
Could Someone help me about WWW-Authenticate and Authorization Header fields
?

  My problem is the differences between RFC2543 ans Henning Schulzrinne SIP
grammar WebPage definitions of both headers. 
Which version should I use ?

  In the Henning Html SIP grammar description, I never seen pgp-challenge
field in a right part of any rule. I have deduce that
auth-params and pgp-params are equivalent.

Are auth-param = token "=" string and pgp-algorithm = "algorithm" "=" (
"md5" | "sha1" |token ) definitions similar ? ( is pgp-algorithm a
auth-params ?) 



Best Regards,

Yannick LE FLOC'H

SIEMENS Réseaux Informatique et Télécommunications (SRIT)
phone: +33 (0) 2 96 48 74 82
mailto:yannick.lefloch@srit.siemens.fr




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan 14 09:45:49 2000
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rfc2543 is always authoritative.

-Jonathan R.

LeFloch Yannick wrote:
> 
> Hi All,
> 
>   I have some difficulties understanding some parts of the SIP grammar
> definitions.
> Could Someone help me about WWW-Authenticate and Authorization Header fields
> ?
> 
>   My problem is the differences between RFC2543 ans Henning Schulzrinne SIP
> grammar WebPage definitions of both headers.
> Which version should I use ?
> 
>   In the Henning Html SIP grammar description, I never seen pgp-challenge
> field in a right part of any rule. I have deduce that
> auth-params and pgp-params are equivalent.
> 
> Are auth-param = token "=" string and pgp-algorithm = "algorithm" "=" (
> "md5" | "sha1" |token ) definitions similar ? ( is pgp-algorithm a
> auth-params ?)
> 
> Best Regards,
> 
> Yannick LE FLOC'H
> 
> SIEMENS Réseaux Informatique et Télécommunications (SRIT)
> phone: +33 (0) 2 96 48 74 82
> mailto:yannick.lefloch@srit.siemens.fr

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan 14 09:52:04 2000
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What happens is what happens if they weren't simultaneous. User 1 makes
a call, then receives one. User 1 can accept the second if he likes.
Same for user 2. The result would be two separate calls (these will have
different Call-IDs as they are initiated by different parties). 

-Jonathan R. 

Rajan Pillai wrote:
> 
> What happens when User 1 sends an invite request to
> User 2 and simultaneously User2 sends an invite
> request to User 1 ? Do both accept each others'
> connections - in which case, each receives 2 copies of
> each message ?
> __________________________________________________
> Do You Yahoo!?
> Talk to your friends online with Yahoo! Messenger.
> http://im.yahoo.com

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Fri, 14 Jan 2000 10:30:42 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE:  overlapping invites
In-reply-to: <20000114070442.6486.qmail@web3305.mail.yahoo.com>
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Rajan Pillai asked:
> What happens when User 1 sends an invite request to
> User 2 and simultaneously User2 sends an invite
> request to User 1 ? Do both accept each others'
> connections - in which case, each receives 2 copies of
> each message ?

My guess:

Same thing that happens when two cell phone users call each other
simultaneously, I suppose. One or both of them refuse to answer the incoming
call because they're busy placing a call.

Of course, display of the From: field in the user interface should allow the
user to detect this situation and apply whatever recovery heuristic they
prefer. Personally, I usually hang up, wait ten to thirty seconds, and call
again -- kind of like Ethernet collision backoff on a human scale.

--
Dean




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Cc: Rajan Pillai <sip_mails@yahoo.com>, sip@lists.research.bell-labs.com
Subject: Re: overlapping invites
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Dean Willis wrote:
> 
> Of course, display of the From: field in the user interface should allow the
> user to detect this situation and apply whatever recovery heuristic they
> prefer. Personally, I usually hang up, wait ten to thirty seconds, and call
> again -- kind of like Ethernet collision backoff on a human scale.

It can even be done automatically:

SIP/2.0 500 Try Later
Retry-After: 30

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Dean Willis wrote:

> Rajan Pillai asked:
> > What happens when User 1 sends an invite request to
> > User 2 and simultaneously User2 sends an invite
> > request to User 1 ? Do both accept each others'
> > connections - in which case, each receives 2 copies of
> > each message ?
>
> My guess:
>
> Same thing that happens when two cell phone users call each other
> simultaneously, I suppose. One or both of them refuse to answer the incoming
> call because they're busy placing a call.
>
> Of course, display of the From: field in the user interface should allow the
> user to detect this situation and apply whatever recovery heuristic they
> prefer. Personally, I usually hang up, wait ten to thirty seconds, and call
> again -- kind of like Ethernet collision backoff on a human scale.
>
> --
> Dean

On my cell phone service, both callers will get busy signals and can then leave

simultaneous voice mail messages to the other. This exact situation happened to
me
yesterday ! It would be a useful feature to detect and short circuit this.



Regards

Marshall Eubanks

T.M. Eubanks
CTO
Multicast Technologies, LLC
P.O. Box 99
Clifton, Virginia 20124
Phone : 703-803-8141
Fax     : 703-222-3250
e-mail : tme@21rst-century.com





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Thomas Marshall Eubanks wrote:
> 
> On my cell phone service, both callers will get busy signals and can then leave
> 
> simultaneous voice mail messages to the other. This exact situation happened to
> me
> yesterday ! It would be a useful feature to detect and short circuit this.

An end system can do this; it doesn't require any protocol support.
Check the From field of the incoming call; if it matches the To field of
a call you have in progress, you are in this glare condition. Handling
is at the discretion of the end system. Any of the following are
supported now in SIP and are your choice:

1. answer so two calls are set up
2. return 500 with retry after
3. redirect to voicemail
4. detect the condition, cancel the call, and automatically answer the
call (might want to use a random timer before doing this so both sides
don't do it at the same time)

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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------------------------------------------------------------------------
                 Call for Papers
             1st IP Telephony workshop
                   IPTel'2000
       12- 13 April 2000 in Berlin, Germany
       http://www.fokus.gmd.de/events/iptel2000/
            iptel2000@fokus.gmd.de

Ext. abstracts submission (~2K words):         31.Jan.2000
Authors notification of acceptance:            29.Feb.2000
Camera-ready abstracts and slides:             31.Mar.2000

         Invited talks include H. Schulzrinne

The objective of the First IP Telephony Workshop is to bring together 
researchers, developers, vendors and service providers working in the
IP telephony area to participate actively in a discussion on recent 
deployment experiences,  innovative results and future directions. 
Topics include but are not limited to: Basic Technologies (SIP, ...),
IP-Telephony Services, Business Deployment, Implementation reports.

         Demonstrations are welcome
------------------------------------------------------------------------



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archow@hss.hns.com wrote:
> 
> Hi,
> Can't we use the general NOTIFY and SUBSCRIBE messages for any
> notifications that you want to implement in SIP ?
> If a UAC wants to get notified for a new Voice mail, it can subscribe to
> the event notifier, and the event notifier can query the
> VM box to look for new mails and notify the user ?
> We had a similar need quite some time ago, and somehow, we did not want to
> use IMAP as the above pure-sip solution looked good - using
> the generic event notificcation service to notify a user of any general
> issue.

The general feedback I have gotten was that people liked the concept of
IMAP for this, but it was a burden on standalone phones.

As a middle compromise, I do agree that a generic subscribe and notify
mechanism for *sip related events only* is not a bad idea. This has been
discussed numerous times, but with no specific proposals. There are
numerous other services which might be enabled by such a generic
mechanism, including call park and pickup, call return, etc.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com





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Hi,
     The call-leg of a SIP message is identified by the call-id, remote
address and the local address. However, since the call-id is unique for a
call-leg, wouldn't a Call-Id comparison suffice to check if two messages
belong to the same Call leg ? Is my assumption about the Call-Id being
unique for a call-leg correct ? I need to do this comparison for stopping
retransmissions of response messages.

Binu





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Hi,
     The RFC says that two instances of a user sharing a SIP address can
use the same call-id with different tags in the From field. In this case
again, wouldn't it be enough to compare just the Call-Ids and the tags (as
against comparing the full address in the from and to headers) to identify
messages belonging to the same call-leg ?

Binu

>Hi,
>     The call-leg of a SIP message is identified by the call-id, remote
>address and the local address. However, since the call-id is unique for a
>call-leg, wouldn't a Call-Id comparison suffice to check if two messages
>belong to the same Call leg ? Is my assumption about the Call-Id being
>unique for a call-leg correct ? I need to do this comparison for stopping
>retransmissions of response messages.
>
>Binu









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The CallLeg is identified by the CallId, and CSeq number.

SIP messages belonging to the same transaction have the same CallId. But,
can have different CSeq numbers.  So, inorder to stop retransmissions, you
would probably have to check for both CallId, and CSeq number to be the
same, which is inessential, the CallLeg.

Hope that helps
sunitha


kbinu@hss.hns.com wrote:

> Hi,
>      The call-leg of a SIP message is identified by the call-id, remote
> address and the local address. However, since the call-id is unique for a
> call-leg, wouldn't a Call-Id comparison suffice to check if two messages
> belong to the same Call leg ? Is my assumption about the Call-Id being
> unique for a call-leg correct ? I need to do this comparison for stopping
> retransmissions of response messages.
>
> Binu

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
The CallLeg is identified by the CallId, and CSeq number.
<p>SIP messages belonging to the same transaction have the same CallId.
But, can have different CSeq numbers.&nbsp; So, inorder to stop retransmissions,
you would probably have to check for both CallId, and CSeq number to be
the same, which is inessential, the CallLeg.
<p>Hope that helps
<br>sunitha
<br>&nbsp;
<p>kbinu@hss.hns.com wrote:
<blockquote TYPE=CITE>Hi,
<br>&nbsp;&nbsp;&nbsp;&nbsp; The call-leg of a SIP message is identified
by the call-id, remote
<br>address and the local address. However, since the call-id is unique
for a
<br>call-leg, wouldn't a Call-Id comparison suffice to check if two messages
<br>belong to the same Call leg ? Is my assumption about the Call-Id being
<br>unique for a call-leg correct ? I need to do this comparison for stopping
<br>retransmissions of response messages.
<p>Binu</blockquote>

<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------E103D26D8A486E66DDB43B0D--




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From: "Liberati, Ernest" <Ernest.Liberati@GD-ES.COM>
Subject: SIP phone to PSTN phone
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
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Re: SIP phone call to PSTN phone via Proxy Server (with location services),
MGC, STP, and circuit switch.

What is the domain value of the SIP URL in the FROM and TO headers, if:
	The SIP phone client registers with the Proxy Server?
	The SIP phone client is not registered with a Proxy Server?

Ernest Liberati
General Dynamics



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 17 12:15:48 2000
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Subject: Re: Call-Leg identification
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Sunitha Kumar wrote:
> 
> The CallLeg is identified by the CallId, and CSeq number.
> 
> SIP messages belonging to the same transaction have the same CallId.
> But, can have different CSeq numbers. 

This is incorrect.

A call is a collection of call-legs. A call-leg is the combination of
remote and local addresses (which include the tags) and Call-ID, but NOT
the CSeq. Within a call leg, each transaction has a different CSeq. The
CSeq space is within the call leg.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 17 12:19:37 2000
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To: kbinu@hss.hns.com
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No. Because you may receive misrouted messages and need to ensure they
are for you. Odds are good it won't be a problem, but the correct
solution is to use the full address.

-Jonathan R.

kbinu@hss.hns.com wrote:
> 
> Hi,
>      The RFC says that two instances of a user sharing a SIP address can
> use the same call-id with different tags in the From field. In this case
> again, wouldn't it be enough to compare just the Call-Ids and the tags (as
> against comparing the full address in the from and to headers) to identify
> messages belonging to the same call-leg ?
> 
> Binu
> 
> >Hi,
> >     The call-leg of a SIP message is identified by the call-id, remote
> >address and the local address. However, since the call-id is unique for a
> >call-leg, wouldn't a Call-Id comparison suffice to check if two messages
> >belong to the same Call leg ? Is my assumption about the Call-Id being
> >unique for a call-leg correct ? I need to do this comparison for stopping
> >retransmissions of response messages.
> >
> >Binu

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 17 12:43:45 2000
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Thanks Jonathan, for correcting me.

sunitha.


Jonathan Rosenberg wrote:

> Sunitha Kumar wrote:
> >
> > The CallLeg is identified by the CallId, and CSeq number.
> >
> > SIP messages belonging to the same transaction have the same CallId.
> > But, can have different CSeq numbers.
>
> This is incorrect.
>
> A call is a collection of call-legs. A call-leg is the combination of
> remote and local addresses (which include the tags) and Call-ID, but NOT
> the CSeq. Within a call leg, each transaction has a different CSeq. The
> CSeq space is within the call leg.
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Thanks Jonathan, for correcting me.
<p>sunitha.
<br>&nbsp;
<p>Jonathan Rosenberg wrote:
<blockquote TYPE=CITE>Sunitha Kumar wrote:
<br>>
<br>> The CallLeg is identified by the CallId, and CSeq number.
<br>>
<br>> SIP messages belonging to the same transaction have the same CallId.
<br>> But, can have different CSeq numbers.
<p>This is incorrect.
<p>A call is a collection of call-legs. A call-leg is the combination of
<br>remote and local addresses (which include the tags) and Call-ID, but
NOT
<br>the CSeq. Within a call leg, each transaction has a different CSeq.
The
<br>CSeq space is within the call leg.
<p>-Jonathan R.
<br>--
<br>Jonathan D. Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
200 Executive Drive
<br>Chief Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Suite 120
<br>dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
West Orange, NJ 07052
<br>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778
<br><a href="http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244
<br><a href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</a></blockquote>

<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------45D0F73E4ACA5456C4C42D8E--




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"Liberati, Ernest" wrote:
> 
> Re: SIP phone call to PSTN phone via Proxy Server (with location services),
> MGC, STP, and circuit switch.
> 
> What is the domain value of the SIP URL in the FROM and TO headers, if:
>         The SIP phone client registers with the Proxy Server?
>         The SIP phone client is not registered with a Proxy Server?

The From field is the name of the user making the call. If that user,
for example, is a customer of Worldcom, it might be:

From: sip:user@wcom.com

This name must usually be configured into the client application on a
SIP phone, just like it is for email.

In the To field, there is going to be a phone number. There is currently
debate about whether a sip URL or tel URL is there, so:

To: tel:5551212

or:

To: sip:5551212@wcom.com

remains to be decided.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 17 17:21:54 2000
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It is given that there could be multiple INVITEs reaching the client,
because of the proxy forking this.And, for each such INVITE reaching the
UAS, it SHOULD send a 200 OK. My question, is should each of this 200 OK
be considered for re-transmission?
And, what happens at the client receiving multiple 200 OK. Should there
be an ACK for each of it, or should just one ACK handle all of this.

Thanks much!


--
Sunitha Kumar



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br>It is given that there could be multiple INVITEs reaching the client,
because of the proxy forking this.And, for each such INVITE reaching the
UAS, it SHOULD send a 200 OK. My question, is should each of this 200 OK
be considered for re-transmission?
<br>And, what happens at the client receiving multiple 200 OK. Should there
be an ACK for each of it, or should just one ACK handle all of this.
<p>Thanks much!
<br>&nbsp;
<pre>--&nbsp;
Sunitha Kumar</pre>
&nbsp;</html>

--------------6F82A6676E138A4885455667--




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Sunitha Kumar wrote:
> 
> 
> It is given that there could be multiple INVITEs reaching the client,
> because of the proxy forking this.And, for each such INVITE reaching
> the UAS, it SHOULD send a 200 OK.

Actually, I don't think thats the best approach. I circulated a note
some time back on handling of "merged requests", as they are called.
See:

http://www.cs.columbia.edu/~jdrosen/sip/multiple_requests.txt

 My question, is should each of this
> 200 OK be considered for re-transmission?
> And, what happens at the client receiving multiple 200 OK.

THe above solution will guarantee that if it receives multiple 200 OK,
they are from different UAS's. Handling of multiple 200 OK has been
discussed on the list extensively. You should ACK each. Beyond that, the
consensus is that there is no one right approach. THe options are:

1. BYE all after the first
2. keep all; ends up being N point to point call legs that are not
bridged
3. use call control to bridge the legs together, resulting in a
conference call

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: Gateways and registration
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All

Parallel question on the reliance of TRIP -- isn't it based on using BGP? If
that is strickly true, what happens to the SIP Device if BGP isn't deployed
within a VoIP network domain that still wants to locate the Gateway?

At 03:33 PM 1/7/2000 -0500, Jonathan Rosenberg wrote:
>
>
>Linden deCarmo wrote:
>> 
>> This is EXACTLY what I'm trying to solve Sean.
>> 
>> I'd prefer to do it in SIP rather than a companion protocol, because we're
>> trying to interface with 3rd party gateways that may/may not talk TRIP.
>
>I don't see why they would be any more likely to behave properly in a
>non-standard usage of REGISTER. TRIP handles keepalives; its keepalive
>mechanism is quite flexible (borrows from BGP, actually), and you also
>get the routing information you need. IMHO, I'd rather solve the general
>problem with a solid solution than make many small extensions to SIP
>here and there to solve only a subset of the real problem.
>
>A complete VoIP solution requires many components. The general approach
>we have taken into IETF is to break the various components into
>independent, manageable pieces, and then define protocols for solving
>each of those pieces. By keeping these pieces standalone and
>independent, we can swap in different versions or alternatives down the
>road without affecting other components. This helps tremendously in
>extensibility and growth; we have seen this in routing protocols, for
>example (separation of intra and inter domain protocols, even though
>both run on a router). It also means we can optimize the solution for
>the problem at hand. 
>
>For many different reasons, some of which have already been outlined
>here, the SIP REGISTER method is not a good tool for gateways to use. 
>
>-Jonathan R.
>
>-- 
>Jonathan D. Rosenberg                       200 Executive Drive
>Chief Scientist                             Suite 120 
>dynamicsoft                                 West Orange, NJ 07052
>jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>

*************************************
"At the end of the day... the most committed win!"

James M. Polk
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5280
www.cisco.com
--=====================_4066839==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>All</div>
<br>
<div>Parallel question on the reliance of TRIP -- isn't it based on using
BGP? If that is strickly true, what happens to the SIP Device if BGP
isn't deployed within a VoIP network domain that still wants to locate
the Gateway?</div>
<br>
<div>At 03:33 PM 1/7/2000 -0500, Jonathan Rosenberg wrote:</div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;Linden deCarmo wrote:</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; This is EXACTLY what I'm trying to solve Sean.</div>
<div>&gt;&gt; </div>
<div>&gt;&gt; I'd prefer to do it in SIP rather than a companion
protocol, because we're</div>
<div>&gt;&gt; trying to interface with 3rd party gateways that may/may
not talk TRIP.</div>
<div>&gt;</div>
<div>&gt;I don't see why they would be any more likely to behave properly
in a</div>
<div>&gt;non-standard usage of REGISTER. TRIP handles keepalives; its
keepalive</div>
<div>&gt;mechanism is quite flexible (borrows from BGP, actually), and
you also</div>
<div>&gt;get the routing information you need. IMHO, I'd rather solve the
general</div>
<div>&gt;problem with a solid solution than make many small extensions to
SIP</div>
<div>&gt;here and there to solve only a subset of the real
problem.</div>
<div>&gt;</div>
<div>&gt;A complete VoIP solution requires many components. The general
approach</div>
<div>&gt;we have taken into IETF is to break the various components
into</div>
<div>&gt;independent, manageable pieces, and then define protocols for
solving</div>
<div>&gt;each of those pieces. By keeping these pieces standalone
and</div>
<div>&gt;independent, we can swap in different versions or alternatives
down the</div>
<div>&gt;road without affecting other components. This helps tremendously
in</div>
<div>&gt;extensibility and growth; we have seen this in routing
protocols, for</div>
<div>&gt;example (separation of intra and inter domain protocols, even
though</div>
<div>&gt;both run on a router). It also means we can optimize the
solution for</div>
<div>&gt;the problem at hand. </div>
<div>&gt;</div>
<div>&gt;For many different reasons, some of which have already been
outlined</div>
<div>&gt;here, the SIP REGISTER method is not a good tool for gateways to
use. </div>
<div>&gt;</div>
<div>&gt;-Jonathan R.</div>
<div>&gt;</div>
<div>&gt;-- </div>
<div>&gt;Jonathan D.
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
200 Executive Drive</div>
<div>&gt;Chief
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Suite 120 </div>
<div>&gt;dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
West Orange, NJ 07052</div>
<div>&gt;jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (732) 741-4778</div>
<div>&gt;<a href="http://www.cs.columbia.edu/~jdrosen" EUDORA=AUTOURL>http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (732) 741-7244</div>
<div>&gt;<a href="http://www.dynamicsoft.com/" EUDORA=AUTOURL>http://www.dynamicsoft.com</a></div>
<div>&gt;</div>
<div>&gt;</div>
<br>

<div align="center">
*************************************<br>
&quot;At the end of the day... the most committed win!&quot;<br>
<br>
</div>
James M. Polk<br>
Sr. Product Manager, Multiservice Architecture and Standards<br>
Enterprise Voice Business Unit<br>
Cisco Systems<br>
Dallas, Texas<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5280<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_4066839==_.ALT--




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Jan 18 13:53:09 2000
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"James M. Polk" wrote:
> 
> All
> 
> Parallel question on the reliance of TRIP -- isn't it based on using
> BGP? If that is strickly true, what happens to the SIP Device if BGP
> isn't deployed within a VoIP network domain that still wants to locate
> the Gateway?

TRIP borrows many ideas from BGP, but it in no way whatsoever requires
BGP to actually be running on routers in the network. TRIP runs at the
application layer, between location servers. It doesn't matter one drop
what the underlying layer 3 routing protocols are.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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"James M. Polk" wrote:
> 
> All
> 
> Parallel question on the reliance of TRIP -- isn't it based on using
> BGP? If that is strickly true, what happens to the SIP Device if BGP
> isn't deployed within a VoIP network domain that still wants to locate
> the Gateway?
> 

The TRIP protocol is based on BGP, meaning it looks alot like BGP.  But
it isn't BGP - ie its not compatible with routers running BGP, you don't
have to have be an AS to run it, etc.  Its deployment is indepent of any
IP routing protocol.  

- Matt



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Subject: Re: SS7-SIP-SS7 interworking, Early ACM
To: Ravandhu_Hariram@mw.3com.com (Ravandhu Hariram)
Date: Tue, 18 Jan 2000 16:08:45 -0600 (CST)
Cc: sip@lists.research.bell-labs.com
In-Reply-To: <86256865.007EA347.00@mwgate02.mw.3com.com> from "Ravandhu Hariram" at Jan 13, 2000 04:59:57 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Ravandhu Hariram writes:
>When SIP is used for SS7 to SS7 interworking, which SIP provisional response can
>be used to carry an Early ACM which just means that the address is complete and
>the called party is being located (an ACM with "No indication" for 'called
>party's status indicator' and without any 'optional backward call indicator')? I
>can not use '100 trying' as it is not passed end-to-end while going through
>proxies. Can I use 183? In that case, what should be the "Session" header field
>value?

That's generally what I had in mind. My current plan is to map any
of the messages from 180 to 199 to ACM at a PSTN gateway, assuming
we've received an IAM and not sent an ACM, CON, or ANM yet.

Since you will (presumably) not be including any SDP in your message,
you don't use a Session: header.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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How or which SIP message is used to do Circuit Block/UnBlock and/or
Circuit Group Block/UnBlock?






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From: =?ISO-8859-1?Q?S=F6ren_Nyckelg=E5rd?= <Soren.M.Nyckelgard@telia.se>
To: "'Nathan Nelson'" <nathan.nelson@inet.com>
Cc: sip@lists.research.bell-labs.com
Subject: RE: SS7 mapping of Circiut Block Message
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Nathan Nelson wrote:
> 
> How or which SIP message is used to do Circuit Block/UnBlock and/or
> Circuit Group Block/UnBlock?

These messages are relevant only in Swiched Circuit Networks like PSTN and
ISDN. They make no sense in a SIP based IP network.

Might your question refer to an operator bypass scenario (POTS-SIP-POTS)
where you wish to block/unblock circuits in the remote POTS node? You can
not do this since POTS-to-SIP gateways act as transit exchanges. The
block/unblock messages you launch from your end hence refer to devices that
physically reside in the adjacent gateway, not in the remote POTS node. SIP
does not carry these messages.

--
Sören Nyckelgård, Telia



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Hi
     When a SIP message carries a multipart MIME payload, what does the
Content-Length header in the SIP message signify ? Does it contain the
overall length of the payload ? Though it seems logical that it should, I
came across illustrative example messages in the BCP-T draft
(draft-zimmerer-mmusic-sip-bcp-t-00, section 2.2.1) where the
Content-Length header values couldnt be related to the payload length.

Binu





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From: Taji Rachid <Rachid.Taji@srit.siemens.fr>
To: sip@lists.research.bell-labs.com
Subject: is Ack always transmitted by TCP ?
Date: Wed, 19 Jan 2000 17:34:48 +0100
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Hi All;

What is the policy for reliability of ACK requests ?

By reading the RFC SIP ( chapter 10.6 ), I have understood that the only way
to ensure this reliability is to send the Ack request by TCP connection. Is
it mandotary to send Ack request by TCP even if  UDP is used for sending the
prior INVITE? 

Can you clarify this point ?

Thanks

Rachid



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 19 12:23:53 2000
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Reply-To: <gcote@awardsolutions.com>
From: "Gary Cote" <gcote@awardsolutions.com>
To: <sip@lists.research.bell-labs.com>
Subject: RE: is Ack always transmitted by TCP ?
Date: Wed, 19 Jan 2000 11:23:11 -0600
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The ACK request certainly may be sent using UDP.
Reliable delivery is assured through a couple conventions.

1. The user agent server continues to retransmit the final
   INVITE response until it receives an ACK request.

2. The user agent client sends the ACK request each time it
   receives a final INVITE response.

Based on those two characteristics, if you actually map out
the message flow, you'll see that the ACK is reliably delivered.
If the ACK is lost in transmission, the user agent server will
retransmit the final INVITE response, which in turn causes the
the user agent client to re-send the ACK request.

 - Gary

--
Gary Cote
AWARD Solutions, Inc.
Richardson, TX


> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of 
> Taji Rachid
> Sent: Wednesday, January 19, 2000 10:35 AM
> To: sip@lists.research.bell-labs.com
> Subject: is Ack always transmitted by TCP ?
> 
> 
> Hi All;
> 
> What is the policy for reliability of ACK requests ?
> 
> By reading the RFC SIP ( chapter 10.6 ), I have understood 
> that the only way
> to ensure this reliability is to send the Ack request by TCP 
> connection. Is
> it mandotary to send Ack request by TCP even if  UDP is used 
> for sending the
> prior INVITE? 
> 
> Can you clarify this point ?
> 
> Thanks
> 
> Rachid
> 
> 




From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 19 12:49:57 2000
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From: Igor Slepchin <islepchin@dynamicsoft.com>
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Subject: Re: Content-Length header and Multipart MIME
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kbinu@hss.hns.com wrote:
> 
> Hi
>      When a SIP message carries a multipart MIME payload, what does the
> Content-Length header in the SIP message signify ? Does it contain the
> overall length of the payload ? Though it seems logical that it should

I believe that's correct. Content-Length has a uniform meaning
regardless of the type of the payload, namely, the total length of that
payload. The example you quote seem to have a totally random value of
Content-Length. BTW, there is another problem in that example: the SIP
message shown contains two Content-Type fields.

> <...>

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 19 16:30:06 2000
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From: Aparna.Vemuri@Level3.com
To: islepchin@dynamicsoft.com, kbinu@hss.hns.com
Cc: sip@lists.research.bell-labs.com
Subject: RE: Content-Length header and Multipart MIME
Date: Wed, 19 Jan 2000 14:24:57 -0700
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Hello,

First off, the LATEST version of the SIP-BCP-T (or SIP-T) as it will
be known henceforth is available at:
http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt

This draft contains a corrected example.

About the Content-Length calculation - it is as explained below:

This is the example that is used in the SIP-T draft. The number of
 octets used in each line are shown at the end of each line in parentheses.

	INVITE sip:13039263142@Den1.level3.com SIP/2.0
	From: sip:13034513355@den3.level3.com
	To: sip:13039263142@Den1.level3.com
	Call-ID: DEN1231999021712095500999@Den1.level3.com
	Content-Length: 377
	Content-Type: multipart/mixed; boundary=unique-boundary-1
                MIME-Version: 1.0
	
	--unique-boundary-1(19)
	Content-Type: application/SDP; charset=ISO-10646(49)

                v=0(3)
	o=ezimmerer 2890844526 2890842807 IN IP4 126.16.64.4(52)
	s=SDP seminar(13)
   	c=IN IP4 MG122.level3.com(25)
	t=2873397496 2873404696(23)
	m=audio 9092 RTP/AVP 0 3 4(26)

	--unique-boundary-1(19)
	Content-type:application/ISUP;version=ETSI1(44)
	Content-Transfer-Encoding: binary(34)

	89 8b 0e 95 1e 1e 1e 06 26 05 0d f5 01 06 10 04 00(17)
	--unique-boundary-1--(21)


b)  Rules used to calculate the Content-Length:
     b.1 Use 2 octets to encode the CRLF sequence at the end of each line
           and the beginning of a blank line.
     b.2 For the MIME headers in the payloads: each character maps onto one
octet.
     b.3 For the SDP payload: each character maps onto one octet according
             to UTF-8 encoding (RFC 2044) used as specified in the SIP-BCP.
     b.4 For the ISUP payload: Since binary encoding is used, each byte is
encoded
             as a byte, and not as a  two-character hex representation.  Hex
digits were
             used in the draft because a literal encoding of those bytes
would have been
             confusing and unreadable.

c) Content-Length calculation:
     c.1 Number of octets in the ISUP payload (no CRLF at the end since it
is 
            binary)
            = 17
     c.2 Number of octets in the MIME header for the ISUP payload (including
            the CRLF at the end of each line)
            = 46+ 36
            = 82
     c.3 Number of octets in the SDP payload (including the CRLFs at the end
            of each of the lines)
            = 5 + 54 + 15 + 27 + 25 + 28
            = 154
     c.4 Number of octets in the MIME header for the SDP payload (including
            the CRLF at the end of the line)
            =51
     c.5 Number of octets in the '--unique-boundary-1' string and newlines
            used throughout:
             = 21 + 21 + 23 + 4*2 = 73
     c.6 The grand-total
             = 17 + 82 +154 + 51 + 73  = 377



Thanks,
Aparna
Level (3) Communications.





-----Original Message-----
From: Igor Slepchin [mailto:islepchin@dynamicsoft.com]
Sent: Wednesday, January 19, 2000 10:47 AM
To: kbinu@hss.hns.com
Cc: sip@lists.research.bell-labs.com
Subject: Re: Content-Length header and Multipart MIME


kbinu@hss.hns.com wrote:
> 
> Hi
>      When a SIP message carries a multipart MIME payload, what does the
> Content-Length header in the SIP message signify ? Does it contain the
> overall length of the payload ? Though it seems logical that it should

I believe that's correct. Content-Length has a uniform meaning
regardless of the type of the payload, namely, the total length of that
payload. The example you quote seem to have a totally random value of
Content-Length. BTW, there is another problem in that example: the SIP
message shown contains two Content-Type fields.

> <...>

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Jan 19 21:53:48 2000
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Aparna.Vemuri@Level3.com wrote:
> 
> Hello,
> 
> First off, the LATEST version of the SIP-BCP-T (or SIP-T) as it will
> be known henceforth is available at:
> http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
> 

Is it really necessary to label some simple additions to SIP with a new
name? We've had the SIP+ disaster before (in terms of confusion) and I'd
rather not repeat this.




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 20 00:15:48 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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First off, its important to mention what has happened regarding SIP-T
within IETF. As you may recall, consensus during the last meeting was
that people were interested in having the SIP-T work complete in IETF.
However, there wasn't a final decision on what working group, and what
specific documents would be produced. The chairs of the SIP group
discussed this with the ADs, and we agreed this work is best handled
within the SIP working group. Furthermore, several documents would be
produced:

1. SIP INFO (already a work item; in fact, an update should appear in
the archives shortly, and I will be issuing a last call on it)
2. ISUP MIME type RFC (proposed standard)
3. ISUP to SIP interworking RFC (proposed standard)
4. SIP profile for this application (currently named SIP-T): proposed

As for the name, since I think we will do proposed standard for it,
SIP-BCP-T is inappropriate. If people have suggestions for alternative
names to SIP-T, we can entertain them (we changed GLP to TRIP at the
last minute in iptel...)

-Jonathan R.

Henning Schulzrinne wrote:
> 
> Aparna.Vemuri@Level3.com wrote:
> >
> > Hello,
> >
> > First off, the LATEST version of the SIP-BCP-T (or SIP-T) as it will
> > be known henceforth is available at:
> > http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
> >
> 
> Is it really necessary to label some simple additions to SIP with a new
> name? We've had the SIP+ disaster before (in terms of confusion) and I'd
> rather not repeat this.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 20 00:44:55 2000
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Folks,

As many of you have noticed, SIP is becoming a very big and very active
working group. To manage this, the chairs are beginning to delegate
subtasks to specific design teams and task forces. These teams will have
leaders who are responsible for moving the work forward, reporting
progress to the group at large, and managing their teams. You have seen
some of this already with the home phone line emulation work. From this
point forward, we will not proceed with new work unless a team is
identified to work on it, and such a team not being overly committed to
other SIP design teams. The ways in which these teams will work will
vary from team to team. Expect to see announcements about them as they
solidify.

The first team I'd like to announce is the SIP Security Task Force. This
group is tasked with evaluating the SIP security model, so that it can
clarified and strengthened in the next version. Of particular importance
is a formal review of threats and an explicit description of a security
model. The team will produce an I-D to be reviewed by the group for
inclusion in the next RFC. If you have very strong security knowledge
and very good knowledge of SIP, and want to participate, please contact
me for more information.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 20 03:44:00 2000
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Date: Thu, 20 Jan 2000 10:40:19 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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Hi,

When a client is retransmitting a non-INVITE method (BYE for instance)
and receives a provisional response it goes on retransmitting it using
T2 interval.

Why does the client have to do this?
If a provisional response has been received some server is taking care
of the request. Why does it retransmit the request again?

If the server went down the client would notice it very soon due to the
ICMP messages, but I do not think this is worthwhile... which are the
reasons behind this retransmission policy?

Thank you,

Gonzalo
-- 
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



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Gonzalo Camarillo wrote:
> 
> When a client is retransmitting a non-INVITE method (BYE for instance)
> and receives a provisional response it goes on retransmitting it using
> T2 interval.
> 
> Why does the client have to do this?
> If a provisional response has been received some server is taking care
> of the request. Why does it retransmit the request again?

The provisional response means the client knows the UAS (or the next-hop
proxy) has received the request.  The request retransmissions following
receipt of a 1xx is necessary because they will trigger retransmissions
of a lost final response.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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--NextPart

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

	Title		: Mandating SIP Extension Support by Servers
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sip-serverfeatures-00.txt
	Pages		: 9
	Date		: 19-Jan-00
	
The Session Initiation Protocol (SIP) defines mechanims that allow a
client to mandate that a server support specific extensions in order
to process a request. However, there is currently no way for a server
to ask for, and a client to indicate, what features a server might
use in a response. This capability is needed for numerous extensions
currently under development, such as provisional response reliability
and the session progress message. This document defines a SIP
extension that allows servers to mandate that clients understand
certain features in order to process a response.

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

Internet-Drafts are also available by anonymous FTP. Login with the username
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--NextPart

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

	Title		: Reliability of Provisional Responses in SIP
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sip-100rel-00.txt
	Pages		: 16
	Date		: 19-Jan-00
	
This document specifies an extension to the Session Initiation
Protocol (SIP) providing reliable provisional response messages. This
extension uses the option tag org.ietf.sip.100rel.

A URL for this Internet-Draft is:
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: The SIP INFO Method
	Author(s)	: S. Donovan
	Filename	: draft-ietf-sip-info-method-01.txt
	Pages		: 10
	Date		: 19-Jan-00
	
This document proposes an extension to the Session Initiation
Protocol (SIP).  This extension adds the INFO method to the SIP
protocol.  The intent of the INFO method is to allow for the carrying
of session related control information that is generated during a
session.  One example of such session control information is ISUP and
ISDN signaling messages used to control telephony call services.
This and other example uses of the INFO method may be standardized in
the future.

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Date: Thu, 20 Jan 2000 09:42:22 -0600
From: Ben Campbell <ben.campbell@wcom.com>
Subject: FW: I-D ACTION:draft-campbell-sip-service-control-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: Control of Service Context using SIP Request-URI
	Author(s)	: B. Campbell, R. Sparks
	Filename	: draft-campbell-sip-service-control-00.txt
	Pages		: 40
	Date		: 19-Jan-00

In a conventional telephony environment, extended service
applications often use call state information, such as calling party,
called party, reason for forward, etc, to infer application context.
In a SIP/2.0 call, much of this information may be either non-
existent or unreliable. This draft proposes a mechanism to
communicate context information to an application.  Under this
proposal, a client or proxy can communicate context through the use
of a distinctive Request-URI. This draft continues with examples of
how this mechanism could be used in a voice mail application.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-campbell-sip-service-control-00.tx
t

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Below is the data which will enable a MIME compliant mail reader
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From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 20 13:00:11 2000
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Date: Thu, 20 Jan 2000 18:56:38 +0100 (MET)
From: mml+siplist@cslab.ericsson.se
To: sip@lists.research.bell-labs.com
Subject: RTP packet size and use of ptime in SIP
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Hi,

Questions in brief:

	  1. How much audio should there be in an RTP packet?

	  2. Does anyone know of an implementation which changes 
	     packet size mid stream (i.e. without another INVITE)?

	  3. Does anyone know of an implementation which refuses
	     an INVITE with a ptime it doesn't like? Does this seem
	     like a reasonable thing to do?

Detail:

	The RTP RFC says: packet size is not specified, but all 
	    examples have 20ms packets

	The AVP RFC says: 20ms is the "default packetisation interval",
	  but implementations should accept anything from 0 to 200ms.

	  The intent seems to be "send any packet size you want, up
	  to 200ms. If you can't decide what packet size you'd
	  like to send, 20ms isn't a bad choice." (Right?)

	The SDP RFC has a "ptime" parameter which may be ignored:

          "It should not be necessary to know ptime to decode RTP or
          vat audio, and it is intended as a recommendation for the
          encoding/packetisation of audio."

Conclusion: 

         Most SIP/RTP implementations (will) use constant-sized
	 packets. They'll get on well with header compression 
	 schemes (e.g. RFC 2508).

	 A few implementations will also pay attention to ptime.
	 (Does anyone apart from me do this?)

         A few implementations might do something else entirely.
	 There's a paper suggesting variable packet sizes (i.e. 
	 packet size varies unpredictably from packet to packet) as 
	 a means for concealing packet loss:

	        http://noc.aic.net/inet98/6e/6e_3.htm 

Comments?

Matt

----
Matthias Läng                          http://www.ericsson.se/cslab/~mml
Ericsson Computer Science Laboratory
Stockholm


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Message-ID: <388776D6.AAF14B04@dynamicsoft.com>
Date: Thu, 20 Jan 2000 15:57:58 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Cc: sip@lists.research.bell-labs.com, rem-conf@es.net
Subject: Re: RTP packet size and use of ptime in SIP
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This rightly belongs on rem-conf, not on sip.



mml+siplist@cslab.ericsson.se wrote:
> 
> Hi,
> 
> Questions in brief:
> 
>           1. How much audio should there be in an RTP packet?

Its flexible; there is a tradeoff between packet overheads (the more
data, the less the overhead) and delay (the more data you put in the
packet, the more delay before it can be sent). 

> 
>           2. Does anyone know of an implementation which changes
>              packet size mid stream (i.e. without another INVITE)?

Actually, rfc1890 says receivers should be allowed to receive anywhere
up to 200ms of data in a single packet:


>   constraints, a higher packetization delay may be appropriate. A
>    receiver should accept packets representing between 0 and 200 ms of
>    audio data. T


> 
>           3. Does anyone know of an implementation which refuses
>              an INVITE with a ptime it doesn't like? Does this seem
>              like a reasonable thing to do?

You don't need to do a re-INVITE to change packetization delays,
actually. See the above. Why would you want to do this?


>         The AVP RFC says: 20ms is the "default packetisation interval",
>           but implementations should accept anything from 0 to 200ms.
> 
>           The intent seems to be "send any packet size you want, up
>           to 200ms. If you can't decide what packet size you'd
>           like to send, 20ms isn't a bad choice." (Right?)

This is for sample based codecs. For frame based codecs, you should be
prepared to receive any number of frames in a packet, up to the rounded
up integer of 200 ms divided by the frame duration (so for g.723.1 is 7
frames). I don't think there is a recommendation of the default
packetization interval.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Thu, 20 Jan 2000 16:06:44 -0500
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Folks,

I am now issuing a working group last call for the INFO method
extension. This last call period starts today, and ends two weeks hence,
February 3. If no comments are received by then, this draft will be
submitted to IESG for publication as proposed standard.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: The SIP INFO Method
	Author(s)	: S. Donovan
	Filename	: draft-ietf-sip-info-method-01.txt
	Pages		: 10
	Date		: 19-Jan-00
	
This document proposes an extension to the Session Initiation
Protocol (SIP).  This extension adds the INFO method to the SIP
protocol.  The intent of the INFO method is to allow for the carrying
of session related control information that is generated during a
session.  One example of such session control information is ISUP and
ISDN signaling messages used to control telephony call services.
This and other example uses of the INFO method may be standardized in
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It is given in the spec, that a To field can be tagged by the callee.
However, can the From field be tagged by the callee, if the caller had
not tagged it?

Thanks!

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
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It is given in the spec, that a To field can be tagged by the callee.
<br>However, can the From field be tagged by the callee, if the caller
had not tagged it?
<p>Thanks!
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------5654634ABF4C76DCB925BEEC--




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In-Reply-To: <38879187.8EA8F646@vovida.com> from "Sunitha Kumar" at Jan 20, 2000 02:51:51 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>It is given in the spec, that a To field can be tagged by the callee.
>However, can the From field be tagged by the callee, if the caller had
>not tagged it?

No.

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan 21 01:22:12 2000
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Folks,

This draft represents the summary of what I understood as consensus
during the last IETF meeting. Although we had discussed the possibility
of adding this stuff directly into the draft version of SIP, I think it
should be a standalone RFC, for the following reasons:

1. we should avoid adding new features into SIP, but rather in
extensions
2. we need this before SIP will be draft standard

This is a small but very important extension to SIP. Provisional
reliability and other up and coming extensions need this. 

So, I am issuing an "extended" working group last call on this, which
will last three weeks, starting today, and ending on Feb. 11.  I'm
making it extended since although we have agreed on this, there are some
small open issues that must be resolved quickly, and people  have not
seen a draft per se on this subject. However, I am doing this as a last
call to expedite the process, and because we have already agreed to it
at the last meeting. There will be another rev of this document to
address the open issues described. Please comment on these asap.

-Jonathan R.


 Subject: 
        I-D ACTION:draft-ietf-sip-serverfeatures-00.txt
   Date: 
        Thu, 20 Jan 2000 06:58:57 -0500
   From: 
        Internet-Drafts@ietf.org
     To: 
        IETF-Announce:;
    CC: 
        sip@lists.research.bell-labs.com



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

        Title           : Mandating SIP Extension Support by Servers
        Author(s)       : J. Rosenberg, H. Schulzrinne
        Filename        : draft-ietf-sip-serverfeatures-00.txt
        Pages           : 9
        Date            : 19-Jan-00
        
The Session Initiation Protocol (SIP) defines mechanims that allow a
client to mandate that a server support specific extensions in order
to process a request. However, there is currently no way for a server
to ask for, and a client to indicate, what features a server might
use in a response. This capability is needed for numerous extensions
currently under development, such as provisional response reliability
and the session progress message. This document defines a SIP
extension that allows servers to mandate that clients understand
certain features in order to process a response.

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



-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Fri, 21 Jan 2000 09:06:14 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: some queries/comments on call scenarios
To: HEMANT AGRAWAL <hemant.agrawal@wipro.com>
Cc: sip@lists.research.bell-labs.com, steven.r.donovan@wcom.com,
        Gonzalo.Camarillo@ericsson.com
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Hermant,

Thanks for your comments on the draft.

The examples given in the draft assume ANSI ISUP, which does not have a CON
message.  You are correct - for non-ANSI ISUP implementations, the CON would be
sent instead of a ANM if no ACM had been sent.

Refer to Section 7.3 and 9. of draft-camarillo-mmusic-sip-isup-bcp-00.txt "Best
Current Practice for ISUP to SIP mapping" by Gonzalo Camarillo, which is the
standard for SIP to ISUP mapping.

Alan Johnston
MCI WorldCom
+1-314-342-7360

HEMANT AGRAWAL wrote:
> 
>  Hi ,
>          I have some queries/comments  on "draft-
> johnston-sip-call-flows-00"
> 
>   In the scenario 5.1.2, Successful PSTN to SIP call, Fast Answer
>    After receiving 200 OK ( f6), GW1 is sending ANM (f9) to user A without
> sending ACM
>    I suppose it should send CON (connect) if it has not sent ACM prior.
>    After sending IAM the ISUP sdl in USER_A will be in wait_for_ACM
> state(Ref. ITU Q.764)
>    so it will not expect ANM to come, however CON can come as setup
> response.
> 
> Please correct me if I am wrong
> 
> Thanks!
> Hemant



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From: "HEMANT AGRAWAL" <hemant.agrawal@wipro.com>
To: <sip@lists.research.bell-labs.com>, <alan.johnston@wcom.com>
Cc: <steven.r.donovan@wcom.com>
References: <388778E4.D4C16E40@dynamicsoft.com>
Subject: some queries/comments on call scenarios
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 Hi ,
         I have some queries/comments  on "draft-
johnston-sip-call-flows-00"

  In the scenario 5.1.2, Successful PSTN to SIP call, Fast Answer
   After receiving 200 OK ( f6), GW1 is sending ANM (f9) to user A without
sending ACM
   I suppose it should send CON (connect) if it has not sent ACM prior.
   After sending IAM the ISUP sdl in USER_A will be in wait_for_ACM
state(Ref. ITU Q.764)
   so it will not expect ANM to come, however CON can come as setup
response.

Please correct me if I am wrong

Thanks!
Hemant








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From: Matthias.Lang@ericsson.com
To: sip@lists.research.bell-labs.com
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: RTP packet size and use of ptime in SIP
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Jonathan Rosenberg writes:

 > >           3. Does anyone know of an implementation which refuses
 > >              an INVITE with a ptime it doesn't like? Does this seem
 > >              like a reasonable thing to do?

 > You don't need to do a re-INVITE to change packetization delays,
 > actually. See the above. Why would you want to do this?

I don't. What I'm mainly concerned about is someone sending me an
INVITE with a ptime I am unable to support.

To make this easier to understand, here's a scenario: I have a gateway
which supports many concurrent calls. The gateway has limited
resources, e.g. my ethernet controllers can send N UDP packets per
second maximum. For the example, let's assume N = 1000.

I already have several calls in progress, and they're consuming 900 of
those 1000 UDP packets/s. If someone sends me an INVITE with a ptime
of 5 ms, I have two choices:

      1. Ignore the requested ptime, and use larger packets to avoid
         giving myself capacity problems

      2. Return 480 "Temorarily Not Available", knowing that I'll be
         able to handle a 5ms packet call eventually.

I don't think there's anything in SIP or SDP which says one of these
is preferable. #1 seems the lesser of two evils.

This maybe also affects attempts to reserve network bandwidth.

Matt



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To: sip@lists.research.bell-labs.com
Subject: RESPONSE to an ACK
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What is the behavior of an UAS when receiving an ACK with an unknown Call-ID
?
(if I understood correctly the RCF, there is no RESPONSE sent to an ACK)
(for all other REQUEST messages (INVITE/BYE etc.) the UAS sends a 404
response)

J. Blanchard



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Blanchard Jacqueline wrote:
> 
> What is the behavior of an UAS when receiving an ACK with an unknown
> Call-ID ?

Drop it. From RFC 2543, section 7.4.18:

7.4.18 481 Call Leg/Transaction Does Not Exist

This status is returned under two conditions: The server received a BYE
request that does not match any existing call leg or the server received
a CANCEL request that does not match any existing transaction. (A server
simply discards an ACK referring to an unknown transaction.)

> (if I understood correctly the RCF, there is no RESPONSE sent to an ACK)

Right.

> (for all other REQUEST messages (INVITE/BYE etc.) the UAS sends a 404
> response)

481.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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Hello,

Does anybody know, where I can get examples of SIP-messages for
testing?

-- 
Best regards,
 Stanislav S. Timinsky              mailto:timinsky@mera.ru





From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 24 10:02:32 2000
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Some "torture" test messages are at:

http://www.cs.columbia.edu/~hgs/sip/bakeoff2/testmsg.html

which test both parser and some basic functions.

Also, check out the call flows document:

http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sparks-sip-service-examples-00.txt



-Jonathan R.

"Stanislav S. Timinsky" wrote:
> 
> Hello,
> 
> Does anybody know, where I can get examples of SIP-messages for
> testing?
> 
> --
> Best regards,
>  Stanislav S. Timinsky              mailto:timinsky@mera.ru

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 24 10:07:36 2000
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2nd sip bakeoff web pages have some test messages.

http://www.cs.columbia.edu/~hgs/sip/bakeoff2/testmsg.html

there may be more elsewhere... i just remembered seeing these.

kevin
"Stanislav S. Timinsky" wrote:
> 
> Hello,
> 
> Does anybody know, where I can get examples of SIP-messages for
> testing?
> 
> --
> Best regards,
>  Stanislav S. Timinsky              mailto:timinsky@mera.ru

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=



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Date: Mon, 24 Jan 2000 09:36:19 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: Sample SIP-messages
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Stanislav S. Timinsky" <timinsky@mera.ru>,
        sip@lists.research.bell-labs.com, prj.men.sip@mera.ru
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Also, there's the "SIP Telephony Call Flow Examples" draft:

http://www.ietf.org/internet-drafts/draft-johnston-sip-call-flows-00.txt

 	Acrobat and PostScript versions containing call flow diagrams:

	http://www.greycouncil.com/sip/drafts/draft-johnston-sip-call-flows-00.pdf
	http://www.greycouncil.com/sip/drafts/draft-johnston-sip-call-flows-00.ps

The next version of this draft will combine the service examples and test
messages as well.

Alan Johnston
MCI WorldCom

Jonathan Rosenberg wrote:
> 
> Some "torture" test messages are at:
> 
> http://www.cs.columbia.edu/~hgs/sip/bakeoff2/testmsg.html
> 
> which test both parser and some basic functions.
> 
> Also, check out the call flows document:
> 
> http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sparks-sip-service-examples-00.txt
> 
> -Jonathan R.
> 
> "Stanislav S. Timinsky" wrote:
> >
> > Hello,
> >
> > Does anybody know, where I can get examples of SIP-messages for
> > testing?
> >
> > --
> > Best regards,
> >  Stanislav S. Timinsky              mailto:timinsky@mera.ru
> 
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



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What you say is true for ITU and ETSI networks, but the authors come from a
US company, MCI Worldcom. In the US, ANSI ISUP is used, and there is no CON
message. Instead, ANM is always sent whether ACM was sent previously or not.

Steve Carr

-----Original Message-----
From: HEMANT AGRAWAL [mailto:hemant.agrawal@wipro.com]
Sent: Friday, January 21, 2000 4:22 AM
To: sip@lists.research.bell-labs.com; alan.johnston@wcom.com
Cc: steven.r.donovan@wcom.com
Subject: some queries/comments on call scenarios



 Hi ,
         I have some queries/comments  on "draft-
johnston-sip-call-flows-00"

  In the scenario 5.1.2, Successful PSTN to SIP call, Fast Answer
   After receiving 200 OK ( f6), GW1 is sending ANM (f9) to user A without
sending ACM
   I suppose it should send CON (connect) if it has not sent ACM prior.
   After sending IAM the ISUP sdl in USER_A will be in wait_for_ACM
state(Ref. ITU Q.764)
   so it will not expect ANM to come, however CON can come as setup
response.

Please correct me if I am wrong

Thanks!
Hemant







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Subject: Re: RESPONSE to an ACK
To: ak@hplb.hpl.hp.com (Anders Kristensen)
Date: Mon, 24 Jan 2000 10:47:57 -0600 (CST)
Cc: Jacqueline.Blanchard@SRIT.siemens.fr (Blanchard Jacqueline),
        sip@lists.research.bell-labs.com
In-Reply-To: <388C5EA5.80F6FFD3@hplb.hpl.hp.com> from "Anders Kristensen" at Jan 24, 2000 02:16:05 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Anders Kristensen writes:
>Blanchard Jacqueline wrote:
>> 
>> What is the behavior of an UAS when receiving an ACK with an unknown
>> Call-ID ?
...
>> (if I understood correctly the RCF, there is no RESPONSE sent to an ACK)
...
>> (for all other REQUEST messages (INVITE/BYE etc.) the UAS sends a 404
>> response)
>
>481.

Responding with a 481 to an INVITE just because it contains
an unrecognised Call-ID wouldn't prove very useful. Think 
about it.

The only reasonable circumstance under which one could
send a 481 in response to an INVITE is if a UAS receives
an INVITE for a call-leg it is not aware of, and there is
already a tag present in the To: field.

481 never makes sense for OPTIONS and REGISTER.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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From: Sunitha Kumar <skumar@vovida.com>
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To: "Stanislav S. Timinsky" <timinsky@mera.ru>
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Stanislav:

The vovida SIP stack provides a driver which is present in the test
directory of the Stack,
THe driver is called Invitetest.cxx.
This has some of the SIP messages constructed.

Another reference is in SIP RFC 2543 , at the end . They have a few
examples.

Hope that helps.

sunitha


"Stanislav S. Timinsky" wrote:

> Hello,
>
> Does anybody know, where I can get examples of SIP-messages for
> testing?
>
> --
> Best regards,
>  Stanislav S. Timinsky             mailto:timinsky@mera.ru

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Stanislav:
<p>The vovida SIP stack provides a driver which is present in the test
directory of the Stack,
<br>THe driver is called Invitetest.cxx.
<br>This has some of the SIP messages constructed.
<p>Another reference is in SIP RFC 2543 , at the end . They have a few
examples.
<p>Hope that helps.
<p>sunitha
<br>&nbsp;
<p>"Stanislav S. Timinsky" wrote:
<blockquote TYPE=CITE>Hello,
<p>Does anybody know, where I can get examples of SIP-messages for
<br>testing?
<p>--
<br>Best regards,
<br>&nbsp;Stanislav S. Timinsky&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;

<a href="mailto:timinsky@mera.ru">mailto:timinsky@mera.ru</a></blockquote>

<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------2FC17A88ABB7B8B1DF244A1C--




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Date: Mon, 24 Jan 2000 11:45:22 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: Comments on call scenarios, call for volunteers . . .
In-reply-to: 
 <153BDA136259D311859900A0C99B5263072152@boca-isbu-nt04.boca.unisphere.com>
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Clearly some examples and expertise with ITU ISUP are needed in the call
flows draft work . . . probably lots of other stuff too.

Please feed ideas to Alan Johnston (alan.johnston@wcom.com) who has
volunteered to coordinate this effort. And of course if you actually want to
help him with the work, please get together with him.

Personally, I think we need to get a more active design team together to
deal with the call flows work -- it's more than any one guy should have to
deal with.

--
Dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr, Steve
> Sent: Monday, January 24, 2000 9:04 AM
> To: 'HEMANT AGRAWAL'; 'sip@lists.research.bell-labs.com';
> 'alan.johnston@wcom.com'
> Cc: 'steven.r.donovan@wcom.com'
> Subject: RE: some queries/comments on call scenarios
>
>
> What you say is true for ITU and ETSI networks, but the authors
> come from a
> US company, MCI Worldcom. In the US, ANSI ISUP is used, and there
> is no CON
> message. Instead, ANM is always sent whether ACM was sent
> previously or not.
>
> Steve Carr
>
> -----Original Message-----
> From: HEMANT AGRAWAL [mailto:hemant.agrawal@wipro.com]
> Sent: Friday, January 21, 2000 4:22 AM
> To: sip@lists.research.bell-labs.com; alan.johnston@wcom.com
> Cc: steven.r.donovan@wcom.com
> Subject: some queries/comments on call scenarios
>
>
>
>  Hi ,
>          I have some queries/comments  on "draft-
> johnston-sip-call-flows-00"
>
>   In the scenario 5.1.2, Successful PSTN to SIP call, Fast Answer
>    After receiving 200 OK ( f6), GW1 is sending ANM (f9) to user A without
> sending ACM
>    I suppose it should send CON (connect) if it has not sent ACM prior.
>    After sending IAM the ISUP sdl in USER_A will be in wait_for_ACM
> state(Ref. ITU Q.764)
>    so it will not expect ANM to come, however CON can come as setup
> response.
>
> Please correct me if I am wrong
>
> Thanks!
> Hemant




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From: "Rick Workman" <rworkman@nortelnetworks.com>
Organization: Nortel Networks
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Has there been any followup discussion on the "SIP for Presence" draft
(draft-rosenburg-sip-pip-00) since its release in Nov. '98, or can I
assume it reflects the current state of this activity? In particular,
has there been any resolution of the four open issues (section 11)?


Rick Workman
Nortel Networks, Ottawa




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Date: Mon, 24 Jan 2000 13:52:42 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Presence is discussed on the impp mailing list. You can find more
information at:

http://www.ietf.org/html.charters/impp-charter.html

I have pushed for this draft to be the basis for impp, with only limited
success. 

-Jonathan R.

Rick Workman wrote:
> 
> Has there been any followup discussion on the "SIP for Presence" draft
> (draft-rosenburg-sip-pip-00) since its release in Nov. '98, or can I
> assume it reflects the current state of this activity? In particular,
> has there been any resolution of the four open issues (section 11)?
> 
> Rick Workman
> Nortel Networks, Ottawa

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hi:

It is given in RFC 2543, that the From field in a response should copy
all the fields from the From
field of the request. However, it can add some other parameters to it if
needed.

My question was, if the request had a 5060 port(which is the default
port) in the msg, can the response omit this parameter

Example:
If  Request has:

From :  <sip:watson@bell-tel.com:5060>

Can response have:
From: <sip:watson@bell-tel.com>

Thanks!

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
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&nbsp;
<br>Hi:
<p>It is given in RFC 2543, that the From field in a response should copy
all the fields from the From
<br>field of the request. However, it can add some other parameters to
it if needed.
<p>My question was, if the request had a 5060 port(which is the default
port) in the msg, can the response omit this parameter
<p>Example:
<br>If&nbsp; Request has:
<p>From :&nbsp; &lt;sip:watson@bell-tel.com:5060>
<p>Can response have:
<br>From: &lt;sip:watson@bell-tel.com>
<p>Thanks!
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------F4262CACFBD8E01433FD4048--




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 24 18:00:06 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Sunitha Kumar <skumar@vovida.com>
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Subject: Re: Question about default port in response
References: <388CCC0E.440AD239@vovida.com>
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This was the subject of discussion during the bakeoff, and on the list
afterwards. The final decision was that URI comparison rules would
define two URIs as being equal if one includes port 5060, and the other
says nothing. 

THis is described in the FAQ and has already been incorporated into the
revised spec:

http://www.cs.columbia.edu/~hgs/sip/faq.html#to

-Jonathan R.

Sunitha Kumar wrote:
> 
> 
> Hi:
> 
> It is given in RFC 2543, that the From field in a response should copy
> all the fields from the From
> field of the request. However, it can add some other parameters to it
> if needed.
> 
> My question was, if the request had a 5060 port(which is the default
> port) in the msg, can the response omit this parameter
> 
> Example:
> If  Request has:
> 
> From :  <sip:watson@bell-tel.com:5060>
> 
> Can response have:
> From: <sip:watson@bell-tel.com>
> 
> Thanks!
> 
> --
> Sunitha Kumar
> Software Engineer
> Vovida Networks
> (408) 957 - 6374
> 
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Jan 25 02:10:10 2000
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To: Aparna.Vemuri@Level3.com
Cc: islepchin@dynamicsoft.com, kbinu@hss.hns.com,
        sip@lists.research.bell-labs.com
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Date: Tue, 25 Jan 2000 12:00:17 +0530
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Hi,
I have a question regarding the computation of Content-Length you
mentioned.

Aparna> Rules used to calculate the Content-Length:
Aparna> b.1 Use 2 octets to encode the CRLF sequence at the end of each
line
Aparna> and the beginning of a blank line.

Does in necessarily have to be two octets ? the sip grammar defines the
separator as
CR | LF | CRLF depending on the target system's interpretation of a new
line.

So isnt't it that whether 1 or 2 octets for end-of-line is taken for C-Len
should vary
depending on what is the line separator one is using ?


Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems






Aparna.Vemuri@Level3.com on 01/20/2000 02:54:57 AM

To:   islepchin@dynamicsoft.com, K Binu/HSSBLR
cc:   sip@lists.research.bell-labs.com

Subject:  RE: Content-Length header and Multipart MIME




Hello,

First off, the LATEST version of the SIP-BCP-T (or SIP-T) as it will
be known henceforth is available at:
http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt

This draft contains a corrected example.

About the Content-Length calculation - it is as explained below:

This is the example that is used in the SIP-T draft. The number of
 octets used in each line are shown at the end of each line in parentheses.

     INVITE sip:13039263142@Den1.level3.com SIP/2.0
     From: sip:13034513355@den3.level3.com
     To: sip:13039263142@Den1.level3.com
     Call-ID: DEN1231999021712095500999@Den1.level3.com
     Content-Length: 377
     Content-Type: multipart/mixed; boundary=unique-boundary-1
                MIME-Version: 1.0

     --unique-boundary-1(19)
     Content-Type: application/SDP; charset=ISO-10646(49)

                v=0(3)
     o=ezimmerer 2890844526 2890842807 IN IP4 126.16.64.4(52)
     s=SDP seminar(13)
     c=IN IP4 MG122.level3.com(25)
     t=2873397496 2873404696(23)
     m=audio 9092 RTP/AVP 0 3 4(26)

     --unique-boundary-1(19)
     Content-type:application/ISUP;version=ETSI1(44)
     Content-Transfer-Encoding: binary(34)

     89 8b 0e 95 1e 1e 1e 06 26 05 0d f5 01 06 10 04 00(17)
     --unique-boundary-1--(21)


b)  Rules used to calculate the Content-Length:
     b.1 Use 2 octets to encode the CRLF sequence at the end of each line
           and the beginning of a blank line.
     b.2 For the MIME headers in the payloads: each character maps onto one
octet.
     b.3 For the SDP payload: each character maps onto one octet according
             to UTF-8 encoding (RFC 2044) used as specified in the SIP-BCP.
     b.4 For the ISUP payload: Since binary encoding is used, each byte is
encoded
             as a byte, and not as a  two-character hex representation.
Hex
digits were
             used in the draft because a literal encoding of those bytes
would have been
             confusing and unreadable.

c) Content-Length calculation:
     c.1 Number of octets in the ISUP payload (no CRLF at the end since it
is
            binary)
            = 17
     c.2 Number of octets in the MIME header for the ISUP payload
(including
            the CRLF at the end of each line)
            = 46+ 36
            = 82
     c.3 Number of octets in the SDP payload (including the CRLFs at the
end
            of each of the lines)
            = 5 + 54 + 15 + 27 + 25 + 28
            = 154
     c.4 Number of octets in the MIME header for the SDP payload (including
            the CRLF at the end of the line)
            =51
     c.5 Number of octets in the '--unique-boundary-1' string and newlines
            used throughout:
             = 21 + 21 + 23 + 4*2 = 73
     c.6 The grand-total
             = 17 + 82 +154 + 51 + 73  = 377



Thanks,
Aparna
Level (3) Communications.





-----Original Message-----
From: Igor Slepchin [mailto:islepchin@dynamicsoft.com]
Sent: Wednesday, January 19, 2000 10:47 AM
To: kbinu@hss.hns.com
Cc: sip@lists.research.bell-labs.com
Subject: Re: Content-Length header and Multipart MIME


kbinu@hss.hns.com wrote:
>
> Hi
>      When a SIP message carries a multipart MIME payload, what does the
> Content-Length header in the SIP message signify ? Does it contain the
> overall length of the payload ? Though it seems logical that it should

I believe that's correct. Content-Length has a uniform meaning
regardless of the type of the payload, namely, the total length of that
payload. The example you quote seem to have a totally random value of
Content-Length. BTW, there is another problem in that example: the SIP
message shown contains two Content-Type fields.

> <...>

---
Igor Slepchin








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From: Samandi Sami <Sami.Samandi@SRIT.siemens.fr>
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Subject: Using SDP to describe a H323 session
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Dear all,

  I want an endpoint to start a SIP session describing a H323 call (e.g the
INVITE is only aimed to start the H323 call). As the  H323 call will not be
under control of SIP (media characteristics are unknown during the SIP
session) I have difficulties to describe the H323 call using SDP. Based on
what I have read in the MMUSIC mailig list archive I want to do the
following :

c=IN IP4 A.B.C.D
m=control 1719 H323 caps 

This mean a call should be addressed to A.B.C.D on port 1719 using Q931
(this is a direct routed call).

am I right or wrong ?

Thank you
Sami






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Subject: Re: Content-Length header and Multipart MIME
To: archow@hss.hns.com
Date: Tue, 25 Jan 2000 09:00:50 -0600 (CST)
Cc: Aparna.Vemuri@Level3.com, islepchin@dynamicsoft.com, kbinu@hss.hns.com,
        sip@lists.research.bell-labs.com
In-Reply-To: <65256871.0023BBD0.00@sampark.hss.hns.com> from "archow@hss.hns.com" at Jan 25, 2000 12:00:17 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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archow@hss.hns.com writes:
>Aparna> Rules used to calculate the Content-Length:
>Aparna> b.1 Use 2 octets to encode the CRLF sequence at the end of each
>line
>Aparna> and the beginning of a blank line.
>
>Does in necessarily have to be two octets ? the sip grammar defines the
>separator as
>CR | LF | CRLF depending on the target system's interpretation of a new
>line.
>
>So isnt't it that whether 1 or 2 octets for end-of-line is taken for C-Len
>should vary
>depending on what is the line separator one is using ?

The "CR | LF | CRLF" is for parsing of messages. In RFC2543, sec 6.6,
it says that "applications MUST follow HTTP common form when generating
[headers]." This denotes a full CRLF at the end of each line.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: sip@lists.research.bell-labs.com
Subject: comments on SIP INFO last call
Date: Tue, 25 Jan 2000 16:49:49 -0600
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A few comments on the draft-ietf-sip-info-method-01.txt:

[1. Introduction]  The introduction says the INFO method merely sends
"optional application layer information".  But some of the uses of the INFO
method (carrying PSTN signaling messages, e.g.) are not really optional.
Maybe optional in the sense of the session state, but not optional in the
sense of getting things to work.

[2.2 Responses to the INFO Request Method]  A 200 OK message must be sent
for an INFO request with no body.  A 200 OK message must be sent if the body
is understood, but there are no rules for processing it.  I guess if the
body is not understood at all (I'm not sure how that differs from the
previous case), the action falls "outside the scope of this document".  (1)
I would like clarified what an "understood" body is, as opposed to a
non-understood body, and how the presence/absence of rules for handling the
body affects this.  Wouldn't a body with no rules defined be "not
understood"?  (2) Would it be correct to at least demand that the server
return some sort of final response on receipt of an INFO message, even if
the determination of the particular response sent is outside the scope of
the document?  The sender ought, I think, to be entitled to receive a final
response of some sort on any INFO method.  (3) Section 2 says the
mid-session information can be sent either as a message body or as message
headers.  If so, why does the response handling vary depending on which of
these ways is chosen?  (So, if the mid-session information is carried in
headers, the draft mandates a 200 OK response, but if the same information
is carried in a body, the response is "outside the scope of this document".)


[X. Reliability]  I think a short reliability section should be added,
similar to that in RFC2543.  Should 200 OK or other final responses be
retransmitted (I think not).  Should a client retransmit an INFO method if a
final response is not received (I think yes).

[2.2 Responses to the INFO Request Method]  (Not really a "comment", but a
"nit".)  The two "tables" are labeled "figures", though they are referred to
as "tables" in the text.  They are consistently called and labeled "tables"
in RFC2543.

Tim Schroeder




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Subject: Comments on "Supported:" header draft
To: jdrosen@dynamicsoft.com, schulzrinne@cs.columbia.edu
Date: Tue, 25 Jan 2000 18:46:45 -0600 (CST)
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From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Re: http://www.ietf.org/internet-drafts/draft-ietf-sip-serverfeatures-00.txt

I've given the "Supported:" header draft a once-over, and have the
following feedback to offer:

1) The paragraph at the bottom of page 4 suggests that:

  "If the protocol mechanisms followed here are operating
   correctly, these features will only be among the set supported by the
   UAC. If that is not the case, the client SHOULD resubmit the request
   as if it were a 421 response."

   This seems somewhat dangerous to me. If the protocol features
   are not operating correctly, you must have a broken implementation
   somewhere. Under those circumstances, the inadvertant introduction
   of an undetected loop may occur. I'd suggest that the appropriate
   behavior for the client in this error situation would be:
   
    - For 1xx, CANCEL the call and be prepared to send a BYE.
    
    - For 2xx, ACK the response and then send a BYE.

    - For 300+, ACK the response and carry on as if the unsupported
      headers listed in the "Require" header are not present.

2) Under 4.2, it makes sense that a server should
   examine the "Required" headers as well. If a feature
   appears in the "Required" headers (but not in the "Supported"
   headers), it can assume that the client supports it.

   This comment applies elsewhere as well (e.g. the overview).

3) The otherwise innocuous attempt to negotiate features using 
   421 has the strong possibilty of being destroyed by intervening
   forking proxies. If my request forks at a proxy to two UAS nodes,
   there is the risk that one will return a response code of 400
   through 420. Since these are numerically lower than 421, they
   will be forwarded upstream, blocking out what may very well be
   a viable call leg from a UAS who attempts to ask for a particular
   feature.

   The only alternative I can propose without some pretty massive
   changes would be to revise your 421 response code to be something
   like 299. This response, of course, would be sent only in response
   to a request containing a "Supported:" header, so there's no
   risk of the client interpreting a 299 response as a generic
   "okay." You can consider it something of a conditional success
   message.

   Since this is a 200 class message, the forking proxy must
   terminate the search and forward the response upstream. The
   client re-launches the INVITE with the appropriate Supported/
   Unsupported headers; the second UAS is then free to service
   the call. One unpleasant side effect is that other clients
   who received forks may ring, stop, and start again as the
   extensions are negotiated. *But*, at least it maximizes the
   chances of terminating a call successfully.  (Can anyone think 
   of a better end around these problems? We could use a 1xx 
   response code, but since the provisional reliable draft relies 
   upon this one, we have something of a chicken-and-egg problem...
   We could overload the meaning of "400," but that seems horrible.)

4) In the message examples, the final message on page 7 (from A to B)
   contains no "Unsupported" header. This is consistent with the text.

   This leads to the strong possibility of loops. It doesn't seem
   too much of a stretch that some faulty code in the server,
   not seeing "com.dynamicsoft.foo" in either a "Supported" or
   "Unsupported" header may decide to once again query the client
   for its supported features. That response may contain only
   the "Unsupported: com.dynamicsoft.foo," without the
   "Supported: com.dynamicsoft.bar." You see how messy this could
   get.

   An easy way to reduce the chance of this would be to say the the 
   client MUST remember all the requested features until a (non-421)
   final response is received and include them in "Supported" and
   "Unsupported" headers in any subsequent INVITE messages. This is 
   very easy to implement.

5) Open issue 1: 

            "The current specification does not support extensions which
             require a proxy to modify responses as they travel along.
             This can probably be added without too much complexity. Is
             this a useful function?"

   Since we cannot predict whether such behavior
   will be useful in the future, I'd propose that it be added
   (since the document points out that such a negotiation can
   be done without too much complexity). It would be a shame to
   get to RFC and only then realise that there's a really cool
   service that we could have implemented if only...

6) Open issue 2: 

	    "Proxy-Require is not used; it doesn't seem necessary. It
             doesn't matter whether the endpoint generating a response
             is a proxy or UAS. Is that OK?"

   This seems okay. However, it does raise an
   interesting issue: if the server requires a feature that also 
   places requirements on the intervening proxies, does it
   indicate this fact explicitly, or is it the client's
   responsibility to know that the feature requires the
   cooperation of proxies? As an example, if the server wants
   reliable provisional responses (for whatever reason),
   and sends back a "Required: org.ietf.sip.100rel" header,
   does the client figure out that its next request needs to
   contain a "Proxy-Required: org.ietf.sip.100rel" header?
   
   Also, what if the server merely wants to place requirements
   on the proxies instead of the client?

7) Open issue 3: 

	    "Do we need the Require header in the final non-421 response
             to indicate what features were finally applied?"

   Yes. Absolutely, yes. The client may exhibit different behavior
   based on the set of features the server finally decides upon.
   Especially, for example, when there is no 421 involved. 
   I may indicate support of a particular feature in my INVITE, but
   without indication from the server, I don't know whether to
   exhibit the behavior required for that extension. Including
   the feature in a 200 even after negotiation is consistent with 
   this behavior (which simplifies implementation).

8) Open issue 4: 

	    "Should we remove org.ietf.sip.supported and have the
             presence of the Supported header alone indicate support for
             this extension?"

   I don't feel strongly one way or the other; if we were voting on
   it, I'd vote for removing the "org.ietf.sip.supported" and having
   the presence of "Supported" be sufficient. My rationale is that
   doing so would make the messages shorter, which is especially
   appropriate when using the "k:" header variant. It seems wildly
   inconsistent to jump through hoops like reducing "To:" to "t:" to 
   save a single byte, but requiring one to add the whole string 
   "org.ietf.sip.supported" (22 bytes -- probably more than you'll 
   save in an entire message by using compact headers) for a full 
   featured client.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Subject: Re: Comments on "Supported:" header draft
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Thanks for your detailed comments!

A few remarks:

"Adam B. Roach" wrote:
> 
> Re: http://www.ietf.org/internet-drafts/draft-ietf-sip-serverfeatures-00.txt
> 
> I've given the "Supported:" header draft a once-over, and have the
> following feedback to offer:
> 
> 1) The paragraph at the bottom of page 4 suggests that:
> 
>   "If the protocol mechanisms followed here are operating
>    correctly, these features will only be among the set supported by the
>    UAC. If that is not the case, the client SHOULD resubmit the request
>    as if it were a 421 response."
> 
>    This seems somewhat dangerous to me. If the protocol features
>    are not operating correctly, you must have a broken implementation
>    somewhere. Under those circumstances, the inadvertant introduction
>    of an undetected loop may occur. I'd suggest that the appropriate
>    behavior for the client in this error situation would be:
> 
>     - For 1xx, CANCEL the call and be prepared to send a BYE.
> 
>     - For 2xx, ACK the response and then send a BYE.
> 
>     - For 300+, ACK the response and carry on as if the unsupported
>       headers listed in the "Require" header are not present.
> 

Seems reasonable. Making failure explicit rather than just hoping that
things will work out also will make it much more likely that bugs like
these will be caught sooner rather than later.

> 2) Under 4.2, it makes sense that a server should
>    examine the "Required" headers as well. If a feature
>    appears in the "Required" headers (but not in the "Supported"
>    headers), it can assume that the client supports it.
> 
>    This comment applies elsewhere as well (e.g. the overview).

Good idea. One would assume that features are symmetric, i.e., we don't
have "send-only" features where the client wants the server to
understand, say, a header, but has no clue what to do if it receives
that same header?

> 
> 3) The otherwise innocuous attempt to negotiate features using
>    421 has the strong possibilty of being destroyed by intervening
>    forking proxies. If my request forks at a proxy to two UAS nodes,
>    there is the risk that one will return a response code of 400
>    through 420. Since these are numerically lower than 421, they
>    will be forwarded upstream, blocking out what may very well be
>    a viable call leg from a UAS who attempts to ask for a particular
>    feature.
> 
>    The only alternative I can propose without some pretty massive
>    changes would be to revise your 421 response code to be something
>    like 299. This response, of course, would be sent only in response
>    to a request containing a "Supported:" header, so there's no
>    risk of the client interpreting a 299 response as a generic
>    "okay." You can consider it something of a conditional success
>    message.
> 

Interesting. This seems apt if the server is prepared to fall back to
not using the feature. Kind of "I'd like to use this feature if possible
- you'll like it, but if you are dumb, the phone call must go through
and I'll take it if I have to". 


>    Since this is a 200 class message, the forking proxy must
>    terminate the search and forward the response upstream. The
>    client re-launches the INVITE with the appropriate Supported/
>    Unsupported headers; the second UAS is then free to service
>    the call. One unpleasant side effect is that other clients
>    who received forks may ring, stop, and start again as the
>    extensions are negotiated. *But*, at least it maximizes the
>    chances of terminating a call successfully.  (Can anyone think
>    of a better end around these problems? We could use a 1xx
>    response code, but since the provisional reliable draft relies
>    upon this one, we have something of a chicken-and-egg problem...
>    We could overload the meaning of "400," but that seems horrible.)
> 
> 4) In the message examples, the final message on page 7 (from A to B)
>    contains no "Unsupported" header. This is consistent with the text.
> 
>    This leads to the strong possibility of loops. It doesn't seem
>    too much of a stretch that some faulty code in the server,
>    not seeing "com.dynamicsoft.foo" in either a "Supported" or
>    "Unsupported" header may decide to once again query the client
>    for its supported features. That response may contain only
>    the "Unsupported: com.dynamicsoft.foo," without the
>    "Supported: com.dynamicsoft.bar." You see how messy this could
>    get.
> 
>    An easy way to reduce the chance of this would be to say the the
>    client MUST remember all the requested features until a (non-421)
>    final response is received and include them in "Supported" and
>    "Unsupported" headers in any subsequent INVITE messages. This is
>    very easy to implement.
> 
> 5) Open issue 1:
> 
>             "The current specification does not support extensions which
>              require a proxy to modify responses as they travel along.
>              This can probably be added without too much complexity. Is
>              this a useful function?"
> 
>    Since we cannot predict whether such behavior
>    will be useful in the future, I'd propose that it be added
>    (since the document points out that such a negotiation can
>    be done without too much complexity). It would be a shame to
>    get to RFC and only then realise that there's a really cool
>    service that we could have implemented if only...
> 
> 6) Open issue 2:
> 
>             "Proxy-Require is not used; it doesn't seem necessary. It
>              doesn't matter whether the endpoint generating a response
>              is a proxy or UAS. Is that OK?"
> 
>    This seems okay. However, it does raise an
>    interesting issue: if the server requires a feature that also
>    places requirements on the intervening proxies, does it
>    indicate this fact explicitly, or is it the client's
>    responsibility to know that the feature requires the
>    cooperation of proxies? As an example, if the server wants
>    reliable provisional responses (for whatever reason),
>    and sends back a "Required: org.ietf.sip.100rel" header,
>    does the client figure out that its next request needs to
>    contain a "Proxy-Required: org.ietf.sip.100rel" header?
> 
>    Also, what if the server merely wants to place requirements
>    on the proxies instead of the client?
> 
> 7) Open issue 3:
> 
>             "Do we need the Require header in the final non-421 response
>              to indicate what features were finally applied?"
> 
>    Yes. Absolutely, yes. The client may exhibit different behavior
>    based on the set of features the server finally decides upon.
>    Especially, for example, when there is no 421 involved.
>    I may indicate support of a particular feature in my INVITE, but
>    without indication from the server, I don't know whether to
>    exhibit the behavior required for that extension. Including
>    the feature in a 200 even after negotiation is consistent with
>    this behavior (which simplifies implementation).

Agreed.

> 
> 8) Open issue 4:
> 
>             "Should we remove org.ietf.sip.supported and have the
>              presence of the Supported header alone indicate support for
>              this extension?"
> 
>    I don't feel strongly one way or the other; if we were voting on
>    it, I'd vote for removing the "org.ietf.sip.supported" and having
>    the presence of "Supported" be sufficient. My rationale is that
>    doing so would make the messages shorter, which is especially
>    appropriate when using the "k:" header variant. It seems wildly
>    inconsistent to jump through hoops like reducing "To:" to "t:" to
>    save a single byte, but requiring one to add the whole string
>    "org.ietf.sip.supported" (22 bytes -- probably more than you'll
>    save in an entire message by using compact headers) for a full
>    featured client.

I made roughly the same argument...


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Subject: Re: Comments on "Supported:" header draft
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
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In-Reply-To: <388E50A9.D024291@cs.columbia.edu> from "Henning Schulzrinne" at Jan 25, 2000 08:40:57 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Henning Schulzrinne writes:
>"Adam B. Roach" wrote:
>>    The only alternative I can propose without some pretty massive
>>    changes would be to revise your 421 response code to be something
>>    like 299. This response, of course, would be sent only in response
>>    to a request containing a "Supported:" header, so there's no
>>    risk of the client interpreting a 299 response as a generic
>>    "okay." You can consider it something of a conditional success
>>    message.
>> 
>
>Interesting. This seems apt if the server is prepared to fall back to
>not using the feature. Kind of "I'd like to use this feature if possible
>- you'll like it, but if you are dumb, the phone call must go through
>and I'll take it if I have to". 

If something goes awry, yes... however, this won't be the normal
case. A server wouldn't be sending a 299 unless a Supported (or k)
header were present in the response. The presence of such a header
would necessarily indicate that the client understands the semantics
of the 299 message -- so a 299 arriving at a client who doesn't
know that it requires another INVITE indicates a faulty server
implementation (or a client who bizarrely sent a Supported header
without knowing what it meant).

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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From: "Donovan, Steven R." <Steven.R.Donovan@wcom.com>
Subject: RE: comments on SIP INFO last call
To: "'Schroeder, Tim'" <schroeder@tri.sbc.com>,
        sip@lists.research.bell-labs.com
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Tim,

Thanks for the thorough review of the draft.  Please see my comments below
preceded by SRD>.

Regards,

Steve

> -----Original Message-----
> From: Schroeder, Tim [mailto:schroeder@tri.sbc.com]
> Sent: Tuesday, January 25, 2000 4:50 PM
> To: sip@lists.research.bell-labs.com
> Subject: comments on SIP INFO last call
> 
> 
> A few comments on the draft-ietf-sip-info-method-01.txt:
> 
> [1. Introduction]  The introduction says the INFO method merely sends
> "optional application layer information".  But some of the 
> uses of the INFO
> method (carrying PSTN signaling messages, e.g.) are not 
> really optional.
> Maybe optional in the sense of the session state, but not 
> optional in the
> sense of getting things to work.

SRD> It is true that INFO is required for SIP to PSTN interworking.
However, the definition of this interworking is outside the scope of the
INFO specification.  There will need be a separate draft that defines the
required behavior of entities impacted by interworking with the PSTN.  Just
as there will need to be a separate specification that documents any other
use of the INFO method.

> 
> [2.2 Responses to the INFO Request Method]  A 200 OK message 
> must be sent
> for an INFO request with no body.  A 200 OK message must be 
> sent if the body
> is understood, but there are no rules for processing it.  I 
> guess if the
> body is not understood at all (I'm not sure how that differs from the
> previous case), the action falls "outside the scope of this 
> document".  (1)
> I would like clarified what an "understood" body is, as opposed to a
> non-understood body, and how the presence/absence of rules 
> for handling the
> body affects this.  Wouldn't a body with no rules defined be "not
> understood"?  (2) Would it be correct to at least demand that 
> the server
> return some sort of final response on receipt of an INFO 
> message, even if
> the determination of the particular response sent is outside 
> the scope of
> the document?  The sender ought, I think, to be entitled to 
> receive a final
> response of some sort on any INFO method.  (3) Section 2 says the
> mid-session information can be sent either as a message body 
> or as message
> headers.  If so, why does the response handling vary 
> depending on which of
> these ways is chosen?  (So, if the mid-session information is 
> carried in
> headers, the draft mandates a 200 OK response, but if the 
> same information
> is carried in a body, the response is "outside the scope of 
> this document".)

SRD> This section was attempting to define the default behavior of 
a UAS upon receipt of the INFO message.  I agree that it is a little 
unclear in the case that a message body is received that it does not
recognize.  As you correctly state, this is the similar to receiving an
INFO message with no message body, although I don't think it is 
quite the same.  It seems that the UAC should be able to known that 
the UAS does not understand the message body sent.  

SRD> Clearly a final response is needed in all cases.  The default 
behavour is to send a 200 OK if the INFO cooresponds to an existing
session at the UAS and there is no message body or the message body
can be processed by the UAS.  In the presence of an message body 
that is not understood the default behavour should be for the UAS to
send a 415 Unsupported Media Type, which is realy saying Unsupported
Message Body.  Otherwise a 481 response is sent.  Any other 
rules as to handling of the INFO message are going to be application
specific.

SRD> I propose changing the first part of section 2.2 to the folloiwng:

"If a server receives an INFO request it MUST send a final
response.  

A 200 OK response MUST be sent by a UAS for an INFO request with
no message body if the INFO request was successfully received for
an existing call.  Beyond that, no additional operations are
required.

Handling of INFO messages that contain message bodies is outside
the scope of this document.  The documents defining the message
bodies will also need to define the SIP protocol rules associated
with those message bodies.

A 481 Call Leg/Transaction Does Not Exist message MUST be sent by 
a UAS if the INFO request does not match any existing call leg.

If a server receives an INFO request with a body it understands,
but it has no knowledge of INFO associated processing rules for
the body, the body MAY be rendered and displayed to the user. The
INFO is responded to with a 200 OK.

If the INFO request contains a body that the server does understand 
then, in the absense of INFO associated processing rules for the 
body, the server MUST respond with a 415 Unsupported Media Type
message."

> 
> 
> [X. Reliability]  I think a short reliability section should be added,
> similar to that in RFC2543.  Should 200 OK or other final responses be
> retransmitted (I think not).  Should a client retransmit an 
> INFO method if a
> final response is not received (I think yes).

SRD> This is addressed in the following from section 2.4:

"Unless stated otherwise, the protocol rules for the INFO
request governing the usage of tags, Route and Record-Route,
retransmission and reliability, CSeq incrementing and message
formatting follow those in [1] as defined for the BYE request."

> 
> [2.2 Responses to the INFO Request Method]  (Not really a 
> "comment", but a
> "nit".)  The two "tables" are labeled "figures", though they 
> are referred to
> as "tables" in the text.  They are consistently called and 
> labeled "tables"
> in RFC2543.

SRD> Oops.  

> 
> Tim Schroeder
> 
> 



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From: Igor Slepchin <islepchin@dynamicsoft.com>
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        sip@lists.research.bell-labs.com
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"Adam B. Roach" wrote:
> 
> 1) The paragraph at the bottom of page 4 suggests that:
>    I'd suggest that the appropriate
>    behavior for the client in this error situation would be:
> 
>     - For 1xx, CANCEL the call and be prepared to send a BYE.
> 
>     - For 2xx, ACK the response and then send a BYE.
> 
>     - For 300+, ACK the response and carry on as if the unsupported
>       headers listed in the "Require" header are not present.

It's theoretically possible for somebody to define an extension, say
com.foo.300is100 that completely redefines the meaning of response codes
and the default behavior may not be expected by the server when that
extension is used. However, I don't think we could do any better than
trying the above mentioned actions. After all, even the meaning of BYE
may be redefined.


> 4) In the message examples, the final message on page 7 (from A to B)
>    contains no "Unsupported" header. This is consistent with the text.
> 
>    This leads to the strong possibility of loops. It doesn't seem
>    too much of a stretch that some faulty code in the server,
>    not seeing "com.dynamicsoft.foo" in either a "Supported" or
>    "Unsupported" header may decide to once again query the client
>    for its supported features. That response may contain only
>    the "Unsupported: com.dynamicsoft.foo," without the
>    "Supported: com.dynamicsoft.bar." You see how messy this could
>    get.
>    An easy way to reduce the chance of this would be to say the the
>    client MUST remember all the requested features until a (non-421)
>    final response is received and include them in "Supported" and
>    "Unsupported" headers in any subsequent INVITE messages. This is
>    very easy to implement.

Wouldn't it be better to require the server to specify _all_ extensions
it may want to apply to the response in the Require header of the first
421 (in the above mentioned example, it would list both
com.dynamicsoft.foo and com.dynamicsoft.bar)? The client will then list
those it supports/does not support in the resubmitted request and the
server will be able to determine which ones it will actually use. It
will also save 1.5 round trip in the example.

---
Igor Slepchin



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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Igor Slepchin wrote:
> 
> "Adam B. Roach" wrote:
> >
> > 1) The paragraph at the bottom of page 4 suggests that:
> >    I'd suggest that the appropriate
> >    behavior for the client in this error situation would be:
> >
> >     - For 1xx, CANCEL the call and be prepared to send a BYE.
> >
> >     - For 2xx, ACK the response and then send a BYE.
> >
> >     - For 300+, ACK the response and carry on as if the unsupported
> >       headers listed in the "Require" header are not present.
> 
> It's theoretically possible for somebody to define an extension, say
> com.foo.300is100 that completely redefines the meaning of response codes
> and the default behavior may not be expected by the server when that
> extension is used. However, I don't think we could do any better than
> trying the above mentioned actions. After all, even the meaning of BYE
> may be redefined.

Anything we can do to *dis*courage people redefining existing behavior
is probably a good idea, as this would seem to be the sure road to
interoperability hell.

> 
> > 4) In the message examples, the final message on page 7 (from A to B)
> >    contains no "Unsupported" header. This is consistent with the text.
> >
> >    This leads to the strong possibility of loops. It doesn't seem
> >    too much of a stretch that some faulty code in the server,
> >    not seeing "com.dynamicsoft.foo" in either a "Supported" or
> >    "Unsupported" header may decide to once again query the client
> >    for its supported features. That response may contain only
> >    the "Unsupported: com.dynamicsoft.foo," without the
> >    "Supported: com.dynamicsoft.bar." You see how messy this could
> >    get.
> >    An easy way to reduce the chance of this would be to say the the
> >    client MUST remember all the requested features until a (non-421)
> >    final response is received and include them in "Supported" and
> >    "Unsupported" headers in any subsequent INVITE messages. This is
> >    very easy to implement.
> 
> Wouldn't it be better to require the server to specify _all_ extensions
> it may want to apply to the response in the Require header of the first
> 421 (in the above mentioned example, it would list both
> com.dynamicsoft.foo and com.dynamicsoft.bar)? The client will then list
> those it supports/does not support in the resubmitted request and the
> server will be able to determine which ones it will actually use. It
> will also save 1.5 round trip in the example.

Given that the number of extensions is probably not dozens, I would
agree that a single indication is far simpler to manage than having them
be cumulative in some way. This goes along with the desire to make
things as stateless as possible and each message as self-contained as
possible.


> 
> ---
> Igor Slepchin

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Subject: Re: Comments on "Supported:" header draft
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
Date: Wed, 26 Jan 2000 15:04:05 -0600 (CST)
Cc: islepchin@dynamicsoft.com (Igor Slepchin), jdrosen@dynamicsoft.com,
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In-Reply-To: <388F5E90.BE73D910@cs.columbia.edu> from "Henning Schulzrinne" at Jan 26, 2000 03:52:32 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Henning Schulzrinne writes:
>Igor Slepchin wrote:
>> Wouldn't it be better to require the server to specify _all_ extensions
>> it may want to apply to the response in the Require header of the first
>> 421 (in the above mentioned example, it would list both
>> com.dynamicsoft.foo and com.dynamicsoft.bar)? The client will then list
>> those it supports/does not support in the resubmitted request and the
>> server will be able to determine which ones it will actually use. It
>> will also save 1.5 round trip in the example.
>
>Given that the number of extensions is probably not dozens, I would
>agree that a single indication is far simpler to manage than having them
>be cumulative in some way. This goes along with the desire to make
>things as stateless as possible and each message as self-contained as
>possible.

Yes; I like this approach better as well. Since there seems to be
support for the point of view that the server sends back the finally
negotiated set of extensions in the final response, it can do its own
internal decision making at that time. I'll summarize what I understand
your proposal to be:

If a server wants to use com.dynamicsoft.foo, but com.dynamicsoft.bar 
would be an acceptable substitute (as in the draft), it would send 
back the 421 (or whatever we finalize) with a Require field specifying 
both. If the client indicates support for com.dynamicsoft.foo *and* 
com.dynamicsoft.bar, the server will include only com.dynamicsoft.foo 
in its final reply "Require" (thus indicating that com.dynamicsoft.bar
is not to be used).

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Subject: RE: Comments on "Supported:" header draft
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Igor Slepchin wrote:
> Wouldn't it be better to require the server to specify _all_ extensions
> it may want to apply to the response in the Require header of the first
> 421 (in the above mentioned example, it would list both
> com.dynamicsoft.foo and com.dynamicsoft.bar)? 

Henning Schulzrinne writes:
> Given that the number of extensions is probably not dozens, I would
> agree that a single indication is far simpler to manage than having them
> be cumulative in some way.

Adam B. Roach writes:
> Yes; I like this approach better as well. Since there seems to be
> support for the point of view that the server sends back the finally
> negotiated set of extensions in the final response, it can do its own
> internal decision making at that time.

I like this better as well.

I believe the server *must* send back the finally negotiated set of
extensions in the final response.  Otherwise, if a client sends an initial
request (not in response to an earlier 421) that contains "Support:
com.dynamicsoft.foo, com.dynamicsoft.bar" (because those are two commonly
requested features), it would have no way of knowing which of the two
possibly incompatible features to use to interpret the response, or even
whether either one of the two features is to be used.

Tim Schroeder



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First off, thanks for your comments, Adam.

"Adam B. Roach" wrote:
> 
> 1) The paragraph at the bottom of page 4 suggests that:
> 
>   "If the protocol mechanisms followed here are operating
>    correctly, these features will only be among the set supported by the
>    UAC. If that is not the case, the client SHOULD resubmit the request
>    as if it were a 421 response."
> 
>    This seems somewhat dangerous to me. If the protocol features
>    are not operating correctly, you must have a broken implementation
>    somewhere. Under those circumstances, the inadvertant introduction
>    of an undetected loop may occur. I'd suggest that the appropriate
>    behavior for the client in this error situation would be:
> 
>     - For 1xx, CANCEL the call and be prepared to send a BYE.
> 
>     - For 2xx, ACK the response and then send a BYE.
> 
>     - For 300+, ACK the response and carry on as if the unsupported
>       headers listed in the "Require" header are not present.

Whilst I agree with Igor that its possible that the feature overrides
the normal behavior of 300+, I agree with Henning this should be
discouraged. Since there is not much else you can do, I agree with Adams
approach.


> 
> 2) Under 4.2, it makes sense that a server should
>    examine the "Required" headers as well. If a feature
>    appears in the "Required" headers (but not in the "Supported"
>    headers), it can assume that the client supports it.
> 
>    This comment applies elsewhere as well (e.g. the overview).

I agree with others that this is a good idea.

> 
> 3) The otherwise innocuous attempt to negotiate features using
>    421 has the strong possibilty of being destroyed by intervening
>    forking proxies. If my request forks at a proxy to two UAS nodes,
>    there is the risk that one will return a response code of 400
>    through 420. Since these are numerically lower than 421, they
>    will be forwarded upstream, blocking out what may very well be
>    a viable call leg from a UAS who attempts to ask for a particular
>    feature.
> 
>    The only alternative I can propose without some pretty massive
>    changes would be to revise your 421 response code to be something
>    like 299. This response, of course, would be sent only in response
>    to a request containing a "Supported:" header, so there's no
>    risk of the client interpreting a 299 response as a generic
>    "okay." You can consider it something of a conditional success
>    message.
> 
>    Since this is a 200 class message, the forking proxy must
>    terminate the search and forward the response upstream. The
>    client re-launches the INVITE with the appropriate Supported/
>    Unsupported headers; the second UAS is then free to service
>    the call. One unpleasant side effect is that other clients
>    who received forks may ring, stop, and start again as the
>    extensions are negotiated. *But*, at least it maximizes the
>    chances of terminating a call successfully.  (Can anyone think
>    of a better end around these problems? We could use a 1xx
>    response code, but since the provisional reliable draft relies
>    upon this one, we have something of a chicken-and-egg problem...
>    We could overload the meaning of "400," but that seems horrible.)


I don't like this approach too much. The problem is that a 2xx response
to INVITE means the call is setup. Even though it should only be sent
when the client indicates it knows that 299 means otherwise, proxies
won't necessarily know. Thus, an intermediate proxy (say running a CPL)
which provides some kind of call forwarding service may get mightily
confused. Also, the 299 may begin billing in intermediate proxies which
run billing systems for gateways. The brief ring and cancellation is
also going to be annoying.

I'm not certain about the right solution. It requires some more thought.
Its broader than just 421; it applies to any response code which can
cause a request to be resubmitted with additional info (401 and 407 are
other examples).

> 
> 4) In the message examples, the final message on page 7 (from A to B)
>    contains no "Unsupported" header. This is consistent with the text.
> 
>    This leads to the strong possibility of loops. It doesn't seem
>    too much of a stretch that some faulty code in the server,
>    not seeing "com.dynamicsoft.foo" in either a "Supported" or
>    "Unsupported" header may decide to once again query the client
>    for its supported features. That response may contain only
>    the "Unsupported: com.dynamicsoft.foo," without the
>    "Supported: com.dynamicsoft.bar." You see how messy this could
>    get.
> 
>    An easy way to reduce the chance of this would be to say the the
>    client MUST remember all the requested features until a (non-421)
>    final response is received and include them in "Supported" and
>    "Unsupported" headers in any subsequent INVITE messages. This is
>    very easy to implement.


Igor later writes:

> Wouldn't it be better to require the server to specify _all_ extensions
> it may want to apply to the response in the Require header of the first
> 421 (in the above mentioned example, it would list both
> com.dynamicsoft.foo and com.dynamicsoft.bar)? The client will then list
> those it supports/does not support in the resubmitted request and the
> server will be able to determine which ones it will actually use. It
> will also save 1.5 round trip in the example.

to which Henning responded:

> Given that the number of extensions is probably not dozens, I would
> agree that a single indication is far simpler to manage than having them
> be cumulative in some way. This goes along with the desire to make
> things as stateless as possible and each message as self-contained as
> possible.


The problem I see here is that:

  1. the server may not know it might need to try bar until after it has
already asked for foo. The decisions about what features to apply may be
the result of programmatic execution after the resubmitted response.
Clearly, when it does know, what Igor suggested is fine (and probably
should be encouraged), but I don't want to mandate it. In that case, I
agree with Adam that the feature lists should be accumulated. We will be
doing this already for credentials, as described in Robert's multi-proxy
authorization draft.

  2. At this rate, we may well have dozens of extensions to SIP :(


> 
> 5) Open issue 1:
> 
>             "The current specification does not support extensions which
>              require a proxy to modify responses as they travel along.
>              This can probably be added without too much complexity. Is
>              this a useful function?"
> 
>    Since we cannot predict whether such behavior
>    will be useful in the future, I'd propose that it be added
>    (since the document points out that such a negotiation can
>    be done without too much complexity). It would be a shame to
>    get to RFC and only then realise that there's a really cool
>    service that we could have implemented if only...

Actually, I would argue the opposite for the same reason. Lets design
extensions that meet specific needs. Until someone comes up with a need,
I'd rather not build it into the protocol.

> 
> 6) Open issue 2:
> 
>             "Proxy-Require is not used; it doesn't seem necessary. It
>              doesn't matter whether the endpoint generating a response
>              is a proxy or UAS. Is that OK?"
> 
>    This seems okay. However, it does raise an
>    interesting issue: if the server requires a feature that also
>    places requirements on the intervening proxies, does it
>    indicate this fact explicitly, or is it the client's
>    responsibility to know that the feature requires the
>    cooperation of proxies? As an example, if the server wants
>    reliable provisional responses (for whatever reason),
>    and sends back a "Required: org.ietf.sip.100rel" header,
>    does the client figure out that its next request needs to
>    contain a "Proxy-Required: org.ietf.sip.100rel" header?

Yes. If the feature is supported, the client should know that if the
feature requires a request to be resubmitted, and this request needs
proxy support, the proxy-require header is needed. This is a bad
example, though, as the new 100 reliability draft doesn't actually
require proxies to understand it.  

> 
>    Also, what if the server merely wants to place requirements
>    on the proxies instead of the client?

Hmm. No way the mechanism supports this easily. Only clients can
resubmit requests. One way to do this would be to define a "stack"
header like Record-Route, and have each proxy push the set of features
it understands in the initial request. Seems overkill though, and can
lead to some very big messages. Might be other, more complex ways to do
this. 

Maybe its useful to list the various combinations of who wants who to do
what:

1. client needs proxies to understand feature to process request
2. client needs UAS to understand feature to process request
3. proxy needs downstream proxy to understand feature to process request
4. proxy needs UAS to understand feature to process request
5. UAS needs proxies to understand feature to process response
6. UAS needs UAC to understand feature to process response
7. proxy needs upstream proxies to understand feature to process
response
8. proxy needs UAC to understand feature to process response

I think that is every possible combination. (1) and (2) are covered by
require and proxy-require in the base spec. (6) and (8) are meant to be
covered by this draft. Can we identify needs for the others? 

> 
> 7) Open issue 3:
> 
>             "Do we need the Require header in the final non-421 response
>              to indicate what features were finally applied?"
> 
>    Yes. Absolutely, yes. The client may exhibit different behavior
>    based on the set of features the server finally decides upon.
>    Especially, for example, when there is no 421 involved.
>    I may indicate support of a particular feature in my INVITE, but
>    without indication from the server, I don't know whether to
>    exhibit the behavior required for that extension. Including
>    the feature in a 200 even after negotiation is consistent with
>    this behavior (which simplifies implementation).

Agreed. 

> 
> 8) Open issue 4:
> 
>             "Should we remove org.ietf.sip.supported and have the
>              presence of the Supported header alone indicate support for
>              this extension?"
> 
>    I don't feel strongly one way or the other; if we were voting on
>    it, I'd vote for removing the "org.ietf.sip.supported" and having
>    the presence of "Supported" be sufficient. My rationale is that
>    doing so would make the messages shorter, which is especially
>    appropriate when using the "k:" header variant. It seems wildly
>    inconsistent to jump through hoops like reducing "To:" to "t:" to
>    save a single byte, but requiring one to add the whole string
>    "org.ietf.sip.supported" (22 bytes -- probably more than you'll
>    save in an entire message by using compact headers) for a full
>    featured client.

I am leaning towards this also; seems Henning agrees also.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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"Schroeder, Tim" wrote:
> 
> I believe the server *must* send back the finally negotiated set of
> extensions in the final response.  Otherwise, if a client sends an initial
> request (not in response to an earlier 421) that contains "Support:
> com.dynamicsoft.foo, com.dynamicsoft.bar" (because those are two commonly
> requested features), it would have no way of knowing which of the two
> possibly incompatible features to use to interpret the response, or even
> whether either one of the two features is to be used.

I think we have consensus on this open issue. Same with presence of
"org.ietf.sip.serverfeatures" in requests (consensus is to boot it). If
you disagree, say so now....

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Adam B. Roach wrote:
> 1) The paragraph at the bottom of page 4 suggests that:
> 
>   "If the protocol mechanisms followed here are operating
>    correctly, these features will only be among the set supported by the
>    UAC. If that is not the case, the client SHOULD resubmit the request
>    as if it were a 421 response."
> 
>    This seems somewhat dangerous to me. If the protocol features
>    are not operating correctly, you must have a broken implementation
>    somewhere. Under those circumstances, the inadvertant introduction
>    of an undetected loop may occur. I'd suggest that the appropriate
>    behavior for the client in this error situation would be:
> 
>     - For 1xx, CANCEL the call and be prepared to send a BYE.
>     - For 2xx, ACK the response and then send a BYE.
>     - For 300+, ACK the response and carry on as if the unsupported
>       headers listed in the "Require" header are not present.

I agree with making the failure more explicit.  If the server does this (and
hence makes the protocol features not operate correctly), it's best to get
out of the session immediately.  I'm not sure about the above approach for
300+ responses though; what exactly does "carry on" mean?  Maybe they should
all be treated as 500 "Server Internal Error".

If all 421 responses are to list the full set of features that might be
used(which I agree is a good idea), these rules would apply to any
subsequent 421 responses, without the need to compare lists or check for
subsets.

> 3) The otherwise innocuous attempt to negotiate features using
>    421 has the strong possibilty of being destroyed by intervening
>    forking proxies.
> 
>    The only alternative I can propose without some pretty massive
>    changes would be to revise your 421 response code to be something
>    like 299. 

Hmm.  How do things work with the 401 Unauthorized response across forking
proxies?  This case seems analogous.  One UAS responds with either 401 (try
again with appropriate authorization) or 421 (try again with the interesting
feature in the Supported header), while another UAS responds with a 400 Bad
Request.  Granted there are a lot more potential responses 400 <= N < 421
than there are 400 <= N < 401.  

One could imagine cases involving 407 Proxy Authentication Required or 402
Payment Required as well; these are basically "try again, but better"
responses that could be blocked by lower numbered but more fatal (404 Not
Found, e.g.) responses.

> 4) In the message examples, the final message on page 7 (from A to B)
>    contains no "Unsupported" header. This is consistent with the text.
> 
>    This leads to the strong possibility of loops. It doesn't seem
>    too much of a stretch that some faulty code in the server,
>    not seeing "com.dynamicsoft.foo" in either a "Supported" or
>    "Unsupported" header may decide to once again query the client
>    for its supported features. That response may contain only
>    the "Unsupported: com.dynamicsoft.foo," without the
>    "Supported: com.dynamicsoft.bar." You see how messy this could
>    get.

If the server must send all the possible features it cares about in a single
421 response, then this problem goes away, right?  Section 4.1 says the
client MUST explicitly put each feature from the 421 into either the
Supported or Unsupported header.  
 
> 5) Open issue 1:
>             "The current specification does not support extensions which
>              require a proxy to modify responses as they travel along.
>              This can probably be added without too much complexity. Is
>              this a useful function?"

Maybe it could be added without too much complexity, but I'm not convinced.
If a proxy is modifying responses, it's processing them.  Then I'm thinking
(off the top of my head) that we need to be able to determine where the
features are required or supported -- in a proxy, or at the user agent?
Wouldn't we end up needing something like Required and Proxy-Required to
distinguish?  Supported and Proxy-Supported, Unsupported and
Proxy-Unsupported?

This could tie in to open issue 2 as well.  Suppose a server wants a feature
from a proxy, and the proxy can support it, but that the client cannot.  The
client will end up responding with an Unsupported header, which will convey
the wrong information to the server.  Alternatively, the proxy could change
the header to Supported, but to do so would have to know that the server was
asking for a proxy feature, not a client feature.

> 7) Open issue 3:
>             "Do we need the Require header in the final non-421 response
>              to indicate what features were finally applied?"

Definitely.  If the client sends Supported: A,B because it supports both,
the server needs to send a response indicating which one was chosen.
Without the Require header, the client would have to assume that neither was
used.

> 8) Open issue 4:
>             "Should we remove org.ietf.sip.supported and have the
>              presence of the Supported header alone indicate support for
>              this extension?"

I like the org.ietf.sip.supported.  It seems odd that the presence of
Supported means something that the presence of Unsupported does not.  Or
would Unsupported in a request also imply org.ietf.sip.supported?  Since
Unsupported already exists in responses, the presence of Unsupported there
could not be used to imply anything.  The fewer "hidden" implications the
better, I think.

Tim Schroeder



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I have established a web site for tracking working group activities. I hope
to use this to document and coordinate our work. It currently has several
sections of information on:

* Meetings, including agendas, presentations, notes and published minutes.
We'll use this area for publishing and polishing future agendas.

* Drafts repository, hopefully with more persistence than the IETF drafts
library. We're also willing to host "rich" drafts (like PS or PDFs) here.

* References, pointers to other work. I need help here . . . that, or I just
point at Henning's page like I did already. He's doing such a great job.

* Tasks, discussion of work efforts both chartered and not. I plan to link
to at least brief discussions of each effort (or to the design team
responsible for furthering the effort).

* Design Teams, subgroups working on chartered tasks. We want to have design
team for each chartered task. I'd like to see each design team have a
mini-home-page, complete with facilitators, rosters, reports, and references
to work.

As usual, this is a work in progress, and will of course develop over time.
Guidance welcome.

--
Dean Willis
SIP WG co-chair.




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Subject: Re: Comments on "Supported:" header draft
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Wed, 26 Jan 2000 17:51:51 -0600 (CST)
Cc: schulzrinne@cs.columbia.edu, sip@lists.research.bell-labs.com
In-Reply-To: <388F777E.725ED206@dynamicsoft.com> from "Jonathan Rosenberg" at Jan 26, 2000 05:38:54 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Jonathan Rosenberg writes:
>>    Also, what if the server merely wants to place requirements
>>    on the proxies instead of the client?
>
>Hmm. No way the mechanism supports this easily. Only clients can
>resubmit requests. One way to do this would be to define a "stack"
>header like Record-Route, and have each proxy push the set of features
>it understands in the initial request. Seems overkill though, and can
>lead to some very big messages. Might be other, more complex ways to do
>this. 

I beleive it can be acheived through very much the same mechanism
as is already defined in the draft.

For example, if I don't see a "Proxy-Require" header in the request
(but see a "Supported" header), I can send back a 421 with a
"Proxy-Require" header in it. The client then (regardless of whether 
it knows what the feature is) re-sends the INVITE with the Proxy-Require 
header mirrored in it.  If it gets a 420 back, it re-sends the request 
with a (new) "Proxy-Unsupported" header containing all of the features
indicated in the "Unsupported" header it got from the 420.  The server 
can use the "Proxy-Unsupported" header to determine which features the 
chain of proxies does not understand.  This adds no new requirements on 
proxies.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Date: Wed, 26 Jan 2000 17:56:54 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Announcement of WG Supplemental Home Page
In-reply-to: <200001262352.RAA24932@b04a24.exu.ericsson.se>
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Ahah, you asker of tricky questions:

The supplemental page is at

http://www.softarmor.com/sipwg/

and is also linked from the official home page at

http://www.ietf.org/html.charters/sip-charter.html

--
Dean, the Befuddled

> -----Original Message-----
> From: Adam B. Roach [mailto:Adam.Roach@Ericsson.com]
> Sent: Wednesday, January 26, 2000 5:53 PM
> To: Dean Willis
> Subject: Re: Announcement of WG Supplemental Home Page
> 
> 
> >I have established a web site for tracking working group activities. 
> 
> Where?
> 
> -- 
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. 
> Arapaho, MS L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 
> 75081 USA
> 



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From owner-sip-outgoing@lists.research.bell-labs.com  Thu Jan 27 01:05:50 2000
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Hi Adam,
I had read this section, and it confuses me a bit.
If this section mentions a MUST then its mandatory that _all_ applications
must append a CRLF only.
Then what is the use for parsing incoming messages for CR, LF or CRLF ?
I guess it would make sense only if there is a possibility that some
application may send only a CR or an LF.
Then the receiver can be lenient on this input. (ie if 6.6 was a SHOULD)

However, if  6.6 states it as a MUST, which implies that a person who sends
CR orLF only is not SIP compliant.
then where is the question of the receiver expecting anything else - and if
it does, it should not accept it as the sender
violated a mandatory clause of the protocol.

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems





"Adam B. Roach" <Adam.Roach@Ericsson.com> on 01/25/2000 08:30:50 PM

To:   archow
cc:   Aparna.Vemuri@Level3.com, islepchin@dynamicsoft.com, K Binu/HSSBLR,
      sip@lists.research.bell-labs.com

Subject:  Re: Content-Length header and Multipart MIME




archow@hss.hns.com writes:
>Aparna> Rules used to calculate the Content-Length:
>Aparna> b.1 Use 2 octets to encode the CRLF sequence at the end of each
>line
>Aparna> and the beginning of a blank line.
>
>Does in necessarily have to be two octets ? the sip grammar defines the
>separator as
>CR | LF | CRLF depending on the target system's interpretation of a new
>line.
>
>So isnt't it that whether 1 or 2 octets for end-of-line is taken for C-Len
>should vary
>depending on what is the line separator one is using ?

The "CR | LF | CRLF" is for parsing of messages. In RFC2543, sec 6.6,
it says that "applications MUST follow HTTP common form when generating
[headers]." This denotes a full CRLF at the end of each line.

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA








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"Adam B. Roach" wrote:
> 
> I beleive it can be acheived through very much the same mechanism
> as is already defined in the draft.
> 
> For example, if I don't see a "Proxy-Require" header in the request
> (but see a "Supported" header), I can send back a 421 with a
> "Proxy-Require" header in it. The client then (regardless of whether
> it knows what the feature is) re-sends the INVITE with the Proxy-Require
> header mirrored in it.  If it gets a 420 back, it re-sends the request
> with a (new) "Proxy-Unsupported" header containing all of the features
> indicated in the "Unsupported" header it got from the 420.  The server
> can use the "Proxy-Unsupported" header to determine which features the
> chain of proxies does not understand.  This adds no new requirements on
> proxies.

This will only get you the set of features not supported by the first
proxy on the path that doesn't support at least one of the features
listed in Proxy-Require. Thats because the request with Proxy-Require is
rejected by the first proxy not supporting one of the features. You
won't learn about features not supported from servers downstream of that
one.

To be done correctly, either a stack mechanism must be used, or some
kind of set operations which cause proxies to add and remove features
from some specific headers in requests as they traverse the proxy path.
I'd still like to understand concrete applications before diving into
this; especially because interoperability begins to be tough when a
feature requires all proxies along the response path to understand it.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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"Donovan, Steven R." wrote:
> 
> > [1. Introduction]  The introduction says the INFO method merely sends
> > "optional application layer information".  But some of the
> > uses of the INFO
> > method (carrying PSTN signaling messages, e.g.) are not
> > really optional.
> > Maybe optional in the sense of the session state, but not
> > optional in the
> > sense of getting things to work.
> 
> SRD> It is true that INFO is required for SIP to PSTN interworking.

Well, its not REQUIRED. We have SIP to PSTN interworking now and its
doing just fine without INFO. The INFO method is only needed when ISUP
transparency is desired. The point this statement is making is that you
cannot use INFO for features which are mandatory in order for the
session to stay up correctly. This is because INFO is an optional
extension; a UAS may terminate a call, set it up, and then receive INFO,
but not understand it. This cannot be a catastrophic call failure.
Perhaps this should be clarified.

> > [2.2 Responses to the INFO Request Method]  A 200 OK message
> > must be sent
> > for an INFO request with no body.  A 200 OK message must be
> > sent if the body
> > is understood, but there are no rules for processing it.  I
> > guess if the
> > body is not understood at all (I'm not sure how that differs from the
> > previous case), the action falls "outside the scope of this
> > document".  (1)
> > I would like clarified what an "understood" body is, as opposed to a
> > non-understood body, and how the presence/absence of rules
> > for handling the
> > body affects this.  Wouldn't a body with no rules defined be "not
> > understood"?  (2) Would it be correct to at least demand that
> > the server
> > return some sort of final response on receipt of an INFO
> > message, even if
> > the determination of the particular response sent is outside
> > the scope of
> > the document?  The sender ought, I think, to be entitled to
> > receive a final
> > response of some sort on any INFO method.  (3) Section 2 says the
> > mid-session information can be sent either as a message body
> > or as message
> > headers.  If so, why does the response handling vary
> > depending on which of
> > these ways is chosen?  (So, if the mid-session information is
> > carried in
> > headers, the draft mandates a 200 OK response, but if the
> > same information
> > is carried in a body, the response is "outside the scope of
> > this document".)
> 
> SRD> This section was attempting to define the default behavior of
> a UAS upon receipt of the INFO message.  I agree that it is a little
> unclear in the case that a message body is received that it does not
> recognize.  As you correctly state, this is the similar to receiving an
> INFO message with no message body, although I don't think it is
> quite the same.  It seems that the UAC should be able to known that
> the UAS does not understand the message body sent.

This particular tidbit was my suggestion, so the fault is mine in its
unclarity. What it means "when an INFO body is understood, but no
specific rules exist for processing", is that the UAS knows how to
process this body when present in other messages, but no specific
behaviors are known for INFO. An example is text/html. When received in
a redirect, a UA knows to display it. What does it mean when its in
INFO? Unless some spec says otherwise, display it also.



> 
> SRD> Clearly a final response is needed in all cases. 

I think the spec already basically says this...

 
> SRD> I propose changing the first part of section 2.2 to the folloiwng:
> 
> "If a server receives an INFO request it MUST send a final
> response.
> 
> A 200 OK response MUST be sent by a UAS for an INFO request with
> no message body if the INFO request was successfully received for
> an existing call.  Beyond that, no additional operations are
> required.
> 
> Handling of INFO messages that contain message bodies is outside
> the scope of this document.  The documents defining the message
> bodies will also need to define the SIP protocol rules associated
> with those message bodies.
> 
> A 481 Call Leg/Transaction Does Not Exist message MUST be sent by
> a UAS if the INFO request does not match any existing call leg.
> 
> If a server receives an INFO request with a body it understands,
> but it has no knowledge of INFO associated processing rules for
> the body, the body MAY be rendered and displayed to the user. The
> INFO is responded to with a 200 OK.
> 
> If the INFO request contains a body that the server does understand
> then, in the absense of INFO associated processing rules for the
> body, the server MUST respond with a 415 Unsupported Media Type
> message."

Looks good to me.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: Content-Length header and Multipart MIME
To: archow@hss.hns.com
Date: Thu, 27 Jan 2000 09:13:11 -0600 (CST)
Cc: Aparna.Vemuri@Level3.com, islepchin@dynamicsoft.com, kbinu@hss.hns.com,
        sip@lists.research.bell-labs.com
In-Reply-To: <65256873.001C9CD2.00@sampark.hss.hns.com> from "archow@hss.hns.com" at Jan 27, 2000 10:42:30 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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archow@hss.hns.com writes:
>I had read this section, and it confuses me a bit.
>If this section mentions a MUST then its mandatory that _all_ applications
>must append a CRLF only.
>Then what is the use for parsing incoming messages for CR, LF or CRLF ?
>I guess it would make sense only if there is a possibility that some
>application may send only a CR or an LF.
>Then the receiver can be lenient on this input. (ie if 6.6 was a SHOULD)
>
>However, if  6.6 states it as a MUST, which implies that a person who sends
>CR orLF only is not SIP compliant.
>then where is the question of the receiver expecting anything else - and if
>it does, it should not accept it as the sender
>violated a mandatory clause of the protocol.

It's one of the overarching philosophies of IETF protocols:
be strict in what you generate, but lenient in what you accept.
That way, for two implementations of a protocol to be
non-interoperable, they must *both* be broken.

This is one of the reasons occasionally cited why IETF protocols
tend to interoperate more sucessfully than those of other
standards bodies.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Date: Thu, 27 Jan 2000 11:04:12 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: Curse of the New Web Site -- A Power Outage
To: IETF SIP <sip@lists.research.bell-labs.com>
Message-id: <001701bf68e8$88c6e220$c2dc23a6@mcit.com>
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Yesterday I announced our new working web site.

Today a snow-blind backhoe operator dug up the power lines.

Guess what that means?

We'll be back on-line shortly.

--
Dean



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Jan 28 06:47:21 2000
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To: IETF-Announce: ;
Cc: sip@lists.research.bell-labs.com
From: Internet-Drafts@ietf.org
Reply-To: Internet-Drafts@ietf.org
Subject: I-D ACTION:draft-ietf-sip-isup-mime-00.txt
Date: Fri, 28 Jan 2000 06:42:32 -0500
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--NextPart

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

	Title		: MIME media types for ISUP and QSIG Objects
	Author(s)	: E. Zimmerer,  A. Vemuri, L. Ong, M. Zonoun,
                          M. Watson
	Filename	: draft-ietf-sip-isup-mime-00.txt
	Pages		: 6
	Date		: 27-Jan-00
	
This document describes MIME types for application/ISUP and 
application/QSIG objects for use in SIP applications, according to 
the rules defined in RFC 2048 [1].  These types can be used to identify 
ISUP and QSIG objects within a SIP message such as INVITE or INFO, 
as might be implemented when using SIP between legacy systems.

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

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sip-isup-mime-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-sip-isup-mime-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20000127130135.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-sip-isup-mime-00.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-sip-isup-mime-00.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20000127130135.I-D@ietf.org>

--OtherAccess--

--NextPart--





From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 05:36:07 2000
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Date: Mon, 31 Jan 2000 13:34:09 +0300
From: "Stanislav S. Timinsky" <timinsky@mera.ru>
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Subject: Question about SIP message
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Hello,

 I have the questions about SIP test message (test1.txt) on
 http://www.cs.columbia.edu/~hgs/sip/bakeoff2/testmsg.html site.

 The text of this message is:

-----------------------------
INVITE sip:vivekg@chair.dnrc.bell-labs.com SIP/2.0
TO :
 sip:vivekg@chair.dnrc.bell-labs.com ;   tag    = 1918181833n
From   : "J Rosenberg \\\"" <sip:jdrosen@lucent.com>
  ;
  tag = 98asjd8
CaLl-Id
 : 0ha0isndaksdj@10.1.1.1
cseq: 8
  INVITE
Via  : SIP  /   2.0 
 /UDP 
    135.180.130.133
Subject :
NewFangledHeader:   newfangled value
 more newfangled value
Content-Type: application/sdp
v:  SIP  / 2.0  / TCP     12.3.4.5   ;
  branch  =   9ikj8  ,
 SIP  /    2.0   / UDP  1.2.3.4   ; hidden   
m:"Quoted string \"\"" <sip:jdrosen@bell-labs.com> ; newparam = newvalue ;
  secondparam = secondvalue  ; q = 0.33
  (((nested comments) and (more)))   ,
 tel:4443322

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC
-----------------------------

1. Why tags in the TO and FROM headers have not hex format?
2. Why SDP body hasn't mandatory fields (session-name-field and
                                         time-fields)?
-- 
Best regards,
 Stanislav S. Timinsky              mailto:timinsky@mera.ru





From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 08:40:31 2000
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Message-ID: <65256877.004AA6C1.00@sampark.hss.hns.com>
Date: Mon, 31 Jan 2000 19:05:23 +0530
Subject: Encryption of From Header
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Hi,
Please refer to section 13.1.1 of RFC 2543

"The From header field SHOULD
   normally remain in the clear, but MAY be encrypted if required, in
   which case some proxies MAY return a 401 (Unauthorized) status if
   they require a From field."

The above indicates that its possible for FROM Header to be encrypted if so
 desired.

Now, please refer to Section 6 Table 4 which clearly mentions From (and
others) to have an "n" status in the enc colum implying
they MUST NOT be encrypted.

So which is the correct statememt ?

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems






From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 09:58:03 2000
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Message-ID: <3895A1D7.6EAD5985@exu.ericsson.se>
Date: Mon, 31 Jan 2000 08:53:11 -0600
From: Michelle Kitchings <Michelle.Kitchings@exu.ericsson.se>
Reply-To: Michelle.Kitchings@ericsson.com
Organization: Ericsson, Inc.
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To: "Schroeder, Tim" <schroeder@tri.sbc.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Comments on "Supported:" header draft
References: <4D45BA2A58A7D3119E050008C7E69E29079062@trimail2.tri.sbc.com>
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"Schroeder, Tim" wrote:
> > 8) Open issue 4:
> >             "Should we remove org.ietf.sip.supported and have the
> >              presence of the Supported header alone indicate support for
> >              this extension?"
> 
> I like the org.ietf.sip.supported.  It seems odd that the presence of
> Supported means something that the presence of Unsupported does not.  Or
> would Unsupported in a request also imply org.ietf.sip.supported?  Since
> Unsupported already exists in responses, the presence of Unsupported there
> could not be used to imply anything.  The fewer "hidden" implications the
> better, I think.


Here is my first post, so bear with me.  :)

If the motivation for making "Supported:" imply "org.ietf.sip.supported" is
because of the extensive length of the feature name, couldn't the feature
lists be interpreted such that any feature without a reverse domain is assumed
to be "org.ietf.sip"?  This would mean that

  supported = org.ietf.sip.supported
  100rel = org.ietf.sip.100rel
  info = org.ietf.sip.info
  etc. ...

This mechanism makes any/all IETF extension cheaper in terms of bandwidth. 

The downfall to my idea is that if a provider (eg, Ericsson) defines an
extension called "info" without using a reverse domain, then the servers could
inadvertantly interpet that as the wrong kind of info--which would happen
anyway if some other provider (eg, MCI) was also using info without reverse
domain.  Perhaps this default domain interpretation would "blackmail"
providers into making sure that they fully-qualify all features?

Cheers,
Michelle :^)

-- 
Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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From: Anders Kristensen <ak@hplb.hpl.hp.com>
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Subject: Re: Comments on "Supported:" header draft
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I like your proposal of omitting the lengthy prefix org.ietf.sip from
IETF sanctioned SIP extensions -- I see no problem with that at all. 
I'd still like to make Supported:" imply support for the "supported"
extension as well, though.

All these little savings add up ya' know.

Regards,
Anders

Michelle Kitchings wrote:
> 
> "Schroeder, Tim" wrote:
> > > 8) Open issue 4:
> > >             "Should we remove org.ietf.sip.supported and have the
> > >              presence of the Supported header alone indicate support for
> > >              this extension?"
> >
> > I like the org.ietf.sip.supported.  It seems odd that the presence of
> > Supported means something that the presence of Unsupported does not.  Or
> > would Unsupported in a request also imply org.ietf.sip.supported?  Since
> > Unsupported already exists in responses, the presence of Unsupported there
> > could not be used to imply anything.  The fewer "hidden" implications the
> > better, I think.
> 
> Here is my first post, so bear with me.  :)
> 
> If the motivation for making "Supported:" imply "org.ietf.sip.supported" is
> because of the extensive length of the feature name, couldn't the feature
> lists be interpreted such that any feature without a reverse domain is assumed
> to be "org.ietf.sip"?  This would mean that
> 
>   supported = org.ietf.sip.supported
>   100rel = org.ietf.sip.100rel
>   info = org.ietf.sip.info
>   etc. ...
> 
> This mechanism makes any/all IETF extension cheaper in terms of bandwidth.
> 
> The downfall to my idea is that if a provider (eg, Ericsson) defines an
> extension called "info" without using a reverse domain, then the servers could
> inadvertantly interpet that as the wrong kind of info--which would happen
> anyway if some other provider (eg, MCI) was also using info without reverse
> domain.  Perhaps this default domain interpretation would "blackmail"
> providers into making sure that they fully-qualify all features?
> 
> Cheers,
> Michelle :^)
> 
> --
> Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
> Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: Comments on "Supported:" header draft
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I think we have clear consensus for having the presence of Supported
imply support for org.ietf.sip.supported.

Issue closed.

-Jonathan R.

Anders Kristensen wrote:
> 
> I like your proposal of omitting the lengthy prefix org.ietf.sip from
> IETF sanctioned SIP extensions -- I see no problem with that at all.
> I'd still like to make Supported:" imply support for the "supported"
> extension as well, though.
> 
> All these little savings add up ya' know.
> 
> Regards,
> Anders
> 
> Michelle Kitchings wrote:
> >
> > "Schroeder, Tim" wrote:
> > > > 8) Open issue 4:
> > > >             "Should we remove org.ietf.sip.supported and have the
> > > >              presence of the Supported header alone indicate support for
> > > >              this extension?"
> > >
> > > I like the org.ietf.sip.supported.  It seems odd that the presence of
> > > Supported means something that the presence of Unsupported does not.  Or
> > > would Unsupported in a request also imply org.ietf.sip.supported?  Since
> > > Unsupported already exists in responses, the presence of Unsupported there
> > > could not be used to imply anything.  The fewer "hidden" implications the
> > > better, I think.
> >
> > Here is my first post, so bear with me.  :)
> >
> > If the motivation for making "Supported:" imply "org.ietf.sip.supported" is
> > because of the extensive length of the feature name, couldn't the feature
> > lists be interpreted such that any feature without a reverse domain is assumed
> > to be "org.ietf.sip"?  This would mean that
> >
> >   supported = org.ietf.sip.supported
> >   100rel = org.ietf.sip.100rel
> >   info = org.ietf.sip.info
> >   etc. ...
> >
> > This mechanism makes any/all IETF extension cheaper in terms of bandwidth.
> >
> > The downfall to my idea is that if a provider (eg, Ericsson) defines an
> > extension called "info" without using a reverse domain, then the servers could
> > inadvertantly interpet that as the wrong kind of info--which would happen
> > anyway if some other provider (eg, MCI) was also using info without reverse
> > domain.  Perhaps this default domain interpretation would "blackmail"
> > providers into making sure that they fully-qualify all features?
> >
> > Cheers,
> > Michelle :^)
> >
> > --
> > Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
> > Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA
> 
> --
> Anders Kristensen <ak@hplb.hpl.hp.com>,
> http://www-uk.hpl.hp.com/people/ak/
> Hewlett-Packard Labs, Bristol, UK

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Jonathan Rosenberg wrote:
> 
> I think we have clear consensus for having the presence of Supported
> imply support for org.ietf.sip.supported.
> 
> Issue closed.

Right. There was another proposal, though: to loose the "org.ietf.sip"
prefix from "official" extensions. 

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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Jonathan,

I agree with presence-imply and wasn't attempting to re-open that particular
issue.

But, the whole discussion made me think of the idea of an implied
"org.ietf.sip" in general.  Do you think the idea has merit as a means to
shortening messages that have Require, Proxy-require, Supported, and
Unsupported?  Like someone said in another post, we could end up having dozens
of IETF extensions.

Cheers,
Michelle :^)


Jonathan Rosenberg wrote:
> 
> I think we have clear consensus for having the presence of Supported
> imply support for org.ietf.sip.supported.
> 
> Issue closed.
> 
> -Jonathan R.
> 
> > Michelle Kitchings wrote:
> > >
...
> > > Here is my first post, so bear with me.  :)
> > >
> > > If the motivation for making "Supported:" imply "org.ietf.sip.supported" is
> > > because of the extensive length of the feature name, couldn't the feature
> > > lists be interpreted such that any feature without a reverse domain is assumed
> > > to be "org.ietf.sip"?  This would mean that
> > >
> > >   supported = org.ietf.sip.supported
> > >   100rel = org.ietf.sip.100rel
> > >   info = org.ietf.sip.info
> > >   etc. ...
> > >
> > > This mechanism makes any/all IETF extension cheaper in terms of bandwidth.
> > >
> > > The downfall to my idea is that if a provider (eg, Ericsson) defines an
> > > extension called "info" without using a reverse domain, then the servers could
> > > inadvertantly interpet that as the wrong kind of info--which would happen
> > > anyway if some other provider (eg, MCI) was also using info without reverse
> > > domain.  Perhaps this default domain interpretation would "blackmail"
> > > providers into making sure that they fully-qualify all features?
> > >
> > > Cheers,
> > > Michelle :^)
> > >
> > > --
> > > Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
> > > Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



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Subject: QoS
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Does someone have information regarding recommendations or future intentions
to improve QoS on a SIP network?



Elly Gazal
System Engineer

Signaling R&D Group
Comverse Network Systems Ltd.

Tel:  972-3-6458994
Fax: 972-3-6458915
Email: Elly_Gazal@icomverse.com




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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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A "SIP Network" does not provide QoS for media traffic. IP networks
provide QoS for media traffic. They use tools such as diffserv, RSVP,
and mpls to accomplish this. Several people have looked at ways in which
standard QoS mechanisms can be integrated cleanly into services signaled
with SIP. Take a look at:

http://www.softarmor.com/sipwg/drafts/draft-dcsgroup-sip-resource-00.txt
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-mmusic-sdp-qos-00.txt
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sinnreich-interdomain-sip-qos-osp-00.txt

Work is currently underway to merge the first two drafts.

-Jonathan R.

"Gazal, Elly" wrote:
> 
> Does someone have information regarding recommendations or future intentions
> to improve QoS on a SIP network?
> 
> Elly Gazal
> System Engineer
> 
> Signaling R&D Group
> Comverse Network Systems Ltd.
> 
> Tel:  972-3-6458994
> Fax: 972-3-6458915
> Email: Elly_Gazal@icomverse.com

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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To: "Gazal, Elly" <Elly_Gazal@icomverse.com>
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Elly,

You could start with a look at the following documents:

"Interdomain IP Communications with QoS, Authorization and Usage Reporting",  H.
Sinnreich, S. Donovan, D. Rawlins, S. Thomas. October 1999
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sinnreich-interdomain-sip-qos-osp-00.txt

"Use of SIP for the Reservation of QoS guaranteed Paths",  M. Gibson and J.
Crowcroft. October 1999.
 http://www.cs.columbia.edu/~hgs/sip/drafts/draft-gibson-sip-qos-resv-00.txt

"Establishing QoS and Security Preconditions for SDP Sessions", J. Rosenberg, H.
Schulzrinne, S. Donovan
 http://www.cs.columbia.edu/~hgs/sip/drafts/draft-ietf-mmusic-sdp-qos-00.txt

"DIAMETER: Policy and Accounting Extension for SIP", Ping Pan, Henning
Schulzrinne and Pat Calhoun
 http://www.cs.columbia.edu/~hgs/sip/drafts/draft-pan-diameter-sip-01.txt


Manu

--
Emmanuel Bertrand
mailto:eb@indigo-software.com
http://www.indigo-software.com



"Gazal, Elly" wrote:

> Does someone have information regarding recommendations or future intentions
> to improve QoS on a SIP network?
>
> Elly Gazal
> System Engineer
>
> Signaling R&D Group
> Comverse Network Systems Ltd.
>
> Tel:  972-3-6458994
> Fax: 972-3-6458915
> Email: Elly_Gazal@icomverse.com







From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 14:50:24 2000
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From: "Buchanan, Paul" <Paul.Buchanan@cwcom.co.uk>
To: sip@lists.research.bell-labs.com
Subject: SIP AND CLID
Date: Mon, 31 Jan 2000 19:48:54 -0000
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Can anyone help me ?  

How will SIP and CLI (Calling Line Identifier) work with each other?

Thanks

Paul Buchanan
Darkness Removal a Speciality 
IP Net Technology Planning
Waterside House, Longshot Lane, Bracknell Berks RG12 1XL
TEL:			01344 704064
FAX:			01344 726304
MOBILE (BEST BET)	0780 833 8143
PA247.com		paulbuchanan@mail.com
"Those who bring sunshine into the lives of others, cannot keep it from
themselves."
*Sir James M. Barrie{1860-1937 British Playwright and clever bloke}




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 18:32:00 2000
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Section 3.1 says:

>>
Note: It is mandatory for SoftSwitches to specify the 'version' 
of  the ISUP message. Proxies, redirect servers, etc., have no 
need to  process/specify this information.

The use of the 'version' parameter allows differentiation 
between  different base ISUP variants. This enables the 
SoftSwitch (also known as a Media Gateway Controller) to 
recognize and parse the message correctly,  or (possibly) to 
reject the message if the  particular ISUP variant is not 
supported.
>>

I'm wondering if it's necessary to bring in these terms (SoftSwitch, media
gateway controller) from an assumed architecture.  The draft stands on its
own just as well without them, if I understand it correctly.

The first paragraph quoted above is saying that the user agents (clients and
servers both) must specify the version, but that the proxy and redirect
servers do not need to specify or process it.  The second paragraph says the
user agents (again, both clients and servers) will use the version to
recognize and parse the message, etc.

I'd like to see this changed, so that the draft is more generally applicable
to any architecture that needs to package ISUP/QSIP within a SIP message,
rather than tying it to a particular SoftSwitch/MGC architecture.

Tim Schroeder



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 19:11:49 2000
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"Stanislav S. Timinsky" wrote:
> 
> <..> 
> 1. Why tags in the TO and FROM headers have not hex format?
> 2. Why SDP body hasn't mandatory fields (session-name-field and
>                                          time-fields)?

I guess this is in line with the general rule that you should strictly
adhere to the protocol when generating your own messages but be lenient
when parsing received messages. This approach greatly increases the
chances of interoperability among different vendors' implementations.

Returning to the example you mentioned:

1. The only real requirement for the tag parameter is its uniqueness and
UUIDs are just a way to enforce this. Thus, having a non-hex tag value
does not represent a major error and should probably be accepted.

2. SDP was initially designed for multicast session announcements over
SAP. Hence, some of the parameters listed as required in RFC2327 are
less vital for SIP session descriptions, and some don't even make much
sense in SIP context. E.g., SDP session name duplicates the
functionality of SIP Subject header, and UAs are likely to assume that
the media exchange starts whenever the call setup is complete and ends
whenever a BYE is sent, thus making time field unnecessary.

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 19:45:48 2000
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Date: Mon, 31 Jan 2000 18:40:28 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: SIP AND CLID
In-reply-to: 
 <A989508D4E92D111AA8F0000F80687AF0AE0EEE6@gbcwcbrtm001.isops.mercury.co.uk>
To: "Buchanan, Paul" <Paul.Buchanan@cwcom.co.uk>,
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There have been several approached put forward. The DCS approach put forward
in

http://www.softarmor.com/sipwg/drafts/draft-dcsgroup-sip-privacy-00.txt

is pretty thorough.

I believe the "IPTEL call flows" draft proposed a simpler approach. Roughly,
it translates to:

1) If there is no presented calling party number, set the From: field to
"UNKNOWN"
2) If the calling party privacy field indicates a private number, set the
From: field to "PRIVATE"
3) If there is a number presented, populate it into the From: field as the
username component. We might better use a Tel: url in this approach.

This simpler approach was taken due to the need to get things working
immediately.

--
Dean Willis

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Buchanan,
> Paul
> Sent: Monday, January 31, 2000 1:49 PM
> To: sip@lists.research.bell-labs.com
> Subject: SIP AND CLID
>
>
> Can anyone help me ?
>
> How will SIP and CLI (Calling Line Identifier) work with each other?
>
> Thanks
>
> Paul Buchanan
> Darkness Removal a Speciality
> IP Net Technology Planning
> Waterside House, Longshot Lane, Bracknell Berks RG12 1XL
> TEL:			01344 704064
> FAX:			01344 726304
> MOBILE (BEST BET)	0780 833 8143
> PA247.com		paulbuchanan@mail.com
> "Those who bring sunshine into the lives of others, cannot keep it from
> themselves."
> *Sir James M. Barrie{1860-1937 British Playwright and clever bloke}
>
>




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 21:26:01 2000
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Date: Mon, 31 Jan 2000 21:22:33 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Cc: "Stanislav S. Timinsky" <timinsky@mera.ru>,
        sip@lists.research.bell-labs.com, prj.men.sip@mera.ru
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Igor Slepchin wrote:
> 

> 
> 2. SDP was initially designed for multicast session announcements over
> SAP. Hence, some of the parameters listed as required in RFC2327 are
> less vital for SIP session descriptions, and some don't even make much
> sense in SIP context. E.g., SDP session name duplicates the
> functionality of SIP Subject header, and UAs are likely to assume that
> the media exchange starts whenever the call setup is complete and ends
> whenever a BYE is sent, thus making time field unnecessary.

While this may be true in current "make-it-look-like-a-phone"
implementations, there are a number of useful applications where SDP and
SIP information convey different information. For example, the spec
gives the example of an invitation to a broadcast session. The subject
of the invitation may well be different from the "Internet TV" session
title. Similarly, for the date, where scheduling calls into the future
can be a rather useful group coordination tool.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Jan 31 23:59:58 2000
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Content-Transfer-Encoding: 7bit

Its also worth noting that rfc2543 does state that if the calling user
wishes to remain anonymous, they use the display name "Anonymous" in the
From field. 

-Jonathan R.

Dean Willis wrote:
> 
> There have been several approached put forward. The DCS approach put forward
> in
> 
> http://www.softarmor.com/sipwg/drafts/draft-dcsgroup-sip-privacy-00.txt
> 
> is pretty thorough.
> 
> I believe the "IPTEL call flows" draft proposed a simpler approach. Roughly,
> it translates to:
> 
> 1) If there is no presented calling party number, set the From: field to
> "UNKNOWN"
> 2) If the calling party privacy field indicates a private number, set the
> From: field to "PRIVATE"
> 3) If there is a number presented, populate it into the From: field as the
> username component. We might better use a Tel: url in this approach.
> 
> This simpler approach was taken due to the need to get things working
> immediately.
> 
> --
> Dean Willis
> 
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Buchanan,
> > Paul
> > Sent: Monday, January 31, 2000 1:49 PM
> > To: sip@lists.research.bell-labs.com
> > Subject: SIP AND CLID
> >
> >
> > Can anyone help me ?
> >
> > How will SIP and CLI (Calling Line Identifier) work with each other?
> >
> > Thanks
> >
> > Paul Buchanan
> > Darkness Removal a Speciality
> > IP Net Technology Planning
> > Waterside House, Longshot Lane, Bracknell Berks RG12 1XL
> > TEL:                  01344 704064
> > FAX:                  01344 726304
> > MOBILE (BEST BET)     0780 833 8143
> > PA247.com             paulbuchanan@mail.com
> > "Those who bring sunshine into the lives of others, cannot keep it from
> > themselves."
> > *Sir James M. Barrie{1860-1937 British Playwright and clever bloke}
> >
> >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



