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From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
References: <3E0B6293.4030600@dynamicsoft.com>
Subject: Re: [Sip] INFO considered harmful
Date: Wed, 1 Jan 2003 11:43:45 -0000
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How about we use new methods to define end-to-end application data, but name
the messages INFO-*.  e.g. INFO-MY-FEATURE, so that they can be readily
identified as application data rather than SIP state affecting methods.
Allow: can then be used to negotiate 'INFO' extensions.


- It's easy to parse.

- It's very easy for a stack to treat it generically.

- It can reuse existing SIP features to do negotiation etc.  (e.g. Allow)

- It could be easily documented in the SIP usage document.



My other thoughts on the subject (and what has led me to the above position
are):

- I think its important to have some mechanism that allows end-to-end
application data exchange that is independent of SIP state.

- Doing this in-band (rather than starting up potentially numerous side
connections) will be desirable for some applications (such as those that use
a small amounts of data).

- It seems appealing to be readily able to tell which methods affect SIP
state, and which is application level data.

- Require:, Supported: and Allow: already negotiation of features and where
possible it would be beneficial to re-use these mechanisms rather than
invent new ones..

- An INFO mechanism does not need to offer any semantics beyond simple data
delivery (although it might be nice to have an in-order delivery service?)
(and the afore mentioned negotiation).

- The main problem with INFO is that it doesn't give a single specific way
of identifying which application a particular INFO method is related to.
Hence it necessitates some sort of heuristic procedure to be used.


Some other ways to move ahead might be to:

- Specify that the application that the INFO corresponds to is defined by
the MIME type of the body.

- Define a new header that defines the application that the INFO corresponds
to.

Pete.

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: <sip@ietf.org>
Sent: 26 December 2002 20:12
Subject: [Sip] INFO considered harmful


> Folks,
>
> For some time, I've been complaining about the continued abuse of the
> INFO method for things that should be done other ways (frequently, not
> using sip at all). I've written up a summary of the problems and a
> proposed path forward. Specifically, I'd like to obsolete INFO and
> replace it with a spec that is approved ONLY for SIP-T.
>
> Until the I-D appears, you can pick up a version at:
>
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From: Peter =?UTF-8?B?UMOkcHBpbmdoYXVz?= <peter.paeppinghaus@siemens.com>
Organization: Siemens AG
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I have posted this earlier, but it seems to have gone
unnoticed.

Hi,

I believe to have stumbled on a buggy example in RFC 3261.

The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
an Expires header, but no expires parameter in the Contact header.

This violates RFC 3261, sec. 10.3, bullet 8:
"Each Contact value MUST feature an "expires" parameter
   indicating its expiration interval chosen by the registrar."

Regards,

Peter Paeppinghaus

-- 
Dr. Peter Päppinghaus
Siemens AG            | Phone:    +49 89 - 722 40065
ICM N PG U ID A1      | Fax:      +49 89 - 722 58726
Hofmannstr. 51        | Visitors: Building 1713 / Room 714
D - 81359 München     | Email:    peter.paeppinghaus@siemens.com

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: Re: Repost: [Sip] Buggy example in RFC 3261
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Sorry for not responding. I did notice this. There is an error here, as 
you point out. What is NOT clear is whether the error is in the example, 
or in the normative text. For REGISTER requests, you can use either the 
Expires header field or the expires contact header field parameter. It 
seems odd that this same rule would not apply to REGISTER responses. My 
personal recollection was that either the Expires header field or the 
expires contact param was OK for either request or response. This 
doesn't match the normative text though.

Does anyone use the Expires header field and NOT the expires contact 
param in REGISTER responses?

-Jonathan R.

Peter Päppinghaus wrote:
> I have posted this earlier, but it seems to have gone
> unnoticed.
> 
> Hi,
> 
> I believe to have stumbled on a buggy example in RFC 3261.
> 
> The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
> an Expires header, but no expires parameter in the Contact header.
> 
> This violates RFC 3261, sec. 10.3, bullet 8:
> "Each Contact value MUST feature an "expires" parameter
>   indicating its expiration interval chosen by the registrar."
> 
> Regards,
> 
> Peter Paeppinghaus
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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Date: Thu, 02 Jan 2003 00:02:13 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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PS, a bugzilla bug was logged against this some time back:
http://www.sipwg.org/sipwg/show_bug.cgi?id=682

-Jonathan R.

Peter Päppinghaus wrote:
> I have posted this earlier, but it seems to have gone
> unnoticed.
> 
> Hi,
> 
> I believe to have stumbled on a buggy example in RFC 3261.
> 
> The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
> an Expires header, but no expires parameter in the Contact header.
> 
> This violates RFC 3261, sec. 10.3, bullet 8:
> "Each Contact value MUST feature an "expires" parameter
>   indicating its expiration interval chosen by the registrar."
> 
> Regards,
> 
> Peter Paeppinghaus
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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response inline.

Amit Bhadoria wrote:
> repost from yesterday...
> 
> ---------- Forwarded message ----------
> Date: Mon, 30 Dec 2002 17:20:50 -0800 (PST)
> From: Amit Bhadoria <bhadoria@cisco.com>
> To: sip@ietf.org
> Subject: [TLS] how to validate multiple domains with single certificate
> 
> hi,
> 
> i have some questions based on rfc 3263 "locating sip services". in
> rfc 3263 it is mentioned that:
>   "For NAPTR records with SIPS protocol fields, (if the server is using
>    a site certificate), the domain name in the query and the domain name
>    in the replacement field MUST both be valid based on the site
>    certificate handed out by the server in the TLS exchange.  Similarly,
>    the domain name in the SRV query and the domain name in the target in
>    the SRV record MUST both be valid based on the same site certificate.
>    Otherwise, an attacker could modify the DNS records to contain
>    replacement values in a different domain, and the client could not
>    validate that this was the desired behavior or the result of an
>    attack."
> 
> now, there are three potentially different domains in concern while
> processing a SIPS request:
> 	1. domain used in NAPTR querry: this is the target domain for this
> 	   request
> 	2. domain mentioned in replacement field of NAPTR records: this
> 	   is the domain used in the SRV querry, obtained from replacement
> 	   field of NAPTR records
> 	3. domain mentioned in target in the SRV record: the domain
> 	   mentioned in the SRV record selected for final A lookup
> 
> is it possible to have all the three domains different, or is it just a
> bad example (due to my limited knowledge of DNS setups), and in practical
> cases the three domains will either be same as TARGET or a subdomain of
> it?

They can be different as far as DNS is concerned, but the point of the 
text is to say that for SIPS URIs, they generally shouldn't be. 
Otherwise, there is this potential DNS attack. I say "generally" because 
it is possible for a single site certificate to be valid for multiple 
domains. In that case, so long as the domains in the DNS were all valid 
for that single certificate, its OK to use different domains. But, this 
is rare I think.


> 
> incase my imagination is correct, and it is possible to have a setup where
> all three domains (from three steps of DNS procedures) are different, then
> how can a client validate these three domains using the same certificate?

It can't. Thats why the domains really can't be different. The only 
exception is if single site certificate can validate multiple domains.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Date: Thu, 02 Jan 2003 11:16:41 +0530
From: "Nataraju A.B." <natarajuab@huawei.com>
Subject: Re: [Sip] A simple query...
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
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Hi, 

Server           =  "Server" HCOLON server-val *(LWS server-val)
 server-val       =  product / comment
product          =  token [SLASH product-version]

product-version  =  token

both product and product version are of type tokens.
if two tokes are in the input what should be considered. I mean product an dprod version OR two product values.

 for ex : Server: HomeServer HS v2
in this case HS is prod version or product2.

Regards,
-------------------------------------------
Nataraju A.B.
Huawei Technologies India Pvt. Ltd.,
Tel : +91-80-5216824/6/7 Xtn 142
-------------------------------------------
  ----- Original Message ----- 
  From: Jonathan Rosenberg 
  To: Nataraju A.B. 
  Cc: sip@ietf.org 
  Sent: Tuesday, 31 December 2002 20-06
  Subject: Re: [Sip] A simple query...


  inline.

  Nataraju A.B. wrote:
  > Hi,
  > I have a query regarding the server/user-agent coding format:
  > The ABNF encoding specifies it as : 
  >  
  > Server           =  "Server" HCOLON server-val *(LWS server-val)
  > server-val       =  product / comment
  > product          =  token [SLASH product-version]
  > product-version  =  token
  >  
  > whereas teh SIP-rfc says it should be
  >  Server: HomeServer v2
  > CanI know the correct format to be used.

  The above is just an example of the server header. It is a valid value 
  according to the BNF, since the BNF allows a series of lws-separated 
  tokens, without any slashes. Thus, either of these is valid:

  Server: HomeServer v2
  Server: HomeServer/v2


  >  
  > If the SIP format is to be used then is it product version is a 
  > mandatory argument.

  Version is not mandatory according to the BNF.

  -Jonathan R.
  -- 
  Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
  Chief Scientist                             First Floor
  dynamicsoft                                 East Hanover, NJ 07936
  jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
  http://www.jdrosen.net                      PHONE: (973) 952-5000
  http://www.dynamicsoft.com

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

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<DIV><FONT color=#008000>Hi, </FONT></DIV>
<DIV><FONT color=#008000></FONT>&nbsp;</DIV>
<DIV><SPAN 
style="FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman'; mso-fareast-font-family: 'MS Mincho'; mso-ansi-language: EN-AU; mso-fareast-language: EN-US; mso-bidi-language: AR-SA">Server<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
</SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>"Server" HCOLON server-val 
*(LWS server-val)</SPAN></DIV>
<DIV><SPAN 
style="FONT-SIZE: 12pt; FONT-FAMILY: 'Times New Roman'; mso-fareast-font-family: 'MS Mincho'; mso-ansi-language: EN-AU; mso-fareast-language: EN-US; mso-bidi-language: AR-SA"><FONT 
color=#008000><FONT color=#000000>&nbsp;</FONT><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow">server-val<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>=<SPAN 
style="mso-spacerun: yes">&nbsp; </SPAN>product / comment<?xml:namespace prefix 
= o ns = "urn:schemas-microsoft-com:office:office" /><o:p></o:p></SPAN>
<P class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow">product<SPAN 
style="mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
</SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; </SPAN>token [SLASH 
product-version]<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN lang=EN-US 
style="FONT-SIZE: 10pt; BACKGROUND: yellow; FONT-FAMILY: 'Courier New'; mso-bidi-font-size: 12.0pt; mso-highlight: yellow">product-version<SPAN 
style="mso-spacerun: yes">&nbsp; </SPAN>=<SPAN style="mso-spacerun: yes">&nbsp; 
</SPAN>token</SPAN></P></FONT></SPAN></DIV>
<DIV><FONT color=#008000>both product and product version are of type 
tokens.</FONT></DIV>
<DIV><FONT color=#008000>if two tokes are in the input what should be 
considered. I mean product an dprod version OR two product values.</FONT></DIV>
<DIV><FONT color=#008000></FONT>&nbsp;</DIV>
<DIV><FONT color=#008000>&nbsp;for ex : <FONT color=#000000>Server: 
</FONT></FONT><FONT color=#ff0000><STRONG>HomeServer <FONT 
color=#0000ff>HS</FONT> v2</STRONG></FONT></DIV>
<DIV>in this case HS is prod version or product2.</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=#008000>Regards,</FONT></DIV>
<DIV>-------------------------------------------<BR>Nataraju A.B.<BR>Huawei 
Technologies India Pvt. Ltd.,<BR>Tel : +91-80-5216824/6/7 Xtn 
142<BR>-------------------------------------------</DIV>
<BLOCKQUOTE 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #008000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=jdrosen@dynamicsoft.com 
  href="mailto:jdrosen@dynamicsoft.com">Jonathan Rosenberg</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=natarajuab@huawei.com 
  href="mailto:natarajuab@huawei.com">Nataraju A.B.</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Cc:</B> <A title=sip@ietf.org 
  href="mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, 31 December 2002 
  20-06</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Sip] A simple 
query...</DIV>
  <DIV><FONT color=#008000></FONT><FONT 
  color=#008000></FONT><BR></DIV>inline.<BR><BR>Nataraju A.B. wrote:<BR>&gt; 
  Hi,<BR>&gt; I have a query regarding the server/user-agent coding 
  format:<BR>&gt; The ABNF encoding specifies it as : <BR>&gt;&nbsp; <BR>&gt; 
  Server&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =&nbsp; 
  "Server" HCOLON server-val *(LWS server-val)<BR>&gt; 
  server-val&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =&nbsp; product / 
  comment<BR>&gt; product&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  =&nbsp; token [SLASH product-version]<BR>&gt; product-version&nbsp; =&nbsp; 
  token<BR>&gt;&nbsp; <BR>&gt; whereas teh SIP-rfc says it should 
  be<BR>&gt;&nbsp; Server: HomeServer v2<BR>&gt; CanI know the correct format to 
  be used.<BR><BR>The above is just an example of the server header. It is a 
  valid value <BR>according to the BNF, since the BNF allows a series of 
  lws-separated <BR>tokens, without any slashes. Thus, either of these is 
  valid:<BR><BR>Server: HomeServer v2<BR>Server: 
  HomeServer/v2<BR><BR><BR>&gt;&nbsp; <BR>&gt; If the SIP format is to be used 
  then is it product version is a <BR>&gt; mandatory argument.<BR><BR>Version is 
  not mandatory according to the BNF.<BR><BR>-Jonathan R.<BR>-- <BR>Jonathan D. 
  Rosenberg, 
  Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  72 Eagle Rock Ave.<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; 
  First 
  Floor<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; 
  East Hanover, NJ 07936<BR><A 
  href="mailto:jdrosen@dynamicsoft.com">jdrosen@dynamicsoft.com</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  FAX:&nbsp;&nbsp; (973) 952-5050<BR><A 
  href="http://www.jdrosen.net">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  PHONE: (973) 952-5000<BR><A 
  href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</A><BR><BR>_______________________________________________<BR>Sip 
  mailing list&nbsp; <A 
  href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</A><BR>This 
  list is for NEW development of the core SIP Protocol<BR>Use <A 
  href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</A> 
  for questions on current sip<BR>Use <A 
  href="mailto:sipping@ietf.org">sipping@ietf.org</A> for new developments on 
  the application of sip</BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_OpowXGA61r7iJ+ETTzIEeA)--
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 11:23:42 -0000 
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Jon,

There may be alternatives to INFO for everything in ANSI ISUP, but in ITU-T
there is the Application Transport Message (APM). The purpose of this is,
...err..., to transparently transport Application layer information.

I think any new method defined for carrying this would end up with identical
semantics to 'INFO restricted to SIP-T' as proposed in Jonathan's draft -
namely 'transport of application layer information, with the restriction
that the application layer is SIP-T'.

If you wanted to restrict its use to APM, this is something you would
restrict at the SIP-T application layer, not in SIP.

...Mark

> -----Original Message-----
> From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
> Sent: 30 December 2002 19:46
> To: 'Jonathan Rosenberg'; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> First, I think it is a really good idea to restrict the 
> applicability of SIP
> INFO method - I very much agree with the problem defined in 
> Section 2 of
> this draft.
> 
> That much said, when it comes to solutions I am torn between 
> option number 3
> and a more radical alternative - deprecating INFO altogether. 
> While it is
> true that INFO was inspired by some of the requirements of 
> SIP-T, over time,
> we've found less and less use for INFO. The original vision 
> for SIP+ (an
> ancestor of SIP-T) relied on INFO for sending virtually every 
> ISUP message,
> but today, there just isn't that much that we do with INFO. 
> Practically
> speaking, we could probably find alternatives to INFO for 
> carrying ISUP data
> in these cases (which include SUS/RES, INF/INR, some 
> provisional response
> cases). Using INFO to send other ISUP messages is frequently 
> misguided, and
> can potentially lead to the same sorts of interoperability 
> problems we have
> with INFO in other arenas. 
> 
> Today, however, RFC3398 does allow the use of INFO in these 
> three cases, so
> it's simplest to let the INFO method to exist (option 3) for 
> the time being
> - but we should be sure to document, even for ISUP 
> transmissions, the cases
> in which it is and is not appropriate. In the long term, 
> though, I think we
> should just get rid of it entirely.
> 
> Jon Peterson
> NeuStar, Inc.
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, December 26, 2002 12:12 PM
> > To: sip@ietf.org
> > Subject: [Sip] INFO considered harmful
> > 
> > 
> > Folks,
> > 
> > For some time, I've been complaining about the continued 
> abuse of the 
> > INFO method for things that should be done other ways 
> > (frequently, not 
> > using sip at all). I've written up a summary of the problems and a 
> > proposed path forward. Specifically, I'd like to obsolete INFO and 
> > replace it with a spec that is approved ONLY for SIP-T.
> > 
> > Until the I-D appears, you can pick up a version at:
> > 
> > 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> > 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> > 
> > Thanks,
> > Jonathan R.
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<P><FONT SIZE=3D2>Jon,</FONT>
</P>

<P><FONT SIZE=3D2>There may be alternatives to INFO for everything in =
ANSI ISUP, but in ITU-T there is the Application Transport Message =
(APM). The purpose of this is, ...err..., to transparently transport =
Application layer information.</FONT></P>

<P><FONT SIZE=3D2>I think any new method defined for carrying this =
would end up with identical semantics to 'INFO restricted to SIP-T' as =
proposed in Jonathan's draft - namely 'transport of application layer =
information, with the restriction that the application layer is =
SIP-T'.</FONT></P>

<P><FONT SIZE=3D2>If you wanted to restrict its use to APM, this is =
something you would restrict at the SIP-T application layer, not in =
SIP.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Peterson, Jon [<A =
HREF=3D"mailto:jon.peterson@neustar.biz">mailto:jon.peterson@neustar.biz=
</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 30 December 2002 19:46</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Jonathan Rosenberg'; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] INFO considered =
harmful</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; First, I think it is a really good idea to =
restrict the </FONT>
<BR><FONT SIZE=3D2>&gt; applicability of SIP</FONT>
<BR><FONT SIZE=3D2>&gt; INFO method - I very much agree with the =
problem defined in </FONT>
<BR><FONT SIZE=3D2>&gt; Section 2 of</FONT>
<BR><FONT SIZE=3D2>&gt; this draft.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; That much said, when it comes to solutions I am =
torn between </FONT>
<BR><FONT SIZE=3D2>&gt; option number 3</FONT>
<BR><FONT SIZE=3D2>&gt; and a more radical alternative - deprecating =
INFO altogether. </FONT>
<BR><FONT SIZE=3D2>&gt; While it is</FONT>
<BR><FONT SIZE=3D2>&gt; true that INFO was inspired by some of the =
requirements of </FONT>
<BR><FONT SIZE=3D2>&gt; SIP-T, over time,</FONT>
<BR><FONT SIZE=3D2>&gt; we've found less and less use for INFO. The =
original vision </FONT>
<BR><FONT SIZE=3D2>&gt; for SIP+ (an</FONT>
<BR><FONT SIZE=3D2>&gt; ancestor of SIP-T) relied on INFO for sending =
virtually every </FONT>
<BR><FONT SIZE=3D2>&gt; ISUP message,</FONT>
<BR><FONT SIZE=3D2>&gt; but today, there just isn't that much that we =
do with INFO. </FONT>
<BR><FONT SIZE=3D2>&gt; Practically</FONT>
<BR><FONT SIZE=3D2>&gt; speaking, we could probably find alternatives =
to INFO for </FONT>
<BR><FONT SIZE=3D2>&gt; carrying ISUP data</FONT>
<BR><FONT SIZE=3D2>&gt; in these cases (which include SUS/RES, INF/INR, =
some </FONT>
<BR><FONT SIZE=3D2>&gt; provisional response</FONT>
<BR><FONT SIZE=3D2>&gt; cases). Using INFO to send other ISUP messages =
is frequently </FONT>
<BR><FONT SIZE=3D2>&gt; misguided, and</FONT>
<BR><FONT SIZE=3D2>&gt; can potentially lead to the same sorts of =
interoperability </FONT>
<BR><FONT SIZE=3D2>&gt; problems we have</FONT>
<BR><FONT SIZE=3D2>&gt; with INFO in other arenas. </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Today, however, RFC3398 does allow the use of =
INFO in these </FONT>
<BR><FONT SIZE=3D2>&gt; three cases, so</FONT>
<BR><FONT SIZE=3D2>&gt; it's simplest to let the INFO method to exist =
(option 3) for </FONT>
<BR><FONT SIZE=3D2>&gt; the time being</FONT>
<BR><FONT SIZE=3D2>&gt; - but we should be sure to document, even for =
ISUP </FONT>
<BR><FONT SIZE=3D2>&gt; transmissions, the cases</FONT>
<BR><FONT SIZE=3D2>&gt; in which it is and is not appropriate. In the =
long term, </FONT>
<BR><FONT SIZE=3D2>&gt; though, I think we</FONT>
<BR><FONT SIZE=3D2>&gt; should just get rid of it entirely.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jon Peterson</FONT>
<BR><FONT SIZE=3D2>&gt; NeuStar, Inc.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sent: Thursday, December 26, 2002 12:12 =
PM</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Subject: [Sip] INFO considered =
harmful</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Folks,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; For some time, I've been complaining about =
the continued </FONT>
<BR><FONT SIZE=3D2>&gt; abuse of the </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; INFO method for things that should be done =
other ways </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; (frequently, not </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; using sip at all). I've written up a =
summary of the problems and a </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; proposed path forward. Specifically, I'd =
like to obsolete INFO and </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; replace it with a spec that is approved =
ONLY for SIP-T.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Until the I-D appears, you can pick up a =
version at:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-0=
0.txt" =
TARGET=3D"_blank">http://www.jdrosen.net/papers/draft-rosenberg-sip-info=
-harmful-00.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-0=
0.html" =
TARGET=3D"_blank">http://www.jdrosen.net/papers/draft-rosenberg-sip-info=
-harmful-00.html</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Thanks,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 72 Eagle Rock Ave.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&gt; &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; East =
Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; &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; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; <A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
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Subject: RE: Repost: [Sip] Buggy example in RFC 3261
Date: Thu, 2 Jan 2003 16:22:15 +0200
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Thread-Topic: Repost: [Sip] Buggy example in RFC 3261
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This is an extract from a discussion I had with Robert Sparks:

> >Yes, that was exactly what I asked. If its not so, what does an Expires
> >header in a 200 OK for a REGISTER mean?
>
>
>As I note above, with the current text an Expire header field has no meaning
>if it appears in a 200 response to a REGISTER request.

http://www.ietf.org/mail-archive/working-groups/sip/current/msg03602.html

Regards,
Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, January 02, 2003 7:01 AM
> To: Peter Päppinghaus
> Cc: sip@ietf.org
> Subject: Re: Repost: [Sip] Buggy example in RFC 3261
> 
> 
> Sorry for not responding. I did notice this. There is an 
> error here, as 
> you point out. What is NOT clear is whether the error is in 
> the example, 
> or in the normative text. For REGISTER requests, you can use 
> either the 
> Expires header field or the expires contact header field 
> parameter. It 
> seems odd that this same rule would not apply to REGISTER 
> responses. My 
> personal recollection was that either the Expires header field or the 
> expires contact param was OK for either request or response. This 
> doesn't match the normative text though.
> 
> Does anyone use the Expires header field and NOT the expires contact 
> param in REGISTER responses?
> 
> -Jonathan R.
> 
> Peter Päppinghaus wrote:
> > I have posted this earlier, but it seems to have gone
> > unnoticed.
> > 
> > Hi,
> > 
> > I believe to have stumbled on a buggy example in RFC 3261.
> > 
> > The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
> > an Expires header, but no expires parameter in the Contact header.
> > 
> > This violates RFC 3261, sec. 10.3, bullet 8:
> > "Each Contact value MUST feature an "expires" parameter
> >   indicating its expiration interval chosen by the registrar."
> > 
> > Regards,
> > 
> > Peter Paeppinghaus
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: Re: [Sip] A simple query...
References: <004401c2a85e$99855700$618d6b80@amer.cisco.com> <3E0375E7.1080608@dynamicsoft.com> <1040415671.949.161.camel@RjS.localdomain> <15888.48291.700746.59246@thomasm-u1.cisco.com> <15889.35402.270711.303036@lohi.eng.song.fi> <004401c2b0ce$627520b0$5d02120a@in.huawei.com> <3E11AB76.5080903@dynamicsoft.com> <003101c2b222$550924d0$5d02120a@in.huawei.com>
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inline.

Nataraju A.B. wrote:
> Hi,
>  
> Server           =  "Server" HCOLON server-val *(LWS server-val)
>  server-val       =  product / comment
> 
> product          =  token [SLASH product-version]
> 
> product-version  =  token
> 
> both product and product version are of type tokens.
> if two tokes are in the input what should be considered. I mean product 
> an dprod version OR two product values.
>  
>  for ex : Server: *HomeServer HS v2*
> in this case HS is prod version or product2.

Its a second product. Version requires a slash. As a result, the example 
in the SIP spec which you quoted would have v2 parse as a product, not a 
version. Thats fine. This field is only useful for display to humans or 
for string comparison, in which case it doesnt matter whether the "v2" 
is interpreted as a product version or a product.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Sachin Shenoy <sachins@netlab.hcltech.com>
Subject: Re: [Sip] Loop detection; Not a bad thing to have
Cc: sip@ietf.org
In-Reply-To: <3E11AC8D.5090700@dynamicsoft.com>
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Please find my comments inline.

>>I have to take your word on this (that loops are rare). But sure it 
>>creates havoc once
>>it happens. (Simple tests where request from UA looped between 2 proxies 
>>ran for more
>>than 10 minutes.
>
>That should not happen. Max-Forwards should be preventing that.
>
>>Where re transmissions of original INVITE from UA, the "Too Many hops" 
>>response and ACK for this just kept on circulating).
>
>The INVITE should not be retransmitted once the final response has arrived.
>
>It seems your problems are the result of faulty software, not an incorrect 
>approach to handling loops.

Yes indeed, as you said, the problem was due to faulty software (or wrong
configuration in this case). Mistake was on my part for having enabled full
fledged debug logging for this test. This slowed down the stack considerably.
When the tests were rerun, with logging turned off, it went through without
even the slightest hiccup. The whole transaction got completed in less than
a second. (UAC only re-transmitted the INVITE once which did not create
any problem).

Thanks
Sachin



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I fully agree with Dean on this one.

One point to consider (I'm on the fence on this one) is whether we would say that INFO is 100% application-specific.  That is, the WG will not publish standard body types.

A reason for doing that is to let people know that INFO really is just, as Dean points out, application-to-application communication.  If you want applications to interoperate, take it either to an Application Area WG or take it outside the IETF.

One reason to not do this is SIP-T should use a method other than INFO.  That said, I don't think there are that many SIP-T implementations extant.


> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, December 31, 2002 1:48 AM
> To: Jonathan Rosenberg; Orit Levin
> Cc: sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> >Orit Levin wrote:
> >>>The big huge difference I have been trying to point out is 
> that EVENTS
> >>>DEFINES SEMANTICS, whereas INFO doesnt.
> >>>MIME provides semantics and syntax that are independent of 
> the type.
> >>>SIP events provides semantics and syntax that are 
> independent of the
> >>>package.
> >>>INFO provides nothing.
> >>
> >>INFO provides "minimal" but very important semantics: 
> asynchronous data
> >>reliably follows the established SIP path.
> >
> >Huh?
> >
> >Every new SIP method is reliable, it inherits the SIP 
> transaction state 
> >machine for non-invite. Follows the established path? Every 
> in-dialog 
> >method would have this property, it is method independent. 
> >Asynchronous?  You can send any method at any time.
> >
> >Thus, nothing you have pointed out above is different 
> between INFO and any 
> >other new method we might introduce.
> 
> I think the key point is that there is a difference between transport 
> protocol semantic and application semantic. The transport 
> protocol semantic 
> expressed in INFO is "Here is some data that will be used by the 
> application. It is not important to the transport protocol, 
> except that the 
> transport protocol is expected to deliver it reliably". INFO 
> says NOTHING 
> about the APPLICATION level semantic -- that's up to what 
> goes IN the INFO 
> payloads. The question is -- do we try to rigidly define 
> application-level 
> semantics here? Do we define that there ARE no such 
> applications possible? 
> Or do we design a framework by which application-level 
> semantics can be 
> expressed outside of the protocol definition?
> 
> And as for "Just add another method to your SIP stack". Let's 
> say I have a 
> mobile phone and the SIP stack is burned into ROM and exposes only a 
> simplistic transactional API. This is only likely to happen, oh, 100 
> million times or so over the next two years . . .  Now, just 
> exactly how is 
> my BREW or JTME application going to go about extending the 
> SIP stack to 
> support another method? Will the evil wireless operator 
> networks even pass 
> another method? Neither is happening anytime soon, I think . . .
> 
> --
> Dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip



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What is the difference between:
	INFO-MUMBLE
	INFO-FOO

and
	MUMBLE
	FOO
???

I don't think we should go down the path of "if you method looks like INFO-%, it is special".  If you are going to parse the first few characters of the method name, you might as well parse the method name.

> -----Original Message-----
> From: Pete Cordell [mailto:pete@tech-know-ware.com]
> Sent: Wednesday, January 01, 2003 6:44 AM
> To: Jonathan Rosenberg; sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> How about we use new methods to define end-to-end application 
> data, but name
> the messages INFO-*.  e.g. INFO-MY-FEATURE, so that they can 
> be readily
> identified as application data rather than SIP state 
> affecting methods.
> Allow: can then be used to negotiate 'INFO' extensions.
> 
> 
> - It's easy to parse.
> 
> - It's very easy for a stack to treat it generically.
> 
> - It can reuse existing SIP features to do negotiation etc.  
> (e.g. Allow)
> 
> - It could be easily documented in the SIP usage document.
> 
> 
> 
> My other thoughts on the subject (and what has led me to the 
> above position
> are):
> 
> - I think its important to have some mechanism that allows end-to-end
> application data exchange that is independent of SIP state.
> 
> - Doing this in-band (rather than starting up potentially 
> numerous side
> connections) will be desirable for some applications (such as 
> those that use
> a small amounts of data).
> 
> - It seems appealing to be readily able to tell which methods 
> affect SIP
> state, and which is application level data.
> 
> - Require:, Supported: and Allow: already negotiation of 
> features and where
> possible it would be beneficial to re-use these mechanisms rather than
> invent new ones..
> 
> - An INFO mechanism does not need to offer any semantics 
> beyond simple data
> delivery (although it might be nice to have an in-order 
> delivery service?)
> (and the afore mentioned negotiation).
> 
> - The main problem with INFO is that it doesn't give a single 
> specific way
> of identifying which application a particular INFO method is 
> related to.
> Hence it necessitates some sort of heuristic procedure to be used.
> 
> 
> Some other ways to move ahead might be to:
> 
> - Specify that the application that the INFO corresponds to 
> is defined by
> the MIME type of the body.
> 
> - Define a new header that defines the application that the 
> INFO corresponds
> to.
> 
> Pete.
> 
> ----- Original Message -----
> From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
> To: <sip@ietf.org>
> Sent: 26 December 2002 20:12
> Subject: [Sip] INFO considered harmful
> 
> 
> > Folks,
> >
> > For some time, I've been complaining about the continued 
> abuse of the
> > INFO method for things that should be done other ways 
> (frequently, not
> > using sip at all). I've written up a summary of the problems and a
> > proposed path forward. Specifically, I'd like to obsolete INFO and
> > replace it with a spec that is approved ONLY for SIP-T.
> >
> > Until the I-D appears, you can pick up a version at:
> >
> > 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.txt
> > 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-info-harmful-00.html
> >
> > Thanks,
> > Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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References: <004401c2a85e$99855700$618d6b80@amer.cisco.com> <3E0375E7.1080608@dynamicsoft.com> <1040415671.949.161.camel@RjS.localdomain> <15888.48291.700746.59246@thomasm-u1.cisco.com> <15889.35402.270711.303036@lohi.eng.song.fi> <004401c2b0ce$627520b0$5d02120a@in.huawei.com> <3E11AB76.5080903@dynamicsoft.com> <003101c2b222$550924d0$5d02120a@in.huawei.com> <3E146845.5020707@dynamicsoft.com>
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One additional comment here. I do appreciate that the example is 
probably confusing, since you would think that "v2" is a version, but 
its not. I've logged that into bugzilla.

-Jonathan R.


Jonathan Rosenberg wrote:
> inline.
> 
> Nataraju A.B. wrote:
> 
>> Hi,
>>  
>> Server           =  "Server" HCOLON server-val *(LWS server-val)
>>  server-val       =  product / comment
>>
>> product          =  token [SLASH product-version]
>>
>> product-version  =  token
>>
>> both product and product version are of type tokens.
>> if two tokes are in the input what should be considered. I mean 
>> product an dprod version OR two product values.
>>  
>>  for ex : Server: *HomeServer HS v2*
>> in this case HS is prod version or product2.
> 
> 
> Its a second product. Version requires a slash. As a result, the example 
> in the SIP spec which you quoted would have v2 parse as a product, not a 
> version. Thats fine. This field is only useful for display to humans or 
> for string comparison, in which case it doesnt matter whether the "v2" 
> is interpreted as a product version or a product.
> 
> -Jonathan R.
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: [Sip] INFO considered harmful
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I don't get it. If you want "100% application
specific", why not pick an unused port and roll
your own protocol? Then all of the crufty SIPism
don't get in the way, and the hue and cry from
this working group ceases immediately. Do the bits 
on the wire keep warmer if they're dressed in SIP?

       Mike

Eric Burger writes:
 > I fully agree with Dean on this one.
 > 
 > One point to consider (I'm on the fence on this one) is whether we would say that INFO is 100% application-specific.  That is, the WG will not publish standard body types.
 > 
 > A reason for doing that is to let people know that INFO really is just, as Dean points out, application-to-application communication.  If you want applications to interoperate, take it either to an Application Area WG or take it outside the IETF.
 > 
 > One reason to not do this is SIP-T should use a method other than INFO.  That said, I don't think there are that many SIP-T implementations extant.
 > 
 > 
 > > -----Original Message-----
 > > From: Dean Willis [mailto:dean.willis@softarmor.com]
 > > Sent: Tuesday, December 31, 2002 1:48 AM
 > > To: Jonathan Rosenberg; Orit Levin
 > > Cc: sip@ietf.org
 > > Subject: Re: [Sip] INFO considered harmful
 > > 
 > > 
 > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
 > > >Orit Levin wrote:
 > > >>>The big huge difference I have been trying to point out is 
 > > that EVENTS
 > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
 > > >>>MIME provides semantics and syntax that are independent of 
 > > the type.
 > > >>>SIP events provides semantics and syntax that are 
 > > independent of the
 > > >>>package.
 > > >>>INFO provides nothing.
 > > >>
 > > >>INFO provides "minimal" but very important semantics: 
 > > asynchronous data
 > > >>reliably follows the established SIP path.
 > > >
 > > >Huh?
 > > >
 > > >Every new SIP method is reliable, it inherits the SIP 
 > > transaction state 
 > > >machine for non-invite. Follows the established path? Every 
 > > in-dialog 
 > > >method would have this property, it is method independent. 
 > > >Asynchronous?  You can send any method at any time.
 > > >
 > > >Thus, nothing you have pointed out above is different 
 > > between INFO and any 
 > > >other new method we might introduce.
 > > 
 > > I think the key point is that there is a difference between transport 
 > > protocol semantic and application semantic. The transport 
 > > protocol semantic 
 > > expressed in INFO is "Here is some data that will be used by the 
 > > application. It is not important to the transport protocol, 
 > > except that the 
 > > transport protocol is expected to deliver it reliably". INFO 
 > > says NOTHING 
 > > about the APPLICATION level semantic -- that's up to what 
 > > goes IN the INFO 
 > > payloads. The question is -- do we try to rigidly define 
 > > application-level 
 > > semantics here? Do we define that there ARE no such 
 > > applications possible? 
 > > Or do we design a framework by which application-level 
 > > semantics can be 
 > > expressed outside of the protocol definition?
 > > 
 > > And as for "Just add another method to your SIP stack". Let's 
 > > say I have a 
 > > mobile phone and the SIP stack is burned into ROM and exposes only a 
 > > simplistic transactional API. This is only likely to happen, oh, 100 
 > > million times or so over the next two years . . .  Now, just 
 > > exactly how is 
 > > my BREW or JTME application going to go about extending the 
 > > SIP stack to 
 > > support another method? Will the evil wireless operator 
 > > networks even pass 
 > > another method? Neither is happening anytime soon, I think . . .
 > > 
 > > --
 > > Dean
 > > 
 > > 
 > > _______________________________________________
 > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > This list is for NEW development of the core SIP Protocol
 > > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > Use sipping@ietf.org for new developments on the application of sip
 > > 
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  2 13:56:37 2003
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From: Gordon Ledgard <gledgard@iperia.com>
To: "'Michael Thomas'" <mat@cisco.com>, Eric Burger <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 13:53:45 -0500 
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I think people are suffering from the delusion that because SIP
has already established a channel of communication between the
transport layer of the SIP entities on the involved machines, that
it then becomes "easier" to get information to the application layer
by tunneling it through INFO on this established channel, and that 
you gain something by the "handshakes" provided "free" by SIP.

In fact, nothing at all has been gained. All of your application
connection problems remain. Michael is right in saying it provides 
no such benefit, and it makes NO difference from an app perspective,
if you tunnel junk through INFO or roll you own protocol over
another port.

Which is not to say INFO isn't useful.


  Gordon

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, January 02, 2003 12:53 PM
> To: Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?
> 
>        Mike
> 
> Eric Burger writes:
>  > I fully agree with Dean on this one.
>  > 
>  > One point to consider (I'm on the fence on this one) is 
> whether we would say that INFO is 100% application-specific.  
> That is, the WG will not publish standard body types.
>  > 
>  > A reason for doing that is to let people know that INFO 
> really is just, as Dean points out, 
> application-to-application communication.  If you want 
> applications to interoperate, take it either to an 
> Application Area WG or take it outside the IETF.
>  > 
>  > One reason to not do this is SIP-T should use a method 
> other than INFO.  That said, I don't think there are that 
> many SIP-T implementations extant.
>  > 
>  > 
>  > > -----Original Message-----
>  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > To: Jonathan Rosenberg; Orit Levin
>  > > Cc: sip@ietf.org
>  > > Subject: Re: [Sip] INFO considered harmful
>  > > 
>  > > 
>  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >Orit Levin wrote:
>  > > >>>The big huge difference I have been trying to point out is 
>  > > that EVENTS
>  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >>>MIME provides semantics and syntax that are independent of 
>  > > the type.
>  > > >>>SIP events provides semantics and syntax that are 
>  > > independent of the
>  > > >>>package.
>  > > >>>INFO provides nothing.
>  > > >>
>  > > >>INFO provides "minimal" but very important semantics: 
>  > > asynchronous data
>  > > >>reliably follows the established SIP path.
>  > > >
>  > > >Huh?
>  > > >
>  > > >Every new SIP method is reliable, it inherits the SIP 
>  > > transaction state 
>  > > >machine for non-invite. Follows the established path? Every 
>  > > in-dialog 
>  > > >method would have this property, it is method independent. 
>  > > >Asynchronous?  You can send any method at any time.
>  > > >
>  > > >Thus, nothing you have pointed out above is different 
>  > > between INFO and any 
>  > > >other new method we might introduce.
>  > > 
>  > > I think the key point is that there is a difference 
> between transport 
>  > > protocol semantic and application semantic. The transport 
>  > > protocol semantic 
>  > > expressed in INFO is "Here is some data that will be used by the 
>  > > application. It is not important to the transport protocol, 
>  > > except that the 
>  > > transport protocol is expected to deliver it reliably". INFO 
>  > > says NOTHING 
>  > > about the APPLICATION level semantic -- that's up to what 
>  > > goes IN the INFO 
>  > > payloads. The question is -- do we try to rigidly define 
>  > > application-level 
>  > > semantics here? Do we define that there ARE no such 
>  > > applications possible? 
>  > > Or do we design a framework by which application-level 
>  > > semantics can be 
>  > > expressed outside of the protocol definition?
>  > > 
>  > > And as for "Just add another method to your SIP stack". Let's 
>  > > say I have a 
>  > > mobile phone and the SIP stack is burned into ROM and 
> exposes only a 
>  > > simplistic transactional API. This is only likely to 
> happen, oh, 100 
>  > > million times or so over the next two years . . .  Now, just 
>  > > exactly how is 
>  > > my BREW or JTME application going to go about extending the 
>  > > SIP stack to 
>  > > support another method? Will the evil wireless operator 
>  > > networks even pass 
>  > > another method? Neither is happening anytime soon, I think . . .
>  > > 
>  > > --
>  > > Dean
>  > > 
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP Protocol
>  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > Use sipping@ietf.org for new developments on the 
> application of sip
>  > > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  2 13:57:18 2003
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From: "George Foti (LMC)" <George.Foti@ericsson.ca>
To: "'Michael Thomas'" <mat@cisco.com>, Eric Burger <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 13:55:13 -0500 
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The application is SIP based, we dont need yet another protocol to convey what needs to be conveyed e2e.
No need to go around the bush.
Lets find *real* arguments against using INFO for that purpopse rather than philosphical  ones.

/gf




-----Original Message-----
From: Michael Thomas [mailto:mat@cisco.com]
Sent: Thursday, January 02, 2003 12:53 PM
To: Eric Burger
Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
Subject: RE: [Sip] INFO considered harmful



I don't get it. If you want "100% application
specific", why not pick an unused port and roll
your own protocol? Then all of the crufty SIPism
don't get in the way, and the hue and cry from
this working group ceases immediately. Do the bits 
on the wire keep warmer if they're dressed in SIP?

       Mike

Eric Burger writes:
 > I fully agree with Dean on this one.
 > 
 > One point to consider (I'm on the fence on this one) is whether we would say that INFO is 100% application-specific.  That is, the WG will not publish standard body types.
 > 
 > A reason for doing that is to let people know that INFO really is just, as Dean points out, application-to-application communication.  If you want applications to interoperate, take it either to an Application Area WG or take it outside the IETF.
 > 
 > One reason to not do this is SIP-T should use a method other than INFO.  That said, I don't think there are that many SIP-T implementations extant.
 > 
 > 
 > > -----Original Message-----
 > > From: Dean Willis [mailto:dean.willis@softarmor.com]
 > > Sent: Tuesday, December 31, 2002 1:48 AM
 > > To: Jonathan Rosenberg; Orit Levin
 > > Cc: sip@ietf.org
 > > Subject: Re: [Sip] INFO considered harmful
 > > 
 > > 
 > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
 > > >Orit Levin wrote:
 > > >>>The big huge difference I have been trying to point out is 
 > > that EVENTS
 > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
 > > >>>MIME provides semantics and syntax that are independent of 
 > > the type.
 > > >>>SIP events provides semantics and syntax that are 
 > > independent of the
 > > >>>package.
 > > >>>INFO provides nothing.
 > > >>
 > > >>INFO provides "minimal" but very important semantics: 
 > > asynchronous data
 > > >>reliably follows the established SIP path.
 > > >
 > > >Huh?
 > > >
 > > >Every new SIP method is reliable, it inherits the SIP 
 > > transaction state 
 > > >machine for non-invite. Follows the established path? Every 
 > > in-dialog 
 > > >method would have this property, it is method independent. 
 > > >Asynchronous?  You can send any method at any time.
 > > >
 > > >Thus, nothing you have pointed out above is different 
 > > between INFO and any 
 > > >other new method we might introduce.
 > > 
 > > I think the key point is that there is a difference between transport 
 > > protocol semantic and application semantic. The transport 
 > > protocol semantic 
 > > expressed in INFO is "Here is some data that will be used by the 
 > > application. It is not important to the transport protocol, 
 > > except that the 
 > > transport protocol is expected to deliver it reliably". INFO 
 > > says NOTHING 
 > > about the APPLICATION level semantic -- that's up to what 
 > > goes IN the INFO 
 > > payloads. The question is -- do we try to rigidly define 
 > > application-level 
 > > semantics here? Do we define that there ARE no such 
 > > applications possible? 
 > > Or do we design a framework by which application-level 
 > > semantics can be 
 > > expressed outside of the protocol definition?
 > > 
 > > And as for "Just add another method to your SIP stack". Let's 
 > > say I have a 
 > > mobile phone and the SIP stack is burned into ROM and exposes only a 
 > > simplistic transactional API. This is only likely to happen, oh, 100 
 > > million times or so over the next two years . . .  Now, just 
 > > exactly how is 
 > > my BREW or JTME application going to go about extending the 
 > > SIP stack to 
 > > support another method? Will the evil wireless operator 
 > > networks even pass 
 > > another method? Neither is happening anytime soon, I think . . .
 > > 
 > > --
 > > Dean
 > > 
 > > 
 > > _______________________________________________
 > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > This list is for NEW development of the core SIP Protocol
 > > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > Use sipping@ietf.org for new developments on the application of sip
 > > 
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  2 14:38:11 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        "'Michael Thomas'"
	 <mat@cisco.com>,
        Eric Burger <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 13:35:38 -0600 
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The argument of "but I'm already holding a hammer and *want*
to treat everything like a nail!" comes up on about an annual
basis, and is rejected outright each time.

Just because you have a SIP stack in your application does
*not* mean that it is appropriate to use that stack for
all general network access. You're allowing laziness to
overwhelm good application design.

/a

> -----Original Message-----
> From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> Sent: Thursday, January 02, 2003 12:55
> To: 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The application is SIP based, we dont need yet another 
> protocol to convey what needs to be conveyed e2e.
> No need to go around the bush.
> Lets find *real* arguments against using INFO for that 
> purpopse rather than philosphical  ones.
> 
> /gf
> 
> 
> 
> 
> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, January 02, 2003 12:53 PM
> To: Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?
> 
>        Mike
> 
> Eric Burger writes:
>  > I fully agree with Dean on this one.
>  > 
>  > One point to consider (I'm on the fence on this one) is 
> whether we would say that INFO is 100% application-specific.  
> That is, the WG will not publish standard body types.
>  > 
>  > A reason for doing that is to let people know that INFO 
> really is just, as Dean points out, 
> application-to-application communication.  If you want 
> applications to interoperate, take it either to an 
> Application Area WG or take it outside the IETF.
>  > 
>  > One reason to not do this is SIP-T should use a method 
> other than INFO.  That said, I don't think there are that 
> many SIP-T implementations extant.
>  > 
>  > 
>  > > -----Original Message-----
>  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > To: Jonathan Rosenberg; Orit Levin
>  > > Cc: sip@ietf.org
>  > > Subject: Re: [Sip] INFO considered harmful
>  > > 
>  > > 
>  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >Orit Levin wrote:
>  > > >>>The big huge difference I have been trying to point out is 
>  > > that EVENTS
>  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >>>MIME provides semantics and syntax that are independent of 
>  > > the type.
>  > > >>>SIP events provides semantics and syntax that are 
>  > > independent of the
>  > > >>>package.
>  > > >>>INFO provides nothing.
>  > > >>
>  > > >>INFO provides "minimal" but very important semantics: 
>  > > asynchronous data
>  > > >>reliably follows the established SIP path.
>  > > >
>  > > >Huh?
>  > > >
>  > > >Every new SIP method is reliable, it inherits the SIP 
>  > > transaction state 
>  > > >machine for non-invite. Follows the established path? Every 
>  > > in-dialog 
>  > > >method would have this property, it is method independent. 
>  > > >Asynchronous?  You can send any method at any time.
>  > > >
>  > > >Thus, nothing you have pointed out above is different 
>  > > between INFO and any 
>  > > >other new method we might introduce.
>  > > 
>  > > I think the key point is that there is a difference 
> between transport 
>  > > protocol semantic and application semantic. The transport 
>  > > protocol semantic 
>  > > expressed in INFO is "Here is some data that will be used by the 
>  > > application. It is not important to the transport protocol, 
>  > > except that the 
>  > > transport protocol is expected to deliver it reliably". INFO 
>  > > says NOTHING 
>  > > about the APPLICATION level semantic -- that's up to what 
>  > > goes IN the INFO 
>  > > payloads. The question is -- do we try to rigidly define 
>  > > application-level 
>  > > semantics here? Do we define that there ARE no such 
>  > > applications possible? 
>  > > Or do we design a framework by which application-level 
>  > > semantics can be 
>  > > expressed outside of the protocol definition?
>  > > 
>  > > And as for "Just add another method to your SIP stack". Let's 
>  > > say I have a 
>  > > mobile phone and the SIP stack is burned into ROM and 
> exposes only a 
>  > > simplistic transactional API. This is only likely to 
> happen, oh, 100 
>  > > million times or so over the next two years . . .  Now, just 
>  > > exactly how is 
>  > > my BREW or JTME application going to go about extending the 
>  > > SIP stack to 
>  > > support another method? Will the evil wireless operator 
>  > > networks even pass 
>  > > another method? Neither is happening anytime soon, I think . . .
>  > > 
>  > > --
>  > > Dean
>  > > 
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP Protocol
>  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > Use sipping@ietf.org for new developments on the 
> application of sip
>  > > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: RE: [Sip] Loop detection; Not a bad thing to have
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> From: Sachin Shenoy [mailto:sachins@netlab.hcltech.com]
> > > I have to take your word on this (that loops are rare). 
> But sure it
> > > creates havoc once
> > > it happens. (Simple tests where request from UA looped 
> between 2 proxies
> > > ran for more
> > > than 10 minutes.
> >
> > That should not happen. Max-Forwards should be preventing that.
> 
> Max-Forwards did work in this case. But didn't prevent the 
> above mentioned
> problem.

It took you *10* *MINUTES* to perform *70* *HOPS*?!?!?

I think you need to seriously work on getting your proxies above
0.1 tps before pursuing this particular problem much further.

/a
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From: "Don Stanwyck" <don@stanwyck.com>
To: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        "'Michael Thomas'" <mat@cisco.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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I guess I have two problems with the statement below.

1) Applications should not be protocol based - they may be protocol
aware.  Unless the SIP stack is rolled into the application, it merely
uses SIP as a lower layer tool to establish communications.  It is a
violation of good design to roll the protocol into the application
itself.
   BTW, if the SIP stack is rolled into the application, then when the
next edition of SIP comes along you will have to rewrite the app.  You
really should have an API that keeps the app from needing to know the
details of the communication establishment protocol.

2) If the apps need to talk directly to each other there are a host of
protocols that are extremely trivial that can do the job.  The apps can
even just open a port over any transport protocol and exchange
information as they please.  There is no reason to bother all the
routing proxies in the network with your e2e private chatter.

Just a thought.
don

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of George Foti (LMC)
> Sent: Thursday, January 02, 2003 11:55 AM
> To: 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The application is SIP based, we dont need yet another 
> protocol to convey what needs to be conveyed e2e. No need to 
> go around the bush. Lets find *real* arguments against using 
> INFO for that purpopse rather than philosphical  ones.
> 
> /gf
> 
> 
> 
> 
> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, January 02, 2003 12:53 PM
> To: Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?
> 
>        Mike
> 
> Eric Burger writes:
>  > I fully agree with Dean on this one.
>  > 
>  > One point to consider (I'm on the fence on this one) is 
> whether we would say that INFO is 100% application-specific.  
> That is, the WG will not publish standard body types.  > 
>  > A reason for doing that is to let people know that INFO 
> really is just, as Dean points out, 
> application-to-application communication.  If you want 
> applications to interoperate, take it either to an 
> Application Area WG or take it outside the IETF.  > 
>  > One reason to not do this is SIP-T should use a method 
> other than INFO.  That said, I don't think there are that 
> many SIP-T implementations extant.  > 
>  > 
>  > > -----Original Message-----
>  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > To: Jonathan Rosenberg; Orit Levin
>  > > Cc: sip@ietf.org
>  > > Subject: Re: [Sip] INFO considered harmful
>  > > 
>  > > 
>  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >Orit Levin wrote:
>  > > >>>The big huge difference I have been trying to point out is 
>  > > that EVENTS
>  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >>>MIME provides semantics and syntax that are independent of 
>  > > the type.
>  > > >>>SIP events provides semantics and syntax that are 
>  > > independent of the
>  > > >>>package.
>  > > >>>INFO provides nothing.
>  > > >>
>  > > >>INFO provides "minimal" but very important semantics: 
>  > > asynchronous data
>  > > >>reliably follows the established SIP path.
>  > > >
>  > > >Huh?
>  > > >
>  > > >Every new SIP method is reliable, it inherits the SIP 
>  > > transaction state 
>  > > >machine for non-invite. Follows the established path? Every 
>  > > in-dialog 
>  > > >method would have this property, it is method independent. 
>  > > >Asynchronous?  You can send any method at any time.
>  > > >
>  > > >Thus, nothing you have pointed out above is different 
>  > > between INFO and any 
>  > > >other new method we might introduce.
>  > > 
>  > > I think the key point is that there is a difference 
> between transport 
>  > > protocol semantic and application semantic. The transport 
>  > > protocol semantic 
>  > > expressed in INFO is "Here is some data that will be used by the 
>  > > application. It is not important to the transport protocol, 
>  > > except that the 
>  > > transport protocol is expected to deliver it reliably". INFO 
>  > > says NOTHING 
>  > > about the APPLICATION level semantic -- that's up to what 
>  > > goes IN the INFO 
>  > > payloads. The question is -- do we try to rigidly define 
>  > > application-level 
>  > > semantics here? Do we define that there ARE no such 
>  > > applications possible? 
>  > > Or do we design a framework by which application-level 
>  > > semantics can be 
>  > > expressed outside of the protocol definition?
>  > > 
>  > > And as for "Just add another method to your SIP stack". Let's 
>  > > say I have a 
>  > > mobile phone and the SIP stack is burned into ROM and 
> exposes only a 
>  > > simplistic transactional API. This is only likely to 
> happen, oh, 100 
>  > > million times or so over the next two years . . .  Now, just 
>  > > exactly how is 
>  > > my BREW or JTME application going to go about extending the 
>  > > SIP stack to 
>  > > support another method? Will the evil wireless operator 
>  > > networks even pass 
>  > > another method? Neither is happening anytime soon, I 
> think . . .  > > 
>  > > --
>  > > Dean
>  > > 
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP 
> Protocol  > > Use sip-implementors@cs.columbia.edu for 
> questions on current sip  > > Use sipping@ietf.org for new 
> developments on the application of sip  > > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on 
> current sip  > Use sipping@ietf.org for new developments on 
> the application of sip _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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



From mailnull@www1.ietf.org  Thu Jan  2 15:36:11 2003
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Date: Thu, 02 Jan 2003 15:33:04 -0500
To: sip@ietf.org
From: "Joel M. Halpern" <joel@stevecrocker.com>
Subject: RE: [Sip] INFO considered harmful
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I think that there is another way to phrase the issue which at least raises 
a fair question.
I have an application.  It uses SIP, and then (sometimes) establishes a 
media session.
However, none of those exchanges provide the IP address of the controlling 
remote application.

Hence, if one application control entity wants to send (or receive) 
information to (from) the other application control entity, the only 
clearly available path is through SIP messages.  Preferably SIP messages 
tied to the existing session.
Presumably, one could mandate a single header for carrying a "control 
entity IP address / port / protocol" but that seems pretty silly.

Yours,
Joel M. Halpern

At 01:35 PM 1/2/2003 -0600, Adam Roach wrote:
>The argument of "but I'm already holding a hammer and *want*
>to treat everything like a nail!" comes up on about an annual
>basis, and is rejected outright each time.
>
>Just because you have a SIP stack in your application does
>*not* mean that it is appropriate to use that stack for
>all general network access. You're allowing laziness to
>overwhelm good application design.
>
>/a
>
> > -----Original Message-----
> > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> > Sent: Thursday, January 02, 2003 12:55
> > To: 'Michael Thomas'; Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> >
> >
> > The application is SIP based, we dont need yet another
> > protocol to convey what needs to be conveyed e2e.
> > No need to go around the bush.
> > Lets find *real* arguments against using INFO for that
> > purpopse rather than philosphical  ones.
> >
> > /gf
> >
> >
> >
> >
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Thursday, January 02, 2003 12:53 PM
> > To: Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> >
> >
> >
> > I don't get it. If you want "100% application
> > specific", why not pick an unused port and roll
> > your own protocol? Then all of the crufty SIPism
> > don't get in the way, and the hue and cry from
> > this working group ceases immediately. Do the bits
> > on the wire keep warmer if they're dressed in SIP?
> >
> >        Mike
> >
> > Eric Burger writes:
> >  > I fully agree with Dean on this one.
> >  >
> >  > One point to consider (I'm on the fence on this one) is
> > whether we would say that INFO is 100% application-specific.
> > That is, the WG will not publish standard body types.
> >  >
> >  > A reason for doing that is to let people know that INFO
> > really is just, as Dean points out,
> > application-to-application communication.  If you want
> > applications to interoperate, take it either to an
> > Application Area WG or take it outside the IETF.
> >  >
> >  > One reason to not do this is SIP-T should use a method
> > other than INFO.  That said, I don't think there are that
> > many SIP-T implementations extant.
> >  >
> >  >
> >  > > -----Original Message-----
> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> >  > > To: Jonathan Rosenberg; Orit Levin
> >  > > Cc: sip@ietf.org
> >  > > Subject: Re: [Sip] INFO considered harmful
> >  > >
> >  > >
> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> >  > > >Orit Levin wrote:
> >  > > >>>The big huge difference I have been trying to point out is
> >  > > that EVENTS
> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> >  > > >>>MIME provides semantics and syntax that are independent of
> >  > > the type.
> >  > > >>>SIP events provides semantics and syntax that are
> >  > > independent of the
> >  > > >>>package.
> >  > > >>>INFO provides nothing.
> >  > > >>
> >  > > >>INFO provides "minimal" but very important semantics:
> >  > > asynchronous data
> >  > > >>reliably follows the established SIP path.
> >  > > >
> >  > > >Huh?
> >  > > >
> >  > > >Every new SIP method is reliable, it inherits the SIP
> >  > > transaction state
> >  > > >machine for non-invite. Follows the established path? Every
> >  > > in-dialog
> >  > > >method would have this property, it is method independent.
> >  > > >Asynchronous?  You can send any method at any time.
> >  > > >
> >  > > >Thus, nothing you have pointed out above is different
> >  > > between INFO and any
> >  > > >other new method we might introduce.
> >  > >
> >  > > I think the key point is that there is a difference
> > between transport
> >  > > protocol semantic and application semantic. The transport
> >  > > protocol semantic
> >  > > expressed in INFO is "Here is some data that will be used by the
> >  > > application. It is not important to the transport protocol,
> >  > > except that the
> >  > > transport protocol is expected to deliver it reliably". INFO
> >  > > says NOTHING
> >  > > about the APPLICATION level semantic -- that's up to what
> >  > > goes IN the INFO
> >  > > payloads. The question is -- do we try to rigidly define
> >  > > application-level
> >  > > semantics here? Do we define that there ARE no such
> >  > > applications possible?
> >  > > Or do we design a framework by which application-level
> >  > > semantics can be
> >  > > expressed outside of the protocol definition?
> >  > >
> >  > > And as for "Just add another method to your SIP stack". Let's
> >  > > say I have a
> >  > > mobile phone and the SIP stack is burned into ROM and
> > exposes only a
> >  > > simplistic transactional API. This is only likely to
> > happen, oh, 100
> >  > > million times or so over the next two years . . .  Now, just
> >  > > exactly how is
> >  > > my BREW or JTME application going to go about extending the
> >  > > SIP stack to
> >  > > support another method? Will the evil wireless operator
> >  > > networks even pass
> >  > > another method? Neither is happening anytime soon, I think . . .
> >  > >
> >  > > --
> >  > > Dean
> >  > >
> >  > >
> >  > > _______________________________________________
> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > > This list is for NEW development of the core SIP Protocol
> >  > > Use sip-implementors@cs.columbia.edu for questions on current sip
> >  > > Use sipping@ietf.org for new developments on the
> > application of sip
> >  > >
> >  > _______________________________________________
> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > This list is for NEW development of the core SIP Protocol
> >  > Use sip-implementors@cs.columbia.edu for questions on current sip
> >  > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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



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Date: Thu, 02 Jan 2003 15:45:30 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Joel,

I am not entirely sure I follow you here. As I started reading, I 
thought you were proposing what I do believe is the right answer here.

If people want to exchange proprietary data between applications that 
are set up using sip, we have a way to do that. This application data 
can just be considered another session type (along with audio and video) 
which SIP can set up and manage for you. So, roll your own protocol 
(say, foo) that does just what you want, and then use SDP to exchange 
the IP/ports and protocol identification infomration needed to make it work.

This way, proxies don't see any of it, and SIP is not being abused in 
any way.

-Jonathan R.

Joel M. Halpern wrote:
> I think that there is another way to phrase the issue which at least 
> raises a fair question.
> I have an application.  It uses SIP, and then (sometimes) establishes a 
> media session.
> However, none of those exchanges provide the IP address of the 
> controlling remote application.
> 
> Hence, if one application control entity wants to send (or receive) 
> information to (from) the other application control entity, the only 
> clearly available path is through SIP messages.  Preferably SIP messages 
> tied to the existing session.
> Presumably, one could mandate a single header for carrying a "control 
> entity IP address / port / protocol" but that seems pretty silly.
> 
> Yours,
> Joel M. Halpern
> 
> At 01:35 PM 1/2/2003 -0600, Adam Roach wrote:
> 
>> The argument of "but I'm already holding a hammer and *want*
>> to treat everything like a nail!" comes up on about an annual
>> basis, and is rejected outright each time.
>>
>> Just because you have a SIP stack in your application does
>> *not* mean that it is appropriate to use that stack for
>> all general network access. You're allowing laziness to
>> overwhelm good application design.
>>
>> /a
>>
>> > -----Original Message-----
>> > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
>> > Sent: Thursday, January 02, 2003 12:55
>> > To: 'Michael Thomas'; Eric Burger
>> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>> > Subject: RE: [Sip] INFO considered harmful
>> >
>> >
>> > The application is SIP based, we dont need yet another
>> > protocol to convey what needs to be conveyed e2e.
>> > No need to go around the bush.
>> > Lets find *real* arguments against using INFO for that
>> > purpopse rather than philosphical  ones.
>> >
>> > /gf
>> >
>> >
>> >
>> >
>> > -----Original Message-----
>> > From: Michael Thomas [mailto:mat@cisco.com]
>> > Sent: Thursday, January 02, 2003 12:53 PM
>> > To: Eric Burger
>> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>> > Subject: RE: [Sip] INFO considered harmful
>> >
>> >
>> >
>> > I don't get it. If you want "100% application
>> > specific", why not pick an unused port and roll
>> > your own protocol? Then all of the crufty SIPism
>> > don't get in the way, and the hue and cry from
>> > this working group ceases immediately. Do the bits
>> > on the wire keep warmer if they're dressed in SIP?
>> >
>> >        Mike
>> >
>> > Eric Burger writes:
>> >  > I fully agree with Dean on this one.
>> >  >
>> >  > One point to consider (I'm on the fence on this one) is
>> > whether we would say that INFO is 100% application-specific.
>> > That is, the WG will not publish standard body types.
>> >  >
>> >  > A reason for doing that is to let people know that INFO
>> > really is just, as Dean points out,
>> > application-to-application communication.  If you want
>> > applications to interoperate, take it either to an
>> > Application Area WG or take it outside the IETF.
>> >  >
>> >  > One reason to not do this is SIP-T should use a method
>> > other than INFO.  That said, I don't think there are that
>> > many SIP-T implementations extant.
>> >  >
>> >  >
>> >  > > -----Original Message-----
>> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
>> >  > > To: Jonathan Rosenberg; Orit Levin
>> >  > > Cc: sip@ietf.org
>> >  > > Subject: Re: [Sip] INFO considered harmful
>> >  > >
>> >  > >
>> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>> >  > > >Orit Levin wrote:
>> >  > > >>>The big huge difference I have been trying to point out is
>> >  > > that EVENTS
>> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>> >  > > >>>MIME provides semantics and syntax that are independent of
>> >  > > the type.
>> >  > > >>>SIP events provides semantics and syntax that are
>> >  > > independent of the
>> >  > > >>>package.
>> >  > > >>>INFO provides nothing.
>> >  > > >>
>> >  > > >>INFO provides "minimal" but very important semantics:
>> >  > > asynchronous data
>> >  > > >>reliably follows the established SIP path.
>> >  > > >
>> >  > > >Huh?
>> >  > > >
>> >  > > >Every new SIP method is reliable, it inherits the SIP
>> >  > > transaction state
>> >  > > >machine for non-invite. Follows the established path? Every
>> >  > > in-dialog
>> >  > > >method would have this property, it is method independent.
>> >  > > >Asynchronous?  You can send any method at any time.
>> >  > > >
>> >  > > >Thus, nothing you have pointed out above is different
>> >  > > between INFO and any
>> >  > > >other new method we might introduce.
>> >  > >
>> >  > > I think the key point is that there is a difference
>> > between transport
>> >  > > protocol semantic and application semantic. The transport
>> >  > > protocol semantic
>> >  > > expressed in INFO is "Here is some data that will be used by the
>> >  > > application. It is not important to the transport protocol,
>> >  > > except that the
>> >  > > transport protocol is expected to deliver it reliably". INFO
>> >  > > says NOTHING
>> >  > > about the APPLICATION level semantic -- that's up to what
>> >  > > goes IN the INFO
>> >  > > payloads. The question is -- do we try to rigidly define
>> >  > > application-level
>> >  > > semantics here? Do we define that there ARE no such
>> >  > > applications possible?
>> >  > > Or do we design a framework by which application-level
>> >  > > semantics can be
>> >  > > expressed outside of the protocol definition?
>> >  > >
>> >  > > And as for "Just add another method to your SIP stack". Let's
>> >  > > say I have a
>> >  > > mobile phone and the SIP stack is burned into ROM and
>> > exposes only a
>> >  > > simplistic transactional API. This is only likely to
>> > happen, oh, 100
>> >  > > million times or so over the next two years . . .  Now, just
>> >  > > exactly how is
>> >  > > my BREW or JTME application going to go about extending the
>> >  > > SIP stack to
>> >  > > support another method? Will the evil wireless operator
>> >  > > networks even pass
>> >  > > another method? Neither is happening anytime soon, I think . . .
>> >  > >
>> >  > > --
>> >  > > Dean
>> >  > >
>> >  > >
>> >  > > _______________________________________________
>> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> >  > > This list is for NEW development of the core SIP Protocol
>> >  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>> >  > > Use sipping@ietf.org for new developments on the
>> > application of sip
>> >  > >
>> >  > _______________________________________________
>> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> >  > This list is for NEW development of the core SIP Protocol
>> >  > Use sip-implementors@cs.columbia.edu for questions on current sip
>> >  > Use sipping@ietf.org for new developments on the application of sip
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol
>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol
>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>> >
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



From mailnull@www1.ietf.org  Thu Jan  2 16:00:14 2003
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Date: Thu, 02 Jan 2003 15:56:53 -0500
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: "Joel M. Halpern" <joel@stevecrocker.com>
Subject: Re: [Sip] INFO considered harmful
Cc: sip@ietf.org
In-Reply-To: <3E14A4EA.8060409@dynamicsoft.com>
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Using an SDP extension to set up the control entity communication path 
would be reasonable.  My point is that simply saying "the applications can 
already communicate" (as some respondents on the list have said) is not 
true, as the normal usage does not permit the entities to communicate.
If the communication is going to be extended, it would seem that the SDP 
based approach is correct.
However, it would seem sensible to have a method to carry smaller pieces of 
information.
This could possible be done with alternative content, or with extended 
headers.  In either case, it would be for the low rate communication that 
would not warrant and SDP exchange to establish the session.  Obviously, 
this is a slippery slope.  We do not want to turn the control plane into a 
generic communication channel.  (I've been there, and been unhappy doing that.)

Yours,
Joel

At 03:45 PM 1/2/2003 -0500, Jonathan Rosenberg wrote:
>Joel,
>
>I am not entirely sure I follow you here. As I started reading, I thought 
>you were proposing what I do believe is the right answer here.
>
>If people want to exchange proprietary data between applications that are 
>set up using sip, we have a way to do that. This application data can just 
>be considered another session type (along with audio and video) which SIP 
>can set up and manage for you. So, roll your own protocol (say, foo) that 
>does just what you want, and then use SDP to exchange the IP/ports and 
>protocol identification infomration needed to make it work.
>
>This way, proxies don't see any of it, and SIP is not being abused in any way.
>
>-Jonathan R.
>
>Joel M. Halpern wrote:
>>I think that there is another way to phrase the issue which at least 
>>raises a fair question.
>>I have an application.  It uses SIP, and then (sometimes) establishes a 
>>media session.
>>However, none of those exchanges provide the IP address of the 
>>controlling remote application.
>>Hence, if one application control entity wants to send (or receive) 
>>information to (from) the other application control entity, the only 
>>clearly available path is through SIP messages.  Preferably SIP messages 
>>tied to the existing session.
>>Presumably, one could mandate a single header for carrying a "control 
>>entity IP address / port / protocol" but that seems pretty silly.
>>Yours,
>>Joel M. Halpern
>>At 01:35 PM 1/2/2003 -0600, Adam Roach wrote:
>>
>>>The argument of "but I'm already holding a hammer and *want*
>>>to treat everything like a nail!" comes up on about an annual
>>>basis, and is rejected outright each time.
>>>
>>>Just because you have a SIP stack in your application does
>>>*not* mean that it is appropriate to use that stack for
>>>all general network access. You're allowing laziness to
>>>overwhelm good application design.
>>>
>>>/a
>>>
>>> > -----Original Message-----
>>> > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
>>> > Sent: Thursday, January 02, 2003 12:55
>>> > To: 'Michael Thomas'; Eric Burger
>>> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>>> > Subject: RE: [Sip] INFO considered harmful
>>> >
>>> >
>>> > The application is SIP based, we dont need yet another
>>> > protocol to convey what needs to be conveyed e2e.
>>> > No need to go around the bush.
>>> > Lets find *real* arguments against using INFO for that
>>> > purpopse rather than philosphical  ones.
>>> >
>>> > /gf
>>> >
>>> >
>>> >
>>> >
>>> > -----Original Message-----
>>> > From: Michael Thomas [mailto:mat@cisco.com]
>>> > Sent: Thursday, January 02, 2003 12:53 PM
>>> > To: Eric Burger
>>> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>>> > Subject: RE: [Sip] INFO considered harmful
>>> >
>>> >
>>> >
>>> > I don't get it. If you want "100% application
>>> > specific", why not pick an unused port and roll
>>> > your own protocol? Then all of the crufty SIPism
>>> > don't get in the way, and the hue and cry from
>>> > this working group ceases immediately. Do the bits
>>> > on the wire keep warmer if they're dressed in SIP?
>>> >
>>> >        Mike
>>> >
>>> > Eric Burger writes:
>>> >  > I fully agree with Dean on this one.
>>> >  >
>>> >  > One point to consider (I'm on the fence on this one) is
>>> > whether we would say that INFO is 100% application-specific.
>>> > That is, the WG will not publish standard body types.
>>> >  >
>>> >  > A reason for doing that is to let people know that INFO
>>> > really is just, as Dean points out,
>>> > application-to-application communication.  If you want
>>> > applications to interoperate, take it either to an
>>> > Application Area WG or take it outside the IETF.
>>> >  >
>>> >  > One reason to not do this is SIP-T should use a method
>>> > other than INFO.  That said, I don't think there are that
>>> > many SIP-T implementations extant.
>>> >  >
>>> >  >
>>> >  > > -----Original Message-----
>>> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>>> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
>>> >  > > To: Jonathan Rosenberg; Orit Levin
>>> >  > > Cc: sip@ietf.org
>>> >  > > Subject: Re: [Sip] INFO considered harmful
>>> >  > >
>>> >  > >
>>> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>>> >  > > >Orit Levin wrote:
>>> >  > > >>>The big huge difference I have been trying to point out is
>>> >  > > that EVENTS
>>> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>>> >  > > >>>MIME provides semantics and syntax that are independent of
>>> >  > > the type.
>>> >  > > >>>SIP events provides semantics and syntax that are
>>> >  > > independent of the
>>> >  > > >>>package.
>>> >  > > >>>INFO provides nothing.
>>> >  > > >>
>>> >  > > >>INFO provides "minimal" but very important semantics:
>>> >  > > asynchronous data
>>> >  > > >>reliably follows the established SIP path.
>>> >  > > >
>>> >  > > >Huh?
>>> >  > > >
>>> >  > > >Every new SIP method is reliable, it inherits the SIP
>>> >  > > transaction state
>>> >  > > >machine for non-invite. Follows the established path? Every
>>> >  > > in-dialog
>>> >  > > >method would have this property, it is method independent.
>>> >  > > >Asynchronous?  You can send any method at any time.
>>> >  > > >
>>> >  > > >Thus, nothing you have pointed out above is different
>>> >  > > between INFO and any
>>> >  > > >other new method we might introduce.
>>> >  > >
>>> >  > > I think the key point is that there is a difference
>>> > between transport
>>> >  > > protocol semantic and application semantic. The transport
>>> >  > > protocol semantic
>>> >  > > expressed in INFO is "Here is some data that will be used by the
>>> >  > > application. It is not important to the transport protocol,
>>> >  > > except that the
>>> >  > > transport protocol is expected to deliver it reliably". INFO
>>> >  > > says NOTHING
>>> >  > > about the APPLICATION level semantic -- that's up to what
>>> >  > > goes IN the INFO
>>> >  > > payloads. The question is -- do we try to rigidly define
>>> >  > > application-level
>>> >  > > semantics here? Do we define that there ARE no such
>>> >  > > applications possible?
>>> >  > > Or do we design a framework by which application-level
>>> >  > > semantics can be
>>> >  > > expressed outside of the protocol definition?
>>> >  > >
>>> >  > > And as for "Just add another method to your SIP stack". Let's
>>> >  > > say I have a
>>> >  > > mobile phone and the SIP stack is burned into ROM and
>>> > exposes only a
>>> >  > > simplistic transactional API. This is only likely to
>>> > happen, oh, 100
>>> >  > > million times or so over the next two years . . .  Now, just
>>> >  > > exactly how is
>>> >  > > my BREW or JTME application going to go about extending the
>>> >  > > SIP stack to
>>> >  > > support another method? Will the evil wireless operator
>>> >  > > networks even pass
>>> >  > > another method? Neither is happening anytime soon, I think . . .
>>> >  > >
>>> >  > > --
>>> >  > > Dean
>>> >  > >
>>> >  > >
>>> >  > > _______________________________________________
>>> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> >  > > This list is for NEW development of the core SIP Protocol
>>> >  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>>> >  > > Use sipping@ietf.org for new developments on the
>>> > application of sip
>>> >  > >
>>> >  > _______________________________________________
>>> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> >  > This list is for NEW development of the core SIP Protocol
>>> >  > Use sip-implementors@cs.columbia.edu for questions on current sip
>>> >  > Use sipping@ietf.org for new developments on the application of sip
>>> > _______________________________________________
>>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> > This list is for NEW development of the core SIP Protocol
>>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>>> > Use sipping@ietf.org for new developments on the application of sip
>>> > _______________________________________________
>>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> > This list is for NEW development of the core SIP Protocol
>>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>>> > Use sipping@ietf.org for new developments on the application of sip
>>> >
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP Protocol
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sipping@ietf.org for new developments on the application of sip
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>
>--
>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.jdrosen.net                      PHONE: (973) 952-5000
>http://www.dynamicsoft.com


_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip



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Frank W. Miller wrote:
> Greetings,
> 
> I would take issue with the basic premise of the draft, which is that the
> SIP and SIPPING groups need to have control over every bit that passes
> through a SIP message.  It is not possible for the standard to anticpate
> every situation that people might want to use the protocol.  It's
> interesting how many times I've heard WG members say, "thats not what I
> intended it for".  The beauty of SIP is that its basic framework and
> semantic level endear it to so many different signaling situations.  The use
> of INFO to supplement the corners of specific applications is a very
> important part of that versatility.

It is the job of the IETF to define protocols that interoperate, that 
scale well, that have good security, that don't congest the internet, 
and so on. Sometimes, there is a need for vendor proprietary extensions 
to such protocols. However, all vendor extensions need to live within a 
"sandbox" that limits the scope of what can be done. Specifically, that 
sandbox has to draw the line around issues of scale, security, and 
congestion control, which affect *other* networks and other vendors. 
IETF protocols are used on networks where other things live too.

The problem with INFO is that it defines no such sandbox. It allows 
anything, and can be used in ways which will cause problems when those 
messages flow through other vendors equipment or run on other networks. 
The clearest example of that is the rate. A record-routing proxy in some 
other domain might get a flood of INFO messages because some app in 
someone elses server thought INFO was a good way to tunnel voice.

Let me take this to the extreme. By your argument, TCP should allow for 
vendor-specified plug ins for congestion control. After all, why should 
IETF prevent me from using something that works well for MY application? 
Indeed, there is lots of research on different congestion control 
algorithms. TCP should allow those to be used! But, would you really 
want someone to be able to plug in a congestion control algorithm that 
basically doesn't do congestion control algorithm? I should hope not.



> 
> JR cites three specific problems with INFO:
> 
> Lack of interoperability: This is not a problem.  Since the INFO contents
> are explicitly outside the standard, this must be assumed by the
> implementor.  If they persist with INFO to implement some signaling
> situtations, there are likely other good reasons for doing so.

You have not properly interpreted my draft. THe interoperability failure 
is that there is know what to know the intent of the INFO method. There 
is no way to signal that this is the "frank's extension usage" as 
opposed to anyone elses.

> 
> Inappropriateness: As I've already stated, I do not feel that the SIP and
> SIPPING working groups should (let alone can) realistically expect that they
> can determine how and when the protocol will be applied in applications at
> this late date.

Please see my above comments on this. I believe that this is a 
fundamental issue about the role of IETF in protocols.

> 
> Incorrectness: Jonathan is eluding to the possiblity of errors being present
> in the protocol extensions.  Well, this is not the SIP and SIPPING groups
> problem.  These extensions are outside the scope of the standard.  If the
> purveyors of the extension design the error, they're responsible for it, not
> the SIP and SIPPING groups.

Except where those extensions impact other servers, vendors or service 
providers. And they can.


> 
> As has already been noted in this thread, there are a number of fine
> implementations (including Dynamicsoft) in the field, many, if not all,
> including a punt to the application of the INFO body.  

Frank, pointing a finger at me and saying "but your company does it!" is 
neither a compelling argument or, IMHO, a civilized way to conduct a 
discussion on the list. It is frequently the case that ones company does 
something that one does not like, because a customer asked, or a partner 
is doing it, or because of one reason or another. Does that mean that we 
should be promoting such things as IETF standards?


> This is very useful
> functionality to implementors and as such it makes very little sense to
> deprecate this method.  Another will take its place as the abused.  For
> example, I've had discussions with people about doing extensions using
> SUBSCRIBE/NOTIFY that had nothing whatever to do with events just because of
> the "political incorrectness" of INFO. 

Unlike INFO, SUB/NOT defines a very well-scoped and limited repetoire of 
behavior for packages. So, while someone may violate the spec and do 
something inappropriate, it would not be IETF compliant. WIth INFO, one 
can do bad things and still claim to be compliant to the specifications.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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inline.

Dean Willis wrote:
> At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> 
>> Orit Levin wrote:
>>
>>>> The big huge difference I have been trying to point out is that EVENTS
>>>> DEFINES SEMANTICS, whereas INFO doesnt.
>>>> MIME provides semantics and syntax that are independent of the type.
>>>> SIP events provides semantics and syntax that are independent of the
>>>> package.
>>>> INFO provides nothing.
>>>
>>>
>>> INFO provides "minimal" but very important semantics: asynchronous data
>>> reliably follows the established SIP path.
>>
>>
>> Huh?
>>
>> Every new SIP method is reliable, it inherits the SIP transaction 
>> state machine for non-invite. Follows the established path? Every 
>> in-dialog method would have this property, it is method independent. 
>> Asynchronous?  You can send any method at any time.
>>
>> Thus, nothing you have pointed out above is different between INFO and 
>> any other new method we might introduce.
> 
> 
> I think the key point is that there is a difference between transport 
> protocol semantic and application semantic. The transport protocol 
> semantic expressed in INFO is "Here is some data that will be used by 
> the application. It is not important to the transport protocol, except 
> that the transport protocol is expected to deliver it reliably". 

Just one moment. SIP is not a transport protocol as its defined in my 
book. Although SIP is generic in that it can support lots of 
applications (conferencing, IVR, gaming, etc.), it is not unbounded. 
SIP's messages provide semantics common to all applications which 
require person-to-person communications. That does NOT make it a 
transport prorotocol for arbitrary data communication.



> And as for "Just add another method to your SIP stack". Let's say I have 
> a mobile phone and the SIP stack is burned into ROM and exposes only a 
> simplistic transactional API. This is only likely to happen, oh, 100 
> million times or so over the next two years . . .  Now, just exactly how 
> is my BREW or JTME application going to go about extending the SIP stack 
> to support another method? Will the evil wireless operator networks even 
> pass another method? Neither is happening anytime soon, I think . . .

The stack has to allow extensibility to pass unknown INFO bodies up to 
the application for handling. Why this is any different than having 
extensibility to handle unknown methods up to the application, is a 
mystery to me.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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To: "'Don Stanwyck'" <don@stanwyck.com>,
        "George Foti (LMC)"
	 <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
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Cc: "'Dean Willis'" <dean.willis@softarmor.com>,
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	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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>There is no reason to bother all the
>routing proxies in the network with your e2e private chatter.

How are they bothered ?
What extra logic do they need?

/gf


-----Original Message-----
From: Don Stanwyck [mailto:don@stanwyck.com]
Sent: Thursday, January 02, 2003 3:13 PM
To: 'George Foti (LMC)'; 'Michael Thomas'; 'Eric Burger'
Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
Subject: RE: [Sip] INFO considered harmful


I guess I have two problems with the statement below.

1) Applications should not be protocol based - they may be protocol
aware.  Unless the SIP stack is rolled into the application, it merely
uses SIP as a lower layer tool to establish communications.  It is a
violation of good design to roll the protocol into the application
itself.
   BTW, if the SIP stack is rolled into the application, then when the
next edition of SIP comes along you will have to rewrite the app.  You
really should have an API that keeps the app from needing to know the
details of the communication establishment protocol.

2) If the apps need to talk directly to each other there are a host of
protocols that are extremely trivial that can do the job.  The apps can
even just open a port over any transport protocol and exchange
information as they please.  There is no reason to bother all the
routing proxies in the network with your e2e private chatter.

Just a thought.
don

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of George Foti (LMC)
> Sent: Thursday, January 02, 2003 11:55 AM
> To: 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The application is SIP based, we dont need yet another 
> protocol to convey what needs to be conveyed e2e. No need to 
> go around the bush. Lets find *real* arguments against using 
> INFO for that purpopse rather than philosphical  ones.
> 
> /gf
> 
> 
> 
> 
> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, January 02, 2003 12:53 PM
> To: Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?
> 
>        Mike
> 
> Eric Burger writes:
>  > I fully agree with Dean on this one.
>  > 
>  > One point to consider (I'm on the fence on this one) is 
> whether we would say that INFO is 100% application-specific.  
> That is, the WG will not publish standard body types.  > 
>  > A reason for doing that is to let people know that INFO 
> really is just, as Dean points out, 
> application-to-application communication.  If you want 
> applications to interoperate, take it either to an 
> Application Area WG or take it outside the IETF.  > 
>  > One reason to not do this is SIP-T should use a method 
> other than INFO.  That said, I don't think there are that 
> many SIP-T implementations extant.  > 
>  > 
>  > > -----Original Message-----
>  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > To: Jonathan Rosenberg; Orit Levin
>  > > Cc: sip@ietf.org
>  > > Subject: Re: [Sip] INFO considered harmful
>  > > 
>  > > 
>  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >Orit Levin wrote:
>  > > >>>The big huge difference I have been trying to point out is 
>  > > that EVENTS
>  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >>>MIME provides semantics and syntax that are independent of 
>  > > the type.
>  > > >>>SIP events provides semantics and syntax that are 
>  > > independent of the
>  > > >>>package.
>  > > >>>INFO provides nothing.
>  > > >>
>  > > >>INFO provides "minimal" but very important semantics: 
>  > > asynchronous data
>  > > >>reliably follows the established SIP path.
>  > > >
>  > > >Huh?
>  > > >
>  > > >Every new SIP method is reliable, it inherits the SIP 
>  > > transaction state 
>  > > >machine for non-invite. Follows the established path? Every 
>  > > in-dialog 
>  > > >method would have this property, it is method independent. 
>  > > >Asynchronous?  You can send any method at any time.
>  > > >
>  > > >Thus, nothing you have pointed out above is different 
>  > > between INFO and any 
>  > > >other new method we might introduce.
>  > > 
>  > > I think the key point is that there is a difference 
> between transport 
>  > > protocol semantic and application semantic. The transport 
>  > > protocol semantic 
>  > > expressed in INFO is "Here is some data that will be used by the 
>  > > application. It is not important to the transport protocol, 
>  > > except that the 
>  > > transport protocol is expected to deliver it reliably". INFO 
>  > > says NOTHING 
>  > > about the APPLICATION level semantic -- that's up to what 
>  > > goes IN the INFO 
>  > > payloads. The question is -- do we try to rigidly define 
>  > > application-level 
>  > > semantics here? Do we define that there ARE no such 
>  > > applications possible? 
>  > > Or do we design a framework by which application-level 
>  > > semantics can be 
>  > > expressed outside of the protocol definition?
>  > > 
>  > > And as for "Just add another method to your SIP stack". Let's 
>  > > say I have a 
>  > > mobile phone and the SIP stack is burned into ROM and 
> exposes only a 
>  > > simplistic transactional API. This is only likely to 
> happen, oh, 100 
>  > > million times or so over the next two years . . .  Now, just 
>  > > exactly how is 
>  > > my BREW or JTME application going to go about extending the 
>  > > SIP stack to 
>  > > support another method? Will the evil wireless operator 
>  > > networks even pass 
>  > > another method? Neither is happening anytime soon, I 
> think . . .  > > 
>  > > --
>  > > Dean
>  > > 
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP 
> Protocol  > > Use sip-implementors@cs.columbia.edu for 
> questions on current sip  > > Use sipping@ietf.org for new 
> developments on the application of sip  > > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on 
> current sip  > Use sipping@ietf.org for new developments on 
> the application of sip _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Message-ID: <32CD630F6CBED411AE180008C7894CBC0C037BFD@lmc37.lmc.ericsson.se>
From: "George Foti (LMC)" <George.Foti@ericsson.ca>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "George Foti (LMC)"
	 <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
        Eric Burger <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 17:04:42 -0500 
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This should not be a philosophical discusion.
If a new SIP method is okay, why is INFO not okay.

What is the difference?
Please explain in technical terms?

/gf

-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Thursday, January 02, 2003 2:36 PM
To: 'George Foti (LMC)'; 'Michael Thomas'; Eric Burger
Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
Subject: RE: [Sip] INFO considered harmful


The argument of "but I'm already holding a hammer and *want*
to treat everything like a nail!" comes up on about an annual
basis, and is rejected outright each time.

Just because you have a SIP stack in your application does
*not* mean that it is appropriate to use that stack for
all general network access. You're allowing laziness to
overwhelm good application design.

/a

> -----Original Message-----
> From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> Sent: Thursday, January 02, 2003 12:55
> To: 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The application is SIP based, we dont need yet another 
> protocol to convey what needs to be conveyed e2e.
> No need to go around the bush.
> Lets find *real* arguments against using INFO for that 
> purpopse rather than philosphical  ones.
> 
> /gf
> 
> 
> 
> 
> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Thursday, January 02, 2003 12:53 PM
> To: Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> 
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?
> 
>        Mike
> 
> Eric Burger writes:
>  > I fully agree with Dean on this one.
>  > 
>  > One point to consider (I'm on the fence on this one) is 
> whether we would say that INFO is 100% application-specific.  
> That is, the WG will not publish standard body types.
>  > 
>  > A reason for doing that is to let people know that INFO 
> really is just, as Dean points out, 
> application-to-application communication.  If you want 
> applications to interoperate, take it either to an 
> Application Area WG or take it outside the IETF.
>  > 
>  > One reason to not do this is SIP-T should use a method 
> other than INFO.  That said, I don't think there are that 
> many SIP-T implementations extant.
>  > 
>  > 
>  > > -----Original Message-----
>  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > To: Jonathan Rosenberg; Orit Levin
>  > > Cc: sip@ietf.org
>  > > Subject: Re: [Sip] INFO considered harmful
>  > > 
>  > > 
>  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >Orit Levin wrote:
>  > > >>>The big huge difference I have been trying to point out is 
>  > > that EVENTS
>  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >>>MIME provides semantics and syntax that are independent of 
>  > > the type.
>  > > >>>SIP events provides semantics and syntax that are 
>  > > independent of the
>  > > >>>package.
>  > > >>>INFO provides nothing.
>  > > >>
>  > > >>INFO provides "minimal" but very important semantics: 
>  > > asynchronous data
>  > > >>reliably follows the established SIP path.
>  > > >
>  > > >Huh?
>  > > >
>  > > >Every new SIP method is reliable, it inherits the SIP 
>  > > transaction state 
>  > > >machine for non-invite. Follows the established path? Every 
>  > > in-dialog 
>  > > >method would have this property, it is method independent. 
>  > > >Asynchronous?  You can send any method at any time.
>  > > >
>  > > >Thus, nothing you have pointed out above is different 
>  > > between INFO and any 
>  > > >other new method we might introduce.
>  > > 
>  > > I think the key point is that there is a difference 
> between transport 
>  > > protocol semantic and application semantic. The transport 
>  > > protocol semantic 
>  > > expressed in INFO is "Here is some data that will be used by the 
>  > > application. It is not important to the transport protocol, 
>  > > except that the 
>  > > transport protocol is expected to deliver it reliably". INFO 
>  > > says NOTHING 
>  > > about the APPLICATION level semantic -- that's up to what 
>  > > goes IN the INFO 
>  > > payloads. The question is -- do we try to rigidly define 
>  > > application-level 
>  > > semantics here? Do we define that there ARE no such 
>  > > applications possible? 
>  > > Or do we design a framework by which application-level 
>  > > semantics can be 
>  > > expressed outside of the protocol definition?
>  > > 
>  > > And as for "Just add another method to your SIP stack". Let's 
>  > > say I have a 
>  > > mobile phone and the SIP stack is burned into ROM and 
> exposes only a 
>  > > simplistic transactional API. This is only likely to 
> happen, oh, 100 
>  > > million times or so over the next two years . . .  Now, just 
>  > > exactly how is 
>  > > my BREW or JTME application going to go about extending the 
>  > > SIP stack to 
>  > > support another method? Will the evil wireless operator 
>  > > networks even pass 
>  > > another method? Neither is happening anytime soon, I think . . .
>  > > 
>  > > --
>  > > Dean
>  > > 
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP Protocol
>  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > Use sipping@ietf.org for new developments on the 
> application of sip
>  > > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  2 17:59:12 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        Adam Roach
	 <adam@dynamicsoft.com>,
        "'Michael Thomas'" <mat@cisco.com>,
        Eric Burger
	 <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 16:56:30 -0600 
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A new method would be okay for new SIP-related functionality.

But that's not what you proposed.

You said, "The application is SIP based, we dont need yet
another protocol to convey what needs to be conveyed e2e,"
which is an argument that we should use SIP as a general
purpose RPC mechanism for any application that already has
a SIP stack.

Doing what you propose with INFO is inappropriate.

Doing what you propose with a new method is inappropriate.

Doing what you propose with SIP is inappropriate.

Since you keep harping on technical, as opposed to
philosophical, arguments, here's a couple:

  - The SIP protocol was designed for session initiation,
    negotiation, and teardown. The timing characteristics
    are not generally appropriate for application control.
    For example, with typical SIP networks, latency and
    jitter of messages are too high to provide any sort
    of reasonable user experience for floor control
    or DTMF transport.

 - Many proxies stay in the signalling path for the purposes
   of providing services. By sending unnecessary application-
   level information throught these proxies, you reduce the
   capacity and increase possible congestion of the network.

I could go on, but this topic has had extensive airing on
the list in years past. There's really no reason to beat
this particular dead horse any longer.

/a

> -----Original Message-----
> From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> Sent: Thursday, January 02, 2003 16:05
> To: 'Adam Roach'; George Foti (LMC); 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> This should not be a philosophical discusion.
> If a new SIP method is okay, why is INFO not okay.
> 
> What is the difference?
> Please explain in technical terms?
> 
> /gf
> 
> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Thursday, January 02, 2003 2:36 PM
> To: 'George Foti (LMC)'; 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The argument of "but I'm already holding a hammer and *want*
> to treat everything like a nail!" comes up on about an annual
> basis, and is rejected outright each time.
> 
> Just because you have a SIP stack in your application does
> *not* mean that it is appropriate to use that stack for
> all general network access. You're allowing laziness to
> overwhelm good application design.
> 
> /a
> 
> > -----Original Message-----
> > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> > Sent: Thursday, January 02, 2003 12:55
> > To: 'Michael Thomas'; Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > The application is SIP based, we dont need yet another 
> > protocol to convey what needs to be conveyed e2e.
> > No need to go around the bush.
> > Lets find *real* arguments against using INFO for that 
> > purpopse rather than philosphical  ones.
> > 
> > /gf
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Thursday, January 02, 2003 12:53 PM
> > To: Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > 
> > I don't get it. If you want "100% application
> > specific", why not pick an unused port and roll
> > your own protocol? Then all of the crufty SIPism
> > don't get in the way, and the hue and cry from
> > this working group ceases immediately. Do the bits 
> > on the wire keep warmer if they're dressed in SIP?
> > 
> >        Mike
> > 
> > Eric Burger writes:
> >  > I fully agree with Dean on this one.
> >  > 
> >  > One point to consider (I'm on the fence on this one) is 
> > whether we would say that INFO is 100% application-specific.  
> > That is, the WG will not publish standard body types.
> >  > 
> >  > A reason for doing that is to let people know that INFO 
> > really is just, as Dean points out, 
> > application-to-application communication.  If you want 
> > applications to interoperate, take it either to an 
> > Application Area WG or take it outside the IETF.
> >  > 
> >  > One reason to not do this is SIP-T should use a method 
> > other than INFO.  That said, I don't think there are that 
> > many SIP-T implementations extant.
> >  > 
> >  > 
> >  > > -----Original Message-----
> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> >  > > To: Jonathan Rosenberg; Orit Levin
> >  > > Cc: sip@ietf.org
> >  > > Subject: Re: [Sip] INFO considered harmful
> >  > > 
> >  > > 
> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> >  > > >Orit Levin wrote:
> >  > > >>>The big huge difference I have been trying to point out is 
> >  > > that EVENTS
> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> >  > > >>>MIME provides semantics and syntax that are independent of 
> >  > > the type.
> >  > > >>>SIP events provides semantics and syntax that are 
> >  > > independent of the
> >  > > >>>package.
> >  > > >>>INFO provides nothing.
> >  > > >>
> >  > > >>INFO provides "minimal" but very important semantics: 
> >  > > asynchronous data
> >  > > >>reliably follows the established SIP path.
> >  > > >
> >  > > >Huh?
> >  > > >
> >  > > >Every new SIP method is reliable, it inherits the SIP 
> >  > > transaction state 
> >  > > >machine for non-invite. Follows the established path? Every 
> >  > > in-dialog 
> >  > > >method would have this property, it is method independent. 
> >  > > >Asynchronous?  You can send any method at any time.
> >  > > >
> >  > > >Thus, nothing you have pointed out above is different 
> >  > > between INFO and any 
> >  > > >other new method we might introduce.
> >  > > 
> >  > > I think the key point is that there is a difference 
> > between transport 
> >  > > protocol semantic and application semantic. The transport 
> >  > > protocol semantic 
> >  > > expressed in INFO is "Here is some data that will be 
> used by the 
> >  > > application. It is not important to the transport protocol, 
> >  > > except that the 
> >  > > transport protocol is expected to deliver it reliably". INFO 
> >  > > says NOTHING 
> >  > > about the APPLICATION level semantic -- that's up to what 
> >  > > goes IN the INFO 
> >  > > payloads. The question is -- do we try to rigidly define 
> >  > > application-level 
> >  > > semantics here? Do we define that there ARE no such 
> >  > > applications possible? 
> >  > > Or do we design a framework by which application-level 
> >  > > semantics can be 
> >  > > expressed outside of the protocol definition?
> >  > > 
> >  > > And as for "Just add another method to your SIP stack". Let's 
> >  > > say I have a 
> >  > > mobile phone and the SIP stack is burned into ROM and 
> > exposes only a 
> >  > > simplistic transactional API. This is only likely to 
> > happen, oh, 100 
> >  > > million times or so over the next two years . . .  Now, just 
> >  > > exactly how is 
> >  > > my BREW or JTME application going to go about extending the 
> >  > > SIP stack to 
> >  > > support another method? Will the evil wireless operator 
> >  > > networks even pass 
> >  > > another method? Neither is happening anytime soon, I 
> think . . .
> >  > > 
> >  > > --
> >  > > Dean
> >  > > 
> >  > > 
> >  > > _______________________________________________
> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > > This list is for NEW development of the core SIP Protocol
> >  > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> >  > > Use sipping@ietf.org for new developments on the 
> > application of sip
> >  > > 
> >  > _______________________________________________
> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > This list is for NEW development of the core SIP Protocol
> >  > Use sip-implementors@cs.columbia.edu for questions on current sip
> >  > Use sipping@ietf.org for new developments on the 
> application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Thu, 2 Jan 2003 16:02:05 -0700
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From: "Don Stanwyck" <don@stanwyck.com>
To: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        "'George Foti (LMC)'" <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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It increases the number of transactions they have to process.  Nothing
in life is free.  SIP Proxies, like other call routing devices, assume a
certain number of transactions per call as part of their design numbers.
This is important for traffic engineering.

If we allow SIP to become the new FTP protocol (and there is nothing
right now that would stop a UA from using SIP-INFO for FTP or FTP like
functions), then the calls/second counts and all other traffic
engineering rules that are needed for proper network operation become
meaningless.

Don

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of George Foti (LMC)
> Sent: Thursday, January 02, 2003 3:03 PM
> To: 'Don Stanwyck'; George Foti (LMC); 'Michael Thomas'; 'Eric Burger'
> Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> >There is no reason to bother all the
> >routing proxies in the network with your e2e private chatter.
> 
> How are they bothered ?
> What extra logic do they need?
> 
> /gf
> 
> 
> -----Original Message-----
> From: Don Stanwyck [mailto:don@stanwyck.com]
> Sent: Thursday, January 02, 2003 3:13 PM
> To: 'George Foti (LMC)'; 'Michael Thomas'; 'Eric Burger'
> Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> I guess I have two problems with the statement below.
> 
> 1) Applications should not be protocol based - they may be 
> protocol aware.  Unless the SIP stack is rolled into the 
> application, it merely uses SIP as a lower layer tool to 
> establish communications.  It is a violation of good design 
> to roll the protocol into the application itself.
>    BTW, if the SIP stack is rolled into the application, then 
> when the next edition of SIP comes along you will have to 
> rewrite the app.  You really should have an API that keeps 
> the app from needing to know the details of the communication 
> establishment protocol.
> 
> 2) If the apps need to talk directly to each other there are 
> a host of protocols that are extremely trivial that can do 
> the job.  The apps can even just open a port over any 
> transport protocol and exchange information as they please.  
> There is no reason to bother all the routing proxies in the 
> network with your e2e private chatter.
> 
> Just a thought.
> don
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of George Foti (LMC)
> > Sent: Thursday, January 02, 2003 11:55 AM
> > To: 'Michael Thomas'; Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > The application is SIP based, we dont need yet another
> > protocol to convey what needs to be conveyed e2e. No need to 
> > go around the bush. Lets find *real* arguments against using 
> > INFO for that purpopse rather than philosphical  ones.
> > 
> > /gf
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Thursday, January 02, 2003 12:53 PM
> > To: Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > 
> > I don't get it. If you want "100% application
> > specific", why not pick an unused port and roll
> > your own protocol? Then all of the crufty SIPism
> > don't get in the way, and the hue and cry from
> > this working group ceases immediately. Do the bits
> > on the wire keep warmer if they're dressed in SIP?
> > 
> >        Mike
> > 
> > Eric Burger writes:
> >  > I fully agree with Dean on this one.
> >  >
> >  > One point to consider (I'm on the fence on this one) is 
> > whether we would say that INFO is 100% application-specific.  
> > That is, the WG will not publish standard body types.  > 
> >  > A reason for doing that is to let people know that INFO 
> > really is just, as Dean points out, 
> > application-to-application communication.  If you want 
> > applications to interoperate, take it either to an 
> > Application Area WG or take it outside the IETF.  > 
> >  > One reason to not do this is SIP-T should use a method 
> > other than INFO.  That said, I don't think there are that 
> > many SIP-T implementations extant.  > 
> >  > 
> >  > > -----Original Message-----
> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> >  > > To: Jonathan Rosenberg; Orit Levin
> >  > > Cc: sip@ietf.org
> >  > > Subject: Re: [Sip] INFO considered harmful
> >  > > 
> >  > > 
> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> >  > > >Orit Levin wrote:
> >  > > >>>The big huge difference I have been trying to point out is 
> >  > > that EVENTS
> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> >  > > >>>MIME provides semantics and syntax that are independent of 
> >  > > the type.
> >  > > >>>SIP events provides semantics and syntax that are 
> >  > > independent of the
> >  > > >>>package.
> >  > > >>>INFO provides nothing.
> >  > > >>
> >  > > >>INFO provides "minimal" but very important semantics: 
> >  > > asynchronous data
> >  > > >>reliably follows the established SIP path.
> >  > > >
> >  > > >Huh?
> >  > > >
> >  > > >Every new SIP method is reliable, it inherits the SIP 
> >  > > transaction state 
> >  > > >machine for non-invite. Follows the established path? Every 
> >  > > in-dialog 
> >  > > >method would have this property, it is method independent. 
> >  > > >Asynchronous?  You can send any method at any time.
> >  > > >
> >  > > >Thus, nothing you have pointed out above is different 
> >  > > between INFO and any 
> >  > > >other new method we might introduce.
> >  > > 
> >  > > I think the key point is that there is a difference 
> > between transport 
> >  > > protocol semantic and application semantic. The transport 
> >  > > protocol semantic 
> >  > > expressed in INFO is "Here is some data that will be 
> used by the 
> >  > > application. It is not important to the transport protocol, 
> >  > > except that the 
> >  > > transport protocol is expected to deliver it reliably". INFO 
> >  > > says NOTHING 
> >  > > about the APPLICATION level semantic -- that's up to what 
> >  > > goes IN the INFO 
> >  > > payloads. The question is -- do we try to rigidly define 
> >  > > application-level 
> >  > > semantics here? Do we define that there ARE no such 
> >  > > applications possible? 
> >  > > Or do we design a framework by which application-level 
> >  > > semantics can be 
> >  > > expressed outside of the protocol definition?
> >  > > 
> >  > > And as for "Just add another method to your SIP stack". Let's 
> >  > > say I have a 
> >  > > mobile phone and the SIP stack is burned into ROM and 
> > exposes only a 
> >  > > simplistic transactional API. This is only likely to 
> > happen, oh, 100 
> >  > > million times or so over the next two years . . .  Now, just 
> >  > > exactly how is 
> >  > > my BREW or JTME application going to go about extending the 
> >  > > SIP stack to 
> >  > > support another method? Will the evil wireless operator 
> >  > > networks even pass 
> >  > > another method? Neither is happening anytime soon, I 
> > think . . .  > > 
> >  > > --
> >  > > Dean
> >  > > 
> >  > > 
> >  > > _______________________________________________
> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > > This list is for NEW development of the core SIP 
> > Protocol  > > Use sip-implementors@cs.columbia.edu for 
> > questions on current sip  > > Use sipping@ietf.org for new 
> > developments on the application of sip  > > 
> >  > _______________________________________________
> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > This list is for NEW development of the core SIP Protocol
> >  > Use sip-implementors@cs.columbia.edu for questions on 
> > current sip  > Use sipping@ietf.org for new developments on 
> > the application of sip 
> _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current 
> > sip Use sipping@ietf.org for new developments on the 
> > application of sip _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current 
> > sip Use sipping@ietf.org for new developments on the 
> > application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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



From mailnull@www1.ietf.org  Thu Jan  2 18:08:20 2003
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Date: Thu, 2 Jan 2003 16:04:57 -0700
Message-ID: <006e01c2b2b3$5efc2c30$0300000a@DONGWLAPTOP>
From: "Don Stanwyck" <don@stanwyck.com>
To: "'Don Stanwyck'" <don@stanwyck.com>,
        "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        "'George Foti (LMC)'" <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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To follow up my own posting (bad form, I know), I thought I should
mention a real life example that worries me.

Many new cell phones have picture taking and picture sending
capabilities.  Some even allow for "movie" modes that send a picture per
second of so.

If the call is set up using SIP and the phone then wants to send all
these photos also over SIP, then the SIP proxies in the network can
become overloaded with what should be a low priority traffic instead of
being used for their design purpose - to process communication session
establishment information.

Don



> -----Original Message-----
> From: Don Stanwyck [mailto:don@stanwyck.com] 
> Sent: Thursday, January 02, 2003 4:02 PM
> To: 'George Foti (LMC)'; 'George Foti (LMC)'; 'Michael 
> Thomas'; 'Eric Burger'
> Cc: 'Dean Willis'; 'Jonathan Rosenberg'; 'sip@ietf.org'
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> It increases the number of transactions they have to process. 
>  Nothing in life is free.  SIP Proxies, like other call 
> routing devices, assume a certain number of transactions per 
> call as part of their design numbers.  This is important for 
> traffic engineering.
> 
> If we allow SIP to become the new FTP protocol (and there is 
> nothing right now that would stop a UA from using SIP-INFO 
> for FTP or FTP like functions), then the calls/second counts 
> and all other traffic engineering rules that are needed for 
> proper network operation become meaningless.
> 
> Don
> 
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of George Foti (LMC)
> > Sent: Thursday, January 02, 2003 3:03 PM
> > To: 'Don Stanwyck'; George Foti (LMC); 'Michael Thomas'; 
> 'Eric Burger'
> > Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > >There is no reason to bother all the
> > >routing proxies in the network with your e2e private chatter.
> > 
> > How are they bothered ?
> > What extra logic do they need?
> > 
> > /gf
> > 
> > 
> > -----Original Message-----
> > From: Don Stanwyck [mailto:don@stanwyck.com]
> > Sent: Thursday, January 02, 2003 3:13 PM
> > To: 'George Foti (LMC)'; 'Michael Thomas'; 'Eric Burger'
> > Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > I guess I have two problems with the statement below.
> > 
> > 1) Applications should not be protocol based - they may be
> > protocol aware.  Unless the SIP stack is rolled into the 
> > application, it merely uses SIP as a lower layer tool to 
> > establish communications.  It is a violation of good design 
> > to roll the protocol into the application itself.
> >    BTW, if the SIP stack is rolled into the application, then 
> > when the next edition of SIP comes along you will have to 
> > rewrite the app.  You really should have an API that keeps 
> > the app from needing to know the details of the communication 
> > establishment protocol.
> > 
> > 2) If the apps need to talk directly to each other there are
> > a host of protocols that are extremely trivial that can do 
> > the job.  The apps can even just open a port over any 
> > transport protocol and exchange information as they please.  
> > There is no reason to bother all the routing proxies in the 
> > network with your e2e private chatter.
> > 
> > Just a thought.
> > don
> > 
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of 
> > > George Foti (LMC)
> > > Sent: Thursday, January 02, 2003 11:55 AM
> > > To: 'Michael Thomas'; Eric Burger
> > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > > 
> > > 
> > > The application is SIP based, we dont need yet another 
> protocol to 
> > > convey what needs to be conveyed e2e. No need to go 
> around the bush. 
> > > Lets find *real* arguments against using INFO for that purpopse 
> > > rather than philosphical  ones.
> > > 
> > > /gf
> > > 
> > > 
> > > 
> > > 
> > > -----Original Message-----
> > > From: Michael Thomas [mailto:mat@cisco.com]
> > > Sent: Thursday, January 02, 2003 12:53 PM
> > > To: Eric Burger
> > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > > 
> > > 
> > > 
> > > I don't get it. If you want "100% application
> > > specific", why not pick an unused port and roll
> > > your own protocol? Then all of the crufty SIPism
> > > don't get in the way, and the hue and cry from
> > > this working group ceases immediately. Do the bits
> > > on the wire keep warmer if they're dressed in SIP?
> > > 
> > >        Mike
> > > 
> > > Eric Burger writes:
> > >  > I fully agree with Dean on this one.
> > >  >
> > >  > One point to consider (I'm on the fence on this one) is
> > > whether we would say that INFO is 100% application-specific.  
> > > That is, the WG will not publish standard body types.  > 
> > >  > A reason for doing that is to let people know that INFO 
> > > really is just, as Dean points out, 
> > > application-to-application communication.  If you want 
> > > applications to interoperate, take it either to an 
> > > Application Area WG or take it outside the IETF.  > 
> > >  > One reason to not do this is SIP-T should use a method 
> > > other than INFO.  That said, I don't think there are that 
> > > many SIP-T implementations extant.  > 
> > >  > 
> > >  > > -----Original Message-----
> > >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> > >  > > To: Jonathan Rosenberg; Orit Levin
> > >  > > Cc: sip@ietf.org
> > >  > > Subject: Re: [Sip] INFO considered harmful
> > >  > > 
> > >  > > 
> > >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> > >  > > >Orit Levin wrote:
> > >  > > >>>The big huge difference I have been trying to 
> point out is 
> > >  > > that EVENTS
> > >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> > >  > > >>>MIME provides semantics and syntax that are 
> independent of 
> > >  > > the type.
> > >  > > >>>SIP events provides semantics and syntax that are 
> > >  > > independent of the
> > >  > > >>>package.
> > >  > > >>>INFO provides nothing.
> > >  > > >>
> > >  > > >>INFO provides "minimal" but very important semantics: 
> > >  > > asynchronous data
> > >  > > >>reliably follows the established SIP path.
> > >  > > >
> > >  > > >Huh?
> > >  > > >
> > >  > > >Every new SIP method is reliable, it inherits the SIP 
> > >  > > transaction state 
> > >  > > >machine for non-invite. Follows the established path? Every 
> > >  > > in-dialog 
> > >  > > >method would have this property, it is method independent. 
> > >  > > >Asynchronous?  You can send any method at any time.
> > >  > > >
> > >  > > >Thus, nothing you have pointed out above is different 
> > >  > > between INFO and any 
> > >  > > >other new method we might introduce.
> > >  > > 
> > >  > > I think the key point is that there is a difference 
> > > between transport 
> > >  > > protocol semantic and application semantic. The transport 
> > >  > > protocol semantic 
> > >  > > expressed in INFO is "Here is some data that will be 
> > used by the
> > >  > > application. It is not important to the transport protocol,
> > >  > > except that the 
> > >  > > transport protocol is expected to deliver it reliably". INFO 
> > >  > > says NOTHING 
> > >  > > about the APPLICATION level semantic -- that's up to what 
> > >  > > goes IN the INFO 
> > >  > > payloads. The question is -- do we try to rigidly define 
> > >  > > application-level 
> > >  > > semantics here? Do we define that there ARE no such 
> > >  > > applications possible? 
> > >  > > Or do we design a framework by which application-level 
> > >  > > semantics can be 
> > >  > > expressed outside of the protocol definition?
> > >  > > 
> > >  > > And as for "Just add another method to your SIP 
> stack". Let's 
> > >  > > say I have a 
> > >  > > mobile phone and the SIP stack is burned into ROM and 
> > > exposes only a 
> > >  > > simplistic transactional API. This is only likely to 
> > > happen, oh, 100 
> > >  > > million times or so over the next two years . . .  Now, just 
> > >  > > exactly how is 
> > >  > > my BREW or JTME application going to go about extending the 
> > >  > > SIP stack to 
> > >  > > support another method? Will the evil wireless operator 
> > >  > > networks even pass 
> > >  > > another method? Neither is happening anytime soon, I 
> > > think . . .  > > 
> > >  > > --
> > >  > > Dean
> > >  > > 
> > >  > > 
> > >  > > _______________________________________________
> > >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >  > > This list is for NEW development of the core SIP 
> > > Protocol  > > Use sip-implementors@cs.columbia.edu for 
> > > questions on current sip  > > Use sipping@ietf.org for new 
> > > developments on the application of sip  > > 
> > >  > _______________________________________________
> > >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >  > This list is for NEW development of the core SIP Protocol
> > >  > Use sip-implementors@cs.columbia.edu for questions on 
> > > current sip  > Use sipping@ietf.org for new developments on 
> > > the application of sip 
> > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sipping@ietf.org for new developments on the application of sip 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use 
> > > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > > sipping@ietf.org for new developments on the application of sip
> > > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current
> > sip Use sipping@ietf.org for new developments on the 
> > application of sip
> > 
> 

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From: "Frank W. Miller" <fmiller@sentito.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
References: <3E0F3B60.A1D37549@lmf.ericsson.se><00c501c2b037$db7e90f0$56fa403f@txdwillis> <15888.45814.401844.797450@thomasm-u1.cisco.com> <006c01c2b055$aab320d0$9f00000a@overhill> <3E14A895.7010804@dynamicsoft.com>
Subject: Re: [Sip] INFO considered harmful
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Thanks for the thoughtful reply.

Comments inline...

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Frank W. Miller" <fmiller@sentito.com>
Cc: <sip@ietf.org>
Sent: Thursday, January 02, 2003 4:01 PM
Subject: Re: [Sip] INFO considered harmful


>
> Sometimes, there is a need for vendor proprietary extensions
> to such protocols. However, all vendor extensions need to live within a
> "sandbox" that limits the scope of what can be done.

I think this is really the crux of the issue for me.  Because this is a
signaling protocol and not a routing or transport protocol, the need to
provide for corner cases and extensions over time is greater.

> The problem with INFO is that it defines no such sandbox.

Absolutely correct and absolutely a potential problem for a variety of
reasons, including clobbering the intermediary servers.  But, the reality
is, application implementors need some kind of app-to-app tunneling for
stuff that isn't explictily handled.  Also a reality is that INFO has been
around for awhile and people are using it.  In lieu of something better, the
mechanism has to be there.  In a way, using INFO can actually be a good
thing because it paints a great big red "A" on the packet as soon as it show
up somewhere.  If it were in some other legal method that was being abused,
the potential for abuse might not be as easy to spot.


> Let me take this to the extreme. By your argument, TCP should allow for
> vendor-specified plug ins for congestion control. After all, why should
> IETF prevent me from using something that works well for MY application?
> Indeed, there is lots of research on different congestion control
> algorithms. TCP should allow those to be used! But, would you really
> want someone to be able to plug in a congestion control algorithm that
> basically doesn't do congestion control algorithm? I should hope not.
>

Your point is well taken, but I would refer back to my earlier statement.
This is a signaling protocol, and has different, higher-level semantic
requirements placed on it.  The need for various congestion control
algorithms (to continue your example) is mitigated by the fact that the
existing protocol covers a huge portion of the requirement space.  This is
not as clear with a signaling protocol.

> >
> > As has already been noted in this thread, there are a number of fine
> > implementations (including Dynamicsoft) in the field, many, if not all,
> > including a punt to the application of the INFO body.
>
> Frank, pointing a finger at me and saying "but your company does it!" is
> neither a compelling argument or, IMHO, a civilized way to conduct a
> discussion on the list. It is frequently the case that ones company does
> something that one does not like, because a customer asked, or a partner
> is doing it, or because of one reason or another. Does that mean that we
> should be promoting such things as IETF standards?
>

My apologies.  I should have realized that just because it might be in your
product, that would not necessarily affect your opinion in the IETF forum.
But that was not really the intent of my comment.  I was simply trying to
point out that the need for tunneling is apparent since all the major
vendors are supporting what their customers have required, nothing more.


>
> > This is very useful
> > functionality to implementors and as such it makes very little sense to
> > deprecate this method.  Another will take its place as the abused.  For
> > example, I've had discussions with people about doing extensions using
> > SUBSCRIBE/NOTIFY that had nothing whatever to do with events just
because of
> > the "political incorrectness" of INFO.
>
> Unlike INFO, SUB/NOT defines a very well-scoped and limited repetoire of
> behavior for packages. So, while someone may violate the spec and do
> something inappropriate, it would not be IETF compliant. WIth INFO, one
> can do bad things and still claim to be compliant to the specifications.
>


So, there it is.  What we really need is a substitute for INFO that works
the way implementors need but also has the appropriate scaling and safety
properties.  I think the substitute consists of two parts.

1) A method/methods that allow for some degree of application-to-application
tunneling
2) An IETF process associated with this method that allows for incorporation
of this tunneling into the mainline protocol over time

Ideally, if 1) is done right, very little syntactic change to extensions
would be necessary for 2) to happen.  The split of SIP into SIP/SIPPING was
a step down the road to 2)  What was missing from that was 1).  If there was
a a 1), the flow of new elements coming into these WGs and subsequently into
the standard might have more rigor and hopefully progress more quickly.

Thanks again,
FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentito.com

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Subject: RE: [Sip] INFO considered harmful
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Right, but my personal bugaboo is modularity.
Like Don's FTPoSIP example, the more junk you
throw into one protocol, the more entangled it
becomes to the point that you have a deep dark
morass. In any case, we already have a real live
kitchen sink protocol: HTTP; you even run IP over
HTTP these days. I'd be surprised to hear that
just about anything that runs SIP doesn't also
have web kruft in it as well.

	 Mike

Adam Roach writes:
 > A new method would be okay for new SIP-related functionality.
 > 
 > But that's not what you proposed.
 > 
 > You said, "The application is SIP based, we dont need yet
 > another protocol to convey what needs to be conveyed e2e,"
 > which is an argument that we should use SIP as a general
 > purpose RPC mechanism for any application that already has
 > a SIP stack.
 > 
 > Doing what you propose with INFO is inappropriate.
 > 
 > Doing what you propose with a new method is inappropriate.
 > 
 > Doing what you propose with SIP is inappropriate.
 > 
 > Since you keep harping on technical, as opposed to
 > philosophical, arguments, here's a couple:
 > 
 >   - The SIP protocol was designed for session initiation,
 >     negotiation, and teardown. The timing characteristics
 >     are not generally appropriate for application control.
 >     For example, with typical SIP networks, latency and
 >     jitter of messages are too high to provide any sort
 >     of reasonable user experience for floor control
 >     or DTMF transport.
 > 
 >  - Many proxies stay in the signalling path for the purposes
 >    of providing services. By sending unnecessary application-
 >    level information throught these proxies, you reduce the
 >    capacity and increase possible congestion of the network.
 > 
 > I could go on, but this topic has had extensive airing on
 > the list in years past. There's really no reason to beat
 > this particular dead horse any longer.
 > 
 > /a
 > 
 > > -----Original Message-----
 > > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
 > > Sent: Thursday, January 02, 2003 16:05
 > > To: 'Adam Roach'; George Foti (LMC); 'Michael Thomas'; Eric Burger
 > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
 > > Subject: RE: [Sip] INFO considered harmful
 > > 
 > > 
 > > This should not be a philosophical discusion.
 > > If a new SIP method is okay, why is INFO not okay.
 > > 
 > > What is the difference?
 > > Please explain in technical terms?
 > > 
 > > /gf
 > > 
 > > -----Original Message-----
 > > From: Adam Roach [mailto:adam@dynamicsoft.com]
 > > Sent: Thursday, January 02, 2003 2:36 PM
 > > To: 'George Foti (LMC)'; 'Michael Thomas'; Eric Burger
 > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
 > > Subject: RE: [Sip] INFO considered harmful
 > > 
 > > 
 > > The argument of "but I'm already holding a hammer and *want*
 > > to treat everything like a nail!" comes up on about an annual
 > > basis, and is rejected outright each time.
 > > 
 > > Just because you have a SIP stack in your application does
 > > *not* mean that it is appropriate to use that stack for
 > > all general network access. You're allowing laziness to
 > > overwhelm good application design.
 > > 
 > > /a
 > > 
 > > > -----Original Message-----
 > > > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
 > > > Sent: Thursday, January 02, 2003 12:55
 > > > To: 'Michael Thomas'; Eric Burger
 > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
 > > > Subject: RE: [Sip] INFO considered harmful
 > > > 
 > > > 
 > > > The application is SIP based, we dont need yet another 
 > > > protocol to convey what needs to be conveyed e2e.
 > > > No need to go around the bush.
 > > > Lets find *real* arguments against using INFO for that 
 > > > purpopse rather than philosphical  ones.
 > > > 
 > > > /gf
 > > > 
 > > > 
 > > > 
 > > > 
 > > > -----Original Message-----
 > > > From: Michael Thomas [mailto:mat@cisco.com]
 > > > Sent: Thursday, January 02, 2003 12:53 PM
 > > > To: Eric Burger
 > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
 > > > Subject: RE: [Sip] INFO considered harmful
 > > > 
 > > > 
 > > > 
 > > > I don't get it. If you want "100% application
 > > > specific", why not pick an unused port and roll
 > > > your own protocol? Then all of the crufty SIPism
 > > > don't get in the way, and the hue and cry from
 > > > this working group ceases immediately. Do the bits 
 > > > on the wire keep warmer if they're dressed in SIP?
 > > > 
 > > >        Mike
 > > > 
 > > > Eric Burger writes:
 > > >  > I fully agree with Dean on this one.
 > > >  > 
 > > >  > One point to consider (I'm on the fence on this one) is 
 > > > whether we would say that INFO is 100% application-specific.  
 > > > That is, the WG will not publish standard body types.
 > > >  > 
 > > >  > A reason for doing that is to let people know that INFO 
 > > > really is just, as Dean points out, 
 > > > application-to-application communication.  If you want 
 > > > applications to interoperate, take it either to an 
 > > > Application Area WG or take it outside the IETF.
 > > >  > 
 > > >  > One reason to not do this is SIP-T should use a method 
 > > > other than INFO.  That said, I don't think there are that 
 > > > many SIP-T implementations extant.
 > > >  > 
 > > >  > 
 > > >  > > -----Original Message-----
 > > >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
 > > >  > > Sent: Tuesday, December 31, 2002 1:48 AM
 > > >  > > To: Jonathan Rosenberg; Orit Levin
 > > >  > > Cc: sip@ietf.org
 > > >  > > Subject: Re: [Sip] INFO considered harmful
 > > >  > > 
 > > >  > > 
 > > >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
 > > >  > > >Orit Levin wrote:
 > > >  > > >>>The big huge difference I have been trying to point out is 
 > > >  > > that EVENTS
 > > >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
 > > >  > > >>>MIME provides semantics and syntax that are independent of 
 > > >  > > the type.
 > > >  > > >>>SIP events provides semantics and syntax that are 
 > > >  > > independent of the
 > > >  > > >>>package.
 > > >  > > >>>INFO provides nothing.
 > > >  > > >>
 > > >  > > >>INFO provides "minimal" but very important semantics: 
 > > >  > > asynchronous data
 > > >  > > >>reliably follows the established SIP path.
 > > >  > > >
 > > >  > > >Huh?
 > > >  > > >
 > > >  > > >Every new SIP method is reliable, it inherits the SIP 
 > > >  > > transaction state 
 > > >  > > >machine for non-invite. Follows the established path? Every 
 > > >  > > in-dialog 
 > > >  > > >method would have this property, it is method independent. 
 > > >  > > >Asynchronous?  You can send any method at any time.
 > > >  > > >
 > > >  > > >Thus, nothing you have pointed out above is different 
 > > >  > > between INFO and any 
 > > >  > > >other new method we might introduce.
 > > >  > > 
 > > >  > > I think the key point is that there is a difference 
 > > > between transport 
 > > >  > > protocol semantic and application semantic. The transport 
 > > >  > > protocol semantic 
 > > >  > > expressed in INFO is "Here is some data that will be 
 > > used by the 
 > > >  > > application. It is not important to the transport protocol, 
 > > >  > > except that the 
 > > >  > > transport protocol is expected to deliver it reliably". INFO 
 > > >  > > says NOTHING 
 > > >  > > about the APPLICATION level semantic -- that's up to what 
 > > >  > > goes IN the INFO 
 > > >  > > payloads. The question is -- do we try to rigidly define 
 > > >  > > application-level 
 > > >  > > semantics here? Do we define that there ARE no such 
 > > >  > > applications possible? 
 > > >  > > Or do we design a framework by which application-level 
 > > >  > > semantics can be 
 > > >  > > expressed outside of the protocol definition?
 > > >  > > 
 > > >  > > And as for "Just add another method to your SIP stack". Let's 
 > > >  > > say I have a 
 > > >  > > mobile phone and the SIP stack is burned into ROM and 
 > > > exposes only a 
 > > >  > > simplistic transactional API. This is only likely to 
 > > > happen, oh, 100 
 > > >  > > million times or so over the next two years . . .  Now, just 
 > > >  > > exactly how is 
 > > >  > > my BREW or JTME application going to go about extending the 
 > > >  > > SIP stack to 
 > > >  > > support another method? Will the evil wireless operator 
 > > >  > > networks even pass 
 > > >  > > another method? Neither is happening anytime soon, I 
 > > think . . .
 > > >  > > 
 > > >  > > --
 > > >  > > Dean
 > > >  > > 
 > > >  > > 
 > > >  > > _______________________________________________
 > > >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > >  > > This list is for NEW development of the core SIP Protocol
 > > >  > > Use sip-implementors@cs.columbia.edu for questions on 
 > > current sip
 > > >  > > Use sipping@ietf.org for new developments on the 
 > > > application of sip
 > > >  > > 
 > > >  > _______________________________________________
 > > >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > >  > This list is for NEW development of the core SIP Protocol
 > > >  > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > >  > Use sipping@ietf.org for new developments on the 
 > > application of sip
 > > > _______________________________________________
 > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > > This list is for NEW development of the core SIP Protocol
 > > > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > > Use sipping@ietf.org for new developments on the application of sip
 > > > _______________________________________________
 > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > > This list is for NEW development of the core SIP Protocol
 > > > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > > Use sipping@ietf.org for new developments on the application of sip
 > > > 
 > > 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Adam Roach <adam@dynamicsoft.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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> From: Frank W. Miller [mailto:fmiller@sentito.com]
>
> [T]he reality is, application implementors need some kind of
> app-to-app tunneling for stuff that isn't explictily handled. 

I'm trying to understand this argument more clearly.

It sounds to me like you need some sort of session of application-
specific information that you can use to communicate between two
nodes.

Perhaps using some sort of protocol to initate this session
would be appropriate. You'll also needs a protocol of some sort
to describe it, probably carried as a payload in this session
initation protocol. Ideally, this session description protocol
could leverage IANA-registered media types to describe the content
of such sessions, so that an infinite number of types of sessions
can be described.

Hmmm...

/a
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From: "George Foti (LMC)" <George.Foti@ericsson.ca>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "George Foti (LMC)"
	 <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
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Subject: RE: [Sip] INFO considered harmful
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>I could go on, but this topic has had extensive airing on
>the list in years past. There's really no reason to beat
>this particular dead horse any longer.

Agreed, but the extremety, the exaggarations and the imaginary, creative non-real abuses and comparisons with other protocols, that have been raised on e-mails about misuse of INFO represents a *sad* distortion of how most common sense people have used it, and contradicts the reality on the ground.
Only constructive counter proposals may move this forward.  

Rgds/gf



-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Thursday, January 02, 2003 5:57 PM
To: 'George Foti (LMC)'; Adam Roach; 'Michael Thomas'; Eric Burger
Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
Subject: RE: [Sip] INFO considered harmful


A new method would be okay for new SIP-related functionality.

But that's not what you proposed.

You said, "The application is SIP based, we dont need yet
another protocol to convey what needs to be conveyed e2e,"
which is an argument that we should use SIP as a general
purpose RPC mechanism for any application that already has
a SIP stack.

Doing what you propose with INFO is inappropriate.

Doing what you propose with a new method is inappropriate.

Doing what you propose with SIP is inappropriate.

Since you keep harping on technical, as opposed to
philosophical, arguments, here's a couple:

  - The SIP protocol was designed for session initiation,
    negotiation, and teardown. The timing characteristics
    are not generally appropriate for application control.
    For example, with typical SIP networks, latency and
    jitter of messages are too high to provide any sort
    of reasonable user experience for floor control
    or DTMF transport.

 - Many proxies stay in the signalling path for the purposes
   of providing services. By sending unnecessary application-
   level information throught these proxies, you reduce the
   capacity and increase possible congestion of the network.

I could go on, but this topic has had extensive airing on
the list in years past. There's really no reason to beat
this particular dead horse any longer.

/a

> -----Original Message-----
> From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> Sent: Thursday, January 02, 2003 16:05
> To: 'Adam Roach'; George Foti (LMC); 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> This should not be a philosophical discusion.
> If a new SIP method is okay, why is INFO not okay.
> 
> What is the difference?
> Please explain in technical terms?
> 
> /gf
> 
> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Thursday, January 02, 2003 2:36 PM
> To: 'George Foti (LMC)'; 'Michael Thomas'; Eric Burger
> Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> The argument of "but I'm already holding a hammer and *want*
> to treat everything like a nail!" comes up on about an annual
> basis, and is rejected outright each time.
> 
> Just because you have a SIP stack in your application does
> *not* mean that it is appropriate to use that stack for
> all general network access. You're allowing laziness to
> overwhelm good application design.
> 
> /a
> 
> > -----Original Message-----
> > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> > Sent: Thursday, January 02, 2003 12:55
> > To: 'Michael Thomas'; Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > The application is SIP based, we dont need yet another 
> > protocol to convey what needs to be conveyed e2e.
> > No need to go around the bush.
> > Lets find *real* arguments against using INFO for that 
> > purpopse rather than philosphical  ones.
> > 
> > /gf
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: Michael Thomas [mailto:mat@cisco.com]
> > Sent: Thursday, January 02, 2003 12:53 PM
> > To: Eric Burger
> > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > 
> > I don't get it. If you want "100% application
> > specific", why not pick an unused port and roll
> > your own protocol? Then all of the crufty SIPism
> > don't get in the way, and the hue and cry from
> > this working group ceases immediately. Do the bits 
> > on the wire keep warmer if they're dressed in SIP?
> > 
> >        Mike
> > 
> > Eric Burger writes:
> >  > I fully agree with Dean on this one.
> >  > 
> >  > One point to consider (I'm on the fence on this one) is 
> > whether we would say that INFO is 100% application-specific.  
> > That is, the WG will not publish standard body types.
> >  > 
> >  > A reason for doing that is to let people know that INFO 
> > really is just, as Dean points out, 
> > application-to-application communication.  If you want 
> > applications to interoperate, take it either to an 
> > Application Area WG or take it outside the IETF.
> >  > 
> >  > One reason to not do this is SIP-T should use a method 
> > other than INFO.  That said, I don't think there are that 
> > many SIP-T implementations extant.
> >  > 
> >  > 
> >  > > -----Original Message-----
> >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> >  > > To: Jonathan Rosenberg; Orit Levin
> >  > > Cc: sip@ietf.org
> >  > > Subject: Re: [Sip] INFO considered harmful
> >  > > 
> >  > > 
> >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> >  > > >Orit Levin wrote:
> >  > > >>>The big huge difference I have been trying to point out is 
> >  > > that EVENTS
> >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> >  > > >>>MIME provides semantics and syntax that are independent of 
> >  > > the type.
> >  > > >>>SIP events provides semantics and syntax that are 
> >  > > independent of the
> >  > > >>>package.
> >  > > >>>INFO provides nothing.
> >  > > >>
> >  > > >>INFO provides "minimal" but very important semantics: 
> >  > > asynchronous data
> >  > > >>reliably follows the established SIP path.
> >  > > >
> >  > > >Huh?
> >  > > >
> >  > > >Every new SIP method is reliable, it inherits the SIP 
> >  > > transaction state 
> >  > > >machine for non-invite. Follows the established path? Every 
> >  > > in-dialog 
> >  > > >method would have this property, it is method independent. 
> >  > > >Asynchronous?  You can send any method at any time.
> >  > > >
> >  > > >Thus, nothing you have pointed out above is different 
> >  > > between INFO and any 
> >  > > >other new method we might introduce.
> >  > > 
> >  > > I think the key point is that there is a difference 
> > between transport 
> >  > > protocol semantic and application semantic. The transport 
> >  > > protocol semantic 
> >  > > expressed in INFO is "Here is some data that will be 
> used by the 
> >  > > application. It is not important to the transport protocol, 
> >  > > except that the 
> >  > > transport protocol is expected to deliver it reliably". INFO 
> >  > > says NOTHING 
> >  > > about the APPLICATION level semantic -- that's up to what 
> >  > > goes IN the INFO 
> >  > > payloads. The question is -- do we try to rigidly define 
> >  > > application-level 
> >  > > semantics here? Do we define that there ARE no such 
> >  > > applications possible? 
> >  > > Or do we design a framework by which application-level 
> >  > > semantics can be 
> >  > > expressed outside of the protocol definition?
> >  > > 
> >  > > And as for "Just add another method to your SIP stack". Let's 
> >  > > say I have a 
> >  > > mobile phone and the SIP stack is burned into ROM and 
> > exposes only a 
> >  > > simplistic transactional API. This is only likely to 
> > happen, oh, 100 
> >  > > million times or so over the next two years . . .  Now, just 
> >  > > exactly how is 
> >  > > my BREW or JTME application going to go about extending the 
> >  > > SIP stack to 
> >  > > support another method? Will the evil wireless operator 
> >  > > networks even pass 
> >  > > another method? Neither is happening anytime soon, I 
> think . . .
> >  > > 
> >  > > --
> >  > > Dean
> >  > > 
> >  > > 
> >  > > _______________________________________________
> >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > > This list is for NEW development of the core SIP Protocol
> >  > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> >  > > Use sipping@ietf.org for new developments on the 
> > application of sip
> >  > > 
> >  > _______________________________________________
> >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >  > This list is for NEW development of the core SIP Protocol
> >  > Use sip-implementors@cs.columbia.edu for questions on current sip
> >  > Use sipping@ietf.org for new developments on the 
> application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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From: Adam Roach <adam@dynamicsoft.com>
To: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        Adam Roach
	 <adam@dynamicsoft.com>,
        "George Foti (LMC)"
	 <George.Foti@lmc.ericsson.se>,
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        Eric Burger <eburger@snowshore.com>
Cc: Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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> -----Original Message-----
> From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
> 
> >I could go on, but this topic has had extensive airing on
> >the list in years past. There's really no reason to beat
> >this particular dead horse any longer.
> 
> Agreed, but the extremety, the exaggarations and the 
> imaginary, creative non-real abuses and comparisons with 
> other protocols, that have been raised on e-mails about 
> misuse of INFO represents a *sad* distortion of how most 
> common sense people have used it, and contradicts the reality 
> on the ground.

Imaginary? Non-real?

Ummm... "DTMF Transport" and "Floor Control" are the two
examples I offered as being inappropriate (along with
a good, technical explanation of why). Both of these have
been seriously suggested as standard extensions to SIP.

I've even seen attempts to deploy them. In a real
carrier environment. End result: scrapping that part
of the system and going back to the drawing board.

/a
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This list is for NEW development of the core SIP Protocol
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From: "Neil Deason" <ndeason@ubiquity.net>
To: "Don Stanwyck" <don@stanwyck.com>,
        "'George Foti \(LMC\)'" <George.Foti@ericsson.ca>,
        "'George Foti \(LMC\)'" <George.Foti@lmc.ericsson.se>,
        "'Michael Thomas'" <mat@cisco.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 16:20:33 -0800
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Hello.

An observation + suggested way forward inline ...

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Don
> Stanwyck
>
> To follow up my own posting (bad form, I know), I thought I should
> mention a real life example that worries me.
>
> Many new cell phones have picture taking and picture sending
> capabilities.  Some even allow for "movie" modes that send a
> picture per
> second of so.
>
> If the call is set up using SIP and the phone then wants to send all
> these photos also over SIP, then the SIP proxies in the network can
> become overloaded with what should be a low priority traffic
> instead of
> being used for their design purpose - to process communication session
> establishment information.

No need to worry about INFO abuse here. Obviously this would be done
using MESSAGE ;)

Actually if we want to police protocol usage closely we had better look
at deprecating MESSAGE too. This method allows SIP to be used as a data
transport protocol, either inside or outside the context of dialog. It
seems naive to think saying usage is expected to be "small in number,
short content, SMS like" etc. is going to make much difference. While
MESSAGE may have slightly more defined method semantics than INFO, I am
not convinced they are really that much more defined. To me it looks
like MESSAGE sematics are more being inferred by the paging-like
application typically associated with it.

Anyway my $0.02 on INFO itself. Personally I am not a huge fan and
obviously many people _really_ don't like it, but one man's meat is
another man's poison. So returning to the original proposal - option
2/"INFO-packages" looks like a way to progress. Providing of course
there is sufficient commitment to keeping INFO around to spur
authorship.

Cheers,
Neil.

> > -----Original Message-----
> > From: Don Stanwyck [mailto:don@stanwyck.com]
> > Sent: Thursday, January 02, 2003 4:02 PM
> > To: 'George Foti (LMC)'; 'George Foti (LMC)'; 'Michael
> > Thomas'; 'Eric Burger'
> > Cc: 'Dean Willis'; 'Jonathan Rosenberg'; 'sip@ietf.org'
> > Subject: RE: [Sip] INFO considered harmful
> >
> >
> > It increases the number of transactions they have to process.
> >  Nothing in life is free.  SIP Proxies, like other call
> > routing devices, assume a certain number of transactions per
> > call as part of their design numbers.  This is important for
> > traffic engineering.
> >
> > If we allow SIP to become the new FTP protocol (and there is
> > nothing right now that would stop a UA from using SIP-INFO
> > for FTP or FTP like functions), then the calls/second counts
> > and all other traffic engineering rules that are needed for
> > proper network operation become meaningless.
> >
> > Don
> >
> > > -----Original Message-----
> > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > > Behalf Of George Foti (LMC)
> > > Sent: Thursday, January 02, 2003 3:03 PM
> > > To: 'Don Stanwyck'; George Foti (LMC); 'Michael Thomas';
> > 'Eric Burger'
> > > Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > >
> > >
> > > >There is no reason to bother all the
> > > >routing proxies in the network with your e2e private chatter.
> > >
> > > How are they bothered ?
> > > What extra logic do they need?
> > >
> > > /gf
> > >
> > >
> > > -----Original Message-----
> > > From: Don Stanwyck [mailto:don@stanwyck.com]
> > > Sent: Thursday, January 02, 2003 3:13 PM
> > > To: 'George Foti (LMC)'; 'Michael Thomas'; 'Eric Burger'
> > > Cc: 'Dean Willis'; 'Jonathan Rosenberg'; sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > >
> > >
> > > I guess I have two problems with the statement below.
> > >
> > > 1) Applications should not be protocol based - they may be
> > > protocol aware.  Unless the SIP stack is rolled into the
> > > application, it merely uses SIP as a lower layer tool to
> > > establish communications.  It is a violation of good design
> > > to roll the protocol into the application itself.
> > >    BTW, if the SIP stack is rolled into the application, then
> > > when the next edition of SIP comes along you will have to
> > > rewrite the app.  You really should have an API that keeps
> > > the app from needing to know the details of the communication
> > > establishment protocol.
> > >
> > > 2) If the apps need to talk directly to each other there are
> > > a host of protocols that are extremely trivial that can do
> > > the job.  The apps can even just open a port over any
> > > transport protocol and exchange information as they please.
> > > There is no reason to bother all the routing proxies in the
> > > network with your e2e private chatter.
> > >
> > > Just a thought.
> > > don
> > >
> > > > -----Original Message-----
> > > > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> Behalf Of
> > > > George Foti (LMC)
> > > > Sent: Thursday, January 02, 2003 11:55 AM
> > > > To: 'Michael Thomas'; Eric Burger
> > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > > > Subject: RE: [Sip] INFO considered harmful
> > > >
> > > >
> > > > The application is SIP based, we dont need yet another
> > protocol to
> > > > convey what needs to be conveyed e2e. No need to go
> > around the bush.
> > > > Lets find *real* arguments against using INFO for that purpopse
> > > > rather than philosphical  ones.
> > > >
> > > > /gf
> > > >
> > > >
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: Michael Thomas [mailto:mat@cisco.com]
> > > > Sent: Thursday, January 02, 2003 12:53 PM
> > > > To: Eric Burger
> > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
> > > > Subject: RE: [Sip] INFO considered harmful
> > > >
> > > >
> > > >
> > > > I don't get it. If you want "100% application
> > > > specific", why not pick an unused port and roll
> > > > your own protocol? Then all of the crufty SIPism
> > > > don't get in the way, and the hue and cry from
> > > > this working group ceases immediately. Do the bits
> > > > on the wire keep warmer if they're dressed in SIP?
> > > >
> > > >        Mike
> > > >
> > > > Eric Burger writes:
> > > >  > I fully agree with Dean on this one.
> > > >  >
> > > >  > One point to consider (I'm on the fence on this one) is
> > > > whether we would say that INFO is 100% application-specific.
> > > > That is, the WG will not publish standard body types.  >
> > > >  > A reason for doing that is to let people know that INFO
> > > > really is just, as Dean points out,
> > > > application-to-application communication.  If you want
> > > > applications to interoperate, take it either to an
> > > > Application Area WG or take it outside the IETF.  >
> > > >  > One reason to not do this is SIP-T should use a method
> > > > other than INFO.  That said, I don't think there are that
> > > > many SIP-T implementations extant.  >
> > > >  >
> > > >  > > -----Original Message-----
> > > >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > > >  > > Sent: Tuesday, December 31, 2002 1:48 AM
> > > >  > > To: Jonathan Rosenberg; Orit Levin
> > > >  > > Cc: sip@ietf.org
> > > >  > > Subject: Re: [Sip] INFO considered harmful
> > > >  > >
> > > >  > >
> > > >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
> > > >  > > >Orit Levin wrote:
> > > >  > > >>>The big huge difference I have been trying to
> > point out is
> > > >  > > that EVENTS
> > > >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
> > > >  > > >>>MIME provides semantics and syntax that are
> > independent of
> > > >  > > the type.
> > > >  > > >>>SIP events provides semantics and syntax that are
> > > >  > > independent of the
> > > >  > > >>>package.
> > > >  > > >>>INFO provides nothing.
> > > >  > > >>
> > > >  > > >>INFO provides "minimal" but very important semantics:
> > > >  > > asynchronous data
> > > >  > > >>reliably follows the established SIP path.
> > > >  > > >
> > > >  > > >Huh?
> > > >  > > >
> > > >  > > >Every new SIP method is reliable, it inherits the SIP
> > > >  > > transaction state
> > > >  > > >machine for non-invite. Follows the established
> path? Every
> > > >  > > in-dialog
> > > >  > > >method would have this property, it is method
> independent.
> > > >  > > >Asynchronous?  You can send any method at any time.
> > > >  > > >
> > > >  > > >Thus, nothing you have pointed out above is different
> > > >  > > between INFO and any
> > > >  > > >other new method we might introduce.
> > > >  > >
> > > >  > > I think the key point is that there is a difference
> > > > between transport
> > > >  > > protocol semantic and application semantic. The transport
> > > >  > > protocol semantic
> > > >  > > expressed in INFO is "Here is some data that will be
> > > used by the
> > > >  > > application. It is not important to the transport protocol,
> > > >  > > except that the
> > > >  > > transport protocol is expected to deliver it
> reliably". INFO
> > > >  > > says NOTHING
> > > >  > > about the APPLICATION level semantic -- that's up to what
> > > >  > > goes IN the INFO
> > > >  > > payloads. The question is -- do we try to rigidly define
> > > >  > > application-level
> > > >  > > semantics here? Do we define that there ARE no such
> > > >  > > applications possible?
> > > >  > > Or do we design a framework by which application-level
> > > >  > > semantics can be
> > > >  > > expressed outside of the protocol definition?
> > > >  > >
> > > >  > > And as for "Just add another method to your SIP
> > stack". Let's
> > > >  > > say I have a
> > > >  > > mobile phone and the SIP stack is burned into ROM and
> > > > exposes only a
> > > >  > > simplistic transactional API. This is only likely to
> > > > happen, oh, 100
> > > >  > > million times or so over the next two years . . .
> Now, just
> > > >  > > exactly how is
> > > >  > > my BREW or JTME application going to go about
> extending the
> > > >  > > SIP stack to
> > > >  > > support another method? Will the evil wireless operator
> > > >  > > networks even pass
> > > >  > > another method? Neither is happening anytime soon, I
> > > > think . . .  > >
> > > >  > > --
> > > >  > > Dean
> > > >  > >
> > > >  > >
> > > >  > > _______________________________________________
> > > >  > > Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip
> > >  > > This list is for NEW development of the core SIP
> > > Protocol  > > Use sip-implementors@cs.columbia.edu for
> > > questions on current sip  > > Use sipping@ietf.org for new
> > > developments on the application of sip  > >
> > >  > _______________________________________________
> > >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >  > This list is for NEW development of the core SIP Protocol
> > >  > Use sip-implementors@cs.columbia.edu for questions on
> > > current sip  > Use sipping@ietf.org for new developments on
> > > the application of sip
> > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> > >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current
> > sip Use sipping@ietf.org for new developments on the
> > application of sip
> >
>

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

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



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Subject: Re: [Sip] INFO considered harmful
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Heh.

What I'm referring to is the need for implementors to build upon the basic
signaling mechanism that SIP provides.  If an application has signaling
needs that are not met with the basic protocol constructs, application
implementors need a way to make it happen.  As others have argued, there are
instances where augmenting the (perhaps third party) SIP stack source code
with new methods does not make sense.  This can be expensive and
time-consuming compared to say, punting an INFO body written in XML to an
open source parser.  The ability to get something like this done quickly
and/or without perturbing the SIP stack code is very important, despite your
sarcasm.

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentito.com


----- Original Message -----
From: "Adam Roach" <adam@dynamicsoft.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>; "Jonathan Rosenberg"
<jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
Sent: Thursday, January 02, 2003 7:00 PM
Subject: RE: [Sip] INFO considered harmful


> > From: Frank W. Miller [mailto:fmiller@sentito.com]
> >
> > [T]he reality is, application implementors need some kind of
> > app-to-app tunneling for stuff that isn't explictily handled.
>
> I'm trying to understand this argument more clearly.
>
> It sounds to me like you need some sort of session of application-
> specific information that you can use to communicate between two
> nodes.
>
> Perhaps using some sort of protocol to initate this session
> would be appropriate. You'll also needs a protocol of some sort
> to describe it, probably carried as a payload in this session
> initation protocol. Ideally, this session description protocol
> could leverage IANA-registered media types to describe the content
> of such sessions, so that an infinite number of types of sessions
> can be described.
>
> Hmmm...
>
> /a
>

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



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Subject: RE: [Sip] INFO considered harmful
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> From: Frank W. Miller [mailto:fmiller@sentito.com]
>
> What I'm referring to is the need for implementors to build 
> upon the basic signaling mechanism that SIP provides.  If an
> application has signaling needs that are not met with the basic
> protocol constructs, application implementors need a way to make
> it happen. As others have argued, there are instances where
> augmenting the (perhaps third party) SIP stack source code with
> new methods does not make sense.

Sorry for being a bit obtuse (and sarcastic). I was trying to
grab your attention, since the earlier statement (made by Jonathan)
that one should use SIP to initiate such sessions apparently failed
to do so.

I wasn't proposing adding new methods. I was proposing using INVITE to
establish a session, and exchanging your application-level information
using such a session.

> This can be expensive and
> time-consuming compared to say, punting an INFO body written in
> XML to an open source parser. The ability to get something
> like this done quickly and/or without perturbing the SIP stack
> code is very important, despite your sarcasm.

Shoving XML over TCP (perhaps using SOAP, if that suits
your fancy) is trivial to code, and the comedia extensions
make it possible. It is all eminently expresible using SDP
as it exists today, and, for any decent programmer with the
most basic understanding of sockets programming (or even a
good abstraction library), is every bit as easy as what you
propose.

Except, of course, that it lacks the "SIP as general-purpose
RPC" abuse factor.

/a
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Neil Deason'" <ndeason@ubiquity.net>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 2 Jan 2003 19:10:44 -0600 
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> From: Neil Deason [mailto:ndeason@ubiquity.net]
> 
> Actually if we want to police protocol usage closely we had 
> better look at deprecating MESSAGE too.

Come back when SIMPLE has finished defining the message sessions,
and I'll throw my voice behind yours in advocating this.

/a
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [Sip] INFO considered harmful
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Actually, there's an important difference: MESSAGE can use the SIP 
features outside an established session, while INFO is within a session 
(and thus there are other SDP-negotiated, options available). Within a 
session, the only real reason for a follow-the-INVITE SIP message is 
that a direct IP-level connection can't be made to work.

 From an efficiency standpoint, if you only have very occasional 
messages, you gain by a "connectionless" approach. After all, an INVITE 
transaction for a single MESSAGE doesn't buy you much and doesn't 
conserve the oh-so-precious proxy cycles. While such short sessions are 
probably not common for human-to-human communication (except spam...), 
they may well be common for the type of things that SMS and pagers are 
popular for: machine-to-human or machine-to-machine communications 
announcing infrequent and unpredictable events ("your flight is now 
leaving on gate 37", "your FedEx just arrived", "Server 42 is on fire").

Adam Roach wrote:
>>From: Neil Deason [mailto:ndeason@ubiquity.net]
>>
>>Actually if we want to police protocol usage closely we had 
>>better look at deprecating MESSAGE too.
> 
> 
> Come back when SIMPLE has finished defining the message sessions,
> and I'll throw my voice behind yours in advocating this.
> 
> /a
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From: "Neil Deason" <ndeason@ubiquity.net>
To: "Adam Roach" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
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> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
>
> > From: Neil Deason [mailto:ndeason@ubiquity.net]
> >
> > Actually if we want to police protocol usage closely we had
> > better look at deprecating MESSAGE too.
>
> Come back when SIMPLE has finished defining the message sessions,
> and I'll throw my voice behind yours in advocating this.

What I was actually advocating here was ppl wanting to keep INFO around
write up something to illuminate INFO packages and provide example
usage.

The MESSAGE cf INFO was just an attempt to highlight an apparent
limitation with the ban INFO approach. Every MIME type in an INFO can be
moved into MESSAGE rather than go away. Maybe through exploring INFO
packages we evolve the collective understanding that this is/isn't a
necessary thing.

Cheers,
Neil.

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inline.

Neil Deason wrote:
>>-----Original Message-----
>>From: Adam Roach [mailto:adam@dynamicsoft.com]
>>
>>
>>>From: Neil Deason [mailto:ndeason@ubiquity.net]
>>>
>>>Actually if we want to police protocol usage closely we had
>>>better look at deprecating MESSAGE too.
>>
>>Come back when SIMPLE has finished defining the message sessions,
>>and I'll throw my voice behind yours in advocating this.
> 
> 
> What I was actually advocating here was ppl wanting to keep INFO around
> write up something to illuminate INFO packages and provide example
> usage.

Well, let me suggest an amendment to that.

It seems that people have a bunch of problems they want to solve, for 
which they are finding INFO a good solution. They have some requirements 
like, "support application to application messaging", and other folks, 
like myself, have requirements for things like congestion control and 
the ability to indicate what kind of usages are supported. It seems like 
RFC 3427 is screaming at us here. Someone needs to write down what 
problems they are trying to solve, and then sipping can debate them. If 
we choose to address them, the sip working group might conclude that it 
should revise INFO to meet those requirements, or it might decide to 
obsolete it in favor of a totally separate solution (like my proposed 
SDP-based approach).

Given the volume on this thread during the holiday season even, there 
should be no shortage of volunteers to write such a requirements doc. 
For once though, I am NOT volunteering ;)

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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CC: Adam Roach <adam@dynamicsoft.com>, "'Neil Deason'" <ndeason@ubiquity.net>,
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Subject: Re: [Sip] INFO considered harmful
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Henning is dead on.

I think we will always need MESSAGE for this reason exactly. It may be, 
that as messaging sessions are completed, and we understand the problem 
space better, we would revise MESSAGE to impose further restrictions on 
it. For now, I think its pretty good as written.

-Jonathan R.

Henning Schulzrinne wrote:
> Actually, there's an important difference: MESSAGE can use the SIP 
> features outside an established session, while INFO is within a session 
> (and thus there are other SDP-negotiated, options available). Within a 
> session, the only real reason for a follow-the-INVITE SIP message is 
> that a direct IP-level connection can't be made to work.
> 
>  From an efficiency standpoint, if you only have very occasional 
> messages, you gain by a "connectionless" approach. After all, an INVITE 
> transaction for a single MESSAGE doesn't buy you much and doesn't 
> conserve the oh-so-precious proxy cycles. While such short sessions are 
> probably not common for human-to-human communication (except spam...), 
> they may well be common for the type of things that SMS and pagers are 
> popular for: machine-to-human or machine-to-machine communications 
> announcing infrequent and unpredictable events ("your flight is now 
> leaving on gate 37", "your FedEx just arrived", "Server 42 is on fire").
> 
> Adam Roach wrote:
> 
>>> From: Neil Deason [mailto:ndeason@ubiquity.net]
>>>
>>> Actually if we want to police protocol usage closely we had better 
>>> look at deprecating MESSAGE too.
>>
>>
>>
>> Come back when SIMPLE has finished defining the message sessions,
>> and I'll throw my voice behind yours in advocating this.
>>
>> /a
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: hisham.khartabil@nokia.com
CC: peter.paeppinghaus@siemens.com, sip@ietf.org
Subject: Re: Repost: [Sip] Buggy example in RFC 3261
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Well, a decision needs to be made one way or another. Since its already 
in 3261, I would propose that we declare the example as buggy, and go 
with the existing normative text, even though it works differently than 
for requests.

-Jonathan R.

hisham.khartabil@nokia.com wrote:
> This is an extract from a discussion I had with Robert Sparks:
> 
> 
>>>Yes, that was exactly what I asked. If its not so, what does an Expires
>>>header in a 200 OK for a REGISTER mean?
>>
>>
>>As I note above, with the current text an Expire header field has no meaning
>>if it appears in a 200 response to a REGISTER request.
> 
> 
> http://www.ietf.org/mail-archive/working-groups/sip/current/msg03602.html
> 
> Regards,
> Hisham
> 
> 
>>-----Original Message-----
>>From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>Sent: Thursday, January 02, 2003 7:01 AM
>>To: Peter Päppinghaus
>>Cc: sip@ietf.org
>>Subject: Re: Repost: [Sip] Buggy example in RFC 3261
>>
>>
>>Sorry for not responding. I did notice this. There is an 
>>error here, as 
>>you point out. What is NOT clear is whether the error is in 
>>the example, 
>>or in the normative text. For REGISTER requests, you can use 
>>either the 
>>Expires header field or the expires contact header field 
>>parameter. It 
>>seems odd that this same rule would not apply to REGISTER 
>>responses. My 
>>personal recollection was that either the Expires header field or the 
>>expires contact param was OK for either request or response. This 
>>doesn't match the normative text though.
>>
>>Does anyone use the Expires header field and NOT the expires contact 
>>param in REGISTER responses?
>>
>>-Jonathan R.
>>
>>Peter Päppinghaus wrote:
>>
>>>I have posted this earlier, but it seems to have gone
>>>unnoticed.
>>>
>>>Hi,
>>>
>>>I believe to have stumbled on a buggy example in RFC 3261.
>>>
>>>The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
>>>an Expires header, but no expires parameter in the Contact header.
>>>
>>>This violates RFC 3261, sec. 10.3, bullet 8:
>>>"Each Contact value MUST feature an "expires" parameter
>>>  indicating its expiration interval chosen by the registrar."
>>>
>>>Regards,
>>>
>>>Peter Paeppinghaus
>>>
>>
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>>Chief Scientist                             First Floor
>>dynamicsoft                                 East Hanover, NJ 07936
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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Date: Fri, 03 Jan 2003 08:19:37 -0500
From: "David R. Oran" <oran@cisco.com>
To: Michael Thomas <mat@cisco.com>, Adam Roach <adam@dynamicsoft.com>
cc: "'George Foti (LMC)'" <George.Foti@ericsson.ca>,
        Eric Burger <eburger@snowshore.com>,
        Dean Willis <dean.willis@softarmor.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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I have nothing substantive to contribute to this discussion, except to 
observe that among my silly "Oran's laws of protocols", about five years 
ago I coined the following law:

"any application protocol which works sufficiently well to be widely 
deployed eventually gets hijacked and used as a transport protocol"

The fact that we are having this discussion is a testament to the success 
of SIP. History also seems to tell us that we may be fighting a losing 
battle in attempting to ban INFO and friends.

Happy New Year, to the whole SI community.

Dave.


--On Thursday, January 02, 2003 3:56 PM -0800 Michael Thomas 
<mat@cisco.com> wrote:

>
> Right, but my personal bugaboo is modularity.
> Like Don's FTPoSIP example, the more junk you
> throw into one protocol, the more entangled it
> becomes to the point that you have a deep dark
> morass. In any case, we already have a real live
> kitchen sink protocol: HTTP; you even run IP over
> HTTP these days. I'd be surprised to hear that
> just about anything that runs SIP doesn't also
> have web kruft in it as well.
>
> 	 Mike
>
> Adam Roach writes:
>  > A new method would be okay for new SIP-related functionality.
>  >
>  > But that's not what you proposed.
>  >
>  > You said, "The application is SIP based, we dont need yet
>  > another protocol to convey what needs to be conveyed e2e,"
>  > which is an argument that we should use SIP as a general
>  > purpose RPC mechanism for any application that already has
>  > a SIP stack.
>  >
>  > Doing what you propose with INFO is inappropriate.
>  >
>  > Doing what you propose with a new method is inappropriate.
>  >
>  > Doing what you propose with SIP is inappropriate.
>  >
>  > Since you keep harping on technical, as opposed to
>  > philosophical, arguments, here's a couple:
>  >
>  >   - The SIP protocol was designed for session initiation,
>  >     negotiation, and teardown. The timing characteristics
>  >     are not generally appropriate for application control.
>  >     For example, with typical SIP networks, latency and
>  >     jitter of messages are too high to provide any sort
>  >     of reasonable user experience for floor control
>  >     or DTMF transport.
>  >
>  >  - Many proxies stay in the signalling path for the purposes
>  >    of providing services. By sending unnecessary application-
>  >    level information throught these proxies, you reduce the
>  >    capacity and increase possible congestion of the network.
>  >
>  > I could go on, but this topic has had extensive airing on
>  > the list in years past. There's really no reason to beat
>  > this particular dead horse any longer.
>  >
>  > /a
>  >
>  > > -----Original Message-----
>  > > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
>  > > Sent: Thursday, January 02, 2003 16:05
>  > > To: 'Adam Roach'; George Foti (LMC); 'Michael Thomas'; Eric Burger
>  > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>  > > Subject: RE: [Sip] INFO considered harmful
>  > >
>  > >
>  > > This should not be a philosophical discusion.
>  > > If a new SIP method is okay, why is INFO not okay.
>  > >
>  > > What is the difference?
>  > > Please explain in technical terms?
>  > >
>  > > /gf
>  > >
>  > > -----Original Message-----
>  > > From: Adam Roach [mailto:adam@dynamicsoft.com]
>  > > Sent: Thursday, January 02, 2003 2:36 PM
>  > > To: 'George Foti (LMC)'; 'Michael Thomas'; Eric Burger
>  > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>  > > Subject: RE: [Sip] INFO considered harmful
>  > >
>  > >
>  > > The argument of "but I'm already holding a hammer and *want*
>  > > to treat everything like a nail!" comes up on about an annual
>  > > basis, and is rejected outright each time.
>  > >
>  > > Just because you have a SIP stack in your application does
>  > > *not* mean that it is appropriate to use that stack for
>  > > all general network access. You're allowing laziness to
>  > > overwhelm good application design.
>  > >
>  > > /a
>  > >
>  > > > -----Original Message-----
>  > > > From: George Foti (LMC) [mailto:George.Foti@ericsson.ca]
>  > > > Sent: Thursday, January 02, 2003 12:55
>  > > > To: 'Michael Thomas'; Eric Burger
>  > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>  > > > Subject: RE: [Sip] INFO considered harmful
>  > > >
>  > > >
>  > > > The application is SIP based, we dont need yet another
>  > > > protocol to convey what needs to be conveyed e2e.
>  > > > No need to go around the bush.
>  > > > Lets find *real* arguments against using INFO for that
>  > > > purpopse rather than philosphical  ones.
>  > > >
>  > > > /gf
>  > > >
>  > > >
>  > > >
>  > > >
>  > > > -----Original Message-----
>  > > > From: Michael Thomas [mailto:mat@cisco.com]
>  > > > Sent: Thursday, January 02, 2003 12:53 PM
>  > > > To: Eric Burger
>  > > > Cc: Dean Willis; Jonathan Rosenberg; sip@ietf.org
>  > > > Subject: RE: [Sip] INFO considered harmful
>  > > >
>  > > >
>  > > >
>  > > > I don't get it. If you want "100% application
>  > > > specific", why not pick an unused port and roll
>  > > > your own protocol? Then all of the crufty SIPism
>  > > > don't get in the way, and the hue and cry from
>  > > > this working group ceases immediately. Do the bits
>  > > > on the wire keep warmer if they're dressed in SIP?
>  > > >
>  > > >        Mike
>  > > >
>  > > > Eric Burger writes:
>  > > >  > I fully agree with Dean on this one.
>  > > >  >
>  > > >  > One point to consider (I'm on the fence on this one) is
>  > > > whether we would say that INFO is 100% application-specific.
>  > > > That is, the WG will not publish standard body types.
>  > > >  >
>  > > >  > A reason for doing that is to let people know that INFO
>  > > > really is just, as Dean points out,
>  > > > application-to-application communication.  If you want
>  > > > applications to interoperate, take it either to an
>  > > > Application Area WG or take it outside the IETF.
>  > > >  >
>  > > >  > One reason to not do this is SIP-T should use a method
>  > > > other than INFO.  That said, I don't think there are that
>  > > > many SIP-T implementations extant.
>  > > >  >
>  > > >  >
>  > > >  > > -----Original Message-----
>  > > >  > > From: Dean Willis [mailto:dean.willis@softarmor.com]
>  > > >  > > Sent: Tuesday, December 31, 2002 1:48 AM
>  > > >  > > To: Jonathan Rosenberg; Orit Levin
>  > > >  > > Cc: sip@ietf.org
>  > > >  > > Subject: Re: [Sip] INFO considered harmful
>  > > >  > >
>  > > >  > >
>  > > >  > > At 12:48 AM 12/31/2002 -0500, Jonathan Rosenberg wrote:
>  > > >  > > >Orit Levin wrote:
>  > > >  > > >>>The big huge difference I have been trying to point out is
>  > > >  > > that EVENTS
>  > > >  > > >>>DEFINES SEMANTICS, whereas INFO doesnt.
>  > > >  > > >>>MIME provides semantics and syntax that are independent of
>  > > >  > > the type.
>  > > >  > > >>>SIP events provides semantics and syntax that are
>  > > >  > > independent of the
>  > > >  > > >>>package.
>  > > >  > > >>>INFO provides nothing.
>  > > >  > > >>
>  > > >  > > >>INFO provides "minimal" but very important semantics:
>  > > >  > > asynchronous data
>  > > >  > > >>reliably follows the established SIP path.
>  > > >  > > >
>  > > >  > > >Huh?
>  > > >  > > >
>  > > >  > > >Every new SIP method is reliable, it inherits the SIP
>  > > >  > > transaction state
>  > > >  > > >machine for non-invite. Follows the established path? Every
>  > > >  > > in-dialog
>  > > >  > > >method would have this property, it is method independent.
>  > > >  > > >Asynchronous?  You can send any method at any time.
>  > > >  > > >
>  > > >  > > >Thus, nothing you have pointed out above is different
>  > > >  > > between INFO and any
>  > > >  > > >other new method we might introduce.
>  > > >  > >
>  > > >  > > I think the key point is that there is a difference
>  > > > between transport
>  > > >  > > protocol semantic and application semantic. The transport
>  > > >  > > protocol semantic
>  > > >  > > expressed in INFO is "Here is some data that will be
>  > > used by the
>  > > >  > > application. It is not important to the transport protocol,
>  > > >  > > except that the
>  > > >  > > transport protocol is expected to deliver it reliably". INFO
>  > > >  > > says NOTHING
>  > > >  > > about the APPLICATION level semantic -- that's up to what
>  > > >  > > goes IN the INFO
>  > > >  > > payloads. The question is -- do we try to rigidly define
>  > > >  > > application-level
>  > > >  > > semantics here? Do we define that there ARE no such
>  > > >  > > applications possible?
>  > > >  > > Or do we design a framework by which application-level
>  > > >  > > semantics can be
>  > > >  > > expressed outside of the protocol definition?
>  > > >  > >
>  > > >  > > And as for "Just add another method to your SIP stack". Let's
>  > > >  > > say I have a
>  > > >  > > mobile phone and the SIP stack is burned into ROM and
>  > > > exposes only a
>  > > >  > > simplistic transactional API. This is only likely to
>  > > > happen, oh, 100
>  > > >  > > million times or so over the next two years . . .  Now, just
>  > > >  > > exactly how is
>  > > >  > > my BREW or JTME application going to go about extending the
>  > > >  > > SIP stack to
>  > > >  > > support another method? Will the evil wireless operator
>  > > >  > > networks even pass
>  > > >  > > another method? Neither is happening anytime soon, I
>  > > think . . .
>  > > >  > >
>  > > >  > > --
>  > > >  > > Dean
>  > > >  > >
>  > > >  > >
>  > > >  > > _______________________________________________
>  > > >  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > >  > > This list is for NEW development of the core SIP Protocol
>  > > >  > > Use sip-implementors@cs.columbia.edu for questions on
>  > > current sip
>  > > >  > > Use sipping@ietf.org for new developments on the
>  > > > application of sip
>  > > >  > >
>  > > >  > _______________________________________________
>  > > >  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > >  > This list is for NEW development of the core SIP Protocol
>  > > >  > Use sip-implementors@cs.columbia.edu for questions on current
> sip  > > >  > Use sipping@ietf.org for new developments on the
>  > > application of sip
>  > > > _______________________________________________
>  > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > > This list is for NEW development of the core SIP Protocol
>  > > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > > Use sipping@ietf.org for new developments on the application of sip
>  > > > _______________________________________________
>  > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > > This list is for NEW development of the core SIP Protocol
>  > > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > > Use sipping@ietf.org for new developments on the application of sip
>  > > >
>  > >
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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And I will support that.

There may be a place for session based messaging, but for all those uses where cellphone SMS constraints work today, MESSAGE seems to be the appropriate solution.

By the way, when we do get session based messaging, it will probably be impossible to write scoping rules that prevent someone sending two MESSAGE requests to the same destination, rather than setting up a session and sending two PDUs within that session. All we will be able to do is provide guidance.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: 03 January 2003 06:19
> To: Henning Schulzrinne
> Cc: Adam Roach; 'Neil Deason'; sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> Henning is dead on.
> 
> I think we will always need MESSAGE for this reason exactly. 
> It may be, 
> that as messaging sessions are completed, and we understand 
> the problem 
> space better, we would revise MESSAGE to impose further 
> restrictions on 
> it. For now, I think its pretty good as written.
> 
> -Jonathan R.
> 
> Henning Schulzrinne wrote:
> > Actually, there's an important difference: MESSAGE can use the SIP 
> > features outside an established session, while INFO is 
> within a session 
> > (and thus there are other SDP-negotiated, options 
> available). Within a 
> > session, the only real reason for a follow-the-INVITE SIP 
> message is 
> > that a direct IP-level connection can't be made to work.
> > 
> >  From an efficiency standpoint, if you only have very occasional 
> > messages, you gain by a "connectionless" approach. After 
> all, an INVITE 
> > transaction for a single MESSAGE doesn't buy you much and doesn't 
> > conserve the oh-so-precious proxy cycles. While such short 
> sessions are 
> > probably not common for human-to-human communication 
> (except spam...), 
> > they may well be common for the type of things that SMS and 
> pagers are 
> > popular for: machine-to-human or machine-to-machine communications 
> > announcing infrequent and unpredictable events ("your flight is now 
> > leaving on gate 37", "your FedEx just arrived", "Server 42 
> is on fire").
> > 
> > Adam Roach wrote:
> > 
> >>> From: Neil Deason [mailto:ndeason@ubiquity.net]
> >>>
> >>> Actually if we want to police protocol usage closely we 
> had better 
> >>> look at deprecating MESSAGE too.
> >>
> >>
> >>
> >> Come back when SIMPLE has finished defining the message sessions,
> >> and I'll throw my voice behind yours in advocating this.
> >>
> >> /a
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Re: [Sip] INFO considered harmful
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David R. Oran wrote:
> I have nothing substantive to contribute to this discussion, except to 
> observe that among my silly "Oran's laws of protocols", about five years 
> ago I coined the following law:
> 
> "any application protocol which works sufficiently well to be widely 
> deployed eventually gets hijacked and used as a transport protocol"

Definitely true.

Please do enumerate the others... would make a nice I-D I think.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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Subject: Re: [Sip] INFO considered harmful
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Jonathan Rosenberg writes:
 > 
 > 
 > David R. Oran wrote:
 > > I have nothing substantive to contribute to this discussion, except to 
 > > observe that among my silly "Oran's laws of protocols", about five years 
 > > ago I coined the following law:
 > > 
 > > "any application protocol which works sufficiently well to be widely 
 > > deployed eventually gets hijacked and used as a transport protocol"
 > 
 > Definitely true.
 > 
 > Please do enumerate the others... would make a nice I-D I think.

   I don't think anybody should be under the
   illusion that SIP won't be misused. What's at
   issue here specifically is whether the WG
   actually sanctions a free for all. One distinct 
   downside is that there may actually be some
   gems that ought to be salvaged from the fetid
   swamp, but that will be extremely difficult if
   they work well enough. It just seems like a 
   poor idea and precedent.

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



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Subject: RE: [Sip] INFO considered harmful
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The ironic part of the example:

SMS is given as one of the reasons that wireless is so cool and why wireless operators can make money.

SMS, in its present form, that is, not what "it could be", but what it IS, is given as an example of how to melt-down a signaling infrastructure.  SMS was supposed to be very infrequent, so it was thought to be OK to run it on the SS7 infrastructure.  No one anticipated the volume of point-to-point traffic SMS would generate.  One reason SMS took a long time to deploy in the US is because the Bellheads running the US wireless networks said, "SMS will overrun our poor STPs -- we can never deploy such a service."

Replace SMS with INFO, replace SS7 with SIP, and replace STP with SIP Proxy.

Not a value judgment, just an interesting look at how people that did the "wrong" thing made a boatload of money and managed not to melt the network (when there is a will and lots of money, a way gets found), and people doing the "right" thing are still trying to figure out if they should go Chapter 11...

> -----Original Message-----
> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> Sent: Friday, January 03, 2003 11:20 AM
> To: Jonathan Rosenberg; Henning Schulzrinne
> Cc: Adam Roach; 'Neil Deason'; sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> And I will support that.
> 
> There may be a place for session based messaging, but for all 
> those uses where cellphone SMS constraints work today, 
> MESSAGE seems to be the appropriate solution.
> 
> By the way, when we do get session based messaging, it will 
> probably be impossible to write scoping rules that prevent 
> someone sending two MESSAGE requests to the same destination, 
> rather than setting up a session and sending two PDUs within 
> that session. All we will be able to do is provide guidance.
> 
> Keith
> 
> Keith Drage
> Lucent Technologies
> Tel: +44 1793 776249
> Email: drage@lucent.com 
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: 03 January 2003 06:19
> > To: Henning Schulzrinne
> > Cc: Adam Roach; 'Neil Deason'; sip@ietf.org
> > Subject: Re: [Sip] INFO considered harmful
> > 
> > 
> > Henning is dead on.
> > 
> > I think we will always need MESSAGE for this reason exactly. 
> > It may be, 
> > that as messaging sessions are completed, and we understand 
> > the problem 
> > space better, we would revise MESSAGE to impose further 
> > restrictions on 
> > it. For now, I think its pretty good as written.
> > 
> > -Jonathan R.
> > 
> > Henning Schulzrinne wrote:
> > > Actually, there's an important difference: MESSAGE can 
> use the SIP 
> > > features outside an established session, while INFO is 
> > within a session 
> > > (and thus there are other SDP-negotiated, options 
> > available). Within a 
> > > session, the only real reason for a follow-the-INVITE SIP 
> > message is 
> > > that a direct IP-level connection can't be made to work.
> > > 
> > >  From an efficiency standpoint, if you only have very occasional 
> > > messages, you gain by a "connectionless" approach. After 
> > all, an INVITE 
> > > transaction for a single MESSAGE doesn't buy you much and doesn't 
> > > conserve the oh-so-precious proxy cycles. While such short 
> > sessions are 
> > > probably not common for human-to-human communication 
> > (except spam...), 
> > > they may well be common for the type of things that SMS and 
> > pagers are 
> > > popular for: machine-to-human or machine-to-machine 
> communications 
> > > announcing infrequent and unpredictable events ("your 
> flight is now 
> > > leaving on gate 37", "your FedEx just arrived", "Server 42 
> > is on fire").
> > > 
> > > Adam Roach wrote:
> > > 
> > >>> From: Neil Deason [mailto:ndeason@ubiquity.net]
> > >>>
> > >>> Actually if we want to police protocol usage closely we 
> > had better 
> > >>> look at deprecating MESSAGE too.
> > >>
> > >>
> > >>
> > >> Come back when SIMPLE has finished defining the message sessions,
> > >> and I'll throw my voice behind yours in advocating this.
> > >>
> > >> /a
> > >> _______________________________________________
> > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > >> This list is for NEW development of the core SIP Protocol
> > >> Use sip-implementors@cs.columbia.edu for questions on current sip
> > >> Use sipping@ietf.org for new developments on the 
> application of sip
> > > 
> > > 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > > 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.jdrosen.net                      PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Juha Heinanen writes:
 > Michael Thomas writes:
 > 
 >  > The problem is that AAA's expect either a CHAP or
 >  > PAP formated authenticator. Digest, while it
 >  > doesn't chip the cleartext password ain't the same
 >  > format so you have a problem. Basic allows you to
 >  > hack up an authenticator at the proxy. Maybe
 >  > EAP+digest is our friend. Or maybe not.
 > 
 > what is wrong with draft-sterman-aaa-sip-00.txt?  we use it with digets
 > authentication without any problems.  leading radius servers support it
 > as well as all good sip proxies.

   Well, it seems to be expried for one... is the jist
   that AAA's have been hacked to deal with digest auth
   types directly (or, say, as an EAP type)? If
   so, that's certainly a possibility, and would
   have the obvious benefit of being useable for
   the web as well. 

       Mike
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Michael Thomas writes:

 >    Well, it seems to be expried for one... is the jist
 >    that AAA's have been hacked to deal with digest auth
 >    types directly (or, say, as an EAP type)? 

i made draft-sterman-aaa-sip-00.txt available as

ftp://lohi.eng.song.fi/tmp/draft-sterman-aaa-sip-00.txt

the directory is not listable.

-- juha

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To add additional perspective: In 2001, 200 billion SMS were sent (for 
GSM, 
http://www.idg.net/idgns/2001/05/29/UPDATE2StudySMSMessagesWillTop200.shtml). 
Comes to about 6400/second. Assuming that a reasonably well tuned server 
can handle 100/messages a second, that requires about 100 servers, 
multiplied by the number of network hops (residential and small-business 
end hops don't matter since they are not likely to be congested). Total 
cost some small multiple of $100k. This total would be sufficient to 
handle the service for all GSM carriers, worldwide. I don't think that's 
a major factor in any protocol debate.

In general, I just don't buy server load arguments. If it's a useful 
service, server cost is usually the least of the problem. In any event, 
one should take this as a fraction of the existing signaling load, not 
as some absolute number. I'm worried about stuff that multiplies the 
signaling load by a factor of 10; I find 20 or 50% load increase 
uninteresting as an argument. (Thus, while I find 'INFO-for-DTMF' 
distasteful, the signaling load is the least of the problems, given that 
most phone calls will see at most a handful of touch tone digits.)

Eric Burger wrote:
> The ironic part of the example:
> 
> SMS is given as one of the reasons that wireless is so cool and why wireless operators can make money.
> 
> SMS, in its present form, that is, not what "it could be", but what it IS, is given as an example of how to melt-down a signaling infrastructure.  SMS was supposed to be very infrequent, so it was thought to be OK to run it on the SS7 infrastructure.  No one anticipated the volume of point-to-point traffic SMS would generate.  One reason SMS took a long time to deploy in the US is because the Bellheads running the US wireless networks said, "SMS will overrun our poor STPs -- we can never deploy such a service."
> 
> Replace SMS with INFO, replace SS7 with SIP, and replace STP with SIP Proxy.
> 
> Not a value judgment, just an interesting look at how people that did the "wrong" thing made a boatload of money and managed not to melt the network (when there is a will and lots of money, a way gets found), and people doing the "right" thing are still trying to figure out if they should go Chapter 11...
> 
> 
>>-----Original Message-----
>>From: Drage, Keith (Keith) [mailto:drage@lucent.com]
>>Sent: Friday, January 03, 2003 11:20 AM
>>To: Jonathan Rosenberg; Henning Schulzrinne
>>Cc: Adam Roach; 'Neil Deason'; sip@ietf.org
>>Subject: RE: [Sip] INFO considered harmful
>>
>>
>>And I will support that.
>>
>>There may be a place for session based messaging, but for all 
>>those uses where cellphone SMS constraints work today, 
>>MESSAGE seems to be the appropriate solution.
>>
>>By the way, when we do get session based messaging, it will 
>>probably be impossible to write scoping rules that prevent 
>>someone sending two MESSAGE requests to the same destination, 
>>rather than setting up a session and sending two PDUs within 
>>that session. All we will be able to do is provide guidance.
>>
>>Keith
>>
>>Keith Drage
>>Lucent Technologies
>>Tel: +44 1793 776249
>>Email: drage@lucent.com 
>>
>>
>>>-----Original Message-----
>>>From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>>Sent: 03 January 2003 06:19
>>>To: Henning Schulzrinne
>>>Cc: Adam Roach; 'Neil Deason'; sip@ietf.org
>>>Subject: Re: [Sip] INFO considered harmful
>>>
>>>
>>>Henning is dead on.
>>>
>>>I think we will always need MESSAGE for this reason exactly. 
>>>It may be, 
>>>that as messaging sessions are completed, and we understand 
>>>the problem 
>>>space better, we would revise MESSAGE to impose further 
>>>restrictions on 
>>>it. For now, I think its pretty good as written.
>>>
>>>-Jonathan R.
>>>
>>>Henning Schulzrinne wrote:
>>>
>>>>Actually, there's an important difference: MESSAGE can 
>>
>>use the SIP 
>>
>>>>features outside an established session, while INFO is 
>>>
>>>within a session 
>>>
>>>>(and thus there are other SDP-negotiated, options 
>>>
>>>available). Within a 
>>>
>>>>session, the only real reason for a follow-the-INVITE SIP 
>>>
>>>message is 
>>>
>>>>that a direct IP-level connection can't be made to work.
>>>>
>>>> From an efficiency standpoint, if you only have very occasional 
>>>>messages, you gain by a "connectionless" approach. After 
>>>
>>>all, an INVITE 
>>>
>>>>transaction for a single MESSAGE doesn't buy you much and doesn't 
>>>>conserve the oh-so-precious proxy cycles. While such short 
>>>
>>>sessions are 
>>>
>>>>probably not common for human-to-human communication 
>>>
>>>(except spam...), 
>>>
>>>>they may well be common for the type of things that SMS and 
>>>
>>>pagers are 
>>>
>>>>popular for: machine-to-human or machine-to-machine 
>>
>>communications 
>>
>>>>announcing infrequent and unpredictable events ("your 
>>
>>flight is now 
>>
>>>>leaving on gate 37", "your FedEx just arrived", "Server 42 
>>>
>>>is on fire").
>>>
>>>>Adam Roach wrote:
>>>>
>>>>
>>>>>>From: Neil Deason [mailto:ndeason@ubiquity.net]
>>>>>>
>>>>>>Actually if we want to police protocol usage closely we 
>>>
>>>had better 
>>>
>>>>>>look at deprecating MESSAGE too.
>>>>>
>>>>>
>>>>>
>>>>>Come back when SIMPLE has finished defining the message sessions,
>>>>>and I'll throw my voice behind yours in advocating this.
>>>>>
>>>>>/a
>>>>>_______________________________________________
>>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>This list is for NEW development of the core SIP Protocol
>>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>Use sipping@ietf.org for new developments on the 
>>
>>application of sip
>>
>>>>
>>>>_______________________________________________
>>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>This list is for NEW development of the core SIP Protocol
>>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>Use sipping@ietf.org for new developments on the 
>>
>>application of sip
>>
>>>-- 
>>>Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>>>Chief Scientist                             First Floor
>>>dynamicsoft                                 East Hanover, NJ 07936
>>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>>http://www.dynamicsoft.com
>>>
>>>_______________________________________________
>>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>This list is for NEW development of the core SIP Protocol
>>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>>Use sipping@ietf.org for new developments on the application of sip
>>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
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Subject: RE: [Sip] INFO considered harmful
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> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>
> In general, I just don't buy server load arguments. If it's a
> useful service, server cost is usually the least of the problem.
> In any event, one should take this as a fraction of the existing
> signaling load, not as some absolute number. I'm worried about
> stuff that multiplies the signaling load by a factor of 10; I find
> 20 or 50% load increase uninteresting as an argument.

You don't buy server load arguments? Real-world experience
notwithstanding?

I'm recalling the rise in popularity of Real Media Player and
its ability to set up an audio stream through an HTTP proxy.
In particular, I'm recalling conversations shortly thereafter
with some friends over in MIS about how they just couldn't add
capacity to the HTTP proxy pool fast enough to keep up with
demand.

Access to external web documents sometimes creeped up to the
order of minutes before everything was finally stabilized.

How long do you guess it will be before Real player
(or a similar product) includes a "SIP Proxy:" field right
next to its "HTTP Proxy:" field for firewall circumvention?

/a
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Adam Roach <adam@dynamicsoft.com>
CC: Eric Burger <eburger@snowshore.com>, sip@ietf.org
Subject: Re: [Sip] INFO considered harmful
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Your example clearly falls into the "> 10" category as the poor web 
server had to deal with every Real audio or video packet. Not that it 
would surprise me, but I don't think anybody *here* has suggested 
RTP-over-INFO so far.

Adam Roach wrote:
>>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>>
>>In general, I just don't buy server load arguments. If it's a
>>useful service, server cost is usually the least of the problem.
>>In any event, one should take this as a fraction of the existing
>>signaling load, not as some absolute number. I'm worried about
>>stuff that multiplies the signaling load by a factor of 10; I find
>>20 or 50% load increase uninteresting as an argument.
> 
> 
> You don't buy server load arguments? Real-world experience
> notwithstanding?
> 
> I'm recalling the rise in popularity of Real Media Player and
> its ability to set up an audio stream through an HTTP proxy.
> In particular, I'm recalling conversations shortly thereafter
> with some friends over in MIS about how they just couldn't add
> capacity to the HTTP proxy pool fast enough to keep up with
> demand.
> 
> Access to external web documents sometimes creeped up to the
> order of minutes before everything was finally stabilized.
> 
> How long do you guess it will be before Real player
> (or a similar product) includes a "SIP Proxy:" field right
> next to its "HTTP Proxy:" field for firewall circumvention?
> 
> /a

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Michael Thomas'" <mat@cisco.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Fri, 3 Jan 2003 15:37:29 -0600
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Mike asked:
> I don't get it. If you want "100% application
> specific", why not pick an unused port and roll
> your own protocol? Then all of the crufty SIPism
> don't get in the way, and the hue and cry from
> this working group ceases immediately. Do the bits 
> on the wire keep warmer if they're dressed in SIP?

Not only do they keep warmer, but they get to take advantage of all those
nifty SIP functions, like rendezvous, delivery delegation, security, and
MOST IMPORTANTLY, name-space correlation with the application naming
provided by SIP. In short, all of the same reasons why PUBLISH-type methods
kep getting suggested.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Fri, 3 Jan 2003 15:37:29 -0600
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> Just one moment. SIP is not a transport protocol as its defined in my 
> book. Although SIP is generic in that it can support lots of 
> applications (conferencing, IVR, gaming, etc.), it is not unbounded. 
> SIP's messages provide semantics common to all applications which 
> require person-to-person communications. That does NOT make it a 
> transport prorotocol for arbitrary data communication.

Well, I think that's exactly where we disagree, and I believe that the vast
majority of people using SIP think of it a lot more like I do. At least the
ones in 3G-space seem to.

Here's what I think I believe:

SIP provides for the establishment of a session (dialog) between two or more
end points, optionally intermediated by proxies, and the exchange of
information between those end points. This information is applied by
applications to provide application functionality. SIP provides several
methods for relaying application information, including relaying a text
message, requesting notification of changes in the state of a named element,
reporting a change in the state of some named element, and so on. It is
either a historical accident that the dialog-initiation request (INVITE) is
non-transparently overloaded with media session negotiation OR that the
media session negotiation method (INVITE) establishes a dialog on which
other requests may be relayed. 

> The stack has to allow extensibility to pass unknown INFO 
> bodies up to 
> the application for handling. Why this is any different than having 
> extensibility to handle unknown methods up to the application, is a 
> mystery to me.

Nope. There's no extension required to the protocol stack to deliver one
type of MIME body versus deliverying another type of MIME body. It's just a
vyte-vector in a buffer. Otherwise, we'd be rewriting sendmail and httpd
every 47 minutes to support a new body type.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
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Subject: RE: [Sip] INFO considered harmful
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Adam said:
> You said, "The application is SIP based, we dont need yet 
> another protocol to convey what needs to be conveyed e2e," 
> which is an argument that we should use SIP as a general 
> purpose RPC mechanism for any application that already has a 
> SIP stack.
> 
> Doing what you propose with INFO is inappropriate.
> 
> Doing what you propose with a new method is inappropriate.
> 
> Doing what you propose with SIP is inappropriate.

The key point is that we already have two applications talking to each other
on a SIP dialog. The SIP intermediaries are providing the routing function
between those applications. The ONLY routing knowledge those applications
have about each other is that expressed in the SIP channel. Therefore, the
existing SIP dialog makes an awfully handy mechanism for exchanging data
between those two aplications.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Don Stanwyck'" <don@stanwyck.com>,
        "'George Foti \(LMC\)'" <George.Foti@ericsson.ca>,
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Don said:
> 2) If the apps need to talk directly to each other there are 
> a host of protocols that are extremely trivial that can do 
> the job.  The apps can even just open a port over any 
> transport protocol and exchange information as they please.  
> There is no reason to bother all the routing proxies in the 
> network with your e2e private chatter.

Not really, because the only thing that one app knows about another is its
SIP URL. Yeah, I suppose you could use another INVITE transaction to
establish a new dialog to convey the e2e stuff, but since we ALREADY have a
dialog established (INFO runs on an existing dialog, right?) it's a hell of
lot easier to just use the existing dialog. Especially given that the
existing dialog has the property of being connected to exactly the right
instance of the remote application.

In short, we can ratehr obviously use SIP to talk to a SIP-enabled
application that we're already talking to using SIP. 

--
Dean


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Dean Willis writes:
 > Mike asked:
 > > I don't get it. If you want "100% application
 > > specific", why not pick an unused port and roll
 > > your own protocol? Then all of the crufty SIPism
 > > don't get in the way, and the hue and cry from
 > > this working group ceases immediately. Do the bits 
 > > on the wire keep warmer if they're dressed in SIP?
 > 
 > Not only do they keep warmer, but they get to take advantage of all those
 > nifty SIP functions, like rendezvous, delivery delegation, security, and
 > MOST IMPORTANTLY, name-space correlation with the application naming
 > provided by SIP. In short, all of the same reasons why PUBLISH-type methods
 > kep getting suggested.

Dean,

It's patently obvious _why_ people want to do this
not unlike Mallory and Everest ("because it's
there!"). The question is whether it's a Good
Idea. The thing that squicks me is the anything
goes attitude parading around as "application
data". Well, SIP is already chock full of various
bits and pieces of "application data" and they
have well defined syntax and semantics. Why is it
OK to have INFO bits which are completely
unspecified but all the rest are well specified?
It's just inviting interoperability problems and
ad hoc standards making. Now I have no doubt that
this is going to happen no matter what, but it
doesn't mean that IETF needs to build and maintain
its kitty litter box.

Others have brought up SMS as an example. In fact,
I think that SMS argues *exactly* the opposite:
that sort of functionality needs to be very well
specified so that interoperablity is achieved.
Leaving it up to individual vendors, or providers
or whatever would be a huge mess. For less
interesting things... well, proprietary methods,
X-blahblahblah's and all of the rest can be hacked
up without any help whatsoever from IETF. If they
become interesting, we can then standardize them.
What I just don't see any value to is an
officially sanctioned free for all. What purpose
does it serve other than people getting to point
to a nice shiny RFC number for their proprietary
extensions?

		Mike
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Frank W. Miller'" <fmiller@sentito.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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My mother said I should stop inflaming things, but here I go again anyhow.
Sorry, Mom.

Adam said, responding to Frank in our long-winded exchange:
> Perhaps using some sort of protocol to initate this session 
> would be appropriate. You'll also needs a protocol of some 
> sort to describe it, probably carried as a payload in this 
> session initation protocol. Ideally, this session description 
> protocol could leverage IANA-registered media types to 
> describe the content of such sessions, so that an infinite 
> number of types of sessions can be described.

Actually, Adam and I just had a conversation about this argument. I think
actually that there is something hiding in here which is THE HEART of almost
every philosophical debate we've ever had in SIP. If we can get this nailed
down now, we'll be preventing a HUGE number of future fights.

Or in short: Boom! Epiphany! I think I understand what we're arguing about.

I'll admit -- Adam's been philosophically consistent in his arguments since
day 1.  He believes that SIP should start sessions, stop sessions, and
nothing else.

Examples:

Lets look at "events". By Adam's logic, the event system should consist of
sending an INVITE (instead of a SUBSCRIBE) to the event source. The SDP body
(Only SDP should be allowed in INVITE) would descibe the events being
requested and the protocol, IP address and port number on which the
subscriber would be listening. The 200 OK response from the event source
would then contain the protocol, IP address and port number to which the
subscriber should send acknowledgements of notifications, assuming that the
notification-transport protocol (clearly, not SIP).

And Whoa! Look at REGISTER. That doesn't set up any kind of session! It's
clearly not a SIP function. We should instead use an INVITE to set up a
"registration session", over which we would then use a registration protocol
to establish registrations.

MESSAGE is clearly divergent. By Adam's logic, only message sessions should
be used. By similar reasoning, INFO is also off, and instead SIP-T should
use INVITE to establish a sigtran session.

This actually all makes sense and hangs together philosophically.

The problem is, the working group diverted from that philosophy a long time
ago. What we have now is a combination of three transactional protocols
(REGISTER, OPTIONS, MESSAGE) with a session-establishment protocol (INVITE)
which supports several in-session request verbs (UPDATE, OPTIONS, ACK, INFO,
MESSAGE, BYE, PRACK, etc.) and another session-establishment protocol
(SUBSCRIBE) which supports an in-session request verb (NOTIFY). We've also
defined an intermediation framework (the SIP Proxy) and a set of end-to-end
and hop-by-hop authentication and integrity protection mechanisms that can
be used for all these protocols. Really, it's the intermediation framework
that gives SIP most of its utility.

Ok, it could be tidier. But it is very powerful, fairly complete, and
actually works. And yes, the combination effectively provides a very useful
transport protocol that does all sorts of things that HTTP alone can't do,
such as intermediated rendezvous. And no, it DOESN'T just do phone calls
(making attempts to model proxy load based on telephone switch metrics
hopelessly bogus).

So my humble suggestion: Get over it. Iron out any kruft that makes SIP less
desirable as a transport protocol (like UDP), and get on with the business
of making stuff that people can use to get their jobs done. Stop complaining
that your proxies are getting loaded down by serving a useful purpose to
somebody, and start understanding what that purpose is and how to charge for
it.

If you don't like it, fine. Let's take out a poll on deprecating REGISTER,
MESSAGE, SUBSCRIBE, INFO, PRACK, SUBSCRIBE, NOTIFY, and all that other
stuff.

Or hey, just toss it all out and add session negotiation to XMPP. 

--
Dean

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INFO = BEEP-like functionality over SIP :-)?

To amplify one part of what Dean said: Part of the functionality that an 
SDP-negotiated channel does not (readily) provide is different degrees 
of specificity. In SDP, I have to include a single IP address or an 
A-record. In SIP, Contact can be significantly more general, since it 
allows different degrees of binding detail and much more resolution 
functionality (SRV, NAPTR), including using the same old generic address 
that started the session and thus redoing the resolution with every 
request. This makes life much easier for sparse sessions with user 
mobility (same user, different device). This matters less when you 
already have a set of media sessions that need to be re-negotiated 
anyway when things change.


Dean Willis wrote:

> 
> SIP provides for the establishment of a session (dialog) between two or more
> end points, optionally intermediated by proxies, and the exchange of
> information between those end points. This information is applied by
> applications to provide application functionality. SIP provides several
> methods for relaying application information, including relaying a text
> message, requesting notification of changes in the state of a named element,
> reporting a change in the state of some named element, and so on. It is
> either a historical accident that the dialog-initiation request (INVITE) is
> non-transparently overloaded with media session negotiation OR that the
> media session negotiation method (INVITE) establishes a dialog on which
> other requests may be relayed. 
> 
> 

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inline.

Henning Schulzrinne wrote:

> In general, I just don't buy server load arguments. If it's a useful 
> service, server cost is usually the least of the problem. In any event, 
> one should take this as a fraction of the existing signaling load, not 
> as some absolute number. I'm worried about stuff that multiplies the 
> signaling load by a factor of 10; I find 20 or 50% load increase 
> uninteresting as an argument.

My firsthand experience with selling stuff to carriers is that this is 
most definitely not the way they see it. Remember, the equipment 
purchases are made up front before the service is activated. If you can 
sell a carrier a solution that requires 50% or even 20% fewer boxes, 
thats a big win. Its not just the equipment cost, its the management 
cost too that increases with the box counts.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Dean,

The lines are not always so black and white. Its not a choice between 
"do anything" and "do nothing but INVITE". There is a line somewhere 
about what is a good use of sip, and what is not. We have some 
documents, even, which help to identify that line (SIP guidelines), and 
no doubt it has moved over time. Indeed, there were even arguments about 
whether BYE belonged in SIP, since clearly its not needed for 
"initiation". However, if you buy the assertion that SIP needs to at 
least manage the session, that clearly includes termination as well as 
initiation. These days, I know of very few folks (only one in fact) that 
would argue against BYE as part of SIP.

I still stand by my assertion that many of the usages of SIP INFO are 
over that line, and that in order to keep the usages in a line we are 
comfortable with, something has to be done.

-Jonathan R.

Dean Willis wrote:
> My mother said I should stop inflaming things, but here I go again anyhow.
> Sorry, Mom.
> 
> Adam said, responding to Frank in our long-winded exchange:
> 
>>Perhaps using some sort of protocol to initate this session 
>>would be appropriate. You'll also needs a protocol of some 
>>sort to describe it, probably carried as a payload in this 
>>session initation protocol. Ideally, this session description 
>>protocol could leverage IANA-registered media types to 
>>describe the content of such sessions, so that an infinite 
>>number of types of sessions can be described.
> 
> 
> Actually, Adam and I just had a conversation about this argument. I think
> actually that there is something hiding in here which is THE HEART of almost
> every philosophical debate we've ever had in SIP. If we can get this nailed
> down now, we'll be preventing a HUGE number of future fights.
> 
> Or in short: Boom! Epiphany! I think I understand what we're arguing about.
> 
> I'll admit -- Adam's been philosophically consistent in his arguments since
> day 1.  He believes that SIP should start sessions, stop sessions, and
> nothing else.
> 
> Examples:
> 
> Lets look at "events". By Adam's logic, the event system should consist of
> sending an INVITE (instead of a SUBSCRIBE) to the event source. The SDP body
> (Only SDP should be allowed in INVITE) would descibe the events being
> requested and the protocol, IP address and port number on which the
> subscriber would be listening. The 200 OK response from the event source
> would then contain the protocol, IP address and port number to which the
> subscriber should send acknowledgements of notifications, assuming that the
> notification-transport protocol (clearly, not SIP).
> 
> And Whoa! Look at REGISTER. That doesn't set up any kind of session! It's
> clearly not a SIP function. We should instead use an INVITE to set up a
> "registration session", over which we would then use a registration protocol
> to establish registrations.
> 
> MESSAGE is clearly divergent. By Adam's logic, only message sessions should
> be used. By similar reasoning, INFO is also off, and instead SIP-T should
> use INVITE to establish a sigtran session.
> 
> This actually all makes sense and hangs together philosophically.
> 
> The problem is, the working group diverted from that philosophy a long time
> ago. What we have now is a combination of three transactional protocols
> (REGISTER, OPTIONS, MESSAGE) with a session-establishment protocol (INVITE)
> which supports several in-session request verbs (UPDATE, OPTIONS, ACK, INFO,
> MESSAGE, BYE, PRACK, etc.) and another session-establishment protocol
> (SUBSCRIBE) which supports an in-session request verb (NOTIFY). We've also
> defined an intermediation framework (the SIP Proxy) and a set of end-to-end
> and hop-by-hop authentication and integrity protection mechanisms that can
> be used for all these protocols. Really, it's the intermediation framework
> that gives SIP most of its utility.
> 
> Ok, it could be tidier. But it is very powerful, fairly complete, and
> actually works. And yes, the combination effectively provides a very useful
> transport protocol that does all sorts of things that HTTP alone can't do,
> such as intermediated rendezvous. And no, it DOESN'T just do phone calls
> (making attempts to model proxy load based on telephone switch metrics
> hopelessly bogus).
> 
> So my humble suggestion: Get over it. Iron out any kruft that makes SIP less
> desirable as a transport protocol (like UDP), and get on with the business
> of making stuff that people can use to get their jobs done. Stop complaining
> that your proxies are getting loaded down by serving a useful purpose to
> somebody, and start understanding what that purpose is and how to charge for
> it.
> 
> If you don't like it, fine. Let's take out a poll on deprecating REGISTER,
> MESSAGE, SUBSCRIBE, INFO, PRACK, SUBSCRIBE, NOTIFY, and all that other
> stuff.
> 
> Or hey, just toss it all out and add session negotiation to XMPP. 
> 
> --
> Dean
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Neil Deason'" <ndeason@ubiquity.net>,
        "'Adam Roach'" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Fri, 3 Jan 2003 17:13:20 -0600
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> What I was actually advocating here was ppl wanting to keep 
> INFO around write up something to illuminate INFO packages 
> and provide example usage.
> 
> The MESSAGE cf INFO was just an attempt to highlight an 
> apparent limitation with the ban INFO approach. Every MIME 
> type in an INFO can be moved into MESSAGE rather than go 
> away. Maybe through exploring INFO packages we evolve the 
> collective understanding that this is/isn't a necessary thing.
> 
> Cheers,
> Neil.

This is an excellent suggestion, and very similar to the one Jonathan made
to me on the phone yesterday (actually, I think he more tersely worded it as
"Send Text!")

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Fri, 3 Jan 2003 17:15:25 -0600
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> The big huge difference I have been trying to point out is 
> that EVENTS 
> DEFINES SEMANTICS, whereas INFO doesn't.

No, EVENTS doesn't define semantics. Event PACKAGES define the semantics of
particular instances of the event methods. All that 3265 gives us is
subscription to and notification of otherwise undefined blobs, and a
framework for defining event packages. It is up to a specific event package
to define a specific blob, including how it is named, how it is encoded, and
what it means. And one of the interesting things is that 3265 and sipchange
establishes a lower threshold-of-review over event packages (which are NOT
considered protocol extensions) than for new SIP methods (which obviously
ARE protocol extensions).

INFO also needs a blob-describing-and-encoding framework, so that the
various uses of INFO can, as appropriate, be effectively documented and
negotiated. 


--
Dean

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Date: Fri, 03 Jan 2003 17:18:13 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] INFO considered harmful
In-reply-to: <3E15B862.6090201@dynamicsoft.com>
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        "'David R. Oran'" <oran@cisco.com>
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>would make a nice I-D I think.

This whole discussion would make a very USEFUL I-D. Anyone contemplating
to do the work?

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Jonathan Rosenberg
> Sent: Friday, January 03, 2003 10:21 AM
> To: David R. Oran
> Cc: Michael Thomas; Adam Roach; 'George Foti (LMC)'; Eric 
> Burger; Dean Willis; sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> 
> 
> David R. Oran wrote:
> > I have nothing substantive to contribute to this 
> discussion, except to
> > observe that among my silly "Oran's laws of protocols", 
> about five years 
> > ago I coined the following law:
> > 
> > "any application protocol which works sufficiently well to be widely
> > deployed eventually gets hijacked and used as a transport protocol"
> 
> Definitely true.
> 
> Please do enumerate the others... would make a nice I-D I think.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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From: "Frank W. Miller" <fmiller@sentito.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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Well, any illusion of reasonableness I may have garnered in the past is
about to be shattered.  I would refer folks to:

http://www.ietf.org/internet-drafts/draft-miller-sip-isup-annex-00.txt

This draft is about to receive a major -01 update that recodes the entire
syntax in XML.  In addition, we (I) have another draft that does something
very similar for ANSI TCAP that I'll be submitting in short order.  I guess
its only appropriate that I bring peoples attention to these drafts under
the current subject heading...

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentito.com

----- Original Message -----
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Neil Deason'" <ndeason@ubiquity.net>; "'Adam Roach'"
<adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Sent: Friday, January 03, 2003 6:13 PM
Subject: RE: [Sip] INFO considered harmful


> > What I was actually advocating here was ppl wanting to keep
> > INFO around write up something to illuminate INFO packages
> > and provide example usage.
> >
> > The MESSAGE cf INFO was just an attempt to highlight an
> > apparent limitation with the ban INFO approach. Every MIME
> > type in an INFO can be moved into MESSAGE rather than go
> > away. Maybe through exploring INFO packages we evolve the
> > collective understanding that this is/isn't a necessary thing.
> >
> > Cheers,
> > Neil.
>
> This is an excellent suggestion, and very similar to the one Jonathan made
> to me on the phone yesterday (actually, I think he more tersely worded it
as
> "Send Text!")
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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Date: Fri, 03 Jan 2003 22:00:44 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Eric Burger <eburger@snowshore.com>, sip@ietf.org
Subject: Re: [Sip] INFO considered harmful
References: <4A3384433CE2AB46A63468CB207E209D2DF635@zoe.office.snowshore.com> <3E15E302.80508@cs.columbia.edu> <3E160FE7.2000907@dynamicsoft.com>
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I'd imagine they do care when they talk to you since that's the cost 
element that they can control during that conversation. They probably 
also try to bargain hard with the copy paper people.

Why would anybody buy extra servers for demand that isn't going to 
materialize for (say) months? Stocking up on servers isn't exactly the 
best return on investment - it's not like OPEC is going to raise their 
prices next year. I suspect that the major cost is not the hardware but 
the per-server software licenses :-)

If you can get a service deployed today rather than waiting for a new 
protocol to be standardized and deployed, and if you're paying interest 
on your 3G license loan, I'd imagine that speed-to-working beats request 
counts.

Since this is all hypotheticals, we can probably agree that server cost 
is just one of many factors. Deployability, system management complexity 
(more protocols=more MIBs), staff training and maturity of the 
alternatives and time-to-market all matter, but are hard to quantify.

Jonathan Rosenberg wrote:
> inline.
> 
> Henning Schulzrinne wrote:
> 
>> In general, I just don't buy server load arguments. If it's a useful 
>> service, server cost is usually the least of the problem. In any 
>> event, one should take this as a fraction of the existing signaling 
>> load, not as some absolute number. I'm worried about stuff that 
>> multiplies the signaling load by a factor of 10; I find 20 or 50% load 
>> increase uninteresting as an argument.
> 
> 
> My firsthand experience with selling stuff to carriers is that this is 
> most definitely not the way they see it. Remember, the equipment 
> purchases are made up front before the service is activated. If you can 
> sell a carrier a solution that requires 50% or even 20% fewer boxes, 
> thats a big win. Its not just the equipment cost, its the management 
> cost too that increases with the box counts.
> 
> -Jonathan R.
> 

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> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>
> system management complexity (more protocols=more MIBs)...

Red herring alert. A kitchen-sink protocol will require
a kitchen-sink MIB. You're not reducing management
complexity by turning a single protocol into a Swiss
army knife; you're just consolidating it.

/a
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Adam Roach <adam@dynamicsoft.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Eric Burger <eburger@snowshore.com>, sip@ietf.org
Subject: Re: [Sip] INFO considered harmful
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If there's no common functionality, I agree. If the non-INFO, non-SIP 
mechanism needs to replicate parts of the SIP functionality, such as 
request routing or proxies, you do end up with extra effort.

Adam Roach wrote:
>>-----Original Message-----
>>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>>
>>system management complexity (more protocols=more MIBs)...
> 
> 
> Red herring alert. A kitchen-sink protocol will require
> a kitchen-sink MIB. You're not reducing management
> complexity by turning a single protocol into a Swiss
> army knife; you're just consolidating it.
> 
> /a

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Date: Sat, 04 Jan 2003 14:08:29 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] INFO considered harmful
In-reply-to: <3E164E5C.50307@cs.columbia.edu>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'Eric Burger'" <eburger@snowshore.com>, sip@ietf.org
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The server hardware cost is probably the least to worry about, rather it
is as mentioned here:

>Deployability, system management, complexity 
> (more protocols=more MIBs), staff training and maturity of the 
> alternatives and time-to-market.

Signaling load is also not too much a problem, if not excessive. I would
rather worry about servers with geographic diversity and the most
effective DNS schema for fast response. 

My two cents, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Henning Schulzrinne
> Sent: Friday, January 03, 2003 9:01 PM
> To: Jonathan Rosenberg
> Cc: Eric Burger; sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> I'd imagine they do care when they talk to you since that's the cost 
> element that they can control during that conversation. They probably 
> also try to bargain hard with the copy paper people.
> 
> Why would anybody buy extra servers for demand that isn't going to 
> materialize for (say) months? Stocking up on servers isn't 
> exactly the 
> best return on investment - it's not like OPEC is going to 
> raise their 
> prices next year. I suspect that the major cost is not the 
> hardware but 
> the per-server software licenses :-)
> 
> If you can get a service deployed today rather than waiting for a new 
> protocol to be standardized and deployed, and if you're 
> paying interest 
> on your 3G license loan, I'd imagine that speed-to-working 
> beats request 
> counts.
> 
> Since this is all hypotheticals, we can probably agree that 
> server cost 
> is just one of many factors. Deployability, system management 
> complexity 
> (more protocols=more MIBs), staff training and maturity of the 
> alternatives and time-to-market all matter, but are hard to quantify.
> 
> Jonathan Rosenberg wrote:
> > inline.
> > 
> > Henning Schulzrinne wrote:
> > 
> >> In general, I just don't buy server load arguments. If 
> it's a useful
> >> service, server cost is usually the least of the problem. In any 
> >> event, one should take this as a fraction of the existing 
> signaling 
> >> load, not as some absolute number. I'm worried about stuff that 
> >> multiplies the signaling load by a factor of 10; I find 20 
> or 50% load 
> >> increase uninteresting as an argument.
> > 
> > 
> > My firsthand experience with selling stuff to carriers is 
> that this is
> > most definitely not the way they see it. Remember, the equipment 
> > purchases are made up front before the service is 
> activated. If you can 
> > sell a carrier a solution that requires 50% or even 20% 
> fewer boxes, 
> > thats a big win. Its not just the equipment cost, its the 
> management 
> > cost too that increases with the box counts.
> > 
> > -Jonathan R.
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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From: "Cullen Jennings" <fluffy@cisco.com>
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I doubt that doing this will help reduce DDOS or DOS attacks. For example
consider a request with a pre loaded route. There's lots more. Anyways, on a
more useful note ...

I'm suspect that we can accomplish what you are suggesting without any
protocol changes. Imagine your scheme would create a VIA that looks like:
Via: SIP/2.0/UDP foo.com:5060;branch=1234;cookie=abcd

Instead just create a branch that has the cookie embedded in it. It would
look like:
Via: SIP/2.0/UDP foo.com:5060;branch=1234-abcd

This trick is being used by proxies to embed other state that they might
want to get back later. You should have a look at HMAC.



Cullen


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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: hisham.khartabil@nokia.com, sip@ietf.org
Subject: Re: Repost: [Sip] Buggy example in RFC 3261
References: <2038BCC78B1AD641891A0D1AE133DBB7FE70A3@esebe019.ntc.nokia.com> <3E152EFF.8080105@dynamicsoft.com>
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Two more remarks inline.

Jonathan Rosenberg wrote:

> Well, a decision needs to be made one way or another. Since its already 
> in 3261, I would propose that we declare the example as buggy, and go 
> with the existing normative text, even though it works differently than 
> for requests.
> 
> -Jonathan R.
> 
> hisham.khartabil@nokia.com wrote:
> 
>> This is an extract from a discussion I had with Robert Sparks:
>>
>>
>>>> Yes, that was exactly what I asked. If its not so, what does an Expires
>>>> header in a 200 OK for a REGISTER mean?
>>>
>>>
>>>
>>> As I note above, with the current text an Expire header field has no 
>>> meaning
>>> if it appears in a 200 response to a REGISTER request.


This remark is not in line with the following text from RFC 3261,
section 10.2.4 Refreshing Bindings:
   "The 200 (OK) response from the registrar contains a list of Contact
    fields enumerating all current bindings.  The UA compares each
    contact address to see if it created the contact address, using
    comparison rules in Section 19.1.4.  If so, it updates the expiration
    time interval according to the expires parameter or, if absent, the
    Expires field value."

This clearly gives a meaning to the Expire header field in
200 for REGISTER.


>>
>> http://www.ietf.org/mail-archive/working-groups/sip/current/msg03602.html
>>
>> Regards,
>> Hisham
>>
>>
>>> -----Original Message-----
>>> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>> Sent: Thursday, January 02, 2003 7:01 AM
>>> To: Peter Päppinghaus
>>> Cc: sip@ietf.org
>>> Subject: Re: Repost: [Sip] Buggy example in RFC 3261
>>>
>>>
>>> Sorry for not responding. I did notice this. There is an error here, 
>>> as you point out. What is NOT clear is whether the error is in the 
>>> example, or in the normative text. For REGISTER requests, you can use 
>>> either the Expires header field or the expires contact header field 
>>> parameter. It seems odd that this same rule would not apply to 
>>> REGISTER responses. My personal recollection was that either the 

To me this asymmetry between REGISTER and its 200 responses does not
seem so odd. On the one hand it is very likely that a UA registering
a set of contact addresses in one REGISTER would suggest the same
expiration interval for all of them. On the other hand the 200 for
such a REGISTER is required to list all current bindings, and there
is no reason why the bindings should all have the same expiration
time.

>>> Expires header field or the expires contact param was OK for either 
>>> request or response. This doesn't match the normative text though.
>>>
>>> Does anyone use the Expires header field and NOT the expires contact 
>>> param in REGISTER responses?
>>>
>>> -Jonathan R.
>>>
>>> Peter Päppinghaus wrote:
>>>
>>>> I have posted this earlier, but it seems to have gone
>>>> unnoticed.
>>>>
>>>> Hi,
>>>>
>>>> I believe to have stumbled on a buggy example in RFC 3261.
>>>>
>>>> The 200 OK in the REGISTER example of RFC 3261, sec. 24.1 contains
>>>> an Expires header, but no expires parameter in the Contact header.
>>>>
>>>> This violates RFC 3261, sec. 10.3, bullet 8:
>>>> "Each Contact value MUST feature an "expires" parameter
>>>>  indicating its expiration interval chosen by the registrar."
>>>>
>>>> Regards,
>>>>
>>>> Peter Paeppinghaus
>>>>
>>>
>>> -- 
>>> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>>> Chief Scientist                             First Floor
>>> dynamicsoft                                 East Hanover, NJ 07936
>>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>>> http://www.dynamicsoft.com

-- 
Dr. Peter Päppinghaus
Siemens AG            | Phone:    +49 89 - 722 40065
ICM N PG U ID A1      | Fax:      +49 89 - 722 58726
Hofmannstr. 51        | Visitors: Building 1713 / Room 714
D - 81359 München     | Email:    peter.paeppinghaus@siemens.com

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From: Sarit Galanos Mekler <Sarit@radvision.com>
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Hi,
If a UA receives BYE and respond with 200, should he keep the BYE
transaction alive
according to timer J or can he terminate the transaction and respond with
481 to BYE retransmissions?
Thanks,
Sarit.
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From: "T A Gray" <tagray@sympatico.ca>
To: <sip@ietf.org>, "TA Gray" <tagray@sympatico.ca>
References: <20030105170000.8967.56963.Mailman@www1.ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Sun, 5 Jan 2003 12:34:20 -0800
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If the provider could charge per INFO message, would this not put an
entirely new light on this discussion. Isn't that the real significance of
SMS rather than the load it puts on the signalling servers.

Tom Gray
Pinetel


> Date: Sat, 04 Jan 2003 14:08:29 -0600
> From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
> Subject: RE: [Sip] INFO considered harmful
> To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
>    "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
> Cc: "'Eric Burger'" <eburger@snowshore.com>, sip@ietf.org
> Organization: WorldCom, Inc.
>
> The server hardware cost is probably the least to worry about, rather it
> is as mentioned here:
>
> >Deployability, system management, complexity
> > (more protocols=more MIBs), staff training and maturity of the
> > alternatives and time-to-market.
>
> Signaling load is also not too much a problem, if not excessive. I would
> rather worry about servers with geographic diversity and the most
> effective DNS schema for fast response.
>
> My two cents, Henry
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On
> > Behalf Of Henning Schulzrinne
> > Sent: Friday, January 03, 2003 9:01 PM
> > To: Jonathan Rosenberg
> > Cc: Eric Burger; sip@ietf.org
> > Subject: Re: [Sip] INFO considered harmful
> >
> >
> > I'd imagine they do care when they talk to you since that's the cost
> > element that they can control during that conversation. They probably
> > also try to bargain hard with the copy paper people.
> >
> > Why would anybody buy extra servers for demand that isn't going to
> > materialize for (say) months? Stocking up on servers isn't
> > exactly the
> > best return on investment - it's not like OPEC is going to
> > raise their
> > prices next year. I suspect that the major cost is not the
> > hardware but
> > the per-server software licenses :-)
> >
> > If you can get a service deployed today rather than waiting for a new
> > protocol to be standardized and deployed, and if you're
> > paying interest
> > on your 3G license loan, I'd imagine that speed-to-working
> > beats request
> > counts.
> >
> > Since this is all hypotheticals, we can probably agree that
> > server cost
> > is just one of many factors. Deployability, system management
> > complexity
> > (more protocols=more MIBs), staff training and maturity of the
> > alternatives and time-to-market all matter, but are hard to quantify.
> >
> > Jonathan Rosenberg wrote:
> > > inline.
> > >
> > > Henning Schulzrinne wrote:
> > >
> > >> In general, I just don't buy server load arguments. If
> > it's a useful
> > >> service, server cost is usually the least of the problem. In any
> > >> event, one should take this as a fraction of the existing
> > signaling
> > >> load, not as some absolute number. I'm worried about stuff that
> > >> multiplies the signaling load by a factor of 10; I find 20
> > or 50% load
> > >> increase uninteresting as an argument.
> > >
> > >
> > > My firsthand experience with selling stuff to carriers is
> > that this is
> > > most definitely not the way they see it. Remember, the equipment
> > > purchases are made up front before the service is
> > activated. If you can
> > > sell a carrier a solution that requires 50% or even 20%
> > fewer boxes,
> > > thats a big win. Its not just the equipment cost, its the
> > management
> > > cost too that increases with the box counts.
> > >
> > > -Jonathan R.
> > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current
> > sip Use sipping@ietf.org for new developments on the
> > application of sip
> >
>


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Hi Cullen,

Thanks for you comments. Please find my comments inline.

> I doubt that doing this will help reduce DDOS or DOS attacks. For example
> consider a request with a pre loaded route. There's lots more. Anyways, on
a
> more useful note ...

Requests with pre-loaded route set can be authenticated at proxies.
This, I think, can ward off DDoS attack using tailor made route set.

For requests there are mechanisms to authenticate and validate it. But
in the case of responses, it seems that, there is no way in which
proxies can authenticate or validate it. The mechanism given in the
draft try to provide one.

> I'm suspect that we can accomplish what you are suggesting without any
> protocol changes. Imagine your scheme would create a VIA that looks like:
> Via: SIP/2.0/UDP foo.com:5060;branch=1234;cookie=abcd
>
> Instead just create a branch that has the cookie embedded in it. It would
> look like:
> Via: SIP/2.0/UDP foo.com:5060;branch=1234-abcd

I had reservations in reusing the branch parameter. That is why I thought a
new
paramter would serve better. Having this parameter should not require any
change in protocol as such. As far as I know Proxies and UAs are supposed
to keep the Via's intact (i.e. they are not supposed to remove any
parameters
from it), so this cookie parameter would be retained when the response
reaches
proxy.

> This trick is being used by proxies to embed other state that they might
> want to get back later. You should have a look at HMAC.

I will do that.

Thanks again.
Sachin


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Date: Sun, 5 Jan 2003 20:05:54 +0200
To: "T A Gray" <tagray@sympatico.ca>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
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T A Gray writes:

 > If the provider could charge per INFO message, would this not put an
 > entirely new light on this discussion. Isn't that the real significance of
 > SMS rather than the load it puts on the signalling servers.

it doesn't make much sense for providers to start charging per info
message, because sip works fine without any provider.  sms is different,
because there is no way for me to send an sms message from a gsm phone
without using the provider's sms service.

-- juha

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Subject: Re: [Sip] INFO considered harmful
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Most people will have to have a proxy agent that will take care of this
concern. However your point is a very valid one in eh argument about the
load on signalling servers.

----- Original Message -----
From: <jh@lohi.eng.song.fi>
To: T A Gray <tagray@sympatico.ca>
Cc: <sip@ietf.org>
Sent: Sunday, January 05, 2003 10:05 AM
Subject: RE: [Sip] INFO considered harmful


> T A Gray writes:
>
>  > If the provider could charge per INFO message, would this not put an
>  > entirely new light on this discussion. Isn't that the real significance
of
>  > SMS rather than the load it puts on the signalling servers.
>
> it doesn't make much sense for providers to start charging per info
> message, because sip works fine without any provider.  sms is different,
> because there is no way for me to send an sms message from a gsm phone
> without using the provider's sms service.
>
> -- juha
>
>

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T A Gray writes:

 > Most people will have to have a proxy agent that will take care of this
 > concern. However your point is a very valid one in eh argument about the
 > load on signalling servers.

sure, similarly most people have an email server somewhere, but very few
actually pay per sent or received email message.

-- juha

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Dean Willis wrote:
>>The big huge difference I have been trying to point out is 
>>that EVENTS 
>>DEFINES SEMANTICS, whereas INFO doesn't.
> 
> 
> No, EVENTS doesn't define semantics. 

Of course it does. There are lots of MAYS, MUSTS and SHOULDs in there. 
There is behavior dealing with subscription refreshes, rules on when to 
send NOTIFY, and so on and so on. How can you say that none of those 
qualify as semantics?


> Event PACKAGES define the semantics of
> particular instances of the event methods. 

Within a well defined framework. 3265 defines clear rules on what each 
package has to specify, and what it can and cannot due ontop of rfc3265.


> INFO also needs a blob-describing-and-encoding framework, so that the
> various uses of INFO can, as appropriate, be effectively documented and
> negotiated. 

Well, I'll try one more time, since I am clearly not getting my point 
across.

RFC3265 defines semantics above and beyond RFC3261. All of those 
MAY/MUST/SHOULD I was talking about. If you had a choice between 
defining a new package, and defining your own sub/not mechanisms, it 
would be a LOT more work to define your own sub/not mechanisms.

Not so for INFO. WHen deciding whether to specify an INFO usage, or a 
new method, there is no additional work to do (specification-wise) when 
specifying a new method as opposed to a new INFO usage. To me, that 
means that INFO is devoid of semantics.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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Subject: Re: Repost: [Sip] Buggy example in RFC 3261
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inline.

Peter Päppinghaus wrote:
> Two more remarks inline.
> 
> Jonathan Rosenberg wrote:
> 
>> Well, a decision needs to be made one way or another. Since its 
>> already in 3261, I would propose that we declare the example as buggy, 
>> and go with the existing normative text, even though it works 
>> differently than for requests.
>>
>> -Jonathan R.
>>
>> hisham.khartabil@nokia.com wrote:
>>
>>> This is an extract from a discussion I had with Robert Sparks:
>>>
>>>
>>>>> Yes, that was exactly what I asked. If its not so, what does an 
>>>>> Expires
>>>>> header in a 200 OK for a REGISTER mean?
>>>>
>>>>
>>>>
>>>>
>>>> As I note above, with the current text an Expire header field has no 
>>>> meaning
>>>> if it appears in a 200 response to a REGISTER request.
>>>
> 
> 
> This remark is not in line with the following text from RFC 3261,
> section 10.2.4 Refreshing Bindings:
>   "The 200 (OK) response from the registrar contains a list of Contact
>    fields enumerating all current bindings.  The UA compares each
>    contact address to see if it created the contact address, using
>    comparison rules in Section 19.1.4.  If so, it updates the expiration
>    time interval according to the expires parameter or, if absent, the
>    Expires field value."
> 
> This clearly gives a meaning to the Expire header field in
> 200 for REGISTER.

Yes, another inconsistency here.


> 
> 
>>>
>>> http://www.ietf.org/mail-archive/working-groups/sip/current/msg03602.html 
>>>
>>>
>>> Regards,
>>> Hisham
>>>
>>>
>>>> -----Original Message-----
>>>> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>>>> Sent: Thursday, January 02, 2003 7:01 AM
>>>> To: Peter Päppinghaus
>>>> Cc: sip@ietf.org
>>>> Subject: Re: Repost: [Sip] Buggy example in RFC 3261
>>>>
>>>>
>>>> Sorry for not responding. I did notice this. There is an error here, 
>>>> as you point out. What is NOT clear is whether the error is in the 
>>>> example, or in the normative text. For REGISTER requests, you can 
>>>> use either the Expires header field or the expires contact header 
>>>> field parameter. It seems odd that this same rule would not apply to 
>>>> REGISTER responses. My personal recollection was that either the 
>>>
> 
> To me this asymmetry between REGISTER and its 200 responses does not
> seem so odd. On the one hand it is very likely that a UA registering
> a set of contact addresses in one REGISTER would suggest the same
> expiration interval for all of them. On the other hand the 200 for
> such a REGISTER is required to list all current bindings, and there
> is no reason why the bindings should all have the same expiration
> time.

True enough, but the same processing would still work in either case.

Anyway, I would still propose to follow my suggestion and go with the 
normative text in bullet 8 of 10.3.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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CC: Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
Subject: Re: [Sip] Anohter possible slution for draft-shenoy-sip-via-validation
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inline.

Sachin Shenoy wrote:
> Hi Cullen,
> 
> Thanks for you comments. Please find my comments inline.
> 
> 
>>I doubt that doing this will help reduce DDOS or DOS attacks. For example
>>consider a request with a pre loaded route. There's lots more. Anyways, on
>> a more useful note ...
> 
> 
> Requests with pre-loaded route set can be authenticated at proxies.
> This, I think, can ward off DDoS attack using tailor made route set.

The responses can be validated too, if you choose to use the security 
mechanisms specified in SIP. SPecifically, TLS connections between 
proxies would prevent the attack you describe.

DOn't get me wrong, I think what you have proposed is generally a fine 
idea and I suspect is something people may already be doing. As cullen 
said, it can be done without standardization. Indeed, you could even use 
a new Via parameter for this, and it would interoperate, since it does 
not require processing by any element but the one which inserted it.

In that regard, its very similar in concept to my predictive nonce idea:
http://www.jdrosen.net/papers/draft-rosenberg-sip-http-pnonce-00.txt
for which we also decided not to standardize - there was no need.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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CC: sip@ietf.org, Antonio.FERRIERI@vodafoneomnitel.it
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Correction inline.

loretosa@vizzavi.it wrote:
 > Hi Antonio,
 >
 > in a REGISTER message,
 >
 > the "To" Header Field contains the address of Record whose
 > registration it to be created.
 >
 > The "Contact" Field MAY be present in REGISTER REQUEST and it may
 > contain "zero or more values" containg address bindings.
 >
 >
 > If Contact Header is absent, the only contact address for the user is
 > the Address present in the "To" Header.

No.

If no Contact header field is present in a REGISTER request, the result 
is NO CHANGE on the current set of bindings.


 >
 >
 > What do you mean about the binding between IP address in VIA field
 > and the Request-URI? I'm not understand the question and what do you
 > mean.

The Via header field is used to route responses only. It is not used to 
route inbound requests.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Subject: RE: [Sip] Re: Restatement: INFO considered harmful
Date: Mon, 6 Jan 2003 10:33:38 -0500
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Thread-Topic: [Sip] Re: Restatement: INFO considered harmful
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From: "Eric Burger" <eburger@snowshore.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Dean Willis" <dean.willis@softarmor.com>
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I actually think the lines are that black and white.

Said differently, we now have three years of experience with SIP.  There's been a lot of pulling and stretching that has made the vision of a simple, text-based protocol for initiating multimedia sessions a lost dream.

A serious question is, "Which has more pages, H.323 et. al. or RFC 3261 et. al.?".

Reading Dean's posting, I was thinking, "It is time for SIP 3.0".  For that matter, for marketing reasons, it would need a different name.  A much simpler protocol that does one thing well (the IETF mantra) would NOT be backwards compatible with the bloat that we have today.

I don't think there is a middle ground.  To mix my metaphors, The line in the sand is being washed away by the tide of features.  Now that we have brushed away the session initiation simplicity (e.g., REGISTER, MESSAGE, SUBSCRIBE, NOTIFY, INFO), I cannot see how one can, with a straight face, say, "My child is cute, but your child is ugly" [substitute method/body for child].

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, January 03, 2003 5:44 PM
> To: Dean Willis
> Cc: 'Adam Roach'; 'Frank W. Miller'; sip@ietf.org
> Subject: [Sip] Re: Restatement: INFO considered harmful
> 
> 
> Dean,
> 
> The lines are not always so black and white. Its not a choice between 
> "do anything" and "do nothing but INVITE". There is a line somewhere 
> about what is a good use of sip, and what is not. We have some 
> documents, even, which help to identify that line (SIP 
> guidelines), and 
> no doubt it has moved over time. Indeed, there were even 
> arguments about 
> whether BYE belonged in SIP, since clearly its not needed for 
> "initiation". However, if you buy the assertion that SIP needs to at 
> least manage the session, that clearly includes termination 
> as well as 
> initiation. These days, I know of very few folks (only one in 
> fact) that 
> would argue against BYE as part of SIP.
> 
> I still stand by my assertion that many of the usages of SIP INFO are 
> over that line, and that in order to keep the usages in a line we are 
> comfortable with, something has to be done.
> 
> -Jonathan R.
> 
> Dean Willis wrote:
> > My mother said I should stop inflaming things, but here I 
> go again anyhow.
> > Sorry, Mom.
> > 
> > Adam said, responding to Frank in our long-winded exchange:
> > 
> >>Perhaps using some sort of protocol to initate this session 
> >>would be appropriate. You'll also needs a protocol of some 
> >>sort to describe it, probably carried as a payload in this 
> >>session initation protocol. Ideally, this session description 
> >>protocol could leverage IANA-registered media types to 
> >>describe the content of such sessions, so that an infinite 
> >>number of types of sessions can be described.
> > 
> > 
> > Actually, Adam and I just had a conversation about this 
> argument. I think
> > actually that there is something hiding in here which is 
> THE HEART of almost
> > every philosophical debate we've ever had in SIP. If we can 
> get this nailed
> > down now, we'll be preventing a HUGE number of future fights.
> > 
> > Or in short: Boom! Epiphany! I think I understand what 
> we're arguing about.
> > 
> > I'll admit -- Adam's been philosophically consistent in his 
> arguments since
> > day 1.  He believes that SIP should start sessions, stop 
> sessions, and
> > nothing else.
> > 
> > Examples:
> > 
> > Lets look at "events". By Adam's logic, the event system 
> should consist of
> > sending an INVITE (instead of a SUBSCRIBE) to the event 
> source. The SDP body
> > (Only SDP should be allowed in INVITE) would descibe the 
> events being
> > requested and the protocol, IP address and port number on which the
> > subscriber would be listening. The 200 OK response from the 
> event source
> > would then contain the protocol, IP address and port number 
> to which the
> > subscriber should send acknowledgements of notifications, 
> assuming that the
> > notification-transport protocol (clearly, not SIP).
> > 
> > And Whoa! Look at REGISTER. That doesn't set up any kind of 
> session! It's
> > clearly not a SIP function. We should instead use an INVITE 
> to set up a
> > "registration session", over which we would then use a 
> registration protocol
> > to establish registrations.
> > 
> > MESSAGE is clearly divergent. By Adam's logic, only message 
> sessions should
> > be used. By similar reasoning, INFO is also off, and 
> instead SIP-T should
> > use INVITE to establish a sigtran session.
> > 
> > This actually all makes sense and hangs together philosophically.
> > 
> > The problem is, the working group diverted from that 
> philosophy a long time
> > ago. What we have now is a combination of three 
> transactional protocols
> > (REGISTER, OPTIONS, MESSAGE) with a session-establishment 
> protocol (INVITE)
> > which supports several in-session request verbs (UPDATE, 
> OPTIONS, ACK, INFO,
> > MESSAGE, BYE, PRACK, etc.) and another 
> session-establishment protocol
> > (SUBSCRIBE) which supports an in-session request verb 
> (NOTIFY). We've also
> > defined an intermediation framework (the SIP Proxy) and a 
> set of end-to-end
> > and hop-by-hop authentication and integrity protection 
> mechanisms that can
> > be used for all these protocols. Really, it's the 
> intermediation framework
> > that gives SIP most of its utility.
> > 
> > Ok, it could be tidier. But it is very powerful, fairly 
> complete, and
> > actually works. And yes, the combination effectively 
> provides a very useful
> > transport protocol that does all sorts of things that HTTP 
> alone can't do,
> > such as intermediated rendezvous. And no, it DOESN'T just 
> do phone calls
> > (making attempts to model proxy load based on telephone 
> switch metrics
> > hopelessly bogus).
> > 
> > So my humble suggestion: Get over it. Iron out any kruft 
> that makes SIP less
> > desirable as a transport protocol (like UDP), and get on 
> with the business
> > of making stuff that people can use to get their jobs done. 
> Stop complaining
> > that your proxies are getting loaded down by serving a 
> useful purpose to
> > somebody, and start understanding what that purpose is and 
> how to charge for
> > it.
> > 
> > If you don't like it, fine. Let's take out a poll on 
> deprecating REGISTER,
> > MESSAGE, SUBSCRIBE, INFO, PRACK, SUBSCRIBE, NOTIFY, and all 
> that other
> > stuff.
> > 
> > Or hey, just toss it all out and add session negotiation to XMPP. 
> > 
> > --
> > Dean
> > 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>,
        Dean Willis
	 <dean.willis@softarmor.com>
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Subject: RE: [Sip] INFO considered harmful
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Looks at first glance like the Generic Transparency Descriptor which has
been proposed within ITU-T Study Group 11 and ANSI T1S1. Just out of
interest, what would be the MIME type for carrying this in SIP ?

Is recoding of ISUP into text format exactly in the scope of the IETF SIP
group ? 

...Mark

> -----Original Message-----
> From: Frank W. Miller [mailto:fmiller@sentito.com]
> Sent: 04 January 2003 00:02
> To: Dean Willis
> Cc: sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> 
> 
> Well, any illusion of reasonableness I may have garnered in 
> the past is
> about to be shattered.  I would refer folks to:
> 
> http://www.ietf.org/internet-drafts/draft-miller-sip-isup-annex-00.txt
> 
> This draft is about to receive a major -01 update that 
> recodes the entire
> syntax in XML.  In addition, we (I) have another draft that 
> does something
> very similar for ANSI TCAP that I'll be submitting in short 
> order.  I guess
> its only appropriate that I bring peoples attention to these 
> drafts under
> the current subject heading...
> 
> FM
> 
> Frank W. Miller, Ph.D.
> Chief Technical Officer
> sentitO Networks, Inc.
> fmiller@sentito.com
> 
> ----- Original Message -----
> From: "Dean Willis" <dean.willis@softarmor.com>
> To: "'Neil Deason'" <ndeason@ubiquity.net>; "'Adam Roach'"
> <adam@dynamicsoft.com>
> Cc: <sip@ietf.org>
> Sent: Friday, January 03, 2003 6:13 PM
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > > What I was actually advocating here was ppl wanting to keep
> > > INFO around write up something to illuminate INFO packages
> > > and provide example usage.
> > >
> > > The MESSAGE cf INFO was just an attempt to highlight an
> > > apparent limitation with the ban INFO approach. Every MIME
> > > type in an INFO can be moved into MESSAGE rather than go
> > > away. Maybe through exploring INFO packages we evolve the
> > > collective understanding that this is/isn't a necessary thing.
> > >
> > > Cheers,
> > > Neil.
> >
> > This is an excellent suggestion, and very similar to the 
> one Jonathan made
> > to me on the phone yesterday (actually, I think he more 
> tersely worded it
> as
> > "Send Text!")
> >
> > --
> > Dean
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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<P><FONT SIZE=3D2>Looks at first glance like the Generic Transparency =
Descriptor which has been proposed within ITU-T Study Group 11 and ANSI =
T1S1. Just out of interest, what would be the MIME type for carrying =
this in SIP ?</FONT></P>

<P><FONT SIZE=3D2>Is recoding of ISUP into text format exactly in the =
scope of the IETF SIP group ? </FONT>
</P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Frank W. Miller [<A =
HREF=3D"mailto:fmiller@sentito.com">mailto:fmiller@sentito.com</A>]</FON=
T>
<BR><FONT SIZE=3D2>&gt; Sent: 04 January 2003 00:02</FONT>
<BR><FONT SIZE=3D2>&gt; To: Dean Willis</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] INFO considered =
harmful</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Well, any illusion of reasonableness I may have =
garnered in </FONT>
<BR><FONT SIZE=3D2>&gt; the past is</FONT>
<BR><FONT SIZE=3D2>&gt; about to be shattered.&nbsp; I would refer =
folks to:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-miller-sip-isup-annex-=
00.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-miller-sip-i=
sup-annex-00.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; This draft is about to receive a major -01 =
update that </FONT>
<BR><FONT SIZE=3D2>&gt; recodes the entire</FONT>
<BR><FONT SIZE=3D2>&gt; syntax in XML.&nbsp; In addition, we (I) have =
another draft that </FONT>
<BR><FONT SIZE=3D2>&gt; does something</FONT>
<BR><FONT SIZE=3D2>&gt; very similar for ANSI TCAP that I'll be =
submitting in short </FONT>
<BR><FONT SIZE=3D2>&gt; order.&nbsp; I guess</FONT>
<BR><FONT SIZE=3D2>&gt; its only appropriate that I bring peoples =
attention to these </FONT>
<BR><FONT SIZE=3D2>&gt; drafts under</FONT>
<BR><FONT SIZE=3D2>&gt; the current subject heading...</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; FM</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Frank W. Miller, Ph.D.</FONT>
<BR><FONT SIZE=3D2>&gt; Chief Technical Officer</FONT>
<BR><FONT SIZE=3D2>&gt; sentitO Networks, Inc.</FONT>
<BR><FONT SIZE=3D2>&gt; fmiller@sentito.com</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; ----- Original Message -----</FONT>
<BR><FONT SIZE=3D2>&gt; From: &quot;Dean Willis&quot; =
&lt;dean.willis@softarmor.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; To: &quot;'Neil Deason'&quot; =
&lt;ndeason@ubiquity.net&gt;; &quot;'Adam Roach'&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; &lt;adam@dynamicsoft.com&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: &lt;sip@ietf.org&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Friday, January 03, 2003 6:13 PM</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] INFO considered =
harmful</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; What I was actually advocating here =
was ppl wanting to keep</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; INFO around write up something to =
illuminate INFO packages</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; and provide example usage.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; The MESSAGE cf INFO was just an =
attempt to highlight an</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; apparent limitation with the ban INFO =
approach. Every MIME</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; type in an INFO can be moved into =
MESSAGE rather than go</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; away. Maybe through exploring INFO =
packages we evolve the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; collective understanding that this =
is/isn't a necessary thing.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Cheers,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &gt; Neil.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This is an excellent suggestion, and very =
similar to the </FONT>
<BR><FONT SIZE=3D2>&gt; one Jonathan made</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; to me on the phone yesterday (actually, I =
think he more </FONT>
<BR><FONT SIZE=3D2>&gt; tersely worded it</FONT>
<BR><FONT SIZE=3D2>&gt; as</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &quot;Send Text!&quot;)</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; --</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Dean</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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References: <A3C2399B2FACD411A54200508BE39C74054F7A43@zwcwd00r.europe.nortel.com>
Subject: Re: [Sip] INFO considered harmful
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RE: [Sip] INFO considered harmfulWe just arbitarily picked:

Content-type: applicaton/isupxml

That is in the -01 version of this document that I referred to earlier.  =
Sorry, I'll get it up on the IETF site as quick as I can, hopefully =
within the week.

It is explictly outside the scope of the group, by choice I believe.  =
But this and the TCAP stuff are good examples of what I've been =
advocating, the need for some mechanism to allow stuff thats "related" =
but not necessarily appropriate (and in this case explictly shunned) for =
the mainline standard.  We actually looked at using SOAP for this (as =
was suggested by Adam) and decided against it for simple expedience =
(read development cost) reasons.  The mechanism present in our =
(third-party) SIP implementation was so compelling coupled with an off =
the shelf XML parser, that it didnt make sense to do anything else...

Furthermore, I would argue that you really never want anything like this =
to be included in the mainline standard for a couple of reasons.  First, =
it is very application specific.  Second, hopefully as SIP gains =
momentum and SS7 loses momentum with the large carriers, this stuff will =
just fade away...

FM

Frank W. Miller
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentitio.com

  ----- Original Message -----=20
  From: Mark Watson=20
  To: 'Frank W. Miller' ; Dean Willis=20
  Cc: sip@ietf.org=20
  Sent: Monday, January 06, 2003 1:07 PM
  Subject: RE: [Sip] INFO considered harmful


  Looks at first glance like the Generic Transparency Descriptor which =
has been proposed within ITU-T Study Group 11 and ANSI T1S1. Just out of =
interest, what would be the MIME type for carrying this in SIP ?

  Is recoding of ISUP into text format exactly in the scope of the IETF =
SIP group ?=20




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] INFO considered harmful</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2722.900" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>We just arbitarily picked:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Content-type: =
applicaton/isupxml</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>That is in the -01 version of this =
document that I=20
referred to earlier.&nbsp; Sorry, I'll get it up on the IETF site as =
quick as I=20
can, hopefully within the week.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>It is explictly outside the scope of =
the group, by=20
choice I believe.&nbsp; But this and the TCAP&nbsp;stuff are good =
examples of=20
what I've been&nbsp;advocating, the need for some mechanism to allow =
stuff thats=20
"related" but not necessarily appropriate (and&nbsp;in this case =
explictly=20
shunned)&nbsp;for the mainline standard.&nbsp; We actually looked at =
using SOAP=20
for this (as was suggested by Adam) and decided against it for simple =
expedience=20
(read development cost) reasons.&nbsp; The mechanism present in our=20
(third-party) SIP implementation was so compelling coupled with an off =
the shelf=20
XML parser, that it didnt make sense to do anything else...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Furthermore, I would argue that you =
really never=20
want anything like this to be included in the mainline standard for a =
couple of=20
reasons.&nbsp; First, it is very application specific.&nbsp; Second, =
hopefully=20
as SIP gains momentum and SS7 loses momentum with the large carriers, =
this stuff=20
will just fade away...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>FM</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Frank W. Miller</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Chief Technical Officer</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>sentitO Networks, Inc.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>fmiller@sentitio.com</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dmwatson@nortelnetworks.com=20
  href=3D"mailto:mwatson@nortelnetworks.com">Mark Watson</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dfmiller@sentito.com=20
  href=3D"mailto:fmiller@sentito.com">'Frank W. Miller'</A> ; <A=20
  title=3Ddean.willis@softarmor.com =
href=3D"mailto:dean.willis@softarmor.com">Dean=20
  Willis</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Monday, January 06, 2003 =
1:07=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> RE: [Sip] INFO =
considered=20
  harmful</DIV>
  <DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial =
size=3D2></FONT><BR></DIV>
  <P><FONT size=3D2>Looks at first glance like the Generic Transparency =
Descriptor=20
  which has been proposed within ITU-T Study Group 11 and ANSI T1S1. =
Just out of=20
  interest, what would be the MIME type for carrying this in SIP =
?</FONT></P>
  <P><FONT size=3D2>Is recoding of ISUP into text format exactly in the =
scope of=20
  the IETF SIP group ? </FONT></P>
  <P><FONT size=3D2></FONT>&nbsp;</P></BLOCKQUOTE></BODY></HTML>

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Subject: [Sip] RFC 3329 on Security Mechanism Agreement for the Session Initiation Protocol (SIP)
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--NextPart


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


        RFC 3329

        Title:      Security Mechanism Agreement for the Session
                    Initiation Protocol (SIP)
        Author(s):  J. Arkko, V. Torvinen, G. Camarillo, A. Niemi,
                    T. Haukka
        Status:     Standards Track
        Date:       January 2003
        Mailbox:    jari.arkko@ericsson.com,
                    vesa.torvinen@ericsson.fi,
                    Gonzalo.Camarillo@ericsson.com,
                    aki.niemi@nokia.com, tao.haukka@nokia.com
        Pages:      24
        Characters: 51503
        Updates/Obsoletes/See Also:   None

        I-D Tag:    draft-ietf-sip-sec-agree-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3329.txt


This document defines new functionality for negotiating the security
mechanisms used between a Session Initiation Protocol (SIP) user
agent and its next-hop SIP entity.  This new functionality
supplements the existing methods of choosing security mechanisms
between SIP entities.

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

This is now a Proposed Standard Protocol.

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

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Submissions for Requests for Comments should be sent to
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Authors, for further information.


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

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Subject: Re: Repost: [Sip] Buggy example in RFC 3261
References: <2038BCC78B1AD641891A0D1AE133DBB7FE70A3@esebe019.ntc.nokia.com> <3E152EFF.8080105@dynamicsoft.com> <3E183A16.8070302@siemens.com> <3E18EE0E.1070700@dynamicsoft.com>
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inline.

Jonathan Rosenberg wrote:

> inline.
> 
> Peter Päppinghaus wrote:
> 
>> Two more remarks inline.
>>
>> Jonathan Rosenberg wrote:
>>
>>> Well, a decision needs to be made one way or another. Since its 
>>> already in 3261, I would propose that we declare the example as 
>>> buggy, and go with the existing normative text, even though it works 
>>> differently than for requests.
>>>
>>> -Jonathan R.


[snip]


>> To me this asymmetry between REGISTER and its 200 responses does not
>> seem so odd. On the one hand it is very likely that a UA registering
>> a set of contact addresses in one REGISTER would suggest the same
>> expiration interval for all of them. On the other hand the 200 for
>> such a REGISTER is required to list all current bindings, and there
>> is no reason why the bindings should all have the same expiration
>> time.
> 
> 
> True enough, but the same processing would still work in either case.
> 
> Anyway, I would still propose to follow my suggestion and go with the 
> normative text in bullet 8 of 10.3.
> 
> -Jonathan R.


This is also my personal preference.

-- 
Dr. Peter Päppinghaus
Siemens AG            | Phone:    +49 89 - 722 40065
ICM N PG U ID A1      | Fax:      +49 89 - 722 58726
Hofmannstr. 51        | Visitors: Building 1713 / Room 714
D - 81359 München     | Email:    peter.paeppinghaus@siemens.com

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Subject: RE: [Sip] INFO considered harmful
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Frank,
 
I guess my concern is that there is an important difference between
transparently carrying ISUP information between 'ISUP capable' devices
(which we have a way to do with SIP-T), and defining new formats for
'ISUP-style' information to be processed by non-ISUP-capable nodes in the
SIP network. This second just sounds like defining extensions to SIP.
 
There's been lots of discussion about processes for defining extensions to
SIP - both standard and 'proprietary/preliminary ...'. Don't want to repeat
these here, but usually there is some opposition to just importing things
from ISUP.
 
Was your intention that this approach be published as Informational, or
rolled in to the GTD work in ITU ?
 
...Mark

-----Original Message-----
From: Frank W. Miller [mailto:fmiller@sentito.com]
Sent: 06 January 2003 18:37
To: Watson, Mark [MOP:EP10:EXCH]
Cc: sip@ietf.org
Subject: Re: [Sip] INFO considered harmful


We just arbitarily picked:
 
Content-type: applicaton/isupxml
 
That is in the -01 version of this document that I referred to earlier.
Sorry, I'll get it up on the IETF site as quick as I can, hopefully within
the week.
 
It is explictly outside the scope of the group, by choice I believe.  But
this and the TCAP stuff are good examples of what I've been advocating, the
need for some mechanism to allow stuff thats "related" but not necessarily
appropriate (and in this case explictly shunned) for the mainline standard.
We actually looked at using SOAP for this (as was suggested by Adam) and
decided against it for simple expedience (read development cost) reasons.
The mechanism present in our (third-party) SIP implementation was so
compelling coupled with an off the shelf XML parser, that it didnt make
sense to do anything else...
 
Furthermore, I would argue that you really never want anything like this to
be included in the mainline standard for a couple of reasons.  First, it is
very application specific.  Second, hopefully as SIP gains momentum and SS7
loses momentum with the large carriers, this stuff will just fade away...
 
FM
 
Frank W. Miller
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentitio.com
 

----- Original Message ----- 
From: Mark Watson <mailto:mwatson@nortelnetworks.com>  
To: 'Frank W.  <mailto:fmiller@sentito.com> Miller' ; Dean Willis
<mailto:dean.willis@softarmor.com>  
Cc: sip@ietf.org <mailto:sip@ietf.org>  
Sent: Monday, January 06, 2003 1:07 PM
Subject: RE: [Sip] INFO considered harmful



Looks at first glance like the Generic Transparency Descriptor which has
been proposed within ITU-T Study Group 11 and ANSI T1S1. Just out of
interest, what would be the MIME type for carrying this in SIP ?

Is recoding of ISUP into text format exactly in the scope of the IETF SIP
group ? 

 


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<TITLE>RE: [Sip] INFO considered harmful</TITLE>

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<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003>Frank,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=037250210-07012003>I 
guess my concern is that there is an important difference between transparently 
carrying ISUP information between 'ISUP capable' devices (which we have a way to 
do with SIP-T), and defining new formats for 'ISUP-style' information to be 
processed by non-ISUP-capable nodes in the SIP network. This second just sounds 
like defining extensions to SIP.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003>There's been lots of discussion about processes for 
defining extensions to SIP - both standard and 'proprietary/preliminary ...'. 
Don't want to repeat these here, but usually there is some opposition to just 
importing things from ISUP.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN class=037250210-07012003>Was 
your intention that this approach be published as Informational, or rolled in to 
the GTD work in ITU ?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Verdana size=1><SPAN 
class=037250210-07012003>...Mark</SPAN></FONT></DIV>
<BLOCKQUOTE dir=ltr 
style="BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Frank W. Miller 
  [mailto:fmiller@sentito.com]<BR><B>Sent:</B> 06 January 2003 
  18:37<BR><B>To:</B> Watson, Mark [MOP:EP10:EXCH]<BR><B>Cc:</B> 
  sip@ietf.org<BR><B>Subject:</B> Re: [Sip] INFO considered 
  harmful<BR><BR></DIV></FONT>
  <DIV><FONT face=Arial size=2>We just arbitarily picked:</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Content-type: applicaton/isupxml</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>That is in the -01 version of this document that 
  I referred to earlier.&nbsp; Sorry, I'll get it up on the IETF site as quick 
  as I can, hopefully within the week.</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>It is explictly outside the scope of the group, 
  by choice I believe.&nbsp; But this and the TCAP&nbsp;stuff are good examples 
  of what I've been&nbsp;advocating, the need for some mechanism to allow stuff 
  thats "related" but not necessarily appropriate (and&nbsp;in this case 
  explictly shunned)&nbsp;for the mainline standard.&nbsp; We actually looked at 
  using SOAP for this (as was suggested by Adam) and decided against it for 
  simple expedience (read development cost) reasons.&nbsp; The mechanism present 
  in our (third-party) SIP implementation was so compelling coupled with an off 
  the shelf XML parser, that it didnt make sense to do anything 
  else...</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Furthermore, I would argue that you really never 
  want anything like this to be included in the mainline standard for a couple 
  of reasons.&nbsp; First, it is very application specific.&nbsp; Second, 
  hopefully as SIP gains momentum and SS7 loses momentum with the large 
  carriers, this stuff will just fade away...</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>FM</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2>Frank W. Miller</FONT></DIV>
  <DIV><FONT face=Arial size=2>Chief Technical Officer</FONT></DIV>
  <DIV><FONT face=Arial size=2>sentitO Networks, Inc.</FONT></DIV>
  <DIV><FONT face=Arial size=2>fmiller@sentitio.com</FONT></DIV>
  <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
  <BLOCKQUOTE dir=ltr 
  style="BORDER-LEFT: #000000 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px; PADDING-RIGHT: 0px">
    <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
    <DIV 
    style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
    <A href="mailto:mwatson@nortelnetworks.com" 
    title=mwatson@nortelnetworks.com>Mark Watson</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>To:</B> <A 
    href="mailto:fmiller@sentito.com" title=fmiller@sentito.com>'Frank W. 
    Miller'</A> ; <A href="mailto:dean.willis@softarmor.com" 
    title=dean.willis@softarmor.com>Dean Willis</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>Cc:</B> <A href="mailto:sip@ietf.org" 
    title=sip@ietf.org>sip@ietf.org</A> </DIV>
    <DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, January 06, 2003 1:07 
    PM</DIV>
    <DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Sip] INFO considered 
    harmful</DIV>
    <DIV><FONT face=Arial size=2></FONT><FONT face=Arial 
size=2></FONT><BR></DIV>
    <P><FONT size=2>Looks at first glance like the Generic Transparency 
    Descriptor which has been proposed within ITU-T Study Group 11 and ANSI 
    T1S1. Just out of interest, what would be the MIME type for carrying this in 
    SIP ?</FONT></P>
    <P><FONT size=2>Is recoding of ISUP into text format exactly in the scope of 
    the IETF SIP group ? </FONT></P>
    <P><FONT size=2></FONT>&nbsp;</P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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	Title		: The Session Initiation Protocol (SIP) INFO Method 
                          Considered Harmful
	Author(s)	: J. Rosenberg
	Filename	: draft-rosenberg-sip-info-harmful-00.txt
	Pages		: 11
	Date		: 2003-1-6
	
The Session Initiation Protocol (SIP) INFO method defines a means for
transporting mid-dialog application layer data between user agents.
Its initial use was to support the transport of ISUP mid-call
messages which could not be mapped to any other SIP request method.
However, since its initial usage for that purpose, INFO has seen
widespread abuse as a means for introducing non-standard and
non-interoperable extensions to SIP.  For this reason, we now believe
INFO should be considered harmful, and therefore, deprecated in its
current form.

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

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RE: [Sip] INFO considered harmful
The main issue with SIP-T is that it requires some portion of the SS7 =
stack to be present in the IP elements of the network.  I see SIP-T as =
being primarily useful as an IP network transport of ISUP between SS7 =
"islands".  If an SS7 element wants to control say a T-1 that happens to =
be SIP terminated, I feel that a translation is a better approach.  It =
limits the need to deploy (very expensive from a dollar and point code =
point of view) ISUP protocol elements.  Also, if done right, it can =
eliminate the country variant problem that is IMHO a huge problem for =
SS7.

As far as my intentions with these drafts, I have none, other than a =
desire for this protocol information to be in the public domain.  I've =
heard rumblings about addressing SIP interaction in the ITU and also =
Telcordia.  I have not participated in the ITU forums actively to date.  =
If IETF takes it up, thats fine.  If there's synergy with ITU, thats =
fine too.  I like IETF because there's a mechanism for making things =
public without having to really go though all the pain of committees.  =
It doesnt mean its sanctioned by IETF but the openness of the forum is =
more appealing to me than some of the other standards groups I've been =
involved with over time.

FM


  ----- Original Message -----=20
  From: Mark Watson=20
  To: 'Frank W. Miller'=20
  Cc: sip@ietf.org=20
  Sent: Tuesday, January 07, 2003 5:04 AM
  Subject: RE: [Sip] INFO considered harmful


  Frank,

  I guess my concern is that there is an important difference between =
transparently carrying ISUP information between 'ISUP capable' devices =
(which we have a way to do with SIP-T), and defining new formats for =
'ISUP-style' information to be processed by non-ISUP-capable nodes in =
the SIP network. This second just sounds like defining extensions to =
SIP.

  There's been lots of discussion about processes for defining =
extensions to SIP - both standard and 'proprietary/preliminary ...'. =
Don't want to repeat these here, but usually there is some opposition to =
just importing things from ISUP.

  Was your intention that this approach be published as Informational, =
or rolled in to the GTD work in ITU ?



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] INFO considered harmful</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2722.900" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>The main issue with SIP-T is that it =
requires some=20
portion of the SS7 stack to be present in the IP elements of the =
network.&nbsp;=20
I see SIP-T as being&nbsp;primarily useful as an&nbsp;IP network =
transport of=20
ISUP between SS7 "islands".&nbsp; If an SS7 element wants to control say =
a T-1=20
that happens to be SIP terminated, I feel that a translation is a better =

approach.&nbsp; It limits the need to deploy (very expensive from a =
dollar and=20
point code point of view) ISUP protocol elements.&nbsp; Also, if done =
right, it=20
can eliminate the country variant problem that is IMHO a huge problem =
for=20
SS7.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>As far as my intentions with these =
drafts, I have=20
none, other than a desire for this protocol information to be in the =
public=20
domain.&nbsp; I've heard rumblings about addressing SIP interaction in =
the ITU=20
and also Telcordia.&nbsp; I have not participated in the ITU forums =
actively to=20
date.&nbsp; If IETF takes it up, thats fine.&nbsp; If there's synergy =
with ITU,=20
thats fine too.&nbsp; I like IETF because there's a mechanism for making =
things=20
public without having to really go though all the pain of =
committees.&nbsp; It=20
doesnt mean its sanctioned by IETF but the openness of the forum is more =

appealing to me than some of the other standards groups I've been =
involved with=20
over time.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>FM</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dmwatson@nortelnetworks.com=20
  href=3D"mailto:mwatson@nortelnetworks.com">Mark Watson</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dfmiller@sentito.com=20
  href=3D"mailto:fmiller@sentito.com">'Frank W. Miller'</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, January 07, 2003 =
5:04=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> RE: [Sip] INFO =
considered=20
  harmful</DIV>
  <DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial =
size=3D2></FONT><FONT=20
  face=3DArial size=3D2></FONT><FONT face=3DArial size=3D2></FONT><FONT =
face=3DArial=20
  size=3D2></FONT><BR></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003>Frank,</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN =
class=3D037250210-07012003>I=20
  guess my concern is that there is an important difference between=20
  transparently carrying ISUP information between 'ISUP capable' devices =
(which=20
  we have a way to do with SIP-T), and defining new formats for =
'ISUP-style'=20
  information to be processed by non-ISUP-capable nodes in the SIP =
network. This=20
  second just sounds like defining extensions to =
SIP.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003>There's been lots of discussion about =
processes for=20
  defining extensions to SIP - both standard and =
'proprietary/preliminary ...'.=20
  Don't want to repeat these here, but usually there is some opposition =
to just=20
  importing things from ISUP.</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003>Was your intention that this approach be =
published as=20
  Informational, or rolled in to the GTD work in ITU =
?</SPAN></FONT></DIV>
  <DIV><FONT face=3DVerdana color=3D#0000ff size=3D1><SPAN=20
  class=3D037250210-07012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_0048_01C2B634.239D2110--

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From: "Pianigiani, Jacopo (Jacopo)" <jpianigiani@lucent.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Mukul Purohit
	 <mpurohit@neomagic.com>
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Subject: RE: [Sip] resource reservation
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Jonathan & Mukul,
as far as I know, referring to 3GPP, Rel.5 (frozen) and rel. 6 (being written almost now) specs the following QoS alternatives mechanisms to pure RSVP reservation mechanism will be available:
a) Go Diameter interface on the GGSN (traffic gating & shaping + CAC) + conditioning of the Radio Access Bearer
b) Diffserv and E/L-LSP QoS transport across RAN (but not for reservation, just for PHB treatment)
c) as you correctly stated, end to end RSVP signalling (inheritance of Signalled SIP QoS parameters)

Jac

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, December 30, 2002 5:37 PM
To: Mukul Purohit
Cc: sip@ietf.org
Subject: Re: [Sip] resource reservation




Mukul Purohit wrote:
> hi,
> I have a basic question after going through RFC 3312, "Integration of 
> Resource Management and SIP".
> 
> Has the actual resource reservation mechanism been specified (mandated 
> or suggested) by IETF?

There is no mandatory resource reservation scheme. The primary one 
specified so far is RSVP. nsis is another one under development. Groups 
like packetcable have specified their own QoS protocols.

> Also, has it been specified by any other body like 3GPP which has chosen 
> SIP for signaling?

Yes, 3gpp has its own mechanisms that are specific to 3gpp. As I 
understand they are built into the GPRS procedures, and reservation 
occurs only on the wireless link.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: [Sip] INFO considered harmful
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From: Frank W. Miller [mailto:fmiller@sentito.com]

> The main issue with SIP-T is that it requires some portion
> of the SS7 stack to be present in the IP elements of the network.

Huh?

I beleive you've completely misunderstood the purpose of SIP-T.
If you have an IP element (pure IP, not your gateway) that's
interpreting the ISUP bodies present in SIP-T messages, you're
violating the spirit of what was intended.

I'm not making a value judgement; I'm just saying that your
reading of the documents appears to have been a bit off.

/a
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From: "Frank W. Miller" <fmiller@sentito.com>
To: "Adam Roach" <adam@dynamicsoft.com>
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I'm not talking about how it was intended for use, I'm talking about how it
can and is being used in some instances (seems to be a theme here lately).

FM


----- Original Message -----
From: "Adam Roach" <adam@dynamicsoft.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>; "Mark Watson"
<mwatson@nortelnetworks.com>
Cc: <sip@ietf.org>
Sent: Tuesday, January 07, 2003 2:44 PM
Subject: RE: [Sip] INFO considered harmful


> From: Frank W. Miller [mailto:fmiller@sentito.com]
>
> > The main issue with SIP-T is that it requires some portion
> > of the SS7 stack to be present in the IP elements of the network.
>
> Huh?
>
> I beleive you've completely misunderstood the purpose of SIP-T.
> If you have an IP element (pure IP, not your gateway) that's
> interpreting the ISUP bodies present in SIP-T messages, you're
> violating the spirit of what was intended.
>


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From mailnull@www1.ietf.org  Tue Jan  7 18:12:46 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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Sorry for being pedantic, but under the circumstances you
describe, it would cause far less confusion and help
limit disinformation if you were to phrase such things
as "The main issue with our proprietary use of SIP-T..."

Your original statement asserted (presumably unintentionally)
that using SIP-T *requires* SIP proxies to understand ISUP,
which is clearly not true.

/a

P.S. The expensive part of your ISUP stacks is not the encoding;
     it's the state machines. The syntactical transformation you
     propose actually gains you very little. ISUP encoding is
     delightfully simple to code. In fact, five years ago, I could
     do it in my head for the most common operations.

> -----Original Message-----
> From: Frank W. Miller [mailto:fmiller@sentito.com]
> Sent: Tuesday, January 07, 2003 16:37
> To: Adam Roach
> Cc: sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> 
> I'm not talking about how it was intended for use, I'm 
> talking about how it
> can and is being used in some instances (seems to be a theme 
> here lately).
> 
> FM
> 
> 
> ----- Original Message -----
> From: "Adam Roach" <adam@dynamicsoft.com>
> To: "'Frank W. Miller'" <fmiller@sentito.com>; "Mark Watson"
> <mwatson@nortelnetworks.com>
> Cc: <sip@ietf.org>
> Sent: Tuesday, January 07, 2003 2:44 PM
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > From: Frank W. Miller [mailto:fmiller@sentito.com]
> >
> > > The main issue with SIP-T is that it requires some portion
> > > of the SS7 stack to be present in the IP elements of the network.
> >
> > Huh?
> >
> > I beleive you've completely misunderstood the purpose of SIP-T.
> > If you have an IP element (pure IP, not your gateway) that's
> > interpreting the ISUP bodies present in SIP-T messages, you're
> > violating the spirit of what was intended.
> >
> 
> 
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If a dialog is already established with a SUBSCRIBE, that SUBSCRIBE is being refreshed (SUBSCRIBE in progress). Can an INVITE, within the same dialog, be sent while the SUBSCRIBE refresh is in progress?

Another question: Can SUBSCRIBEs overlap in the following 2 scenarios?

Scenario 1:

   (1) SUBSCRIBE event:presence
---------------------------->

   (2) SUBSCRIBE event:presence
---------------------------->

(1) and (2) are requests within the same dialog and are the same association with the dialog.
Can (2) be sent before (1) completes. This is like INVITEs overlapping.

Scenario 2:

   (3) SUBSCRIBE event:presence
---------------------------->

   (4) SUBSCRIBE event:winfo
---------------------------->

(3) and (4) are requests within the same dialog but are in different associations with the dialog.
Can (4) be sent before (3) completes. This is like INVITE and SUBSCRIBE overlapping, I think.
A similar example of this is 2 SUBSCRIBEs with the same event package but different event ids.

Thanks,
Hisham
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Subject: [Sip] SIP de-registration
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All

Is it valid to reject a de-registration request?

For example 'If a  SIP terminal sends a de-registration request to a network
while it still has an active session established, and the network
rejects this request , is this correct behaviour.


Jo Higgens
Orange UK



*******************************************************************************
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Orange PCS Limited is a subsidiary of Orange SA and is registered in England No
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*******************************************************************************

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Date: Wed, 08 Jan 2003 17:20:38 +0530
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] SIP de-registration
In-reply-to: <80256CA8.00401720.00@ruddick>
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Any registrar can reject a de-registration request anytime.  But rejecting a
de-registration for the fact that a session is active is a local
implementation.  In general the registration is used to locate an entity and
theoritically after the session is established, the registration is
immaterial.
Cheers,
Prasanna

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
joanne.higgens@orange.co.uk
Sent: Wednesday, January 08, 2003 5:12 PM
To: sip@ietf.org
Subject: [Sip] SIP de-registration



All

Is it valid to reject a de-registration request?

For example 'If a  SIP terminal sends a de-registration request to a network
while it still has an active session established, and the network
rejects this request , is this correct behaviour.


Jo Higgens
Orange UK



****************************************************************************
***
Important.
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and
may be confidential and/or legally privileged. Any opinions expressed in
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you
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anyone; please delete/destroy and inform the sender immediately.

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Orange may monitor all incoming and outgoing emails in line with current
legislation.  Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus free.

Orange PCS Limited is a subsidiary of Orange SA and is registered in England
No
2178917, with its address at St James Court, Great Park Road, Almondsbury
Park,
Bradley Stoke, Bristol BS32 4QJ.
****************************************************************************
***

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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        Mukul Purohit
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Subject: RE: [Sip] resource reservation
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Well, for the avoidance of confusion:

1) GPRS terminals need to perform a GPRS-specific resource reservation
mechanims to get resources on the radio network/GPRS core network. This is
known as a PDP Context. 3GPP IMS uses the "Integration of Resource
Management and SIP" mechanisms to delay completion of the SIP session
establishment until this GPRS-specific procedure has been completed.

2) Various mechanism can be used within the GPRS network/RAN to ensure that
the traffic receives the QoS agreed in the PDP Context establishment
negotiation. This could include Diffserv/MPLS but is internal to the GPRS
network and not visible to the terminal

3) The Go interface is COPS-PR, not Diameter. It is used to authorise PDP
Context establishements which are related to SIP sessions.

4) The terminal is free to attempt RSVP reservations, but these will have no
effect on the GPRS network, unless the Layer 2 in the terminal is trapping
them and firing off appropriate PDP Context requests. The RSVP may of course
affect 'downstream' networks outside GPRS.

...Mark

> -----Original Message-----
> From: Pianigiani, Jacopo (Jacopo) [mailto:jpianigiani@lucent.com]
> Sent: 07 January 2003 17:15
> To: 'Jonathan Rosenberg'; Mukul Purohit
> Cc: sip@ietf.org
> Subject: RE: [Sip] resource reservation
> 
> 
> Jonathan & Mukul,
> as far as I know, referring to 3GPP, Rel.5 (frozen) and rel. 
> 6 (being written almost now) specs the following QoS 
> alternatives mechanisms to pure RSVP reservation mechanism 
> will be available:
> a) Go Diameter interface on the GGSN (traffic gating & 
> shaping + CAC) + conditioning of the Radio Access Bearer
> b) Diffserv and E/L-LSP QoS transport across RAN (but not for 
> reservation, just for PHB treatment)
> c) as you correctly stated, end to end RSVP signalling 
> (inheritance of Signalled SIP QoS parameters)
> 
> Jac
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, December 30, 2002 5:37 PM
> To: Mukul Purohit
> Cc: sip@ietf.org
> Subject: Re: [Sip] resource reservation
> 
> 
> 
> 
> Mukul Purohit wrote:
> > hi,
> > I have a basic question after going through RFC 3312, 
> "Integration of 
> > Resource Management and SIP".
> > 
> > Has the actual resource reservation mechanism been 
> specified (mandated 
> > or suggested) by IETF?
> 
> There is no mandatory resource reservation scheme. The primary one 
> specified so far is RSVP. nsis is another one under 
> development. Groups 
> like packetcable have specified their own QoS protocols.
> 
> > Also, has it been specified by any other body like 3GPP 
> which has chosen 
> > SIP for signaling?
> 
> Yes, 3gpp has its own mechanisms that are specific to 3gpp. As I 
> understand they are built into the GPRS procedures, and reservation 
> occurs only on the wireless link.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

------_=_NextPart_001_01C2B70C.1B9DB5AA
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] resource reservation</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Well, for the avoidance of confusion:</FONT>
</P>

<P><FONT SIZE=3D2>1) GPRS terminals need to perform a GPRS-specific =
resource reservation mechanims to get resources on the radio =
network/GPRS core network. This is known as a PDP Context. 3GPP IMS =
uses the &quot;Integration of Resource Management and SIP&quot; =
mechanisms to delay completion of the SIP session establishment until =
this GPRS-specific procedure has been completed.</FONT></P>

<P><FONT SIZE=3D2>2) Various mechanism can be used within the GPRS =
network/RAN to ensure that the traffic receives the QoS agreed in the =
PDP Context establishment negotiation. This could include Diffserv/MPLS =
but is internal to the GPRS network and not visible to the =
terminal</FONT></P>

<P><FONT SIZE=3D2>3) The Go interface is COPS-PR, not Diameter. It is =
used to authorise PDP Context establishements which are related to SIP =
sessions.</FONT></P>

<P><FONT SIZE=3D2>4) The terminal is free to attempt RSVP reservations, =
but these will have no effect on the GPRS network, unless the Layer 2 =
in the terminal is trapping them and firing off appropriate PDP Context =
requests. The RSVP may of course affect 'downstream' networks outside =
GPRS.</FONT></P>

<P><FONT SIZE=3D2>...Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Pianigiani, Jacopo (Jacopo) [<A =
HREF=3D"mailto:jpianigiani@lucent.com">mailto:jpianigiani@lucent.com</A>=
]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 07 January 2003 17:15</FONT>
<BR><FONT SIZE=3D2>&gt; To: 'Jonathan Rosenberg'; Mukul Purohit</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: RE: [Sip] resource reservation</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan &amp; Mukul,</FONT>
<BR><FONT SIZE=3D2>&gt; as far as I know, referring to 3GPP, Rel.5 =
(frozen) and rel. </FONT>
<BR><FONT SIZE=3D2>&gt; 6 (being written almost now) specs the =
following QoS </FONT>
<BR><FONT SIZE=3D2>&gt; alternatives mechanisms to pure RSVP =
reservation mechanism </FONT>
<BR><FONT SIZE=3D2>&gt; will be available:</FONT>
<BR><FONT SIZE=3D2>&gt; a) Go Diameter interface on the GGSN (traffic =
gating &amp; </FONT>
<BR><FONT SIZE=3D2>&gt; shaping + CAC) + conditioning of the Radio =
Access Bearer</FONT>
<BR><FONT SIZE=3D2>&gt; b) Diffserv and E/L-LSP QoS transport across =
RAN (but not for </FONT>
<BR><FONT SIZE=3D2>&gt; reservation, just for PHB treatment)</FONT>
<BR><FONT SIZE=3D2>&gt; c) as you correctly stated, end to end RSVP =
signalling </FONT>
<BR><FONT SIZE=3D2>&gt; (inheritance of Signalled SIP QoS =
parameters)</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Jac</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Monday, December 30, 2002 5:37 PM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Mukul Purohit</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] resource reservation</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Mukul Purohit wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; hi,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; I have a basic question after going =
through RFC 3312, </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;Integration of </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Resource Management and SIP&quot;.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Has the actual resource reservation =
mechanism been </FONT>
<BR><FONT SIZE=3D2>&gt; specified (mandated </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; or suggested) by IETF?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; There is no mandatory resource reservation =
scheme. The primary one </FONT>
<BR><FONT SIZE=3D2>&gt; specified so far is RSVP. nsis is another one =
under </FONT>
<BR><FONT SIZE=3D2>&gt; development. Groups </FONT>
<BR><FONT SIZE=3D2>&gt; like packetcable have specified their own QoS =
protocols.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Also, has it been specified by any other =
body like 3GPP </FONT>
<BR><FONT SIZE=3D2>&gt; which has chosen </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; SIP for signaling?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Yes, 3gpp has its own mechanisms that are =
specific to 3gpp. As I </FONT>
<BR><FONT SIZE=3D2>&gt; understand they are built into the GPRS =
procedures, and reservation </FONT>
<BR><FONT SIZE=3D2>&gt; occurs only on the wireless link.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; -- </FONT>
<BR><FONT SIZE=3D2>&gt; Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 72 Eagle Rock Ave.</FONT>
<BR><FONT SIZE=3D2>&gt; Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; First Floor</FONT>
<BR><FONT SIZE=3D2>&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; East =
Hanover, NJ 07936</FONT>
<BR><FONT SIZE=3D2>&gt; =
jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
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Date: Wed, 08 Jan 2003 09:08:29 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [Sip] resource reservation
In-reply-to: 
 <A3C2399B2FACD411A54200508BE39C74054F7A62@zwcwd00r.europe.nortel.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Pianigiani, Jacopo (Jacopo)'" <jpianigiani@lucent.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Mukul Purohit'" <mpurohit@neomagic.com>
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>3) The Go interface is COPS-PR, not Diameter.
yes, and RADIUS in wireline systems. Also not DIAMETER.

Thanks, Henry

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of Mark
Watson
Sent: Wednesday, January 08, 2003 5:50 AM
To: 'Pianigiani, Jacopo (Jacopo)'; 'Jonathan Rosenberg'; Mukul Purohit
Cc: sip@ietf.org
Subject: RE: [Sip] resource reservation


Well, for the avoidance of confusion: 
1) GPRS terminals need to perform a GPRS-specific resource reservation
mechanims to get resources on the radio network/GPRS core network. This
is known as a PDP Context. 3GPP IMS uses the "Integration of Resource
Management and SIP" mechanisms to delay completion of the SIP session
establishment until this GPRS-specific procedure has been completed.
2) Various mechanism can be used within the GPRS network/RAN to ensure
that the traffic receives the QoS agreed in the PDP Context
establishment negotiation. This could include Diffserv/MPLS but is
internal to the GPRS network and not visible to the terminal
3) The Go interface is COPS-PR, not Diameter. It is used to authorise
PDP Context establishements which are related to SIP sessions.
4) The terminal is free to attempt RSVP reservations, but these will
have no effect on the GPRS network, unless the Layer 2 in the terminal
is trapping them and firing off appropriate PDP Context requests. The
RSVP may of course affect 'downstream' networks outside GPRS.
...Mark 
> -----Original Message----- 
> From: Pianigiani, Jacopo (Jacopo) [mailto:jpianigiani@lucent.com] 
> Sent: 07 January 2003 17:15 
> To: 'Jonathan Rosenberg'; Mukul Purohit 
> Cc: sip@ietf.org 
> Subject: RE: [Sip] resource reservation 
> 
> 
> Jonathan & Mukul, 
> as far as I know, referring to 3GPP, Rel.5 (frozen) and rel. 
> 6 (being written almost now) specs the following QoS 
> alternatives mechanisms to pure RSVP reservation mechanism 
> will be available: 
> a) Go Diameter interface on the GGSN (traffic gating & 
> shaping + CAC) + conditioning of the Radio Access Bearer 
> b) Diffserv and E/L-LSP QoS transport across RAN (but not for 
> reservation, just for PHB treatment) 
> c) as you correctly stated, end to end RSVP signalling 
> (inheritance of Signalled SIP QoS parameters) 
> 
> Jac 
> 
> -----Original Message----- 
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
> Sent: Monday, December 30, 2002 5:37 PM 
> To: Mukul Purohit 
> Cc: sip@ietf.org 
> Subject: Re: [Sip] resource reservation 
> 
> 
> 
> 
> Mukul Purohit wrote: 
> > hi, 
> > I have a basic question after going through RFC 3312, 
> "Integration of 
> > Resource Management and SIP". 
> > 
> > Has the actual resource reservation mechanism been 
> specified (mandated 
> > or suggested) by IETF? 
> 
> There is no mandatory resource reservation scheme. The primary one 
> specified so far is RSVP. nsis is another one under 
> development. Groups 
> like packetcable have specified their own QoS protocols. 
> 
> > Also, has it been specified by any other body like 3GPP 
> which has chosen 
> > SIP for signaling? 
> 
> Yes, 3gpp has its own mechanisms that are specific to 3gpp. As I 
> understand they are built into the GPRS procedures, and reservation 
> occurs only on the wireless link. 
> 
> -Jonathan R. 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave. 
> Chief Scientist                             First Floor 
> dynamicsoft                                 East Hanover, NJ 07936 
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050 
> http://www.jdrosen.net                      PHONE: (973) 952-5000 
> http://www.dynamicsoft.com 
> 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip 
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip 
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Pianigiani, Jacopo (Jacopo)" <jpianigiani@lucent.com>
To: "'Mark Watson'" <mwatson@nortelnetworks.com>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] resource reservation
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Mark,
I've written some thoughts inline along your mail. 
Thanks for your comments and regards.

Jac
 -----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Wednesday, January 08, 2003 12:50 PM
To: 'Pianigiani, Jacopo (Jacopo)'; 'Jonathan Rosenberg'; Mukul Purohit
Cc: sip@ietf.org
Subject: RE: [Sip] resource reservation

Well, for the avoidance of confusion: 

1) GPRS terminals need to perform a GPRS-specific resource reservation mechanims to get resources on the radio network/GPRS core network. This is known as a PDP Context. 3GPP IMS uses the "Integration of Resource Management and SIP" mechanisms to delay completion of the SIP session establishment until this GPRS-specific procedure has been completed. 

2) Various mechanism can be used within the GPRS network/RAN to ensure that the traffic receives the QoS agreed in the PDP Context establishment negotiation. This could include Diffserv/MPLS but is internal to the GPRS network and not visible to the terminal     

 3) The Go interface is COPS-PR, not Diameter. It is used to authorise PDP Context establishements which are related to SIP sessions. 

Sorry for the confusion in writing. What i meant is that both Go (COPS-PR) and Diameter are used netween IMS and HSS and IMS and GGSN (Diameter is used to perform AAA function toward the HSS and this is applied whenever a PDP context with specific QoS constraint is requested from the IMS then Go implements the policy decisions. ). 

4) The terminal is free to attempt RSVP reservations, but these will have no effect on the GPRS network, unless the Layer 2 in the terminal is trapping them and firing off appropriate PDP Context requests. The RSVP may of course affect 'downstream' networks outside GPRS. 

Agreed. But if we consider a scenario where there is a mechanism to reserve QoS for SIP sessions is in place within the GPRS network, then it could be relevant to have a method to extend QoS from the edge of the GPRS network (GGSN) onward, to the other end (external bearer transport QoS and interworking of QoS parameters across distinct QoS domains)

...Mark 

> -----Original Message----- 
> From: Pianigiani, Jacopo (Jacopo) [ mailto:jpianigiani@lucent.com <mailto:jpianigiani@lucent.com> ] 
> Sent: 07 January 2003 17:15 
> To: 'Jonathan Rosenberg'; Mukul Purohit 
> Cc: sip@ietf.org 
> Subject: RE: [Sip] resource reservation 
> 
> 
> Jonathan & Mukul, 
> as far as I know, referring to 3GPP, Rel.5 (frozen) and rel. 
> 6 (being written almost now) specs the following QoS 
> alternatives mechanisms to pure RSVP reservation mechanism 
> will be available: 
> a) Go Diameter interface on the GGSN (traffic gating & 
> shaping + CAC) + conditioning of the Radio Access Bearer 
> b) Diffserv and E/L-LSP QoS transport across RAN (but not for 
> reservation, just for PHB treatment) 
> c) as you correctly stated, end to end RSVP signalling 
> (inheritance of Signalled SIP QoS parameters) 
> 
> Jac 
> 
> -----Original Message----- 
> From: Jonathan Rosenberg [ mailto:jdrosen@dynamicsoft.com <mailto:jdrosen@dynamicsoft.com> ] 
> Sent: Monday, December 30, 2002 5:37 PM 
> To: Mukul Purohit 
> Cc: sip@ietf.org 
> Subject: Re: [Sip] resource reservation 
> 
> 
> 
> 
> Mukul Purohit wrote: 
> > hi, 
> > I have a basic question after going through RFC 3312, 
> "Integration of 
> > Resource Management and SIP". 
> > 
> > Has the actual resource reservation mechanism been 
> specified (mandated 
> > or suggested) by IETF? 
> 
> There is no mandatory resource reservation scheme. The primary one 
> specified so far is RSVP. nsis is another one under 
> development. Groups 
> like packetcable have specified their own QoS protocols. 
> 
> > Also, has it been specified by any other body like 3GPP 
> which has chosen 
> > SIP for signaling? 
> 
> Yes, 3gpp has its own mechanisms that are specific to 3gpp. As I 
> understand they are built into the GPRS procedures, and reservation 
> occurs only on the wireless link. 
> 
> -Jonathan R. 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave. 
> Chief Scientist                             First Floor 
> dynamicsoft                                 East Hanover, NJ 07936 
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050 
> http://www.jdrosen.net <http://www.jdrosen.net>                       PHONE: (973) 952-5000 
> http://www.dynamicsoft.com <http://www.dynamicsoft.com>  
> 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip <https://www1.ietf.org/mailman/listinfo/sip>  
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip <https://www1.ietf.org/mailman/listinfo/sip>  
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip 
> 


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<TITLE>RE: [Sip] resource reservation</TITLE>

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<DIV><FONT face=Tahoma><FONT size=2><SPAN class=412015714-08012003><FONT 
face="Trebuchet MS" color=#0000ff>Mark,</FONT></SPAN></FONT></FONT><FONT 
face=Tahoma><FONT size=2><SPAN class=412015714-08012003><FONT 
face="Trebuchet MS" color=#0000ff></FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT face=Tahoma><FONT size=2><SPAN class=412015714-08012003><FONT 
face="Trebuchet MS" color=#0000ff>I've written some thoughts inline along your 
mail. </FONT></SPAN></FONT></FONT></DIV>
<DIV><FONT face=Tahoma><FONT size=2><SPAN class=412015714-08012003><FONT 
face="Trebuchet MS" color=#0000ff>Thanks for your comments and 
regards.</FONT></SPAN></FONT></FONT></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma><FONT 
  color=#0000ff size=2><SPAN 
  class=412015714-08012003>Jac</SPAN></FONT></FONT></DIV>
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma><FONT 
  size=2><SPAN class=412015714-08012003>&nbsp;</SPAN>-----Original 
  Message-----<BR><B>From:</B> Mark Watson 
  [mailto:mwatson@nortelnetworks.com]<BR><B>Sent:</B> Wednesday, January 08, 
  2003 12:50 PM<BR><B>To:</B> 'Pianigiani, Jacopo (Jacopo)'; 'Jonathan 
  Rosenberg'; Mukul Purohit<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> RE: 
  [Sip] resource reservation</DIV></FONT></FONT>
  <P><FONT size=2>Well, for the avoidance of confusion:</FONT> </P>
  <P><FONT size=2>1) GPRS terminals need to perform a GPRS-specific resource 
  reservation mechanims to get resources on the radio network/GPRS core network. 
  This is known as a PDP Context. 3GPP IMS uses the "Integration of Resource 
  Management and SIP" mechanisms to delay completion of the SIP session 
  establishment until this GPRS-specific procedure has been completed.<SPAN 
  class=412015714-08012003><FONT face="Trebuchet MS" 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></P>
  <P><FONT size=2>2) Various mechanism can be used within the GPRS network/RAN 
  to ensure that the traffic receives the QoS agreed in the PDP Context 
  establishment negotiation. This could include Diffserv/MPLS but is internal to 
  the GPRS network and not visible to the terminal<FONT 
  face="Trebuchet MS"><FONT color=#0000ff><SPAN 
  class=412015714-08012003>&nbsp;</SPAN></FONT></FONT></FONT><FONT size=2><FONT 
  face="Trebuchet MS"><FONT color=#0000ff><SPAN 
  class=412015714-08012003>&nbsp;</SPAN><SPAN 
  class=412015714-08012003>&nbsp;</SPAN></FONT></FONT><SPAN 
  class=412015714-08012003>&nbsp;</SPAN><SPAN 
  class=412015714-08012003>&nbsp;</SPAN></FONT></P>
  <P><FONT size=2><SPAN class=412015714-08012003>&nbsp;</SPAN>3) The Go 
  interface is COPS-PR, not Diameter. It is used to authorise PDP Context 
  establishements which are related to SIP sessions.<SPAN 
  class=412015714-08012003><FONT face="Trebuchet MS" 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></P>
  <P><FONT size=2><SPAN class=412015714-08012003><FONT face="Trebuchet MS" 
  color=#0000ff>Sorry for the confusion in writing. What i meant is that 
  </FONT></SPAN></FONT><FONT size=2><SPAN class=412015714-08012003><FONT 
  face="Trebuchet MS" color=#0000ff>both Go (COPS-PR) and Diameter are 
  used&nbsp;netween IMS and HSS and IMS and GGSN (Diameter is used to 
  perform&nbsp;AAA function toward the&nbsp;HSS and this is applied whenever a 
  PDP context with specific QoS constraint is requested from the IMS then Go 
  implements the policy decisions. ). </FONT></SPAN></FONT></P>
  <P><FONT size=2>4) The terminal is free to attempt RSVP reservations, but 
  these will have no effect on the GPRS network, unless the Layer 2 in the 
  terminal is trapping them and firing off appropriate PDP Context requests. The 
  RSVP may of course affect 'downstream' networks outside GPRS.<SPAN 
  class=412015714-08012003><FONT face="Trebuchet MS" 
  color=#0000ff>&nbsp;</FONT></SPAN></FONT></P>
  <P><FONT face="Trebuchet MS" color=#0000ff size=2><SPAN 
  class=412015714-08012003>Agreed. But if we consider a scenario where there is 
  a mechanism to reserve QoS for SIP sessions is in place within the GPRS 
  network, then it could be relevant to have a method to extend QoS from the 
  edge of the GPRS network (GGSN) onward, to the other end (external bearer 
  transport QoS and interworking of QoS parameters across distinct QoS 
  domains)</SPAN></FONT></P>
  <P><FONT size=2>...Mark</FONT> </P>
  <P><FONT size=2>&gt; -----Original Message-----</FONT> <BR><FONT size=2>&gt; 
  From: Pianigiani, Jacopo (Jacopo) [<A 
  href="mailto:jpianigiani@lucent.com">mailto:jpianigiani@lucent.com</A>]</FONT> 
  <BR><FONT size=2>&gt; Sent: 07 January 2003 17:15</FONT> <BR><FONT size=2>&gt; 
  To: 'Jonathan Rosenberg'; Mukul Purohit</FONT> <BR><FONT size=2>&gt; Cc: 
  sip@ietf.org</FONT> <BR><FONT size=2>&gt; Subject: RE: [Sip] resource 
  reservation</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; Jonathan &amp; Mukul,</FONT> <BR><FONT 
  size=2>&gt; as far as I know, referring to 3GPP, Rel.5 (frozen) and rel. 
  </FONT><BR><FONT size=2>&gt; 6 (being written almost now) specs the following 
  QoS </FONT><BR><FONT size=2>&gt; alternatives mechanisms to pure RSVP 
  reservation mechanism </FONT><BR><FONT size=2>&gt; will be available:</FONT> 
  <BR><FONT size=2>&gt; a) Go Diameter interface on the GGSN (traffic gating 
  &amp; </FONT><BR><FONT size=2>&gt; shaping + CAC) + conditioning of the Radio 
  Access Bearer</FONT> <BR><FONT size=2>&gt; b) Diffserv and E/L-LSP QoS 
  transport across RAN (but not for </FONT><BR><FONT size=2>&gt; reservation, 
  just for PHB treatment)</FONT> <BR><FONT size=2>&gt; c) as you correctly 
  stated, end to end RSVP signalling </FONT><BR><FONT size=2>&gt; (inheritance 
  of Signalled SIP QoS parameters)</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; Jac</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; 
  -----Original Message-----</FONT> <BR><FONT size=2>&gt; From: Jonathan 
  Rosenberg [<A 
  href="mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</A>]</FONT> 
  <BR><FONT size=2>&gt; Sent: Monday, December 30, 2002 5:37 PM</FONT> <BR><FONT 
  size=2>&gt; To: Mukul Purohit</FONT> <BR><FONT size=2>&gt; Cc: 
  sip@ietf.org</FONT> <BR><FONT size=2>&gt; Subject: Re: [Sip] resource 
  reservation</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; </FONT><BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; Mukul Purohit wrote:</FONT> <BR><FONT size=2>&gt; &gt; hi,</FONT> 
  <BR><FONT size=2>&gt; &gt; I have a basic question after going through RFC 
  3312, </FONT><BR><FONT size=2>&gt; "Integration of </FONT><BR><FONT 
  size=2>&gt; &gt; Resource Management and SIP".</FONT> <BR><FONT size=2>&gt; 
  &gt; </FONT><BR><FONT size=2>&gt; &gt; Has the actual resource reservation 
  mechanism been </FONT><BR><FONT size=2>&gt; specified (mandated 
  </FONT><BR><FONT size=2>&gt; &gt; or suggested) by IETF?</FONT> <BR><FONT 
  size=2>&gt; </FONT><BR><FONT size=2>&gt; There is no mandatory resource 
  reservation scheme. The primary one </FONT><BR><FONT size=2>&gt; specified so 
  far is RSVP. nsis is another one under </FONT><BR><FONT size=2>&gt; 
  development. Groups </FONT><BR><FONT size=2>&gt; like packetcable have 
  specified their own QoS protocols.</FONT> <BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; &gt; Also, has it been specified by any other 
  body like 3GPP </FONT><BR><FONT size=2>&gt; which has chosen </FONT><BR><FONT 
  size=2>&gt; &gt; SIP for signaling?</FONT> <BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; Yes, 3gpp has its own mechanisms that are 
  specific to 3gpp. As I </FONT><BR><FONT size=2>&gt; understand they are built 
  into the GPRS procedures, and reservation </FONT><BR><FONT size=2>&gt; occurs 
  only on the wireless link.</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; -Jonathan R.</FONT> <BR><FONT size=2>&gt; </FONT><BR><FONT 
  size=2>&gt; -- </FONT><BR><FONT size=2>&gt; Jonathan D. Rosenberg, 
  Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  72 Eagle Rock Ave.</FONT> <BR><FONT size=2>&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; 
  First Floor</FONT> <BR><FONT size=2>&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; 
  East Hanover, NJ 07936</FONT> <BR><FONT size=2>&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; (973) 952-5050</FONT> <BR><FONT size=2>&gt; <A 
  href="http://www.jdrosen.net" 
  target=_blank>http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  PHONE: (973) 952-5000</FONT> <BR><FONT size=2>&gt; <A 
  href="http://www.dynamicsoft.com" 
  target=_blank>http://www.dynamicsoft.com</A></FONT> <BR><FONT size=2>&gt; 
  </FONT><BR><FONT size=2>&gt; 
  _______________________________________________</FONT> <BR><FONT size=2>&gt; 
  Sip mailing list&nbsp; <A href="https://www1.ietf.org/mailman/listinfo/sip" 
  target=_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> <BR><FONT 
  size=2>&gt; This list is for NEW development of the core SIP Protocol</FONT> 
  <BR><FONT size=2>&gt; Use sip-implementors@cs.columbia.edu for questions on 
  current sip</FONT> <BR><FONT size=2>&gt; Use sipping@ietf.org for new 
  developments on the application of sip</FONT> <BR><FONT size=2>&gt; 
  _______________________________________________</FONT> <BR><FONT size=2>&gt; 
  Sip mailing list&nbsp; <A href="https://www1.ietf.org/mailman/listinfo/sip" 
  target=_blank>https://www1.ietf.org/mailman/listinfo/sip</A></FONT> <BR><FONT 
  size=2>&gt; This list is for NEW development of the core SIP Protocol</FONT> 
  <BR><FONT size=2>&gt; Use sip-implementors@cs.columbia.edu for questions on 
  current sip</FONT> <BR><FONT size=2>&gt; Use sipping@ietf.org for new 
  developments on the application of sip</FONT> <BR><FONT size=2>&gt; 
</FONT></P></BLOCKQUOTE></BODY></HTML>

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From: "Tom-PT Taylor" <taylor@nortelnetworks.com>
To: "'Frank W. Miller'" <fmiller@sentito.com>,
        Dean Willis
	 <dean.willis@softarmor.com>
Cc: sip@ietf.org, "James McEachern" <jmce@nortelnetworks.com>,
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Subject: RE: [Sip] INFO considered harmful
Date: Wed, 8 Jan 2003 11:02:53 -0500 
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I don't know if there is a relationship, but Cisco brought just such a
text-encoding proposal to ITU-T Study Group 11.  I very much agree that if
this proposal goes forward, it should use XML rather than some arbitrary
TL-1-like language.

> -----Original Message-----
> From: Frank W. Miller [mailto:fmiller@sentito.com] 
> Sent: Friday, January 03, 2003 7:02 PM
> To: Dean Willis
> Cc: sip@ietf.org
> Subject: Re: [Sip] INFO considered harmful
> 
> 
> 
> 
> Well, any illusion of reasonableness I may have garnered in 
> the past is about to be shattered.  I would refer folks to:
> 
> http://www.ietf.org/internet-drafts/draft-miller-sip-isup-annex-00.txt
> 
> This draft is about to receive a major -01 update that 
> recodes the entire syntax in XML.  In addition, we (I) have 
> another draft that does something very similar for ANSI TCAP 
> that I'll be submitting in short order.  I guess its only 
> appropriate that I bring peoples attention to these drafts 
> under the current subject heading...
> 
> FM
> 
> Frank W. Miller, Ph.D.
> Chief Technical Officer
> sentitO Networks, Inc.
> fmiller@sentito.com
> 
> ----- Original Message -----
> From: "Dean Willis" <dean.willis@softarmor.com>
> To: "'Neil Deason'" <ndeason@ubiquity.net>; "'Adam Roach'" 
> <adam@dynamicsoft.com>
> Cc: <sip@ietf.org>
> Sent: Friday, January 03, 2003 6:13 PM
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > > What I was actually advocating here was ppl wanting to keep INFO 
> > > around write up something to illuminate INFO packages and provide 
> > > example usage.
> > >
> > > The MESSAGE cf INFO was just an attempt to highlight an apparent 
> > > limitation with the ban INFO approach. Every MIME type in an INFO 
> > > can be moved into MESSAGE rather than go away. Maybe through 
> > > exploring INFO packages we evolve the collective 
> understanding that 
> > > this is/isn't a necessary thing.
> > >
> > > Cheers,
> > > Neil.
> >
> > This is an excellent suggestion, and very similar to the 
> one Jonathan 
> > made to me on the phone yesterday (actually, I think he 
> more tersely 
> > worded it
> as
> > "Send Text!")
> >
> > --
> > Dean
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> >
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 
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From: "Louis Alexander" <louis.alexander@cmg.com>
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Subject: [Sip] Bulk registration
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Hello Everyone,
    Can anyone point me to a draft that proposes a method for bulk
registration?  That is, a UA performing a bulk registration for a domain
address, or a telephone number wild card?  
 
Louis G. Alexander
Software Engineer
CMG Wireless Data Solutions
+1 804 935 4707 (SNR)
louis.alexander@cmg.com
 

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
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<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D222240220-08012003><FONT face=3DArial size=3D2>Hello=20
Everyone,</FONT></SPAN></DIV>
<DIV><SPAN class=3D222240220-08012003>&nbsp;&nbsp;&nbsp; <FONT =
face=3DArial=20
size=3D2>Can anyone point me to a draft that proposes a method for bulk=20
registration?&nbsp; That is, a UA performing a bulk registration for a =
domain=20
address, or a telephone number wild card?&nbsp; </FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Louis G. =
Alexander</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>Software =
Engineer</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>CMG Wireless Data =
Solutions</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial size=3D2>+1 804 935 4707 =
(SNR)</FONT></DIV>
<DIV align=3Dleft><FONT face=3DArial =
size=3D2>louis.alexander@cmg.com</FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From mailnull@www1.ietf.org  Wed Jan  8 16:27:50 2003
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Date: Wed, 08 Jan 2003 22:25:00 +0100
To: Henry Sinnreich <Henry.Sinnreich@wcom.com>,
        "'Mark Watson'" <mwatson@nortelnetworks.com>,
        "'Pianigiani, Jacopo (Jacopo)'" <jpianigiani@lucent.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
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From: Jiri Kuthan <jiri@iptel.org>
Subject: RE: [Sip] resource reservation
Cc: sip@ietf.org
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At 04:08 PM 1/8/2003, Henry Sinnreich wrote:
>>3) The Go interface is COPS-PR, not Diameter.
>yes, and RADIUS in wireline systems. Also not DIAMETER.

Let my add my two cents: quite many our users are interesterd
in use of RADIUS for SIP accounting, and there is zero interest
in diameter. The reason is there is a widely deployed RADIUS
base. To operators, diameter benefits are apparently less
compelling than the upgrade inconvenience. I'm sort of
surprised, that the WG has ignored this situation and has
not produced any document on use of RADIUS for SIP.

-Jiri  

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From: Adam Roach <adam@dynamicsoft.com>
To: "'Louis Alexander'" <louis.alexander@cmg.com>, sip@ietf.org
Subject: RE: [Sip] Bulk registration
Date: Wed, 8 Jan 2003 16:32:49 -0600 
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What you want is TRIP, not SIP.

See <http://www.ietf.org/internet-drafts/draft-ietf-iptel-trip-09.txt>.

/a

-----Original Message-----
From: Louis Alexander [mailto:louis.alexander@cmg.com]
Sent: Wednesday, January 08, 2003 14:04
To: sip@ietf.org
Subject: [Sip] Bulk registration


Hello Everyone,
    Can anyone point me to a draft that proposes a method for bulk
registration?  That is, a UA performing a bulk registration for a domain
address, or a telephone number wild card?  

Louis G. Alexander
Software Engineer
CMG Wireless Data Solutions
+1 804 935 4707 (SNR)
louis.alexander@cmg.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Date: Wed, 08 Jan 2003 17:52:01 -0500
From: "David R. Oran" <oran@cisco.com>
To: Louis Alexander <louis.alexander@cmg.com>, sip@ietf.org
Subject: Re: [Sip] Bulk registration
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--On Wednesday, January 08, 2003 3:03 PM -0500 Louis Alexander 
<louis.alexander@cmg.com> wrote:

>
> Hello Everyone,
>     Can anyone point me to a draft that proposes a method for bulk
> registration?  That is, a UA performing a bulk registration for a domain
> address, or a telephone number wild card?

There is none, because SIP does not support "bulk registrations". There are 
very good reasons why. This comes up periodically, so perhaps we should 
have an entry on this subject in the SIP FAQ if there is not one already.

Instead, you probably want to look at TRIP or TGRP depending on exactly 
what you want to accomplish.


> Louis G. Alexander
> Software Engineer
> CMG Wireless Data Solutions
> +1 804 935 4707 (SNR)
> louis.alexander@cmg.com
>

------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Prasanna Venkatesh'" <prasanna@huawei.com>,
        <joanne.higgens@orange.co.uk>, <sip@ietf.org>
Subject: RE: [Sip] SIP de-registration
Date: Wed, 8 Jan 2003 16:54:53 -0600
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Prasanna said:
>Jo said:
>> Is it valid to reject a de-registration request?
>> 
>> For example 'If a  SIP terminal sends a de-registration 
>> request to a network while it still has an active session 
>> established, and the network rejects this request , is this 
>> correct behaviour.

> Any registrar can reject a de-registration request anytime.  
> But rejecting a de-registration for the fact that a session 
> is active is a local implementation.  In general the 
> registration is used to locate an entity and theoritically 
> after the session is established, the registration is 
> immaterial. Cheers, Prasanna

I believe Prasanna is correct here, in general. SIP does not view REGISTER
as a "log on for service" -- rather, it is an address binding function that
makes a temporary association between a (presumbaly temporary) local
instance address and a (presumbaly long-lived) address-of-record. In
particular, SIP does not require that a UA have a current registration or
execute a REGISTER transaction before executing an INVITE transaction
request.

3GPP (and consequently 3GPP2) have however layered some extra roles on top
of the SIP REGISTER transaction. In these environments, REGISTER is a sort
of "log on" operation in that it establishes the security association
between a client and registrar that will be used for (some) future
interactions between that client and registrar. 

Consequently, answering the question that Jo asked requires knowing whether
the 3GPP system discards the security association established by a REGISTER
request if the registration created by that request is destroyed.

The answer, unfortunately, is "I don't know. The IETF doesn't know. Maybe
somebody more conversant with that particular bit of 3GPPP knows (and
they're probably reading this)." But in short, it is an environment specific
question related to a specific implementation having done something just a
bit different from generic SIP. In straight SIP, the registrar can terminate
a registration without impacting any sessions that a UA is participating in.
In fact, the registrar usually doesn't know or care whether a UA has or will
ever have participated in a session.

My guess is that this would probably be implementation-depdendent in the
S-CSCF, and that if I were designing an S-CSCF, I'd have it not discard
SA-state until after all transacations on that SA had terminated. A second
choice would be to design it to use network-initiated-termination procedures
to force-terminate any sessions before discarding the security associations.

--
dean

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From mailnull@www1.ietf.org  Wed Jan  8 17:58:06 2003
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Wed, 8 Jan 2003 16:54:53 -0600
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> > No, EVENTS doesn't define semantics.
> 
> Of course it does. There are lots of MAYS, MUSTS and SHOULDs 
> in there. 
> There is behavior dealing with subscription refreshes, rules 
> on when to 
> send NOTIFY, and so on and so on. How can you say that none of those 
> qualify as semantics?


That's called a "protocol specification", not a specification of the
semanticsd of the delivered content.

What does a NOTIFY mean? Does it mean that I have voice mail and I should
poll it? Does it mean that somebody has subscribed to my presence and I
should adjust my authorization policy? Does it mean that my laundry is done
and I should go pick it up? Does it mean that the IRS is going to audit my
books and I should flee for the border?  

You can't answer this question, in general, because those semantics are NOT
defined in RFC 3265. They're defined in event packages.

Similarly, we complain because the INFO spec doen't tell us what an INFO
means. Should we convert the body to ISUP? Should we execute the operator
now? Should the phone explode? The INFO spec should no more answer this than
does the Events spec. The semantic encoding is a separate problem.

Now, here's the difference. I don't think you're talking about the presence
of semantic information. I think you're talking about the absence of a
mechanism for documenting the semantic information. Event packages give us
this mechanism for events. We don't have anything like that for INFO. You're
not talking semantics -- you're talking tools for exchanging meta-semantics.

> 
> > Event PACKAGES define the semantics of
> > particular instances of the event methods.
> 
> Within a well defined framework. 3265 defines clear rules on 
> what each 
> package has to specify, and what it can and cannot due ontop 
> of rfc3265.
> 
> 
> > INFO also needs a blob-describing-and-encoding framework, 
> so that the 
> > various uses of INFO can, as appropriate, be effectively documented 
> > and negotiated.
> 
> Well, I'll try one more time, since I am clearly not getting my point 
> across.
> 
> RFC3265 defines semantics above and beyond RFC3261. All of those 
> MAY/MUST/SHOULD I was talking about. If you had a choice between 
> defining a new package, and defining your own sub/not mechanisms, it 
> would be a LOT more work to define your own sub/not mechanisms.

huh?

It would be much more work to independently reimplement the functionality of
INFO than it is to encode some information into MIME and ship it via INFO,
especially given that the main functionality of INFO is correlated
resolution with an existing dialog (identical rendezvous).

> Not so for INFO. WHen deciding whether to specify an INFO usage, or a 
> new method, there is no additional work to do 
> (specification-wise) when 
> specifying a new method as opposed to a new INFO usage. To me, that 
> means that INFO is devoid of semantics.

No, it just means that the process for defining a new INFO usage is broken.

--
Dean

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Date: Wed, 08 Jan 2003 20:58:41 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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There is a long-expired document on this topic; 
draft-schulzrinne-sipping-radius-accounting-00.txt

You can find it in the musty corners of 
http://www.cs.columbia.edu/sip/drafts/

By reading it, you will lose all your political correctness points.

Jiri Kuthan wrote:
> At 04:08 PM 1/8/2003, Henry Sinnreich wrote:
> 
>>>3) The Go interface is COPS-PR, not Diameter.
>>
>>yes, and RADIUS in wireline systems. Also not DIAMETER.
> 
> 
> Let my add my two cents: quite many our users are interesterd
> in use of RADIUS for SIP accounting, and there is zero interest
> in diameter. The reason is there is a widely deployed RADIUS
> base. To operators, diameter benefits are apparently less
> compelling than the upgrade inconvenience. I'm sort of
> surprised, that the WG has ignored this situation and has
> not produced any document on use of RADIUS for SIP.
> 
> -Jiri  
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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hi,

i have some questions based on rfc 3263 "locating sip services". in
rfc 3263 it is mentioned that:
  "For NAPTR records with SIPS protocol fields, (if the server is using
   a site certificate), the domain name in the query and the domain name
   in the replacement field MUST both be valid based on the site
   certificate handed out by the server in the TLS exchange.  Similarly,
   the domain name in the SRV query and the domain name in the target in
   the SRV record MUST both be valid based on the same site certificate.
   Otherwise, an attacker could modify the DNS records to contain
   replacement values in a different domain, and the client could not
   validate that this was the desired behavior or the result of an
   attack."

now, there are three potentially different domains in concern while
processing a SIPS request:
	1. domain used in NAPTR querry: this is the target domain for this
	   request
	2. domain mentioned in replacement field of NAPTR records: this
	   is the domain used in the SRV querry, obtained from replacement
	   field of NAPTR records
	3. domain mentioned in target in the SRV record: the domain
	   mentioned in the SRV record selected for final A lookup

is it possible to have all the three domains different, or is it just a
bad example (due to my limited knowledge of DNS setups), and in practical
cases the three domains will either be same as TARGET or a subdomain of
it?

incase my imagination is correct, and it is possible to have a setup where
all three domains (from three steps of DNS procedures) are different, then
how can a client validate these three domains using the same certificate?

i'd appreciate your comments/explanation to my querries.

thanks,
-amit.
----------------------------------------------------------------------------
I have hardly ever known a mathematician who was capable of reasoning.
   -- gnulib

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Subject: [Sip] Sip REGISTRATION
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Hi

I am new to SIP Registration process. I have the following doubt:

In a UA REGISTERation process with a proxy/registrar, the
Unauthourized(407) is posed against the To OR the From header of the
Register request? i.e, the Registrar  tries to authorize the to
hreader URI or the From header URI ( in the case when ther are
different as in for a third party registration )?

Please help!
Thanks
CS

__________________________________
csa@postmark.net

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Thread-Topic: free SIP stack?
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Hi there,

Could anybody please recommend a good and free SIP stack?

Regards,

Yuyong
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Subject: [Sip] about md5 algorithm for authorization header
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HELLO SIR,

I read document in RFC 1321 .But i have some ploblem i
don't understand in
request-digest  =<"> < KD ( H(A1), unq(nonce-value)
":" H(A2) ) ><">


request-digest  = "KD(H(A1),unq(nonce-value) ":" H(A2)
)"

A1 = unq(username-value) ":" unq(realm-value) ":"
passwd

A2 = Method ":" digest-uri-value

what KD?,H?

weather This  equation is correct or not ?

KD(H(A1), unq(nonce-value) ":" H(A2)) = MD5(H

(A1), unq(nonce-value) ":" H(A2))

Thanking you..

Regards
krishna

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From: KANUMURI Sreeram <SKanumur@covansys.com>
To: "'Yuyong Zhang'" <yzhang@cedarpointcom.com>, sip@ietf.org
Subject: RE: [Sip] free SIP stack?
Date: Thu, 9 Jan 2003 02:12:21 -0500 
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see www.vovida.org you can get the source code in C++


-Sreeram

-----Original Message-----
From: Yuyong Zhang [mailto:yzhang@cedarpointcom.com]
Sent: Friday, January 03, 2003 11:22 PM
To: sip@ietf.org
Subject: [Sip] free SIP stack?


Hi there,

Could anybody please recommend a good and free SIP stack?

Regards,

Yuyong
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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_______________________________________________
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From mailnull@www1.ietf.org  Thu Jan  9 03:42:11 2003
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From: John J C <john@hirp.iisc.ernet.in>
To: KANUMURI Sreeram <SKanumur@covansys.com>
cc: "'Yuyong Zhang'" <yzhang@cedarpointcom.com>, <sip@ietf.org>
Subject: RE: [Sip] free SIP stack?
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try osip. The source code is in C . U can get it at the following URL.

http://freesoftware.fsf.org/download/osip/libosip-0.8.0.tar.gz


John

				        John J Chooracken
                                        Senior Engineer - R&D
                                        HFCL-IISC Research Program
                                        IISc Campus - Bangalore - 560 012
                                        Voice: (+91)-80-3562062
                                        Email: john@hirp.iisc.ernet.in
                                               chooracken@yahoo.com






On Thu, 9 Jan 2003, KANUMURI Sreeram wrote:

>
> see www.vovida.org you can get the source code in C++
>
>
> -Sreeram
>
> -----Original Message-----
> From: Yuyong Zhang [mailto:yzhang@cedarpointcom.com]
> Sent: Friday, January 03, 2003 11:22 PM
> To: sip@ietf.org
> Subject: [Sip] free SIP stack?
>
>
> Hi there,
>
> Could anybody please recommend a good and free SIP stack?
>
> Regards,
>
> Yuyong
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

-- 


_______________________________________________
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Hi,

> Your example clearly falls into the "> 10" category as the poor web
> server had to deal with every Real audio or video packet. Not that it
> would surprise me, but I don't think anybody *here* has suggested
> RTP-over-INFO so far.

Isn't that part of the missuse problem we are talking about here - that
someone actually COULD start using INFO for sending RTP (or whatever else
that would creating frequent message sending)?

Regards,

Christer Holmberg
Ericsson Finland



>
>
> Adam Roach wrote:
> >>From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> >>
> >>In general, I just don't buy server load arguments. If it's a
> >>useful service, server cost is usually the least of the problem.
> >>In any event, one should take this as a fraction of the existing
> >>signaling load, not as some absolute number. I'm worried about
> >>stuff that multiplies the signaling load by a factor of 10; I find
> >>20 or 50% load increase uninteresting as an argument.
> >
> >
> > You don't buy server load arguments? Real-world experience
> > notwithstanding?
> >
> > I'm recalling the rise in popularity of Real Media Player and
> > its ability to set up an audio stream through an HTTP proxy.
> > In particular, I'm recalling conversations shortly thereafter
> > with some friends over in MIS about how they just couldn't add
> > capacity to the HTTP proxy pool fast enough to keep up with
> > demand.
> >
> > Access to external web documents sometimes creeped up to the
> > order of minutes before everything was finally stabilized.
> >
> > How long do you guess it will be before Real player
> > (or a similar product) includes a "SIP Proxy:" field right
> > next to its "HTTP Proxy:" field for firewall circumvention?
> >
> > /a
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: Re: [Sip] INFO considered harmful
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Hi,

> Actually, there's an important difference: MESSAGE can use the SIP
> features outside an established session, while INFO is within a session
> (and thus there are other SDP-negotiated, options available). Within a
> session, the only real reason for a follow-the-INVITE SIP message is
> that a direct IP-level connection can't be made to work.

[CHH]  Please remember that in SIP-T there are needs to send INFO BEFORE the
200 OK has been sent for INVITE, which means the SDP negoatiation MAY not be
finished yet. I don't know about any other mechanisms using INFO which
require to be able to do the same, but just to keep in mind...

Regards,

Christer Holmberg
Ericsson Finland


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Date: Thu, 09 Jan 2003 09:24:13 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] I-D ACTION:draft-rosenberg-sip-info-harmful-00.txt
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Jonathan,

I would appreciate some clarification on the scope of solution 3, i.e., the
scope of INFO usage for SIP-T. INFO usage for SIP-T is for transparently
conveying ISUP messages across a SIP network between two ISUP gateways.
There is a similar requirement for conveying other protocols between peer
gateways: QSIG (for which a MIME type is already defined) and also DPNSS
(for which there is a proposal for a MIME type). Both of these require the
transport of messages of the protocol concerned at times when no other SIP
request or response is being sent, and INFO would be the obvious candidate
for this. It would be inappropriate to invent different solutions for what
is basically the same problem but just different transported protocols.

John Elwell
e-mail: mailto:john.elwell@siemens.com

> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: 07 January 2003 13:29
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-rosenberg-sip-info-harmful-00.txt
> 
> 
> A New Internet-Draft is available from the on-line 
> Internet-Drafts directories.
> 
> 
> 	Title		: The Session Initiation Protocol (SIP) 
> INFO Method 
>                           Considered Harmful
> 	Author(s)	: J. Rosenberg
> 	Filename	: draft-rosenberg-sip-info-harmful-00.txt
> 	Pages		: 11
> 	Date		: 2003-1-6
> 	
> The Session Initiation Protocol (SIP) INFO method defines a means for
> transporting mid-dialog application layer data between user agents.
> Its initial use was to support the transport of ISUP mid-call
> messages which could not be mapped to any other SIP request method.
> However, since its initial usage for that purpose, INFO has seen
> widespread abuse as a means for introducing non-standard and
> non-interoperable extensions to SIP.  For this reason, we now believe
> INFO should be considered harmful, and therefore, deprecated in its
> current form.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-info-h
armful-00.txt

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From mailnull@www1.ietf.org  Thu Jan  9 04:40:48 2003
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To: Henning Schulzrinne <hgs@cs.columbia.edu>, Jiri Kuthan <jiri@iptel.org>
From: Jiri Kuthan <jiri.kuthan@fokus.fraunhofer.de>
Subject: radius (was Re: [Sip] resource reservation
Cc: sip@ietf.org
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Actually, we have implemented it and it is fine. I'm just 
concerned about lack of a standard -- that makes interaction
of accounting apps with SIP servers difficult. I would really
like to see it as a WG item.

-Jiri

At 02:58 AM 1/9/2003, Henning Schulzrinne wrote:
>There is a long-expired document on this topic; draft-schulzrinne-sipping-radius-accounting-00.txt
>
>You can find it in the musty corners of http://www.cs.columbia.edu/sip/drafts/
>
>By reading it, you will lose all your political correctness points.
>
>Jiri Kuthan wrote:
>>At 04:08 PM 1/8/2003, Henry Sinnreich wrote:
>>
>>>>3) The Go interface is COPS-PR, not Diameter.
>>>
>>>yes, and RADIUS in wireline systems. Also not DIAMETER.
>>
>>Let my add my two cents: quite many our users are interesterd
>>in use of RADIUS for SIP accounting, and there is zero interest
>>in diameter. The reason is there is a widely deployed RADIUS
>>base. To operators, diameter benefits are apparently less
>>compelling than the upgrade inconvenience. I'm sort of
>>surprised, that the WG has ignored this situation and has
>>not produced any document on use of RADIUS for SIP.
>>-Jiri  
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

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Date: Thu, 09 Jan 2003 10:57:40 +0100
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CC: KANUMURI Sreeram <SKanumur@covansys.com>,
        "'Yuyong Zhang'" <yzhang@cedarpointcom.com>, sip@ietf.org
Subject: Re: [Sip] free SIP stack?
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This is a very old version of osip. The latest is:
http://ftp.gnu.org/gnu/osip/libosip-0.9.3.tar.gz

Thanks
Aymeric

John J C wrote:
> 
> try osip. The source code is in C . U can get it at the following URL.
> 
> http://freesoftware.fsf.org/download/osip/libosip-0.8.0.tar.gz
> 
> John
> 
>                                         John J Chooracken
>                                         Senior Engineer - R&D
>                                         HFCL-IISC Research Program
>                                         IISc Campus - Bangalore - 560 012
>                                         Voice: (+91)-80-3562062
>                                         Email: john@hirp.iisc.ernet.in
>                                                chooracken@yahoo.com
> 
> On Thu, 9 Jan 2003, KANUMURI Sreeram wrote:
> 
> >
> > see www.vovida.org you can get the source code in C++
> >
> >
> > -Sreeram
> >
> > -----Original Message-----
> > From: Yuyong Zhang [mailto:yzhang@cedarpointcom.com]
> > Sent: Friday, January 03, 2003 11:22 PM
> > To: sip@ietf.org
> > Subject: [Sip] free SIP stack?
> >
> >
> > Hi there,
> >
> > Could anybody please recommend a good and free SIP stack?
> >
> > Regards,
> >
> > Yuyong
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> 
> --
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Jonathan Rosenberg'"
	 <jdrosen@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 10:50:17 -0000 
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Maybe Adam wrote the original slightly stronger than I would, but the NOTIFY method itself does carry semantics.

Those semantics are that it is carrying information defined as an event. Obviously you need to read the relevant headers and the body to find out what sort of event, but until you read the string "NOTIFY" at the start of the received PDU, you do not know you are looking for an event.

Obviously there are also normative requirements to understanding how it fits in a dialog, but I agree, that is part of the protocol specification rather than semantics of the method.

As such it is not carried unstructured information (INFO), or information defined as fulfilling instant messaging requirements (MESSAGE).

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 08 January 2003 22:55
> To: 'Jonathan Rosenberg'
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > > No, EVENTS doesn't define semantics.
> > 
> > Of course it does. There are lots of MAYS, MUSTS and SHOULDs 
> > in there. 
> > There is behavior dealing with subscription refreshes, rules 
> > on when to 
> > send NOTIFY, and so on and so on. How can you say that none 
> of those 
> > qualify as semantics?
> 
> 
> That's called a "protocol specification", not a specification of the
> semanticsd of the delivered content.
> 
> What does a NOTIFY mean? Does it mean that I have voice mail 
> and I should
> poll it? Does it mean that somebody has subscribed to my 
> presence and I
> should adjust my authorization policy? Does it mean that my 
> laundry is done
> and I should go pick it up? Does it mean that the IRS is 
> going to audit my
> books and I should flee for the border?  
> 
> You can't answer this question, in general, because those 
> semantics are NOT
> defined in RFC 3265. They're defined in event packages.
> 
> Similarly, we complain because the INFO spec doen't tell us 
> what an INFO
> means. Should we convert the body to ISUP? Should we execute 
> the operator
> now? Should the phone explode? The INFO spec should no more 
> answer this than
> does the Events spec. The semantic encoding is a separate problem.
> 
> Now, here's the difference. I don't think you're talking 
> about the presence
> of semantic information. I think you're talking about the absence of a
> mechanism for documenting the semantic information. Event 
> packages give us
> this mechanism for events. We don't have anything like that 
> for INFO. You're
> not talking semantics -- you're talking tools for exchanging 
> meta-semantics.
> 
> > 
> > > Event PACKAGES define the semantics of
> > > particular instances of the event methods.
> > 
> > Within a well defined framework. 3265 defines clear rules on 
> > what each 
> > package has to specify, and what it can and cannot due ontop 
> > of rfc3265.
> > 
> > 
> > > INFO also needs a blob-describing-and-encoding framework, 
> > so that the 
> > > various uses of INFO can, as appropriate, be effectively 
> documented 
> > > and negotiated.
> > 
> > Well, I'll try one more time, since I am clearly not 
> getting my point 
> > across.
> > 
> > RFC3265 defines semantics above and beyond RFC3261. All of those 
> > MAY/MUST/SHOULD I was talking about. If you had a choice between 
> > defining a new package, and defining your own sub/not 
> mechanisms, it 
> > would be a LOT more work to define your own sub/not mechanisms.
> 
> huh?
> 
> It would be much more work to independently reimplement the 
> functionality of
> INFO than it is to encode some information into MIME and ship 
> it via INFO,
> especially given that the main functionality of INFO is correlated
> resolution with an existing dialog (identical rendezvous).
> 
> > Not so for INFO. WHen deciding whether to specify an INFO 
> usage, or a 
> > new method, there is no additional work to do 
> > (specification-wise) when 
> > specifying a new method as opposed to a new INFO usage. To me, that 
> > means that INFO is devoid of semantics.
> 
> No, it just means that the process for defining a new INFO 
> usage is broken.
> 
> --
> Dean
> 
> _______________________________________________
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> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: Re: [Sip] INFO considered harmful
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The same is true for any message. Heck, you could probably encapsulate 
RTP in ACK or (intentionally rejected) INVITE, if ACK or INVITE doesn't 
get charged.

Christer Holmberg wrote:
> Hi,
> 
> 
>>Your example clearly falls into the "> 10" category as the poor web
>>server had to deal with every Real audio or video packet. Not that it
>>would surprise me, but I don't think anybody *here* has suggested
>>RTP-over-INFO so far.
> 
> 
> Isn't that part of the missuse problem we are talking about here - that
> someone actually COULD start using INFO for sending RTP (or whatever else
> that would creating frequent message sending)?
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 

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Hi,

> The same is true for any message. Heck, you could probably encapsulate
> RTP in ACK or (intentionally rejected) INVITE, if ACK or INVITE doesn't
> get charged.

That was exactly my point in the beginning of the thread, when I asked if we
need to define what kind of message bodies/information we are allowed to
include in INVITE and UPDATE.

Regards,

Christer Holmberg
Ericsson Finland


>
>
> Christer Holmberg wrote:
> > Hi,
> >
> >
> >>Your example clearly falls into the "> 10" category as the poor web
> >>server had to deal with every Real audio or video packet. Not that it
> >>would surprise me, but I don't think anybody *here* has suggested
> >>RTP-over-INFO so far.
> >
> >
> > Isn't that part of the missuse problem we are talking about here - that
> > someone actually COULD start using INFO for sending RTP (or whatever else
> > that would creating frequent message sending)?
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >

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--NextPart

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


	Title		: Carrying TCAP in SIP Messages (SIP-TCAP)
	Author(s)	: F. Miller et al.
	Filename	: draft-miller-sip-tcap-00.txt
	Pages		: 0
	Date		: 2003-1-8
	
SIP-TCAP is a mechanism by which an XML representation of TCAP
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Hi,
  When a UAS received a malformed Invite Request,
  it would send back a 400 Response after parsing
  the message. It would not (or may not be able to) 
  create a Transaction for the same.
  The Client on recieving this Response would respond
  with an ACK.

  The UAS would drop this ACK as an unhandled message,
  and would be reflected as an error on the same.

  The question is - Do we need a particular response code
  which would tell the client that an ACK for this should
  not be sent? 
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Date: Thu, 09 Jan 2003 10:24:49 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
CC: sip@ietf.org
Subject: Re: [Sip] INFO considered harmful
References: <9BF66EBF6BEFD942915B4D4D45C051F3A643C6@DYN-TX-EXCH-001.dynamicsoft.com> <3E15F0ED.7070002@cs.columbia.edu> <3E1D36BD.65F07CB6@lmf.ericsson.se> <3E1D6F7B.6060304@cs.columbia.edu> <3E1D7475.64386290@lmf.ericsson.se>
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The good guys don't need the restriction, the bad guys will ignore it. 
This all comes down to economic incentives: if it costs less to send a 
message of type foo than type bar (where 'costs less' may mean that the 
alternative has infinite cost, since the firewall won't let it pass), 
users will find that interesting, whether we proscribe it or not.

We don't worry about what you can attach to an email message, even 
though this also consumes precious server resources. (We obviously 
should and do worry what you do to the shared network resources.) If a 
proxy wants to reject all INFO messager or rate-limit a particular 
sender or restrict the size or content of messages, that's its privilege.

Christer Holmberg wrote:
> Hi,
> 
> 
>>The same is true for any message. Heck, you could probably encapsulate
>>RTP in ACK or (intentionally rejected) INVITE, if ACK or INVITE doesn't
>>get charged.
> 
> 
> That was exactly my point in the beginning of the thread, when I asked if we
> need to define what kind of message bodies/information we are allowed to
> include in INVITE and UPDATE.
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 

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Hi,

> The good guys don't need the restriction, the bad guys will ignore it.
> This all comes down to economic incentives: if it costs less to send a
> message of type foo than type bar (where 'costs less' may mean that the
> alternative has infinite cost, since the firewall won't let it pass),
> users will find that interesting, whether we proscribe it or not.
>
> We don't worry about what you can attach to an email message, even
> though this also consumes precious server resources. (We obviously
> should and do worry what you do to the shared network resources.) If a
> proxy wants to reject all INFO messager or rate-limit a particular
> sender or restrict the size or content of messages, that's its privilege.

If that is the solution, I see no reason for this whole thread :) There is no idea
to put any restrictions on INFO either, because the bad guys will ignore it
anyway...

Regards,

Christer Holmberg
Ericsson Finland


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To: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>,
        Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 10:15:53 -0600 
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> If that is the solution, I see no reason for this whole 
> thread :) There is no idea to put any restrictions on INFO
> either, because the bad guys will ignore it anyway...

Yes, people will do it. Unless we take up arms and start a massive
intelligence effort, there's nothing we can do about that simple
fact.

The overarching question is: should the IETF lend these
noninteroperable, potentially network-crippling approaches
credibility? Allison? Does the IESG generally condone that
sort of thing?

/a
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From: "Roberto Tealdi" <roby.tea@tin.it>
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Subject: [SIP] Authentication based on SIP request
Date: Thu, 9 Jan 2003 18:57:34 +0100
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Hi to All

I'm new in SIP and I have a question:

In draft "draft-ietf-sipping-aaa-req-01" is specified that the Home AAA =
server must be able to authenticate a user based on any SIP Request =
(except CANCEL): this means that ANY SIP request message (from client to =
sip proxy) must contains a user's credentials, for example an =
authentication token?=20

Thanks a lot.
Roby.


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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charset=3Diso-8859-1">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi to All</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I'm new in SIP and I have a =
question:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>In draft =
"draft-ietf-sipping-aaa-req-01" is=20
specified that the Home AAA server must be able to authenticate a user =
based on=20
any SIP Request (except CANCEL): this means&nbsp;that&nbsp;ANY SIP=20
request&nbsp;message (from client to sip proxy) must contains a user's=20
credentials,&nbsp;for example an authentication token? </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks a lot.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Roby.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0015_01C2B810.F84A94A0--


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From mailnull@www1.ietf.org  Thu Jan  9 14:34:56 2003
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Date: Thu, 09 Jan 2003 13:33:34 -0600
From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: radius (was Re: [Sip] resource reservation
In-reply-to: <5.2.0.9.0.20030109103823.02cfee60@mailhost.fokus.gmd.de>
To: "'Jiri Kuthan'" <jiri.kuthan@fokus.fraunhofer.de>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Jiri Kuthan'" <jiri@iptel.org>
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>I would really 
>like to see it as a WG item.

Yes, it would be a good idea to start with
>draft-schulzrinne-sipping-radius-accounting-00.txt> and make it a WG
item. This is very useful.

Thanks, Henry

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On 
> Behalf Of Jiri Kuthan
> Sent: Thursday, January 09, 2003 3:41 AM
> To: Henning Schulzrinne; Jiri Kuthan
> Cc: sip@ietf.org
> Subject: radius (was Re: [Sip] resource reservation
> 
> 
> Actually, we have implemented it and it is fine. I'm just 
> concerned about lack of a standard -- that makes interaction
> of accounting apps with SIP servers difficult. I would really 
> like to see it as a WG item.
> 
> -Jiri
> 
> At 02:58 AM 1/9/2003, Henning Schulzrinne wrote:
> >There is a long-expired document on this topic; 
> >draft-schulzrinne-sipping-radius-accounting-00.txt
> >
> >You can find it in the musty corners of 
> >http://www.cs.columbia.edu/sip/drafts/
> >
> >By reading it, you will lose all your political correctness points.
> >
> >Jiri Kuthan wrote:
> >>At 04:08 PM 1/8/2003, Henry Sinnreich wrote:
> >>
> >>>>3) The Go interface is COPS-PR, not Diameter.
> >>>
> >>>yes, and RADIUS in wireline systems. Also not DIAMETER.
> >>
> >>Let my add my two cents: quite many our users are 
> interesterd in use 
> >>of RADIUS for SIP accounting, and there is zero interest in 
> diameter. 
> >>The reason is there is a widely deployed RADIUS base. To operators, 
> >>diameter benefits are apparently less compelling than the upgrade 
> >>inconvenience. I'm sort of surprised, that the WG has ignored this 
> >>situation and has not produced any document on use of 
> RADIUS for SIP.
> >>-Jiri  
> >>_______________________________________________
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP Protocol
> >>Use sip-implementors@cs.columbia.edu for questions on current sip
> >>Use sipping@ietf.org for new developments on the application of sip
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> >sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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



From mailnull@www1.ietf.org  Thu Jan  9 14:36:13 2003
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Subject: RE: [Sip] INFO considered harmful
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Funny, none of the existing uses of INFO has melted the network.

Counterexamples?

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Thursday, January 09, 2003 11:16 AM
> To: 'Christer Holmberg'; Henning Schulzrinne
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > If that is the solution, I see no reason for this whole 
> > thread :) There is no idea to put any restrictions on INFO
> > either, because the bad guys will ignore it anyway...
> 
> Yes, people will do it. Unless we take up arms and start a massive
> intelligence effort, there's nothing we can do about that simple
> fact.
> 
> The overarching question is: should the IETF lend these
> noninteroperable, potentially network-crippling approaches
> credibility? Allison? Does the IESG generally condone that
> sort of thing?
> 
> /a
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  9 15:48:14 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Eric Burger'" <eburger@snowshore.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 14:47:48 -0600 
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I said "potentially network-crippling," for the exact reason
that I don't have examples of this happening specifically
with SIP. What I do have, as I've cited before, is examples
of the whacked-out kind of things that people will do and
have done to appliation-level protocols. The potential exists,
and there is a demonstrated will to exploit it.

I'll repeat the example here.

I'm recalling the rise in popularity of Real Media Player and
its ability to set up an audio stream through an HTTP proxy.
In particular, I'm recalling conversations shortly thereafter
with some friends over in MIS about how they just couldn't add
capacity to the HTTP proxy pool fast enough to keep up with
demand.

Access to external web documents sometimes creeped up to the
order of minutes before everything was finally stabilized.
It would be difficult to argue that such a mode of operation
represents anything other than a crippled network.

How long do you guess it will be before Real player
(or a similar product) includes a "SIP Proxy:" field right
next to its "HTTP Proxy:" field for firewall circumvention?

And, lest we lose sight of why I'm saying this in the
first place (this conversation has shown signs of extreme
ADHD): Yes, people *will* abuse the protocol. There's
nothing we can do about that simple fact. It is, however,
irresponsible for the IETF to lend these noninteroperable,
potentially network-crippling approaches credibility.

/a

> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Thursday, January 09, 2003 13:37
> To: Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> Funny, none of the existing uses of INFO has melted the network.
> 
> Counterexamples?
> 
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > Sent: Thursday, January 09, 2003 11:16 AM
> > To: 'Christer Holmberg'; Henning Schulzrinne
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > > If that is the solution, I see no reason for this whole 
> > > thread :) There is no idea to put any restrictions on INFO
> > > either, because the bad guys will ignore it anyway...
> > 
> > Yes, people will do it. Unless we take up arms and start a massive
> > intelligence effort, there's nothing we can do about that simple
> > fact.
> > 
> > The overarching question is: should the IETF lend these
> > noninteroperable, potentially network-crippling approaches
> > credibility? Allison? Does the IESG generally condone that
> > sort of thing?
> > 
> > /a
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  9 16:34:34 2003
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Subject: RE: [Sip] INFO considered harmful
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I guess the rat hole is "noninteroperable, potentially network-crippling approaches credibility."  A number of drafts have proposed interoperable, non-network-crippling payloads for INFO.

Rewind 20 years.  Replace INFO with UDP.  The arguments are IDENTICAL.  Unless you want to keep X.25 alive, who today would live without UDP?  RTP would be bumming :-)

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Thursday, January 09, 2003 3:48 PM
> To: Eric Burger; Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> I said "potentially network-crippling," for the exact reason
> that I don't have examples of this happening specifically
> with SIP. What I do have, as I've cited before, is examples
> of the whacked-out kind of things that people will do and
> have done to appliation-level protocols. The potential exists,
> and there is a demonstrated will to exploit it.
> 
> I'll repeat the example here.
> 
> I'm recalling the rise in popularity of Real Media Player and
> its ability to set up an audio stream through an HTTP proxy.
> In particular, I'm recalling conversations shortly thereafter
> with some friends over in MIS about how they just couldn't add
> capacity to the HTTP proxy pool fast enough to keep up with
> demand.
> 
> Access to external web documents sometimes creeped up to the
> order of minutes before everything was finally stabilized.
> It would be difficult to argue that such a mode of operation
> represents anything other than a crippled network.
> 
> How long do you guess it will be before Real player
> (or a similar product) includes a "SIP Proxy:" field right
> next to its "HTTP Proxy:" field for firewall circumvention?
> 
> And, lest we lose sight of why I'm saying this in the
> first place (this conversation has shown signs of extreme
> ADHD): Yes, people *will* abuse the protocol. There's
> nothing we can do about that simple fact. It is, however,
> irresponsible for the IETF to lend these noninteroperable,
> potentially network-crippling approaches credibility.
> 
> /a
> 
> > -----Original Message-----
> > From: Eric Burger [mailto:eburger@snowshore.com]
> > Sent: Thursday, January 09, 2003 13:37
> > To: Adam Roach
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > Funny, none of the existing uses of INFO has melted the network.
> > 
> > Counterexamples?
> > 
> > > -----Original Message-----
> > > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > > Sent: Thursday, January 09, 2003 11:16 AM
> > > To: 'Christer Holmberg'; Henning Schulzrinne
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > > 
> > > 
> > > > If that is the solution, I see no reason for this whole 
> > > > thread :) There is no idea to put any restrictions on INFO
> > > > either, because the bad guys will ignore it anyway...
> > > 
> > > Yes, people will do it. Unless we take up arms and start a massive
> > > intelligence effort, there's nothing we can do about that simple
> > > fact.
> > > 
> > > The overarching question is: should the IETF lend these
> > > noninteroperable, potentially network-crippling approaches
> > > credibility? Allison? Does the IESG generally condone that
> > > sort of thing?
> > > 
> > > /a
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > > 
> > 
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  9 17:16:48 2003
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Eric Burger'" <eburger@snowshore.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 16:17:44 -0600 
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The #1 issue is that the INFO RFC, today, just says,
"here's a method; do with it what you will. Everything's
allowed."

On re-reading my arguments, all I'm doing is reinforcing
this point, which I beleive is relatively uncontentious.
(Relative, that is, to...)

The more subtle point is whether to turn the free-for-all
into an organized effort that involves IANA registration,
or whether to completely close the doors.

That leads us to...

> A number of 
> drafts have proposed interoperable, non-network-crippling 
> payloads for INFO.

Okay, there's your draft, which is apparently just an informational
publication, instead of any proposal for a standard.

Beyond, that, RFC 2967 is mentioned only in the following
internet drafts:

- draft-ietf-sip-serverfeatures-05
- draft-gurbani-sin-02
  Only mention INFO as an example method

- draft-ietf-sip-message-01
  Only mentions INFO in the context of "MESSAGE has this in
  common with INFO".

- draft-ietf-sipping-sigcomp-sip-dictionary-05
  Mentions INFO because, well, it's a dictionary that includes
  all the SIP methods

- draft-garcia-sip-associated-uri-04
- draft-henrikson-sip-original-dialog-id-02
- draft-ietf-sip-join-00
- draft-ietf-sip-replaces-02
- draft-ietf-sip-callerprefs-07
  Only mention INFO in an attempt to be complete when
  expanding table 2 in RFC 3261

- draft-mahy-sipping-signaled-digits-01
  Uses INFO for RTP transport, but questions this practice
  in the open issues section

By my count, that's exactly one draft that's currently proposing
to use INFO for something nonproprietary, and even that draft isn't
so certain that INFO is the right approach.

Did I miss something?

Where are the applications that must take advantage of INFO to
work? INFO has been an RFC for over two years now, and was first
published as an ID almost four years ago. Has that been
insufficient time to come up with the killer app that makes keeping
INFO around worth the effort of updating it instead of scrapping it?

/a

> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Thursday, January 09, 2003 15:34
> To: Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> I guess the rat hole is "noninteroperable, potentially 
> network-crippling approaches credibility."  A number of 
> drafts have proposed interoperable, non-network-crippling 
> payloads for INFO.
> 
> Rewind 20 years.  Replace INFO with UDP.  The arguments are 
> IDENTICAL.  Unless you want to keep X.25 alive, who today 
> would live without UDP?  RTP would be bumming :-)
> 
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > Sent: Thursday, January 09, 2003 3:48 PM
> > To: Eric Burger; Adam Roach
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > I said "potentially network-crippling," for the exact reason
> > that I don't have examples of this happening specifically
> > with SIP. What I do have, as I've cited before, is examples
> > of the whacked-out kind of things that people will do and
> > have done to appliation-level protocols. The potential exists,
> > and there is a demonstrated will to exploit it.
> > 
> > I'll repeat the example here.
> > 
> > I'm recalling the rise in popularity of Real Media Player and
> > its ability to set up an audio stream through an HTTP proxy.
> > In particular, I'm recalling conversations shortly thereafter
> > with some friends over in MIS about how they just couldn't add
> > capacity to the HTTP proxy pool fast enough to keep up with
> > demand.
> > 
> > Access to external web documents sometimes creeped up to the
> > order of minutes before everything was finally stabilized.
> > It would be difficult to argue that such a mode of operation
> > represents anything other than a crippled network.
> > 
> > How long do you guess it will be before Real player
> > (or a similar product) includes a "SIP Proxy:" field right
> > next to its "HTTP Proxy:" field for firewall circumvention?
> > 
> > And, lest we lose sight of why I'm saying this in the
> > first place (this conversation has shown signs of extreme
> > ADHD): Yes, people *will* abuse the protocol. There's
> > nothing we can do about that simple fact. It is, however,
> > irresponsible for the IETF to lend these noninteroperable,
> > potentially network-crippling approaches credibility.
> > 
> > /a
> > 
> > > -----Original Message-----
> > > From: Eric Burger [mailto:eburger@snowshore.com]
> > > Sent: Thursday, January 09, 2003 13:37
> > > To: Adam Roach
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > > 
> > > 
> > > Funny, none of the existing uses of INFO has melted the network.
> > > 
> > > Counterexamples?
> > > 
> > > > -----Original Message-----
> > > > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > > > Sent: Thursday, January 09, 2003 11:16 AM
> > > > To: 'Christer Holmberg'; Henning Schulzrinne
> > > > Cc: sip@ietf.org
> > > > Subject: RE: [Sip] INFO considered harmful
> > > > 
> > > > 
> > > > > If that is the solution, I see no reason for this whole 
> > > > > thread :) There is no idea to put any restrictions on INFO
> > > > > either, because the bad guys will ignore it anyway...
> > > > 
> > > > Yes, people will do it. Unless we take up arms and 
> start a massive
> > > > intelligence effort, there's nothing we can do about that simple
> > > > fact.
> > > > 
> > > > The overarching question is: should the IETF lend these
> > > > noninteroperable, potentially network-crippling approaches
> > > > credibility? Allison? Does the IESG generally condone that
> > > > sort of thing?
> > > > 
> > > > /a
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > > 
> > > 
> > 
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Eric Burger'" <eburger@snowshore.com>,
        Adam Roach
	 <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 18:04:57 -0500 
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Comparing the projected use of SIP INFO with UDP is very apt. However, we
already have a UDP, and we don't need SIP to become a more cumbersome
surrogate.

This argument has gone around and around to little profit. There is no
question that SIP can be used as a transport protocol, and we cannot prevent
implementers from doing that. Indeed, any protocol can be misapplied. The
real argument here is just about what the specifications should present as
the correct application of the protocol.

While you can argue, Eric, that SIP has lost sight of its original goals,
and therefore that it doesn't matter what ugliness we introduce now, I
disagree. There are two functions that SIP is intended to provide -
discovery and session management. Generic data delivery is not one of these
functions. Your arguments that SIP is bloated, or what have you, would
encourage us to reduce the scope of the standard, not to condone mistakes
that clearly conflict with SIP's mission.

I also do not agree that SIP needs any method that is designed for
experimentation. You are welcome to implement any experiments you would like
- but the standard does not need to detail any particular way of
experimenting with SIP in order to enable you to do so. If we did believe
this was necessary, we could create an EXPERIMENTAL method - clearly, we
chose not to give INFO that name for some reason. We do not need to
standardize a method for experimentation because standards exist to foster
interoperability - experiments will necessarily run between experimental
user agents that do not need a standard specification describing their
experimental behavior.

It is hard for me to understand why the unregulated use of INFO is being
defended so vehemently. A correction to INFO that regulated its usage, and
provided some guidelines for what can legitimately be standardized as uses
of the INFO method, would be a big improvement over what we have now, and it
would not conflict with any future appropriate use of the method. Is even
that being rejected here?

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Eric Burger [mailto:eburger@snowshore.com]
> Sent: Thursday, January 09, 2003 1:34 PM
> To: Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> I guess the rat hole is "noninteroperable, potentially 
> network-crippling approaches credibility."  A number of 
> drafts have proposed interoperable, non-network-crippling 
> payloads for INFO.
> 
> Rewind 20 years.  Replace INFO with UDP.  The arguments are 
> IDENTICAL.  Unless you want to keep X.25 alive, who today 
> would live without UDP?  RTP would be bumming :-)
> 
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > Sent: Thursday, January 09, 2003 3:48 PM
> > To: Eric Burger; Adam Roach
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] INFO considered harmful
> > 
> > 
> > I said "potentially network-crippling," for the exact reason
> > that I don't have examples of this happening specifically
> > with SIP. What I do have, as I've cited before, is examples
> > of the whacked-out kind of things that people will do and
> > have done to appliation-level protocols. The potential exists,
> > and there is a demonstrated will to exploit it.
> > 
> > I'll repeat the example here.
> > 
> > I'm recalling the rise in popularity of Real Media Player and
> > its ability to set up an audio stream through an HTTP proxy.
> > In particular, I'm recalling conversations shortly thereafter
> > with some friends over in MIS about how they just couldn't add
> > capacity to the HTTP proxy pool fast enough to keep up with
> > demand.
> > 
> > Access to external web documents sometimes creeped up to the
> > order of minutes before everything was finally stabilized.
> > It would be difficult to argue that such a mode of operation
> > represents anything other than a crippled network.
> > 
> > How long do you guess it will be before Real player
> > (or a similar product) includes a "SIP Proxy:" field right
> > next to its "HTTP Proxy:" field for firewall circumvention?
> > 
> > And, lest we lose sight of why I'm saying this in the
> > first place (this conversation has shown signs of extreme
> > ADHD): Yes, people *will* abuse the protocol. There's
> > nothing we can do about that simple fact. It is, however,
> > irresponsible for the IETF to lend these noninteroperable,
> > potentially network-crippling approaches credibility.
> > 
> > /a
> > 
> > > -----Original Message-----
> > > From: Eric Burger [mailto:eburger@snowshore.com]
> > > Sent: Thursday, January 09, 2003 13:37
> > > To: Adam Roach
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] INFO considered harmful
> > > 
> > > 
> > > Funny, none of the existing uses of INFO has melted the network.
> > > 
> > > Counterexamples?
> > > 
> > > > -----Original Message-----
> > > > From: Adam Roach [mailto:adam@dynamicsoft.com]
> > > > Sent: Thursday, January 09, 2003 11:16 AM
> > > > To: 'Christer Holmberg'; Henning Schulzrinne
> > > > Cc: sip@ietf.org
> > > > Subject: RE: [Sip] INFO considered harmful
> > > > 
> > > > 
> > > > > If that is the solution, I see no reason for this whole 
> > > > > thread :) There is no idea to put any restrictions on INFO
> > > > > either, because the bad guys will ignore it anyway...
> > > > 
> > > > Yes, people will do it. Unless we take up arms and 
> start a massive
> > > > intelligence effort, there's nothing we can do about that simple
> > > > fact.
> > > > 
> > > > The overarching question is: should the IETF lend these
> > > > noninteroperable, potentially network-crippling approaches
> > > > credibility? Allison? Does the IESG generally condone that
> > > > sort of thing?
> > > > 
> > > > /a
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the 
> > application of sip
> > > > 
> > > 
> > 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Thu Jan  9 19:13:36 2003
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	Thu, 9 Jan 2003 18:13:55 -0600
From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Eric Burger'" <eburger@snowshore.com>,
        "'Adam Roach'" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 18:13:49 -0600
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> Jon says:
> There are two 
> functions that SIP is intended to provide - discovery and 
> session management. Generic data delivery is not one of these 
> functions. 

Perhaps you overlook the "rendezvous" function, which is not quite the same
as "discovery". The utility of INFO is that it delivers data to the
endpoints of a specific dialog, while respecting the SIP-level anonymity,
security, and routing functionality of that dialog. Nothing BUT an in-dialog
SIP method has that property, as we discovered in the extended discssion of
binding HTTP URLS to SIP.


> It is hard for me to understand why the unregulated use of 
> INFO is being defended so vehemently. A correction to INFO 
> that regulated its usage, and provided some guidelines for 
> what can legitimately be standardized as uses of the INFO 
> method, would be a big improvement over what we have now, and 
> it would not conflict with any future appropriate use of the 
> method. Is even that being rejected here?


No, what's being rejected out-of-hand is the notion that INFO is horribly
dangerous and busily eating the internet like a giant space-amoeba,
requiring us immediately to allocate resources to expurgate it. I suspect
that the participants are simply enjoying the debate, much as my dog enjoys
chewing a piece of rawhide.

There is probably an undercurrent of debate on exactly what sorts of
constraints, guidances, and registrations should be applied, but that
discussion could probably be readily diverteed into a "send text"
instruction.

--
Dean

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



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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 18:13:49 -0600
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> The overarching question is: should the IETF lend these 
> noninteroperable, potentially network-crippling approaches 
> credibility? Allison? Does the IESG generally condone that 
> sort of thing?

My, that's seriously humorous rhetoric. You continued deluge of non
interoperable network crippling bytes has overwhelmed my mail server far
more quickly than the tens of unsolicited (and often unknown and
indecipherable) MIME objects that get delivered to my mailbox each day, and
I'm sure it does more harm to the network than the thousands of MIME objects
not (specified by HTTP) that I retrieve with HTYP each day . . .

We need to get a little perspective here, folks.

--
Dean

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To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
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Subject: RE: [Sip] INFO considered harmful
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Yep. Several university hacker groups tunneled IP over DNS back when
firewalls first reared their eugly heads in the early '90s. You know, it
doesn't specifically say "Thou shalt not tunnel IP over domain xfer
requests" in the DNS spec, and I'm sure if it had, that they would not have
done this.

--
Dean

> The same is true for any message. Heck, you could probably 
> encapsulate 
> RTP in ACK or (intentionally rejected) INVITE, if ACK or 
> INVITE doesn't 
> get charged.
> 
> Christer Holmberg wrote:
> > Hi,
> > 
> > 
> >>Your example clearly falls into the "> 10" category as the poor web 
> >>server had to deal with every Real audio or video packet. 
> Not that it 
> >>would surprise me, but I don't think anybody *here* has suggested 
> >>RTP-over-INFO so far.
> > 

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
Date: Thu, 9 Jan 2003 18:13:49 -0600
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> Where are the applications that must take advantage of INFO 
> to work? INFO has been an RFC for over two years now, and was 
> first published as an ID almost four years ago. Has that been 
> insufficient time to come up with the killer app that makes 
> keeping INFO around worth the effort of updating it instead 
> of scrapping it?

I suspect that they (the applications) are buried behind the chanted
rhetoric of the fanatic disciples of the anti-INFO cult, famed for their
monologous exposition and persistent derogatory exculpation of infidels
proposing the heretical practice of INFOlogy in the time-honored tradition
that theocracies throughout history have used to suppress dissent.

Or maybe the protocol has just gotten sufficiently convoluted that nobody is
really exercising it with new applications?

Or maybe they're using it and not telling you . . . Yes that's it, everybody
is using INFO and deliberately hiding it. It's deliberate, and they're doing
it just to spite you (Adam). In fact, if you check the phone on your desk
that you use every day, I'll bet you find that it's committing the heresy of
DTMF-INFO on every dialed call. There's no place to hide, Adam, INFO is
coming for you . . .

Seriously, is this thread pointless, or what?

--
Dean

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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Eric Burger'"
	 <eburger@snowshore.com>,
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Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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A small note below.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, January 09, 2003 4:14 PM
> To: 'Peterson, Jon'; 'Eric Burger'; 'Adam Roach'
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> > Jon says:
> > There are two 
> > functions that SIP is intended to provide - discovery and 
> > session management. Generic data delivery is not one of these 
> > functions. 
> 
> Perhaps you overlook the "rendezvous" function, which is not quite the
same
> as "discovery". The utility of INFO is that it delivers data to the
> endpoints of a specific dialog, while respecting the SIP-level anonymity,
> security, and routing functionality of that dialog. Nothing BUT an
in-dialog
> SIP method has that property, as we discovered in the extended discssion
of
> binding HTTP URLS to SIP.


By discovery I meant what one would generally consider to be the rendezvous
function - all of the registration and events mechanisms, URI resolution and
so on that goes into allowing user agents to find the right place and time
on the network to establish a dialog with their targets. SIP connects user
agents in a dialog, however, so that they can have a session. Sessions are
the way SIP delivers data between endpoints in a specific dialog. The
anonymity, security and routing of session data can indeed run afoul of
problems - but we don't solve those problems for, say, RTP by sending RTP
over INFO. MESSAGE has already been discussed in this thread, so I won't
rehash it (MESSAGE is not a dialog, etc) - suffice it to say that I think
the unregulated standardization of INFO gives SIP some capabilities that it
doesn't have otherwise, which don't serve either of SIP's core functions,
which confuse and complicate SIP's applicability. I agree that INFO is not a
space amoeba, but it is a superfluous bifurcation of SIP's direction that we
can prune without diminishing the protocol's core assets.

[snip]
> 
> --
> Dean
> 
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Date: Thu, 09 Jan 2003 21:04:14 -0500
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Before I get misunderstood: I don't suggest standardizing applications 
(of INFO or anything else) that shouldn't be. We don't have to 
legitimize dumb designs. On the other hand, I'm not aware of any 
(successfull) attempt to limit Content-Type to a specific, approved set.

Also, generally the IETF has worried about network load. Thus, mandating 
various rules for use of TCP or congestion-control protocols seems 
perfectly in line with experience. Prohibiting things for the benefit of 
CPU load has no precedent that I can think of.

Adam Roach wrote:
>>If that is the solution, I see no reason for this whole 
>>thread :) There is no idea to put any restrictions on INFO
>>either, because the bad guys will ignore it anyway...
> 
> 
> Yes, people will do it. Unless we take up arms and start a massive
> intelligence effort, there's nothing we can do about that simple
> fact.
> 
> The overarching question is: should the IETF lend these
> noninteroperable, potentially network-crippling approaches
> credibility? Allison? Does the IESG generally condone that
> sort of thing?
> 
> /a

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Eric Burger'" <eburger@snowshore.com>,
        "'Adam Roach'" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO considered harmful
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Jon said: 
> By discovery I meant what one would generally consider to be 
> the rendezvous function - all of the registration and events 
> mechanisms, URI resolution and so on that goes into allowing 
> user agents to find the right place and time on the network 
> to establish a dialog with their targets. SIP connects user 
> agents in a dialog, however, so that they can have a session. 
> Sessions are the way SIP delivers data between endpoints in a 
> specific dialog. The anonymity, security and routing of 
> session data can indeed run afoul of problems - but we don't 
> solve those problems for, say, RTP by sending RTP over INFO. 
> MESSAGE has already been discussed in this thread, so I won't 
> rehash it (MESSAGE is not a dialog, etc) - suffice it to say 
> that I think the unregulated standardization of INFO gives 
> SIP some capabilities that it doesn't have otherwise, which 
> don't serve either of SIP's core functions, which confuse and 
> complicate SIP's applicability. I agree that INFO is not a 
> space amoeba, but it is a superfluous bifurcation of SIP's 
> direction that we can prune without diminishing the 
> protocol's core assets.

We solve the rendezvous/discovery problems for events and message with SIP
routing, and this functionality of SIP is clearly applicable to
application-data such as ISUP processing in SIP-T (currently done via INFO
routing). I really don't understand what part of "the data carried in INFO
has the semantic of being associated with the application driving this
dialog, not the SIP state machine itself" that isn't clear. The SIP layer
fundamentally neither knows nor cares what that application is.

--
Dean

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From: "Jesske, R" <R.Jesske@telekom.de>
To: sip@ietf.org, fmiller@sentito.com
Cc: "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>,
        "Tolksdorf, Christian" <Christian.Tolksdorf@telekom.de>,
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Hi,
In a Mail regarding the issue: "Re: [Sip] INFO considered harmful" you pointed to a draft regarding a textual representation of SS7 in SIP INFO.
After a short view at your draft I'm confused now. You are describing for SS7 Blocking (BLO) messages a text description to transport BLO over a SIP network. Why?
The SS7 BLO/UBL Messages are only exchanged between two exchanges within the SS7 world to say the circuits are now blocked/unblocked. Within SIP there are no Circuits, why do you need such kind of messages within the SIP world. 

Best Regards

Roland

Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38a12
Section T38; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:    +49 6151 83-4577 
 <mailto:r.jesske@telekom.de>


Well, any illusion of reasonableness I may have garnered in the past is
about to be shattered.  I would refer folks to:

http://www.ietf.org/internet-drafts/draft-miller-sip-isup-annex-00.txt

This draft is about to receive a major -01 update that recodes the entire
syntax in XML.  In addition, we (I) have another draft that does something
very similar for ANSI TCAP that I'll be submitting in short order.  I guess
its only appropriate that I bring peoples attention to these drafts under
the current subject heading...

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Networks, Inc.
fmiller@sentito.com







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From: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'Peterson, Jon'" <jon.peterson@neustar.biz>,
        "'Eric Burger'"
	 <eburger@snowshore.com>,
        Adam Roach <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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Hi all,

Typically wanting to avoid these types of debate, I thought I  would just
respond to this one point in Jon's posting (I've deleted the rest of the
thread), which I think gets to the crux of the primary concern.

----beginning of thread snipped----

>It is hard for me to understand why the unregulated use of INFO is being
>defended so vehemently. A correction to INFO that regulated its usage, and
>provided some guidelines for what can legitimately be standardized as uses
>of the INFO method, would be a big improvement over what we have now, and
it
>would not conflict with any future appropriate use of the method. Is even
>that being rejected here?

>Jon Peterson
>NeuStar, Inc.

----rest of thread deleted-----


I think the initial reaction and what folks were originally most concerned
over was the deprecation of the use of INFO (for other than SIP-T), which
was the preferred approach identified in Jonathan's draft.  I would think
that most people would not object to some guidelines around the use of INFO,
although I would think many would be more in favor of a 1.5 versus the
"completeness" proposed by the second approach in the draft. Basically, I'm
suggesting providing a venue (i.e. as part of SIPPING WG charter, perhaps)
to review individual usages of INFO.  The reality is that if you want your
use of INFO to interwork with a 3rd party, some level of specification is
required and in the end some of the usages might indeed converge to a widely
deployed approach or perhaps put the spotlight on the need for a new method.


As others have suggested, deprecating INFO or being too stringent in
enforcing it's usage would likely move the policing of the protocol problem
elsewhere, perhaps where the guidelines are less stringent (e.g. the lack of
registration with IANA is only discouraged).  And, as others have said the
"bad" guys will likely to continue using it regardless (not caring that it
impacts their particular "seal of IETF approval"). 

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Mary Barnes'" <mbarnes@nortelnetworks.com>,
        "'Peterson, Jon'"
	 <jon.peterson@neustar.biz>,
        "'Eric Burger'" <eburger@snowshore.com>,
        Adam Roach <adam@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
Date: Fri, 10 Jan 2003 15:20:38 -0000
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Perhaps it would be appropriate to have the same review mechanism and restrictions on INFO usage as provided for P-headers and event definitions. 

If the sip changes draft is still active, maybe we need some text in there. 

There would additionally need to be some text in the sip guidelines document.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Mary Barnes [mailto:mbarnes@nortelnetworks.com]
> Sent: 10 January 2003 14:46
> To: 'Peterson, Jon'; 'Eric Burger'; Adam Roach
> Cc: sip@ietf.org
> Subject: RE: [Sip] INFO considered harmful
> 
> 
> Hi all,
> 
> Typically wanting to avoid these types of debate, I thought I 
>  would just
> respond to this one point in Jon's posting (I've deleted the 
> rest of the
> thread), which I think gets to the crux of the primary concern.
> 
> ----beginning of thread snipped----
> 
> >It is hard for me to understand why the unregulated use of 
> INFO is being
> >defended so vehemently. A correction to INFO that regulated 
> its usage, and
> >provided some guidelines for what can legitimately be 
> standardized as uses
> >of the INFO method, would be a big improvement over what we 
> have now, and
> it
> >would not conflict with any future appropriate use of the 
> method. Is even
> >that being rejected here?
> 
> >Jon Peterson
> >NeuStar, Inc.
> 
> ----rest of thread deleted-----
> 
> 
> I think the initial reaction and what folks were originally 
> most concerned
> over was the deprecation of the use of INFO (for other than 
> SIP-T), which
> was the preferred approach identified in Jonathan's draft.  I 
> would think
> that most people would not object to some guidelines around 
> the use of INFO,
> although I would think many would be more in favor of a 1.5 versus the
> "completeness" proposed by the second approach in the draft. 
> Basically, I'm
> suggesting providing a venue (i.e. as part of SIPPING WG 
> charter, perhaps)
> to review individual usages of INFO.  The reality is that if 
> you want your
> use of INFO to interwork with a 3rd party, some level of 
> specification is
> required and in the end some of the usages might indeed 
> converge to a widely
> deployed approach or perhaps put the spotlight on the need 
> for a new method.
> 
> 
> As others have suggested, deprecating INFO or being too stringent in
> enforcing it's usage would likely move the policing of the 
> protocol problem
> elsewhere, perhaps where the guidelines are less stringent 
> (e.g. the lack of
> registration with IANA is only discouraged).  And, as others 
> have said the
> "bad" guys will likely to continue using it regardless (not 
> caring that it
> impacts their particular "seal of IETF approval"). 
> 
> Regards,
> Mary H. Barnes
> mbarnes@nortelnetworks.com
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sip@ietf.org, sip-implementors@cs.columbia.edu
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Subject: [Sip] SIPIT registration closing soon
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If you are planning to attend SIPIT 12 but have not yet registered,
please do so soon. Registration closes January 20th.

See http://www.hotsip.com/sipit12 for more information.

RjS



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To: "Jesske, R" <R.Jesske@telekom.de>
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----- Original Message -----
From: "Jesske, R" <R.Jesske@telekom.de>
To: <sip@ietf.org>; <fmiller@sentito.com>
Cc: "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>; "Tolksdorf, Christian"
<Christian.Tolksdorf@telekom.de>; "Alexeitsev, D" <D.Alexeitsev@telekom.de>;
<Robert.Benz@telekom.de>; "Pinker, Gerold" <Gerold.Pinker@telekom.de>
Sent: Friday, January 10, 2003 7:41 AM
Subject: Question draft-miller-sip-isup-annex-00.txt


> In a Mail regarding the issue: "Re: [Sip] INFO considered harmful" you
pointed to a draft regarding a textual representation of SS7 in SIP INFO.
> After a short view at your draft I'm confused now. You are describing for
SS7 Blocking (BLO) messages a text description to transport BLO over a SIP
network. Why?
> The SS7 BLO/UBL Messages are only exchanged between two exchanges within
the SS7 world to say the circuits are now blocked/unblocked. Within SIP
there are no Circuits, why do you need such kind of messages within the SIP
world.
>


We're seeing an architectural evolution whereby TDM access lines like T's in
the U.S. are are terminated at the access point in favor of SIP controlled
transport.  What this does is inject SIP "islands" in an existing TDM access
delivery network.  The key to this work (the ISUP and recently release TCAP
drafts) is to prevent those "islands" from having to deal with any kind of
SS7 binary protocol (read pay for SS7 stacks) inside the SIP "island".  Its
strictly a cost argument.  The idea is that existing SS7 elements can still
be used to provision channels but the SIP "islands" don't have to pay for an
SS7 (both $$ and point codes, whose name space is more congested the IPV4).

If you have some specific implementation level questions, I suggest we take
the discussion off the IETF board...

FM



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        Adam Roach
	 <adam@dynamicsoft.com>,
        "'Eric Burger'" <eburger@snowshore.com>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO considered harmful
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> It's deliberate, and they're doing it just to spite you (Adam).
> In fact, if you check the phone on your desk that you use every
> day, I'll bet you find that it's committing the heresy of DTMF-INFO
> on every dialed call.

Just checked. AVT tones payload, actually.

/a
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To: "Vijay Gaur" <vgaur@aastra.com>, <sip@ietf.org>
References: <F924CEFBBF62D611A0C600D0B76ED0371635CD@cvxmail.ana.aastra.com>
Subject: Re: [Sip] Handling for ACK for 400 Response
Date: Fri, 10 Jan 2003 17:41:41 -0500
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inline
----- Original Message -----
From: "Vijay Gaur" <vgaur@aastra.com>


> Hi,
>   When a UAS received a malformed Invite Request,
>   it would send back a 400 Response after parsing
>   the message. It would not (or may not be able to)
>   create a Transaction for the same.
>   The Client on recieving this Response would respond
>   with an ACK.
>
>   The UAS would drop this ACK as an unhandled message,
>   and would be reflected as an error on the same.

The ACK is just dropped, there is no error.

>
>   The question is - Do we need a particular response code
>   which would tell the client that an ACK for this should
>   not be sent?
>
I don't think that is necessary. The ACK is really harmless. The UAS can
respond statelessly (as described in section 8.2.7) for the malformed
request.

As far as the unhandled ACK, the UAS could form the to-tag of the response
such that the UAS could recognize it in the ACK and just ignore the ACK.

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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Can INVITE overlap SUBSCRIBE
Date: Fri, 10 Jan 2003 19:59:12 -0500
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> If a dialog is already established with a 
> SUBSCRIBE, that SUBSCRIBE is being refreshed 
> (SUBSCRIBE in progress). Can an INVITE, within 
> the same dialog, be sent while the SUBSCRIBE 
> refresh is in progress?

Yes.  However the requests might be received
out of order.  Thus it might lead to a 500 
response for the subscribe.

However caution should be observed when 
overlapping requests towards devices that
authenticate each request with a new nonce
and only remember the last nonce.  Such a
situation can lead to a continual challenge
situation where UAC appears to keep sending 
non-latest nonce.
 
> Another question: Can SUBSCRIBEs overlap in
>  the following 2 scenarios?
> 
> Scenario 1:
> 
>    (1) SUBSCRIBE event:presence
> ---------------------------->
> 
>    (2) SUBSCRIBE event:presence
> ---------------------------->
> 
> (1) and (2) are requests within the same dialog 
> and are the same association with the dialog.
> Can (2) be sent before (1) completes. This is 
> like INVITEs overlapping.

Yes, assuming the dialog has already been establish.  
However the requests might be received out of order.  
Thus it might lead to a 500 response for 1.

> Scenario 2:
> 
>    (3) SUBSCRIBE event:presence
> ---------------------------->
> 
>    (4) SUBSCRIBE event:winfo
> ---------------------------->
> 
> (3) and (4) are requests within the same dialog 
> but are in different associations with the dialog.
> Can (4) be sent before (3) completes. This is like 
> INVITE and SUBSCRIBE overlapping, I think.

I'm not positive about what you mean by different 
associations.  Thus I'll assume you mean different 
event types.

Assuming the dialog has already been establish,
yes it is legal to send.  However the requests might 
be received out of order.  Thus it might lead to a 
500 response for 3.

> A similar example of this is 2 SUBSCRIBEs with the 
> same event package but different event ids.

Yes it is legal to send.  However the requests might 
be received out of order.  Thus it might lead to a 
500 response for 3.

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Subject: RE: [Sip] Can INVITE overlap SUBSCRIBE
Date: Mon, 13 Jan 2003 10:50:01 +0200
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Thread-Topic: [Sip] Can INVITE overlap SUBSCRIBE
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To avoid all this confusion, would it be a good idea to forbid overlapping of target refresh requests within a dialog? i.e. requests that create dialogs are not allowed to overlap. In this way, we avoid the 500 response and the confusion of the race condition of the responses as well that might carry different remote targets.

Regards,
Hisham

> -----Original Message-----
> From: ext Brett Tate [mailto:brett@broadsoft.com]
> Sent: Saturday, January 11, 2003 2:59 AM
> To: sip@ietf.org
> Subject: RE: [Sip] Can INVITE overlap SUBSCRIBE
> 
> 
> > If a dialog is already established with a 
> > SUBSCRIBE, that SUBSCRIBE is being refreshed 
> > (SUBSCRIBE in progress). Can an INVITE, within 
> > the same dialog, be sent while the SUBSCRIBE 
> > refresh is in progress?
> 
> Yes.  However the requests might be received
> out of order.  Thus it might lead to a 500 
> response for the subscribe.
> 
> However caution should be observed when 
> overlapping requests towards devices that
> authenticate each request with a new nonce
> and only remember the last nonce.  Such a
> situation can lead to a continual challenge
> situation where UAC appears to keep sending 
> non-latest nonce.
>  
> > Another question: Can SUBSCRIBEs overlap in
> >  the following 2 scenarios?
> > 
> > Scenario 1:
> > 
> >    (1) SUBSCRIBE event:presence
> > ---------------------------->
> > 
> >    (2) SUBSCRIBE event:presence
> > ---------------------------->
> > 
> > (1) and (2) are requests within the same dialog 
> > and are the same association with the dialog.
> > Can (2) be sent before (1) completes. This is 
> > like INVITEs overlapping.
> 
> Yes, assuming the dialog has already been establish.  
> However the requests might be received out of order.  
> Thus it might lead to a 500 response for 1.
> 
> > Scenario 2:
> > 
> >    (3) SUBSCRIBE event:presence
> > ---------------------------->
> > 
> >    (4) SUBSCRIBE event:winfo
> > ---------------------------->
> > 
> > (3) and (4) are requests within the same dialog 
> > but are in different associations with the dialog.
> > Can (4) be sent before (3) completes. This is like 
> > INVITE and SUBSCRIBE overlapping, I think.
> 
> I'm not positive about what you mean by different 
> associations.  Thus I'll assume you mean different 
> event types.
> 
> Assuming the dialog has already been establish,
> yes it is legal to send.  However the requests might 
> be received out of order.  Thus it might lead to a 
> 500 response for 3.
> 
> > A similar example of this is 2 SUBSCRIBEs with the 
> > same event package but different event ids.
> 
> Yes it is legal to send.  However the requests might 
> be received out of order.  Thus it might lead to a 
> 500 response for 3.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Can INVITE overlap SUBSCRIBE
Date: Mon, 13 Jan 2003 11:59:02 -0500
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> To avoid all this confusion, would it be a 
> good idea to forbid overlapping of target 
> refresh requests within a dialog? 
> i.e. requests that create dialogs are not 
> allowed to overlap. In this way, we avoid 
> the 500 response and the confusion of the 
> race condition of the responses as well that 
> might carry different remote targets.

RFC 3261 section 12.2.2 already defines what
must happen when requests are received out
of order; a 500 response MUST be sent.  
Thus retargeting requests from the same
direction are already accommodated.

There are numerous race conditions associated 
with devices using responses to re-target.
Device Y should not be confident that a
re-target response will be processed correctly
by device X when device Y is currently awaiting 
a response for request containing the Contact of
prior target.  Using a retarget sequence number 
or Date header potentially could help solve the 
problem; however it is likely not worth the
extra specification to define it.

Some issues with dialog sharing and retargeting
were captured within the following:

http://www.ietf.org/mail-archive/working-groups/sip/current/msg06767.html


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From mailnull@www1.ietf.org  Mon Jan 13 21:21:53 2003
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From: "Dean Willis" <dean.willis@softarmor.com>
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Date: Mon, 13 Jan 2003 20:21:45 -0600
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Subject: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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It only seemed appropriate to follow up my previous assumably eloquent but
frequently misspelled exhortations by following my own advice and "sending
text".

The resulting draft has been submitted to the archive and should be
announced in due course. In the interim, you may wish to retrieve it from:

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.html

or

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.txt

Or, if you just LIKE looking at XML source or wish to fix some of my
probable glaring errors,

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.xml


I'd like to caveat that this is a rough first draft -- I just sat down and
banged it out this afternoon, and only Adam Roach has seen any of it so far,
and that was in small bits.

thanks

--
Dean

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From: "Frank W. Miller" <fmiller@sentito.com>
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: <sip@ietf.org>
References: <003301c2bb73$b6528650$fe0c0c42@txdwillis>
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Tue, 14 Jan 2003 09:44:50 -0500
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Looks like a good compromise.

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com
----- Original Message -----
From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Sent: Monday, January 13, 2003 9:21 PM
Subject: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00


>
> It only seemed appropriate to follow up my previous assumably eloquent but
> frequently misspelled exhortations by following my own advice and "sending
> text".
>
> The resulting draft has been submitted to the archive and should be
> announced in due course. In the interim, you may wish to retrieve it from:
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.html
>
> or
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.txt
>
> Or, if you just LIKE looking at XML source or wish to fix some of my
> probable glaring errors,
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.xml
>
>
> I'd like to caveat that this is a rough first draft -- I just sat down and
> banged it out this afternoon, and only Adam Roach has seen any of it so
far,
> and that was in small bits.
>
> thanks
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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From: Idnani Ajaykumar-AIDNANI1 <Ajaykumar.Idnani@motorola.com>
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Hi All,
 
I have a very simple question about the operation of Redirect Servers. Me
and colleague out here can not seem to convince each other on this, so I
thought I will email all the SIP experts here :-).
 
When a Redirect Server gets a request, it is supposed to return back the
Contacts for the called party. Where does it pick the address of the called
party from. Is it the To address or the Request-URI? The assumption here is
that the Redirect Server is co-located with the Registrar, and has access to
the database of the Registrar. When the Redirect Server is searching within
this database, what does it use as the key for searching - To Address or the
Request-URI.
 
Appreciate your help.
 
Thanks and Regards
- Ajay

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>I have a very simple question about the operation of
Redirect Servers. Me and colleague out here can not seem to convince =
each other
on this, so I thought I will email all the SIP experts <span =
class=3DGramE>here </span></span></font><font
face=3DWingdings><span =
style=3D'font-family:Wingdings;mso-ascii-font-family:Arial;
mso-hansi-font-family:Arial;mso-bidi-font-family:Arial;mso-char-type:sym=
bol;
mso-symbol-font-family:Wingdings;mso-no-proof:yes'><span =
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symbol;mso-symbol-font-family:Wingdings'>J</span></span></font><font
face=3DArial><span =
style=3D'font-family:Arial'>.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>When a Redirect Server gets a request, it is =
supposed to
return back the Contacts for the called party. Where does it pick the =
address
of the called party <span class=3DGramE>from.</span> Is it the <span =
class=3DGramE>To</span>
address or the Request-URI? The assumption here is that the Redirect =
Server is co-located
with the Registrar, and has access to the database of the Registrar. =
When the
Redirect Server is searching within this database, what does it use as =
the key
for searching - To Address or the Request-<span =
class=3DGramE>URI.</span><o:p></o:p></span></font></p>

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

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

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

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

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

</div>

</body>

</html>

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



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Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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Hi,

To negotiate what INFO can be used for between nodes is good thing, and at least
on a theoretical level it solves one issue.

But, is the biggest problem we have today end-nodes receiving INFOs with data
they don't know to handle? And, how is this going to solve the problem of having
people using INFO for whatever purpose? All they have to do is to define a
package for their who-knows-what data, right? Or, does defining a package allow
one to use INFO for whatever he/she wants?

Or, do we accept Henning's rule about the "bad guys", and do NOT try to define
any rules on exactly what kind of data INFO should be used for in the first
place?

I am not saying that this draft should solve anything, but I would like to know
exactly which issues regarding INFO we are going to try solving :)

Regards,

Christer Holmberg
Ericsson Finland





Dean Willis wrote:

> It only seemed appropriate to follow up my previous assumably eloquent but
> frequently misspelled exhortations by following my own advice and "sending
> text".
>
> The resulting draft has been submitted to the archive and should be
> announced in due course. In the interim, you may wish to retrieve it from:
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.html
>
> or
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.txt
>
> Or, if you just LIKE looking at XML source or wish to fix some of my
> probable glaring errors,
>
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.xml
>
> I'd like to caveat that this is a rough first draft -- I just sat down and
> banged it out this afternoon, and only Adam Roach has seen any of it so far,
> and that was in small bits.
>
> thanks
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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



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Subject: RE: [Sip] Question about Redirect Server
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To: Idnani Ajaykumar-AIDNANI1 <Ajaykumar.Idnani@motorola.com>, sip@ietf.org
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Hi,
As per the RFC-3261 on the first hop it is expected that the To and the
Request-URI are same.  In this case your query does not have any problem.
If some intermediate proxy has changed the request-URI then, the request-uri
being the latest should be used to search the registrar.  However it depends
on the design of the registry database also.
If the database caters only to one domain, then just the user part of the
request-uri is taken and the address can be resolved.
If the database caters to more than one domain, and if by chance the host
part of the request-uri is an IP then, the host part of the To along with
the user part of the request-uri is taken as the key.
Cheers,
Prasanna

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Idnani
Ajaykumar-AIDNANI1
Sent: Tuesday, January 14, 2003 8:23 PM
To: sip@ietf.org
Subject: [Sip] Question about Redirect Server


Hi All,

I have a very simple question about the operation of Redirect Servers. Me
and colleague out here can not seem to convince each other on this, so I
thought I will email all the SIP experts here J.

When a Redirect Server gets a request, it is supposed to return back the
Contacts for the called party. Where does it pick the address of the called
party from. Is it the To address or the Request-URI? The assumption here is
that the Redirect Server is co-located with the Registrar, and has access to
the database of the Registrar. When the Redirect Server is searching within
this database, what does it use as the key for searching - To Address or the
Request-URI.

Appreciate your help.

Thanks and Regards
- Ajay

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Tue, 14 Jan 2003 16:26:31 -0000
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In principle, the idea looks good to me.

I would note that the proposal invalidates existing usages of the INFO method, as presumably all usages of INFO carry the new header Info-Type and the dialog should presumably be created carrying the Allow-Info header. Assuming we allow this constraint, which is OK by me but some people who have already implemented it might shout, then it would seem better to document this by obsoleting the existing version of INFO (RFC 2976) and creating a new one; this would allow the method specification to be aligned with RFC 3261 rather than RFC 2543 (and also allow its interaction with all the other new RFCs to be specified - although I assume this interaction is fairly minimal - the only impact I can see is the inclusion of the Allow-Events header in INFO).

From this proposal there is possibly a minor impact on RFC 3398. This document requires a gateway to support the INFO method. Presumably it would be appropropriate that the gateway now advertises the type of INFO usage that it supports (i.e. ISUP encapsulation), and this would be a requirement that should be in RFC 3398 rather than elsewhere.

Strangely enough, we seem to have absolutely no examples of the use of INFO to carry mid call information, so no changes are generated as a result of this. However the advertising of the capability would have impact on http://www.ietf.org/internet-drafts/draft-ietf-sipping-pstn-call-flows-01.txt, as might be expected.

The example in section 4.1 of RFC 3204 would also be invalidated by the proposal as it stands.

Detailed comments:

1)	I assume that reference [5] should be to RFC 3261, rather than RFC 3162. Correction required both in the body of the document and in the references themselves.

2)	In section 4 and 5, I am unsure why the Supported header needs to be in all requests. Surely INFO can only be used in an INVITE dialog, therefore it is sufficient to put it in the INVITE (and of course OPTIONS). Also why is the same requirement duplicated in both sections. Perhaps the use of the option-tags should be brought out to a separate section, e.g. incorporated into 7.1. Note that there is a typo at the end of the clause 5 text on the option-tags. It is also not practice to put header names in quotation marks.

3)	Figure 1 and figure 2 should really be tables rather than figures, and should show all the methods currently defined - see RFC 3455 for an example.

4)	The Allow-Events header only appears in requests. I assume the same considerations apply to the new Allow-Info header and this should be appropriately indicated in figure 1.

5)	In figure 2, surely the Info-Type header only applies to requests and this should be indicated appropriately, as only the INFO request can carry the data.

6)	The IANA considerations section is missing the Info-Type header.

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 14 January 2003 02:22
> To: sip@ietf.org
> Subject: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> 
> It only seemed appropriate to follow up my previous assumably 
> eloquent but
> frequently misspelled exhortations by following my own advice 
> and "sending
> text".
> 
> The resulting draft has been submitted to the archive and should be
> announced in due course. In the interim, you may wish to 
> retrieve it from:
> 
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopac
kage-00.html

or

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.txt

Or, if you just LIKE looking at XML source or wish to fix some of my
probable glaring errors,

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.xml


I'd like to caveat that this is a rough first draft -- I just sat down and
banged it out this afternoon, and only Adam Roach has seen any of it so far,
and that was in small bits.

thanks

--
Dean

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CC: Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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> To negotiate what INFO can be used for between nodes is good thing, and at least
> on a theoretical level it solves one issue.

It would be nice to motivate how this differs and intersects with the 
Content-* indication and negotiation mechanism. After all, if INFO 
transports objects that are opaque to the "SIP layer" (e.g., session 
state), the logical indication of use is via the various Content-* 
indications.

> 
> But, is the biggest problem we have today end-nodes receiving INFOs with data
> they don't know to handle? And, how is this going to solve the problem of having
> people using INFO for whatever purpose? All they have to do is to define a
> package for their who-knows-what data, right? Or, does defining a package allow
> one to use INFO for whatever he/she wants?

Beyond ISUP and maybe DTMF, what is INFO used for today, by both the 
good and bad guys? What proposals are out there? Last time this was 
discussed, the list seemed pretty short. I think we agree that DTMF has 
better, non-INFO solutions.

With events, we have a whole set of examples and applications to draw 
on. Building a mechanism on an example space of one seems a bit dangerous.

> 
> Or, do we accept Henning's rule about the "bad guys", and do NOT try to define
> any rules on exactly what kind of data INFO should be used for in the first
> place?

I do think it would be helpful if this draft or some other draft 
indicated where SIP INFO *may* be appropriate. As discussed, this seems 
to involve things where SIP routing and proxies are needed or useful, 
for example.

> 
> I am not saying that this draft should solve anything, but I would like to know
> exactly which issues regarding INFO we are going to try solving :)
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> 
> 
> 
> 
> Dean Willis wrote:
> 
> 
>>It only seemed appropriate to follow up my previous assumably eloquent but
>>frequently misspelled exhortations by following my own advice and "sending
>>text".
>>
>>The resulting draft has been submitted to the archive and should be
>>announced in due course. In the interim, you may wish to retrieve it from:
>>
>>http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.html
>>
>

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Drage, Keith \(Keith\)'" <drage@lucent.com>, <sip@ietf.org>
Date: Tue, 14 Jan 2003 12:57:31 -0600
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Subject: [Sip] Table 2 and  INFO Packaging --  draft-willis-sip-infopackage-00
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Keith said:
> 3)	Figure 1 and figure 2 should really be tables rather 
> than figures, and should show all the methods currently 
> defined - see RFC 3455 for an example.
. . . 
> 5)	In figure 2, surely the Info-Type header only applies 
> to requests and this should be indicated appropriately, as 
> only the INFO request can carry the data.

We have SO got to stop trying to to update these tables. It's a cubic
expansion problem, and I don't think it's actually possible to get right, as
new stuff appears between the writing and the publishing.

Any ideas on alternatives? Can we simply stop tracking this? Does it need to
be turned into some kind of IANA registry? How could we approach the problem
if we made it a registry?


--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Tue, 14 Jan 2003 12:57:31 -0600
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> To negotiate what INFO can be used for between nodes is good 
> thing, and at least on a theoretical level it solves one issue.
> 
> But, is the biggest problem we have today end-nodes receiving 
> INFOs with data they don't know to handle? 

Well, the "I don't know what this means" problem is really there. SIP-T
basically requires configuration to "know" that the other end understands
and wants ISUP. That's a little bit clumsy.

> And, how is this 
> going to solve the problem of having people using INFO for 
> whatever purpose? All they have to do is to define a package 
> for their who-knows-what data, right? Or, does defining a 
> package allow one to use INFO for whatever he/she wants?


Welll, this draft basically induces the same sort of review process that is
in place for P-headers and event packages -- to get a registered
info-package "name", you have to document this usage as an RFC (any track)
that receives expert review from SIP. The goal here is to help make sure
that anything widely deployed is at least documented in a reasonable
fashion.

My basic philosphy is that if something is going be be a kitchen sink, we
should at least make it a sturdy kitchen sink, try to keep it from getting
plugged up with garbage, and establish some communications between the users
of the sink.
 
> Or, do we accept Henning's rule about the "bad guys", and do 
> NOT try to define any rules on exactly what kind of data INFO 
> should be used for in the first place?

Personally, I don't really care what kind of data INFO gets used for. Market
reality is going to shape the usages towards "stuff that works" -- for
example, I believe the world has realized that using INFO for floor control
over 3G wireless networks just doesn't work. But somebody had to try it
before we really knew.  It seems to me that INFO is reasonably useful
wherever one needs to send application data along a relationship path
established by a SIP dialog, AND that data has no particular relationship
with the SIP state machine, AND the timing, load, and capacity
characteristics of the SIP network are appropriate for that sort of
application. People will probably come up with some downright weird usages.
Some may work. Some probably won't. But as long as they don't step on each
other, they're probably only hurting themselves. This draft (like the
congestion safety work) is about keeping people from stepping on each other.


--
Dean

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
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Subject: [Sip] RE: Table 2 and  INFO Packaging --  draft-willis-sip-infopackage-
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The problem is that these tables are the only place where some of this information appears. By all means find some other mechanism, but the information is still in general necessary.

For example, my comment 5) that the header only applies to a request could quite easily be written in text, but is normally shown only by an "R" in the tables. However it is specified, it still seems to me to be a fairly important bit of information, and if it is not followed could lead to a number of incompatibilities wehre one side provides the information and the other side says it is unnecessary to follow it.

In general, there is a very common pattern to the entries in the tables. One possible solution would be to replace the existing method columns by a set of columns that relate to

-	initial method of dialog
-	target refresh for a dialog
-	subsequent method of dialog
-	method outside dialog

unfortunately ACK and CANCEL seem to have their own special set of inconsistencies with regard to the use of various headers, but this can probably be dealt with entirely in RFC 3261. Similarly REGISTER creates its own problems, but could be dealt with by saying that new headers are handled as "method outside dialog" unless explicit requirements are stated elsewhere.

Such a change would get rid of the 2nd dimension, where the table extends further to the right as new methods are added, and may actually introduce a bit more consistency into the usages.

Note that the where and proxy columns should continue to be used as is.

[Aside - alternatively we could just stop adding new methods and use INFO for everything :-) ]

Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 14 January 2003 18:58
> To: 'Drage, Keith (Keith)'; sip@ietf.org
> Subject: Table 2 and INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> Keith said:
> > 3)	Figure 1 and figure 2 should really be tables rather 
> > than figures, and should show all the methods currently 
> > defined - see RFC 3455 for an example.
> . . . 
> > 5)	In figure 2, surely the Info-Type header only applies 
> > to requests and this should be indicated appropriately, as 
> > only the INFO request can carry the data.
> 
> We have SO got to stop trying to to update these tables. It's a cubic
> expansion problem, and I don't think it's actually possible 
> to get right, as
> new stuff appears between the writing and the publishing.
> 
> Any ideas on alternatives? Can we simply stop tracking this? 
> Does it need to
> be turned into some kind of IANA registry? How could we 
> approach the problem
> if we made it a registry?
> 
> 
> --
> Dean
> 
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thanks juha,
i also reposted in the morgue on softarmor

http://kevlar.softarmor.com/sipping/drafts/morgue/draft-sterman-aaa- 
sip-00.txt

thx,
-r

On Friday, January 3, 2003, at 11:03 AM, jh@lohi.eng.song.fi wrote:

> Michael Thomas writes:
>
>>    Well, it seems to be expried for one... is the jist
>>    that AAA's have been hacked to deal with digest auth
>>    types directly (or, say, as an EAP type)?
>
> i made draft-sterman-aaa-sip-00.txt available as
>
> ftp://lohi.eng.song.fi/tmp/draft-sterman-aaa-sip-00.txt
>
> the directory is not listable.
>
> -- juha
>
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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Henning said:
> It would be nice to motivate how this differs and intersects with the 
> Content-* indication and negotiation mechanism. After all, if INFO 
> transports objects that are opaque to the "SIP layer" (e.g., session 
> state), the logical indication of use is via the various Content-* 
> indications.


I'm not sure I understand this. Are you saying that we could replace INFO
usage with a Content-Indirected MIME type?

> Beyond ISUP and maybe DTMF, what is INFO used for today, by both the 
> good and bad guys? What proposals are out there? Last time this was 
> discussed, the list seemed pretty short. I think we agree 
> that DTMF has 
> better, non-INFO solutions.

Well, people have also used or mentioned using INFO for 

* conference control (instructing a conference server to add and delete
legs)

* floor control in a conference bridge (which has timing issues for 3G)

* gaming applications (like transmitting chess moves in parallel to a voice
call)

* pushing user interface components, like data entry forms, keypad layouts
(kpml type stuff), and java applets

* delivering SOAP

* delivering vCards

* delivering calendar data

Some of these are probably bogus. Others may be better-implemented other
ways. All, if implemented usingINFO, could benefit from negotiation (Does
the other guy SUPPORT sending chess moves?) and documentation (Does she mean
the same thing I mean when I send her a application/xchess INFO . . .


--
dean

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Date: Tue, 14 Jan 2003 15:28:59 -0500
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Dean Willis <dean.willis@softarmor.com>
CC: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>, sip@ietf.org
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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> I'm not sure I understand this. Are you saying that we could replace INFO
> usage with a Content-Indirected MIME type?

No. We currently negotiate above-SIP content with Content-Type, 
Content-Language and all the other Content-Headers, just like HTTP, SMTP 
and other MIME users. If the body contains message/isup, it is an ISUP 
message. It is not clear to me what value an additional, semi-orthogonal 
negotiation mechanism offers. (Semi-orthogonal, since most INFO types 
will have exactly one valid Content-Type.)

> Well, people have also used or mentioned using INFO for 
> 
> * conference control (instructing a conference server to add and delete
> legs)
> 
> * floor control in a conference bridge (which has timing issues for 3G)
> 
> * gaming applications (like transmitting chess moves in parallel to a voice
> call)
> 
> * pushing user interface components, like data entry forms, keypad layouts
> (kpml type stuff), and java applets
> 
> * delivering SOAP
> 
> * delivering vCards
> 
> * delivering calendar data
> 
> Some of these are probably bogus. Others may be better-implemented other
> ways. All, if implemented usingINFO, could benefit from negotiation (Does
> the other guy SUPPORT sending chess moves?) and documentation (Does she mean
> the same thing I mean when I send her a application/xchess INFO . . .

It seems that the meaning should precisely depend on the content type 
and the content disposition.

> 
> 
> --
> dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>
Cc: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>, <sip@ietf.org>
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Tue, 14 Jan 2003 17:40:04 -0600
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> could replace 
> > INFO usage with a Content-Indirected MIME type?
> 
> No. We currently negotiate above-SIP content with Content-Type, 
> Content-Language and all the other Content-Headers, just like 
> HTTP, SMTP 
> and other MIME users. If the body contains message/isup, it 
> is an ISUP 
> message. It is not clear to me what value an additional, 
> semi-orthogonal 
> negotiation mechanism offers. (Semi-orthogonal, since most INFO types 
> will have exactly one valid Content-Type.)

Ah. Actually, that is the direction that I initially started taking things
in the draft -- basically, I associated MIME content-types with usages, and
didn't have the Info-Type. Then I briefly talked it over with Adam, and he
talked me into the separate semantic indicator. 

I believe that the assertion that "most INFO types will have exactly one
valid Content-Type" is the breaker. Let's look at text. Is the text I just
sent you indicative of a move in the chess game we are playing? Is it
something I want you to post on a web page for me? Should you just read it?
Is it a javascript that you should use to remap your keyboard? Is it C code
that you should compile and execute? Is it a credit card number that you
should debit for the price of this call? Why the heck AM I sending it to
you? 

There are probably much better examples, but I think that a one-to-one
mapping between content-type and expected function is more constraining than
the way I think people are trying to use INFO, perhaps overly constraining.

--
dean


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To: Henning Schulzrinne <hgs@cs.columbia.edu>
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Henning,

>> 
>> But, is the biggest problem we have today end-nodes receiving INFOs 
with data
>> they don't know to handle? And, how is this going to solve the problem 
of having
>> people using INFO for whatever purpose? All they have to do is to 
define a
>> package for their who-knows-what data, right? Or, does defining a 
package allow
>> one to use INFO for whatever he/she wants?
>
>Beyond ISUP and maybe DTMF, what is INFO used for today, by both the 
>good and bad guys? What proposals are out there? Last time this was 
>discussed, the list seemed pretty short. I think we agree that DTMF has 
>better, non-INFO solutions.

Like for ISUP, ECMA is also in the process of defining the use of INFO for 
QSIG.

Regards,

Frank

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--NextPart

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


	Title		: Packaging and Negotiation of INFO Methods for the 
                          Session Initiation Protocol (SIP)
	Author(s)	: D. Willis
	Filename	: draft-willis-sip-infopackage-00.txt
	Pages		: 11
	Date		: 2003-1-14
	
The SIP INFO method (RFC 2976) establishes a method by which
applications may transfer application-specific information within a
SIP dialog.  However, RFC 2976 does not provide a mechanism for
describing and documenting an application of INFO, nor does it
provide a mechanism by which applications may negotiate such uses.
This document provides a framework for documenting and naming
specific uses of INFO (called INFO packages), for registering those
package names with IANA, and for negotiating the support for various
INFO packages between applications using SIP.

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From mailnull@www1.ietf.org  Wed Jan 15 09:00:27 2003
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Hi,

I appologise if this has been discussed before, but at least I didn't
find it in my mail box (I admit I didn't check the archieve on the web).

Anyway...

Chapter 18.2.2 in RFC3261 talks about the routing of responses, using
the Via maddr, received and sent-by parameters.

My question is regarding maddr. When reading the text, I get the picture
that only IP addresses are used in maddr. At least the text doesn't
mention the SRV RFC, which it does for sent-by, and it says that the
default port is 5060. However, the ABNF defines maddr as "host", which
can also be a domain name.

So, is there a bug in the ABNF, or have I missunderstood something?

Regards,

Christer Holmberg
Ericsson Finland

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From mailnull@www1.ietf.org  Wed Jan 15 09:33:38 2003
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>,
        "'Henning Schulzrinne'"
	 <hgs@cs.columbia.edu>
Cc: sip@ietf.org
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Wed, 15 Jan 2003 14:34:31 -0000
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I nearly made the same comment as Henning when I made my comments, but decided I did not know enough about Content-Type etc usage to make the comment sensibly.

I would like to see some more discussion on this as I am not yet convinced either way.

If we end up following Deans proposal for a new header Info-Type, then at least we will need to capture some of this discussion in order that:

-	specifiers of new packages make sensible choices of values for Info-Type
-	implementers choose values of Content-* that are consistent with the Info-Type value.

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 14 January 2003 23:40
> To: 'Henning Schulzrinne'
> Cc: 'Christer Holmberg'; sip@ietf.org
> Subject: RE: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> > could replace 
> > > INFO usage with a Content-Indirected MIME type?
> > 
> > No. We currently negotiate above-SIP content with Content-Type, 
> > Content-Language and all the other Content-Headers, just like 
> > HTTP, SMTP 
> > and other MIME users. If the body contains message/isup, it 
> > is an ISUP 
> > message. It is not clear to me what value an additional, 
> > semi-orthogonal 
> > negotiation mechanism offers. (Semi-orthogonal, since most 
> INFO types 
> > will have exactly one valid Content-Type.)
> 
> Ah. Actually, that is the direction that I initially started 
> taking things
> in the draft -- basically, I associated MIME content-types 
> with usages, and
> didn't have the Info-Type. Then I briefly talked it over with 
> Adam, and he
> talked me into the separate semantic indicator. 
> 
> I believe that the assertion that "most INFO types will have 
> exactly one
> valid Content-Type" is the breaker. Let's look at text. Is 
> the text I just
> sent you indicative of a move in the chess game we are playing? Is it
> something I want you to post on a web page for me? Should you 
> just read it?
> Is it a javascript that you should use to remap your 
> keyboard? Is it C code
> that you should compile and execute? Is it a credit card 
> number that you
> should debit for the price of this call? Why the heck AM I 
> sending it to
> you? 
> 
> There are probably much better examples, but I think that a one-to-one
> mapping between content-type and expected function is more 
> constraining than
> the way I think people are trying to use INFO, perhaps overly 
> constraining.
> 
> --
> dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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>>Ah. Actually, that is the direction that I initially started 
>>taking things
>>in the draft -- basically, I associated MIME content-types 
>>with usages, and
>>didn't have the Info-Type. Then I briefly talked it over with 
>>Adam, and he
>>talked me into the separate semantic indicator. 
>>
>>I believe that the assertion that "most INFO types will have 
>>exactly one
>>valid Content-Type" is the breaker. Let's look at text. Is 
>>the text I just
>>sent you indicative of a move in the chess game we are playing? Is it
>>something I want you to post on a web page for me? Should you 
>>just read it?
>>Is it a javascript that you should use to remap your 
>>keyboard? Is it C code
>>that you should compile and execute? Is it a credit card 

The mistake would be specifying all of them as text/plain. They should 
be specified as the correct media type to begin with. After all, I may 
want to save the content and saving a C file as .txt instead of .c is 
not very helpful.

If this is application data, it should be labeled as application/chess.

In addition, there is Content-Disposition to look at.


>>number that you
>>should debit for the price of this call? Why the heck AM I 
>>sending it to
>>you? 
>>
>>There are probably much better examples, but I think that a one-to-one
>>mapping between content-type and expected function is more 
>>constraining than
>>the way I think people are trying to use INFO, perhaps overly 
>>constraining.
>>
>>--
>>dean
>>
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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



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From: "Tom-PT Taylor" <taylor@nortelnetworks.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "Drage, Keith (Keith)"
	 <drage@lucent.com>
Cc: "'Dean Willis'" <dean.willis@softarmor.com>, sip@ietf.org
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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I really agree with Henning's viewpoint.  The mechanisms are already there.
Quite frankly, I don't want to have to redo my SIP-T implementation of INFO
unless a really good case can be made for it.

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu] 
> Sent: Wednesday, January 15, 2003 10:13 AM
> To: Drage, Keith (Keith)
> Cc: 'Dean Willis'; sip@ietf.org
> Subject: Re: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> 
> >>Ah. Actually, that is the direction that I initially started
> >>taking things
> >>in the draft -- basically, I associated MIME content-types 
> >>with usages, and
> >>didn't have the Info-Type. Then I briefly talked it over with 
> >>Adam, and he
> >>talked me into the separate semantic indicator. 
> >>
> >>I believe that the assertion that "most INFO types will have
> >>exactly one
> >>valid Content-Type" is the breaker. Let's look at text. Is 
> >>the text I just
> >>sent you indicative of a move in the chess game we are 
> playing? Is it
> >>something I want you to post on a web page for me? Should you 
> >>just read it?
> >>Is it a javascript that you should use to remap your 
> >>keyboard? Is it C code
> >>that you should compile and execute? Is it a credit card 
> 
> The mistake would be specifying all of them as text/plain. 
> They should 
> be specified as the correct media type to begin with. After 
> all, I may 
> want to save the content and saving a C file as .txt instead of .c is 
> not very helpful.
> 
> If this is application data, it should be labeled as 
> application/chess.
> 
> In addition, there is Content-Disposition to look at.
> 
> 
> >>number that you
> >>should debit for the price of this call? Why the heck AM I
> >>sending it to
> >>you? 
> >>
> >>There are probably much better examples, but I think that a 
> one-to-one 
> >>mapping between content-type and expected function is more 
> >>constraining than the way I think people are trying to use INFO, 
> >>perhaps overly constraining.
> >>
> >>--
> >>dean
> >>
> >>
> >>_______________________________________________
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP Protocol
> >>Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> >>sipping@ietf.org for new developments on the application of sip
> >>
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip
> 

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<TITLE>RE: [Sip] INFO Packaging --  =
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<P><FONT SIZE=3D2>I really agree with Henning's viewpoint.&nbsp; The =
mechanisms are already there.&nbsp; Quite frankly, I don't want to have =
to redo my SIP-T implementation of INFO unless a really good case can =
be made for it.</FONT></P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Henning Schulzrinne [<A =
HREF=3D"mailto:hgs@cs.columbia.edu">mailto:hgs@cs.columbia.edu</A>] =
</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: Wednesday, January 15, 2003 10:13 =
AM</FONT>
<BR><FONT SIZE=3D2>&gt; To: Drage, Keith (Keith)</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: 'Dean Willis'; sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] INFO Packaging -- =
draft-willis-sip-infopackage-00</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;Ah. Actually, that is the direction =
that I initially started</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;taking things</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;in the draft -- basically, I associated =
MIME content-types </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;with usages, and</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;didn't have the Info-Type. Then I =
briefly talked it over with </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;Adam, and he</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;talked me into the separate semantic =
indicator. </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;I believe that the assertion that =
&quot;most INFO types will have</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;exactly one</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;valid Content-Type&quot; is the =
breaker. Let's look at text. Is </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;the text I just</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;sent you indicative of a move in the =
chess game we are </FONT>
<BR><FONT SIZE=3D2>&gt; playing? Is it</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;something I want you to post on a web =
page for me? Should you </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;just read it?</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;Is it a javascript that you should use =
to remap your </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;keyboard? Is it C code</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;that you should compile and execute? Is =
it a credit card </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; The mistake would be specifying all of them as =
text/plain. </FONT>
<BR><FONT SIZE=3D2>&gt; They should </FONT>
<BR><FONT SIZE=3D2>&gt; be specified as the correct media type to begin =
with. After </FONT>
<BR><FONT SIZE=3D2>&gt; all, I may </FONT>
<BR><FONT SIZE=3D2>&gt; want to save the content and saving a C file as =
.txt instead of .c is </FONT>
<BR><FONT SIZE=3D2>&gt; not very helpful.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; If this is application data, it should be =
labeled as </FONT>
<BR><FONT SIZE=3D2>&gt; application/chess.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; In addition, there is Content-Disposition to =
look at.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;number that you</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;should debit for the price of this =
call? Why the heck AM I</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;sending it to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;you? </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;There are probably much better =
examples, but I think that a </FONT>
<BR><FONT SIZE=3D2>&gt; one-to-one </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;mapping between content-type and =
expected function is more </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;constraining than the way I think =
people are trying to use INFO, </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;perhaps overly constraining.</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;--</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;dean</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; =
&gt;&gt;_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;Use sip-implementors@cs.columbia.edu =
for questions on </FONT>
<BR><FONT SIZE=3D2>&gt; current sip Use </FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;sipping@ietf.org for new developments =
on the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; &gt; This list is for NEW development of the =
core SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Use sip-implementors@cs.columbia.edu for =
questions on </FONT>
<BR><FONT SIZE=3D2>&gt; current sip Use </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current </FONT>
<BR><FONT SIZE=3D2>&gt; sip Use sipping@ietf.org for new developments =
on the </FONT>
<BR><FONT SIZE=3D2>&gt; application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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Date: Tue, 14 Jan 2003 18:39:29 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Dean Willis <dean.willis@softarmor.com>
CC: "'Prasanna Venkatesh'" <prasanna@huawei.com>, joanne.higgens@orange.co.uk,
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Subject: Re: [Sip] SIP de-registration
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Catching up after a week of "real work":

I have one bit to add to what Dean is saying. If a registrar should
reject a request because, say, there is an "active session" and the
policy of the registrar prohibits de-registration during an active
session, you really must use an appropriate response code to have a hope
of interop. The appropriate response is a 500 (not any 500 class). The
reason is that the 500 can contain a Retry-After, informing the UA that
it should try again later. Otherwise, when the registration is rejected,
what should it do? Should it try again? When should it try again?

If, however, the intent is that the registration is accepted, but will
take affect later, that is something quite different, and would require
some standardization activity to make interoperable.

Of course, that said, I agree with Dean that the intention of REGISTER
is merely to establish an address binding, and not to be a "log in" in
the sense that it is frequently mis-interpreted.

-Jonathan R.

Dean Willis wrote:
 > Prasanna said:
 >
 >>Jo said:
 >>
 >>>Is it valid to reject a de-registration request?
 >>>
 >>>For example 'If a  SIP terminal sends a de-registration
 >>>request to a network while it still has an active session
 >>>established, and the network rejects this request , is this
 >>>correct behaviour.
 >>
 >
 >>Any registrar can reject a de-registration request anytime.
 >>But rejecting a de-registration for the fact that a session
 >>is active is a local implementation.  In general the
 >>registration is used to locate an entity and theoritically
 >>after the session is established, the registration is
 >>immaterial. Cheers, Prasanna
 >
 >
 > I believe Prasanna is correct here, in general. SIP does not view 
REGISTER
 > as a "log on for service" -- rather, it is an address binding 
function that
 > makes a temporary association between a (presumbaly temporary) local
 > instance address and a (presumbaly long-lived) address-of-record. In
 > particular, SIP does not require that a UA have a current registration or
 > execute a REGISTER transaction before executing an INVITE transaction
 > request.
 >
 > 3GPP (and consequently 3GPP2) have however layered some extra roles 
on top
 > of the SIP REGISTER transaction. In these environments, REGISTER is a 
sort
 > of "log on" operation in that it establishes the security association
 > between a client and registrar that will be used for (some) future
 > interactions between that client and registrar.
 >
 > Consequently, answering the question that Jo asked requires knowing 
whether
 > the 3GPP system discards the security association established by a 
REGISTER
 > request if the registration created by that request is destroyed.
 >
 > The answer, unfortunately, is "I don't know. The IETF doesn't know. Maybe
 > somebody more conversant with that particular bit of 3GPPP knows (and
 > they're probably reading this)." But in short, it is an environment 
specific
 > question related to a specific implementation having done something 
just a
 > bit different from generic SIP. In straight SIP, the registrar can 
terminate
 > a registration without impacting any sessions that a UA is 
participating in.
 > In fact, the registrar usually doesn't know or care whether a UA has 
or will
 > ever have participated in a session.
 >
 > My guess is that this would probably be implementation-depdendent in the
 > S-CSCF, and that if I were designing an S-CSCF, I'd have it not discard
 > SA-state until after all transacations on that SA had terminated. A 
second
 > choice would be to design it to use network-initiated-termination 
procedures
 > to force-terminate any sessions before discarding the security 
associations.
 >
 > --
 > dean
 >
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip
 >

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        "'Drage, Keith \(Keith\)'" <drage@lucent.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Wed, 15 Jan 2003 14:04:59 -0600
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> The mistake would be specifying all of them as text/plain. 
> They should 
> be specified as the correct media type to begin with. After 
> all, I may 
> want to save the content and saving a C file as .txt instead of .c is 
> not very helpful.
> 
> If this is application data, it should be labeled as 
> application/chess.
> 
> In addition, there is Content-Disposition to look at.

Consider XML. Do we end up with a different MIME registration for every
different XML schema? Or would we have to re-extend INFO to say "If the data
you've gotten is XML, then you must parse the schema declaration to know
which application reistered on this dialog gets the data?"

--
Dean

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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Via maddr
Date: Wed, 15 Jan 2003 16:25:37 -0500
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> I appologise if this has been discussed
> before, but at least I didn't find it in
> my mail box (I admit I didn't check the
> archieve on the web).

I posted a related question to sip-implementors;
however I never got a reply.

http://lists.cs.columbia.edu/pipermail/sip-implementors/2002-October/004052.
html

> Chapter 18.2.2 in RFC3261 talks about the
> routing of responses, using the Via maddr,
> received and sent-by parameters.
>
> My question is regarding maddr. When reading
> the text, I get the picture that only
> IP addresses are used in maddr. At least the
> text doesn't mention the SRV RFC, which it does
> for sent-by, and it says that the default port
> is 5060. However, the ABNF defines maddr as
> "host", which can also be a domain name.

maddr can be a domain name.  The text appears
to disable the ability to perform an SRV query
associated with the Via maddr.  However I never
got an answer concerning if this was intentional.

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To: Dean Willis <dean.willis@softarmor.com>
CC: "'Drage, Keith \(Keith\)'" <drage@lucent.com>, sip@ietf.org
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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> Consider XML. Do we end up with a different MIME registration for every
> different XML schema? Or would we have to re-extend INFO to say "If the data
> you've gotten is XML, then you must parse the schema declaration to know
> which application reistered on this dialog gets the data?"

I thought we used things like application/xml+cpl for different schema?

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Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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----- Original Message ----- 
From: "Drage, Keith (Keith)" <drage@lucent.com>
> I would note that the proposal invalidates existing usages
> of the INFO method, as presumably all usages of INFO
> carry the new header Info-Type and the dialog should
> presumably be created carrying the Allow-Info header.

Note well that we had a completely congruent situation
with SIP events (PINT had already made it to RFC),
and the solution was both straightforward and
uncontroversial.

Dean: you should probably add a section that's a cut,
paste, and tiny edit from section 3.3.8 of RFC 3265.

/a

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References: <001f01c2bc06$d4609400$fe0c0c42@txdwillis> <3E24730B.2050707@cs.columbia.edu>
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Wed, 15 Jan 2003 17:32:17 -0600
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From: "Henning Schulzrinne" <hgs@cs.columbia.edu>
> 
> No. We currently negotiate above-SIP content with Content-Type, 
> Content-Language and all the other Content-Headers, just like HTTP, SMTP 
> and other MIME users. If the body contains message/isup, it is an ISUP 
> message. It is not clear to me what value an additional, semi-orthogonal 
> negotiation mechanism offers. (Semi-orthogonal, since most INFO types 
> will have exactly one valid Content-Type.)

Not so sure about this. To take your example, we already have proposals
for application/isup and application/vnd.sentito.isup-text (or whatever
draft-miller-sip-isup-annex-00 plans to propose).

/a

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Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Wed, 15 Jan 2003 17:36:02 -0600
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RE: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
You don't have to re-do it, as long as we grandfather its usage like we did
with
PINT in RFC 3265.

/a

----- Original Message -----
From: Tom-PT Taylor
To: 'Henning Schulzrinne' ; Drage, Keith (Keith)
Cc: 'Dean Willis' ; sip@ietf.org
Sent: Wednesday, January 15, 2003 9:36
Subject: RE: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00


I really agree with Henning's viewpoint.  The mechanisms are already there.
Quite frankly, I don't want to have to redo my SIP-T implementation of INFO
unless a really good case can be made for it.


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From: "Adam Roach" <adam@dynamicsoft.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>, <sip@ietf.org>
References: <001f01c2bc06$d4609400$fe0c0c42@txdwillis> <3E24730B.2050707@cs.columbia.edu>
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Wed, 15 Jan 2003 18:02:25 -0600
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"Henning Schulzrinne" <hgs@cs.columbia.edu> writes:

> We currently negotiate above-SIP content with Content-Type, 
> Content-Language and all the other Content-Headers, just like HTTP, SMTP 
> and other MIME users. If the body contains message/isup, it is an ISUP 
> message. It is not clear to me what value an additional, semi-orthogonal 
> negotiation mechanism offers.

Upon further reflection, I realise that my previous response addressed
the wrong half of this objection (multiple content types per application,
instead of multiple applications per content type).

Take, for example, the content type of "image/jpeg".

It is quite conceivable that someone might define an INFO
package of "app-icon" that uses "image/jpeg" to send
a 32x32 pixel image that should be used as the icon for the
application for the duration of the session.

It is also completely beleivable that someone else might
define an INFO package of "set-pointer" that uses "image/jpeg"
to send a 16x16 bitmap that is to be used as the mouse pointer
when said pointer is in the window associated with the
application.

Now, I'm not advocating these usages of INFO or even the
ability to do either of these things with SIP, but I'm certain
that you will agree that these are the *sort* of content-type
collisions that you will encounter if you don't have the
orthagonal mechanism that Dean includes in his draft.

/a

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Subject: [Sip] Comments  RFC 3329 Security Mechanism Agreement for SIP
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Hi Gonzalo & company, 
Some comments on RFC 3329:

*	Section 2.4 Security mechanisms:

You dont specify the behavior w.r.to a proxy when it comes to headers such as Security-Client. You only specify the proxy behavior towards the Require and Proxy-Require headers. 
Also,  what should the proxy do to a Supported header if it is present
*	Section 2.3.2 Server initiated :
It is stated:
"A server that by policy requires the use of this specification and receives a request that does not have the sec-agree option tag in a Require, Proxy-Require or Supported header field MUST return a 421 (Extension Required) response"
I read the text that it mandates the presence of the 3 headers for the server to accept the request.
Is that the case?
If the answer is yes, then I dont see that it is mandated in the RFC that the client MUST include Supported header. The strength used for the client w.r.to Supported header is SHOULD.
If the answer is no, then please clarify the text to eliminate any ambinguity.
*	Section 2.6
The table indicates that a proxy can add a Security-Client, and a Security-Verify. This looks like an error.
Also, I beleive that 421, 494 response can add a Security-Server.
Please verify the table
Rgds/gf




 



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charset=3DISO-8859-1">
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<TITLE>Comments  RFC 3329 Security Mechanism Agreement for SIP</TITLE>
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<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hi Gonzalo &amp; company, </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Some comments on RFC 3329:</FONT>
</P>

<UL><LI><FONT SIZE=3D2 FACE=3D"Arial">Section 2.4 Security =
mechanisms:</FONT></LI>
<BR>
</UL>
<P><FONT SIZE=3D2 FACE=3D"Arial">You dont specify the behavior w.r.to a =
proxy when it comes to headers such as Security-Client. You only =
specify the proxy behavior towards the Require and Proxy-Require =
headers. </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Also,&nbsp; what should the proxy do =
to a Supported header if it is present</FONT>

<UL><LI><FONT SIZE=3D2 FACE=3D"Arial">Section 2.3.2 Server initiated =
:</FONT></LI>
</UL>
<P><FONT SIZE=3D2 FACE=3D"Arial">It is stated:</FONT>
<BR><FONT FACE=3D"Times New Roman">&quot;A server that by policy =
requires the use of this specification and receives a request that does =
not have the sec-agree option tag in a Require, Proxy-Require or =
Supported header field MUST return a 421 (Extension Required) =
response&quot;</FONT></P>

<P><FONT FACE=3D"Times New Roman">I read the text that it mandates the =
presence of the 3 headers for the server to accept the request.</FONT>
<BR><FONT FACE=3D"Times New Roman">Is that the case?</FONT>
<BR><FONT FACE=3D"Times New Roman">If the answer is yes, then I dont =
see that it is mandated in the RFC that the client MUST include =
Supported header. The strength used for the client w.r.to Supported =
header is SHOULD.</FONT></P>

<P><FONT FACE=3D"Times New Roman">If the answer is no, then please =
clarify the text to eliminate any ambinguity.</FONT>

<UL><LI><FONT FACE=3D"Times New Roman">Section 2.6</FONT></LI>
</UL>
<P><FONT FACE=3D"Times New Roman">The table indicates that a proxy can =
add a Security-Client, and a Security-Verify. This looks like an =
error.</FONT>
<BR><FONT FACE=3D"Times New Roman">Also, I beleive that 421, 494 =
response can add a Security-Server.</FONT>
<BR><FONT FACE=3D"Times New Roman">Please verify the table</FONT>
<BR><FONT FACE=3D"Times New Roman">Rgds/gf</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT FACE=3D"Times New Roman">&nbsp;</FONT>
</P>
<BR>

</BODY>
</HTML>
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CC: Dean Willis <dean.willis@softarmor.com>,
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        sip@ietf.org
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
References: <001f01c2bc06$d4609400$fe0c0c42@txdwillis> <3E24730B.2050707@cs.columbia.edu> <047801c2bcf2$aecc0930$bd346b83@dynamicsoft.com>
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Consider the interaction with Content-Disposition.

Adam Roach wrote:
> "Henning Schulzrinne" <hgs@cs.columbia.edu> writes:
> 
> 
>>We currently negotiate above-SIP content with Content-Type, 
>>Content-Language and all the other Content-Headers, just like HTTP, SMTP 
>>and other MIME users. If the body contains message/isup, it is an ISUP 
>>message. It is not clear to me what value an additional, semi-orthogonal 
>>negotiation mechanism offers.
> 
> 
> Upon further reflection, I realise that my previous response addressed
> the wrong half of this objection (multiple content types per application,
> instead of multiple applications per content type).
> 
> Take, for example, the content type of "image/jpeg".
> 
> It is quite conceivable that someone might define an INFO
> package of "app-icon" that uses "image/jpeg" to send
> a 32x32 pixel image that should be used as the icon for the
> application for the duration of the session.
> 
> It is also completely beleivable that someone else might
> define an INFO package of "set-pointer" that uses "image/jpeg"
> to send a 16x16 bitmap that is to be used as the mouse pointer
> when said pointer is in the window associated with the
> application.

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Subject: [Sip] Extension to Assure Congestion Safety
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Hi

are there news about the Congestion problem ?
I read some weeks ago that work is frozen. 
But your draft expires at the end of January. 
Will be some change in new release? If it will be resubmitted.

About the change with TCP as Transport Protocol default,
someone can tell me some news?

Sal
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My apologies, I missed this one...

The difference here is that TCAP binary messages do not traverse the IP
portion of the network.  The impetus is similar to that for translating the
ISUP maintenance commands, SS7 stacks cost a lot (both $$ and point codes).
It makes more sense to implement a simple text representation that traverses
the IP portion of the network and is converted to binary TCAP at the gateway
to the SS7 cloud.  Also, from a philosophical point of view, since I believe
that SIP will supplant SS7 over the long run, limiting the need to implement
SS7 in new network elements is "a good thing".  This last point is of
course, just MHO.

FM


Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com


----- Original Message -----
From: <hbhondwe@hss.hns.com>
To: <sipping@ietf.org>
Cc: <fmiller@sentito.com>
Sent: Thursday, January 16, 2003 8:48 AM
Subject: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt : Reposting


>
>
> Reposting this mail as it did not receive a response on the list the first
> time.
>
>
>
>
>
> hbhondwe@hss.hns.com on 01/10/2003 11:13:02 AM
>
> To:   sipping@ietf.org
> cc:    (bcc: Harsh Bhondwe/HSSBLR)
>
> Subject:  [Sipping] Comments: I-D ACTION:draft-miller-sip-tcap-00.txt
>
>
>
>
>
>
> Hi ,
>
> I have a fundamental question on the the usage of SIP for
> transporting TCAP messages.
>
> If I understand correctly the figure 1 in the draft, is
> something like the following:
>
>
> SIP Element           Signaling Gateway         SS7 Node
>
> TCAP applications                               PSTN Services
> TCAP                                            TCAP
> SIP--------IP----------SIP||SS7--------SS7------SS7
>
>
> Here a TCAP peer will be running on the SIP Element and will
> use SIP to exchange TCAP messages with peers(i.e other TCAPs)
> in the PSTN network.
>
> However, SIGTRAN protocols defined also by the SIGTRAN
> Working group of the  IETF have been especially designed
> for transporting Trunk/Access signaling over IP.
>
> So, could one not better accomplish this  by the
> following configuration using SIGTRAN?
>
>
> SIP Element                   Signaling Gateway        PSTN Node
>
> TCAP applications                                   PSTN Services
> TCAP                                                   TCAP
> SIGTRAN---------IP-------------SIGTRAN||SS7 ---SS7-----SS7
> (SUA over SCTP)                (SUA over SCTP)
>
>
> The difference of course is that in this case the SIP
> Element and the Signaling Gateway will have to (also)
> support SIGTRAN.They could continue to support SIP as well.
>
> Or am i missing something?
>
> thanks
> harsh
>
>
>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>
>
>
>
>

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Subject: Re: [Sip] Extension to Assure Congestion Safety
From: Dean Willis <dean.willis@softarmor.com>
To: loretosa@vizzavi.it
Cc: bcampbell@dynamicsoft.com, sip@ietf.org
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On Thu, 2003-01-16 at 06:36, loretosa@vizzavi.it wrote:
> Hi
> 
> are there news about the Congestion problem ?
> I read some weeks ago that work is frozen. 
> But your draft expires at the end of January. 
> Will be some change in new release? If it will be resubmitted.
> 
> About the change with TCP as Transport Protocol default,
> someone can tell me some news?
> 

Thanks, Sal. Actually, we haven't talked much about this lately, and we
probably should.

When conversation last stopped, we had just realized that although we
could use this sort of mechanism in the forward direction, it doesn't do
anything about the reverse direction.

So, let's say somebody S/MIME signs a 200 OK, making it perhaps 4KB in
size (That doesn't necessarily happen, but it MIGHT if they include
their cert). Since this is a response, it MUST follow the path laid out
in the Via header field, even if this takes it across non-congestion
safe UDP links.

Is this a problem that we should worry about?

If so, can we address it short of jut deprecating UDP? 

One approach that came up is essentially tagging the Via values so that
we know which are congestion safe. Under this approach, the receiving UA
would be able to recognize that the large response it is generating
could not be returned safely. What should it do then? Could such a
mechanism be made backward compatible?

How much additional triggerwork are we willing to put into the protocol
just to continue accommodating UDP? I know a number of people believe
that a lot of the complexity of SIP stems from its concurrent support
for UDP and TCP, and that we'd be better off just ripping the UDP out
and consequently greatly simplifying the state machines than adding more
complexity to the core protocol.

So, we don't really have a group consensus about where to go with this,
and we need one. 

Ladies, gentleman, and those to whom these terms are irrelevant, the
floor is open for discussion.

--
Dean


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Hi,

When reading the latest (-07) of the 2806bis draft, I see that the "+"
prefix is mandatory for both global- and local numbers.

I assume this means that if a SIP-URI has the user=phone parameter set,
the userpart is not valid unless there is a "+" prefix (since RFC3261
says the userpart shall be parsed as a TEL-URL telephone-uri in this
case)?

The reason I am confused about this because in the examples (chapter 6)
in 2806bis there is no "+" sign in front of the local numbers. Is this a
leftover from previous versions, when there were no "+" in front of
local numbers, or is there a bug in the ABNF?

I also wonder why there has to be a "+" sign in front of a local number
in the first place? When reading chapter 5.1.3 and 5.1.4 I find nowhere
an explanation for this (5.1.3 even says that global numbers are
identified by the "+" prefix).

Regards,

Christer Holmberg
Ericsson Finland

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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Subject: [Sip] TEL-URL local numbers
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looks to me that the syntax has a bug, i.e., local numbers don't need
start with +.

i have another question.  why is context mandatory in local numbers?  i
would like to make it optional and if it is missing, the context would
be derived from the from uri.

-- juha

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Hi,

> looks to me that the syntax has a bug, i.e., local numbers don't need
> start with +.

Local numbers MUST NOT start with "+", I'd say.

I also wonder a little about the syntax for local numbers:

local-number          =  local-number-digits *par context *par

I wonder why context is "surrounded" by optional pars. I guess there is
nothing wrong with it from an ABNF point of view, but wouldn't it look
better to have the opional pars either before or after the context? That is
how it used to be, so I wonder what the reason for changing this is?

> i have another question.  why is context mandatory in local numbers?  i
> would like to make it optional and if it is missing, the context would
> be derived from the from uri.

I was going to ask the same, but forgot, so good that you brought it up :)
In previous versions of the draft the context is optional, so I also wonder
what the reason for making it mandatory is?

Regards,

Christer Holmberg
Ericsson Finland


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[This discussion should be on iptel, not sip.]

Yes, it's a bug which crept in due to other revisions. It will be fixed 
in 08.

jh@lohi.eng.song.fi wrote:
> looks to me that the syntax has a bug, i.e., local numbers don't need
> start with +.
> 
> i have another question.  why is context mandatory in local numbers?  i
> would like to make it optional and if it is missing, the context would
> be derived from the from uri.

How would it be uniquely and unambiguously derived?

> 
> -- juha

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To: Henning Schulzrinne <hgs@cs.columbia.edu>
Cc: Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org,
        antti.vaha-sipila@nokia.com, iptel@ietf.org
Subject: Re: [Sip] TEL-URL local numbers
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Henning Schulzrinne writes:

 > > i have another question.  why is context mandatory in local numbers?  i
 > > would like to make it optional and if it is missing, the context would
 > > be derived from the from uri.
 > 
 > How would it be uniquely and unambiguously derived?

if my from uri is +358-3-1234567 and i make a call to 03-2345678, the
user agent or proxy could figure out that the missing country code is
+358.  this is normal dial plan stuff.

-- juha


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CC: Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org,
        antti.vaha-sipila@nokia.com, iptel@ietf.org
Subject: Re: [Sip] TEL-URL local numbers
References: <3E27EF16.B7312454@lmf.ericsson.se>	<15911.63537.627963.460641@harjus.eng.song.fi>	<3E27FF63.4030601@cs.columbia.edu> <15912.432.180748.674189@harjus.eng.song.fi>
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> if my from uri is +358-3-1234567 and i make a call to 03-2345678, the
> user agent or proxy could figure out that the missing country code is
> +358.  this is normal dial plan stuff.

tel URIs, as has been discussed a few times, do no incorporate dial 
plans. They are *generated* by dial plans.

> 

> -- juha
> 

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Cc: Christer Holmberg <christer.holmberg@lmf.ericsson.se>, sip@ietf.org,
        antti.vaha-sipila@nokia.com, iptel@ietf.org
Subject: Re: [Sip] TEL-URL local numbers
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Henning Schulzrinne writes:

 > > if my from uri is +358-3-1234567 and i make a call to 03-2345678, the
 > > user agent or proxy could figure out that the missing country code is
 > > +358.  this is normal dial plan stuff.
 > 
 > tel URIs, as has been discussed a few times, do no incorporate dial 
 > plans. They are *generated* by dial plans.

if the tel uri must be always fully qualified then the ua must have a
dial plan and generate the fully qualified tel uri.  in some cases it
would be easier to let the proxy to run the dial plan code on the
message and fully qualify the tel uri.

-- juha


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] TEL-URL local numbers
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		<15911.63537.627963.460641@harjus.eng.song.fi>
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Hi,

>  >>if my from uri is +358-3-1234567 and i make a call to 03-2345678, the
>  >>user agent or proxy could figure out that the missing country code is
>  >>+358.  this is normal dial plan stuff.
>  >
>  >tel URIs, as has been discussed a few times, do no incorporate dial
>  >plans. They are *generated* by dial plans.
>
> if the tel uri must be always fully qualified then the ua must have a
> dial plan and generate the fully qualified tel uri.  in some cases it
> would be easier to let the proxy to run the dial plan code on the
> message and fully qualify the tel uri.

I agree with Juha. I don't think we have any requirements on having dial plans
in the SIP phones, do we? I know there is a discussion ongoing about it, but
still...

Regards,

Christer Holmberg
Ericsson Finland


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Subject: [Sip] any restrictions on characters in user part of sip uri?
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the syntax of user part of sip uri is:

user = 1*( unreserved / escaped / user-unreserved )
user-unreserved = "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"

escaped is %hexhex and can thus be used to include any character in
user.  is it really so that there is no restrictions on what chars may
be include in user using escaped?  for example, is %0000 valid in user?
if so, it makes implementations "interesting".

-- juha

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Hi,

by seeking a solution for a technical problem we should not consider philosophical approaches.

For the transport of SS7-signalling information the SIGTRAN protocols have been especially defined by the SIGTRAN
working group of the IETF. In this scenario SUA over SCTP could transport the TCAP information. For the time being PSTNs still exist and in converged scenarios we should use the same straight forward approach defined by the SIGTRAN group for each SS7 user part - why special treatment for TCAP???.

Why defining another approach when everythhing neccesary is already defined. Or are there doubts about the maturity in the SIGTRAN protocols?

You won´t need additional point codes and even the textbased TCAP-Endpoint in the IP cloud will cost some bucks.

Some questions on the proposed solution: 
-- How is adressing beeing resolved? 
-- TCAP dialoges are mainly used for call unrelated signalling, means no dialog exists. But one has to have an open SIP-Dialog before exchanging Info-Messages. How is that going to be resolved? 
-- For what kind of services this should be used? Is this solution anticipated for call completion services instead of usage of subscribe/notify?

Thanks

Joachim

-----Ursprüngliche Nachricht-----
Von: Frank W. Miller [mailto:fmiller@sentito.com]
Gesendet: Donnerstag, 16. Januar 2003 16:06
An: hbhondwe@hss.hns.com
Cc: sip@ietf.org
Betreff: [Sip] Re: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt :
Reposting



My apologies, I missed this one...

The difference here is that TCAP binary messages do not traverse the IP
portion of the network.  The impetus is similar to that for translating the
ISUP maintenance commands, SS7 stacks cost a lot (both $$ and point codes).
It makes more sense to implement a simple text representation that traverses
the IP portion of the network and is converted to binary TCAP at the gateway
to the SS7 cloud.  Also, from a philosophical point of view, since I believe
that SIP will supplant SS7 over the long run, limiting the need to implement
SS7 in new network elements is "a good thing".  This last point is of
course, just MHO.

FM


Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com


----- Original Message -----
From: <hbhondwe@hss.hns.com>
To: <sipping@ietf.org>
Cc: <fmiller@sentito.com>
Sent: Thursday, January 16, 2003 8:48 AM
Subject: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt : Reposting


>
>
> Reposting this mail as it did not receive a response on the list the first
> time.
>
>
>
>
>
> hbhondwe@hss.hns.com on 01/10/2003 11:13:02 AM
>
> To:   sipping@ietf.org
> cc:    (bcc: Harsh Bhondwe/HSSBLR)
>
> Subject:  [Sipping] Comments: I-D ACTION:draft-miller-sip-tcap-00.txt
>
>
>
>
>
>
> Hi ,
>
> I have a fundamental question on the the usage of SIP for
> transporting TCAP messages.
>
> If I understand correctly the figure 1 in the draft, is
> something like the following:
>
>
> SIP Element           Signaling Gateway         SS7 Node
>
> TCAP applications                               PSTN Services
> TCAP                                            TCAP
> SIP--------IP----------SIP||SS7--------SS7------SS7
>
>
> Here a TCAP peer will be running on the SIP Element and will
> use SIP to exchange TCAP messages with peers(i.e other TCAPs)
> in the PSTN network.
>
> However, SIGTRAN protocols defined also by the SIGTRAN
> Working group of the  IETF have been especially designed
> for transporting Trunk/Access signaling over IP.
>
> So, could one not better accomplish this  by the
> following configuration using SIGTRAN?
>
>
> SIP Element                   Signaling Gateway        PSTN Node
>
> TCAP applications                                   PSTN Services
> TCAP                                                   TCAP
> SIGTRAN---------IP-------------SIGTRAN||SS7 ---SS7-----SS7
> (SUA over SCTP)                (SUA over SCTP)
>
>
> The difference of course is that in this case the SIP
> Element and the Signaling Gateway will have to (also)
> support SIGTRAN.They could continue to support SIP as well.
>
> Or am i missing something?
>
> thanks
> harsh
>
>
>
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>
>
>
>
>

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To: "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>
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Thanks for the questions!

Comments inline.


> For the transport of SS7-signalling information the SIGTRAN protocols have
been especially defined by the SIGTRAN
> working group of the IETF. In this scenario SUA over SCTP could transport
the TCAP information. For the time being PSTNs still exist and in converged
scenarios we should use the same straight forward approach defined by the
SIGTRAN group for each SS7 user part - why special treatment for TCAP???.
>

We have a similar mechanism for the ISUP maintenance commands.  An early
version of that draft is on the IETF site and a major update that changes
the INFO body contents to an XML representation will be released within
days.  TCAP is not special in this regard.  IMHO, ISUP and TCAP are the best
place to do translations because the "semantic level" appears to me to be
the closest to where SIP session semantic level is.  These two works are
designed to augment the basic SIP-ISUP translation draft to include the
ability to control PSTN access trunks (e.g. PRI's and IMTs on T-1s) from the
PSTN and to do service delivery from existing PSTN elements (i.e. SCPs) to
SIP endpoints, both without having to implement SS7 protocol stacks in the
SIP elements.  The goal is make sharper the demarcation between SIP and SS7.
This is primarily a cost argument.  SS7 stacks are more expensive in the
marketplace than SIP stacks, so limiting the number of SS7 deployments saves
costs for the carriers.


> Why defining another approach when everythhing neccesary is already
defined. Or are there doubts about the maturity in the SIGTRAN protocols?
>

No, SIGTRAN is very mature and very useful for certain applications.  This
is just a different approach.  I see no reason why tunneling and translation
approaches cannot coexist in IETF standards.  Both have their applications.


> Some questions on the proposed solution:
> -- How is adressing beeing resolved?

One of the nice things about translation is that it limits addressing
problems to the mapping of addresses.  You can do about anything you want on
the SIP side as long as the mappings at the SIP UA and the GW are handled
appropriately.  Addressing is certainly an area that is ripe for discussion
in any standardization activity.


> -- TCAP dialoges are mainly used for call unrelated signalling, means no
dialog exists. But one has to have an open SIP-Dialog before exchanging
Info-Messages. How is that going to be resolved?

The element with the UA maintains a persistent session with the GW for TCAP
interactions.  I believe this is discussed in the draft.


> -- For what kind of services this should be used? Is this solution
anticipated for call completion services instead of usage of
subscribe/notify?
>

There are several service examples detailed in the draft.  Calling Name
Delivery is the one that leaps to mind.

FM


Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com

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Subject: RE: [Sip] any restrictions on characters in user part of sip uri?
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Juha and all:

Comment:

What about using escaped is %hexhex for support to unicode which can be used for languages such as Japanese.

Regards

-- Jean-Luc
-----Original Message-----
From: ext jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
Sent: Friday, January 17, 2003 5:29 AM
To: sip@ietf.org
Subject: [Sip] any restrictions on characters in user part of sip uri?


the syntax of user part of sip uri is:

user = 1*( unreserved / escaped / user-unreserved )
user-unreserved = "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"

escaped is %hexhex and can thus be used to include any character in
user.  is it really so that there is no restrictions on what chars may
be include in user using escaped?  for example, is %0000 valid in user?
if so, it makes implementations "interesting".

-- juha

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From: "Elisabetta Carrara (EAB)" <Elisabetta.Carrara@era.ericsson.se>
To: "'Steffen Fries'" <steffen.fries@siemens.com>, sip@ietf.org
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Subject: [Sip] RE: SRTP and MIKEY usage in SIP
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Hi Steffen
the public-key based methods are most suitable for
client-to-client communications. The pre-shared could
be for ex a good choice for client(s)-to-server 
communications.
So, for SIP the public-key based may be more suitable,
generally speaking.

Cheers,
/E

 



> -----Original Message-----
> From: Steffen Fries [mailto:steffen.fries@siemens.com]
> Sent: den 17 januari 2003 16:20
> To: sip@ietf.org
> Cc: Elisabetta.Carrara@era.ericsson.se
> Subject: SRTP and MIKEY usage in SIP 
> 
> 
> Hi,
> 
> I'm not quite sure, which working group fits best for this 
> question, thus I'm starting with the SIP WG ;-)
> 
> When SIP and SRTP are used in conjunction, an appropriate
> key management is necessary for SRTP. Within the MSEC WG 
> MIKEY has been defined, which is thought to be used (also) 
> for SRTP. 
> 
> MIKEY and an related draft (draft-ietf-msec-MIKEY-DHHMAC-
> 01.txt) offer 4 different key management methods based on:
> - pre-shared secrets
> - public key encryption
> - Diffie Hellman protected with signatures
> - Diffie Hellman protected with pre-shared secrets
> 
> When MIKEY and SRTP are to be used in a SIP environment, 
> what would be a suitable choice out of the four options?
> 
> I'm not sure if the pre-shared secret pased methods are 
> suitable, since this would assume, that all users who want to 
> communicate need to exchange a shared secret before. Well, this 
> could be done by puting a shared secret in the SIP message and 
> securing this by S/MIME, but then MIKEY would be protected by 
> symmetric methods, although asymmetric technology was used to 
> secure the shared secret transport, namely S/MIME. One could 
> also use the certificates and private keys to secure MIKEY 
> right from the beginning. 
> 
> There might be scenarios where the symmetric case is 
> appropriate, but I'm not sure if this is a rather general case.
> 
> Is MIKEY generally considered for key management in SIP or will 
> this rather be done using draft-baugher-mmusic-sdpmediasec-
> 00.txt secured by S/MIME?
> 
> The usage of MIKEY and SRTP is especially interesting in 
> conjunction with other multimedia protocols like H.323. Voice 
> encryption and associated key management could be performed 
> across the different signaling protocols.
> 
> Was there already a discussion related to this question, which 
> I may be missed? 
> 
> Regards
>         Steffen
> 
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Subject: [Sip] SRTP and MIKEY usage in SIP
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Hi,

I'm not quite sure, which working group fits best for this 
question, thus I'm starting with the SIP WG ;-)

When SIP and SRTP are used in conjunction, an appropriate
key management is necessary for SRTP. Within the MSEC WG 
MIKEY has been defined, which is thought to be used (also) 
for SRTP. 

MIKEY and an related draft (draft-ietf-msec-MIKEY-DHHMAC-
01.txt) offer 4 different key management methods based on:
- pre-shared secrets
- public key encryption
- Diffie Hellman protected with signatures
- Diffie Hellman protected with pre-shared secrets

When MIKEY and SRTP are to be used in a SIP environment, 
what would be a suitable choice out of the four options?

I'm not sure if the pre-shared secret pased methods are 
suitable, since this would assume, that all users who want to 
communicate need to exchange a shared secret before. Well, this 
could be done by puting a shared secret in the SIP message and 
securing this by S/MIME, but then MIKEY would be protected by 
symmetric methods, although asymmetric technology was used to 
secure the shared secret transport, namely S/MIME. One could 
also use the certificates and private keys to secure MIKEY 
right from the beginning. 

There might be scenarios where the symmetric case is 
appropriate, but I'm not sure if this is a rather general case.

Is MIKEY generally considered for key management in SIP or will 
this rather be done using draft-baugher-mmusic-sdpmediasec-
00.txt secured by S/MIME?

The usage of MIKEY and SRTP is especially interesting in 
conjunction with other multimedia protocols like H.323. Voice 
encryption and associated key management could be performed 
across the different signaling protocols.

Was there already a discussion related to this question, which 
I may be missed? 

Regards
        Steffen

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Hi,
Don't forget to register with the 12th SIPit event.
The registration time has been extended until January 24th
to give the last minute planners some more time.

Everyone implementing products that are up to date
with the most recent SIMPLE work should consider to participate since
there will be breakout sessions 
testing against these specifications to let interoperability become a
reality also among presence and IM systems!

Registration and other useful information is found at:
http://www.hotsip.com/sipit12/

If you have questions, contact sipit12@hotsip.com

Welcome!
The SIPit 12 team.
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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Hi,
Don't forget to register with the 12th SIPit event.
The registration time has been extended until January 24th
to give the last minute planners some more time.

Everyone implementing products that are up to date
with the most recent SIMPLE work should consider to participate since
there will be breakout sessions 
testing against these specifications to let interoperability become a
reality also among presence and IM systems!

Registration and other useful information is found at:
http://www.hotsip.com/sipit12/

If you have questions, contact sipit12@hotsip.com

Welcome!
The SIPit 12 team.
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From mailnull@www1.ietf.org  Mon Jan 20 12:53:56 2003
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Hi all,

about SIP and Congetion Safety, i've observed there isn't a great interest.

I don't know why.
I think that will be a giant problem. 

And the mainly question for it is: what protocol use for transport SIP signals ???


others wg in IETF are working about congestion control algorithms for real-time 
or, more  generally,  delay sensitive applications... 
see DCCP (Datagram Congestion Control Protocol  -  http://www.ietf.org/html.charters/dccp-charter.html).

sometime ago there was a proposal about  SCTP use for Instant Messaging with SIP...

in other mail i read a proposal about Connection Transport Protocol as default protocol for SIP

in the "Session Initiation Protocol Extension to Assure Congestion Safety" draft-ietf-sip-congestsafe-00
there's only a proposal for define and check a CONGESTION-SAFE hop... but it isn't enough 


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



From mailnull@www1.ietf.org  Mon Jan 20 14:22:05 2003
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From: Amit Bhadoria <bhadoria@cisco.com>
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Subject: [Sip] TLS and "sip:" scheme
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hi,

on page 152 section 19.1.2, rfc3261 says that default port for
"sip:" using TLS over TCP is 5061. does this mean that a url with
"sip:" scheme and transport=tls is a valid url, i.e., is the following url
admissible:

	sip:bhadoria@cisco.com;transport=tls

i understand that use of transport=tls has been deprecated, but due to the
above mentioned guidelines in rfc, do we still need to accept urls with
sip: scheme and transport=tls?

thanks.

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



From mailnull@www1.ietf.org  Tue Jan 21 04:02:36 2003
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Subject: RE: [Sip] SIP and Congestion Safety
Date: Tue, 21 Jan 2003 10:59:06 +0200
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Please take a look at

http://www.ietf.org/internet-drafts/draft-khartabil-sip-congestionsafe-ci-01.txt

I'm planning to update this to include a UA posting (eg: contents for PUBLISH or MESSAGE).

Regards,
Hisham

> -----Original Message-----
> From: ext loretosa@vizzavi.it [mailto:loretosa@vizzavi.it]
> Sent: Monday, January 20, 2003 7:55 PM
> To: sip@ietf.org
> Subject: [Sip] SIP and Congestion Safety
> 
> 
> 
> Hi all,
> 
> about SIP and Congetion Safety, i've observed there isn't a 
> great interest.
> 
> I don't know why.
> I think that will be a giant problem. 
> 
> And the mainly question for it is: what protocol use for 
> transport SIP signals ???
> 
> 
> others wg in IETF are working about congestion control 
> algorithms for real-time 
> or, more  generally,  delay sensitive applications... 
> see DCCP (Datagram Congestion Control Protocol  -  
http://www.ietf.org/html.charters/dccp-charter.html).

sometime ago there was a proposal about  SCTP use for Instant Messaging with SIP...

in other mail i read a proposal about Connection Transport Protocol as default protocol for SIP

in the "Session Initiation Protocol Extension to Assure Congestion Safety" draft-ietf-sip-congestsafe-00
there's only a proposal for define and check a CONGESTION-SAFE hop... but it isn't enough 


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



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Roberto,

It means that you can challenge any request (but CANCEL) in order to get
credentials for the UAC.

Gonzalo

Roberto Tealdi wrote:
> 
> Hi to All
> 
> I'm new in SIP and I have a question:
> 
> In draft "draft-ietf-sipping-aaa-req-01" is specified that the Home
> AAA server must be able to authenticate a user based on any SIP
> Request (except CANCEL): this means that ANY SIP request message (from
> client to sip proxy) must contains a user's credentials, for example
> an authentication token?
> 
> Thanks a lot.
> Roby.
> 

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Amit Bhadoria'" <bhadoria@cisco.com>, sip@ietf.org
Subject: RE: [Sip] TLS and "sip:" scheme
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For backwards compatibility, it might be worth supporting that construction,
sure. But that URL is not considered RFC3261 compliant - RFC3261
implementations shouldn't generate it, anyway. The section in 19.1.2, I
suspect, refers to sending a request for a SIP URI scheme (and no
;transport=tls parameter) over TLS, which is permissable - there's no
requirement to append ;transport=tls to the Request-URI when you send a SIP
request over TLS.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Amit Bhadoria [mailto:bhadoria@cisco.com]
> Sent: Monday, January 20, 2003 11:22 AM
> To: sip@ietf.org
> Subject: [Sip] TLS and "sip:" scheme
> 
> 
> hi,
> 
> on page 152 section 19.1.2, rfc3261 says that default port for
> "sip:" using TLS over TCP is 5061. does this mean that a url with
> "sip:" scheme and transport=tls is a valid url, i.e., is the 
> following url
> admissible:
> 
> 	sip:bhadoria@cisco.com;transport=tls
> 
> i understand that use of transport=tls has been deprecated, but due to the
> above mentioned guidelines in rfc, do we still need to accept urls with
> sip: scheme and transport=tls?
> 
> thanks.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: [Sip] IEEE JSAC - All-IP WIRELESS NETWORKS , CALL FOR PAPERS
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			CALL FOR PAPERS

	   IEEE Journal on Selected Areas in Communications
		       ALL-IP WIRELESS NETWORKS

IP  (Internet Protocol),  which is  already a  universal network-layer
protocol  for  wireline  packet  networks,  is  becoming  a  promising
universal  network-layer  protocol   over  all  wireless  systems.  IP
provides a  globally successful  open infrastructure for  creating and
providing services  and applications. An all-IP  wireless and wireline
network could  make wireless networks more robust,  scalable, and cost
effective. It will also  enable the abundant applications and software
technologies  developed  for IP  networks  to  be  used over  wireless
networks. Today's  many different wireless systems,  ranging from PANs
(Personal  Area  Networks), wireless  LANs  (Local  Area Networks)  to
wide-area cellular  systems, are often not compatible  with each other
which makes it  difficult for a user to roam from  one radio system to
another. No wireless technology has  emerged as a common and long-term
universal solution. With  IP as the common network  layer protocol, an
IP-based  mobile device  (with multiple  radio interfaces  or software
defined radio) could roam between different wireless systems.

The papers  in this issue  will focus on state-of-the-art  research in
all-IP wireless networks. Submission  for Beyond 3G (B3G) systems will
be included. We solicit papers covering a variety of topics including,
but not limited to, the following topics:

  o   Systems and Architecture 
  o   Access Protocols 
  o   Transport Protocols 
  o   Signaling Protocols 
  o   Resource Management 
  o   Power Control and Management 
  o   QoS Provisioning 
  o   Security 
  o   Mobile VoIP 
  o   Video over Wireless Internet 
  o   Network Performance Analysis 
  o   Internetworking 
  o   System Integration 

Only  original and  unpublished  research papers  will be  considered.
Authors should  follow the IEEE  J-SAC manuscript format  described in
the Information for  Authors on the inside back cover  of any issue of
J-SAC.   There will be  one round  of reviews  and acceptance  will be
limited to papers needing only moderate revisions. Prospective authors
should submit a pdf version of their complete manuscript (which should
be  compressed  if   the  file  size  exceeds  1   Mbyte)  online  via
http://wire.cs.nthu.edu.tw/~jsac according to the following timetable:

      Manuscript Submission:   FEBRUARY 1, 2003
      Acceptance Notification: June 1, 2003
      Final Manuscript Due:    September 1, 2003
      Publication:             1st Quarter 2004


      Prathima Agrawal
      Telcordia Technologies
      Room 1J244B
      445 South St
      Morristown, NJ 07960 USA

      Tao Zhang
      Telcordia Technologies
      Room 1J214B
      445 South St
      Morristown, NJ 07960 USA

      Cormac J. Sreenan
      Dept of Computer Science
      University College Cork
      Cork, Ireland

      Jyh-Cheng Chen
      Dept of Computer Science and
      Inst of Communications Engineering
      National Tsing Hua Univ
      Hsinchu, Taiwan


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From mailnull@www1.ietf.org  Wed Jan 22 03:34:54 2003
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Hi Hisham,

I've read your draft "draft-khartabil-sip-congestionsafe-ci-01.txt"
It's an interesting draft, but not cover all my questions.

In any case about your draft i've some observations:

in the section -5.0-
You write that the proxy affect for Congestion Safe in 3GPP architecture is S-CSCF.
I think that P-CSCF is the most critical Proxy.
This one is the closest proxy to the UA. And in a Visited UA
scenario, I think that it's preferable, for an Operator 3G, not forward information about his link capacity to another Operator 3G.
Morover I think that are also performance problem for which P-CSCF have know information about Max-Size SIP message that UAS can handle.

In the section 5.5. "Home Proxy Receiving 413 Response from UA"
What's advantage  for a UA to reject a request, also if is a large SIP request, when it's already received? When arrived to UA, the large message already has congestioned the link.

The main problem is avoided and prevent the congestion link.





Sal


> Please take a look at
> 
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-congestionsafe-ci-01.txt
> 
> I'm planning to update this to include a UA posting (eg: contents for PUBLISH or MESSAGE).
> 
> Regards,
> Hisham
> 
> > -----Original Message-----
> > From: ext loretosa@vizzavi.it [mailto:loretosa@vizzavi.it]
> > Sent: Monday, January 20, 2003 7:55 PM
> > To: sip@ietf.org
> > Subject: [Sip] SIP and Congestion Safety
> > 
> > 
> > 
> > Hi all,
> > 
> > about SIP and Congetion Safety, i've observed there isn't a 
> > great interest.
> > 
> > I don't know why.
> > I think that will be a giant problem. 
> > 
> > And the mainly question for it is: what protocol use for 
> > transport SIP signals ???
> > 
> > 
> > others wg in IETF are working about congestion control 
> > algorithms for real-time 
> > or, more  generally,  delay sensitive applications... 
> > see DCCP (Datagram Congestion Control Protocol  -  
> http://www.ietf.org/html.charters/dccp-charter.html).
> 
> sometime ago there was a proposal about  SCTP use for Instant Messaging with SIP...
> 
> in other mail i read a proposal about Connection Transport Protocol as default protocol for SIP
> 
> in the "Session Initiation Protocol Extension to Assure Congestion Safety" draft-ietf-sip-congestsafe-00
> there's only a proposal for define and check a CONGESTION-SAFE hop... but it isn't enough 
> 
> 
> Sal
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

 Per registrarti gratuitamente a Vodafone Mail vai su www.190.it 


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From: "Vishal Verma" <vishal@sonim-india.com>
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Hi ,
  what are the compact header notations for Max-Forward,CSeq fields ?.Is
there any draft available which lists down sip fields vs their compact
header notations ?

regards, vishal

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Subject: RE: RE: [Sip] SIP and Congestion Safety
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> -----Original Message-----
> From: ext loretosa@vodafone.it [mailto:loretosa@vodafone.it]
> Sent: Wednesday, January 22, 2003 10:31 AM
> To: Khartabil Hisham (NMP/Helsinki); sip@ietf.org
> Subject: Ri: RE: [Sip] SIP and Congestion Safety
> 
> 
> 
> Hi Hisham,
> 
> I've read your draft "draft-khartabil-sip-congestionsafe-ci-01.txt"
> It's an interesting draft, but not cover all my questions.
> 
> In any case about your draft i've some observations:
> 
> in the section -5.0-
> You write that the proxy affect for Congestion Safe in 3GPP 
> architecture is S-CSCF.
> I think that P-CSCF is the most critical Proxy.
> This one is the closest proxy to the UA. And in a Visited UA
> scenario, I think that it's preferable, for an Operator 3G, 
> not forward information about his link capacity to another 
> Operator 3G.

The document does say "the S-CSCF closest to the UA". I think the use of the word "home" confused you.

> Morover I think that are also performance problem for which 
> P-CSCF have know information about Max-Size SIP message that 
> UAS can handle.

Perhaps, but to my knowledge, S-CSCFs have the intelligence. P-CSCFs are nothing but outbound proxies to the sending UA.

In any case, where 3GPP decides to put the intelligence is out of the scope of this document. The sentence you refer to is just an example (and perhaps should be removed).

> 
> In the section 5.5. "Home Proxy Receiving 413 Response from UA"
> What's advantage  for a UA to reject a request, also if is a 
> large SIP request, when it's already received? When arrived 
> to UA, the large message already has congestioned the link.
> 
> The main problem is avoided and prevent the congestion link.

The solution is not just limited to the congestion of the network, it also tries to handle out-of-memory situations for a terminal, even when using TCP. If a terminal is receiving TCP packets for a sip message and knows it can't handle anymore packets, it can stop receiving and send 413 error response.

Regards,
Hisham

> 
> 
> 
> 
> 
> Sal
> 
> 
> > Please take a look at
> > 
> > 
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-conges
> tionsafe-ci-01.txt
> > 
> > I'm planning to update this to include a UA posting (eg: 
> contents for PUBLISH or MESSAGE).
> > 
> > Regards,
> > Hisham
> > 
> > > -----Original Message-----
> > > From: ext loretosa@vizzavi.it [mailto:loretosa@vizzavi.it]
> > > Sent: Monday, January 20, 2003 7:55 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] SIP and Congestion Safety
> > > 
> > > 
> > > 
> > > Hi all,
> > > 
> > > about SIP and Congetion Safety, i've observed there isn't a 
> > > great interest.
> > > 
> > > I don't know why.
> > > I think that will be a giant problem. 
> > > 
> > > And the mainly question for it is: what protocol use for 
> > > transport SIP signals ???
> > > 
> > > 
> > > others wg in IETF are working about congestion control 
> > > algorithms for real-time 
> > > or, more  generally,  delay sensitive applications... 
> > > see DCCP (Datagram Congestion Control Protocol  -  
> > http://www.ietf.org/html.charters/dccp-charter.html).
> > 
> > sometime ago there was a proposal about  SCTP use for 
> Instant Messaging with SIP...
> > 
> > in other mail i read a proposal about Connection Transport 
> Protocol as default protocol for SIP
> > 
> > in the "Session Initiation Protocol Extension to Assure 
> Congestion Safety" draft-ietf-sip-congestsafe-00
> > there's only a proposal for define and check a 
> CONGESTION-SAFE hop... but it isn't enough 
> > 
> > 
> > Sal
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
>  Per registrarti gratuitamente a Vodafone Mail vai su www.190.it 
> 
> 
> 
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From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] query for compact headers
Date: Wed, 22 Jan 2003 10:22:15 -0500
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> what are the compact header notations for 
> Max-Forward, CSeq fields ?.

Neither has a compact form.  The CSeq name 
was considered small enough.  The Max-Forwards 
was not a required header within rfc 2543; 
thus I guess that it was not considered common 
enough for a compact form.  The compact form
was not added to rfc 3261 since it would not
be backward compatible with rfc 2543.

> Is there any draft available which lists 
> down sip fields vs their compact header 
> notations ?

The compact forms are listed with the
header definition within the associated 
rfc's.

The following link consolidates the definitions
from various drafts and rfc's.

http://www.iana.org/assignments/sip-parameters

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From: "Seonman Kim" <seonman@funtv.com>
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Subject: [Sip] Multiple SDP descriptions for conferencing
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Hi,

I am trying to implement a SIP-based multi-point conferencing based on
"Centralized Signaling, Distributed Media" model
(draft-ietf-sipping-conferencing-models-01.txt). In this model, each
participant  UA gets updated SDP descriptions using "re-INVITE" whenever a
user enters to (or exits from) the conference. If the number of participants
becomes more than two,
then the re-INVITE message should contain two or more SDP descriptions.
For example, if user A, B and C are in the conference, then a re-INVITE to
user A should contain two SDP description, one for user B and the other for
user C.

My question here is: how to describe multiple SDP descriptions in the
content body
of INVITE (or 200 OK) message? Do we just concatenate them?
For example, if the conference server sends re-INVITE to user A with SDPs of
user B and C, the content body will look like this:

v=0
o=user-b 0 0 IN IP4 192.168.1.100
s=session
c=IN IP4 192.168.1.100
b=CT:1000
t=0 0
m=audio 30001 RTP/AVP 0
a=rtpmap:0 PCMU/8000
v=0
o=user-c 0 0 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
b=CT:1000
t=0 0
m=audio 30002 RTP/AVP 0
a=rtpmap:0 PCMU/8000

Is this correct implementation for multiple SDP sessions?
I could not find any descriptions in RTF 3261 that say UA should handle
multiple SDP sessions.

Thanks,
Seonman

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From: "Roberto Tealdi" <roby.tea@tin.it>
To: <sip@ietf.org>
Subject: [SIP] JAIN SIP open source user agents
Date: Thu, 23 Jan 2003 18:15:41 +0100
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Hi All,

does anyone indicate me a free open source user agent implementation =
based on JAIN SIP?

Thanks a lot.

Roby.

------=_NextPart_000_0061_01C2C30B.702D0390
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi All,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>does anyone indicate me a free open =
source user=20
agent implementation based on JAIN SIP?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks a lot.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Roby.</FONT></DIV></BODY></HTML>

------=_NextPart_000_0061_01C2C30B.702D0390--


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Subject: [Sip] RFC 3313 on Private Session Initiation Protocol (SIP) Extensions for Media Authorization
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--NextPart


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


        RFC 3313

        Title:      Private Session Initiation Protocol (SIP)
                    Extensions for Media Authorization
        Author(s):  W. Marshall, Ed.
        Status:     Informational
        Date:       January 2003
        Mailbox:    wtm@research.att.com
        Pages:      16
        Characters: 36866
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-sip-call-auth-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3313.txt


This document describes the need for Quality of Service (QoS) and
media authorization and defines a Session Initiation Protocol (SIP)
extension that can be used to integrate QoS admission control with
call signaling and help guard against denial of service attacks.  The
use of this extension is only applicable in administrative domains, or
among federations of administrative domains with previously
agreed-upon policies, where both the SIP proxy authorizing the QoS,
and the policy control of the underlying network providing the QoS,
belong to that administrative domain or federation of domains.

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

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USC/Information Sciences Institute

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From mailnull@www1.ietf.org  Thu Jan 23 17:05:57 2003
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Date: Thu, 23 Jan 2003 15:59:52 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [Sip] Multiple SDP descriptions for conferencing
In-reply-to: <01b201c2c25f$473f5610$400015ac@skim>
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To: Seonman Kim <seonman@funtv.com>, sip@ietf.org
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Hi Seonman,

You will find more updated work on SIP conferencing with the SIPPING 
Conferencing Design Team:

      http://www.softarmor.com/sipping/teams/conf/

The framework document includes a discussion of the this model:

      http://www.ietf.org/internet-drafts/draft-rosenberg-sipping-conferencing-framework-00.txt

Instead of multiple SDP message bodies, the re-INVITE would add an extra 
media line for each participant that enters the conference.  See below for 
an example.

For other valid offer/answer examples of SDP, see:

      http://www.ietf.org/internet-drafts/draft-johnston-mmusic-offer-answer-examples-00.txt

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com


At 01:43 PM 1/22/2003 -0800, Seonman Kim wrote:
>Hi,
>
>I am trying to implement a SIP-based multi-point conferencing based on
>"Centralized Signaling, Distributed Media" model
>(draft-ietf-sipping-conferencing-models-01.txt). In this model, each
>participant  UA gets updated SDP descriptions using "re-INVITE" whenever a
>user enters to (or exits from) the conference. If the number of participants
>becomes more than two,
>then the re-INVITE message should contain two or more SDP descriptions.
>For example, if user A, B and C are in the conference, then a re-INVITE to
>user A should contain two SDP description, one for user B and the other for
>user C.
>
>My question here is: how to describe multiple SDP descriptions in the
>content body
>of INVITE (or 200 OK) message? Do we just concatenate them?
>For example, if the conference server sends re-INVITE to user A with SDPs of
>user B and C, the content body will look like this:
>
>v=0
>o=user-b 0 0 IN IP4 192.168.1.100
>s=session
>c=IN IP4 192.168.1.100
>b=CT:1000
>t=0 0
>m=audio 30001 RTP/AVP 0
>a=rtpmap:0 PCMU/8000
>v=0
>o=user-c 0 0 IN IP4 192.168.1.101
>s=session
>c=IN IP4 192.168.1.101
>b=CT:1000
>t=0 0
>m=audio 30002 RTP/AVP 0
>a=rtpmap:0 PCMU/8000
>
>Is this correct implementation for multiple SDP sessions?

No, I think it would look like this:

v=0
o=user-b 0 0 IN IP4 192.168.1.100
s=session
b=CT:1000
t=0 0
m=audio 30001 RTP/AVP 0
c=IN IP4 192.168.1.100
a=rtpmap:0 PCMU/8000
m=audio 30002 RTP/AVP 0
c=IN IP4 192.168.1.101
a=rtpmap:0 PCMU/8000

>I could not find any descriptions in RTF 3261 that say UA should handle
>multiple SDP sessions.
>
>Thanks,
>Seonman
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Date: Thu, 23 Jan 2003 20:23:23 -0600
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Subject: [Sip] FW: SIP patent infringement warning
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FYI

-----Original Message-----
From: pierobon [mailto:pierobon@sympatico.ca] 
Sent: Thursday, January 23, 2003 7:06 PM
To: dean.willis@softarmor.com; rohan@cisco.com; jon.peterson@neustar.biz
Cc: sob@harvard.edu; mankin@psg.com; dromasca@avaya.com; sip@ietf.org
Subject: SIP patent infringement warning 


Dear Sirs,

Having examined your specs and use of SIP, I'm must point out that your 
company may be infringing on my patent rights.

You can verify for yourself by examining the patent at 
http://massive-array-cellular-system.com

Thanks,

Victor Pierobon
416 221-2340
Pierobon@attcanada.ca


  To:

Dean Willis
Rohan Mahy
Jon Peterson
Scott Bradner
Allison Mankin
Allison Mankin
Dan Romascanu
  



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From: Gao Qing <qing.gao@siemens.com>
To: sip@ietf.org
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------_=_NextPart_001_01C2C38D.EE69A060
Content-Type: text/plain;
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Dear Sirs,

Who can explain me what exactly is "target set" in RFC3261? Is there a
definition of it somewhere?

Thanks in advance!

Gao Qing

------_=_NextPart_001_01C2C38D.EE69A060
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<P><FONT SIZE=3D2 FACE=3D"Courier New">Dear Sirs,<BR>
<BR>
</FONT><FONT SIZE=3D2 FACE=3D"Courier New">Who can explain me what =
exactly is &quot;target set&quot; in RFC3261? Is there a definition of =
it somewhere?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Thanks in advance!</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Gao Qing</FONT>
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_______________________________________________
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From: "Roberto Tealdi" <roby.tea@tin.it>
To: "Tom Scott" <telecomtom@vedatel.com>
Cc: <sip@ietf.org>
References: <3E307E20.50001@vedatel.com>
Subject: Re: [SIP] JAIN SIP open source user agents
Date: Fri, 24 Jan 2003 11:12:58 +0100
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Try http://www.siptrex.net/downloads/

Roberto.


----- Original Message ----- 
From: "Tom Scott" <telecomtom@vedatel.com>
To: "Roberto Tealdi" <roby.tea@tin.it>
Sent: Friday, January 24, 2003 12:43 AM
Subject: [Fwd: [SIP] JAIN SIP open source user agents]


> Roberto,
> 
> If you find an free open source UA based on SIP JAIN, please post
> the location on the list or send to me.
> 
> -- TIA, TT
> 
> 
> -------- Original Message --------
> Subject: [SIP] JAIN SIP open source user agents
> Date: Thu, 23 Jan 2003 18:15:41 +0100
> From: "Roberto Tealdi" <roby.tea@tin.it>
> To: <sip@ietf.org>
> 
> 
> 
> Hi All,
> 
> does anyone indicate me a free open source user agent implementation
> based on JAIN SIP?
> 
> Thanks a lot.
> 
> Roby.
> 
> 


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From: "Nataraju A.B.(huawei)" <natarajuab@huawei.com>
Subject: Re: [Sip] target set
To: Gao Qing <qing.gao@siemens.com>, sip@ietf.org
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target setHi, 
I guess the target set (contact addressses) means the set of URI's returned by the redirect server.
these are the request uri for the next request send by teh UAC to reach the intended user.
Regards,
-------------------------------------------
Nataraju A.B.
Huawei Technologies India Pvt. Ltd.,
Tel : +91-80-5217152 / 4 Xtn 142
-------------------------------------------
  ----- Original Message ----- 
  From: Gao Qing 
  To: sip@ietf.org 
  Sent: Friday, January 24, 2003 3-19
  Subject: [Sip] target set


  Dear Sirs,

  Who can explain me what exactly is "target set" in RFC3261? Is there a definition of it somewhere? 

  Thanks in advance! 

  Gao Qing 


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<DIV><FONT color=#008000>Hi, </FONT></DIV>
<DIV><FONT color=#008000>I guess the target set (contact addressses) means the 
set of URI's returned by the redirect server.</FONT></DIV>
<DIV><FONT color=#008000>these are the request uri for the next request send by 
teh UAC to reach the intended user.</FONT></DIV>
<DIV><FONT color=#008000>Regards,</FONT></DIV>
<DIV>-------------------------------------------<BR>Nataraju A.B.<BR>Huawei 
Technologies India Pvt. Ltd.,<BR>Tel : +91-80-5217152 / 4 Xtn 
142<BR>-------------------------------------------</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #008000 2px solid; MARGIN-RIGHT: 0px">
  <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=qing.gao@siemens.com href="mailto:qing.gao@siemens.com">Gao Qing</A> 
  </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=sip@ietf.org 
  href="mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, January 24, 2003 3-19</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> [Sip] target set</DIV>
  <DIV><BR></DIV>
  <P><FONT face="Courier New" size=2>Dear Sirs,<BR><BR></FONT><FONT 
  face="Courier New" size=2>Who can explain me what exactly is "target set" in 
  RFC3261? Is there a definition of it somewhere?</FONT> </P>
  <P><FONT face="Courier New" size=2>Thanks in advance!</FONT> </P>
  <P><FONT face="Courier New" size=2>Gao Qing</FONT> 
</P></BLOCKQUOTE></BODY></HTML>

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From: "Ranjit Kumar Avasarala" <ranjit.avasarala@bplmail.com>
To: "Gao Qing" <qing.gao@siemens.com>, <sip@ietf.org>
Subject: RE: [Sip] target set
Date: Fri, 24 Jan 2003 17:06:35 +0530
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target setHI

'target set' is a set of uris which the client sets to try incase it
receives a 3xx response.
so to start with the client will have only one uri in the target set ile the
target it intends to reach.

now say it receives 3xx (300 Moved) response then it receives a new URI in
the contact header. so it can place this URI in its target set and send the
request to that URI.

so this way the target set grows.

-Ranjit
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Gao Qing
  Sent: Friday, January 24, 2003 3:20 PM
  To: sip@ietf.org
  Subject: [Sip] target set


  Dear Sirs,

  Who can explain me what exactly is "target set" in RFC3261? Is there a
definition of it somewhere?

  Thanks in advance!

  Gao Qing


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>target set</TITLE>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003>HI</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003>'target set' is a set of uris which the =
client sets to=20
try incase it receives a 3xx response. </SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D156573211-24012003>so to=20
start with the client will have only one uri in the target set ile the =
target it=20
intends to reach.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D156573211-24012003>now=20
say it receives 3xx (300 Moved) response then it receives a new URI in =
the=20
contact header. so it can place this URI in its target set and send the =
request=20
to that URI.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D156573211-24012003>so=20
this way the target set grows.</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D156573211-24012003>-Ranjit</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> sip-admin@ietf.org =

  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Gao =
Qing<BR><B>Sent:</B>=20
  Friday, January 24, 2003 3:20 PM<BR><B>To:</B> =
sip@ietf.org<BR><B>Subject:</B>=20
  [Sip] target set<BR><BR></DIV></FONT>
  <P><FONT face=3D"Courier New" size=3D2>Dear Sirs,<BR><BR></FONT><FONT=20
  face=3D"Courier New" size=3D2>Who can explain me what exactly is =
"target set" in=20
  RFC3261? Is there a definition of it somewhere?</FONT> </P>
  <P><FONT face=3D"Courier New" size=3D2>Thanks in advance!</FONT> </P>
  <P><FONT face=3D"Courier New" size=3D2>Gao Qing</FONT>=20
</P></BLOCKQUOTE></BODY></HTML>

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From: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>
To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Date: Fri, 24 Jan 2003 11:56:48 +0000
Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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At 1:11 pm +0200 24/1/03, Christer Holmberg wrote:
>Hi,
>
>>  At 3:23 pm +0200 17/1/03, jh@lohi.eng.song.fi wrote:
>>  >Henning Schulzrinne writes:
>>  >
>>  >  > > if my from uri is +358-3-1234567 and i make a call to 03-2345678, the
>>  >  > > user agent or proxy could figure out that the missing country code is
>>  >  > > +358.  this is normal dial plan stuff.
>>  >  >
>>  >  > tel URIs, as has been discussed a few times, do no incorporate dial
>>  >  > plans. They are *generated* by dial plans.
>>  >
>>  >if the tel uri must be always fully qualified then the ua must have a
>>  >dial plan and generate the fully qualified tel uri.  in some cases it
>>  >would be easier to let the proxy to run the dial plan code on the
>>  >message and fully qualify the tel uri.
>>  >
>>  >-- juha
>  >[Lawrence said:]
>  > Hi Folks,
>>    ...sigh...
>>
>>  If I receive a tel URL containing a local number, without
>>  and idea of the context *in the place that the URL was constructed*
>>  then I don't know whether or not I'm in that context.
>
>[CHH] In this case you could reject it. If the sender didn't insert it, and
>there was no intermediate node inserting it, there is a network architecture
>problem.
>
>My point, I guess Juha's too, was not that there should not be a context when
>the URL reaches the UAS, but that it MAY be inserted by another node than the
>UAC.
>
<snip>
>I agree on that. However, my SIP phone may not know to which context they
>belong, for whatever reason, but the outbound proxy they communicate with DO
>know.

To which I reply::

Aha! Thanks for the clarification - I'm being slow today.
   This is a pure SIP issue - usage of tel URLs in SIP, and request
modification by intermediate nodes. Interesting issue.

However, IMHO there are other uses for tel URLs, so I'd suggest that
draft-antti-rfc2916bis is correct in mandating this behaviour.

If there's a separate draft for "SIP usage of tel URLs" that qualifies
the main tel URL RFC, relaxing the requirement, that's fine; I'm not sure
if it's a SIP or SIPPING job, but I think that's where it lives.

I just don't want to nail this down in the general 2916 update.

all the best,
    Lawrence
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Date: Fri, 24 Jan 2003 18:09:32 +0530
From: "Nataraju A.B.(huawei)" <natarajuab@huawei.com>
Subject: Re: [Sip] target set
To: ranjit.avasarala@bplmail.com, Gao Qing <qing.gao@siemens.com>,
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target set----- Original Message ----- 
  From: Ranjit Kumar Avasarala 
  To: Gao Qing ; sip@ietf.org 
  Sent: Friday, January 24, 2003 5-06
  Subject: RE: [Sip] target set


  HI

  'target set' is a set of uris which the client sets to try incase it receives a 3xx response. 
  so to start with the client will have only one uri in the target set ile the target it intends to reach.

  This sentance is quiet confusing. In the initial request what the UAC tries out is not the target set (there is only one URI). its the URI which UAC intends to reach out. the target set (contact addresses in 3xx response ) implies the URI's where the target user could be reached. these addresses can be contacted in either linear or parallel search technique.

  now say it receives 3xx (300 Moved) response then it receives a new URI in the contact header. so it can place this URI in its target set and send the request to that URI.

  so this way the target set grows.

  -Ranjit
    -----Original Message-----
    From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Gao Qing
    Sent: Friday, January 24, 2003 3:20 PM
    To: sip@ietf.org
    Subject: [Sip] target set


    Dear Sirs,

    Who can explain me what exactly is "target set" in RFC3261? Is there a definition of it somewhere? 

    Thanks in advance! 

    Gao Qing 

    Regards,
    -------------------------------------------
    Nataraju A.B.
    Huawei Technologies India Pvt. Ltd.,
    Tel : +91-80-5217152 / 4 Xtn 142
    -------------------------------------------

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  <DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV 
  style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B> 
  <A title=ranjit.avasarala@bplmail.com 
  href="mailto:ranjit.avasarala@bplmail.com">Ranjit Kumar Avasarala</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>To:</B> <A title=qing.gao@siemens.com 
  href="mailto:qing.gao@siemens.com">Gao Qing</A> ; <A title=sip@ietf.org 
  href="mailto:sip@ietf.org">sip@ietf.org</A> </DIV>
  <DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, January 24, 2003 5-06</DIV>
  <DIV style="FONT: 10pt arial"><B>Subject:</B> RE: [Sip] target set</DIV>
  <DIV><FONT color=#008000></FONT><FONT color=#008000></FONT><BR></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003>HI</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003>'target set' is a set of uris which the client sets 
  to try incase it receives a 3xx response. </SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN class=156573211-24012003>so 
  to start with the client will have only one uri in the target set ile the 
  target it intends to reach.</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#ff0000 size=2><SPAN 
  class=156573211-24012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#ff0000 size=2><SPAN class=156573211-24012003>This 
  sentance is quiet confusing. In the initial request what the UAC tries out is 
  not the target set (there is only one URI). its the URI which UAC intends to 
  reach out. the target set (contact addresses in 3xx response ) implies the 
  URI's where the target user could be reached. these addresses can be contacted 
  in either linear or parallel search technique.</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN class=156573211-24012003>now 
  say it receives 3xx (300 Moved) response then it receives a new URI in the 
  contact header. so it can place this URI in its target set and send the 
  request to that URI.</SPAN></FONT></DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN class=156573211-24012003>so 
  this way the target set grows.</SPAN></FONT></DIV>
  <DIV><FONT color=#008000></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial color=#0000ff size=2><SPAN 
  class=156573211-24012003>-Ranjit</SPAN></FONT></DIV>
  <BLOCKQUOTE style="MARGIN-RIGHT: 0px">
    <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
    size=2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org 
    [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Gao Qing<BR><B>Sent:</B> 
    Friday, January 24, 2003 3:20 PM<BR><B>To:</B> 
    sip@ietf.org<BR><B>Subject:</B> [Sip] target set<BR><BR></DIV></FONT>
    <P><FONT face="Courier New" size=2>Dear Sirs,<BR><BR></FONT><FONT 
    face="Courier New" size=2>Who can explain me what exactly is "target set" in 
    RFC3261? Is there a definition of it somewhere?</FONT> </P>
    <P><FONT face="Courier New" size=2>Thanks in advance!</FONT> </P>
    <P><FONT face="Courier New" size=2>Gao Qing</FONT> </P>
    <DIV><FONT color=#008000>Regards,</FONT></DIV>
    <DIV>-------------------------------------------<BR>Nataraju A.B.<BR>Huawei 
    Technologies India Pvt. Ltd.,<BR>Tel : +91-80-5217152 / 4 Xtn 
    142<BR>-------------------------------------------</DIV></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_V+DIqRKwyKwlgcQRooJPqg)--
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Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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H,


> >>If I receive a tel URL containing a local number, without
> >>and idea of the context *in the place that the URL was constructed*
> >>then I don't know whether or not I'm in that context.
> >
> >[CHH] In this case you could reject it. If the sender didn't insert it, and
> >there was no intermediate node inserting it, there is a network architecture
> >problem.
> >
> >My point, I guess Juha's too, was not that there should not be a context when
> >the URL reaches the UAS, but that it MAY be inserted by another node than the
> >UAC.
> >
> <snip>
> >I agree on that. However, my SIP phone may not know to which context they
> >belong, for whatever reason, but the outbound proxy they communicate with DO
> >know.
>
> To which I reply::
>
> Aha! Thanks for the clarification - I'm being slow today.
>    This is a pure SIP issue - usage of tel URLs in SIP, and request
> modification by intermediate nodes. Interesting issue.
>
> However, IMHO there are other uses for tel URLs, so I'd suggest that
> draft-antti-rfc2916bis is correct in mandating this behaviour.
>
> If there's a separate draft for "SIP usage of tel URLs" that qualifies
> the main tel URL RFC, relaxing the requirement, that's fine; I'm not sure
> if it's a SIP or SIPPING job, but I think that's where it lives.
>
> I just don't want to nail this down in the general 2916 update.

[CHH] 2916 doesn't have to say anything about SIP. 2916 can make the context
optional, and "users" (eg SIP) and network architectures can then mandate the use of
the context. There may also be other parameters, which are optional in 2916, but
still are mandatory for specific users/architectures/applications.

Another option, which you mentioned, is to write a separate "SIP usage of tel URL"
draft (this is pretty much what has been done for SDP in H.248/Megaco), but I am not
sure that is the best solution for this specific purpose.

Have a nice weekend!

Regards,

Christer Holmberg
Ericsson Finland


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From: "Conroy, Lawrence (SMTP)" <lwc@roke.co.uk>
To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Date: Fri, 24 Jan 2003 13:55:17 +0000
Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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Hi folks,
<snip>
comments below - lwc -
At 3:37 pm +0200 24/1/03, Christer Holmberg wrote:
>  > <snip>
>>  >I agree on that. However, my SIP phone may not know to which context they
>>  >belong, for whatever reason, but the outbound proxy they 
>>communicate with DO
>>  >know.
>>
>>  To which I reply::
>>
>>  Aha! Thanks for the clarification - I'm being slow today.
>>     This is a pure SIP issue - usage of tel URLs in SIP, and request
>>  modification by intermediate nodes. Interesting issue.
>>
>>  However, IMHO there are other uses for tel URLs, so I'd suggest that
>>  draft-antti-rfc2916bis is correct in mandating this behaviour.
>>
>>  If there's a separate draft for "SIP usage of tel URLs" that qualifies
>>  the main tel URL RFC, relaxing the requirement, that's fine; I'm not sure
>>  if it's a SIP or SIPPING job, but I think that's where it lives.
>>
>>  I just don't want to nail this down in the general 2916 update.
>
>[CHH] 2916 doesn't have to say anything about SIP. 2916 can make the context
>optional, and "users" (eg SIP) and network architectures can then 
>mandate the use of
>the context. There may also be other parameters, which are optional 
>in 2916, but
>still are mandatory for specific users/architectures/applications.
>Another option, which you mentioned, is to write a separate "SIP 
>usage of tel URL"
>draft (this is pretty much what has been done for SDP in 
>H.248/Megaco), but I am not
>sure that is the best solution for this specific purpose.
>
>Have a nice weekend!
>

- lwc -
[BTW, folks, apologies for cross-posting - this seems to relevant both to
SIP and IPTEL]

A SIP proxy can have a mechanism to insert a context, so disambiguating the
tel URL. So can other systems, I suspect.
However... if there is no such mechanism, then the DEFAULT behaviour is to
require a context as part of a "local number" tel URL (or bounce the URL).

I think that this "default" should be in RFC2916bis, whilst the relaxation
of the rule should be in the areas where there are mechanisms to avoid the
potential mistakes. Otherwise we can have unsafe but standard-conforming
implementations. My opinion only, but I need to be convinced.

   Nice weekend to one and all.
     Lawrence
-- 
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From mailnull@www1.ietf.org  Fri Jan 24 09:09:52 2003
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Subject: RE: RE: [Sip] SIP and Congestion Safety
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Hi,

I'm assuming that it won't be possible , for various reasons,
to deprecate UDP. Therefore even though it may not be possible
to assure 100% congestion safety when UDP is used, some
mechanisms such as those already presented in 2 existing drafts
may be used to prevent congestion to some extent.

Both the drafts(draft-khartabil-sip-congestionsafe-ci-01.txt
and draft-ietf-sip-congestsafe-00) propose using a max-size
header to indicate to the previous hop SIP entity the max
size that it(the SIP entity sending the 403, 503 response)
would allow on the next hop. Both the drats also assume that
the proxies know the MTU of the next hop by some means.

Its not very difficult to encounter the following problem
in this mechanism:
UA1--(MTU1)--P1--(MTU2)---P2--(MTU3)--P3--(MTU4)--P4--(MTU5)--P5--(MTU6)--U
A2

MTU1>MTU2>MTU3>MTU4>MTU5>MTU6

In such a case it might take UA1, 6 attempts to finally
get a request through to UA2. And since the MTU for a
link is dynamic this could get even worse.

Considering such a scenario, I was wondering if it might be
a good idea to adopt a mechanism similar to the one used
for the session timer, i.e allow proxies in the path to
insert or update a min MTU similar to the way the session
timer value is inserted/updated. Proxies need not reject
requests that exceed the MTU of the next hop(hence
avoiding the problem enumurated above). And like with
session timer, the UA that initiated the request would
get to know the min MTU for the complete path in the
response, instead of having to converge on a value
through trial and error.

Comments?

thanks
Harsh




hisham.khartabil@nokia.com on 01/22/2003 08:14:23 PM

To:   loretosa@vodafone.it, sip@ietf.org
cc:    (bcc: Harsh Bhondwe/HSSBLR)

Subject:  RE: RE: [Sip] SIP and Congestion Safety






> -----Original Message-----
> From: ext loretosa@vodafone.it [mailto:loretosa@vodafone.it]
> Sent: Wednesday, January 22, 2003 10:31 AM
> To: Khartabil Hisham (NMP/Helsinki); sip@ietf.org
> Subject: Ri: RE: [Sip] SIP and Congestion Safety
>
>
>
> Hi Hisham,
>
> I've read your draft "draft-khartabil-sip-congestionsafe-ci-01.txt"
> It's an interesting draft, but not cover all my questions.
>
> In any case about your draft i've some observations:
>
> in the section -5.0-
> You write that the proxy affect for Congestion Safe in 3GPP
> architecture is S-CSCF.
> I think that P-CSCF is the most critical Proxy.
> This one is the closest proxy to the UA. And in a Visited UA
> scenario, I think that it's preferable, for an Operator 3G,
> not forward information about his link capacity to another
> Operator 3G.

The document does say "the S-CSCF closest to the UA". I think the use of
the word "home" confused you.

> Morover I think that are also performance problem for which
> P-CSCF have know information about Max-Size SIP message that
> UAS can handle.

Perhaps, but to my knowledge, S-CSCFs have the intelligence. P-CSCFs are
nothing but outbound proxies to the sending UA.

In any case, where 3GPP decides to put the intelligence is out of the scope
of this document. The sentence you refer to is just an example (and perhaps
should be removed).

>
> In the section 5.5. "Home Proxy Receiving 413 Response from UA"
> What's advantage  for a UA to reject a request, also if is a
> large SIP request, when it's already received? When arrived
> to UA, the large message already has congestioned the link.
>
> The main problem is avoided and prevent the congestion link.

The solution is not just limited to the congestion of the network, it also
tries to handle out-of-memory situations for a terminal, even when using
TCP. If a terminal is receiving TCP packets for a sip message and knows it
can't handle anymore packets, it can stop receiving and send 413 error
response.

Regards,
Hisham

>
>
>
>
>
> Sal
>
>
> > Please take a look at
> >
> >
> http://www.ietf.org/internet-drafts/draft-khartabil-sip-conges
> tionsafe-ci-01.txt
> >
> > I'm planning to update this to include a UA posting (eg:
> contents for PUBLISH or MESSAGE).
> >
> > Regards,
> > Hisham
> >
> > > -----Original Message-----
> > > From: ext loretosa@vizzavi.it [mailto:loretosa@vizzavi.it]
> > > Sent: Monday, January 20, 2003 7:55 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] SIP and Congestion Safety
> > >
> > >
> > >
> > > Hi all,
> > >
> > > about SIP and Congetion Safety, i've observed there isn't a
> > > great interest.
> > >
> > > I don't know why.
> > > I think that will be a giant problem.
> > >
> > > And the mainly question for it is: what protocol use for
> > > transport SIP signals ???
> > >
> > >
> > > others wg in IETF are working about congestion control
> > > algorithms for real-time
> > > or, more  generally,  delay sensitive applications...
> > > see DCCP (Datagram Congestion Control Protocol  -
> > http://www.ietf.org/html.charters/dccp-charter.html).
> >
> > sometime ago there was a proposal about  SCTP use for
> Instant Messaging with SIP...
> >
> > in other mail i read a proposal about Connection Transport
> Protocol as default protocol for SIP
> >
> > in the "Session Initiation Protocol Extension to Assure
> Congestion Safety" draft-ietf-sip-congestsafe-00
> > there's only a proposal for define and check a
> CONGESTION-SAFE hop... but it isn't enough
> >
> >
> > Sal
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
>  Per registrarti gratuitamente a Vodafone Mail vai su www.190.it
>
>
>
_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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_______________________________________________
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From: Richard Shockey <richard@shockey.us>
Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
Cc: sip@ietf.org, iptel@ietf.org
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>>
>>[CHH] 2916 doesn't have to say anything about SIP. 2916 can make the context
>>optional, and "users" (eg SIP) and network architectures can then mandate 
>>the use of
>>the context. There may also be other parameters, which are optional in 
>>2916, but
>>still are mandatory for specific users/architectures/applications.
>>Another option, which you mentioned, is to write a separate "SIP usage of 
>>tel URL"
>>draft (this is pretty much what has been done for SDP in H.248/Megaco), 
>>but I am not
>>sure that is the best solution for this specific purpose.
>>
>>Have a nice weekend!

Well some of this discussion might be relevant for the ENUM WG as well but 
I do not see any need for 2916 to have any information about context at 
all. This thread is somewhat confusing to me.

What 2916 is explicit about is that is dealing with fully qualified e164 
numbers only. You will have noticed that references or examples of partial 
e164 numbers have been removed from 2916bis 03.  Queries to partial e164 
numbers ( local numbers ) or private dialing plans are assumed not to be in 
the context of e164.arpa at all.

UAC ore the relevant proxyshould be able to understand their context and 
reformulate the URL into the e164 number before any other action is taken.

As some one else mentioned this is old stuff to most of us... as a old 
developer of outbound IVR systems we always had to have a local text 
configuration file.

>- lwc -
>[BTW, folks, apologies for cross-posting - this seems to relevant both to
>SIP and IPTEL]
>
>A SIP proxy can have a mechanism to insert a context, so disambiguating the
>tel URL. So can other systems, I suspect.
>However... if there is no such mechanism, then the DEFAULT behaviour is to
>require a context as part of a "local number" tel URL (or bounce the URL).
>
>I think that this "default" should be in RFC2916bis, whilst the relaxation
>of the rule should be in the areas where there are mechanisms to avoid the
>potential mistakes. Otherwise we can have unsafe but standard-conforming
>implementations. My opinion only, but I need to be convinced.

For my information are you suggesting that this behavior be documented in a 
enumservice field registration for the tel URL or as part of the overall 
use of ENUM in SIP.

I dont see how documenting this in 2916bis is relevant..or am I missing 
something here



>   Nice weekend to one and all.
>     Lawrence
>--
>-----------------------------------------------------------------------
>Roke Manor Research    : This information is provided "as is" and is not
><mailto:lwc@roke.co.uk>: intended to create any contractual or legal
><tel:+441794833666>    : relationship.


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

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Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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At 10:21 am -0500 24/1/03, Richard Shockey wrote:
><snip>
>Well some of this discussion might be relevant for the ENUM WG as 
>well but I do not see any need for 2916 to have any information 
>about context at all. This thread is somewhat confusing to me.
<snip>

To which I reply:

Arghh!!!!
     My bad - for 2916bis in the thread, please read 2806bis.
It's time for the weekend. That will teach me to respond to mailing
lists during a teleconference - duh!

Sorry for the confusion - we return you to your normal service.

atb,  Lawrence
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Subject: RE: RE: [Sip] SIP and Congestion Safety
From: Dean Willis <dean.willis@softarmor.com>
To: hbhondwe@hss.hns.com
Cc: sip@ietf.org
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On Fri, 2003-01-24 at 08:04, hbhondwe@hss.hns.com wrote:
> Both the drafts(draft-khartabil-sip-congestionsafe-ci-01.txt
> and draft-ietf-sip-congestsafe-00) propose using a max-size
> header to indicate to the previous hop SIP entity the max
> size that it(the SIP entity sending the 403, 503 response)
> would allow on the next hop. Both the drats also assume that
> the proxies know the MTU of the next hop by some means.
> 
> Its not very difficult to encounter the following problem
> in this mechanism:
> UA1--(MTU1)--P1--(MTU2)---P2--(MTU3)--P3--(MTU4)--P4--(MTU5)--P5--(MTU6)--U
> A2
> 
> MTU1>MTU2>MTU3>MTU4>MTU5>MTU6
> 
> In such a case it might take UA1, 6 attempts to finally
> get a request through to UA2. And since the MTU for a
> link is dynamic this could get even worse.

yep. Actually, it's worse than that, since as you point out it is
possible for dynamic rerouting to occur.  That makes the absolute worst
case even more entertaining. Assuming that every reduction reduces the
size of the message by 1 byte, then the worst case number of retries is
given by the equation Size(Max) - Size(Min), where Max is the largest
message ever sent, and Min is the smallest MTU allowed. Assuming common
values of 1300 and 576, that gives us more than 700 retries before a
message gets through. But at least it's deterministic.


> Considering such a scenario, I was wondering if it might be
> a good idea to adopt a mechanism similar to the one used
> for the session timer, i.e allow proxies in the path to
> insert or update a min MTU similar to the way the session
> timer value is inserted/updated. Proxies need not reject
> requests that exceed the MTU of the next hop(hence
> avoiding the problem enumurated above). And like with
> session timer, the UA that initiated the request would
> get to know the min MTU for the complete path in the
> response, instead of having to converge on a value
> through trial and error.


If I understand this, you're proposing that we complete one round-trip
in order to establish a route size for future use within the dialog, and
that we assume that the route does not vary over the course of the
dialog.
 
The main problem here -- what if the first request is 64kB long and
blows up on the UDP network just past the first proxy?

And of course, there's the problem of route dynamicity.

Here's an alternative proposal. Rather than negotiating size, could we
"just say no"?

Any proxy that finds it cannot congestion-safely transmit a message
simply kicks back a 5xx "cannot send safely" response, and it's up to
the sender to find another route?

--
Dean

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Subject: RE: RE: [Sip] SIP and Congestion Safety
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> -----Original Message-----
> From: ext Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Friday, January 24, 2003 6:45 PM
> To: hbhondwe@hss.hns.com
> Cc: sip@ietf.org
> Subject: RE: RE: [Sip] SIP and Congestion Safety
> 
> 
> On Fri, 2003-01-24 at 08:04, hbhondwe@hss.hns.com wrote:
> > Both the drafts(draft-khartabil-sip-congestionsafe-ci-01.txt
> > and draft-ietf-sip-congestsafe-00) propose using a max-size
> > header to indicate to the previous hop SIP entity the max
> > size that it(the SIP entity sending the 403, 503 response)
> > would allow on the next hop. Both the drats also assume that
> > the proxies know the MTU of the next hop by some means.
> > 
> > Its not very difficult to encounter the following problem
> > in this mechanism:
> > 
> UA1--(MTU1)--P1--(MTU2)---P2--(MTU3)--P3--(MTU4)--P4--(MTU5)--
> P5--(MTU6)--U
> > A2
> > 
> > MTU1>MTU2>MTU3>MTU4>MTU5>MTU6
> > 
> > In such a case it might take UA1, 6 attempts to finally
> > get a request through to UA2. And since the MTU for a
> > link is dynamic this could get even worse.
> 
> yep. Actually, it's worse than that, since as you point out it is
> possible for dynamic rerouting to occur.  That makes the 
> absolute worst
> case even more entertaining. Assuming that every reduction reduces the
> size of the message by 1 byte, then the worst case number of 
> retries is
> given by the equation Size(Max) - Size(Min), where Max is the largest
> message ever sent, and Min is the smallest MTU allowed. 
> Assuming common
> values of 1300 and 576, that gives us more than 700 retries before a
> message gets through. But at least it's deterministic.


I believe the drafts do say the max MTU en route to the UAS. How that en route is discovered is outside the scope of sip, IMO.


> 
> 
> > Considering such a scenario, I was wondering if it might be
> > a good idea to adopt a mechanism similar to the one used
> > for the session timer, i.e allow proxies in the path to
> > insert or update a min MTU similar to the way the session
> > timer value is inserted/updated. Proxies need not reject
> > requests that exceed the MTU of the next hop(hence
> > avoiding the problem enumurated above). And like with
> > session timer, the UA that initiated the request would
> > get to know the min MTU for the complete path in the
> > response, instead of having to converge on a value
> > through trial and error.
> 
> 
> If I understand this, you're proposing that we complete one round-trip
> in order to establish a route size for future use within the 
> dialog, and
> that we assume that the route does not vary over the course of the
> dialog.
>  
> The main problem here -- what if the first request is 64kB long and
> blows up on the UDP network just past the first proxy?
> 
> And of course, there's the problem of route dynamicity.
> 
> Here's an alternative proposal. Rather than negotiating size, could we
> "just say no"?
> 
> Any proxy that finds it cannot congestion-safely transmit a message
> simply kicks back a 5xx "cannot send safely" response, and it's up to
> the sender to find another route?

What's wrong with content indirection?

/Hisham


> 
> --
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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Hi,

> >>[CHH] 2916 doesn't have to say anything about SIP. 2916 can make the >>context
> optional, and "users" (eg SIP) and network architectures can then >>mandate the
> use of
> >>the context. There may also be other parameters, which are optional in
> >>2916, but
> >>still are mandatory for specific users/architectures/applications.
> >>Another option, which you mentioned, is to write a separate "SIP usage of
> >>tel URL"
> >>draft (this is pretty much what has been done for SDP in H.248/Megaco),
> >>but I am not
> >>sure that is the best solution for this specific purpose.
> >>
> >>Have a nice weekend!
>
> Well some of this discussion might be relevant for the ENUM WG as well but
> I do not see any need for 2916 to have any information about context at
> all. This thread is somewhat confusing to me.

[CHH] I was actually thinking about that. If I have a E.164 number, do an ENUM DNS
lookup, and receive a TEL-URL with a local number in the lookup answer, shall that
TEL-URL contain the context, or shall my node add it?

> What 2916 is explicit about is that is dealing with fully qualified e164
> numbers only. You will have noticed that references or examples of partial
> e164 numbers have been removed from 2916bis 03.  Queries to partial e164
> numbers ( local numbers ) or private dialing plans are assumed not to be in
> the context of e164.arpa at all.

[CHH] So, possible TEL-URL(s) received in the ENUM lookup MUST be in international
format?

Regards,

Christer Holmberg
Ericsson Finland


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Any comments on this?

Regards,

Christer Holmberg
Ericsson Finland

Brett Tate wrote:

> > I appologise if this has been discussed
> > before, but at least I didn't find it in
> > my mail box (I admit I didn't check the
> > archieve on the web).
>
> I posted a related question to sip-implementors;
> however I never got a reply.
>
> http://lists.cs.columbia.edu/pipermail/sip-implementors/2002-October/004052.
> html
>
> > Chapter 18.2.2 in RFC3261 talks about the
> > routing of responses, using the Via maddr,
> > received and sent-by parameters.
> >
> > My question is regarding maddr. When reading
> > the text, I get the picture that only
> > IP addresses are used in maddr. At least the
> > text doesn't mention the SRV RFC, which it does
> > for sent-by, and it says that the default port
> > is 5060. However, the ABNF defines maddr as
> > "host", which can also be a domain name.
>
> maddr can be a domain name.  The text appears
> to disable the ability to perform an SRV query
> associated with the Via maddr.  However I never
> got an answer concerning if this was intentional.
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: Re: [Iptel] Re: [Sip] TEL-URL local numbers
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>
> >
> > Well some of this discussion might be relevant for the ENUM WG as well but
> > I do not see any need for 2916 to have any information about context at
> > all. This thread is somewhat confusing to me.
>
>[CHH] I was actually thinking about that. If I have a E.164 number, do an 
>ENUM DNS
>lookup, and receive a TEL-URL with a local number in the lookup answer, 
>shall that
>TEL-URL contain the context, or shall my node add it?

why would ENUM return a local number is my question? What is the 
application scenario you have in mind ..

For Instance the presumption is that you would have a return of URL' one 
sip maybe one tel
this is the old call forwarding application ..if you cannot reach sip: then 
try tel: easy to do with the modification of the preference field..then 
certainly the tel should be a FQe164.



> > What 2916 is explicit about is that is dealing with fully qualified e164
> > numbers only. You will have noticed that references or examples of partial
> > e164 numbers have been removed from 2916bis 03.  Queries to partial e164
> > numbers ( local numbers ) or private dialing plans are assumed not to be in
> > the context of e164.arpa at all.
>
>[CHH] So, possible TEL-URL(s) received in the ENUM lookup MUST be in 
>international
>format?

Thats certainly what I would recommend and that has always been the 
parameters of 2916 no partial e164 numbers . but you could point to a 
private dialing plan as in a domain other than e164.arpa.

In fact you can create any terminal or non terminal redirection you want if 
you write up a RFC describing it using the procedure in 2916bis


>Regards,
>
>Christer Holmberg
>Ericsson Finland


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey, Senior Manager, Strategic Technology Initiatives
NeuStar Inc.
46000 Center Oak Plaza  -   Sterling, VA  20166
Voice +1 571.434.5651 Cell : +1 314.503.0640,  Fax: +1 815.333.1237
<mailto:richard@shockey.us> or <mailto:richard.shockey@neustar.biz>
  <http://www.neustar.biz> ; <http://www.enum.org>
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

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From mailnull@www1.ietf.org  Fri Jan 24 16:52:42 2003
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This answer is not accurate.

Servers always use the request URI in determining next hops, looking up 
in registration databases, and so on. The To and request-uri can be 
divergent at the very first hop due to security concerns, for example.

Furthermore, a server can only perform a location database lookup on the 
request uri if it is authoritatively responsible for the domain in the 
request uri.

-Jonathan R.

Prasanna Venkatesh wrote:
> Hi,
> As per the RFC-3261 on the first hop it is expected that the To and the
> Request-URI are same.  In this case your query does not have any problem.
> If some intermediate proxy has changed the request-URI then, the request-uri
> being the latest should be used to search the registrar.  However it depends
> on the design of the registry database also.
> If the database caters only to one domain, then just the user part of the
> request-uri is taken and the address can be resolved.
> If the database caters to more than one domain, and if by chance the host
> part of the request-uri is an IP then, the host part of the To along with
> the user part of the request-uri is taken as the key.
> Cheers,
> Prasanna
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Idnani
> Ajaykumar-AIDNANI1
> Sent: Tuesday, January 14, 2003 8:23 PM
> To: sip@ietf.org
> Subject: [Sip] Question about Redirect Server
> 
> 
> Hi All,
> 
> I have a very simple question about the operation of Redirect Servers. Me
> and colleague out here can not seem to convince each other on this, so I
> thought I will email all the SIP experts here J.
> 
> When a Redirect Server gets a request, it is supposed to return back the
> Contacts for the called party. Where does it pick the address of the called
> party from. Is it the To address or the Request-URI? The assumption here is
> that the Redirect Server is co-located with the Registrar, and has access to
> the database of the Registrar. When the Redirect Server is searching within
> this database, what does it use as the key for searching - To Address or the
> Request-URI.
> 
> Appreciate your help.
> 
> Thanks and Regards
> - Ajay
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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From mailnull@www1.ietf.org  Fri Jan 24 17:02:57 2003
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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
CC: brett@broadsoft.com, sip@ietf.org
Subject: Re: [Sip] Via maddr
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The intent of 18.2.2 is indeed NOT to use the SRV processing. In fact, 
if you look at 18.1, the only way that the maddr gets set is through 
multicast, and thus the intent is that maddr is really for multicast. 
THis need not be an IP address, since DNS names can refer to multicast 
addresses (sip.mcast.net).

Originally, the maddr in the URI was the same - meant just for 
multicast. However, in the absence of a suitable loose routing 
mechanism, it got continually abused until the common case was for that, 
and not for multicast at all. Doing that properly required SRV 
processing. Now, we have thankfully deprecated the usage of maddr for 
that purpose, but older rfc2543 elements might construct such URIs, and 
so RFC3263 continues to perform the SRV processing for the maddr in the 
request URI, when present.

Thus the asymetry.

-Jonathan R.

Christer Holmberg wrote:
> Any comments on this?
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> Brett Tate wrote:
> 
> 
>>>I appologise if this has been discussed
>>>before, but at least I didn't find it in
>>>my mail box (I admit I didn't check the
>>>archieve on the web).
>>
>>I posted a related question to sip-implementors;
>>however I never got a reply.
>>
>>http://lists.cs.columbia.edu/pipermail/sip-implementors/2002-October/004052.
>>html
>>
>>
>>>Chapter 18.2.2 in RFC3261 talks about the
>>>routing of responses, using the Via maddr,
>>>received and sent-by parameters.
>>>
>>>My question is regarding maddr. When reading
>>>the text, I get the picture that only
>>>IP addresses are used in maddr. At least the
>>>text doesn't mention the SRV RFC, which it does
>>>for sent-by, and it says that the default port
>>>is 5060. However, the ABNF defines maddr as
>>>"host", which can also be a domain name.
>>
>>maddr can be a domain name.  The text appears
>>to disable the ability to perform an SRV query
>>associated with the Via maddr.  However I never
>>got an answer concerning if this was intentional.
>>
>>_______________________________________________
>>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP Protocol
>>Use sip-implementors@cs.columbia.edu for questions on current sip
>>Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



From mailnull@www1.ietf.org  Fri Jan 24 17:12:23 2003
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To: "Nataraju A.B.(huawei)" <natarajuab@huawei.com>
CC: ranjit.avasarala@bplmail.com, Gao Qing <qing.gao@siemens.com>,
        sip@ietf.org
Subject: Re: [Sip] target set
References: <GGEKLACDJCBAOANBIIIFEEMPCDAA.ranjit.avasarala@bplmail.com> <001701c2c3a5$a65f9fa0$5d02120a@in.huawei.com>
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The target set is a set of URIs, used by proxies or UAC, which represent 
a set of destinations that are to be tried. A proxy creates such a set 
by contacting location services, but it can be changed by many factors 
through the duration of a transaction, such as a redirect response. A 
UAC creates a target set when it is redirected to multipicity of contacts.

-Jonathan R.

Nataraju A.B.(huawei) wrote:
> 
>     ----- Original Message -----
>     *From:* Ranjit Kumar Avasarala <mailto:ranjit.avasarala@bplmail.com>
>     *To:* Gao Qing <mailto:qing.gao@siemens.com> ; sip@ietf.org
>     <mailto:sip@ietf.org>
>     *Sent:* Friday, January 24, 2003 5-06
>     *Subject:* RE: [Sip] target set
> 
>     HI
>      
>     'target set' is a set of uris which the client sets to try incase it
>     receives a 3xx response.
>     so to start with the client will have only one uri in the target set
>     ile the target it intends to reach.
>      
>     This sentance is quiet confusing. In the initial request what the
>     UAC tries out is not the target set (there is only one URI). its the
>     URI which UAC intends to reach out. the target set (contact
>     addresses in 3xx response ) implies the URI's where the target user
>     could be reached. these addresses can be contacted in either linear
>     or parallel search technique.
>      
>     now say it receives 3xx (300 Moved) response then it receives a new
>     URI in the contact header. so it can place this URI in its target
>     set and send the request to that URI.
>      
>     so this way the target set grows.
>      
>     -Ranjit
> 
>         -----Original Message-----
>         *From:* sip-admin@ietf.org [mailto:sip-admin@ietf.org]*On Behalf
>         Of *Gao Qing
>         *Sent:* Friday, January 24, 2003 3:20 PM
>         *To:* sip@ietf.org
>         *Subject:* [Sip] target set
> 
>         Dear Sirs,
> 
>         Who can explain me what exactly is "target set" in RFC3261? Is
>         there a definition of it somewhere?
> 
>         Thanks in advance!
> 
>         Gao Qing
> 
>         Regards,
>         -------------------------------------------
>         Nataraju A.B.
>         Huawei Technologies India Pvt. Ltd.,
>         Tel : +91-80-5217152 / 4 Xtn 142
>         -------------------------------------------
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



From mailnull@www1.ietf.org  Fri Jan 24 17:26:22 2003
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To: jean-luc.bouthemy@nokia.com
CC: jh@lohi.eng.song.fi, sip@ietf.org
Subject: Re: [Sip] any restrictions on characters in user part of sip uri?
References: <8F698F0DD0B49A48BDA9F451A7A68CAE019E6764@daebe001.americas.nokia.com>
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Well, the spec is currently quiet on this point. According to rfc2396, 
definitions of URI schemes are supposed to define the character set that 
is used to map from the octet sequence to the original character 
sequence. SIP doesn't say what this is. It ought to, though, and clearly 
UTF-8 is the appropriate choice. That would allow you to escape-encode 
Japanese characters, for example.

This has been logged in bugzilla for some time:
http://www.sipwg.org/sipwg/show_bug.cgi?id=678

Concerning Juha's example, UTF-8 has the property that 00 never occurs 
in a UTF-8 sequence except to represent NULL, so you shuoldn't be seeing 
it except in the "normal" US-ASCII way.

-Jonathan R.




jean-luc.bouthemy@nokia.com wrote:
 > Juha and all:
 >
 > Comment:
 >
 > What about using escaped is %hexhex for support to unicode which can
 > be used for languages such as Japanese.
 >
 > Regards
 >
 > -- Jean-Luc -----Original Message----- From: ext jh@lohi.eng.song.fi
 > [mailto:jh@lohi.eng.song.fi] Sent: Friday, January 17, 2003 5:29 AM
 > To: sip@ietf.org Subject: [Sip] any restrictions on characters in
 > user part of sip uri?
 >
 >
 > the syntax of user part of sip uri is:
 >
 > user = 1*( unreserved / escaped / user-unreserved ) user-unreserved =
 > "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"
 >
 > escaped is %hexhex and can thus be used to include any character in
 > user.  is it really so that there is no restrictions on what chars
 > may be include in user using escaped?  for example, is %0000 valid in
 > user? if so, it makes implementations "interesting".
 >
 > -- juha
 >
 > _______________________________________________ Sip mailing list
 > https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
 > development of the core SIP Protocol Use
 > sip-implementors@cs.columbia.edu for questions on current sip Use
 > sipping@ietf.org for new developments on the application of sip
 > _______________________________________________ Sip mailing list
 > https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
 > development of the core SIP Protocol Use
 > sip-implementors@cs.columbia.edu for questions on current sip Use
 > sipping@ietf.org for new developments on the application of sip
 >

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Subject: Re: [Sip] Re: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt :
 Reposting
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You won't be surprised to hear me say that I think this is an entirely 
inappropriate use of SIP, whether you use the INFO method or not.

There is nothing in your usage that derives any benefit from SIP. There 
are no sessions, no calls, no nothing. You are merely using an INFO 
method as a way to send data from A to B. TCAP is closer to an RPC 
protocol than anything else, and SIP is certainly NOT an RPC protocol.

Tunneling ISUP over SIP (SIP-T) does make a lot of sense, since ISUP, 
functionally, is very close to SIP. We are therefore able to use SIP 
semantics for the exchange, and augment the flows with additional data 
(the tunneled ISUP message) that provides information of interest to 
gateways. The key to the utility of SIP-T is that its still SIP.

If you want an XML encoding of TCAP, I have no beef with that concept, 
but its not SIP. Have you considered SOAP? SOAP over HTTP seems a much 
better match.

-Jonathan R.



Frank W. Miller wrote:
> Thanks for the questions!
> 
> Comments inline.
> 
> 
> 
>>For the transport of SS7-signalling information the SIGTRAN protocols have
> 
> been especially defined by the SIGTRAN
> 
>>working group of the IETF. In this scenario SUA over SCTP could transport
> 
> the TCAP information. For the time being PSTNs still exist and in converged
> scenarios we should use the same straight forward approach defined by the
> SIGTRAN group for each SS7 user part - why special treatment for TCAP???.
> 
> 
> We have a similar mechanism for the ISUP maintenance commands.  An early
> version of that draft is on the IETF site and a major update that changes
> the INFO body contents to an XML representation will be released within
> days.  TCAP is not special in this regard.  IMHO, ISUP and TCAP are the best
> place to do translations because the "semantic level" appears to me to be
> the closest to where SIP session semantic level is.  These two works are
> designed to augment the basic SIP-ISUP translation draft to include the
> ability to control PSTN access trunks (e.g. PRI's and IMTs on T-1s) from the
> PSTN and to do service delivery from existing PSTN elements (i.e. SCPs) to
> SIP endpoints, both without having to implement SS7 protocol stacks in the
> SIP elements.  The goal is make sharper the demarcation between SIP and SS7.
> This is primarily a cost argument.  SS7 stacks are more expensive in the
> marketplace than SIP stacks, so limiting the number of SS7 deployments saves
> costs for the carriers.
> 
> 
> 
>>Why defining another approach when everythhing neccesary is already
> 
> defined. Or are there doubts about the maturity in the SIGTRAN protocols?
> 
> 
> No, SIGTRAN is very mature and very useful for certain applications.  This
> is just a different approach.  I see no reason why tunneling and translation
> approaches cannot coexist in IETF standards.  Both have their applications.
> 
> 
> 
>>Some questions on the proposed solution:
>>-- How is adressing beeing resolved?
> 
> 
> One of the nice things about translation is that it limits addressing
> problems to the mapping of addresses.  You can do about anything you want on
> the SIP side as long as the mappings at the SIP UA and the GW are handled
> appropriately.  Addressing is certainly an area that is ripe for discussion
> in any standardization activity.
> 
> 
> 
>>-- TCAP dialoges are mainly used for call unrelated signalling, means no
> 
> dialog exists. But one has to have an open SIP-Dialog before exchanging
> Info-Messages. How is that going to be resolved?
> 
> The element with the UA maintains a persistent session with the GW for TCAP
> interactions.  I believe this is discussed in the draft.
> 
> 
> 
>>-- For what kind of services this should be used? Is this solution
> 
> anticipated for call completion services instead of usage of
> subscribe/notify?
> 
> 
> There are several service examples detailed in the draft.  Calling Name
> Delivery is the one that leaps to mind.
> 
> FM
> 
> 
> Frank W. Miller, Ph.D.
> Chief Technical Officer
> sentitO Network, Inc.
> fmiller@sentito.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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CC: Christer Holmberg <christer.holmberg@lmf.ericsson.se>,
        Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
Subject: Re: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
References: <003301c2bb73$b6528650$fe0c0c42@txdwillis> <3E243454.3B30B59F@lmf.ericsson.se> <3E24398F.8090205@cs.columbia.edu>
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The point I have been trying to make all along here is similar to what 
Henning is saying. Everything in this draft can also be done with the 
Allow mechanism, and then you use the method registry for "usages".

Now, Dean has argued that this is somehow fundamentally different due to 
some perceived software layering problem in implementations, but 
functionally speaking, its the same.

THis is what I mean when I have been saying "INFO has no semantics, 
whereas rfc3265 does". There is protocol functionality provided by 
SUB/NOT, independent of any package, and above rfc3261, which makes it 
useful. Not so here.

-Jonathan R.

Henning Schulzrinne wrote:
>> To negotiate what INFO can be used for between nodes is good thing, 
>> and at least
>> on a theoretical level it solves one issue.
> 
> 
> It would be nice to motivate how this differs and intersects with the 
> Content-* indication and negotiation mechanism. After all, if INFO 
> transports objects that are opaque to the "SIP layer" (e.g., session 
> state), the logical indication of use is via the various Content-* 
> indications.
> 
>>
>> But, is the biggest problem we have today end-nodes receiving INFOs 
>> with data
>> they don't know to handle? And, how is this going to solve the problem 
>> of having
>> people using INFO for whatever purpose? All they have to do is to 
>> define a
>> package for their who-knows-what data, right? Or, does defining a 
>> package allow
>> one to use INFO for whatever he/she wants?
> 
> 
> Beyond ISUP and maybe DTMF, what is INFO used for today, by both the 
> good and bad guys? What proposals are out there? Last time this was 
> discussed, the list seemed pretty short. I think we agree that DTMF has 
> better, non-INFO solutions.
> 
> With events, we have a whole set of examples and applications to draw 
> on. Building a mechanism on an example space of one seems a bit dangerous.
> 
>>
>> Or, do we accept Henning's rule about the "bad guys", and do NOT try 
>> to define
>> any rules on exactly what kind of data INFO should be used for in the 
>> first
>> place?
> 
> 
> I do think it would be helpful if this draft or some other draft 
> indicated where SIP INFO *may* be appropriate. As discussed, this seems 
> to involve things where SIP routing and proxies are needed or useful, 
> for example.
> 
>>
>> I am not saying that this draft should solve anything, but I would 
>> like to know
>> exactly which issues regarding INFO we are going to try solving :)
>>
>> Regards,
>>
>> Christer Holmberg
>> Ericsson Finland
>>
>>
>>
>>
>>
>> Dean Willis wrote:
>>
>>
>>> It only seemed appropriate to follow up my previous assumably 
>>> eloquent but
>>> frequently misspelled exhortations by following my own advice and 
>>> "sending
>>> text".
>>>
>>> The resulting draft has been submitted to the archive and should be
>>> announced in due course. In the interim, you may wish to retrieve it 
>>> from:
>>>
>>> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.html 
>>>
>>>
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

_______________________________________________
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Subject: [Sip] trimming "implicit preferences for media and priority" from caller
 prefs
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I have been working on yet-another revision of caller preferences, in my 
continuing attempts to get this thing finished. Just when I thought it 
was done, and all the changes were happily applied, I ran into another 
issue. Grr.

The issue is an interaction between the implicit preferences mechanism 
in the draft, and the desire to reach a user wherever they are.

Anyway, let me explain the problem by an example. Lets say I send an 
INVITE with audio and video in the SDP. The called party has two 
devices, a PC w/ audio and video, and a  cell phone with audio. They are 
not at their PC right now, but are available by cell. The desired 
behavior, and the intent of the caller (since they did not specify a 
preference), is that the call would reach both devices and ring both, 
and the user answers at the cell phone with audio only. However, the 
implicit preferences will cause the PC to ring first (because its a 
better match to the "implicit preference" to have video), and then the 
cell phone. THis will increase call setup time and worsen the user 
experience.

This problem is a specific case of a more genreal issue, which is that 
the implicit caller preferences mechanism is at odds with the proposed 
fix to HERFP (see 
http://www.jdrosen.net/papers/draft-rosenberg-sip-unify-00.txt if you 
forget what HERFP is). They cannot co-exist. The whole idea with the 
HERFP fix is to let the request go everywhere it can, since the called 
party could be at any one of the devices. But, the caller prefs implicit 
preferences would prioritize and select only the matching devices. The 
called party may not be at that device, though.

The essence of the problem, I think, is that there is no such thing as 
implicit preferences. Just because I list audio and video in my SDP, 
does not mean I want the call to be routed to a device which supports 
those. It just means that those are the media types I want to try to use 
for this call. Generally, danger lurks wherever the network tries to 
assume something about caller intent, when it should not have assumed it.

So, my proposed solution is to remove the implicit preferences 
computation for media types, and also for priority (sections 7.2.2.1 and 
7.2.2.4 of caller prefs -07). The priority one is good to remove since 
we were never sure it was right to begin with. That would leave implicit 
preferences computations ONLY for methods, languages and event packages. 
There are issues there too, but of a somewhat different sort, which I 
will post in a separate note.

So, unless someone screams really soon, consider those two features removed.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Thanks Jonathan for the response.

Your reaction is indeed expected.  I would make only two comments on your
response.  First, I do not agree with your unequivocal statement that TCAP
has nothing to do with a session.  There are several applications of TCAP
(CNAME and LNP come to mind) where there is clearly at least a fledgling
session in progress when TCAP is required.  Second, I've tried to get across
to other responders that there is a practical issue that is very important
here, that being cost.  Current PSTN protocol implementations are very
expensive to implement.  By this, I mean that the cost of protocol stacks
must be passed onto the end customer in some form.  I believe that this is
at least in part due to the fact that these protocols are "closed" when
compared with SIP.  The cost-effectiveness of SIP yields a desire to limit
the use of the PSTN protocol implementations in order to minimize the cost
of the equipment into which they are built.  This is why tunneling can be
problematic, it requires at least some part of these protocol stacks to be
in place on a larger base of equipment.

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Frank W. Miller" <fmiller@sentito.com>
Cc: "Poetzl, Joachim" <Joachim.Poetzl@telekom.de>; <sip@ietf.org>
Sent: Friday, January 24, 2003 10:40 PM
Subject: Re: [Sip] Re: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt :
Reposting


> You won't be surprised to hear me say that I think this is an entirely
> inappropriate use of SIP, whether you use the INFO method or not.
>
> There is nothing in your usage that derives any benefit from SIP. There
> are no sessions, no calls, no nothing. You are merely using an INFO
> method as a way to send data from A to B. TCAP is closer to an RPC
> protocol than anything else, and SIP is certainly NOT an RPC protocol.
>
> Tunneling ISUP over SIP (SIP-T) does make a lot of sense, since ISUP,
> functionally, is very close to SIP. We are therefore able to use SIP
> semantics for the exchange, and augment the flows with additional data
> (the tunneled ISUP message) that provides information of interest to
> gateways. The key to the utility of SIP-T is that its still SIP.
>


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"Frank W. Miller" <fmiller@sentito.com> writes:

> First, I do not agree with your unequivocal statement that TCAP
> has nothing to do with a session.  There are several applications of TCAP
> (CNAME and LNP come to mind) where there is clearly at least a fledgling
> session in progress when TCAP is required.

For those readers not familiar with TCAP, the above argument
is analogous to claiming something like, "There are several applications
of DNS (SIP SRV comes to mind) where there is clearly at least a
fledgling session in progress when DNS is required". CNAME and
LNP are nothing more than database dips.

Now, imagine someone using this argument as justification for
a proposal to tunnel XML formatted DNS queries over SIP.

> Second, I've tried to get across
> to other responders that there is a practical issue that is very important
> here, that being cost.  Current PSTN protocol implementations are very
> expensive to implement.  By this, I mean that the cost of protocol stacks
> must be passed onto the end customer in some form.

And you've steadfastly ignored  responses pointing out
that the expensive part of PSTN protocol stacks is *not*
the syntactical decoding, but the implementation of the
necessary protocol semantics. A syntactic transformation gains
you little or nothing in terms of implementation complexity
and total cost of solution.

/a

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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@ietf.org
Subject: Re: [Sip] trimming "implicit preferences for media and priority"
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Caller preferences seems to mandate being explicit rather than assuming 
things, so I agree that this should go. This is partially a lack of 
expressiveness of SDP (and the media preferences in caller-prefs), which 
can't distinguish between three subtly different things:

- capabilities that the device has (things that are possible)

- things the user would like to do now (which may be smaller or larger 
than the current capabilities);

- things that the user might want to do later (e.g., after reaching a 
higher-bandwidth area or transferring a conference mid-call to a more 
capable device).

Also, this is really a symptom of two non-commensurate quantities 
(preference for multiple media and preference for quickly reaching the 
destination) being at odds. Without a script, we can't really express 
all the subleties here, from "I'm willing to wait for the most desirable 
device to become available" at one extreme to "all devices being 
available, I choose X" at the other.

The problem would not occur if only devices where the user is available 
are reachable. In this example, the correct solution would be that the 
PC would not be in the reachable set at all.

Jonathan Rosenberg wrote:
> I have been working on yet-another revision of caller preferences, in my 
> continuing attempts to get this thing finished. Just when I thought it 
> was done, and all the changes were happily applied, I ran into another 
> issue. Grr.
> 
> The issue is an interaction between the implicit preferences mechanism 
> in the draft, and the desire to reach a user wherever they are.
> 
> Anyway, let me explain the problem by an example. Lets say I send an 
> INVITE with audio and video in the SDP. The called party has two 
> devices, a PC w/ audio and video, and a  cell phone with audio. They are 
> not at their PC right now, but are available by cell. The desired 
> behavior, and the intent of the caller (since they did not specify a 
> preference), is that the call would reach both devices and ring both, 
> and the user answers at the cell phone with audio only. However, the 
> implicit preferences will cause the PC to ring first (because its a 
> better match to the "implicit preference" to have video), and then the 
> cell phone. THis will increase call setup time and worsen the user 
> experience.
> 
> This problem is a specific case of a more genreal issue, which is that 
> the implicit caller preferences mechanism is at odds with the proposed 
> fix to HERFP (see 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-unify-00.txt if you 
> forget what HERFP is). They cannot co-exist. The whole idea with the 
> HERFP fix is to let the request go everywhere it can, since the called 
> party could be at any one of the devices. But, the caller prefs implicit 
> preferences would prioritize and select only the matching devices. The 
> called party may not be at that device, though.
> 
> The essence of the problem, I think, is that there is no such thing as 
> implicit preferences. Just because I list audio and video in my SDP, 
> does not mean I want the call to be routed to a device which supports 
> those. It just means that those are the media types I want to try to use 
> for this call. Generally, danger lurks wherever the network tries to 
> assume something about caller intent, when it should not have assumed it.
> 
> So, my proposed solution is to remove the implicit preferences 
> computation for media types, and also for priority (sections 7.2.2.1 and 
> 7.2.2.4 of caller prefs -07). The priority one is good to remove since 
> we were never sure it was right to begin with. That would leave implicit 
> preferences computations ONLY for methods, languages and event packages. 
> There are issues there too, but of a somewhat different sort, which I 
> will post in a separate note.
> 
> So, unless someone screams really soon, consider those two features 
> removed.
> 
> -Jonathan R.
> 

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Subject: Re: [Sip] Re: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt : Reposting
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WRT to sessions, while I can make at least some kind of weak argument about
TCAP.  Its clear that using a similar mechanism for ISUP maintenance
commands can be characterized exactly as Jonathan has described.  The only
reason for using SIP at all is the fact that its already there.  Its there
because its being used to establish sessions for phone calls.  ISUP and TCAP
are used in the PSTN for establishing calls/maintaining trunks and service
creation.  My view is that this underlying semantic similarity is enough to
justify the use of SIP.  I'd go even further, IMHO the acceptance of SIP as
a signaling protocol for the big carriers will not be embraced
wholeheartedly until these "corners" can be addressed.  Its clear that any
installation of SIP for signaling in the incumbents will have to
interoperate with existing SS7 signaling networks for the forseeable future
and that means all of SS7, not just call setup and teardown.
Interoperability can of course be addressed by tunneling as has been
described.  It can also be addressed by translation at the SIP/SS7 border.
This translation could be done to something other than SIP, e.g. SOAP as you
and Jonathan have mentioned, but why?  Why should I add another protocol,
with its missing functionality btw, when I can just do it all in the context
of SIP?

>
> And you've steadfastly ignored  responses pointing out
> that the expensive part of PSTN protocol stacks is *not*
> the syntactical decoding, but the implementation of the
> necessary protocol semantics. A syntactic transformation gains
> you little or nothing in terms of implementation complexity
> and total cost of solution.
>

You are correct, the implemenation of the state machines associated with
these features is an expensive proposition.  However, the state machine
remains the same regardless of whether you're en/decoding a binary encoded
TCAP message or an XML encoded version of the same message in an INFO body.
The difference is I don't have to pay an SS7 vendor as many times for
licenses to SS7 stack instances.  This is a very significant cost savings.

FM

Frank W. Miller, Ph.D.
Chief Technical Officer
sentitO Network, Inc.
fmiller@sentito.com

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Subject: Re: [Sip] Re: Comments: I-D ACTION:draft-miller-sip-tcap-00.txt : Reposting
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"Frank W. Miller" <fmiller@sentito.com> writes:
> Its clear that any
> installation of SIP for signaling in the incumbents will have to
> interoperate with existing SS7 signaling networks for the forseeable
future
> and that means all of SS7, not just call setup and teardown.
> Interoperability can of course be addressed by tunneling as has been
> described.  It can also be addressed by translation at the SIP/SS7 border.

The translation you are describing, though, has no benefit.
It's not like you're suggesting that the commands are translated
into something that performs analogous operations in the SIP
network; you're just suggesting that we perform syntactic
translations, slap them in a SIP INFO, and send them on
their way. There's no benefit over tunneling.

> This translation could be done to something other than SIP, e.g. SOAP as
you
> and Jonathan have mentioned, but why?  Why should I add another protocol,
> with its missing functionality btw, when I can just do it all in the
context
> of SIP?

I'm still trying to build a conceptual model of the netowork you have
in mind; however, based on the descriptions you give in this note,
it sounds like what you really want is probably sigtran. The goal
there, of course, was explicitly to not have any missing functionality
when carrying SS7 information over an IP network.

And I have long said that using SIP to establish sigtran sessions
would be an elegant solution to that sort of problem.

> > And you've steadfastly ignored  responses pointing out
> > that the expensive part of PSTN protocol stacks is *not*
> > the syntactical decoding, but the implementation of the
> > necessary protocol semantics. A syntactic transformation gains
> > you little or nothing in terms of implementation complexity
> > and total cost of solution.
>
> You are correct, the implemenation of the state machines associated with
> these features is an expensive proposition.

No, not the state machines of the features -- the state machines of the
*protocols* themselves. In SIP terms, you're considering changing
out the parser and re-coding the application without taking into
consideration all the other "stuff" in between: the "stuff" that makes
the SIP spec 269 pages instead of a 10-page listing of ABNF.

> However, the state machine
> remains the same regardless of whether you're en/decoding a binary encoded
> TCAP message or an XML encoded version of the same message in an INFO
body.
> The difference is I don't have to pay an SS7 vendor as many times for
> licenses to SS7 stack instances.  This is a very significant cost savings.

It sounds to me like you're saying, "by inventing  new syntax that
no one has implemented yet, I'll be forced to develop a stack
in-house, at a substantial savings over licensing a third party stack."

With your own assertion that it's just as complicated regardless
of the syntax chosen, isn't it clear that you can develop an actual
honest-to-God TCAP stack just as cheaply in-house as you can
implement your own "TCAP-with-a-syntax-transform"?

/a

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--=====================_787146697==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed


Hi Everyone,

I am looking for a real time scenario in which 305(Use proxy) message is 
generated by a redirect server.
I find it hard to think of an example where a pure redirect server would 
send a USE PROXY instead of simply issuing a 300-302 messages.

It is also said in the RFC that the 305(use proxy) "MUST ONLY" be generated 
by user agent servers. This is a bit unclear to me.

thanks.

Rajesh k.

--=====================_787146697==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<br>
Hi Everyone,<br>
<br>
I am looking for a real time scenario in which
<font color="#FF0000">305(Use proxy)</font> message is generated by a
redirect server.<br>
I find it hard to think of an example where a pure redirect server would 
send a USE PROXY instead of simply issuing a  300-302 messages.<br>
<br>
It is also said in the RFC that the 305(use proxy) &quot;MUST ONLY&quot;
be generated by user agent servers. This is a bit unclear to me. <br>
<br>
thanks.<br>
<br>
Rajesh k.<br>
</html>

--=====================_787146697==_.ALT--

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Date: Thu, 23 Jan 2003 21:06:22 -0400
To: <dean.willis@softarmor.com>, <rohan@cisco.com>, <jon.peterson@neustar.biz>
From: pierobon <pierobon@sympatico.ca>
Cc: <sob@harvard.edu>, <mankin@psg.com>, <dromasca@avaya.com>, sip@ietf.org
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Subject: [Sip] SIP patent infringement warning
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Dear Sirs,

Having examined your specs and use of SIP, I'm must point out that your 
company may be infringing on my patent rights.

You can verify for yourself by examining the patent at 
http://massive-array-cellular-system.com

Thanks,

Victor Pierobon
416 221-2340
Pierobon@attcanada.ca


  To:

Dean Willis
Rohan Mahy
Jon Peterson
Scott Bradner
Allison Mankin
Allison Mankin
Dan Romascanu
  


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Date: Mon, 27 Jan 2003 14:27:17 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Error reporting in caller prefs
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This is the second note of two that describes some recently encountered
issues while trying to wrap caller prefs.

The issue here is error reporting.

Lets say a caller lists a bunch of caller preferences, which include
Reject-Contacts and Require-Contacts (now removed in -08 and replaced
with a require parameter on Accept-Contact, but I'll use the -07 
terminology). It is entirely
possible that these preferences will result in ALL registered contacts
getting discarded. The current behavior will be that the proxy acts as
if there were no registered contacts, and returns a 480. Now, this means
that the caller has no way to determine if the request failed because of
their preferences, or because there really were no registered contacts.

At first, I thought that this was a bad thing. But, thinking abuot it
more, I concluded it was a good thing, for privacy reasons. The proxy is
the one performing the caller preferences computation. The proxy will
frequently not be the one authenticating the caller, or authorizing them
to make calls or learn information abuot the called party. But, if a
proxy reported information on call failures because of preference
mismatches, a client would be able to "probe" the network using caller
preferences, and learn detailed ifnormation on the capabilities of the
devices associated with a user, without authentication. The current
behavior prevents that, and I think thats good.

OK, so whats the problem? Well, its those implicit preferences again.
Lets say a user sends a SUBSCRIBE request, without caller preferences
parameters. The called party has one UA, and it doesnt support
SUBSCRIBE. Its registration to its proxy indicated such. The proxy
supports caller preferences. So, it wuold compute an implicit preference
to route the request to a UA that supports SUBSCRIBE, and give it the 
require strength (meaning, contacts which don't support SUBSCRIBE are 
discarded). Since none do, the
one registered contact is discarded, and the caller receives a 480.

A similar thing happens for event packages.

This is a change in behavior if caller preferences were not used, in
which case either (1) the UAS would authenticate and authorize the
caller, and then return a 405 with the allowed methods, (2) the UAS
would not authenticate or authorize the UAC, and thus genreate a 403 or 401.

As a result, even though the caller did not use caller preferences, they
have suffered from a loss of information, just because the proxy used
caller prefs. The 405 would be useful for the UAC to perhaps try another
method, or report the failure to the user. I dont mind a loss of
information with EXPLICIT preferences, since the caller knows what they
are getting into. But for implicit, we shouldn't be changing basic SIP
behavior as seen by the UAC.

There are a small number of solutions:

1. eliminate Require-Contact entirely (or, in -08 terms, eliminate the 
require parameter to Accept-Contact). This solves the problem, because 
the proxy will simply set the un-matching contacts to have a q-value of 
zero, and try them last. In the above example, the one registered 
contact remains with q=0, and is tried, resulting in a 405.

2.eliminate implicit preferences that use the Require-Contact semantic.
So, if the request fails with a 480, its because the client explicitly 
asked for  soemthing they couldnt get.

3. If a proxy uses caller prefs, and the result is the eliminatin of all 
contacts, AND if the reason they were eliminated was that there was an 
implicit Require-Contact for a specific method or event package, the 
proxy generates a 405 or 489 as if it were the UAS.

4. modify the semantics of require-contact so that, in the case where
all contacts would be eliminated, instead of eliminating them, it sets
the q-value to zero. This avoids the error condition.


I didn't see any other approaches. None of these seem perfect:

1. Approach 1 is a non-starter; we have applications which need the 
Require semantic.

2. Approach 2 is better than approach 1. However, it will result in 
SUBSCRIBE and other methods being sent to a UAS that indicated that it 
didn't support those methods. On the list, Juha had complained loudly 
about this, and this thread was one of the motivations for adding 
require-contact. Now, the client can add the explicit Require-Contact 
with this solution, but that adds somewhat of a burden.

3. Appraoch 3 has a potential security issue. The proxy will generate a 
405 or 489 on behalf of the UAS. So, if the UAS would generally not want 
to reveal this information (supported methods and event packages) unless 
the UAC would be authorized and authenticated, this approach will "leak" 
this information. One might argue that if the UAS uses caller prefs, and 
tells the proxy which methods and event packages it supports, it has 
forfeited its right to explicitly authenticate and authorize the caller 
before this information is divulged. If the UAS has these privacy 
concerns, it should not register these caller preferences parameters.

4. Approach 4 has similar problems to approach 2.


I am inclined to go with approach 3, adding the appropriate discussions 
on the privacy considerations. The specific algorithm I would propose is 
this. If the caller prefs selection process results in zero contact, the 
proxy re-runs the algorithms without the implicit preferences. If now, 
there are contacts remaining, it concludes that the implicit preferences 
caused the problem. It then checks to see if the problem was methods or 
event packages, and generates a 405 or 489 accordingly.

Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From mailnull@www1.ietf.org  Mon Jan 27 16:36:02 2003
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From: "Adam Roach" <adam@dynamicsoft.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@ietf.org>
References: <3E358815.2050806@dynamicsoft.com>
Subject: Re: [Sip] Error reporting in caller prefs
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"Jonathan Rosenberg" <jdrosen@dynamicsoft.com> writes:
> I am inclined to go with approach 3, adding the appropriate discussions 
> on the privacy considerations. The specific algorithm I would propose is 
> this. If the caller prefs selection process results in zero contact, the 
> proxy re-runs the algorithms without the implicit preferences. If now, 
> there are contacts remaining, it concludes that the implicit preferences 
> caused the problem. It then checks to see if the problem was methods or 
> event packages, and generates a 405 or 489 accordingly.

Two comments; one con, one "interesting":

1. It seems to me that this approach adds a substantial burden
   to the proxy.

2. This has the interesting effect of forcing the proxy to
   aggregate the methods and the event-packages
   available for an address-of-record. (e.g. if I have
   a client registered that supports MESSAGE,
   and one that supports INVITE, the proxy-generated
   405 will ostensibly indicate both INVITE and MESSAGE
   as acceptable methods.) Of course, this may well exacerbate
   the security problems you discuss -- but it also might
   be a useful feature. It also has the odd consequence that
   I can get a more accurate picture of your capabilities
   by intentionally sending an invalid method than
   I can by sending an OPTIONS...

/a

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Date: Tue, 28 Jan 2003 14:32:49 +0530
From: Prasanna Venkatesh <prasanna@huawei.com>
Subject: RE: [Sip] Regarding 305 Redirection message ....
In-reply-to: <4.3.2.7.2.20030127162940.027bfb40@desh.cisco.com>
To: "Rajesh Kalshetty (rkalshet)" <rkalshet@cisco.com>, sip@ietf.org
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Hi,
    In a simple scenario where an UA is behind an "authenticating" proxy,
and where the UA trusts requests from the proxy only, if an INVITE is
directly send to the UA, it may request the sender of the UA to go through
the proxy to get authenticated and the trust relationship.
    305 is a requirement for specific network toplogies and an UAS which
sends this is never a redirect server.

Cheers,
Prasanna
  -----Original Message-----
  From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Rajesh
Kalshetty (rkalshet)
  Sent: Monday, January 27, 2003 4:56 PM
  To: sip@ietf.org
  Subject: [Sip] Regarding 305 Redirection message ....



  Hi Everyone,

  I am looking for a real time scenario in which 305(Use proxy) message is
generated by a redirect server.
  I find it hard to think of an example where a pure redirect server would
send a USE PROXY instead of simply issuing a 300-302 messages.

  It is also said in the RFC that the 305(use proxy) "MUST ONLY" be
generated by user agent servers. This is a bit unclear to me.

  thanks.

  Rajesh k.


--Boundary_(ID_oOc8nb7+50T8v9hOOIxUnQ)
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=us-ascii" http-equiv=Content-Type>
<META content="MSHTML 5.00.3103.1000" name=GENERATOR></HEAD>
<BODY>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003>Hi,</SPAN></FONT></DIV>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003>&nbsp;&nbsp;&nbsp; In a simple scenario where an UA is 
behind an "authenticating" proxy, and where the UA trusts requests from the 
proxy only, if an INVITE is directly send to the UA, it may request the sender 
of the UA to go through the proxy to get authenticated and the trust 
relationship.</SPAN></FONT></DIV>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003>&nbsp;&nbsp;&nbsp; 305 is a requirement for specific 
network toplogies and an UAS which sends this is&nbsp;never a redirect 
server.</SPAN></FONT></DIV>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003>Cheers,</SPAN></FONT></DIV>
<DIV><FONT color=#808080 face=Lydian><SPAN 
class=188412007-28012003>Prasanna</SPAN></FONT></DIV>
<BLOCKQUOTE>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> sip-admin@ietf.org 
  [mailto:sip-admin@ietf.org]<B>On Behalf Of </B>Rajesh Kalshetty 
  (rkalshet)<BR><B>Sent:</B> Monday, January 27, 2003 4:56 PM<BR><B>To:</B> 
  sip@ietf.org<BR><B>Subject:</B> [Sip] Regarding 305 Redirection message 
  ....<BR><BR></DIV></FONT><BR>Hi Everyone,<BR><BR>I am looking for a real time 
  scenario in which <FONT color=#ff0000>305(Use proxy)</FONT> message is 
  generated by a redirect server.<BR>I find it hard to think of an example where 
  a pure redirect server would send a USE PROXY instead of simply issuing a 
  300-302 messages.<BR><BR>It is also said in the RFC that the 305(use proxy) 
  "MUST ONLY" be generated by user agent servers. This is a bit unclear to me. 
  <BR><BR>thanks.<BR><BR>Rajesh k.<BR></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_oOc8nb7+50T8v9hOOIxUnQ)--
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Hi,

The behavior or a SIP client and server in IPv4 is quite clear and
analogous in IPv6.
However, the behavior of SIP clients and servers in an environment with
both IPv4 and IPv6 is to me unclear (dual-stack). Do anyone know if
there has been any work done about this, and in that case what the
result of that work was?

Best regards,
Carl

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<P><SPAN LANG=3D"sv"><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT></SPAN>
</P>

<P><SPAN LANG=3D"sv"><FONT SIZE=3D2 FACE=3D"Arial">The behavior or a SIP =
client and server in IPv4 is quite clear and analogous in =
IPv6.</FONT></SPAN>

<BR><SPAN LANG=3D"sv"><FONT SIZE=3D2 FACE=3D"Arial">However, the =
behavior of SIP clients and servers in an environment with both IPv4 and =
IPv6 is to me unclear (dual-stack). Do anyone know if there has been any =
work done about this, and in that case what the result of that work =
was?</FONT></SPAN></P>

<P><SPAN LANG=3D"sv"><FONT SIZE=3D2 FACE=3D"Arial">Best =
regards,</FONT></SPAN>

<BR><SPAN LANG=3D"sv"><FONT SIZE=3D2 =
FACE=3D"Arial">Carl</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN>
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Subject: RE: [Sip] Error reporting in caller prefs
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Can all this be fixed with a Require/Supported tag indicating that a UA understands the caller preferences extensions? Basically eliminating implicit preferences.

Regards,
Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Monday, January 27, 2003 9:27 PM
> To: sip@ietf.org
> Subject: [Sip] Error reporting in caller prefs
> 
> 
> This is the second note of two that describes some recently 
> encountered
> issues while trying to wrap caller prefs.
> 
> The issue here is error reporting.
> 
> Lets say a caller lists a bunch of caller preferences, which include
> Reject-Contacts and Require-Contacts (now removed in -08 and replaced
> with a require parameter on Accept-Contact, but I'll use the -07 
> terminology). It is entirely
> possible that these preferences will result in ALL registered contacts
> getting discarded. The current behavior will be that the proxy acts as
> if there were no registered contacts, and returns a 480. Now, 
> this means
> that the caller has no way to determine if the request failed 
> because of
> their preferences, or because there really were no registered 
> contacts.
> 
> At first, I thought that this was a bad thing. But, thinking abuot it
> more, I concluded it was a good thing, for privacy reasons. 
> The proxy is
> the one performing the caller preferences computation. The proxy will
> frequently not be the one authenticating the caller, or 
> authorizing them
> to make calls or learn information abuot the called party. But, if a
> proxy reported information on call failures because of preference
> mismatches, a client would be able to "probe" the network using caller
> preferences, and learn detailed ifnormation on the capabilities of the
> devices associated with a user, without authentication. The current
> behavior prevents that, and I think thats good.
> 
> OK, so whats the problem? Well, its those implicit preferences again.
> Lets say a user sends a SUBSCRIBE request, without caller preferences
> parameters. The called party has one UA, and it doesnt support
> SUBSCRIBE. Its registration to its proxy indicated such. The proxy
> supports caller preferences. So, it wuold compute an implicit 
> preference
> to route the request to a UA that supports SUBSCRIBE, and give it the 
> require strength (meaning, contacts which don't support SUBSCRIBE are 
> discarded). Since none do, the
> one registered contact is discarded, and the caller receives a 480.
> 
> A similar thing happens for event packages.
> 
> This is a change in behavior if caller preferences were not used, in
> which case either (1) the UAS would authenticate and authorize the
> caller, and then return a 405 with the allowed methods, (2) the UAS
> would not authenticate or authorize the UAC, and thus 
> genreate a 403 or 401.
> 
> As a result, even though the caller did not use caller 
> preferences, they
> have suffered from a loss of information, just because the proxy used
> caller prefs. The 405 would be useful for the UAC to perhaps 
> try another
> method, or report the failure to the user. I dont mind a loss of
> information with EXPLICIT preferences, since the caller knows 
> what they
> are getting into. But for implicit, we shouldn't be changing basic SIP
> behavior as seen by the UAC.
> 
> There are a small number of solutions:
> 
> 1. eliminate Require-Contact entirely (or, in -08 terms, 
> eliminate the 
> require parameter to Accept-Contact). This solves the 
> problem, because 
> the proxy will simply set the un-matching contacts to have a 
> q-value of 
> zero, and try them last. In the above example, the one registered 
> contact remains with q=0, and is tried, resulting in a 405.
> 
> 2.eliminate implicit preferences that use the Require-Contact 
> semantic.
> So, if the request fails with a 480, its because the client 
> explicitly 
> asked for  soemthing they couldnt get.
> 
> 3. If a proxy uses caller prefs, and the result is the 
> eliminatin of all 
> contacts, AND if the reason they were eliminated was that 
> there was an 
> implicit Require-Contact for a specific method or event package, the 
> proxy generates a 405 or 489 as if it were the UAS.
> 
> 4. modify the semantics of require-contact so that, in the case where
> all contacts would be eliminated, instead of eliminating them, it sets
> the q-value to zero. This avoids the error condition.
> 
> 
> I didn't see any other approaches. None of these seem perfect:
> 
> 1. Approach 1 is a non-starter; we have applications which need the 
> Require semantic.
> 
> 2. Approach 2 is better than approach 1. However, it will result in 
> SUBSCRIBE and other methods being sent to a UAS that 
> indicated that it 
> didn't support those methods. On the list, Juha had complained loudly 
> about this, and this thread was one of the motivations for adding 
> require-contact. Now, the client can add the explicit Require-Contact 
> with this solution, but that adds somewhat of a burden.
> 
> 3. Appraoch 3 has a potential security issue. The proxy will 
> generate a 
> 405 or 489 on behalf of the UAS. So, if the UAS would 
> generally not want 
> to reveal this information (supported methods and event 
> packages) unless 
> the UAC would be authorized and authenticated, this approach 
> will "leak" 
> this information. One might argue that if the UAS uses caller 
> prefs, and 
> tells the proxy which methods and event packages it supports, it has 
> forfeited its right to explicitly authenticate and authorize 
> the caller 
> before this information is divulged. If the UAS has these privacy 
> concerns, it should not register these caller preferences parameters.
> 
> 4. Approach 4 has similar problems to approach 2.
> 
> 
> I am inclined to go with approach 3, adding the appropriate 
> discussions 
> on the privacy considerations. The specific algorithm I would 
> propose is 
> this. If the caller prefs selection process results in zero 
> contact, the 
> proxy re-runs the algorithms without the implicit 
> preferences. If now, 
> there are contacts remaining, it concludes that the implicit 
> preferences 
> caused the problem. It then checks to see if the problem was 
> methods or 
> event packages, and generates a 405 or 489 accordingly.
> 
> Comments?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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From mailnull@www1.ietf.org  Tue Jan 28 18:09:53 2003
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Implicit preferences has the benefit that it requires only one "side" to 
support it - the UAS and its proxy. Thus, if the terminating domain 
desires that SUBSCRIBE requests are routed to the PCs and not to the 
phones, it can do that by deploying caller prefs without the 
subscriber/watcher domain deploying it. This particular feature of 
caller prefs - only routing requests to UAs which support that method, 
is particularly useful. There was a long thread on the list about this 
feature, with some strong advocates behind it.

-Jonathan R.



hisham.khartabil@nokia.com wrote:
 > Can all this be fixed with a Require/Supported tag indicating that a
 > UA understands the caller preferences extensions? Basically
 > eliminating implicit preferences.
 >
 > Regards, Hisham
 >
 >
 >> -----Original Message----- From: ext Jonathan Rosenberg
 >> [mailto:jdrosen@dynamicsoft.com] Sent: Monday, January 27, 2003
 >> 9:27 PM To: sip@ietf.org Subject: [Sip] Error reporting in caller
 >> prefs
 >>
 >>
 >> This is the second note of two that describes some recently
 >> encountered issues while trying to wrap caller prefs.
 >>
 >> The issue here is error reporting.
 >>
 >> Lets say a caller lists a bunch of caller preferences, which
 >> include Reject-Contacts and Require-Contacts (now removed in -08
 >> and replaced with a require parameter on Accept-Contact, but I'll
 >> use the -07 terminology). It is entirely possible that these
 >> preferences will result in ALL registered contacts getting
 >> discarded. The current behavior will be that the proxy acts as if
 >> there were no registered contacts, and returns a 480. Now, this
 >> means that the caller has no way to determine if the request failed
 >>  because of their preferences, or because there really were no
 >> registered contacts.
 >>
 >> At first, I thought that this was a bad thing. But, thinking abuot
 >> it more, I concluded it was a good thing, for privacy reasons. The
 >> proxy is the one performing the caller preferences computation. The
 >> proxy will frequently not be the one authenticating the caller, or
 >>  authorizing them to make calls or learn information abuot the
 >> called party. But, if a proxy reported information on call failures
 >> because of preference mismatches, a client would be able to "probe"
 >> the network using caller preferences, and learn detailed
 >> ifnormation on the capabilities of the devices associated with a
 >> user, without authentication. The current behavior prevents that,
 >> and I think thats good.
 >>
 >> OK, so whats the problem? Well, its those implicit preferences
 >> again. Lets say a user sends a SUBSCRIBE request, without caller
 >> preferences parameters. The called party has one UA, and it doesnt
 >> support SUBSCRIBE. Its registration to its proxy indicated such.
 >> The proxy supports caller preferences. So, it wuold compute an
 >> implicit preference to route the request to a UA that supports
 >> SUBSCRIBE, and give it the require strength (meaning, contacts
 >> which don't support SUBSCRIBE are discarded). Since none do, the
 >> one registered contact is discarded, and the caller receives a 480.
 >>
 >>
 >> A similar thing happens for event packages.
 >>
 >> This is a change in behavior if caller preferences were not used,
 >> in which case either (1) the UAS would authenticate and authorize
 >> the caller, and then return a 405 with the allowed methods, (2) the
 >> UAS would not authenticate or authorize the UAC, and thus genreate
 >> a 403 or 401.
 >>
 >> As a result, even though the caller did not use caller preferences,
 >> they have suffered from a loss of information, just because the
 >> proxy used caller prefs. The 405 would be useful for the UAC to
 >> perhaps try another method, or report the failure to the user. I
 >> dont mind a loss of information with EXPLICIT preferences, since
 >> the caller knows what they are getting into. But for implicit, we
 >> shouldn't be changing basic SIP behavior as seen by the UAC.
 >>
 >> There are a small number of solutions:
 >>
 >> 1. eliminate Require-Contact entirely (or, in -08 terms, eliminate
 >> the require parameter to Accept-Contact). This solves the problem,
 >> because the proxy will simply set the un-matching contacts to have
 >> a q-value of zero, and try them last. In the above example, the one
 >> registered contact remains with q=0, and is tried, resulting in a
 >> 405.
 >>
 >> 2.eliminate implicit preferences that use the Require-Contact
 >> semantic. So, if the request fails with a 480, its because the
 >> client explicitly asked for  soemthing they couldnt get.
 >>
 >> 3. If a proxy uses caller prefs, and the result is the eliminatin
 >> of all contacts, AND if the reason they were eliminated was that
 >> there was an implicit Require-Contact for a specific method or
 >> event package, the proxy generates a 405 or 489 as if it were the
 >> UAS.
 >>
 >> 4. modify the semantics of require-contact so that, in the case
 >> where all contacts would be eliminated, instead of eliminating
 >> them, it sets the q-value to zero. This avoids the error condition.
 >>
 >>
 >>
 >> I didn't see any other approaches. None of these seem perfect:
 >>
 >> 1. Approach 1 is a non-starter; we have applications which need the
 >>  Require semantic.
 >>
 >> 2. Approach 2 is better than approach 1. However, it will result in
 >>  SUBSCRIBE and other methods being sent to a UAS that indicated
 >> that it didn't support those methods. On the list, Juha had
 >> complained loudly about this, and this thread was one of the
 >> motivations for adding require-contact. Now, the client can add the
 >> explicit Require-Contact with this solution, but that adds somewhat
 >> of a burden.
 >>
 >> 3. Appraoch 3 has a potential security issue. The proxy will
 >> generate a 405 or 489 on behalf of the UAS. So, if the UAS would
 >> generally not want to reveal this information (supported methods
 >> and event packages) unless the UAC would be authorized and
 >> authenticated, this approach will "leak" this information. One
 >> might argue that if the UAS uses caller prefs, and tells the proxy
 >> which methods and event packages it supports, it has forfeited its
 >> right to explicitly authenticate and authorize the caller before
 >> this information is divulged. If the UAS has these privacy
 >> concerns, it should not register these caller preferences
 >> parameters.
 >>
 >> 4. Approach 4 has similar problems to approach 2.
 >>
 >>
 >> I am inclined to go with approach 3, adding the appropriate
 >> discussions on the privacy considerations. The specific algorithm I
 >> would propose is this. If the caller prefs selection process
 >> results in zero contact, the proxy re-runs the algorithms without
 >> the implicit preferences. If now, there are contacts remaining, it
 >> concludes that the implicit preferences caused the problem. It then
 >> checks to see if the problem was methods or event packages, and
 >> generates a 405 or 489 accordingly.
 >>
 >> Comments?
 >>
 >> -Jonathan R. -- Jonathan D. Rosenberg, Ph.D.                72
 >> Eagle Rock Ave. Chief Scientist                             First
 >> Floor dynamicsoft                                 East Hanover, NJ
 >> 07936 jdrosen@dynamicsoft.com                     FAX:   (973)
 >> 952-5050 http://www.jdrosen.net                      PHONE: (973)
 >> 952-5000 http://www.dynamicsoft.com
 >>
 >>
 >> _______________________________________________ Sip mailing list
 >> https://www1.ietf.org/mailman/listinfo/sip This list is for NEW
 >> development of the core SIP Protocol Use
 >> sip-implementors@cs.columbia.edu for questions on current sip Use
 >> sipping@ietf.org for new developments on the application of sip
 >>
 >
 >

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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



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inline.

Adam Roach wrote:
> "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> writes:
> 
>>I am inclined to go with approach 3, adding the appropriate discussions 
>>on the privacy considerations. The specific algorithm I would propose is 
>>this. If the caller prefs selection process results in zero contact, the 
>>proxy re-runs the algorithms without the implicit preferences. If now, 
>>there are contacts remaining, it concludes that the implicit preferences 
>>caused the problem. It then checks to see if the problem was methods or 
>>event packages, and generates a 405 or 489 accordingly.
> 
> 
> Two comments; one con, one "interesting":
> 
> 1. It seems to me that this approach adds a substantial burden
>    to the proxy.

Well, its definitely more than zero. Caller prefs as it stands is, by 
definition, something a proxy spends cycles doing. How much more it will 
take to add this piece of processing, I am not certain.

> 
> 2. This has the interesting effect of forcing the proxy to
>    aggregate the methods and the event-packages
>    available for an address-of-record. (e.g. if I have
>    a client registered that supports MESSAGE,
>    and one that supports INVITE, the proxy-generated
>    405 will ostensibly indicate both INVITE and MESSAGE
>    as acceptable methods.) Of course, this may well exacerbate
>    the security problems you discuss -- but it also might
>    be a useful feature. It also has the odd consequence that
>    I can get a more accurate picture of your capabilities
>    by intentionally sending an invalid method than
>    I can by sending an OPTIONS...

Interesting indeed.

I would rather not reveal information which would otherwise not be 
revealed. I'd propose that the proxy reveal only the supported methods 
of the UA that would have otherwise received the request.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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I would actually recommend a BEEP package to carry TCAP instead.

thanks,
-rohan

On Friday, January 24, 2003, at 07:40 PM, Jonathan Rosenberg wrote:

> You won't be surprised to hear me say that I think this is an entirely 
> inappropriate use of SIP, whether you use the INFO method or not.
>
> There is nothing in your usage that derives any benefit from SIP. 
> There are no sessions, no calls, no nothing. You are merely using an 
> INFO method as a way to send data from A to B. TCAP is closer to an 
> RPC protocol than anything else, and SIP is certainly NOT an RPC 
> protocol.
>
> Tunneling ISUP over SIP (SIP-T) does make a lot of sense, since ISUP, 
> functionally, is very close to SIP. We are therefore able to use SIP 
> semantics for the exchange, and augment the flows with additional data 
> (the tunneled ISUP message) that provides information of interest to 
> gateways. The key to the utility of SIP-T is that its still SIP.
>
> If you want an XML encoding of TCAP, I have no beef with that concept, 
> but its not SIP. Have you considered SOAP? SOAP over HTTP seems a much 
> better match.
>
> -Jonathan R.
>
>
>
> Frank W. Miller wrote:
>> Thanks for the questions!
>> Comments inline.
>>> For the transport of SS7-signalling information the SIGTRAN 
>>> protocols have
>> been especially defined by the SIGTRAN
>>> working group of the IETF. In this scenario SUA over SCTP could 
>>> transport
>> the TCAP information. For the time being PSTNs still exist and in 
>> converged
>> scenarios we should use the same straight forward approach defined by 
>> the
>> SIGTRAN group for each SS7 user part - why special treatment for 
>> TCAP???.
>> We have a similar mechanism for the ISUP maintenance commands.  An 
>> early
>> version of that draft is on the IETF site and a major update that 
>> changes
>> the INFO body contents to an XML representation will be released 
>> within
>> days.  TCAP is not special in this regard.  IMHO, ISUP and TCAP are 
>> the best
>> place to do translations because the "semantic level" appears to me 
>> to be
>> the closest to where SIP session semantic level is.  These two works 
>> are
>> designed to augment the basic SIP-ISUP translation draft to include 
>> the
>> ability to control PSTN access trunks (e.g. PRI's and IMTs on T-1s) 
>> from the
>> PSTN and to do service delivery from existing PSTN elements (i.e. 
>> SCPs) to
>> SIP endpoints, both without having to implement SS7 protocol stacks 
>> in the
>> SIP elements.  The goal is make sharper the demarcation between SIP 
>> and SS7.
>> This is primarily a cost argument.  SS7 stacks are more expensive in 
>> the
>> marketplace than SIP stacks, so limiting the number of SS7 
>> deployments saves
>> costs for the carriers.
>>> Why defining another approach when everythhing neccesary is already
>> defined. Or are there doubts about the maturity in the SIGTRAN 
>> protocols?
>> No, SIGTRAN is very mature and very useful for certain applications.  
>> This
>> is just a different approach.  I see no reason why tunneling and 
>> translation
>> approaches cannot coexist in IETF standards.  Both have their 
>> applications.
>>> Some questions on the proposed solution:
>>> -- How is adressing beeing resolved?
>> One of the nice things about translation is that it limits addressing
>> problems to the mapping of addresses.  You can do about anything you 
>> want on
>> the SIP side as long as the mappings at the SIP UA and the GW are 
>> handled
>> appropriately.  Addressing is certainly an area that is ripe for 
>> discussion
>> in any standardization activity.
>>> -- TCAP dialoges are mainly used for call unrelated signalling, 
>>> means no
>> dialog exists. But one has to have an open SIP-Dialog before 
>> exchanging
>> Info-Messages. How is that going to be resolved?
>> The element with the UA maintains a persistent session with the GW 
>> for TCAP
>> interactions.  I believe this is discussed in the draft.
>>> -- For what kind of services this should be used? Is this solution
>> anticipated for call completion services instead of usage of
>> subscribe/notify?
>> There are several service examples detailed in the draft.  Calling 
>> Name
>> Delivery is the one that leaps to mind.
>> FM
>> Frank W. Miller, Ph.D.
>> Chief Technical Officer
>> sentitO Network, Inc.
>> fmiller@sentito.com
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: Re: [Sip] trimming "implicit preferences for media and priority" from caller prefs
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I agree.  I believe it also unsafe to use implicit schemes (sip:, tel:, 
sips:, etc...).

thx,
-r


On Friday, January 24, 2003, at 09:49 PM, Jonathan Rosenberg wrote:

> I have been working on yet-another revision of caller preferences, in 
> my continuing attempts to get this thing finished. Just when I thought 
> it was done, and all the changes were happily applied, I ran into 
> another issue. Grr.
>
> The issue is an interaction between the implicit preferences mechanism 
> in the draft, and the desire to reach a user wherever they are.
>
> Anyway, let me explain the problem by an example. Lets say I send an 
> INVITE with audio and video in the SDP. The called party has two 
> devices, a PC w/ audio and video, and a  cell phone with audio. They 
> are not at their PC right now, but are available by cell. The desired 
> behavior, and the intent of the caller (since they did not specify a 
> preference), is that the call would reach both devices and ring both, 
> and the user answers at the cell phone with audio only. However, the 
> implicit preferences will cause the PC to ring first (because its a 
> better match to the "implicit preference" to have video), and then the 
> cell phone. THis will increase call setup time and worsen the user 
> experience.
>
> This problem is a specific case of a more genreal issue, which is that 
> the implicit caller preferences mechanism is at odds with the proposed 
> fix to HERFP (see 
> http://www.jdrosen.net/papers/draft-rosenberg-sip-unify-00.txt if you 
> forget what HERFP is). They cannot co-exist. The whole idea with the 
> HERFP fix is to let the request go everywhere it can, since the called 
> party could be at any one of the devices. But, the caller prefs 
> implicit preferences would prioritize and select only the matching 
> devices. The called party may not be at that device, though.
>
> The essence of the problem, I think, is that there is no such thing as 
> implicit preferences. Just because I list audio and video in my SDP, 
> does not mean I want the call to be routed to a device which supports 
> those. It just means that those are the media types I want to try to 
> use for this call. Generally, danger lurks wherever the network tries 
> to assume something about caller intent, when it should not have 
> assumed it.
>
> So, my proposed solution is to remove the implicit preferences 
> computation for media types, and also for priority (sections 7.2.2.1 
> and 7.2.2.4 of caller prefs -07). The priority one is good to remove 
> since we were never sure it was right to begin with. That would leave 
> implicit preferences computations ONLY for methods, languages and 
> event packages. There are issues there too, but of a somewhat 
> different sort, which I will post in a separate note.
>
> So, unless someone screams really soon, consider those two features 
> removed.
>
> -Jonathan R.
>
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: Re: [Sip] Error reporting in caller prefs
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Hi,

I vote for option 4 with a subtle variation.  Accept-contact w/ require 
will still match contacts (with q=0) if all the explicit preferences 
match but the implicit ones do not.  The difference is that you still 
get a 480 if you eliminated all contacts using an *explicit* match, but 
an ordinary 405 or 489 if all contacts were eliminated using implicit 
preferences only.

thanks,
-rohan


On Monday, January 27, 2003, at 11:27 AM, Jonathan Rosenberg wrote:

> This is the second note of two that describes some recently encountered
> issues while trying to wrap caller prefs.
>
> The issue here is error reporting.
>
> Lets say a caller lists a bunch of caller preferences, which include
> Reject-Contacts and Require-Contacts (now removed in -08 and replaced
> with a require parameter on Accept-Contact, but I'll use the -07 
> terminology). It is entirely
> possible that these preferences will result in ALL registered contacts
> getting discarded. The current behavior will be that the proxy acts as
> if there were no registered contacts, and returns a 480. Now, this 
> means
> that the caller has no way to determine if the request failed because 
> of
> their preferences, or because there really were no registered contacts.
>
> At first, I thought that this was a bad thing. But, thinking abuot it
> more, I concluded it was a good thing, for privacy reasons. The proxy 
> is
> the one performing the caller preferences computation. The proxy will
> frequently not be the one authenticating the caller, or authorizing 
> them
> to make calls or learn information abuot the called party. But, if a
> proxy reported information on call failures because of preference
> mismatches, a client would be able to "probe" the network using caller
> preferences, and learn detailed ifnormation on the capabilities of the
> devices associated with a user, without authentication. The current
> behavior prevents that, and I think thats good.
>
> OK, so whats the problem? Well, its those implicit preferences again.
> Lets say a user sends a SUBSCRIBE request, without caller preferences
> parameters. The called party has one UA, and it doesnt support
> SUBSCRIBE. Its registration to its proxy indicated such. The proxy
> supports caller preferences. So, it wuold compute an implicit 
> preference
> to route the request to a UA that supports SUBSCRIBE, and give it the 
> require strength (meaning, contacts which don't support SUBSCRIBE are 
> discarded). Since none do, the
> one registered contact is discarded, and the caller receives a 480.
>
> A similar thing happens for event packages.
>
> This is a change in behavior if caller preferences were not used, in
> which case either (1) the UAS would authenticate and authorize the
> caller, and then return a 405 with the allowed methods, (2) the UAS
> would not authenticate or authorize the UAC, and thus genreate a 403 
> or 401.
>
> As a result, even though the caller did not use caller preferences, 
> they
> have suffered from a loss of information, just because the proxy used
> caller prefs. The 405 would be useful for the UAC to perhaps try 
> another
> method, or report the failure to the user. I dont mind a loss of
> information with EXPLICIT preferences, since the caller knows what they
> are getting into. But for implicit, we shouldn't be changing basic SIP
> behavior as seen by the UAC.
>
> There are a small number of solutions:
>
> 1. eliminate Require-Contact entirely (or, in -08 terms, eliminate the 
> require parameter to Accept-Contact). This solves the problem, because 
> the proxy will simply set the un-matching contacts to have a q-value 
> of zero, and try them last. In the above example, the one registered 
> contact remains with q=0, and is tried, resulting in a 405.
>
> 2.eliminate implicit preferences that use the Require-Contact semantic.
> So, if the request fails with a 480, its because the client explicitly 
> asked for  soemthing they couldnt get.
>
> 3. If a proxy uses caller prefs, and the result is the eliminatin of 
> all contacts, AND if the reason they were eliminated was that there 
> was an implicit Require-Contact for a specific method or event 
> package, the proxy generates a 405 or 489 as if it were the UAS.
>
> 4. modify the semantics of require-contact so that, in the case where
> all contacts would be eliminated, instead of eliminating them, it sets
> the q-value to zero. This avoids the error condition.
>
>
> I didn't see any other approaches. None of these seem perfect:
>
> 1. Approach 1 is a non-starter; we have applications which need the 
> Require semantic.
>
> 2. Approach 2 is better than approach 1. However, it will result in 
> SUBSCRIBE and other methods being sent to a UAS that indicated that it 
> didn't support those methods. On the list, Juha had complained loudly 
> about this, and this thread was one of the motivations for adding 
> require-contact. Now, the client can add the explicit Require-Contact 
> with this solution, but that adds somewhat of a burden.
>
> 3. Appraoch 3 has a potential security issue. The proxy will generate 
> a 405 or 489 on behalf of the UAS. So, if the UAS would generally not 
> want to reveal this information (supported methods and event packages) 
> unless the UAC would be authorized and authenticated, this approach 
> will "leak" this information. One might argue that if the UAS uses 
> caller prefs, and tells the proxy which methods and event packages it 
> supports, it has forfeited its right to explicitly authenticate and 
> authorize the caller before this information is divulged. If the UAS 
> has these privacy concerns, it should not register these caller 
> preferences parameters.
>
> 4. Approach 4 has similar problems to approach 2.
>
>
> I am inclined to go with approach 3, adding the appropriate 
> discussions on the privacy considerations. The specific algorithm I 
> would propose is this. If the caller prefs selection process results 
> in zero contact, the proxy re-runs the algorithms without the implicit 
> preferences. If now, there are contacts remaining, it concludes that 
> the implicit preferences caused the problem. It then checks to see if 
> the problem was methods or event packages, and generates a 405 or 489 
> accordingly.
>
> Comments?
>
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: [Sip] TCP sides and TLS sides
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Hi,
When implementing TLS as a secure means of communications for "sips" scheme, 
I have the following question:
Does the TCP client has to be (or can I assume it is) the TLS client in the 
handshake process?
Can it be that a TCP client will assume the role of a TLS server in the 
handshake process (or vice versa)?

Thanks in Advance,
James S. Ford.

_________________________________________________________________
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Subject: Re: [Sip] TCP sides and TLS sides
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From: Eric Rescorla <ekr@rtfm.com>
Date: 29 Jan 2003 09:48:28 -0800
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"James Ford" <james_s_ford@hotmail.com> writes:
> When implementing TLS as a secure means of communications for "sips"
> scheme, I have the following question:
> 
> Does the TCP client has to be (or can I assume it is) the TLS client
> in the handshake process?
The concept of TCP client isn't necessarily as clear as one might like
since the TCP specification allows for simultaneous open. RFC 793 in
fact doesn't even contain the word "client".

Which TLS peer takes the role of "client" must be specified by the
TLS-using protocol. However, in most cases one party does the TCP
active OPEN and that's the TLS client and the other does the TCP
passive OPEN and that's the TLS server.

You could, however, write a protocol that specified things
differently. SIPS is not such a protocol.

-Ekr

-- 
[Eric Rescorla                                   ekr@rtfm.com]
                http://www.rtfm.com/
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Rohan Mahy wrote:
> Hi,
> 
> I vote for option 4 with a subtle variation.  Accept-contact w/ require 
> will still match contacts (with q=0) if all the explicit preferences 
> match but the implicit ones do not.  The difference is that you still 
> get a 480 if you eliminated all contacts using an *explicit* match, but 
> an ordinary 405 or 489 if all contacts were eliminated using implicit 
> preferences only.

I don't quite follow the second sentence above, but I do understand the 
desired property in the third. Indeed, in my analysis below, I had 
simply declared approach 4 as being the same as 2, but its not so. In 
the case where there is at least one matching contact, approach 4 will 
make sure that implicit preferences are honored at require strength, so 
that a UA which doesn't support SUBSCRIBE never gets one if there is a 
UA that does, for example. The only case that a UA will get a method it 
doesnt support is when none of the registered contacts support that 
method. I think thats actually a reasonable compromise. It also has less 
burden on proxies than my original proposal to go with option 3.

So, I am now inclined to agree with Rohan and go with option 4.

-Jonathan R.

> On Monday, January 27, 2003, at 11:27 AM, Jonathan Rosenberg wrote:
> 
>> This is the second note of two that describes some recently encountered
>> issues while trying to wrap caller prefs.
>>
>> The issue here is error reporting.
>>
>> Lets say a caller lists a bunch of caller preferences, which include
>> Reject-Contacts and Require-Contacts (now removed in -08 and replaced
>> with a require parameter on Accept-Contact, but I'll use the -07 
>> terminology). It is entirely
>> possible that these preferences will result in ALL registered contacts
>> getting discarded. The current behavior will be that the proxy acts as
>> if there were no registered contacts, and returns a 480. Now, this means
>> that the caller has no way to determine if the request failed because of
>> their preferences, or because there really were no registered contacts.
>>
>> At first, I thought that this was a bad thing. But, thinking abuot it
>> more, I concluded it was a good thing, for privacy reasons. The proxy is
>> the one performing the caller preferences computation. The proxy will
>> frequently not be the one authenticating the caller, or authorizing them
>> to make calls or learn information abuot the called party. But, if a
>> proxy reported information on call failures because of preference
>> mismatches, a client would be able to "probe" the network using caller
>> preferences, and learn detailed ifnormation on the capabilities of the
>> devices associated with a user, without authentication. The current
>> behavior prevents that, and I think thats good.
>>
>> OK, so whats the problem? Well, its those implicit preferences again.
>> Lets say a user sends a SUBSCRIBE request, without caller preferences
>> parameters. The called party has one UA, and it doesnt support
>> SUBSCRIBE. Its registration to its proxy indicated such. The proxy
>> supports caller preferences. So, it wuold compute an implicit preference
>> to route the request to a UA that supports SUBSCRIBE, and give it the 
>> require strength (meaning, contacts which don't support SUBSCRIBE are 
>> discarded). Since none do, the
>> one registered contact is discarded, and the caller receives a 480.
>>
>> A similar thing happens for event packages.
>>
>> This is a change in behavior if caller preferences were not used, in
>> which case either (1) the UAS would authenticate and authorize the
>> caller, and then return a 405 with the allowed methods, (2) the UAS
>> would not authenticate or authorize the UAC, and thus genreate a 403 
>> or 401.
>>
>> As a result, even though the caller did not use caller preferences, they
>> have suffered from a loss of information, just because the proxy used
>> caller prefs. The 405 would be useful for the UAC to perhaps try another
>> method, or report the failure to the user. I dont mind a loss of
>> information with EXPLICIT preferences, since the caller knows what they
>> are getting into. But for implicit, we shouldn't be changing basic SIP
>> behavior as seen by the UAC.
>>
>> There are a small number of solutions:
>>
>> 1. eliminate Require-Contact entirely (or, in -08 terms, eliminate the 
>> require parameter to Accept-Contact). This solves the problem, because 
>> the proxy will simply set the un-matching contacts to have a q-value 
>> of zero, and try them last. In the above example, the one registered 
>> contact remains with q=0, and is tried, resulting in a 405.
>>
>> 2.eliminate implicit preferences that use the Require-Contact semantic.
>> So, if the request fails with a 480, its because the client explicitly 
>> asked for  soemthing they couldnt get.
>>
>> 3. If a proxy uses caller prefs, and the result is the eliminatin of 
>> all contacts, AND if the reason they were eliminated was that there 
>> was an implicit Require-Contact for a specific method or event 
>> package, the proxy generates a 405 or 489 as if it were the UAS.
>>
>> 4. modify the semantics of require-contact so that, in the case where
>> all contacts would be eliminated, instead of eliminating them, it sets
>> the q-value to zero. This avoids the error condition.
>>
>>
>> I didn't see any other approaches. None of these seem perfect:
>>
>> 1. Approach 1 is a non-starter; we have applications which need the 
>> Require semantic.
>>
>> 2. Approach 2 is better than approach 1. However, it will result in 
>> SUBSCRIBE and other methods being sent to a UAS that indicated that it 
>> didn't support those methods. On the list, Juha had complained loudly 
>> about this, and this thread was one of the motivations for adding 
>> require-contact. Now, the client can add the explicit Require-Contact 
>> with this solution, but that adds somewhat of a burden.
>>
>> 3. Appraoch 3 has a potential security issue. The proxy will generate 
>> a 405 or 489 on behalf of the UAS. So, if the UAS would generally not 
>> want to reveal this information (supported methods and event packages) 
>> unless the UAC would be authorized and authenticated, this approach 
>> will "leak" this information. One might argue that if the UAS uses 
>> caller prefs, and tells the proxy which methods and event packages it 
>> supports, it has forfeited its right to explicitly authenticate and 
>> authorize the caller before this information is divulged. If the UAS 
>> has these privacy concerns, it should not register these caller 
>> preferences parameters.
>>
>> 4. Approach 4 has similar problems to approach 2.
>>
>>
>> I am inclined to go with approach 3, adding the appropriate 
>> discussions on the privacy considerations. The specific algorithm I 
>> would propose is this. If the caller prefs selection process results 
>> in zero contact, the proxy re-runs the algorithms without the implicit 
>> preferences. If now, there are contacts remaining, it concludes that 
>> the implicit preferences caused the problem. It then checks to see if 
>> the problem was methods or event packages, and generates a 405 or 489 
>> accordingly.
>>
>> Comments?
>>
>> -Jonathan R.
>> -- 
>> Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
>> Chief Scientist                             First Floor
>> dynamicsoft                                 East Hanover, NJ 07936
>> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>> http://www.jdrosen.net                      PHONE: (973) 952-5000
>> http://www.dynamicsoft.com
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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inline.

Henning Schulzrinne wrote:
> Caller preferences seems to mandate being explicit rather than assuming 
> things, so I agree that this should go. This is partially a lack of 
> expressiveness of SDP (and the media preferences in caller-prefs), which 
> can't distinguish between three subtly different things:
> 
> - capabilities that the device has (things that are possible)
> 
> - things the user would like to do now (which may be smaller or larger 
> than the current capabilities);
> 
> - things that the user might want to do later (e.g., after reaching a 
> higher-bandwidth area or transferring a conference mid-call to a more 
> capable device).
> 
> Also, this is really a symptom of two non-commensurate quantities 
> (preference for multiple media and preference for quickly reaching the 
> destination) being at odds.

Right. This is exactly what I was trying to say.


> Without a script, we can't really express 
> all the subleties here, from "I'm willing to wait for the most desirable 
> device to become available" at one extreme to "all devices being 
> available, I choose X" at the other.

Indeed. WHich is why I think its good to not try and infer what the user 
is tryign to do. That would only get in the way of some kind of explicit 
script or other mechanism developed to achieve the desired processing.

ANyway, given Rohan's and Henning's agreement, I think we are closed on 
this particular point.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: RE: RE: [Sip] SIP and Congestion Safety
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Hi Dean,

Are you suggesting,in your alternative proposal, that proxies that send
back a 5xx response will not indicate any MTU in it?
If so, then we should also place a restriction that MTU should not be
used as a basis for such an action. Otherwise the originator of the
request will have no clue what is it supposed to do, retry the request
or reduce the size and if so to what value?
However with UDP , MTU does seem a valid basis for preventing
congestion and it may not be possible to place such a restriction.

This is assuming that congestion  due to UDP messages could be
trigerred both due to fragmentation as well as due to a high message
rate.

How about the following?

UA1-------->P1(MTU1)--->P2.......Pn------->UA2(MTU2).

Let checks on MTU be enforced only at the entry and exit points. P1
will reject requests exceeding MTU1, and indicate MTU1 in the 5xx
sent back to UA1. Let the intermediate proxies pass requests without
checking on the MTU of their next hop. At the edge proxy, Pn, let
there be another check based on the max MTU size that UA2 is willing
to accept(such as using the mechanism specified in  Hisham's draft).
That way the second rejection could occur at the edge proxy.
This limits the number of attempts to at most 2, and prevents large
messages from entering the network in the first place.

As an addition, if we want  to allow intermediate proxies to limit
message sizes to the MTUs for the intermediate paths, we might consider
having a mechanism to establish route size for a dialog. Let proxies
not reject messages exceeding the next hop MTU. Agreed that
this does not take care of route dynamicity nor does it take care of
dynamic MTU changes, but attempting to address these issues would
amount to trying to redefine a PMTU discovery mechanism at the
application level.

thanks
Harsh Bhondwe
Hughes Software Systems




Dean Willis <dean.willis@softarmor.com> on 01/24/2003 10:15:14 PM

To:   Harsh Bhondwe/HSSBLR
cc:   sip@ietf.org

Subject:  RE: RE: [Sip] SIP and Congestion Safety




On Fri, 2003-01-24 at 08:04, hbhondwe@hss.hns.com wrote:
> Both the drafts(draft-khartabil-sip-congestionsafe-ci-01.txt
> and draft-ietf-sip-congestsafe-00) propose using a max-size
> header to indicate to the previous hop SIP entity the max
> size that it(the SIP entity sending the 403, 503 response)
> would allow on the next hop. Both the drats also assume that
> the proxies know the MTU of the next hop by some means.
>
> Its not very difficult to encounter the following problem
> in this mechanism:
>
UA1--(MTU1)--P1--(MTU2)---P2--(MTU3)--P3--(MTU4)--P4--(MTU5)--P5--(MTU6)--U
> A2
>
> MTU1>MTU2>MTU3>MTU4>MTU5>MTU6
>
> In such a case it might take UA1, 6 attempts to finally
> get a request through to UA2. And since the MTU for a
> link is dynamic this could get even worse.

yep. Actually, it's worse than that, since as you point out it is
possible for dynamic rerouting to occur.  That makes the absolute worst
case even more entertaining. Assuming that every reduction reduces the
size of the message by 1 byte, then the worst case number of retries is
given by the equation Size(Max) - Size(Min), where Max is the largest
message ever sent, and Min is the smallest MTU allowed. Assuming common
values of 1300 and 576, that gives us more than 700 retries before a
message gets through. But at least it's deterministic.


> Considering such a scenario, I was wondering if it might be
> a good idea to adopt a mechanism similar to the one used
> for the session timer, i.e allow proxies in the path to
> insert or update a min MTU similar to the way the session
> timer value is inserted/updated. Proxies need not reject
> requests that exceed the MTU of the next hop(hence
> avoiding the problem enumurated above). And like with
> session timer, the UA that initiated the request would
> get to know the min MTU for the complete path in the
> response, instead of having to converge on a value
> through trial and error.


If I understand this, you're proposing that we complete one round-trip
in order to establish a route size for future use within the dialog, and
that we assume that the route does not vary over the course of the
dialog.

The main problem here -- what if the first request is 64kB long and
blows up on the UDP network just past the first proxy?

And of course, there's the problem of route dynamicity.

Here's an alternative proposal. Rather than negotiating size, could we
"just say no"?

Any proxy that finds it cannot congestion-safely transmit a message
simply kicks back a 5xx "cannot send safely" response, and it's up to
the sender to find another route?

--
Dean

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Subject: RE: [Sip] Extension to Assure Congestion Safety
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I'll bite :-)

How long will it take to setup a call in a far-from-the-antenna 3G scenario using TCP?  That is, at 14.4kb/s, I'm looking at least 3RTT's just to say "INVITE".

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, January 16, 2003 11:26 AM
> To: loretosa@vizzavi.it
> Cc: bcampbell@dynamicsoft.com; sip@ietf.org
> Subject: Re: [Sip] Extension to Assure Congestion Safety
> 
> 
> On Thu, 2003-01-16 at 06:36, loretosa@vizzavi.it wrote:
> > Hi
> > 
> > are there news about the Congestion problem ?
> > I read some weeks ago that work is frozen. 
> > But your draft expires at the end of January. 
> > Will be some change in new release? If it will be resubmitted.
> > 
> > About the change with TCP as Transport Protocol default,
> > someone can tell me some news?
> > 
> 
> Thanks, Sal. Actually, we haven't talked much about this 
> lately, and we
> probably should.
> 
> When conversation last stopped, we had just realized that although we
> could use this sort of mechanism in the forward direction, it 
> doesn't do
> anything about the reverse direction.
> 
> So, let's say somebody S/MIME signs a 200 OK, making it perhaps 4KB in
> size (That doesn't necessarily happen, but it MIGHT if they include
> their cert). Since this is a response, it MUST follow the 
> path laid out
> in the Via header field, even if this takes it across non-congestion
> safe UDP links.
> 
> Is this a problem that we should worry about?
> 
> If so, can we address it short of jut deprecating UDP? 
> 
> One approach that came up is essentially tagging the Via 
> values so that
> we know which are congestion safe. Under this approach, the 
> receiving UA
> would be able to recognize that the large response it is generating
> could not be returned safely. What should it do then? Could such a
> mechanism be made backward compatible?
> 
> How much additional triggerwork are we willing to put into 
> the protocol
> just to continue accommodating UDP? I know a number of people believe
> that a lot of the complexity of SIP stems from its concurrent support
> for UDP and TCP, and that we'd be better off just ripping the UDP out
> and consequently greatly simplifying the state machines than 
> adding more
> complexity to the core protocol.
> 
> So, we don't really have a group consensus about where to go 
> with this,
> and we need one. 
> 
> Ladies, gentleman, and those to whom these terms are irrelevant, the
> floor is open for discussion.
> 
> --
> Dean
> 
> 
> _______________________________________________
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> 
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Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Thu, 30 Jan 2003 10:44:40 -0500
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If this goes forward, I would also suggest that wording to the effect of "INFO packages MUST describe congestion control mechanisms, or why congestion control mechanisms will not be an issue (e.g., we're talking about only ever sending a single, small message)".

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, January 13, 2003 9:22 PM
> To: sip@ietf.org
> Subject: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> 
> It only seemed appropriate to follow up my previous assumably 
> eloquent but
> frequently misspelled exhortations by following my own advice 
> and "sending
> text".
> 
> The resulting draft has been submitted to the archive and should be
> announced in due course. In the interim, you may wish to 
> retrieve it from:
> 
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopac
> kage-00.html
> 
> or
> 
> http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopac
kage-00.txt

Or, if you just LIKE looking at XML source or wish to fix some of my
probable glaring errors,

http://www.softarmor.com/sipwg/drafts/draft-willis-sip-infopackage-00.xml


I'd like to caveat that this is a rough first draft -- I just sat down and
banged it out this afternoon, and only Adam Roach has seen any of it so far,
and that was in small bits.

thanks

--
Dean

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Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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[Lot's of discussion; I'm just catching up.]

The whole issue of 'what this message means' versus 'what this object is' was dealt with to great extent in the VPIM work group.

Adam gave the excellent example of "what does image/jpeg mean?".  We grappled with "what does audio/basic mean?".  The answer is you absolutely cannot determine what the message context (meaning) is from the types of bodies in the message.

Moreover, registering lots of application/mumbles doesn't work either.  There is little value to registering a bunch of things that have the same semantics.  That also has interaction with things like multipart/alternative or multipart/mixed.

You can find a *very* detailed explanation of this problem, and the solution, in RFC 3458.

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Wednesday, January 15, 2003 7:02 PM
> To: Henning Schulzrinne; Dean Willis
> Cc: 'Christer Holmberg'; sip@ietf.org
> Subject: Re: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00
> 
> 
> "Henning Schulzrinne" <hgs@cs.columbia.edu> writes:
> 
> > We currently negotiate above-SIP content with Content-Type, 
> > Content-Language and all the other Content-Headers, just 
> like HTTP, SMTP 
> > and other MIME users. If the body contains message/isup, it 
> is an ISUP 
> > message. It is not clear to me what value an additional, 
> semi-orthogonal 
> > negotiation mechanism offers.
> 
> Upon further reflection, I realise that my previous response addressed
> the wrong half of this objection (multiple content types per 
> application,
> instead of multiple applications per content type).
> 
> Take, for example, the content type of "image/jpeg".
> 
> It is quite conceivable that someone might define an INFO
> package of "app-icon" that uses "image/jpeg" to send
> a 32x32 pixel image that should be used as the icon for the
> application for the duration of the session.
> 
> It is also completely beleivable that someone else might
> define an INFO package of "set-pointer" that uses "image/jpeg"
> to send a 16x16 bitmap that is to be used as the mouse pointer
> when said pointer is in the window associated with the
> application.
> 
> Now, I'm not advocating these usages of INFO or even the
> ability to do either of these things with SIP, but I'm certain
> that you will agree that these are the *sort* of content-type
> collisions that you will encounter if you don't have the
> orthagonal mechanism that Dean includes in his draft.
> 
> /a
> 
> _______________________________________________
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Subject: RE: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
Date: Thu, 30 Jan 2003 10:44:40 -0500
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Thread-Topic: [Sip] INFO Packaging --  draft-willis-sip-infopackage-00
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From: "Eric Burger" <eburger@snowshore.com>
To: "Tom-PT Taylor" <taylor@nortelnetworks.com>
Cc: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
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Don't worry: your existing SIP-T implementations will still work.  They won't ask for or accept the info-package offer, which means that raw INFO should still work.


-----Original Message-----
From: Tom-PT Taylor [mailto:taylor@nortelnetworks.com]
Sent: Wednesday, January 15, 2003 10:36 AM
To: 'Henning Schulzrinne'; Drage, Keith (Keith)
Cc: 'Dean Willis'; sip@ietf.org
Subject: RE: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00


I really agree with Henning's viewpoint.  The mechanisms are already there.  Quite frankly, I don't want to have to redo my SIP-T implementation of INFO unless a really good case can be made for it.
> -----Original Message----- 
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu] 
> Sent: Wednesday, January 15, 2003 10:13 AM 
> To: Drage, Keith (Keith) 
> Cc: 'Dean Willis'; sip@ietf.org 
> Subject: Re: [Sip] INFO Packaging -- draft-willis-sip-infopackage-00 
> 
> 
> 
> >>Ah. Actually, that is the direction that I initially started 
> >>taking things 
> >>in the draft -- basically, I associated MIME content-types 
> >>with usages, and 
> >>didn't have the Info-Type. Then I briefly talked it over with 
> >>Adam, and he 
> >>talked me into the separate semantic indicator. 
> >> 
> >>I believe that the assertion that "most INFO types will have 
> >>exactly one 
> >>valid Content-Type" is the breaker. Let's look at text. Is 
> >>the text I just 
> >>sent you indicative of a move in the chess game we are 
> playing? Is it 
> >>something I want you to post on a web page for me? Should you 
> >>just read it? 
> >>Is it a javascript that you should use to remap your 
> >>keyboard? Is it C code 
> >>that you should compile and execute? Is it a credit card 
> 
> The mistake would be specifying all of them as text/plain. 
> They should 
> be specified as the correct media type to begin with. After 
> all, I may 
> want to save the content and saving a C file as .txt instead of .c is 
> not very helpful. 
> 
> If this is application data, it should be labeled as 
> application/chess. 
> 
> In addition, there is Content-Disposition to look at. 
> 
> 
> >>number that you 
> >>should debit for the price of this call? Why the heck AM I 
> >>sending it to 
> >>you? 
> >> 
> >>There are probably much better examples, but I think that a 
> one-to-one 
> >>mapping between content-type and expected function is more 
> >>constraining than the way I think people are trying to use INFO, 
> >>perhaps overly constraining. 
> >> 
> >>-- 
> >>dean 
> >> 
> >> 
> >>_______________________________________________ 
> >>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip 
> >>This list is for NEW development of the core SIP Protocol 
> >>Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> >>sipping@ietf.org for new developments on the application of sip 
> >> 
> > 
> > _______________________________________________ 
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip 
> > This list is for NEW development of the core SIP Protocol 
> > Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > sipping@ietf.org for new developments on the application of sip 
> 
> _______________________________________________ 
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip 
> This list is for NEW development of the core SIP Protocol 
> Use sip-implementors@cs.columbia.edu for questions on current 
> sip Use sipping@ietf.org for new developments on the 
> application of sip 
> 
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Eric Burger'" <eburger@snowshore.com>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Extension to Assure Congestion Safety
Date: Thu, 30 Jan 2003 12:50:19 -0600
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Eric chewed on:
> I'll bite :-)
> 
> How long will it take to setup a call in a 
> far-from-the-antenna 3G scenario using TCP?  That is, at 
> 14.4kb/s, I'm looking at least 3RTT's just to say "INVITE".

Well, consider that you should be using a long-lived TCP session established
back when you turned your phone on. Consequently, there's little further
overhead for TCP -- just some ack fields in the TCP headers on the INVITE,
200OK, and ACK. There's probably a final null-ack TCP packet on the ACK
message, but that's after call setup has completed.

Given this, the constraint is still the three-way handshake of SIP, which
means you have at least 3 RTTs whether you use UDP or TCP. 

So, let's say each request, using compression, is 400 bytes (that's 3200
bits). At 14.4, that makes a total serialization delay of 222 ms per each
mobile link. Two mobile links give us 444ms. Add 150ms of core network
delays (50ms each hop), and we're at 372ms. Not too shabby so far.

Now, add aother 50 ms each for the P-CSCF, I-CSCF, S-CSCF, I-CSCF, S-CSCF,
P-CSCF processing and we get another 300 ms for each hop, or 900ms. Add a
little more for topology hiding gateways and the like, and we're at around
1800 ms.

So much for the math. The problem is, current 3G systems have a lot more
transmission latency than one would think based on the bandwidth. Assuming
there's an active connection, it still tends to be a couple of hundred ms
before a mobile can actually send anything. Consequently, real world
performance with today's system adds another 2 seconds or so to the call
setup. We're working on that from the network side. On the other hand,
that's still not any worse than the 10-15 second call setup dealys I always
have on my TDMA phone . . .

Now, lets think about the specific impact of TCP vs UDP here. The BIG delays
in SIP setup happen if a request or response gets lost and we use the SIP
backoff timers to resend. TCP, giving us a smoother-running network, is less
likely to give us a network in which packets get lost. Furthermore, its
recovery timers (especially in the wireless-tuned TCP) are likely to give
smoother, quicker recovery than does SIP in the event of a loss-burst. So,
all in all, I don't think we lose anything by using TCP to the mobile, and
we potentially gain quite a bit. Server-to-server operations have to think
about head-of-line blocking issues, and that's wehre SCTP comes in as a
better alternative.

--
Dean

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From: "Dean Willis" <dean.willis@softarmor.com>
To: <hbhondwe@hss.hns.com>
Cc: <sip@ietf.org>
Subject: RE: RE: [Sip] SIP and Congestion Safety
Date: Thu, 30 Jan 2003 12:50:19 -0600
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> Hi Dean,
> 
> Are you suggesting,in your alternative proposal, that proxies
> that send back a 5xx response will not indicate any MTU in 
> it? If so, then we should also place a restriction that MTU 
> should not be used as a basis for such an action. Otherwise 
> the originator of the request will have no clue what is it 
> supposed to do, retry the request or reduce the size and if 
> so to what value? However with UDP , MTU does seem a valid 
> basis for preventing congestion and it may not be possible to 
> place such a restriction.

Right. MTU is not a basis in response processing, because responses MUST
follow the path of the request. The comparison factor I'm suggesting is "if
the response is bigger than the request AND the request was not marked
congestion-safe, send a 5XX "cannot respond safely". 

 
> This is assuming that congestion  due to UDP messages could
> be trigerred both due to fragmentation as well as due to a 
> high message rate.
> 
> How about the following?
> 
> UA1-------->P1(MTU1)--->P2.......Pn------->UA2(MTU2).
> 
> Let checks on MTU be enforced only at the entry and exit
> points. P1 will reject requests exceeding MTU1, and indicate 
> MTU1 in the 5xx sent back to UA1. Let the intermediate 
> proxies pass requests without checking on the MTU of their 
> next hop. At the edge proxy, Pn, let there be another check 
> based on the max MTU size that UA2 is willing to accept(such 
> as using the mechanism specified in  Hisham's draft). That 
> way the second rejection could occur at the edge proxy. This 
> limits the number of attempts to at most 2, and prevents 
> large messages from entering the network in the first place.
> 
> As an addition, if we want  to allow intermediate proxies to
> limit message sizes to the MTUs for the intermediate paths, 
> we might consider having a mechanism to establish route size 
> for a dialog. Let proxies not reject messages exceeding the 
> next hop MTU. Agreed that this does not take care of route 
> dynamicity nor does it take care of dynamic MTU changes, but 
> attempting to address these issues would amount to trying to 
> redefine a PMTU discovery mechanism at the application level.
> 

Your suggestions seem applicable to REQUEST processing, although I don't see
a real gain over the somewhat simpler approach in
draft-ietf-sip-congestsafe-00. The big piece missing from that draft was the
RESPONSE processing. I also continue to worry about path dynamicity. Part of
the difference may be a matter of intent. Personally, I really don't want to
reinvent TCP flow control on top of UDP for SIP. The TCP folks have been
working on that problem for a long time already. I'm more interested in a
way to keep UDP implementations, where we think we MUST have them, from
breaking things.

So, open question -- exactly what is it we're trying to solve here, and how
much mechanism is it reasonable to have?

--
Dean

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Thread-Topic: RequestUri to Host Matching and proxy servers
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: <sip@ietf.org>
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Updating the sip mib ID with Kevin & Kavitha, we have a quick question
re: Request-URI and host matching for proxy servers.  Is there any
application where not requiring request-uri host matching makes sense
for a proxy?
We thought about the outbound proxy case where requests are usually
targetted for external proxies, therefore request-uri to host matching
may potentially not be the default action.

We have a sipRequestUriHostMatching mib object (true/false value) that
is currently applicable to UAs and proxies and we're wondering whether
to keep it as a common configuration object or move it to the UA config
section of the mib.

Additionally, if request-URI to host matching is enforced, is it ok for
the object description to state something like:
"If the value of this object is 'true', then the server requires a
match, and if the RequestURI doesn't match the server's host name, a
Location Service may be used to obtain information about a callee's
possible location(s) or a 404 Not Found status code is returned by the
server."

Thanks,
Jean-Francois.

   sipRequestUriHostMatching OBJECT-TYPE    
       SYNTAX     TruthValue    
       MAX-ACCESS read-write    
       
       STATUS     current    
       DESCRIPTION    
            "This object specifies whether or not the host in    
             RequestURI must match server's host name.    
       
             If the value of this object is 'true', then the 
             server requires a match, and if the RequestURI 
             doesn't match the server's host name, a Location 
             Service may be used to obtain information about 
             a callee's possible location(s) or a 
             404 Not Found status code is returned by the 
             server.
             If the value is 'false', then no match is required."    
       REFERENCE    
            "RFC 3261, Section 8.2.2.1"    
       ::= { sipCommonCfgEntry 8 }    
_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip



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From: "KumarS" <knowsip@fastmail.fm>
To: sip@ietf.org
Date: Thu, 30 Jan 2003 22:03:47 -0800
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Subject: [Sip] Re-Invite supported SIP implementation
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Hi All,

I am looking for Re-Invite supported SIP implementation.
May I get any pointer to Re-Invite (rfc 3261) implemented downloadable
SIP code.

Thanks in advance
Suman
-- 
  KumarS
  knowsip@fastmail.fm

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From: margaretmary <margaret_mary@dexceldesigns.com>
To: <sip@ietf.org>
Date: Fri, 31 Jan 2003 19:10:20 +0530
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Subject: [Sip] query on redirect server
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This is a multi-part message in MIME format.

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Hello  sir,

With respect to draft-ietf-sip-rfc2543bis-09.ps for Redirect server " A =
redirect server does not issue any SIP requests or its own. Ater =
receiving a request other than CANCEL, the server either refuses the =
request or gathers the list of alternative locations from the location =
service and returns a final response of class 3xx".
This statement  tells for all methods of (INVITE, ACK, BYE, REGISTER, =
OPTIONS) other then CANCEL or only for INVITE.

Its quite confusing.
Please try to provide a solution for this.


Regards,
Margaret,


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2920.0" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hello&nbsp; sir,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>With respect to =
draft-ietf-sip-rfc2543bis-09.ps for=20
Redirect server " A redirect server does not issue any SIP requests or =
its own.=20
<STRONG>Ater receiving a request other than CANCEL, the server either =
refuses=20
the request or gathers the list of alternative locations from the =
location=20
service and returns a final response of class =
3xx</STRONG>".</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This statement&nbsp; tells for all =
methods of=20
(INVITE, ACK, BYE, REGISTER, OPTIONS) other then CANCEL or only for=20
INVITE.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Its quite confusing.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Please try to provide a solution for=20
this.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>Regards,<BR>Margaret,<BR></FONT></DIV></BODY></HTML>

<br>--------------------------------------------------------------<br>Dexcel Electronics Designs (P) Ltd., Bangalore, India<br>
------=_NextPart_000_0007_01C2C95C.66051AE0--

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Use sipping@ietf.org for new developments on the application of sip



